From chenct1976 at sina.com Thu Mar 1 02:26:10 2012 From: chenct1976 at sina.com (chenct1976 at sina.com) Date: Thu, 01 Mar 2012 09:26:10 +0800 Subject: [FFmpeg-user] A question: ffplay can not play from the given time Message-ID: <20120301012610.9B36673E301@webmail.sinamail.sina.com.cn> Hello everybody, I played the file Wildlife.wmv(Win7 OS Sample Videos) by using ffplay comand, but it can not play from the given seconds. OS: Win7 32Bit + Wingw ffplay version: V0.10 Command line: ./ffplay_g -i Wildlife.wmv -ss 2 Expected result: play the video file from the 2nd second. Test result: play from 0 second. Is it a ffplay's bug? or the tested file has problem? Thanks in advance! Tom.chen attach the test log: 2012/03/01-09:09: ffplay version 0.10 2012/03/01-09:09: Copyright (c) 2003-2012 the FFmpeg developers 2012/03/01-09:09: built on Jan 29 2012 17:50:46 with gcc 4.5.2 2012/03/01-09:09: configuration: --prefix=/ffmpeg/release10 --enable-debug --disable-static --enable-shared --enable-gpl --enable-version3 --enable-avfilter --enable-memalign-hack --enable-avisynth --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libfaac --enable-nonfree --enable-libspeex --enable-libopenjpeg --enable-libxavs --enable-libvpx --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libschroedinger --enable-libcelt --enable-frei0r --enable-libdirac --disable-decoder=libdirac --extra-cflags=-I/ffmpeg/olibs9/include --extra-ldflags=-L/ffmpeg/olibs9/lib --extra-libs=/mingw/lib/libdl.a 2012/03/01-09:09: libavutil 51. 34.101 / 51. 34.101 2012/03/01-09:09: libavcodec 53. 60.100 / 53. 60.100 2012/03/01-09:09: libavformat 53. 31.100 / 53. 31.100 2012/03/01-09:09: libavdevice 53. 4.100 / 53. 4.100 2012/03/01-09:09: libavfilter 2. 60.100 / 2. 60.100 2012/03/01-09:09: libswscale 2. 1.100 / 2. 1.100 2012/03/01-09:09: libswresample 0. 6.100 / 0. 6.100 2012/03/01-09:09: libpostproc 52. 0.100 / 52. 0.100 2012/03/01-09:09: Input #0, asf, from 'Wildlife.wmv': 2012/03/01-09:09: Metadata: 2012/03/01-09:09: SfOriginalFPS : 299 2012/03/01-09:09: WMFSDKVersion : 11.0.6001.7000 2012/03/01-09:09: WMFSDKNeeded : 0.0.0.0000 2012/03/01-09:09: IsVBR : 0 2012/03/01-09:09: title : Wildlife in HD 2012/03/01-09:09: copyright : ? 2008 Microsoft Corporation 2012/03/01-09:09: comment : Footage: Small World Productions, Inc; Tourism New Zealand | Producer: Gary F. Spradling | Music: Steve Ball 2012/03/01-09:09: Duration: 2012/03/01-09:09: 00:00:30.09 2012/03/01-09:09: start: 2012/03/01-09:09: 0.000000 2012/03/01-09:09: bitrate: 2012/03/01-09:09: 6977 kb/s 2012/03/01-09:09: Stream #0:0 2012/03/01-09:09: (eng) 2012/03/01-09:09: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, 2 channels, s16, 192 kb/s 2012/03/01-09:09: Stream #0:1 2012/03/01-09:09: (eng) 2012/03/01-09:09: Video: vc1 (Advanced) (WVC1 / 0x31435657), yuv420p, 1280x720, 5942 kb/s 2012/03/01-09:09: 29.97 tbr 2012/03/01-09:09: 1k tbn 2012/03/01-09:09: 1k tbc From adam at the-adam.com Thu Mar 1 02:41:52 2012 From: adam at the-adam.com (Adam N. Rosenberg) Date: Wed, 29 Feb 2012 18:41:52 -0700 (MST) Subject: [FFmpeg-user] Vimeo uploading - may be off topic Message-ID: Hi Vimeo fans, This may be off topic, but maybe somebody can take pity on me. I have 284 aviation videos that I would like to upload to Vimeo. (I used FFMPEG to add music soundtracks with help from this group.) I could upload the AVI files one at a time and type my video titles and descriptions, one at a time, or I could write a script on Linux. I know how to write C code and stuff like that, I can cobble together HTML and blunder my way through really-basic PHP code, but the kind of interactive reactions that run Vimeo applications are beyond my current knowledge. Their documentation presumes one already has a working knowledge of how to use Vimeo applications. I haven't a clue, so I really need baby-steps instructions. I downloaded their PHP file for uploading and I ran Vimeo's web page that gave me a Consumer Key and a Consumer Secret, but I don't know what I'm supposed to do next. If I can get one PHP script to upload one video with title and description, then I can write 284 scripts with a C program from my file list and submit them. Any help would be appreciated. Adam N. Rosenberg mailto:adam at the-adam.com http://www.the-adam.com From dheianevans at gmail.com Thu Mar 1 08:05:08 2012 From: dheianevans at gmail.com (Ian Evans) Date: Thu, 1 Mar 2012 02:05:08 -0500 Subject: [FFmpeg-user] converting 1080 to 720 mpeg2 In-Reply-To: References: <4F4C98CD.1090102@googlemail.com> <4F4C9C17.8020008@googlemail.com> <4F4C9F47.80604@googlemail.com> Message-ID: On Wed, Feb 29, 2012 at 5:18 PM, Andy Sheen wrote: >> This is old, please try current git head. > Teehee... I've been waiting for that. > > And change all the syntax... There should be no issues with map x:y - > it's been there for yonks and always worked when I've needed it - > except when I needed to demux 0:24 and it only mapped 0:20. The only > problem was the new ffmpeg introduced a 100ms delay in the audio but > you can work around that too ;). > > Try a -map 0:0 infront of the video decode (after the -f filetype...) first. Okay...updated to the latest from git tonight. Ran this command line: ffmpeg -i 1662_20120226204000.mpg -t 00:5:00.000 -f mpegts -map 0:0 -vcodec mpeg2video -qscale 2 -vf 'yadif=1,scale=1280:720' -r 60000/1001 -map 0:1 -acodec copy -map 0:2 -acodec copy video-01.mpeg.ts Here's the info of the test file it generated: ffmpeg version N-38403-gd2101bf Copyright (c) 2000-2012 the FFmpeg developers built on Feb 29 2012 23:37:15 with gcc 4.4.5 configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-version3 --enable-x11grab libavutil 51. 41.100 / 51. 41.100 libavcodec 54. 4.100 / 54. 4.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 62.101 / 2. 62.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, mpegts, from 'video-01.mpeg.ts': Duration: 00:04:59.98, start: 1.400000, bitrate: 24036 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 104857 kb/s, 59.96 fps, 59.94 tbr, 90k tbn, 119.88 tbc Stream #0:1[0x101]: Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, 5.1(side), s16, 384 kb/s Stream #0:2[0x102]: Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, stereo, s16, 128 kb/s So am I ready to cat it to the 720 file? What step occurs after that? Thanks to everyone for the help. From dave.bevan at bbc.co.uk Thu Mar 1 09:10:13 2012 From: dave.bevan at bbc.co.uk (Dave Bevan) Date: Thu, 1 Mar 2012 08:10:13 -0000 Subject: [FFmpeg-user] Vimeo uploading - may be off topic In-Reply-To: References: Message-ID: > Subject: [FFmpeg-user] Vimeo uploading - may be off topic ... > Any help would be appreciated. > Adam N. Rosenberg I reckon http://vimeo.com/api/docs/upload has everything you need to know. --D. http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From chenct1976 at sina.com Thu Mar 1 09:33:04 2012 From: chenct1976 at sina.com (chenct1976 at sina.com) Date: Thu, 01 Mar 2012 16:33:04 +0800 Subject: [FFmpeg-user] Will you pls give me some advice? // A question: ffplay can not play from the given time Message-ID: <20120301083304.14B92E8C5C1@webmail.sinamail.sina.com.cn> Hello everybody, I played the file Wildlife.wmv(Win7 OS Sample Videos) by using ffplay comand, but it can not play from the given seconds. OS: Win7 32Bit + Wingw ffplay version: V0.10 Command line: ./ffplay_g -i Wildlife.wmv -ss 2 Expected result: play the video file from the 2nd second. Test result: play from 0 second. Is it a ffplay's bug? or the tested file has problem? Thanks in advance! Tom.chen attach the test log: 2012/03/01-09:09: ffplay version 0.10 2012/03/01-09:09: Copyright (c) 2003-2012 the FFmpeg developers 2012/03/01-09:09: built on Jan 29 2012 17:50:46 with gcc 4.5.2 2012/03/01-09:09: configuration: --prefix=/ffmpeg/release10 --enable-debug --disable-static --enable-shared --enable-gpl --enable-version3 --enable-avfilter --enable-memalign-hack --enable-avisynth --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libfaac --enable-nonfree --enable-libspeex --enable-libopenjpeg --enable-libxavs --enable-libvpx --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libschroedinger --enable-libcelt --enable-frei0r --enable-libdirac --disable-decoder=libdirac --extra-cflags=-I/ffmpeg/olibs9/include --extra-ldflags=-L/ffmpeg/olibs9/lib --extra-libs=/mingw/lib/libdl.a 2012/03/01-09:09: libavutil 51. 34.101 / 51. 34.101 2012/03/01-09:09: libavcodec 53. 60.100 / 53. 60.100 2012/03/01-09:09: libavformat 53. 31.100 / 53. 31.100 2012/03/01-09:09: libavdevice 53. 4.100 / 53. 4.100 2012/03/01-09:09: libavfilter 2. 60.100 / 2. 60.100 2012/03/01-09:09: libswscale 2. 1.100 / 2. 1.100 2012/03/01-09:09: libswresample 0. 6.100 / 0. 6.100 2012/03/01-09:09: libpostproc 52. 0.100 / 52. 0.100 2012/03/01-09:09: Input #0, asf, from 'Wildlife.wmv': 2012/03/01-09:09: Metadata: 2012/03/01-09:09: SfOriginalFPS : 299 2012/03/01-09:09: WMFSDKVersion : 11.0.6001.7000 2012/03/01-09:09: WMFSDKNeeded : 0.0.0.0000 2012/03/01-09:09: IsVBR : 0 2012/03/01-09:09: title : Wildlife in HD 2012/03/01-09:09: copyright : ? 2008 Microsoft Corporation 2012/03/01-09:09: comment : Footage: Small World Productions, Inc; Tourism New Zealand | Producer: Gary F. Spradling | Music: Steve Ball 2012/03/01-09:09: Duration: 2012/03/01-09:09: 00:00:30.09 2012/03/01-09:09: start: 2012/03/01-09:09: 0.000000 2012/03/01-09:09: bitrate: 2012/03/01-09:09: 6977 kb/s 2012/03/01-09:09: Stream #0:0 2012/03/01-09:09: (eng) 2012/03/01-09:09: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, 2 channels, s16, 192 kb/s 2012/03/01-09:09: Stream #0:1 2012/03/01-09:09: (eng) 2012/03/01-09:09: Video: vc1 (Advanced) (WVC1 / 0x31435657), yuv420p, 1280x720, 5942 kb/s 2012/03/01-09:09: 29.97 tbr 2012/03/01-09:09: 1k tbn 2012/03/01-09:09: 1k tbc From pavel at sokolov.me Thu Mar 1 10:12:51 2012 From: pavel at sokolov.me (Pavel Sokolov) Date: Thu, 01 Mar 2012 13:12:51 +0400 Subject: [FFmpeg-user] AVI/AC3 -> VOB/LPCM In-Reply-To: References: <4F4E5D37.5000802@sokolov.me> Message-ID: <4F4F3D93.6020202@sokolov.me> 29.02.2012 23:10, Andrey Utkin ?????: > 2012/2/29 Pavel Sokolov: >> Hi all! >> >> How can I do remux of the AVI with AC3 stereo to the VOB/LPCM >> >> When I tried the next cmd I hear bad audio (sh-sh-sh-sh): >> ffmpeg -i video-mpeg4_720x544-audio_ac3_48000_stereo.avi -vcodec copy >> -acodec pcm_s16be -sample_fmt s16 -f vob test1.vob >> >> What I'm doing wrong? > Full uncut console output is missing... > Also publish your source file, can help. > Sound is bad on PC also, or just on your specific hardware? > file: http://sokolov.me/tmp/video-mpeg4_720x544-audio_ac3_48000_stereo.avi ffmpeg log: [vob @ 0xdf5be0] buffer underflow i=1 bufi=4026 size=6144 Last message repeated 34 times [vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it [vob @ 0xdf5be0] buffer underflow i=1 bufi=4026 size=6144 [vob @ 0xdf5be0] buffer underflow i=1 bufi=6043 size=6144 [vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it [vob @ 0xdf5be0] buffer underflow i=1 bufi=6043 size=6144 [vob @ 0xdf5be0] buffer underflow i=1 bufi=3928 size=6144 Last message repeated 3 times [vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it [vob @ 0xdf5be0] buffer underflow i=1 bufi=3928 size=6144 [vob @ 0xdf5be0] buffer underflow i=1 bufi=5945 size=6144 [vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it [vob @ 0xdf5be0] buffer underflow i=1 bufi=5945 size=6144 [vob @ 0xdf5be0] buffer underflow i=1 bufi=3830 size=6144 [vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it [vob @ 0xdf5be0] buffer underflow i=1 bufi=3830 size=6144 [vob @ 0xdf5be0] buffer underflow i=1 bufi=5847 size=6144 [vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it [vob @ 0xdf5be0] buffer underflow i=1 bufi=5847 size=6144 [vob @ 0xdf5be0] buffer underflow i=1 bufi=3732 size=6144 [vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it [vob @ 0xdf5be0] buffer underflow i=1 bufi=3732 size=6144 [vob @ 0xdf5be0] buffer underflow i=1 bufi=5749 size=6144 [vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it From andrey.krieger.utkin at gmail.com Thu Mar 1 10:54:58 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Thu, 1 Mar 2012 11:54:58 +0200 Subject: [FFmpeg-user] Will you pls give me some advice? // A question: ffplay can not play from the given time In-Reply-To: <20120301083304.14B92E8C5C1@webmail.sinamail.sina.com.cn> References: <20120301083304.14B92E8C5C1@webmail.sinamail.sina.com.cn> Message-ID: 2012/3/1 : > Hello everybody, > I played the file Wildlife.wmv(Win7 OS Sample Videos) by using ffplay comand, but it can not play from the given seconds. > OS: Win7 32Bit + Wingw > ffplay version: V0.10 > Command line: ./ffplay_g -i Wildlife.wmv -ss 2 > Expected result: play the video file from the 2nd second. > Test result: play from 0 second. As far as i know, you can seek (jump in time) only to keyframe location. Keyframes can be located with interval of several seconds. Thus you request to start from 2 seconds offset, but ffmpeg plays from closest keyframe that is at 0 seconds offset. I think for now you can pass your file through ffmpeg, to cut off unneeded beginning of file. Or write application using ffmpeg API that will skip video not by seeking, but by decoding every frame and not displaying unneeded ones. -- Andrey Utkin From ssmanian at yahoo.com Thu Mar 1 03:26:46 2012 From: ssmanian at yahoo.com (Senthil) Date: Wed, 29 Feb 2012 18:26:46 -0800 (PST) Subject: [FFmpeg-user] Unable to play VOB file with audio. Message-ID: <1330568806375-4433823.post@n4.nabble.com> I have a vob file that plays just fine but no audio as my DLNA TV does not support the DTS format Input #0, mpeg, from 'test.VOB': Duration: 00:18:50.66, start: 1148.989333, bitrate: 7595 kb/s Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 9000 kb/s, 27.21 fps, 25 tbr, 90k tbn, 50 tbc Stream #0:1[0x89]: Audio: dts (DTS), 48000 Hz, 5.1(side), s16, 768 kb/s Stream #0:2[0x80]: Audio: ac3, 48000 Hz, 5.1(side), s16, 448 kb/s At least one output file must be specified As you can see, the first stream is DTS which is not supported but the second stream is supported. So what I want to do is create another VOB file and use the video as is and the second stream from the audio. So I did, /volume1/@appstore/Serviio/bin/ffmpeg -i test.VOB -i test.VOB -map 0:0 -map 0:2 -vcode c copy -acodec copy -y test-copyav.VOB The resulting file now does not play video but the audio is fine. (Reverse of the original issue). The output of the new file is: Seems stream 1 codec frame rate differs from container frame rate: 50.00 (50/1) -> 50.00 (50/1) Input #0, mpeg, from 'test-copyav.VOB': Duration: 00:18:51.14, start: 1.042667, bitrate: 6780 kb/s Stream #0:0[0x80]: Audio: ac3, 48000 Hz, 5.1(side), s16, 448 kb/s Stream #0:1[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 9000 kb/s, 27.14 fps, *50 tbr,* 90k tbn, 50 tbc At least one output file must be specified The only difference i see is the TBR is 50 in the new file and 25 on the original. Is that the reason why it is not playing video? If so, is there an option to force the tbr value. Am I doing something wrong in the above step? Thanks for any help. Senthil -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Unable-to-play-VOB-file-with-audio-tp4433823p4433823.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From wodfer at gmail.com Thu Mar 1 12:51:58 2012 From: wodfer at gmail.com (Andy Wodfer) Date: Thu, 1 Mar 2012 12:51:58 +0100 Subject: [FFmpeg-user] Problems encoding to DV25 .avi file Message-ID: Hi, I need some input on a problem I have. I'm using the latest 0.10 version of ffmpeg compiled from source on a FreeBSD 8.2 machine. The source file is a Prores 422HQ file (1280x720 @ 50fps 720P). I want to make this into a DV25 .avi file that is playable in Windows Media Player and that will work in Ie. Adobe Premiere Pro 2.0 on a WinXP machine. Ie: A Windows compatible DV25 file. I've tried several options, but I can't make it work. "-target pal-dv" gives me a file that plays fine in VLC player, but not in Windows Media Player. Adobe Premiere reports unsupported format and won't import it. "-vcodec dvvideo -s 720x576 -pix_fmt yuv420p -r 25 -acodec pcm_s16le -ac 2 -ar 48000" works if I choose .mov as a container. It opens in Quicktime and VLC and import to Adobe Premiere works fine. Aspect ratio is also fine. If I change the container to .avi it won't play back in Windows Media Player. Any ideas of what I can try next? Thanks!! Best regards, Andy From de.techno at gmail.com Thu Mar 1 13:50:03 2012 From: de.techno at gmail.com (dE .) Date: Thu, 01 Mar 2012 18:20:03 +0530 Subject: [FFmpeg-user] Need help with ffmpeg In-Reply-To: References: Message-ID: <4F4F707B.7040206@gmail.com> On 02/29/12 21:01, Carl Eugen Hoyos wrote: > Aashish vivekanand gmail.com> writes: > >> link: http://www.mediafire.com/?3ri05yd2wej5wrm > Which program plays this file? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user File command says that it's data.... From sheen.andy at googlemail.com Thu Mar 1 14:40:58 2012 From: sheen.andy at googlemail.com (Andy Sheen) Date: Thu, 1 Mar 2012 13:40:58 +0000 Subject: [FFmpeg-user] converting 1080 to 720 mpeg2 In-Reply-To: References: <4F4C98CD.1090102@googlemail.com> <4F4C9C17.8020008@googlemail.com> <4F4C9F47.80604@googlemail.com> Message-ID: On 1 March 2012 07:05, Ian Evans wrote: > Okay...updated to the latest from git tonight. Ran this command line: > ... > > So am I ready to cat it to the 720 file? What step occurs after that? > > Have you tried it? If it works - you should just be able to play it. I'm wondering about how the transition will be handled. The TS file format will use PIDs and these will change when you change streams. It may be better to remux (after concatenation) to something like a .mkv file. From bostjan.strojan at gmail.com Thu Mar 1 15:26:58 2012 From: bostjan.strojan at gmail.com (=?UTF-8?Q?Bo=C5=A1tjan_Strojan?=) Date: Thu, 1 Mar 2012 15:26:58 +0100 Subject: [FFmpeg-user] Scene detection In-Reply-To: <1330531015081-4431928.post@n4.nabble.com> References: <354784.24180.qm@web110607.mail.gq1.yahoo.com> <1312726584077-3724931.post@n4.nabble.com> <39659E17-5BE5-4C7C-8E1E-0E79DB34CCE5@avpreserve.com> <20110808161254.88776wg7sdrgb946@webmail.tuwien.ac.at> <20110808222557.GC9924@geppetto> <1330531015081-4431928.post@n4.nabble.com> Message-ID: On Wed, Feb 29, 2012 at 4:56 PM, atigian wrote: > When I don't have an EDL, my approach is to do the scene detection during > the encoding using keyint, min-keyint and scenecut, thus inserting a key > frame in every scene change. Once I've encoded the file I just extract all > key frames: > ffmpeg -vf select="eq(pict_type\,PICT_TYPE_I)" -i myvideo.mp4 -vsync 2 -s > 73x41 -f image2 thumbnails-%02d.jpeg > > I also create a text file with the timecodes of the key frames by adding: > -loglevel debug 2>&1 | grep "pict_type:I -> select:1" | cut -d " " -f 6 - > > keyframe-timecodes.txt > > And I use this file to sync the thumbnails of the scene changes with a video > player: > http://www.videoproductionslondon.com/blog/scene-change-detection-during-encoding-key-frame-extraction-code > http://www.videoproductionslondon.com/blog/scene-change-detection-during-encoding-key-frame-extraction-code Thanks, much appreciated. b. From adam at the-adam.com Thu Mar 1 15:41:04 2012 From: adam at the-adam.com (Adam N. Rosenberg) Date: Thu, 1 Mar 2012 07:41:04 -0700 (MST) Subject: [FFmpeg-user] Vimeo uploading - may be off topic In-Reply-To: References: Message-ID: On Thu, 1 Mar 2012, Dave Bevan wrote: >> Subject: [FFmpeg-user] Vimeo uploading - may be off topic > ... >> Any help would be appreciated. >> Adam N. Rosenberg > > I reckon http://vimeo.com/api/docs/upload has everything you need to > know. > > --D. Hi Dave, I reckon it does have everything, except the basic stuff of how to setup and to run the programs. They figure you already know that. I'm looking for real baby steps for a first-time Vimeo programmer. 1. Go the http://vimeo.com/setup-auth 2. Type your username, password, and description of the application 3. Press the GET KEY button. 4. Write down code and secret code. 5. Download PHP file from http://vimeo.com/get-upload-code 6. Add the following lines to the bottom ... Once I can do that for one video, then, yes, I can figure out how to do the other 283 of them. Adam N. Rosenberg mailto:adam at the-adam.com http://www.the-adam.com From de.techno at gmail.com Thu Mar 1 16:12:51 2012 From: de.techno at gmail.com (dE .) Date: Thu, 01 Mar 2012 20:42:51 +0530 Subject: [FFmpeg-user] Unable to play VOB file with audio. In-Reply-To: <1330568806375-4433823.post@n4.nabble.com> References: <1330568806375-4433823.post@n4.nabble.com> Message-ID: <4F4F91F3.4020507@gmail.com> On 03/01/12 07:56, Senthil wrote: > Stream #0:0[0x80]: Audio: ac3, 48000 Hz, 5.1(side), s16, 448 kb/s > Stream #0:1[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [SAR Usually the first stream (0.0) is Video. Try moving it to the first stream and make audio the second. From tevans.uk at googlemail.com Thu Mar 1 16:25:51 2012 From: tevans.uk at googlemail.com (Tom Evans) Date: Thu, 1 Mar 2012 15:25:51 +0000 Subject: [FFmpeg-user] Vimeo uploading - may be off topic In-Reply-To: References: Message-ID: On Thu, Mar 1, 2012 at 2:41 PM, Adam N. Rosenberg wrote: > Hi Dave, > > ? ?I reckon it does have everything, except the basic stuff of how > to setup and to run the programs. ?They figure you already know that. > I'm looking for real baby steps for a first-time Vimeo programmer. > Might I humbly suggest if you are looking for vimeo support, the ffmpeg users mailing list is not the best place to look. If you were looking for baby steps for a first time ffmpeg programmer, you would have been exactly right... Cheers Tom From longxd at gmail.com Thu Mar 1 16:59:36 2012 From: longxd at gmail.com (X. Long) Date: Thu, 1 Mar 2012 10:59:36 -0500 Subject: [FFmpeg-user] Where to find the version ffmpeg version N-31774-g6c4e9ca Message-ID: I run into a problem of frame alignment between different version of ffmpeg. I have a video decoded by someone else by the above version with some information generated after processing the image. Now I also decoded the video and processed the image and another type of information generated. I want to marry the two types of information but first I need to find the correct frame correspondence, which I find a difficult task. I am using the newest version of ffmpeg on windows, and the old version someone else used is attached below, where can I download the exact version of the software for: ffmpeg version N-31774-g6c4e9ca? Thanks Cindy ------------------------------------- ffmpeg version N-31774-g6c4e9ca, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 6 2011 22:22:11 with gcc 4.6.1 configuration: --enable-gpl --enable-version3 --enable-memalign-hack --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 11. 1 / 51. 11. 1 libavcodec 53. 9. 1 / 53. 9. 1 libavformat 53. 6. 0 / 53. 6. 0 libavdevice 53. 2. 0 / 53. 2. 0 libavfilter 2. 28. 0 / 2. 28. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [wmv3 @ 01DDAB00] Extra data: 8 bits left, value: 0 Input #0, asf, from ?20110826151009068.wmv': Metadata: WMFSDKVersion : 12.0.7601.17514 WMFSDKNeeded : 0.0.0.0000 IsVBR : 1 VBR Peak : 50010.0000 Buffer Average : 63940.0000 Duration: 00:25:57.94, start: 0.000000, bitrate: 38455 kb/s Stream #0.0(eng): Video: wmv3 (Main), yuv420p, 1600x1200, 49480 kb/s, 1k tbr, 1k tbn, 1k tbc At least one output file must be specified From nicolas.george at normalesup.org Thu Mar 1 17:13:07 2012 From: nicolas.george at normalesup.org (Nicolas George) Date: Thu, 1 Mar 2012 17:13:07 +0100 Subject: [FFmpeg-user] Where to find the version ffmpeg version N-31774-g6c4e9ca In-Reply-To: References: Message-ID: <20120301161307.GA19772@phare.normalesup.org> Le duodi 12 vent?se, an CCXX, X. Long a ?crit?: > I am using the newest version of ffmpeg on windows, and the old > version someone else used is attached below, where can I download the > exact version of the software for: ffmpeg version N-31774-g6c4e9ca? The strange string at the end is the Git short hash, prefixed by "g", it identifies the version exactly. If you have a complete git clone of the source tree, the following command will put the tree at that version: git checkout 6c4e9ca (for references, it dates back to Sun Aug 7 00:32:11 2011 +0200) Alternatively, you can browse the gitweb repository on . I do not know how to look for a particular commit, though, but putting "h=6c4e9ca" in the URL seems to do the trick. Then you can ask for a snapshot. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From ffmpeg-user at dogtoe.com Thu Mar 1 20:07:09 2012 From: ffmpeg-user at dogtoe.com (Brian Johnson) Date: Thu, 1 Mar 2012 11:07:09 -0800 Subject: [FFmpeg-user] Problem "live" transcoding via piped input / Ceton InfiniTV 4 Message-ID: The command I am using: ffmpeg -i /dev/ctn91xx_mpeg0_0 -async 2 -acodec libmp3lame -ac 2 -vcodec libx264 -s 1280x720 -aspect 16:9 -f mpegts file.ts The file /dev/ctn91xx_mpeg0_0 is actually the device (Ceton InfiniTV 4) itself where it returns a mpeg transport stream, basically a named pipe containing the raw audio and video. When I run the command below, I get tons of errors which repeats forever: ES packet size mismatch.0 size=???? 797kB time=00:00:04.23 bitrate=1541.1kbits/s dup=25 drop=0 [ac3 @ 0x8393340] frame CRC mismatch [mpeg2video @ 0x834e680] ac-tex damaged at 0 28 [mpeg2video @ 0x834e680] ac-tex damaged at 2 1 [mpeg2video @ 0x834e680] ac-tex damaged at 0 27 [mpeg2video @ 0x834e680] ac-tex damaged at 0 29 [mpeg2video @ 0x834e680] Warning MVs not available [mpeg2video @ 0x834e680] concealing 1276 DC, 1276 AC, 1276 MV errors [mpegts @ 0x8287aa0] PES packet size mismatch frame sync error 34 q=29.0 size=??? 1202kB time=00:00:05.76 bitrate=1709.0kbits/s dup=145 drop=0 Error while decoding stream #0:27 ac-tex damaged at 4 2929.0 size=??? 1383kB time=00:00:09.17 bitrate=1235.1kbits/s dup=145 drop=0 [mpeg2video @ 0x834e680] mb incr damaged [mpeg2video @ 0x834e680] ac-tex damaged at 35 29 [mpeg2video @ 0x834e680] concealing 880 DC, 880 AC, 880 MV errors [mpegts @ 0x8287aa0] PES packet size mismatch [ac3 @ 0x8393340] frame CRC mismatch [mpegts @ 0x8287aa0] PES packet size mismatch ??? Last message repeated 1 times [ac3 @ 0x8393340] frame sync error Error while decoding stream #0:27 [mpeg2video @ 0x834e680] invalid cbp at 0 12 [mpeg2video @ 0x834e680] mb incr damaged [mpeg2video @ 0x834e680] invalid mb type in B Frame at 3 24 [mpeg2video @ 0x834e680] 00 motion_type at 1 25 [mpeg2video @ 0x834e680] 00 motion_type at 0 26 [mpeg2video @ 0x834e680] invalid mb type in B Frame at 16 27 What is very interesting is that lower quality video transcodes (i.e. 320x240) has the same errors as above but only lasts about 1-2 seconds, and then output looks normal frame= 1235 fps= 35 q=29.0 size= 1985kB time=00:00:39.53 bitrate= 411.2kbits/s dup=131 drop=0 I believe this to be related to and in my own research, I found that this may have been a regression from an older version, but again that is pure speculation on my part. Any ideas? :) - Brian From longxd at gmail.com Thu Mar 1 20:51:27 2012 From: longxd at gmail.com (X. Long) Date: Thu, 1 Mar 2012 14:51:27 -0500 Subject: [FFmpeg-user] Where to find the version ffmpeg version N-31774-g6c4e9ca In-Reply-To: <20120301161307.GA19772@phare.normalesup.org> References: <20120301161307.GA19772@phare.normalesup.org> Message-ID: Nicolas: Thanks for the reply. It is very helpful. I am able to find the source code.. But I want to run it on windows, is there a built based on this version of the source code, or is it a easy way to build this so that the software can run on windows. Thanks Cindy 2012/3/1 Nicolas George : > Le duodi 12 vent?se, an CCXX, X. Long a ?crit?: >> I am using the newest version of ffmpeg on windows, and the old >> version someone else used is attached below, where can I download the >> exact version of the software for: ffmpeg version N-31774-g6c4e9ca? > > The strange string at the end is the Git short hash, prefixed by "g", it > identifies the version exactly. If you have a complete git clone of the > source tree, the following command will put the tree at that version: > > git checkout 6c4e9ca > > (for references, it dates back to Sun Aug 7 00:32:11 2011 +0200) > > Alternatively, you can browse the gitweb repository on http://source.ffmpeg.org/ >. I do not know how to look for a particular > commit, though, but putting "h=6c4e9ca" in the URL seems to do the trick. > Then you can ask for a snapshot. > > Regards, > > -- > ?Nicolas George > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.11 (GNU/Linux) > > iEYEARECAAYFAk9PoBMACgkQsGPZlzblTJOVoQCdEvFQGnTAYVoBPOpA9iuSR9BP > ZvUAni2ISZ/HKeP2Ioc5TH93NyqRT40a > =28Oz > -----END PGP SIGNATURE----- > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From wodfer at gmail.com Thu Mar 1 23:36:54 2012 From: wodfer at gmail.com (Andy Wodfer) Date: Thu, 1 Mar 2012 23:36:54 +0100 Subject: [FFmpeg-user] Problems encoding to DV25 .avi file In-Reply-To: References: Message-ID: On Thu, Mar 1, 2012 at 12:51 PM, Andy Wodfer wrote: > Hi, > I need some input on a problem I have. > > I'm using the latest 0.10 version of ffmpeg compiled from source on a > FreeBSD 8.2 machine. > > The source file is a Prores 422HQ file (1280x720 @ 50fps 720P). > > I want to make this into a DV25 .avi file that is playable in Windows > Media Player and that will work in Ie. Adobe Premiere Pro 2.0 on a WinXP > machine. Ie: A Windows compatible DV25 file. > > I've tried several options, but I can't make it work. > > "-target pal-dv" gives me a file that plays fine in VLC player, but not in > Windows Media Player. Adobe Premiere reports unsupported format and won't > import it. > > "-vcodec dvvideo -s 720x576 -pix_fmt yuv420p -r 25 -acodec pcm_s16le -ac > 2 -ar 48000" works if I choose .mov as a container. It opens in Quicktime > and VLC and import to Adobe Premiere works fine. Aspect ratio is also fine. > > If I change the container to .avi it won't play back in Windows Media > Player. > > Any ideas of what I can try next? > > Thanks!! > Best regards, > Andy > I seem to be making some progress with these parameters: -vcodec dvvideo -s 720x576 -vtag dvsd -pix_fmt yuv420p -r 25 -f avi -acodec pcm_s16le -ac 2 -ar 4 8000 However, when playing back in WMP (which finally works) the 16x9 image is squeezed into a 4x3 frame. I need to make my parameters flexible so they can take either 4x3 or 16x9 and keep the aspect ratio on the output files, even if it's 16x9 HD downscaled to PAL DV. Any ideas? From batguano999 at hotmail.com Fri Mar 2 00:01:38 2012 From: batguano999 at hotmail.com (bat guano) Date: Thu, 1 Mar 2012 23:01:38 +0000 Subject: [FFmpeg-user] Where to find the version ffmpeg version N-31774-g6c4e9ca In-Reply-To: References: , <20120301161307.GA19772@phare.normalesup.org>, Message-ID: ---------------------------------------- .. But I want to run it on windows, is there a built based on this > version of the source code Hi Windows build ffmpeg-git-6c4e9ca-win32-static.7z? 06-Aug-2011 It's available from here ---> http://ffmpeg.zeranoe.com/builds/win32/static/? ;-) From renlifeng at gmail.com Fri Mar 2 06:23:56 2012 From: renlifeng at gmail.com (Lifeng Ren) Date: Fri, 2 Mar 2012 13:23:56 +0800 Subject: [FFmpeg-user] [ffplay] after changing to another audio stream, audio will become out of sync with video Message-ID: hi, I want to report that after changing to another audio stream, audio and video becomes out of sync. steps to reproduce: $ ffplay video-with-2-audio-stream.mpg a (press a to toggle audio stream) I also find this can be avoided by patching ffplay.c as such. diff --git a/ffplay.c b/ffplay.c index 93097e1..ac31ce6 100644 --- a/ffplay.c +++ b/ffplay.c @@ -2725,6 +2725,7 @@ static void stream_cycle_channel(VideoState *is, int codec_type) AVFormatContext *ic = is->ic; int start_index, stream_index; AVStream *st; + double pos; if (codec_type == AVMEDIA_TYPE_VIDEO) start_index = is->video_stream; @@ -2765,8 +2766,10 @@ static void stream_cycle_channel(VideoState *is, int codec_type) } } the_end: + pos = get_master_clock(is); stream_component_close(is, start_index); stream_component_open(is, stream_index); + stream_seek(is, (int64_t)(pos * AV_TIME_BASE), 0, 0); } From nichot20 at yahoo.com Fri Mar 2 09:11:05 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Fri, 02 Mar 2012 08:11:05 +0000 Subject: [FFmpeg-user] Problems encoding to DV25 .avi file In-Reply-To: References: Message-ID: <4F508099.9020602@yahoo.com> On 01/03/12 22:36, Andy Wodfer wrote: > On Thu, Mar 1, 2012 at 12:51 PM, Andy Wodfer wrote: > >> Hi, >> I need some input on a problem I have. >> >> I'm using the latest 0.10 version of ffmpeg compiled from source on a >> FreeBSD 8.2 machine. >> 0.10 is not the latest, use git HEAD if you want that. >> The source file is a Prores 422HQ file (1280x720 @ 50fps 720P). >> >> I want to make this into a DV25 .avi file that is playable in Windows >> Media Player and that will work in Ie. Adobe Premiere Pro 2.0 on a WinXP >> machine. Ie: A Windows compatible DV25 file. >> >> I've tried several options, but I can't make it work. >> >> "-target pal-dv" gives me a file that plays fine in VLC player, but not in >> Windows Media Player. Adobe Premiere reports unsupported format and won't >> import it. >> >> "-vcodec dvvideo -s 720x576 -pix_fmt yuv420p -r 25 -acodec pcm_s16le -ac >> 2 -ar 48000" works if I choose .mov as a container. It opens in Quicktime >> and VLC and import to Adobe Premiere works fine. Aspect ratio is also fine. >> >> If I change the container to .avi it won't play back in Windows Media >> Player. [..] > I seem to be making some progress with these parameters: > > -vcodec dvvideo -s 720x576 -vtag dvsd -pix_fmt yuv420p -r 25 -f avi -acodec > pcm_s16le -ac 2 -ar 4 > 8000 > > However, when playing back in WMP (which finally works) the 16x9 image is > squeezed into a 4x3 frame. > > I need to make my parameters flexible so they can take either 4x3 or 16x9 > and keep the aspect ratio on the output files, even if it's 16x9 HD > downscaled to PAL DV. Two thoughts. You may need to use the setdar filter to force the display aspect ratio. I'm not sure if WMP recognises this setting in AVI's. Premi?re Pro seems to be OK with it. -- Tim From de.techno at gmail.com Fri Mar 2 10:19:59 2012 From: de.techno at gmail.com (dE .) Date: Fri, 02 Mar 2012 14:49:59 +0530 Subject: [FFmpeg-user] Problem "live" transcoding via piped input / Ceton InfiniTV 4 In-Reply-To: References: Message-ID: <4F5090BF.8000405@gmail.com> On 03/02/12 00:37, Brian Johnson wrote: > The command I am using: ffmpeg -i /dev/ctn91xx_mpeg0_0 -async 2 > -acodec libmp3lame -ac 2 -vcodec libx264 -s 1280x720 -aspect 16:9 -f > mpegts file.ts > > The file /dev/ctn91xx_mpeg0_0 is actually the device (Ceton InfiniTV > 4) itself where it returns a mpeg transport stream, basically a named > pipe containing the raw audio and video. > > When I run the command below, I get tons of errors which repeats forever: > > ES packet size mismatch.0 size= 797kB time=00:00:04.23 > bitrate=1541.1kbits/s dup=25 drop=0 > [ac3 @ 0x8393340] frame CRC mismatch > [mpeg2video @ 0x834e680] ac-tex damaged at 0 28 > [mpeg2video @ 0x834e680] ac-tex damaged at 2 1 > [mpeg2video @ 0x834e680] ac-tex damaged at 0 27 > [mpeg2video @ 0x834e680] ac-tex damaged at 0 29 > [mpeg2video @ 0x834e680] Warning MVs not available > [mpeg2video @ 0x834e680] concealing 1276 DC, 1276 AC, 1276 MV errors > [mpegts @ 0x8287aa0] PES packet size mismatch > frame sync error 34 q=29.0 size= 1202kB time=00:00:05.76 > bitrate=1709.0kbits/s dup=145 drop=0 > Error while decoding stream #0:27 > ac-tex damaged at 4 2929.0 size= 1383kB time=00:00:09.17 > bitrate=1235.1kbits/s dup=145 drop=0 > [mpeg2video @ 0x834e680] mb incr damaged > [mpeg2video @ 0x834e680] ac-tex damaged at 35 29 > [mpeg2video @ 0x834e680] concealing 880 DC, 880 AC, 880 MV errors > [mpegts @ 0x8287aa0] PES packet size mismatch > [ac3 @ 0x8393340] frame CRC mismatch > [mpegts @ 0x8287aa0] PES packet size mismatch > Last message repeated 1 times > [ac3 @ 0x8393340] frame sync error > Error while decoding stream #0:27 > [mpeg2video @ 0x834e680] invalid cbp at 0 12 > [mpeg2video @ 0x834e680] mb incr damaged > [mpeg2video @ 0x834e680] invalid mb type in B Frame at 3 24 > [mpeg2video @ 0x834e680] 00 motion_type at 1 25 > [mpeg2video @ 0x834e680] 00 motion_type at 0 26 > [mpeg2video @ 0x834e680] invalid mb type in B Frame at 16 27 > > What is very interesting is that lower quality video transcodes (i.e. > 320x240) has the same errors as above but only lasts about 1-2 > seconds, and then output looks normal > > frame= 1235 fps= 35 q=29.0 size= 1985kB time=00:00:39.53 bitrate= > 411.2kbits/s dup=131 drop=0 > > I believe this to be related to > > and in my own research, I found that this may have been a regression > from an older version, but again that is pure speculation on my part. > > Any ideas? :) > > - Brian > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Are you using the latest? This may be cause your PC is not as fast. Try using -threads and -preset ultrafast. From pavel at sokolov.me Fri Mar 2 12:11:53 2012 From: pavel at sokolov.me (Pavel Sokolov) Date: Fri, 02 Mar 2012 15:11:53 +0400 Subject: [FFmpeg-user] AVI/AC3 -> VOB/LPCM In-Reply-To: <4F4F3D93.6020202@sokolov.me> References: <4F4E5D37.5000802@sokolov.me> <4F4F3D93.6020202@sokolov.me> Message-ID: <4F50AAF9.9010402@sokolov.me> 01.03.2012 13:12, Pavel Sokolov ?????: > > 29.02.2012 23:10, Andrey Utkin ?????: >> 2012/2/29 Pavel Sokolov: >>> Hi all! >>> >>> How can I do remux of the AVI with AC3 stereo to the VOB/LPCM >>> >>> When I tried the next cmd I hear bad audio (sh-sh-sh-sh): >>> ffmpeg -i video-mpeg4_720x544-audio_ac3_48000_stereo.avi -vcodec copy >>> -acodec pcm_s16be -sample_fmt s16 -f vob test1.vob >>> >>> What I'm doing wrong? >> Full uncut console output is missing... >> Also publish your source file, can help. >> Sound is bad on PC also, or just on your specific hardware? >> > file: > http://sokolov.me/tmp/video-mpeg4_720x544-audio_ac3_48000_stereo.avi > > ffmpeg log: > > [vob @ 0xdf5be0] buffer underflow i=1 bufi=4026 size=6144 > Last message repeated 34 times > [vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it > [vob @ 0xdf5be0] buffer underflow i=1 bufi=4026 size=6144 > [vob @ 0xdf5be0] buffer underflow i=1 bufi=6043 size=6144 > ... Is somebody know how to solve this issue? -- With best regards, Pavel A. Sokolov mobile: +7(921)419-1819 skype: pavel_a_sokolov From wodfer at gmail.com Fri Mar 2 12:14:39 2012 From: wodfer at gmail.com (Andy Wodfer) Date: Fri, 2 Mar 2012 12:14:39 +0100 Subject: [FFmpeg-user] Problems encoding to DV25 .avi file In-Reply-To: <4F508099.9020602@yahoo.com> References: <4F508099.9020602@yahoo.com> Message-ID: On Fri, Mar 2, 2012 at 9:11 AM, Tim Nicholson wrote: > On 01/03/12 22:36, Andy Wodfer wrote: > > On Thu, Mar 1, 2012 at 12:51 PM, Andy Wodfer wrote: > > > >> Hi, > >> I need some input on a problem I have. > >> > >> I'm using the latest 0.10 version of ffmpeg compiled from source on a > >> FreeBSD 8.2 machine. > >> > > 0.10 is not the latest, use git HEAD if you want that. > > >> The source file is a Prores 422HQ file (1280x720 @ 50fps 720P). > >> > >> I want to make this into a DV25 .avi file that is playable in Windows > >> Media Player and that will work in Ie. Adobe Premiere Pro 2.0 on a WinXP > >> machine. Ie: A Windows compatible DV25 file. > >> > >> I've tried several options, but I can't make it work. > >> > >> "-target pal-dv" gives me a file that plays fine in VLC player, but not > in > >> Windows Media Player. Adobe Premiere reports unsupported format and > won't > >> import it. > >> > >> "-vcodec dvvideo -s 720x576 -pix_fmt yuv420p -r 25 -acodec pcm_s16le > -ac > >> 2 -ar 48000" works if I choose .mov as a container. It opens in > Quicktime > >> and VLC and import to Adobe Premiere works fine. Aspect ratio is also > fine. > >> > >> If I change the container to .avi it won't play back in Windows Media > >> Player. > > [..] > > > I seem to be making some progress with these parameters: > > > > -vcodec dvvideo -s 720x576 -vtag dvsd -pix_fmt yuv420p -r 25 -f avi > -acodec > > pcm_s16le -ac 2 -ar 4 > > 8000 > > > > However, when playing back in WMP (which finally works) the 16x9 image is > > squeezed into a 4x3 frame. > > > > I need to make my parameters flexible so they can take either 4x3 or 16x9 > > and keep the aspect ratio on the output files, even if it's 16x9 HD > > downscaled to PAL DV. > > Two thoughts. > > You may need to use the setdar filter to force the display aspect ratio. > I'm not sure if WMP recognises this setting in AVI's. Premi?re Pro seems > to be > OK with it. Thanks for your input Tim! I tested some more with the file I generated with the -vtag and it seems to work perfectly in Premiere. Correct aspect and no error messages. I seem to recollect that WMP has had problems playing back 16x9 DV material from a DV25 .avi file before. /Andy From yogesh.bit2006 at gmail.com Fri Mar 2 14:42:48 2012 From: yogesh.bit2006 at gmail.com (Yogesh Tyagi) Date: Fri, 2 Mar 2012 19:12:48 +0530 Subject: [FFmpeg-user] how to enable hardware acceleration in ffmpeg Message-ID: Hi, Is there any command line option in ffmpeg to enable hardware acceleration for h264 decoding.I have configured and compiled ffmpeg with" --enable-vaapi --enable-hwaccel=h264_vaapi" options but ffmpeg is not invoking decode_slice function of vaapi_h264.c. Thanks and Regards, Yogesh From tevans.uk at googlemail.com Fri Mar 2 15:29:13 2012 From: tevans.uk at googlemail.com (Tom Evans) Date: Fri, 2 Mar 2012 14:29:13 +0000 Subject: [FFmpeg-user] how to enable hardware acceleration in ffmpeg In-Reply-To: References: Message-ID: On Fri, Mar 2, 2012 at 1:42 PM, Yogesh Tyagi wrote: > Hi, > > Is there any command line option in ffmpeg to enable hardware acceleration > for h264 decoding.I have configured and compiled ffmpeg with" > --enable-vaapi --enable-hwaccel=h264_vaapi" options but ffmpeg is not > invoking ?decode_slice function of vaapi_h264.c. > As I understand it, vdpau and vaapi are useful for decoding when you want to display that decoded content. It's not a usable API, eg for decoding video as a step for subsequently re-encoding it (if that makes sense). IE, you can use vdpau/vaapi to display media using hwaccel, but you can't use it to accelerate decoding whilst encoding. Cheers Tom From exampte01 at hotmail.fr Fri Mar 2 18:15:34 2012 From: exampte01 at hotmail.fr (aple ex) Date: Fri, 2 Mar 2012 18:15:34 +0100 Subject: [FFmpeg-user] Problem transcoding from MP4 (H264 - AAC) to OGG (DIRAC - FLAC) In-Reply-To: <4F46A788.9070406@gmail.com> References: <4F46A788.9070406@gmail.com> Message-ID: > Date: Thu, 23 Feb 2012 17:54:32 -0300 > From: agprus at gmail.com > To: ffmpeg-user at ffmpeg.org > Subject: [FFmpeg-user] Problem transcoding from MP4 (H264 - AAC) to OGG (DIRAC - FLAC) > > Hello, > > I am trying to do a lossy-to-lossless transcoding from an MP4 file (with > one H264 video stream and one AAC audio stream) to an OGG file (with a > DIRAC video stream and a FLAC audio stream). The command line I am using > is this: > > ffmpeg -i input.mp4 -vcodec libschroedinger -q:v 0 -acodec flac output.ogg > > and I am getting the following errors (complete output is at the end): > > [ogg @ 0000000001F29570] Unsupported codec id in stream 0 > Could not write header for output file #0 (incorrect codec parameters ?) > > Does anybody know an appropriate way to do this kind of transcoding? > > Thanks in advance, > Alexis > > ffmpeg version N-38148-gb6ff81d Copyright (c) 2000-2012 the FFmpeg > developers > built on Feb 23 2012 12:30:44 with gcc 4.6.2 > configuration: --enable-gpl --enable-version3 --disable-w32threads > --enable-runtime-cpudetect --enable-avisynth --enable-bzlib > --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-libfreetype --enable-libgsm --enable-libmp3lame > --enable-libopenjpeg --enable-librtmp --enable-libschroedinger > --enable-libspeex --enable-libtheora --enable-libvo-aacenc > --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx > --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib > libavutil 51. 40.100 / 51. 40.100 > libavcodec 54. 4.100 / 54. 4.100 > libavformat 54. 1.100 / 54. 1.100 > libavdevice 53. 4.100 / 53. 4.100 > libavfilter 2. 62.101 / 2. 62.101 > libswscale 2. 1.100 / 2. 1.100 > libswresample 0. 7.100 / 0. 7.100 > libpostproc 52. 0.100 / 52. 0.100 > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'h264aac1.mp4': > Metadata: > major_brand : isom > minor_version : 512 > compatible_brands: isomiso2avc1mp41 > creation_time : 2010-02-16 19:06:52 > encoder : Lavf52.36.0 > Duration: 00:03:40.03, start: 0.000000, bitrate: 2571 kb/s > Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / > 0x31637661), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 2537 kb/s, 30 fps, 30 > tbr, 30 tbn, 60 tbc > Metadata: > creation_time : 2010-02-16 19:06:52 > handler_name : VideoHandler > Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 11025 Hz, stereo, > s16, 31 kb/s > Metadata: > creation_time : 2010-02-16 19:06:52 > handler_name : > [buffer @ 0000000001F29F30] w:640 h:480 pixfmt:yuv420p tb:1/1000000 > sar:1/1 sws_param: > [ogg @ 0000000001F29570] Unsupported codec id in stream 0 > Output #0, ogg, to 'h264aac.ogg': > Metadata: > major_brand : isom > minor_version : 512 > compatible_brands: isomiso2avc1mp41 > creation_time : 2010-02-16 19:06:52 > encoder : Lavf54.1.100 > Stream #0:0(und): Video: dirac, yuv420p, 640x480 [SAR 1:1 DAR 4:3], > q=2-31, 200 kb/s, 30 tbn, 30 tbc > Metadata: > creation_time : 2010-02-16 19:06:52 > handler_name : VideoHandler > Stream #0:1(und): Audio: flac, 11025 Hz, stereo, s16, 128 kb/s > Metadata: > creation_time : 2010-02-16 19:06:52 > handler_name : > Stream mapping: > Stream #0:0 -> #0:0 (h264 -> libschroedinger) > Stream #0:1 -> #0:1 (aac -> flac) > Could not write header for output file #0 (incorrect codec parameters ?) Hi according to their website dirac codec is supposed to be contained by ogg, see here : http://diracvideo.org/schrodinger-faq/ (specially Question 6) I have tested myself and I obtain the same result. I can provide my console output if necessary but dirac can be encoded in mpegts container. Regards From pobrejuanito at gmail.com Fri Mar 2 18:38:15 2012 From: pobrejuanito at gmail.com (pobre) Date: Fri, 2 Mar 2012 09:38:15 -0800 (PST) Subject: [FFmpeg-user] Generate 64kbps audio-only MPEGTS from MP4 Message-ID: <1330709895585-4439211.post@n4.nabble.com> One of the Apple App Store's requirement for video apps is having a baseline 64kbps audio-only stream. I am having trouble figuring out how to convert my mp4 videos into mpegts to include audio-only. Any hints as to how to generate 64kbps audio-only mpegts from mp4? It would be nice to have a black screen and only audio playing which totals 64kbps. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Generate-64kbps-audio-only-MPEGTS-from-MP4-tp4439211p4439211.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From tomnmillie at yahoo.com Fri Mar 2 18:49:40 2012 From: tomnmillie at yahoo.com (tomnmillie) Date: Fri, 2 Mar 2012 09:49:40 -0800 (PST) Subject: [FFmpeg-user] Invalid data found when processing input In-Reply-To: References: <1330467369591-4429788.post@n4.nabble.com> <4F4DEADF.90300@gmail.com> <1330535287541-4432184.post@n4.nabble.com> <1330542930467-4432632.post@n4.nabble.com> Message-ID: <1330710580121-4439243.post@n4.nabble.com> Thanks so much Carl! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Invalid-data-found-when-processing-input-tp4429788p4439243.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From andrey.krieger.utkin at gmail.com Fri Mar 2 19:00:34 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Fri, 2 Mar 2012 20:00:34 +0200 Subject: [FFmpeg-user] Generate 64kbps audio-only MPEGTS from MP4 In-Reply-To: <1330709895585-4439211.post@n4.nabble.com> References: <1330709895585-4439211.post@n4.nabble.com> Message-ID: 2012/3/2 pobre : > One of the Apple App Store's requirement for video apps is having a baseline > 64kbps audio-only stream. ?I am having trouble figuring out how to convert > my mp4 videos into mpegts to include audio-only. > > Any hints as to how to generate 64kbps audio-only mpegts from mp4? ?It would > be nice to have a black screen and only audio playing which totals 64kbps. Sth like ffmpeg -i yourfile.mp4 -vn -acodec libfaac -b:a 64k out.ts -- Andrey Utkin From rosko at thirdiris.com Thu Mar 1 21:22:35 2012 From: rosko at thirdiris.com (Steve Roskowski) Date: Thu, 1 Mar 2012 12:22:35 -0800 Subject: [FFmpeg-user] live transcoding flv files Message-ID: I have an http server which generates FLV format video "on the fly", which allows flash to play them back in realtime - live streaming. I would like to use ffmpeg to transcode them, specifically to lower bit rates for mobile clients. I cannot figure out how to get ffmpeg to do the transcoding in realtime. FFMPEG seems to download the entire source file before beginning to generate output. VLC shows the exact same behavior - video does not start playing till the entire flv file is downloaded... which on a live stream is forever. is this something I am missing in the options/configuration, or is this something specific to the flv demuxer? typical ffmpeg command ffmpeg -i http://myserver/live_stream1/flv -b 64k -vcodec libx264 foo.flv works fine, but only starts rendering the output file when the live stream is closed. -- Steve Roskowski From funkyirish at gmail.com Thu Mar 1 21:45:34 2012 From: funkyirish at gmail.com (booglyboo) Date: Thu, 1 Mar 2012 12:45:34 -0800 (PST) Subject: [FFmpeg-user] bgra to yuv Message-ID: <1330634734197-4436338.post@n4.nabble.com> I've been trying to convert a 1920x1080 BGRA 30 fps video (no sound) which is in the avi container into a yuv file for the purpose of using dirac for compression. I've tried to use the information at http://ffmpeg-users.933282.n4.nabble.com/YUV-with-alpha-channel-td4351120.html, but the framerate messes up and the file size changes so dramatically that I don't believe the conversion was lossless. To clarify: I tried the exact command and got a .nut file which seemed too small, then I changed the -pix_fmt yuva420p to -pix_fmt yuv420p and tried to output a .avi file. This resulted in something that was garbage. I have tried many different things to convert this file including some software that claimed it would do it but nothing preserves the video. I could use some help if anyone has the time. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/bgra-to-yuv-tp4436338p4436338.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From pobrejuanito at gmail.com Fri Mar 2 19:21:09 2012 From: pobrejuanito at gmail.com (pobre) Date: Fri, 2 Mar 2012 10:21:09 -0800 (PST) Subject: [FFmpeg-user] Generate 64kbps audio-only MPEGTS from MP4 In-Reply-To: References: <1330709895585-4439211.post@n4.nabble.com> Message-ID: <1330712469869-4439344.post@n4.nabble.com> Gosh! Thank you! I added -f mpegts for the segmenter to break it up! ffmpeg -i myvideo.mp4 -f mpegts -vn -acodec libfaac -b:a 64k And it works! Is there a way I can add an image to the audio playing? I'll be submitting it to Apple Store to see if I meet their requirement. Thanks! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Generate-64kbps-audio-only-MPEGTS-from-MP4-tp4439211p4439344.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From andrey.krieger.utkin at gmail.com Fri Mar 2 19:27:25 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Fri, 2 Mar 2012 20:27:25 +0200 Subject: [FFmpeg-user] Generate 64kbps audio-only MPEGTS from MP4 In-Reply-To: <1330712469869-4439344.post@n4.nabble.com> References: <1330709895585-4439211.post@n4.nabble.com> <1330712469869-4439344.post@n4.nabble.com> Message-ID: 2012/3/2 pobre : > Gosh! Thank you! I added -f mpegts for the segmenter to break it up! > > ffmpeg -i myvideo.mp4 -f mpegts -vn -acodec libfaac -b:a 64k > > And it works! > > Is there a way I can add an image to the audio playing? > > > I'll be submitting it to Apple Store to see if I meet their requirement. Don't exactly what you got (without image) is what is required (audio only?) I'm not an expert in apple-compliant streams, but... "Image" you add will convert into video stream, which will take its part of bitrate, and result will be certainly not just 64kbit audio-only stream. -- Andrey Utkin From maurice at cmdrkey.com Fri Mar 2 19:24:54 2012 From: maurice at cmdrkey.com (Maurice Randall) Date: Fri, 02 Mar 2012 13:24:54 -0500 Subject: [FFmpeg-user] live transcoding flv files In-Reply-To: References: Message-ID: <3de204bf13c0f1f35d80f19ecdc272e9@web.speedwaymail.net> On 2012-03-01 15:22, Steve Roskowski wrote: > I cannot figure out how to get ffmpeg to do the transcoding in > realtime. > FFMPEG seems to download the entire source file before beginning to > generate output. VLC shows the exact same behavior - video does not > start > playing till the entire flv file is downloaded... which on a live > stream is > forever. > ffmpeg -i http://myserver/live_stream1/flv -b 64k -vcodec libx264 > foo.flv > > works fine, but only starts rendering the output file when the live > stream > is closed. Is it possible to fetch the flv's from the same server using the rtmp protocol? If so, that would do what you are after. If only http can be used, maybe someone else can come up with an answer for you. -Maurice From squeaky at sdf.org Fri Mar 2 20:31:41 2012 From: squeaky at sdf.org (Gary Taylor) Date: Fri, 2 Mar 2012 11:31:41 -0800 Subject: [FFmpeg-user] Getting a point in time as known by ffmpeg? Message-ID: <20120302193141.GC6507@SDF.ORG> When I watch a video (in xine or gxine for example)I see a value displayed which is how long the video has been playing. If I find a scene where I want to process the video starting at 00:00:40.000 as displayed in xine, in ffmpeg I have to experiment to find that same spot. It's not at 00:00:40.000, but instead 00:00:63.000. It's like the notion of time in ffmpeg is different than what is displayed by the players. Is this user error, should I be viewing them in something else, or is there a way to use ffplay to see display the time? fmpeg version N-37798-gcd1c12b Copyright (c) 2000-2012 the FFmpeg developers built on Feb 11 2012 23:02:52 with gcc 4.4.3 Thanks, Gary -- SDF Public Access UNIX System - http://sdf.lonestar.org From lou at lrcd.com Fri Mar 2 20:40:44 2012 From: lou at lrcd.com (Lou) Date: Fri, 2 Mar 2012 10:40:44 -0900 Subject: [FFmpeg-user] Getting a point in time as known by ffmpeg? In-Reply-To: <20120302193141.GC6507@SDF.ORG> References: <20120302193141.GC6507@SDF.ORG> Message-ID: <20120302104044.4e9c8666@lrcd.com> On Fri, 2 Mar 2012 11:31:41 -0800 Gary Taylor wrote: > When I watch a video (in xine or gxine for example)I see a > value displayed which is how long the video has been > playing. If I find a scene where I want to process the > video starting at 00:00:40.000 as displayed in xine, in > ffmpeg I have to experiment to find that same spot. It's not > at 00:00:40.000, but instead 00:00:63.000. > > It's like the notion of time in ffmpeg is different than > what is displayed by the players. Is this user error, > should I be viewing them in something else, or is there a > way to use ffplay to see display the time? > > fmpeg version N-37798-gcd1c12b Copyright (c) 2000-2012 the > FFmpeg developers > built on Feb 11 2012 23:02:52 with gcc 4.4.3 > > > Thanks, > Gary > ffplay shows the current time in the last line: $ ffplay input.mkv ... 3.05 A-V: 0.002 fd= 1 aq= 320KB vq= 5357KB sq= 0B f=0/0 f=0/0 In this example 3.05 seconds has elapsed. Also note that the behavior of the -ss option can change depending if it is used as an input or output option. From coniophora at gmail.com Fri Mar 2 21:00:34 2012 From: coniophora at gmail.com (Jim Worrall) Date: Fri, 2 Mar 2012 13:00:34 -0700 Subject: [FFmpeg-user] Where to find the version ffmpeg version N-31774-g6c4e9ca In-Reply-To: <20120301161307.GA19772@phare.normalesup.org> References: <20120301161307.GA19772@phare.normalesup.org> Message-ID: <68C2A4F9-CEE3-484C-9B6E-DDD3918C0DFC@gmail.com> On Mar 1, 2012, at 9:13 AM, Nicolas George wrote: > Le duodi 12 vent?se, an CCXX, X. Long a ?crit : >> I am using the newest version of ffmpeg on windows, and the old >> version someone else used is attached below, where can I download the >> exact version of the software for: ffmpeg version N-31774-g6c4e9ca? > > The strange string at the end is the Git short hash, prefixed by "g", it > identifies the version exactly. If you have a complete git clone of the > source tree, the following command will put the tree at that version: OK, I learned something there. So what is the significance of the N-31774 part of the version? From sheen.andy at googlemail.com Fri Mar 2 21:04:16 2012 From: sheen.andy at googlemail.com (Andy Sheen) Date: Fri, 02 Mar 2012 20:04:16 +0000 Subject: [FFmpeg-user] Where to find the version ffmpeg version N-31774-g6c4e9ca In-Reply-To: <68C2A4F9-CEE3-484C-9B6E-DDD3918C0DFC@gmail.com> References: <20120301161307.GA19772@phare.normalesup.org> <68C2A4F9-CEE3-484C-9B6E-DDD3918C0DFC@gmail.com> Message-ID: <4F5127C0.4080905@googlemail.com> Jim Worrall wrote on Fri 02 Mar at 20:00 UK time > > On Mar 1, 2012, at 9:13 AM, Nicolas George wrote: > >> Le duodi 12 vent?se, an CCXX, X. Long a ?crit : >>> I am using the newest version of ffmpeg on windows, and the old >>> version someone else used is attached below, where can I download the >>> exact version of the software for: ffmpeg version N-31774-g6c4e9ca? >> >> The strange string at the end is the Git short hash, prefixed by "g", it >> identifies the version exactly. If you have a complete git clone of the >> source tree, the following command will put the tree at that version: > > OK, I learned something there. So what is the significance of the > N-31774 part of the version? > Commit number. As the hash is effectively a random tag, the commit number lets you know whether you are using an earlier or later version. The one I use is version N-32611-gd55b06b so I know that is newer than the N-31774-g6c4e9ca version. From andrey.krieger.utkin at gmail.com Fri Mar 2 21:25:26 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Fri, 2 Mar 2012 22:25:26 +0200 Subject: [FFmpeg-user] live transcoding flv files In-Reply-To: References: Message-ID: 2012/3/1 Steve Roskowski : > I have an http server which generates FLV format video "on the fly", which > allows flash to play them back in realtime - live streaming. > > I would like to use ffmpeg to transcode them, specifically to lower bit > rates for mobile clients. > > I cannot figure out how to get ffmpeg to do the transcoding in realtime. > ?FFMPEG seems to download the entire source file before beginning to > generate output. ?VLC shows the exact same behavior - video does not start > playing till the entire flv file is downloaded... which on a live stream is > forever. > > is this something I am missing in the options/configuration, or is this > something specific to the flv demuxer? > > typical ffmpeg command > > ffmpeg -i http://myserver/live_stream1/flv -b 64k -vcodec libx264 foo.flv > > works fine, but only starts rendering the output file when the live stream > is closed. ffmpeg utility does not render anything at all. It just transcodes. And it will process a file or a stream as it reads it (on the fly), exactly as you need. So you must have confused something, or missed to tell to us sth, like how do you playback it. -- Andrey Utkin From andrey.krieger.utkin at gmail.com Fri Mar 2 21:30:08 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Fri, 2 Mar 2012 22:30:08 +0200 Subject: [FFmpeg-user] Getting a point in time as known by ffmpeg? In-Reply-To: <20120302193141.GC6507@SDF.ORG> References: <20120302193141.GC6507@SDF.ORG> Message-ID: 2012/3/2 Gary Taylor : > When I watch a video (in xine or gxine for example)I see a > value displayed which is how long the video has been > playing. ?If I find a scene where I want to process the > video starting at 00:00:40.000 as displayed in xine, in > ffmpeg I have to experiment to find that same spot. It's not > at 00:00:40.000, but instead 00:00:63.000. > > It's like the notion of time in ffmpeg is different than > what is displayed by the players. ?Is this user error, > should I be viewing them in something else, or is there a > way to use ffplay to see display the time? > > fmpeg version N-37798-gcd1c12b Copyright (c) 2000-2012 the > FFmpeg developers > ?built on Feb 11 2012 23:02:52 with gcc 4.4.3 There's such thing as "start time". Timestamps may go not from zero. ff{play,mpeg,probe} shows it. Most probably, you should just shift desired time point by start time value. -- Andrey Utkin From cehoyos at ag.or.at Fri Mar 2 21:43:43 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 2 Mar 2012 20:43:43 +0000 (UTC) Subject: [FFmpeg-user] bgra to yuv References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: booglyboo gmail.com> writes: > I've been trying to convert a 1920x1080 BGRA 30 fps video (no sound) which is > in the avi container into a yuv file for the purpose of using dirac for > compression. [...] > but the framerate messes up and the file size changes so dramatically that I > don't believe the conversion was lossless. I don't think you can (easily) convert lossless from bgra to yuv, at least not to yuv420p. Please try to explain what you want to do (FFmpeg supports dirac encoding via external library, so while I don't think you will get satisfying results with dirac, there is no need to go through an uncompressed file), then provide the command line you used together with complete, uncut console output and explain what went wrong. Carl Eugen From cehoyos at ag.or.at Fri Mar 2 21:45:28 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 2 Mar 2012 20:45:28 +0000 (UTC) Subject: [FFmpeg-user] [ffplay] after changing to another audio stream, audio will become out of sync with video References: Message-ID: Lifeng Ren gmail.com> writes: > I also find this can be avoided by patching ffplay.c as such. Please send patches to ffmpeg-devel Carl Eugen From cehoyos at ag.or.at Fri Mar 2 21:53:37 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 2 Mar 2012 20:53:37 +0000 (UTC) Subject: [FFmpeg-user] converting 1080 to 720 mpeg2 References: <4F4C98CD.1090102@googlemail.com> <4F4C9C17.8020008@googlemail.com> <4F4C9F47.80604@googlemail.com> Message-ID: Andy Sheen googlemail.com> writes: > The only problem was the new ffmpeg introduced a 100ms delay > in the audio but you can work around that too ;). I don't think this is a known problem for any developer (and please understand that unknown problems usually get not fixed while known regressions should be fixed fast). Carl Eugen From cehoyos at ag.or.at Fri Mar 2 21:47:41 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 2 Mar 2012 20:47:41 +0000 (UTC) Subject: [FFmpeg-user] Where to find the version ffmpeg version N-31774-g6c4e9ca References: Message-ID: X. Long gmail.com> writes: > I have a video decoded by someone else by the above version with some > information generated after processing the image. Now I also decoded > the video and processed the image and another type of information > generated. If you believe you found a regression, please report it either here or on trac. And please remember that (all) old versions of FFmpeg are known to contain bugs, some of them security relevant! Carl Eugen From cehoyos at ag.or.at Fri Mar 2 22:03:15 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 2 Mar 2012 21:03:15 +0000 (UTC) Subject: [FFmpeg-user] AVI/AC3 -> VOB/LPCM References: <4F4E5D37.5000802@sokolov.me> <4F4F3D93.6020202@sokolov.me> <4F50AAF9.9010402@sokolov.me> Message-ID: Pavel Sokolov sokolov.me> writes: > Is somebody know how to solve this issue? If you need help, please always post your command line together with complete, uncut console output and explain what is wrong with the output file. Carl Eugen From sheen.andy at googlemail.com Fri Mar 2 22:08:51 2012 From: sheen.andy at googlemail.com (Andy Sheen) Date: Fri, 02 Mar 2012 21:08:51 +0000 Subject: [FFmpeg-user] converting 1080 to 720 mpeg2 In-Reply-To: References: <4F4C98CD.1090102@googlemail.com> <4F4C9C17.8020008@googlemail.com> <4F4C9F47.80604@googlemail.com> Message-ID: <4F5136E3.3030907@googlemail.com> Carl Eugen Hoyos wrote on Fri 02 Mar at 20:53 UK time > Andy Sheen googlemail.com> writes: > >> The only problem was the new ffmpeg introduced a 100ms delay >> in the audio but you can work around that too ;). > > I don't think this is a known problem for any developer > (and please understand that unknown problems usually get not fixed > while known regressions should be fixed fast). > I do a lot of reencoding from UK broadcast .ts files to .mkv files. I do a 2 pass reencode of video to a lower bitrate .x264, mux out the audio and then remux with mkvmerge. I currently use mkvmerge to mux to .mkv. The version I use is: mkvmerge v4.7.0 ('Just Like You Imagined') built on Apr 21 2011 01:13:14 For me, simply upgrading from: FFmpeg version SVN-r26400, Copyright (c) 2000-2011 the FFmpeg developers built on Jan 18 2011 04:07:05 with gcc 4.4.2 to: ffmpeg version N-32611-gd55b06b, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 15 2011 00:26:45 with gcc 4.6.1 caused me to have to change my mkvmerge command line from: mkvmerge -o file.mkv file.h264 --language 0:eng file.ac3 to mkvmerge -o file.mkv file.h264 --language 0:eng --sync 0:100 file.ac3 It may well be some sort of bug in mkvmerge, but the only thing changed was ffmpeg and this applies across all channels and all re-encodes... > Carl Eugen > Andy From funkyirish at gmail.com Fri Mar 2 23:07:16 2012 From: funkyirish at gmail.com (Josh long) Date: Fri, 2 Mar 2012 16:07:16 -0600 Subject: [FFmpeg-user] bgra to yuv In-Reply-To: References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: I've been given raw bgra files from a client in a school project and want to compress them with dirac, so that I can later parallelize the code. On Fri, Mar 2, 2012 at 2:43 PM, Carl Eugen Hoyos wrote: > booglyboo gmail.com> writes: > > > I've been trying to convert a 1920x1080 BGRA 30 fps video (no sound) > which is > > in the avi container into a yuv file for the purpose of using dirac for > > compression. > > [...] > > but the framerate messes up and the file size changes so dramatically > that I > > don't believe the conversion was lossless. > > I don't think you can (easily) convert lossless from bgra to yuv, > at least not to yuv420p. > > Please try to explain what you want to do (FFmpeg supports dirac encoding > via external library, so while I don't think you will get satisfying > results with dirac, there is no need to go through an uncompressed > file), then provide the command line you used together with complete, > uncut console output and explain what went wrong. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From squeaky at sdf.org Sat Mar 3 01:47:53 2012 From: squeaky at sdf.org (Gary Taylor) Date: Fri, 2 Mar 2012 16:47:53 -0800 Subject: [FFmpeg-user] Getting a point in time as known by ffmpeg? In-Reply-To: <20120302104044.4e9c8666@lrcd.com> References: <20120302193141.GC6507@SDF.ORG> <20120302104044.4e9c8666@lrcd.com> Message-ID: <20120303004753.GA19541@SDF.ORG> On Fri, Mar 02, 2012 at 10:40:44AM -0900, Lou wrote: > > ffplay shows the current time in the last line: > > $ ffplay input.mkv > ... > 3.05 A-V: 0.002 fd= 1 aq= 320KB vq= 5357KB sq= 0B f=0/0 f=0/0 > > In this example 3.05 seconds has elapsed. > > Also note that the behavior of the -ss option can change depending if it > is used as an input or output option. Thanks for the responses here. I can use $ ffplay -stats movie.mpg and it's just what I need. Gary From de.techno at gmail.com Sat Mar 3 07:19:21 2012 From: de.techno at gmail.com (dE .) Date: Sat, 03 Mar 2012 11:49:21 +0530 Subject: [FFmpeg-user] Problem transcoding from MP4 (H264 - AAC) to OGG (DIRAC - FLAC) In-Reply-To: References: <4F46A788.9070406@gmail.com> Message-ID: <4F51B7E9.5080005@gmail.com> On 03/02/12 22:45, aple ex wrote: > > >> Date: Thu, 23 Feb 2012 17:54:32 -0300 >> From: agprus at gmail.com >> To: ffmpeg-user at ffmpeg.org >> Subject: [FFmpeg-user] Problem transcoding from MP4 (H264 - AAC) to OGG (DIRAC - FLAC) >> >> Hello, >> >> I am trying to do a lossy-to-lossless transcoding from an MP4 file (with >> one H264 video stream and one AAC audio stream) to an OGG file (with a >> DIRAC video stream and a FLAC audio stream). The command line I am using >> is this: >> >> ffmpeg -i input.mp4 -vcodec libschroedinger -q:v 0 -acodec flac output.ogg >> >> and I am getting the following errors (complete output is at the end): >> >> [ogg @ 0000000001F29570] Unsupported codec id in stream 0 >> Could not write header for output file #0 (incorrect codec parameters ?) >> >> Does anybody know an appropriate way to do this kind of transcoding? >> >> Thanks in advance, >> Alexis >> >> ffmpeg version N-38148-gb6ff81d Copyright (c) 2000-2012 the FFmpeg >> developers >> built on Feb 23 2012 12:30:44 with gcc 4.6.2 >> configuration: --enable-gpl --enable-version3 --disable-w32threads >> --enable-runtime-cpudetect --enable-avisynth --enable-bzlib >> --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb >> --enable-libfreetype --enable-libgsm --enable-libmp3lame >> --enable-libopenjpeg --enable-librtmp --enable-libschroedinger >> --enable-libspeex --enable-libtheora --enable-libvo-aacenc >> --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx >> --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib >> libavutil 51. 40.100 / 51. 40.100 >> libavcodec 54. 4.100 / 54. 4.100 >> libavformat 54. 1.100 / 54. 1.100 >> libavdevice 53. 4.100 / 53. 4.100 >> libavfilter 2. 62.101 / 2. 62.101 >> libswscale 2. 1.100 / 2. 1.100 >> libswresample 0. 7.100 / 0. 7.100 >> libpostproc 52. 0.100 / 52. 0.100 >> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'h264aac1.mp4': >> Metadata: >> major_brand : isom >> minor_version : 512 >> compatible_brands: isomiso2avc1mp41 >> creation_time : 2010-02-16 19:06:52 >> encoder : Lavf52.36.0 >> Duration: 00:03:40.03, start: 0.000000, bitrate: 2571 kb/s >> Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / >> 0x31637661), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 2537 kb/s, 30 fps, 30 >> tbr, 30 tbn, 60 tbc >> Metadata: >> creation_time : 2010-02-16 19:06:52 >> handler_name : VideoHandler >> Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 11025 Hz, stereo, >> s16, 31 kb/s >> Metadata: >> creation_time : 2010-02-16 19:06:52 >> handler_name : >> [buffer @ 0000000001F29F30] w:640 h:480 pixfmt:yuv420p tb:1/1000000 >> sar:1/1 sws_param: >> [ogg @ 0000000001F29570] Unsupported codec id in stream 0 >> Output #0, ogg, to 'h264aac.ogg': >> Metadata: >> major_brand : isom >> minor_version : 512 >> compatible_brands: isomiso2avc1mp41 >> creation_time : 2010-02-16 19:06:52 >> encoder : Lavf54.1.100 >> Stream #0:0(und): Video: dirac, yuv420p, 640x480 [SAR 1:1 DAR 4:3], >> q=2-31, 200 kb/s, 30 tbn, 30 tbc >> Metadata: >> creation_time : 2010-02-16 19:06:52 >> handler_name : VideoHandler >> Stream #0:1(und): Audio: flac, 11025 Hz, stereo, s16, 128 kb/s >> Metadata: >> creation_time : 2010-02-16 19:06:52 >> handler_name : >> Stream mapping: >> Stream #0:0 -> #0:0 (h264 -> libschroedinger) >> Stream #0:1 -> #0:1 (aac -> flac) >> Could not write header for output file #0 (incorrect codec parameters ?) > Hi according to their website dirac codec is supposed to be contained by ogg, see here : > http://diracvideo.org/schrodinger-faq/ (specially Question 6) > I have tested myself and I obtain the same result. I can provide my console output if necessary but dirac can be encoded in mpegts container. > Regards > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Yes, use libtheora instead. From what I've experimented, only theora can be stored in ogg. And actually, since it contains a video, I think it should be ogv. From de.techno at gmail.com Sat Mar 3 07:22:38 2012 From: de.techno at gmail.com (dE .) Date: Sat, 03 Mar 2012 11:52:38 +0530 Subject: [FFmpeg-user] live transcoding flv files In-Reply-To: References: Message-ID: <4F51B8AE.80905@gmail.com> On 03/02/12 01:52, Steve Roskowski wrote: > I have an http server which generates FLV format video "on the fly", which > allows flash to play them back in realtime - live streaming. > > I would like to use ffmpeg to transcode them, specifically to lower bit > rates for mobile clients. > > I cannot figure out how to get ffmpeg to do the transcoding in realtime. > FFMPEG seems to download the entire source file before beginning to > generate output. VLC shows the exact same behavior - video does not start > playing till the entire flv file is downloaded... which on a live stream is > forever. > > is this something I am missing in the options/configuration, or is this > something specific to the flv demuxer? > > typical ffmpeg command > > ffmpeg -i http://myserver/live_stream1/flv -b 64k -vcodec libx264 foo.flv > > works fine, but only starts rendering the output file when the live stream > is closed. > If you want to stream the output, it does have to be to a file. Use a pipe... maybe. From cehoyos at ag.or.at Sat Mar 3 08:11:31 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 3 Mar 2012 07:11:31 +0000 (UTC) Subject: [FFmpeg-user] bgra to yuv References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: Josh long gmail.com> writes: > I've been given raw bgra files from a client in a school project > and want to compress them with dirac, so that I can later > parallelize the code. ffmpeg -i input -vcodec libschroedinger out.avi But please post the command line you tried together with complete, uncut console output and explain what goes wrong. Please do not top-post here, it is considered rude, Carl Eugen From yogesh.bit2006 at gmail.com Sat Mar 3 08:19:50 2012 From: yogesh.bit2006 at gmail.com (Yogesh Tyagi) Date: Sat, 3 Mar 2012 12:49:50 +0530 Subject: [FFmpeg-user] how to enable hardware acceleration in ffmpeg In-Reply-To: References: Message-ID: Hi, Yes I want to use vaapi to display media using hwaccel.Is there any option by which I can switch b/w software decoding and hwaccel at run time? Thanks and Regards, Yogesh On Fri, Mar 2, 2012 at 7:59 PM, Tom Evans wrote: > On Fri, Mar 2, 2012 at 1:42 PM, Yogesh Tyagi > wrote: > > Hi, > > > > Is there any command line option in ffmpeg to enable hardware > acceleration > > for h264 decoding.I have configured and compiled ffmpeg with" > > --enable-vaapi --enable-hwaccel=h264_vaapi" options but ffmpeg is not > > invoking decode_slice function of vaapi_h264.c. > > > > As I understand it, vdpau and vaapi are useful for decoding when you > want to display that decoded content. It's not a usable API, eg for > decoding video as a step for subsequently re-encoding it (if that > makes sense). > > IE, you can use vdpau/vaapi to display media using hwaccel, but you > can't use it to accelerate decoding whilst encoding. > > Cheers > > Tom > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Thanks and Regards, Yogesh Tyagi Mobile No: 09911814030 From ds at viddiga.com Sat Mar 3 11:30:14 2012 From: ds at viddiga.com (David Scravaglieri) Date: Sat, 3 Mar 2012 11:30:14 +0100 Subject: [FFmpeg-user] MPEGTS buffers Message-ID: <4D727825-B080-41F7-A901-19CDAE1E916E@viddiga.com> Hi, Does someone could give me indications in order to decode MPEGTS live streaming with ffmpeg libraries (avcodec, avformat...). I know how to do with a file but I don't know how to do this with live stream where format and data are hidden by the MPEGTS layer. David. From officemab at gmail.com Sun Mar 4 01:12:10 2012 From: officemab at gmail.com (funtastic) Date: Sun, 4 Mar 2012 01:12:10 +0100 Subject: [FFmpeg-user] video delay after audio fix Message-ID: Hi, if I open a specific file with Totem video player (ver. 3.0.1) on Ubuntu the audio is very choppy (video fine). With VLC player audio and video is good but the video stream starts several seconds after the audio... (with Totem, audio and video starts at the same time) I tried to fix the file by applying the following command: ffmpeg -i share.mp4 -sameq output.mp4 After that, Totem plays the audio of the file correctly but now the video is delayed and starts several seconds after the audio (same situation as with VLC before). Audio and video are in sync (when both have started) so adding a simple offset will not improve the situation... Does somebody know the reason for that and how to fix the file? Regards! From ffmpeg-user at dogtoe.com Sun Mar 4 02:14:37 2012 From: ffmpeg-user at dogtoe.com (Brian Johnson) Date: Sat, 3 Mar 2012 17:14:37 -0800 Subject: [FFmpeg-user] Problem "live" transcoding via piped input / Ceton InfiniTV 4 In-Reply-To: <4F5090BF.8000405@gmail.com> References: <4F5090BF.8000405@gmail.com> Message-ID: -preset ultrafast seems to help out tremendously, but I have a quad core CPU and even with -threads 4, 8, 10, etc I can't get ffmpeg to utilize the entire CPU (seems to max out at about 190%). - Brian On Fri, Mar 2, 2012 at 1:19 AM, dE . wrote: > On 03/02/12 00:37, Brian Johnson wrote: >> >> The command I am using: ffmpeg -i /dev/ctn91xx_mpeg0_0 -async 2 >> -acodec libmp3lame -ac 2 -vcodec libx264 -s 1280x720 -aspect 16:9 -f >> mpegts file.ts >> >> The file /dev/ctn91xx_mpeg0_0 is actually the device (Ceton InfiniTV >> 4) itself where it returns a mpeg transport stream, basically a named >> pipe containing the raw audio and video. >> >> When I run the command below, I get tons of errors which repeats forever: >> >> ES packet size mismatch.0 size= ? ? 797kB time=00:00:04.23 >> bitrate=1541.1kbits/s dup=25 drop=0 >> [ac3 @ 0x8393340] frame CRC mismatch >> [mpeg2video @ 0x834e680] ac-tex damaged at 0 28 >> [mpeg2video @ 0x834e680] ac-tex damaged at 2 1 >> [mpeg2video @ 0x834e680] ac-tex damaged at 0 27 >> [mpeg2video @ 0x834e680] ac-tex damaged at 0 29 >> [mpeg2video @ 0x834e680] Warning MVs not available >> [mpeg2video @ 0x834e680] concealing 1276 DC, 1276 AC, 1276 MV errors >> [mpegts @ 0x8287aa0] PES packet size mismatch >> frame sync error 34 q=29.0 size= ? ?1202kB time=00:00:05.76 >> bitrate=1709.0kbits/s dup=145 drop=0 >> Error while decoding stream #0:27 >> ac-tex damaged at 4 2929.0 size= ? ?1383kB time=00:00:09.17 >> bitrate=1235.1kbits/s dup=145 drop=0 >> [mpeg2video @ 0x834e680] mb incr damaged >> [mpeg2video @ 0x834e680] ac-tex damaged at 35 29 >> [mpeg2video @ 0x834e680] concealing 880 DC, 880 AC, 880 MV errors >> [mpegts @ 0x8287aa0] PES packet size mismatch >> [ac3 @ 0x8393340] frame CRC mismatch >> [mpegts @ 0x8287aa0] PES packet size mismatch >> ? ? Last message repeated 1 times >> [ac3 @ 0x8393340] frame sync error >> Error while decoding stream #0:27 >> [mpeg2video @ 0x834e680] invalid cbp at 0 12 >> [mpeg2video @ 0x834e680] mb incr damaged >> [mpeg2video @ 0x834e680] invalid mb type in B Frame at 3 24 >> [mpeg2video @ 0x834e680] 00 motion_type at 1 25 >> [mpeg2video @ 0x834e680] 00 motion_type at 0 26 >> [mpeg2video @ 0x834e680] invalid mb type in B Frame at 16 27 >> >> What is very interesting is that lower quality video transcodes (i.e. >> 320x240) has the same errors as above but only lasts about 1-2 >> seconds, and then output looks normal >> >> frame= 1235 fps= 35 q=29.0 size= ? ?1985kB time=00:00:39.53 bitrate= >> 411.2kbits/s dup=131 drop=0 >> >> I believe this to be related to >> >> and in my own research, I found that this may have been a regression >> from an older version, but again that is pure speculation on my part. >> >> Any ideas? :) >> >> - Brian >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > Are you using the latest? This may be cause your PC is not as fast. Try > using -threads and -preset ultrafast. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From worthspending at gmail.com Sun Mar 4 03:42:29 2012 From: worthspending at gmail.com (worthspending) Date: Sat, 3 Mar 2012 20:42:29 -0600 Subject: [FFmpeg-user] audio / video separates files Message-ID: Is there any way to use ffmpeg to output the audio and video to separate files during a screen capture? I know I can split the audio and video afterwards, however, why take the extra step if I don't need to. From lou at lrcd.com Sun Mar 4 05:38:17 2012 From: lou at lrcd.com (Lou) Date: Sat, 3 Mar 2012 19:38:17 -0900 Subject: [FFmpeg-user] audio / video separates files In-Reply-To: References: Message-ID: <20120303193817.5d712fec@lrcd.com> On Sat, 3 Mar 2012 20:42:29 -0600 worthspending wrote: > Is there any way to use ffmpeg to output the audio and video to separate > files during a screen capture? I know I can split the audio and video > afterwards, however, why take the extra step if I don't need to. Yes. Verbose example: ffmpeg -f alsa -i hw:0,0 -f x11grab -r 30 -s 1024x768 -i :0.0 -c:v \ libx264 -preset ultrafast -crf 0 -an -y output.mkv -c:a pcm_s16le \ output.wav From cehoyos at ag.or.at Sun Mar 4 13:29:45 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 4 Mar 2012 12:29:45 +0000 (UTC) Subject: [FFmpeg-user] video delay after audio fix References: Message-ID: funtastic gmail.com> writes: > I tried to fix the file by applying the following command: > ffmpeg -i share.mp4 -sameq output.mp4 Complete, uncut console output missing. Does the original sample play correct with MPlayer or ffplay? Carl Eugen From officemab at gmail.com Sun Mar 4 15:23:44 2012 From: officemab at gmail.com (funtastic) Date: Sun, 4 Mar 2012 15:23:44 +0100 Subject: [FFmpeg-user] video delay after audio fix In-Reply-To: References: Message-ID: Am 4. M?rz 2012 13:29 schrieb Carl Eugen Hoyos : > funtastic gmail.com> writes: > > > I tried to fix the file by applying the following command: > > ffmpeg -i share.mp4 -sameq output.mp4 > > Complete, uncut console output missing. > Does the original sample play correct with MPlayer or ffplay? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Sorry for the missing details... (will provide them now in greater detail) Please see the output of "ffmpeg -i share.mp4 -sameq output.mp4" here: http://pastebin.com/n40GJG3s ffplay: Playing the video with ffplay shows correct audio and video BUT with a kind of slow motion for the first several seconds.... (audio and video are in sync after that) MPlayer: Plays the video but here, audio and video are NOT in sync.... I obtained the share.mp4 video file with the following command: ffmpeg -i rtsp://url.mp4 -vcodec copy -acodec copy -ss 0.00001 share.mp4 Thats the URL (rtsp://url.mp4) I use: http://pastebin.com/JWdXKiZc Note: I need the -vcodec copy and -acodec copy options since my system is too weak to perform heavy calculations during download... (so please consider this as necessary) One interessing aspect: When I execute the "ffmpeg -i share.mp4 -sameq output.mp4" command, ffmpeg responds with: "multiple edit list entries, a/v desync might occur, patch welcome" Could this be a part of the problem and how to resolve the issue??? It would be great if it would be possible to download the correct stream with the -vcodec copy -acodec copy options... Excited to hear your thoughts... Regards Martin From mediastream at gmail.com Sun Mar 4 18:05:17 2012 From: mediastream at gmail.com (Dennis) Date: Sun, 4 Mar 2012 12:05:17 -0500 Subject: [FFmpeg-user] Problem "live" transcoding via piped input / Ceton InfiniTV 4 In-Reply-To: References: <4F5090BF.8000405@gmail.com> Message-ID: Try putting -threads 4 in front of -i, it will implement multithreading for decoding. On 3/3/12, Brian Johnson wrote: > -preset ultrafast seems to help out tremendously, but I have a quad > core CPU and even with -threads 4, 8, 10, etc I can't get ffmpeg to > utilize the entire CPU (seems to max out at about 190%). > > - Brian > > On Fri, Mar 2, 2012 at 1:19 AM, dE . wrote: >> On 03/02/12 00:37, Brian Johnson wrote: >>> >>> The command I am using: ffmpeg -i /dev/ctn91xx_mpeg0_0 -async 2 >>> -acodec libmp3lame -ac 2 -vcodec libx264 -s 1280x720 -aspect 16:9 -f >>> mpegts file.ts >>> >>> The file /dev/ctn91xx_mpeg0_0 is actually the device (Ceton InfiniTV >>> 4) itself where it returns a mpeg transport stream, basically a named >>> pipe containing the raw audio and video. >>> >>> When I run the command below, I get tons of errors which repeats forever: >>> >>> ES packet size mismatch.0 size= ? ? 797kB time=00:00:04.23 >>> bitrate=1541.1kbits/s dup=25 drop=0 >>> [ac3 @ 0x8393340] frame CRC mismatch >>> [mpeg2video @ 0x834e680] ac-tex damaged at 0 28 >>> [mpeg2video @ 0x834e680] ac-tex damaged at 2 1 >>> [mpeg2video @ 0x834e680] ac-tex damaged at 0 27 >>> [mpeg2video @ 0x834e680] ac-tex damaged at 0 29 >>> [mpeg2video @ 0x834e680] Warning MVs not available >>> [mpeg2video @ 0x834e680] concealing 1276 DC, 1276 AC, 1276 MV errors >>> [mpegts @ 0x8287aa0] PES packet size mismatch >>> frame sync error 34 q=29.0 size= ? ?1202kB time=00:00:05.76 >>> bitrate=1709.0kbits/s dup=145 drop=0 >>> Error while decoding stream #0:27 >>> ac-tex damaged at 4 2929.0 size= ? ?1383kB time=00:00:09.17 >>> bitrate=1235.1kbits/s dup=145 drop=0 >>> [mpeg2video @ 0x834e680] mb incr damaged >>> [mpeg2video @ 0x834e680] ac-tex damaged at 35 29 >>> [mpeg2video @ 0x834e680] concealing 880 DC, 880 AC, 880 MV errors >>> [mpegts @ 0x8287aa0] PES packet size mismatch >>> [ac3 @ 0x8393340] frame CRC mismatch >>> [mpegts @ 0x8287aa0] PES packet size mismatch >>> ? ? Last message repeated 1 times >>> [ac3 @ 0x8393340] frame sync error >>> Error while decoding stream #0:27 >>> [mpeg2video @ 0x834e680] invalid cbp at 0 12 >>> [mpeg2video @ 0x834e680] mb incr damaged >>> [mpeg2video @ 0x834e680] invalid mb type in B Frame at 3 24 >>> [mpeg2video @ 0x834e680] 00 motion_type at 1 25 >>> [mpeg2video @ 0x834e680] 00 motion_type at 0 26 >>> [mpeg2video @ 0x834e680] invalid mb type in B Frame at 16 27 >>> >>> What is very interesting is that lower quality video transcodes (i.e. >>> 320x240) has the same errors as above but only lasts about 1-2 >>> seconds, and then output looks normal >>> >>> frame= 1235 fps= 35 q=29.0 size= ? ?1985kB time=00:00:39.53 bitrate= >>> 411.2kbits/s dup=131 drop=0 >>> >>> I believe this to be related to >>> >>> and in my own research, I found that this may have been a regression >>> from an older version, but again that is pure speculation on my part. >>> >>> Any ideas? :) >>> >>> - Brian >>> _______________________________________________ >>> ffmpeg-user mailing list >>> ffmpeg-user at ffmpeg.org >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> >> >> Are you using the latest? This may be cause your PC is not as fast. Try >> using -threads and -preset ultrafast. >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From voncampeg at netscape.net Sun Mar 4 18:54:52 2012 From: voncampeg at netscape.net (Gord von Campe) Date: Sun, 04 Mar 2012 18:54:52 +0100 Subject: [FFmpeg-user] Invalid audio in resulting MPEG-TS Message-ID: <4F53AC6C.10206@netscape.net> Hello, When I try to mux a video (MPEG-2) and audio (WAV, pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s) file into a MPEG-TS container, the resulting file has an invalid audio stream (the FFmpeg command used is "ffmpeg -fflags +genpts -i Video.m2v -i Audio.wav -c:v copy -c:a copy -f mpegts MuxedAV.ts" and completes without error). After muxing, the audio stream in the new file is not recognized by FFmpeg (see below) or MediaInfo, and the MPEG-TS file plays without sound in MPlayer or Media Player Classic. Muxing the same input video and audio files using tsMuxeR results in a valid file with intact audio. How can I make this work with FFmpeg? I am using the following ffmpeg version on Windows: ffmpeg version N-38292-ga4c22e3 Copyright (c) 2000-2012 the FFmpeg developers built on Feb 27 2012 14:50:39 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 41.100 / 51. 41.100 libavcodec 54. 4.100 / 54. 4.100 libavformat 54. 1.100 / 54. 1.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 62.101 / 2. 62.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 Output of "ffmpeg -i MuxedAV.ts": [...] [mpegts @ 0217A7C0] probed stream 1 failed [mpegts @ 0217A7C0] max_analyze_duration 5000000 reached at 5005000 [mpegts @ 0217A7C0] decoding for stream 1 failed [mpegts @ 0217A7C0] Could not find codec parameters (Unknown: none ([6][0][0][0] / 0x0006)) Input #0, mpegts, from 'MuxedAV.ts': Duration: 00:00:59.99, start: 1.400000, bitrate: 6400 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p, 720x480 [SAR 32:27 DAR 16:9], 7500 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0:1[0x101]: Unknown: none ([6][0][0][0] / 0x0006) Thank you in advance for your help! Gord From cehoyos at ag.or.at Sun Mar 4 19:20:15 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 4 Mar 2012 18:20:15 +0000 (UTC) Subject: [FFmpeg-user] Invalid audio in resulting MPEG-TS References: <4F53AC6C.10206@netscape.net> Message-ID: Gord von Campe netscape.net> writes: > When I try to mux a video (MPEG-2) and audio (WAV, pcm_s16le > ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s) file into > a MPEG-TS container, Do you have any indication that mpeg-ts does support pcm_s16le? Do you have a sample file? Or do you want to encode uncompressed Bluray audio? Carl Eugen From cehoyos at ag.or.at Sun Mar 4 19:22:16 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 4 Mar 2012 18:22:16 +0000 (UTC) Subject: [FFmpeg-user] video delay after audio fix References: Message-ID: funtastic gmail.com> writes: > Sorry for the missing details... (will provide them now in greater detail) > Please see the output of "ffmpeg -i share.mp4 -sameq output.mp4" > here: http://pastebin.com/n40GJG3s Please provide complete, uncut console output (that is everything that ffmpeg prints on the console) together with the command line you used here on this mailing lists, external resources may disappear. Carl Eugen From voncampeg at netscape.net Sun Mar 4 19:42:03 2012 From: voncampeg at netscape.net (Gord von Campe) Date: Sun, 04 Mar 2012 19:42:03 +0100 Subject: [FFmpeg-user] Invalid audio in resulting MPEG-TS In-Reply-To: References: <4F53AC6C.10206@netscape.net> Message-ID: <4F53B77B.2080907@netscape.net> On 04-03-2012 19:20, Carl Eugen Hoyos wrote: > Gord von Campe netscape.net> writes: > >> When I try to mux a video (MPEG-2) and audio (WAV, pcm_s16le >> ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s) file into >> a MPEG-TS container, > > Do you have any indication that mpeg-ts does support pcm_s16le? > Do you have a sample file? > > Or do you want to encode uncompressed Bluray audio? > > Carl Eugen The original files are from an opera: besides the mpeg video, there are ac3 (5.1) and wav (stereo) audio files. I try to produce a MPEG-TS file retaining the original ac3 and wav audio for local streaming (Popcorn Hour). I don't really know if mpeg-ts supports pcm_s16le, but since tsMuxeR produces a playable file (on MPlayer or Media Player Classic at least) and that file is recognized correctly by FFmpeg and MediaInfo, I assumed that this is the case... The files are quite large, but I could produce a short clip if needed: where should I send these files? Gord From voncampeg at netscape.net Sun Mar 4 21:06:26 2012 From: voncampeg at netscape.net (Gord von Campe) Date: Sun, 04 Mar 2012 21:06:26 +0100 Subject: [FFmpeg-user] Invalid audio in resulting MPEG-TS In-Reply-To: References: <4F53AC6C.10206@netscape.net> Message-ID: <4F53CB42.50601@netscape.net> On 04-03-2012 19:20, Carl Eugen Hoyos wrote: > Gord von Campe netscape.net> writes: > >> When I try to mux a video (MPEG-2) and audio (WAV, pcm_s16le >> ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s) file into >> a MPEG-TS container, > > Do you have any indication that mpeg-ts does support pcm_s16le? > Do you have a sample file? > > Or do you want to encode uncompressed Bluray audio? > > Carl Eugen OK, apparently you are right that uncompressed (WAV, pcm_s16le) might not be supported in mpeg-ts. Although tsMuxeR creates such a file, and the individual streams are recognized by FFmpeg and MediaInfo, the WAV/PCM stream cannot be played back (only video, no audio). The confusion here comes from the fact that in my first attempt I muxed both AC3 and WAV files into the mpeg-ts stream and it seemed that it was possible to switch from one stream to the other, when in fact the player was only cycling the AC3 stream: sorry, my bad! Gord From officemab at gmail.com Sun Mar 4 21:18:05 2012 From: officemab at gmail.com (funtastic) Date: Sun, 4 Mar 2012 21:18:05 +0100 Subject: [FFmpeg-user] video delay after audio fix In-Reply-To: References: Message-ID: Am 4. M?rz 2012 19:22 schrieb Carl Eugen Hoyos : > funtastic gmail.com> writes: > > > Sorry for the missing details... (will provide them now in greater > detail) > > Please see the output of "ffmpeg -i share.mp4 -sameq output.mp4" > > here: http://pastebin.com/n40GJG3s > > Please provide complete, uncut console output (that is everything that > ffmpeg prints on the console) together with the command line you used here > on this mailing lists, external resources may disappear. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Sorry for that but: I don't want to distribute private material or stuff that is not intended for public distribution. And: I don't want to paste stuff that could be under copyright by somebody else.... Nevertheless, the same effects I described can be seen with the following command: ffmpeg -i rtsp://184.72.239.149/vod/mp4:BigBuckBunny_175k.mov -vcodec copy -acodec copy -ss 0.0001 -t 30 -y share.mp4 (BigBuckBunny is a free to distribute video from the Blender Foundation | www.blender.org under the Creative Commons Attribution 3.0licencse) (Big Buck Bunny video ? Blender Foundation | www.bigbuckbunny.org) Complete console output: ffmpeg -i rtsp://184.72.239.149/vod/mp4:BigBuckBunny_175k.mov -vcodec copy -acodec copy -ss 0.0001 -t 30 -y share.mp4 ffmpeg version 0.7.3-4:0.7.3-0ubuntu0.11.10.1, Copyright (c) 2000-2011 the Libav developers built on Jan 4 2012 16:21:50 with gcc 4.6.1 configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static WARNING: library configuration mismatch avutil configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avcodec configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avformat configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avdevice configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avfilter configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay swscale configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay postproc configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libavutil 51. 7. 0 / 51. 7. 0 libavcodec 53. 6. 0 / 53. 6. 0 libavformat 53. 3. 0 / 53. 3. 0 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 4. 0 / 2. 4. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 52. 0. 0 / 52. 0. 0 [rtsp @ 0x9d362a0] Estimating duration from bitrate, this may be inaccurate Seems stream 1 codec frame rate differs from container frame rate: 48.00 (48/1) -> 1000.00 (1000/1) Input #0, rtsp, from 'rtsp://184.72.239.149/vod/mp4:BigBuckBunny_175k.mov': Metadata: title : BigBuckBunny_175k.mov Duration: 00:09:56.45, start: 0.000000, bitrate: N/A Stream #0.0: Audio: aac, 48000 Hz, stereo, s16 Stream #0.1: Video: h264 (Constrained Baseline), yuv420p, 240x160, 24 fps, 1k tbr, 90k tbn, 48 tbc Output #0, mp4, to 'share.mp4': Metadata: title : BigBuckBunny_175k.mov encoder : Lavf53.3.0 Stream #0.0: Video: ![0][0][0] / 0x0021, yuv420p, 240x160, q=2-31, 24 tbn, 24 tbc Stream #0.1: Audio: aac, 48000 Hz, stereo Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Press ctrl-c to stop encoding frame= 718 fps= 25 q=-1.0 Lsize= 714kB time=30.02 bitrate= 195.0kbits/s video:341kB audio:358kB global headers:0kB muxing overhead 2.147997% ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ With Totem, the video shows the choppy audio as described... other players work better. After applying the following command, the video gets its video delay at the beginning (although this would be a tiny problem with this special example...) Other streams show some really bad video delay.... Please see the "multiple edit list entries, a/v desync might occur, patch welcome" message I talked about earlier. Output for: ffmpeg -i share.mp4 -sameq output.mp4 ffmpeg version 0.7.3-4:0.7.3-0ubuntu0.11.10.1, Copyright (c) 2000-2011 the Libav developers built on Jan 4 2012 16:21:50 with gcc 4.6.1 configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static WARNING: library configuration mismatch avutil configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avcodec configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avformat configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avdevice configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avfilter configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay swscale configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay postproc configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libavutil 51. 7. 0 / 51. 7. 0 libavcodec 53. 6. 0 / 53. 6. 0 libavformat 53. 3. 0 / 53. 3. 0 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 4. 0 / 2. 4. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 52. 0. 0 / 52. 0. 0 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8d192a0] multiple edit list entries, a/v desync might occur, patch welcome Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'share.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 1970-01-01 00:00:00 title : BigBuckBunny_175k.mov encoder : Lavf53.3.0 Duration: 00:00:30.04, start: 0.000000, bitrate: 194 kb/s Stream #0.0(und): Video: h264 (Constrained Baseline), yuv420p, 240x160, 93 kb/s, 23.90 fps, 24 tbr, 24 tbn, 48 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(und): Audio: aac, 48000 Hz, stereo, s16, 97 kb/s Metadata: creation_time : 1970-01-01 00:00:00 File 'output.mp4' already exists. Overwrite ? [y/N] y [buffer @ 0x8d21740] w:240 h:160 pixfmt:yuv420p Output #0, mp4, to 'output.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 1970-01-01 00:00:00 title : BigBuckBunny_175k.mov encoder : Lavf53.3.0 Stream #0.0(und): Video: mpeg4, yuv420p, 240x160, q=2-31, 200 kb/s, 24 tbn, 24 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(und): Audio: aac, 48000 Hz, stereo, s16, 64 kb/s Metadata: creation_time : 1970-01-01 00:00:00 Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press ctrl-c to stop encoding Multiple frames in a packet from stream 1 frame= 720 fps=219 q=0.0 Lsize= 1997kB time=29.99 bitrate= 545.3kbits/s dup=2 drop=0 video:1737kB audio:241kB global headers:0kB muxing overhead 0.913233% +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Thats all I can provide.... Regards, Martin From andrey.krieger.utkin at gmail.com Mon Mar 5 01:01:02 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Mon, 5 Mar 2012 02:01:02 +0200 Subject: [FFmpeg-user] video delay after audio fix In-Reply-To: References: Message-ID: Hi Martin. We're glad to be helpful for ffmpeg users. But from my point you have described many different issues, not closely related to each other. It makes hard to understand you and to help you. >From your first post, i got the only thing that there's no player that plays your video correctly (you mentioned Totem, VLC, FFplay and all they do not give required behaviour). I assume there's sth wrong with your file. You can fix the file by cutting the earlier-starting stream - audio, in your case. It will require you to split file into separate files - audio only and video only, then cutting beginning from audio file, then joining back. It will be simple ffmpeg commands. If you have difficulties with making up these commands, post here. -- Andrey Utkin From smarrocco at ringsidecreative.com Mon Mar 5 01:48:41 2012 From: smarrocco at ringsidecreative.com (Sam Marrocco) Date: Sun, 04 Mar 2012 19:48:41 -0500 Subject: [FFmpeg-user] Change startframe number of DPX sequence? Message-ID: <4F540D69.2050508@ringsidecreative.com> During ffmpeg conversion of Prores QT mov files to DPX the default condition is that frame numbering starts with 1 using and output name of MyFile_%04d.dpx. When using source qt files containing timecode it is desirable to have the timecode, translated into framenumbers, fixed in the output dpx files. Is there a flag or property that can be set to force the output image sequence to begin numbering at an integer other than 1? I can perform the timecode frame calculation myself but would like to avoid a "manual" renumbering of the output files after transcoding if possible. -- Sam Marrocco Chief Technical Officer 248-548-2500 Main 248-910-3344 Cell "Just because no one understands you doesn't make you an artist." RINGSIDE CREATIVE | INTEGRATED MEDIA STUDIO? http://www.ringsidecreative.com Find us on Facebook. Please consider the environment before printing this email. From jmcintyre at dfsoftware.com Mon Mar 5 04:02:22 2012 From: jmcintyre at dfsoftware.com (Jared McIntyre) Date: Sun, 4 Mar 2012 20:02:22 -0700 Subject: [FFmpeg-user] Framerate Question Message-ID: <468DBF07-6AF0-44D4-B97B-78F593A20493@dfsoftware.com> I'm re-encodeing broadcast video. The source was apparently 720 30fps, but is being broadcast 60fps where every frame appears twice. I thought I'd try to save some space by re-encoding and using 30fps to cut out every other frame. I used the following command: ffmpeg -i in.mpg -acodec libfaac -ab 160k -crf 23.1 -vcodec libx264 -tune animation -preset faster -profile baseline -f mp4 -r 29.97 out.mp4 I was a bit surprised to find out that if I used -r 29.97 I would get a file with half the frames, but a larger size than if I hadn't done so. I'm sure I'm misunderstanding something, but is this the expected behavior? Here is the log data in case about the encodeing: Seems stream 0 codec frame rate differs from container frame rate: 119.88 (120000/1001) -> 59.96 (90000/1501) Input #0, mpeg, from 'in.mpg': Duration: 00:02:30.91, start: 0.328367, bitrate: 11748 kb/s Stream #0.0[0x1e0]: Video: mpeg2video (Main), yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 38810 kb/s, 59.96 fps, 59.96 tbr, 90k tbn, 119.88 tbc Stream #0.1[0x80]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s [buffer @ 0x101a33900] w:1280 h:720 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param: [libx264 @ 0x10204a000] using SAR=1/1 [libx264 @ 0x10204a000] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.1 Cache64 [libx264 @ 0x10204a000] profile Constrained Baseline, level 3.1 [libx264 @ 0x10204a000] 264 - core 115 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=0 ref=4 deblock=1:1:1 analyse=0x1:0x111 me=hex subme=4 psy=1 psy_rd=0.40:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=3 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=20 rc=crf mbtree=1 crf=23.1 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:0.60 Output #0, mp4, to 'out.mp4': Metadata: encoder : Lavf52.110.0 Stream #0.0: Video: libx264, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 2997 tbn, 29.97 tbc Stream #0.1: Audio: libfaac, 48000 Hz, 5.1, s16, 160 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 From mediastream at gmail.com Mon Mar 5 05:06:32 2012 From: mediastream at gmail.com (Dennis) Date: Sun, 4 Mar 2012 23:06:32 -0500 Subject: [FFmpeg-user] Framerate Question In-Reply-To: <468DBF07-6AF0-44D4-B97B-78F593A20493@dfsoftware.com> References: <468DBF07-6AF0-44D4-B97B-78F593A20493@dfsoftware.com> Message-ID: Its could be a decoder flag that signals to video renderer to play each frame twice, that's done after decoding of the frame is completed. On 3/4/12, Jared McIntyre wrote: > I'm re-encodeing broadcast video. The source was apparently 720 30fps, but > is being broadcast 60fps where every frame appears twice. I thought I'd try > to save some space by re-encoding and using 30fps to cut out every other > frame. I used the following command: > > ffmpeg -i in.mpg -acodec libfaac -ab 160k -crf 23.1 -vcodec libx264 -tune > animation -preset faster -profile baseline -f mp4 -r 29.97 out.mp4 > > I was a bit surprised to find out that if I used -r 29.97 I would get a file > with half the frames, but a larger size than if I hadn't done so. I'm sure > I'm misunderstanding something, but is this the expected behavior? Here is > the log data in case about the encodeing: > > Seems stream 0 codec frame rate differs from container frame rate: 119.88 > (120000/1001) -> 59.96 (90000/1501) > Input #0, mpeg, from 'in.mpg': > Duration: 00:02:30.91, start: 0.328367, bitrate: 11748 kb/s > Stream #0.0[0x1e0]: Video: mpeg2video (Main), yuv420p, 1280x720 [PAR 1:1 > DAR 16:9], 38810 kb/s, 59.96 fps, 59.96 tbr, 90k tbn, 119.88 tbc > Stream #0.1[0x80]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s > [buffer @ 0x101a33900] w:1280 h:720 pixfmt:yuv420p tb:1/1000000 sar:1/1 > sws_param: > [libx264 @ 0x10204a000] using SAR=1/1 > [libx264 @ 0x10204a000] using cpu capabilities: MMX2 SSE2Fast SSSE3 > FastShuffle SSE4.1 Cache64 > [libx264 @ 0x10204a000] profile Constrained Baseline, level 3.1 > [libx264 @ 0x10204a000] 264 - core 115 - H.264/MPEG-4 AVC codec - Copyleft > 2003-2011 - http://www.videolan.org/x264.html - options: cabac=0 ref=4 > deblock=1:1:1 analyse=0x1:0x111 me=hex subme=4 psy=1 psy_rd=0.40:0.00 > mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 > fast_pskip=1 chroma_qp_offset=0 threads=3 sliced_threads=0 nr=0 decimate=1 > interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 > keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=20 rc=crf > mbtree=1 crf=23.1 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 > aq=1:0.60 > Output #0, mp4, to 'out.mp4': > Metadata: > encoder : Lavf52.110.0 > Stream #0.0: Video: libx264, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], > q=2-31, 200 kb/s, 2997 tbn, 29.97 tbc > Stream #0.1: Audio: libfaac, 48000 Hz, 5.1, s16, 160 kb/s > Stream mapping: > Stream #0.0 -> #0.0 > Stream #0.1 -> #0.1 > > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From jmcintyre at dfsoftware.com Mon Mar 5 05:19:19 2012 From: jmcintyre at dfsoftware.com (Jared McIntyre) Date: Sun, 4 Mar 2012 21:19:19 -0700 Subject: [FFmpeg-user] Framerate Question In-Reply-To: References: <468DBF07-6AF0-44D4-B97B-78F593A20493@dfsoftware.com> Message-ID: <568F6786-AB7C-47BA-91FF-B07F0681D28C@dfsoftware.com> I guess it is possible, but there are subtle artifact differences between the frames if you look close enough (especially when they fade to and from commercial). On Mar 4, 2012, at 9:06 PM, Dennis wrote: > Its could be a decoder flag that signals to video renderer to play > each frame twice, that's done after decoding of the frame is > completed. > > On 3/4/12, Jared McIntyre wrote: >> I'm re-encodeing broadcast video. The source was apparently 720 30fps, but >> is being broadcast 60fps where every frame appears twice. I thought I'd try >> to save some space by re-encoding and using 30fps to cut out every other >> frame. I used the following command: >> >> ffmpeg -i in.mpg -acodec libfaac -ab 160k -crf 23.1 -vcodec libx264 -tune >> animation -preset faster -profile baseline -f mp4 -r 29.97 out.mp4 >> >> I was a bit surprised to find out that if I used -r 29.97 I would get a file >> with half the frames, but a larger size than if I hadn't done so. I'm sure >> I'm misunderstanding something, but is this the expected behavior? Here is >> the log data in case about the encodeing: >> >> Seems stream 0 codec frame rate differs from container frame rate: 119.88 >> (120000/1001) -> 59.96 (90000/1501) >> Input #0, mpeg, from 'in.mpg': >> Duration: 00:02:30.91, start: 0.328367, bitrate: 11748 kb/s >> Stream #0.0[0x1e0]: Video: mpeg2video (Main), yuv420p, 1280x720 [PAR 1:1 >> DAR 16:9], 38810 kb/s, 59.96 fps, 59.96 tbr, 90k tbn, 119.88 tbc >> Stream #0.1[0x80]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s >> [buffer @ 0x101a33900] w:1280 h:720 pixfmt:yuv420p tb:1/1000000 sar:1/1 >> sws_param: >> [libx264 @ 0x10204a000] using SAR=1/1 >> [libx264 @ 0x10204a000] using cpu capabilities: MMX2 SSE2Fast SSSE3 >> FastShuffle SSE4.1 Cache64 >> [libx264 @ 0x10204a000] profile Constrained Baseline, level 3.1 >> [libx264 @ 0x10204a000] 264 - core 115 - H.264/MPEG-4 AVC codec - Copyleft >> 2003-2011 - http://www.videolan.org/x264.html - options: cabac=0 ref=4 >> deblock=1:1:1 analyse=0x1:0x111 me=hex subme=4 psy=1 psy_rd=0.40:0.00 >> mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 >> fast_pskip=1 chroma_qp_offset=0 threads=3 sliced_threads=0 nr=0 decimate=1 >> interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 >> keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=20 rc=crf >> mbtree=1 crf=23.1 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 >> aq=1:0.60 >> Output #0, mp4, to 'out.mp4': >> Metadata: >> encoder : Lavf52.110.0 >> Stream #0.0: Video: libx264, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], >> q=2-31, 200 kb/s, 2997 tbn, 29.97 tbc >> Stream #0.1: Audio: libfaac, 48000 Hz, 5.1, s16, 160 kb/s >> Stream mapping: >> Stream #0.0 -> #0.0 >> Stream #0.1 -> #0.1 >> >> >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From pavel at sokolov.me Mon Mar 5 09:23:56 2012 From: pavel at sokolov.me (Pavel Sokolov) Date: Mon, 05 Mar 2012 12:23:56 +0400 Subject: [FFmpeg-user] Invalid audio in resulting MPEG-TS In-Reply-To: References: <4F53AC6C.10206@netscape.net> Message-ID: <4F54781C.9030602@sokolov.me> >> When I try to mux a video (MPEG-2) and audio (WAV, pcm_s16le >> ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s) file into >> a MPEG-TS container, If you want to put raw pcm like pcm_s16le to the mpeg container it calls "LPCM". As I know in ffmpeg LPCM is working only for muxers from mpegenc.c (VOB, VCD, etc) throw the private stream 1. MPEG-TS in ffmpeg has separate muxer (mpegtsenc.c) and there is no code for support LPCM inside it. So, try to remux your file to VOB or DVD. P.S. I think, that you must use pcm_s16be instead of pcm_s16le. -- With best regards, Pavel A. Sokolov mobile: +7(921)419-1819 skype: pavel_a_sokolov From andrey.krieger.utkin at gmail.com Mon Mar 5 11:47:01 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Mon, 5 Mar 2012 12:47:01 +0200 Subject: [FFmpeg-user] Framerate Question In-Reply-To: <468DBF07-6AF0-44D4-B97B-78F593A20493@dfsoftware.com> References: <468DBF07-6AF0-44D4-B97B-78F593A20493@dfsoftware.com> Message-ID: 2012/3/5 Jared McIntyre : > I'm re-encodeing broadcast video. The source was apparently 720 30fps, but is being broadcast 60fps where every frame appears twice. I thought I'd try to save some space by re-encoding and using 30fps to cut out every other frame. I used the following command: I believe x264 encoder is able to compress efficiently in this case, without such a hack. > ffmpeg -i in.mpg -acodec libfaac -ab 160k -crf 23.1 -vcodec libx264 -tune animation -preset faster -profile baseline -f mp4 -r 29.97 out.mp4 > > I was a bit surprised to find out that if I used -r 29.97 I would get a file with half the frames, but a larger size than if I hadn't done so. I'm sure I'm misunderstanding something, but is this the expected behavior? Here is the log data in case about the encodeing: What's the intention of -tune animation flag? To get video of desired bitrate, give that bitrate as a parameter. I'd suggest to not change framerate, and just tune encoder options to have satisfying output size or quality. Start from simply transcoding it without any options, just setting desired bitrate. -- Andrey Utkin From oussama.stiti at gmail.com Mon Mar 5 14:09:46 2012 From: oussama.stiti at gmail.com (Oussama Stiti) Date: Mon, 5 Mar 2012 22:09:46 +0900 Subject: [FFmpeg-user] Detection of defective frames with ffmpeg Message-ID: Hello, I'm beginner with ffmpeg, and i'm trying to detect defective frames from a video, and to extract from them important informations like slice #, frame ID, frame type, referenced frames... Please if someone knows how to do this, it would be very helpful for me. Cheers Ps: I'm running on Ubuntu 12.04 LTS From zhangzh at net-east.com Mon Mar 5 08:23:30 2012 From: zhangzh at net-east.com (zhangzh) Date: Mon, 5 Mar 2012 15:23:30 +0800 Subject: [FFmpeg-user] how to add a srt subtitle file into a mp4 Message-ID: <2012030515232996858811@net-east.com> hello I'm a new user of ffmpeg . I just want to add a srt subtitle file into a mp4 , but I failed . So how to do it ? D:\tv>ffmpeg -i hehe.mp4 -attach hehe.srt -metadata:s:0 mimetype=application/x-truetype-font -vcodec copy -acodec copy hehe2.mp4 ffmpeg version N-37208-g01fcbdf Copyright (c) 2000-2012 the FFmpeg developers built on Jan 27 2012 18:34:52 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencor -libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enabl able-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 34.101 / 51. 34.101 libavcodec 53. 60.100 / 53. 60.100 libavformat 53. 31.100 / 53. 31.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 60.100 / 2. 60.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'hehe.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 title : artist : encoder : Lavf52.93.0 copyright : duricom Duration: 00:17:16.86, start: 0.000000, bitrate: 367 kb/s Stream #0:0(und): Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D), yuv420p, 320x240 [SAR 1:1 DAR 4:3], 307 kb/s, 30 fps, 30 tbr, 30 tbn, 30 tbc Metadata: creation_time : 1970-01-01 00:00:00 handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 32000 Hz, stereo, s16, 56 kb/s Metadata: creation_time : 1970-01-01 00:00:00 handler_name : ret 0, stream_spec 0 [mp4 @ 022BB020] track 2: could not find tag, codec not currently supported in container Output #0, mp4, to 'hehe2.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 title : artist : copyright : duricom encoder : Lavf53.31.100 Stream #0:0(und): Video: mpeg4 ( [0][0][0] / 0x0020), yuv420p, 320x240 [SAR 1:1 DAR 4:3], q=2-31, 307 kb/s, 30 fps, 30 tbn, 30 tbc Metadata: creation_time : 1970-01-01 00:00:00 handler_name : VideoHandler mimetype : application/x-truetype-font Stream #0:1(und): Audio: aac (@[0][0][0] / 0x0040), 32000 Hz, stereo, 56 kb/s Metadata: creation_time : 1970-01-01 00:00:00 handler_name : Stream #0:2: Attachment: none Metadata: filename : hehe.srt Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) File hehe.srt -> Stream #0:2 Could not write header for output file #0 (incorrect codec parameters ?) zhangzh From cehoyos at ag.or.at Mon Mar 5 19:23:16 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 5 Mar 2012 18:23:16 +0000 (UTC) Subject: [FFmpeg-user] Framerate Question References: <468DBF07-6AF0-44D4-B97B-78F593A20493@dfsoftware.com> Message-ID: Andrey Utkin gmail.com> writes: > 2012/3/5 Jared McIntyre dfsoftware.com>: > > I'm re-encodeing broadcast video. The source was apparently > > 720 30fps, but is being broadcast 60fps where every frame > > appears twice. I thought I'd try to save some space by > > re-encoding and using 30fps to cut > > out every other frame. I used the following command: > > I believe x264 encoder is able to compress efficiently in > this case, without such a hack. I don't think that is possible for the actual use-case (where the frames show subtle differences, I get the same here for several tv channels). -r 30 / 30000/1001 is correct afaict if you have to re-encode, the size of the resulting video only depends on the settings, and I believe it is unlikely you will save space if you don't want to loose quality. Carl Eugen From officemab at gmail.com Mon Mar 5 19:28:44 2012 From: officemab at gmail.com (funtastic) Date: Mon, 5 Mar 2012 19:28:44 +0100 Subject: [FFmpeg-user] video delay after audio fix In-Reply-To: References: Message-ID: Am 5. M?rz 2012 01:01 schrieb Andrey Utkin : > Hi Martin. > We're glad to be helpful for ffmpeg users. > But from my point you have described many different issues, not > closely related to each other. It makes hard to understand you and to > help you. > From your first post, i got the only thing that there's no player that > plays your video correctly (you mentioned Totem, VLC, FFplay and all > they do not give required behaviour). I assume there's sth wrong with > your file. > You can fix the file by cutting the earlier-starting stream - audio, > in your case. > It will require you to split file into separate files - audio only and > video only, then cutting beginning from audio file, then joining back. > It will be simple ffmpeg commands. If you have difficulties with > making up these commands, post here. > > -- > Andrey Utkin > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > hi again. You are right the question is too abstract... so I try to precise the whole thing: The problem can be narrowed down to one specific "problem" regarding the download of a RTSP stream from a Wowza server (if this should be relevant to solve the problem). For this task I use: ffmpeg -i rtsp://184.72.239.149/vod/mp4:BigBuckBunny_175k.mov -vcodec copy -acodec copy -ss 0.0001 -t 30 -y share.mp4 (please see the output from the mailing thread... I posted everything) Two explanations: 1. I have to use -vcodec copy and -acodec copy because I don't want to encode the video during download in order to achieve the intended framerate (on a very weak computer) 2. -ss 0.0001 is a hack to overcome problems with bad timestamps at the beginning of the video... (shame on me...) Please take this 2 things as necessary. When I run the command everything works fine but the ffmpeg output shows one particualar line: Stream #0.0: Video: ![0][0][0] / 0x0021, yuv420p, 240x160, q=2-31, 24 tbn, 24 tbc (I posted the complete console output earlier) Since I believe that this line is the reason for all further problems I would like to know how the ffmpeg command from before can be used to specify a certain format (instead of getting this format list...) Remember: i have to use -vcodec copy / -acodec copy... It would be ok to tunnel the stream via http or anything else... I only want to download the stream with as minimal encoding effort as possible (-vcodec copy/ -acodec copy has given the best results for weak hardware so far)... I really appreciate every tipp or hint to support my idea.... but I don't know if this is possible... I really hope that the question is obvious this time; Regards Martin From cehoyos at ag.or.at Mon Mar 5 19:29:37 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 5 Mar 2012 18:29:37 +0000 (UTC) Subject: [FFmpeg-user] video delay after audio fix References: Message-ID: funtastic gmail.com> writes: > ffmpeg version 0.7.3-4:0.7.3-0ubuntu0.11.10.1, Copyright (c) > 2000-2011 the Libav developers This version of FFmpeg is intentionally broken (by the distributors), it contains several hundred known bugs (fixed in current FFmpeg), some of them security relevant. We therefore cannot support this version, please see http://ffmpeg.org/download.html for supported versions. Carl Eugen From cehoyos at ag.or.at Mon Mar 5 19:26:20 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 5 Mar 2012 18:26:20 +0000 (UTC) Subject: [FFmpeg-user] how to add a srt subtitle file into a mp4 References: <2012030515232996858811@net-east.com> Message-ID: zhangzh net-east.com> writes: > I'm a new user of ffmpeg . I just want to add a srt subtitle > file into a mp4 , but I failed . > [mp4 @ 022BB020] track 2: could not find tag, codec not > currently supported in container Patch welcome, the alternative is to wait. Carl Eugen From robert at theMakers.com Mon Mar 5 19:32:42 2012 From: robert at theMakers.com (Robert Reinhardt) Date: Mon, 5 Mar 2012 18:32:42 +0000 Subject: [FFmpeg-user] how to add a srt subtitle file into a mp4 In-Reply-To: References: <2012030515232996858811@net-east.com>, Message-ID: <2D405CD275952E49B92B7F48B3A0308A2B83C996@nakedex.flaction.com> I haven't explored using FFmpeg for this purpose, but MP4Box can remux an SRT file into an existing MP4 asset. I'm a big fan of MP4Box. I used it on the WelcomeBC.ca site's Newcomers Guide to create the subtitled assets for iOS deployment and Flash desktop deployment. -Robert Robert Reinhardt The difference knowledge + experience makes | Consultant @ [theMAKERS] { work: http://www.theMakers.com } { video: http://videoRx.com } { blog: http://probablyjustme.com } ________________________________________ From: ffmpeg-user-bounces at ffmpeg.org [ffmpeg-user-bounces at ffmpeg.org] on behalf of Carl Eugen Hoyos [cehoyos at ag.or.at] Sent: Monday, March 05, 2012 10:26 AM To: ffmpeg-user at ffmpeg.org Subject: Re: [FFmpeg-user] how to add a srt subtitle file into a mp4 zhangzh net-east.com> writes: > I'm a new user of ffmpeg . I just want to add a srt subtitle > file into a mp4 , but I failed . > [mp4 @ 022BB020] track 2: could not find tag, codec not > currently supported in container Patch welcome, the alternative is to wait. Carl Eugen _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From hguth at listingsmagic.com Tue Mar 6 07:11:49 2012 From: hguth at listingsmagic.com (Hans) Date: Mon, 05 Mar 2012 23:11:49 -0700 Subject: [FFmpeg-user] Is it possible to create the Peel away effect like the one in this video Message-ID: <4F55AAA5.6020106@listingsmagic.com> I was just wondering if/how it would be possible to create an effect like the one shown in the video on this page with ffmpeg? Can after effects transitions be created using ffmpeg? http://www.aihouqi.com/after-effects/5790-videohive-peel-away-129865.html Thanks From jmcintyre at dfsoftware.com Tue Mar 6 07:47:20 2012 From: jmcintyre at dfsoftware.com (Jared McIntyre) Date: Mon, 5 Mar 2012 23:47:20 -0700 Subject: [FFmpeg-user] Framerate Question In-Reply-To: References: <468DBF07-6AF0-44D4-B97B-78F593A20493@dfsoftware.com> Message-ID: <4586E3B2-3254-4FB1-9B15-83394F956409@dfsoftware.com> On Mar 5, 2012, at 3:47 AM, Andrey Utkin wrote: > I believe x264 encoder is able to compress efficiently in this case, > without such a hack. It certainly appears that this scenario doesn't make much difference on file size, which is quite impressive, though I'm not sure I agree with the "hack" characterization pf the test. >> ffmpeg -i in.mpg -acodec libfaac -ab 160k -crf 23.1 -vcodec libx264 -tune animation -preset faster -profile baseline -f mp4 -r 29.97 out.mp4 >> >> I was a bit surprised to find out that if I used -r 29.97 I would get a file with half the frames, but a larger size than if I hadn't done so. I'm sure I'm misunderstanding something, but is this the expected behavior? Here is the log data in case about the encodeing: > > What's the intention of -tune animation flag? The flag is unrelated to the issue, but it was in the command I was using during the test, so I included it here. Probably should have left it out so as not to muddy the waters. > To get video of desired bitrate, give that bitrate as a parameter. > > I'd suggest to not change framerate, and just tune encoder options to > have satisfying output size or quality. > Start from simply transcoding it without any options, just setting > desired bitrate. That is certainly the standard way to go about it. This was just an unexpected outcome of a test I was performing on a very specific use case and I wanted to make sure I wasn't misunderstanding the use of the frame rate flag and its affect on the final output. From andrey.krieger.utkin at gmail.com Tue Mar 6 10:59:10 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Tue, 6 Mar 2012 11:59:10 +0200 Subject: [FFmpeg-user] video delay after audio fix In-Reply-To: References: Message-ID: 2012/3/5 funtastic : > ffmpeg -i rtsp://184.72.239.149/vod/mp4:BigBuckBunny_175k.mov -vcodec copy > -acodec copy -ss 0.0001 -t 30 -y share.mp4 AFAIK -ss without transcoding shows bad precision/accuracy results/ because it seeks to closest key frame. This can result in +/- several seconds from desired time point. I'd like somebody else to check/correct me, though. Regarding root of you problem, bad timestamps from Wowza :) I also have this sort of problem with RTMP/FLV streams from wowza. I try to workaround it using video filter select='isnan(prev_selected_pts)+gt(pts,prev_selected_pts)' which should not allow non-monotonity of decoded frames pts. If you feel you understand the problem, and cannot solve it with ffmpeg utility, you can go deeper and make up an application that uses ffmpeg API. Hope this helps. PS Carl is right, if you use libav.org version, go to list libav-tools at libav.org , there are people to help, too. To ask for help here, use ffmpeg.org last version. -- Andrey Utkin From andrey.krieger.utkin at gmail.com Tue Mar 6 11:09:43 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Tue, 6 Mar 2012 12:09:43 +0200 Subject: [FFmpeg-user] Is it possible to create the Peel away effect like the one in this video In-Reply-To: <4F55AAA5.6020106@listingsmagic.com> References: <4F55AAA5.6020106@listingsmagic.com> Message-ID: 2012/3/6 Hans : > I was just wondering if/how it would be possible to create an effect like > the one shown in the video on this page with ffmpeg? ?Can after effects > transitions be created using ffmpeg? > > http://www.aihouqi.com/after-effects/5790-videohive-peel-away-129865.html Yes, possible. It would be a cute video filter. See libavfilter/ subdir in project sources. You can also ask in libav-user at ffmpeg.org, libav-api at libav.org for help on implementing such filter using ffmpeg API. I've checked list of video filters that are already in ffmpeg, there looks to be no such filter built in. But it is probable that such filter is existing, or is easier to implement with frei0r effect framework, which is pluggable via ffmpeg. I'm not so keen on frei0r, so this is just a speculation on what should be researched by you. -- Andrey Utkin From ajaxhe at gmail.com Tue Mar 6 12:57:36 2012 From: ajaxhe at gmail.com (ajaxhe) Date: Tue, 6 Mar 2012 03:57:36 -0800 (PST) Subject: [FFmpeg-user] How to the use vf_drawtext.c in my program Message-ID: <1331035056939-4449601.post@n4.nabble.com> Now, I want to add the system datetime information in per video frame, then store as a .mp4 file. I have known vf_drawtext.c can finish this task, but any examples to introduce the usage of APIs in drawtext.c? thanks! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-to-the-use-vf-drawtext-c-in-my-program-tp4449601p4449601.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From andrey.krieger.utkin at gmail.com Tue Mar 6 13:56:00 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Tue, 6 Mar 2012 14:56:00 +0200 Subject: [FFmpeg-user] How to the use vf_drawtext.c in my program In-Reply-To: <1331035056939-4449601.post@n4.nabble.com> References: <1331035056939-4449601.post@n4.nabble.com> Message-ID: 2012/3/6 ajaxhe : > Now, I want to add the system datetime information in per video frame, then > store as a .mp4 file. I have known vf_drawtext.c can finish this task, but > any examples to introduce the usage of APIs in drawtext.c? http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/examples/filtering_video.c;h=2ca6a0549712c8ae4f780570ae3e72c7d151b933;hb=refs/heads/master -- Andrey Utkin From bostjan.strojan at gmail.com Tue Mar 6 15:25:30 2012 From: bostjan.strojan at gmail.com (=?UTF-8?Q?Bo=C5=A1tjan_Strojan?=) Date: Tue, 6 Mar 2012 15:25:30 +0100 Subject: [FFmpeg-user] utvideo status? Message-ID: Hi, a. What is the status of utvideo encoder right now? b. Does ut support 10bit stuff as well? thanks, b. From cehoyos at ag.or.at Tue Mar 6 15:48:06 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 6 Mar 2012 14:48:06 +0000 (UTC) Subject: [FFmpeg-user] utvideo status? References: Message-ID: Bo?tjan Strojan gmail.com> writes: > a. What is the status of utvideo encoder right now? If it does not work for you, please report with a command line and complete, uncut console output. > b. Does ut support 10bit stuff as well? I don't think so, do you have any indication that utvideo supports 10 bit? Carl Eugen From ajaxhe at gmail.com Tue Mar 6 12:37:01 2012 From: ajaxhe at gmail.com (ajaxhe) Date: Tue, 6 Mar 2012 03:37:01 -0800 (PST) Subject: [FFmpeg-user] How to the use drawtext.c in my program Message-ID: <1331033821104-4449535.post@n4.nabble.com> Now, I want to add the system datetime information in per video frame, then store as a .mp4 file. I have known drawtext.c can finish this task, but any example to use the APIs in drawtext.c? thanks! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-to-the-use-drawtext-c-in-my-program-tp4449535p4449535.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From cehoyos at ag.or.at Tue Mar 6 20:35:05 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 6 Mar 2012 19:35:05 +0000 (UTC) Subject: [FFmpeg-user] Converting 16-bit TIFFs to Quicktime Motion-JPEG References: Message-ID: Jeremy Oddo jerfu.com> writes: > Does ffmpeg support 16-bit TIFFs? This should work fine now, if you still have problems, please provide a sample! Carl Eugen From madzhugin at yandex.ru Wed Mar 7 00:02:58 2012 From: madzhugin at yandex.ru (=?koi8-r?B?7cHE1tXHyc4g4czFy9PBzsTS?=) Date: Wed, 07 Mar 2012 03:02:58 +0400 Subject: [FFmpeg-user] Fwd: problem with analyzeduration Message-ID: <5431331074978@web73.yandex.ru> -------- ???????????? ????????? -------- 27.02.2012, 23:05, "???????? ?????????" : Hi. I have faced the following problem - i have to make a photo with webcam and i have to do it quickly. But while running ffmpeg analizes input stream too long - about 2-3 sec. I tried to use the option "-analyzeduration 0", however, it didn't give any result - ffmpeg analizes input stream anyway. I also tried to describe input stream totally, but it didn't give any result again. I use the following command: "ffmpeg -f video4linux2,v4l2 -analyzeduration 0 -s 640x480 -qscale 0 -r 30 -pix_fmt yuyv422 -i /dev/video0 -f image2 -c:v mjpeg -pix_fmt yuvj422p -vframes 1 out.jpg" I get out: -------------------------------------------------------------------------------------------------------------------------------- "ffmpeg version 0.9.1-4:0.9.1-0ubuntu1~jon1~lucid1, Copyright (c) 2000-2012 the FFmpeg developers ??built on Jan ?7 2012 15:05:17 with gcc 4.4.3 ??configuration: --extra-version='4:0.9.1-0ubuntu1~jon1~lucid1' --arch=i386 --prefix=/usr --disable-stripping --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-libfreetype --enable-gpl --enable-postproc --enable-x11grab --enable-libdc1394 --enable-shared --disable-static ??WARNING: library configuration mismatch ??avutil ?????configuration: --extra-version='4:0.9.1ubuntu1~jon1~lucid1' --arch=i386 --prefix=/usr --disable-stripping --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-libfreetype --enable-frei0r --enable-libopenjpeg --enable-gpl --enable-postproc --enable-x11grab --enable-libdirac --enable-libmp3lame --enable-libx264 --enable-libopencore-amrnb --enable-version3 --enable-libopencore-amrwb --enable-version3 --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay --disable-ffprobe --disable-ffserver --disable-avconv ??avcodec ????configuration: --extra-version='4:0.9.1ubuntu1~jon1~lucid1' --arch=i386 --prefix=/usr --disable-stripping --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-libfreetype --enable-frei0r --enable-libopenjpeg --enable-gpl --enable-postproc --enable-x11grab --enable-libdirac --enable-libmp3lame --enable-libx264 --enable-libopencore-amrnb --enable-version3 --enable-libopencore-amrwb --enable-version3 --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay --disable-ffprobe --disable-ffserver --disable-avconv ??avformat ???configuration: --extra-version='4:0.9.1-0ubuntu1~jon1~lucid1' --arch=i386 --prefix=/usr --disable-stripping --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-libfreetype --enable-gpl --enable-postproc --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay --disable-ffprobe --disable-ffserver --disable-avconv ??avdevice ???configuration: --extra-version='4:0.9.1-0ubuntu1~jon1~lucid1' --arch=i386 --prefix=/usr --disable-stripping --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-libfreetype --enable-gpl --enable-postproc --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay --disable-ffprobe --disable-ffserver --disable-avconv ??avfilter ???configuration: --extra-version='4:0.9.1-0ubuntu1~jon1~lucid1' --arch=i386 --prefix=/usr --disable-stripping --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-libfreetype --enable-gpl --enable-postproc --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay --disable-ffprobe --disable-ffserver --disable-avconv ??swscale ????configuration: --extra-version='4:0.9.1-0ubuntu1~jon1~lucid1' --arch=i386 --prefix=/usr --disable-stripping --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-libfreetype --enable-gpl --enable-postproc --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay --disable-ffprobe --disable-ffserver --disable-avconv ??postproc ???configuration: --extra-version='4:0.9.1-0ubuntu1~jon1~lucid1' --arch=i386 --prefix=/usr --disable-stripping --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-libfreetype --enable-gpl --enable-postproc --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay --disable-ffprobe --disable-ffserver --disable-avconv ??libavutil ???51. 32. 0 / 51. 32. 0 ??libavcodec ??53. 42. 4 / 53. 42. 4 ??libavformat ?53. 24. 2 / 53. 24. 2 ??libavdevice ?53. ?4. 0 / 53. ?4. 0 ??libavfilter ??2. 53. 0 / ?2. 53. 0 ??libswscale ???2. ?1. 0 / ?2. ?1. 0 ??libpostproc ?52. ?0. 0 / 52. ?0. 0 [video4linux2,v4l2 @ 0x972b580] Estimating duration from bitrate, this may be inaccurate Input #0, video4linux2,v4l2, from '/dev/video0': ??Duration: N/A, start: 455998.312997, bitrate: 147456 kb/s ????Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 640x480, 147456 kb/s, 30 tbr, 1000k tbn, 30 tbc [buffer @ 0x972bb60] w:640 h:480 pixfmt:yuyv422 tb:1/1000000 sar:0/1 sws_param: [buffersink @ 0x972c000] auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'out' [scale @ 0x972faa0] w:640 h:480 fmt:yuyv422 -> w:640 h:480 fmt:yuvj422p flags:0x4 Output #0, image2, to 'out.jpg': ??Metadata: ????encoder ????????: Lavf53.24.2 ????Stream #0:0: Video: mjpeg, yuvj422p, 640x480, q=2-31, 200 kb/s, 90k tbn, 30 tbc Stream mapping: ??Stream #0:0 -> #0:0 (rawvideo -> mjpeg) Press [q] to stop, [?] for help frame= ???1 fps= ?0 q=1.6 Lsize= ??????0kB time=00:00:00.03 bitrate= ??0.0kbits/s video:29kB audio:0kB global headers:0kB muxing overhead -100.000000% -------------------------------------------------------------------------------------------------------------------------------- Whats wrong with it? P.S. -------------------------------------------------------------------------------------------------------------------------------- $ uname -a Linux kiosk 3.2.5-030205-generic #201202061401 SMP Mon Feb 6 19:11:06 UTC 2012 i686 GNU/Linux -------------------------------------------------------------------------------------------------------------------------------- $ lsusb .. Bus 001 Device 004: ID 0458:7060 KYE Systems Corp. (Mouse Systems) .. -------------------------------------------------------------------------------------------------------------------------------- $ modinfo uvcvideo filename: ??????/lib/modules/3.2.5-030205-generic/kernel/drivers/media/video/uvc/uvcvideo.ko version: ???????1.1.1 license: ???????GPL description: ???USB Video Class driver author: ????????Laurent Pinchart srcversion: ????8C7099B998ADA2049B68BD6 alias: ?????????usb:v*p*d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v1C4Fp3000d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v1B3Bp2951d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v19ABp1000d00*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v19ABp1000d01[0-1]*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v19ABp1000d012[0-6]dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v199Ep8102d*dc*dsc*dp*icFFisc01ip00* alias: ?????????usb:v18ECp3290d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v18ECp3288d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v18ECp3188d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v18CDpCAFEd*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v1871p0306d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v17EFp480Bd*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v17DCp0202d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v174Fp8A34d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v174Fp8A33d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v174Fp8A31d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v174Fp8A12d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v174Fp5931d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v174Fp5212d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v152Dp0310d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v13D3p5103d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v0E8Dp0004d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v0AC8p3420d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v0AC8p3410d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v0AC8p332Dd*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v06F8p300Cd*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v05E3p0505d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v05C8p0403d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v05ACp8501d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v058Fp3820d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v04F2pB071d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v046Dp08C7d*dc*dsc*dp*icFFisc01ip00* alias: ?????????usb:v046Dp08C6d*dc*dsc*dp*icFFisc01ip00* alias: ?????????usb:v046Dp08C5d*dc*dsc*dp*icFFisc01ip00* alias: ?????????usb:v046Dp08C3d*dc*dsc*dp*icFFisc01ip00* alias: ?????????usb:v046Dp08C2d*dc*dsc*dp*icFFisc01ip00* alias: ?????????usb:v046Dp08C1d*dc*dsc*dp*icFFisc01ip00* alias: ?????????usb:v045Ep0723d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v045Ep00F8d*dc*dsc*dp*ic0Eisc01ip00* alias: ?????????usb:v0458p706Ed*dc*dsc*dp*ic0Eisc01ip00* depends: ???????videodev intree: ????????Y vermagic: ??????3.2.5-030205-generic SMP mod_unload modversions 686 parm: ??????????clock:Video buffers timestamp clock parm: ??????????nodrop:Don't drop incomplete frames (uint) parm: ??????????quirks:Forced device quirks (uint) parm: ??????????trace:Trace level bitmask (uint) parm: ??????????timeout:Streaming control requests timeout (uint) -------- ?????????? ????????????? ????????? -------- From alok+ffmpeg at mbaproductions.com Wed Mar 7 05:59:21 2012 From: alok+ffmpeg at mbaproductions.com (Alok) Date: Wed, 7 Mar 2012 05:59:21 +0100 Subject: [FFmpeg-user] ProRes Invalid pixel format string In-Reply-To: References: Message-ID: Hello, I'm trying to use ffmpeg to crop, rotate, and encode ProRes 422 videos into JPEG Image sequences. I'm running on Mac OS X 10.7, and have tried the following configurations, based on other threads I've found: --enable-libmp3lame --enable-shared --disable-mmx --arch=x86_64 --enable-libmp3lame --enable-shared --disable-mmx --arch=x86_64 --enable-gpl --enable-version3 I've also tried adding -vf "format=yuv:420p" to the command without success. Using any of these configurations yields a result similar to this: sneezy:ffmpeg admin$ ffmpeg -i "video.mov" -an -f image2 -vf "crop 4090:3106:6:0" -vf "transpose=1" "imgseq_%05d.jpg" FFmpeg version SVN-r26402, Copyright (c) 2000-2011 the FFmpeg developers built on Mar 6 2012 20:48:57 with llvm_gcc 4.2.1 (Based on Apple Inc. build 5658) (LLVM build 2336.9.00) configuration: --enable-libmp3lame --enable-shared --disable-mmx --arch=x86_64 --enable-gpl --enable-version3 libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 29.97 (30000/1001) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mov': Metadata: major_brand : qt minor_version : 0 compatible_brands: qt creation_time : 2012-03-03 01:41:16 Duration: 00:02:38.05, start: 0.000000, bitrate: 678316 kb/s Stream #0.0(und): Video: Apple ProRes 422, 4096x3112, 670989 kb/s, PAR 1:1 DAR 512:389, 29.97 fps, 29.97 tbr, 30k tbn, 30k tbc Metadata: creation_time : 2012-03-03 01:41:17 Stream #0.1(und): Audio: pcm_s24le, 48000 Hz, 6 channels, s32, 6912 kb/s Metadata: creation_time : 2012-03-03 01:41:17 Stream #0.2(und): Data: tmcd / 0x64636D74 Metadata: creation_time : 2012-03-03 01:41:17 [buffer @ 0x7fbf5bc04340] Invalid pixel format string '-1' Error opening filters! Any thoughts? From joddo at jerfu.com Wed Mar 7 07:23:08 2012 From: joddo at jerfu.com (Jeremy Oddo) Date: Tue, 6 Mar 2012 22:23:08 -0800 Subject: [FFmpeg-user] Converting 16-bit TIFFs to Quicktime Motion-JPEG In-Reply-To: References: Message-ID: That's great news Carl. I'll give it a shot tomorrow (time permitting). On Tue, Mar 6, 2012 at 11:35 AM, Carl Eugen Hoyos wrote: > Jeremy Oddo jerfu.com> writes: > >> Does ffmpeg support 16-bit TIFFs? > > This should work fine now, if you still have problems, > please provide a sample! > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From cehoyos at ag.or.at Wed Mar 7 08:12:02 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 7 Mar 2012 07:12:02 +0000 (UTC) Subject: [FFmpeg-user] ProRes Invalid pixel format string References: Message-ID: Alok mbaproductions.com> writes: > FFmpeg version SVN-r26402, Copyright (c) 2000-2011 This is ancient and does not support ProRes, please use current git head. Carl Eugen From lytc at vega.com.vn Wed Mar 7 09:24:45 2012 From: lytc at vega.com.vn (Ly Tran Cong) Date: Wed, 7 Mar 2012 15:24:45 +0700 Subject: [FFmpeg-user] ISMV muxer example Message-ID: Hi, In ffmpeg 0.10, I see that ffmpeg now support ISMV muxer but I don't know how to use it. Can any body can tell me an example command of ISMV (Smooth Streaming) muxer. Thanks -- Regards, Tran Ly Vega Corporation 98 Hoang Quoc Viet Str, Hanoi, Vietnam Tel: 84 4 755 4190 Fax: 84 4 755 4190 Mobile: 84 91 487 1115 www.vega.com.vn; www.clip.vn; www.chacha.vn; www.ringring.vn From pb at das-werkstatt.com Wed Mar 7 09:52:23 2012 From: pb at das-werkstatt.com (Peter B.) Date: Wed, 07 Mar 2012 09:52:23 +0100 Subject: [FFmpeg-user] Technical specification for FFv1? Message-ID: <20120307095223.1929961ws9nn0s7b@webmail.tuwien.ac.at> Hello, As some on this list might already know, we at the Austrian Mediathek (=national audio/video archive) are using FFv1 for long-term storage of our digitized video material. Now, we've received questions about the standardization status of FFv1, and as far as I know: there is none. The only specification of FFv1 I know about is: a) The code itself [1] b) A short overview-text [2] (still up-to-date?) Since FFv1 has turned out to currently be *the* codec for video archiving (compared to JPEG2k and Uncompressed), it would be great if other archives would/could use it, too. Long story short: Is there any proper documentation, which could be used to propose it for standardization? Or to improve faith that its bitstream can be opened in x years from now? Any feedback / information welcome! Thanks in advance, Peter B. == References: [1] http://git.videolan.org/?p=ffmpeg.git;a=blob_plain;f=libavcodec/ffv1.c;hb=HEAD [2] http://www1.mplayerhq.hu/~michael/ffv1.html From andrey.krieger.utkin at gmail.com Wed Mar 7 11:14:21 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Wed, 7 Mar 2012 12:14:21 +0200 Subject: [FFmpeg-user] Fwd: problem with analyzeduration In-Reply-To: <5431331074978@web73.yandex.ru> References: <5431331074978@web73.yandex.ru> Message-ID: 7 ????? 2012??. 1:02 ???????????? ???????? ????????? ???????: > Hi. > I have faced the following problem - i have to make a photo with webcam and i have to do it quickly. > But while running ffmpeg analizes input stream too long - about 2-3 sec. > I tried to use the option "-analyzeduration 0", however, it didn't give any result - ffmpeg analizes input stream anyway. > I also tried to describe input stream totally, but it didn't give any result again. > > I use the following command: > "ffmpeg -f video4linux2,v4l2 -analyzeduration 0 -s 640x480 -qscale 0 -r 30 -pix_fmt yuyv422 -i /dev/video0 -f image2 -c:v mjpeg -pix_fmt yuvj422p -vframes 1 out.jpg" Try adding -probesize 0 -fpsprobesize 0 -- Andrey Utkin From drabner at zoobe.com Wed Mar 7 13:39:15 2012 From: drabner at zoobe.com (Drabner) Date: Wed, 7 Mar 2012 04:39:15 -0800 (PST) Subject: [FFmpeg-user] Combine unordered images to video (+MP3 audio) Message-ID: <1331123955470-4453148.post@n4.nabble.com> I have also posted this problem on stackoverflow (http://stackoverflow.com/questions/9600384/combine-unordered-images-to-video-mp3-audio) and will add the answer there, if I receive one on the mailing list - and vice versa. I did a few jobs with ffmpeg and also created my own dll, using ffmpeg API directly, but for my next project, I need to be able to combine multiple images together to a video with ffmpeg, also adding a mp3 sound clip. I know that you can do that for ordered images like this: For images: image001.jpg image002.jpg image003.jpg etc... ffmpeg command line: ffmpeg -f image2 -i img%03d.jpg -i sound.mp3 output.mpg But in our project, we do not have the images ordered like that. Instead, which images to use for the video in which order is determined at runtime (one image for each frame of a video at 30fps). So a video with 10 frames, for example, could have to consist of the following order of images: image001.jpg image002.jpg image111.jpg image012.jpg imageFun.jpg image001.jpg image002.jpg imageFun.jpg image055.jpg imageEnd.jpg How would I do that using ffmpeg? This part of the documentation (http://ffmpeg.org/ffmpeg.html#toc-image2-2) doesn't exactly help me here. I really don't want to resort to using the ffmpeg API directly from C/C++, but fear that I have to if that is not possible "natively". *Addition:* If that is not possible with ffmpeg, but with some other software (that runs on Linux and can be controlled from command line) - I'm all ears! ;) -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Combine-unordered-images-to-video-MP3-audio-tp4453148p4453148.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From andrey.krieger.utkin at gmail.com Wed Mar 7 13:46:17 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Wed, 7 Mar 2012 14:46:17 +0200 Subject: [FFmpeg-user] Combine unordered images to video (+MP3 audio) In-Reply-To: <1331123955470-4453148.post@n4.nabble.com> References: <1331123955470-4453148.post@n4.nabble.com> Message-ID: 2012/3/7 Drabner : > But in our project, we do not have the images ordered like that. Instead, > which images to use for the video in which order is determined at runtime > (one image for each frame of a video at 30fps). > > So a video with 10 frames, for example, could have to consist of the > following order of images: > image001.jpg image002.jpg image111.jpg image012.jpg imageFun.jpg > image001.jpg image002.jpg imageFun.jpg image055.jpg imageEnd.jpg Just copy or symlink your images to another dir with other names, to have names form an order. If not acceptable - then yes, you have to use API. -- Andrey Utkin From drabner at zoobe.com Wed Mar 7 14:01:40 2012 From: drabner at zoobe.com (Drabner) Date: Wed, 7 Mar 2012 05:01:40 -0800 (PST) Subject: [FFmpeg-user] Combine unordered images to video (+MP3 audio) In-Reply-To: References: <1331123955470-4453148.post@n4.nabble.com> Message-ID: <1331125300427-4453209.post@n4.nabble.com> Damn. Well, we'll have to try out if that affects performance or something else too much. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Combine-unordered-images-to-video-MP3-audio-tp4453148p4453209.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From singthesorrow at gmx.de Wed Mar 7 14:20:24 2012 From: singthesorrow at gmx.de (Flipp) Date: Wed, 7 Mar 2012 05:20:24 -0800 (PST) Subject: [FFmpeg-user] Use libavformat to write raw MPEG4 from raw RTP data Message-ID: <1331126424692-4453244.post@n4.nabble.com> Hello, from an IP camera, I get a MPEG4 Stream over RTP. This stream, as well as the preceding RTSP communication gets logged by an external hardware. This means I am actually not reading from a real RTP stream as far more I am reading the raw data out of it. In my data I have the SDP Information ( e.g. the config field ) and also I have all the RTP messages. As far as I know I need to parse all RTP frames for their Marker Bit and put always one whole Frame ( until Marker Bit = 1 ) into a buffer and then pass it to the lib. Now my problem is that I need some kind of example to get my mp4 file written - all examples I found are accessing a file. I just can't find out which functions I would need and how I would pass the data appropriately to them. Maybe you could supply me with some links or examples. Regards -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Use-libavformat-to-write-raw-MPEG4-from-raw-RTP-data-tp4453244p4453244.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From longxd at gmail.com Wed Mar 7 18:13:22 2012 From: longxd at gmail.com (X. Long) Date: Wed, 7 Mar 2012 12:13:22 -0500 Subject: [FFmpeg-user] Where to find the version ffmpeg version N-31774-g6c4e9ca In-Reply-To: References: <20120301161307.GA19772@phare.normalesup.org> Message-ID: bat: Thanks a lot. This really helped me out. Cindy 2012/3/1 bat guano : > > > > ---------------------------------------- > .. But I want to run it on windows, is there a built based on this >> version of the source code > > Hi > Windows build ffmpeg-git-6c4e9ca-win32-static.7z? 06-Aug-2011 > It's available from here ---> http://ffmpeg.zeranoe.com/builds/win32/static/ > ;-) > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From chabboud at yahoo.com Wed Mar 7 21:08:45 2012 From: chabboud at yahoo.com (charley a) Date: Wed, 7 Mar 2012 12:08:45 -0800 (PST) Subject: [FFmpeg-user] Real-time H.264 to TS transcoding? Message-ID: <1331150925.52452.YahooMailNeo@web160101.mail.bf1.yahoo.com> I have been using ffmpeg to convert complete h.264 files to mpeg-2 transport streams using the command:?ffmpeg -i input_file.h264 -vcodec copy -acodec copy output_file.ts. What I would like to do is be able to convert h.264 data as it is coming in via?dedicated?Ethernet in real-time. Before going off and writing my own transcoder, can this be easily done with ffmpeg (assuming I can already capture h.264 elementary bitstream and dump it a file on the hard drive)? So: h.264 -> file -> ffmpeg -> ts From brown at mrvideo.vidiot.com Wed Mar 7 23:32:30 2012 From: brown at mrvideo.vidiot.com (Mike Brown) Date: Wed, 7 Mar 2012 16:32:30 -0600 Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck In-Reply-To: <4F3C161A.10701@alcatel-lucent.com> References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <4F3C161A.10701@alcatel-lucent.com> Message-ID: <20120307223230.GX19913@mrvideo.vidiot.com> On Wed, Feb 15, 2012 at 03:31:22PM -0500, Mike Scheutzow wrote: Any update on getting this stream comverted to WAV? MB > Mike Brown wrote: >> On to the next issue. >> >> I have a D-VHS deck that will firewire the analog video to MPEG-2 video >> conversion to my computer, via TSReader. Within the mux is a MPEG audio >> stream and a LPCM audio stream. >> > >> [mpegts @ 0214A9E0] decoding for stream 2 failed >> [mpegts @ 0214A9E0] Could not find codec parameters (Audio: aac_latm (BSSD / 0x44535342), 0 channels, s16) > > From the source, the mpegts demux doesn't appear to recognize a stream type > of 0x83 and a registration descriptor "BSSD". So this stream type is not > implemented yet. > > Does TSReader document the format that they're putting into that stream? > Often these formats have some kind of frame header to specify word size, > number of channels to the decoder. > > > Mike Scheutzow > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- e-mail: vidiot at vidiot.com | vidiot at vidiot.net /~\ The ASCII 6082066843 at email.uscc.net (140 char limit) \ / Ribbon Campaign Visit - URL: http://vidiot.com/ X Against http://vidiot.net/ / \ HTML Email From bcardarella at gmail.com Thu Mar 8 00:05:50 2012 From: bcardarella at gmail.com (Brian Cardarella) Date: Wed, 7 Mar 2012 18:05:50 -0500 Subject: [FFmpeg-user] ffmpeg and rmvb conversions Message-ID: Hi, I'm sure this has been brought up before but I'm not sure how to search the mailing list so my apologies. I have a rmvb (RealMedia Variable Bitrate) file that I'd like to convert. All of my attempts has resulted in a jittery video. Some research has told me that this is the result of the variable frame rate and it probably needs to be compensated for. I'm not quite sure how to handle that in ffmpeg. Any pointers would be greatly appreciated. Thank-you! - Brian From michaelni at gmx.at Thu Mar 8 05:18:15 2012 From: michaelni at gmx.at (Michael Niedermayer) Date: Thu, 8 Mar 2012 05:18:15 +0100 Subject: [FFmpeg-user] Technical specification for FFv1? In-Reply-To: <20120307095223.1929961ws9nn0s7b@webmail.tuwien.ac.at> References: <20120307095223.1929961ws9nn0s7b@webmail.tuwien.ac.at> Message-ID: <20120308041815.GA22962@kiste2> On Wed, Mar 07, 2012 at 09:52:23AM +0100, Peter B. wrote: > Hello, > > As some on this list might already know, we at the Austrian > Mediathek (=national audio/video archive) are using FFv1 for > long-term storage of our digitized video material. > > Now, we've received questions about the standardization status of > FFv1, and as far as I know: there is none. > > The only specification of FFv1 I know about is: > a) The code itself [1] > b) A short overview-text [2] (still up-to-date?) > > Since FFv1 has turned out to currently be *the* codec for video > archiving (compared to JPEG2k and Uncompressed), it would be great > if other archives would/could use it, too. > > Long story short: > Is there any proper documentation, which could be used to propose it > for standardization? Or to improve faith that its bitstream can be > opened in x years from now? FFmpeg supports opening almost every format, form the most obscure and dead formats from 20 and more years ago. To the most recent cutting edge. I think any fear that we would drop support for FFv1 is quite unfounded. But if we really lost all sanity, and our servers would be all bombed by NATO. Still everyone who ever checked out a git version does have not just one checkout, he has _every_ revission that has been ever pushed into public by us. Thus he can always checkout and compile that version with which his video was encoded and decode it to YUV or RGB RAW. This is very hypothetical, we wont drop support. That said, FFmpeg is open source and free software, theres no risk of being locked into a format that cant be converted all of a sudden everyone has the code to decode it ... [...] -- Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB There will always be a question for which you do not know the correct awnser. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From yogesh.bit2006 at gmail.com Thu Mar 8 07:24:08 2012 From: yogesh.bit2006 at gmail.com (Yogesh Tyagi) Date: Thu, 8 Mar 2012 11:54:08 +0530 Subject: [FFmpeg-user] how to enable hardware acceleration in ffmpeg In-Reply-To: References: Message-ID: Hi, I wanted to know whether VAAPI can return the decode data back to application or the decoded data has to be displayed using VAAPI(by vaPutSurface call).I am trying to plugin a hardware decoder(provided by a separate media driver for a board) as a replacement of VAAPI into FFmpeg.Will I have to display decoded data using X library or can I use other display system to display this data. Thanks and Regards, Yogesh On Sat, Mar 3, 2012 at 12:49 PM, Yogesh Tyagi wrote: > Hi, > > Yes I want to use vaapi to display media using hwaccel.Is there any > option by which I can switch b/w software decoding and hwaccel at run time? > > Thanks and Regards, > Yogesh > > > > > On Fri, Mar 2, 2012 at 7:59 PM, Tom Evans wrote: > >> On Fri, Mar 2, 2012 at 1:42 PM, Yogesh Tyagi >> wrote: >> > Hi, >> > >> > Is there any command line option in ffmpeg to enable hardware >> acceleration >> > for h264 decoding.I have configured and compiled ffmpeg with" >> > --enable-vaapi --enable-hwaccel=h264_vaapi" options but ffmpeg is not >> > invoking decode_slice function of vaapi_h264.c. >> > >> >> As I understand it, vdpau and vaapi are useful for decoding when you >> want to display that decoded content. It's not a usable API, eg for >> decoding video as a step for subsequently re-encoding it (if that >> makes sense). >> >> IE, you can use vdpau/vaapi to display media using hwaccel, but you >> can't use it to accelerate decoding whilst encoding. >> >> Cheers >> >> Tom >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > > > -- > Thanks and Regards, > Yogesh Tyagi > Mobile No: 09911814030 > -- Thanks and Regards, Yogesh Tyagi Mobile No: 09911814030 From cehoyos at ag.or.at Thu Mar 8 07:55:43 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 8 Mar 2012 06:55:43 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg and rmvb conversions References: Message-ID: Brian Cardarella gmail.com> writes: > I have a rmvb (RealMedia Variable Bitrate) file that I'd like to > convert. All of my attempts has resulted in a jittery video. Command line and complete, uncut ouput missing. Carl Eugen From cehoyos at ag.or.at Thu Mar 8 07:59:38 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 8 Mar 2012 06:59:38 +0000 (UTC) Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <4F3C161A.10701@alcatel-lucent.com> <20120307223230.GX19913@mrvideo.vidiot.com> Message-ID: Mike Brown mrvideo.vidiot.com> writes: > Any update on getting this stream comverted to WAV? You will either have to test the patch I posted in this thread or provide longer samples (possibly with intended output). Please do not top-post here, it is considered rude, Carl Eugen From andrey.krieger.utkin at gmail.com Thu Mar 8 08:15:27 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Thu, 8 Mar 2012 09:15:27 +0200 Subject: [FFmpeg-user] Real-time H.264 to TS transcoding? In-Reply-To: <1331150925.52452.YahooMailNeo@web160101.mail.bf1.yahoo.com> References: <1331150925.52452.YahooMailNeo@web160101.mail.bf1.yahoo.com> Message-ID: 2012/3/7 charley a : > I have been using ffmpeg to convert complete h.264 files to mpeg-2 transport streams using the command:?ffmpeg -i input_file.h264 -vcodec copy -acodec copy output_file.ts. > > What I would like to do is be able to convert h.264 data as it is coming in via?dedicated?Ethernet in real-time. Before going off and writing my own transcoder, can this be easily done with ffmpeg (assuming I can already capture h.264 elementary bitstream and dump it a file on the hard drive)? Yes. Just give it URL as input file. -- Andrey Utkin From oussama.stiti at gmail.com Thu Mar 8 08:18:47 2012 From: oussama.stiti at gmail.com (Oussama Stiti) Date: Thu, 8 Mar 2012 16:18:47 +0900 Subject: [FFmpeg-user] How to detect distorded frames with ffmpeg Message-ID: Hello, I'm beginner with ffmpeg, and i'm trying to detect defective frames from a video, and to extract from them important informations like slice #, frame ID, frame type, referenced frames... Please if someone knows how to do this, it would be very helpful for me. Kind Regards Ps: I'm running on Ubuntu 12.04 LTS -- *Oussama Stiti* E-mail: oussama.stiti at gmail.com From cehoyos at ag.or.at Thu Mar 8 08:22:00 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 8 Mar 2012 07:22:00 +0000 (UTC) Subject: [FFmpeg-user] How to detect distorded frames with ffmpeg References: Message-ID: Oussama Stiti gmail.com> writes: > I'm beginner with ffmpeg, and i'm trying to detect defective frames Please define "defective frames" / "distorted frames". Carl Eugen From oussama.stiti at gmail.com Thu Mar 8 08:47:04 2012 From: oussama.stiti at gmail.com (Oussama Stiti) Date: Thu, 8 Mar 2012 16:47:04 +0900 Subject: [FFmpeg-user] How to detect distorded frames with ffmpeg In-Reply-To: References: Message-ID: Hi Carl, Thank you for responding, actually, what i mean by "distorded frames" is a corrupted, or a missing frame. I'm working on an analyzer of video quality in real time, and i have to know exactly which frames are defective. Could ffmpeg, provide me with such information ? Regards. From brown at mrvideo.vidiot.com Thu Mar 8 08:50:46 2012 From: brown at mrvideo.vidiot.com (Mike Brown) Date: Thu, 8 Mar 2012 01:50:46 -0600 Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck In-Reply-To: References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <4F3C161A.10701@alcatel-lucent.com> <20120307223230.GX19913@mrvideo.vidiot.com> Message-ID: <20120308075046.GE24977@mrvideo.vidiot.com> On Thu, Mar 08, 2012 at 06:59:38AM +0000, Carl Eugen Hoyos wrote: > Mike Brown mrvideo.vidiot.com> writes: > > > Any update on getting this stream comverted to WAV? > > You will either have to test the patch I posted in this thread > or provide longer samples (possibly with intended output). Never saw the patch. As I mentioned in a reply in a later response that mentioned the patch, I said that I do not compile on MS systems, only on Linux/Unix. That is assuming that "patch" means that a source patch was provided. If you provided a binary, I never saw a link to it. Another member of the group responded that he tested the patch and the output was distorted. At that point I replied to the query as to what the audio was. I'm curious as to why the short piece that I provided wasn't good enough. I contain 1kHz tone, which would be easy to detect if it was decoded wrong. My intended output is 2ch WAV. > Please do not top-post here, it is considered rude, Carl Eugen It was done instead of burying the single line deep into the posting. If I was responding to particular paragraphs/lines, like in this posting, and placed the response at the top, then yes, I would agree with you. MB -- e-mail: vidiot at vidiot.com | vidiot at vidiot.net /~\ The ASCII 6082066843 at email.uscc.net (140 char limit) \ / Ribbon Campaign Visit - URL: http://vidiot.com/ X Against http://vidiot.net/ / \ HTML Email From cehoyos at ag.or.at Thu Mar 8 09:29:14 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 8 Mar 2012 08:29:14 +0000 (UTC) Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <4F3C161A.10701@alcatel-lucent.com> <20120307223230.GX19913@mrvideo.vidiot.com> <20120308075046.GE24977@mrvideo.vidiot.com> Message-ID: Mike Brown mrvideo.vidiot.com> writes: > > You will either have to test the patch I posted in this thread > > or provide longer samples (possibly with intended output). > > Never saw the patch. http://thread.gmane.org/gmane.comp.video.ffmpeg.user/35474/focus=35592 Carl Eugen From brown at mrvideo.vidiot.com Thu Mar 8 09:48:39 2012 From: brown at mrvideo.vidiot.com (Mike Brown) Date: Thu, 8 Mar 2012 02:48:39 -0600 Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck In-Reply-To: References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <4F3C161A.10701@alcatel-lucent.com> <20120307223230.GX19913@mrvideo.vidiot.com> <20120308075046.GE24977@mrvideo.vidiot.com> Message-ID: <20120308084839.GI24977@mrvideo.vidiot.com> On Thu, Mar 08, 2012 at 08:29:14AM +0000, Carl Eugen Hoyos wrote: > Mike Brown mrvideo.vidiot.com> writes: > > > > You will either have to test the patch I posted in this thread > > > or provide longer samples (possibly with intended output). > > > > Never saw the patch. > > http://thread.gmane.org/gmane.comp.video.ffmpeg.user/35474/focus=35592 As I figured in my previous posting. It is a source patch file, which I can do nothing with. I can only work with executables on my XP boxes. As Mike Scheutzow indicated on 2/22, it didn't work. The audio was distorted. MB -- e-mail: vidiot at vidiot.com | vidiot at vidiot.net /~\ The ASCII 6082066843 at email.uscc.net (140 char limit) \ / Ribbon Campaign Visit - URL: http://vidiot.com/ X Against http://vidiot.net/ / \ HTML Email From drabner at zoobe.com Thu Mar 8 10:07:34 2012 From: drabner at zoobe.com (Drabner) Date: Thu, 8 Mar 2012 01:07:34 -0800 (PST) Subject: [FFmpeg-user] Combine unordered images to video (+MP3 audio) In-Reply-To: <1331125300427-4453209.post@n4.nabble.com> References: <1331123955470-4453148.post@n4.nabble.com> <1331125300427-4453209.post@n4.nabble.com> Message-ID: <1331197654857-4455934.post@n4.nabble.com> There is indeed a better way to do this, via piping! A user on stackoverflow ( http://stackoverflow.com/questions/9600384/combine-unordered-images-to-video-mp3-audio ) posted that answer. I am now using the following command do do what I want to achieve: *type "list of jpeg images here" | ffmpeg -s 640x360 -f image2pipe -vcodec mjpeg -r 30 -i - -i audioMp3.data -y video.mp4* -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Combine-unordered-images-to-video-MP3-audio-tp4453148p4455934.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From pb at das-werkstatt.com Thu Mar 8 10:51:20 2012 From: pb at das-werkstatt.com (Peter B.) Date: Thu, 08 Mar 2012 10:51:20 +0100 Subject: [FFmpeg-user] Technical specification for FFv1? In-Reply-To: <20120308041815.GA22962@kiste2> References: <20120307095223.1929961ws9nn0s7b@webmail.tuwien.ac.at> <20120308041815.GA22962@kiste2> Message-ID: <20120308105120.11554yn483yy7xug@webmail.tuwien.ac.at> Zitat von Michael Niedermayer : > That said, FFmpeg is open source and free software, theres no risk > of being locked into a format that cant be converted all of a sudden > everyone has the code to decode it ... ...and that's exactly one of my top-5 arguments *why* I'm supporting and promoting Free Software and Open Formats within the archive domain! :) However, I'm *not* afraid of you dropping support of FFv1, or that I'm not able to keep the code alive to open the videos. It's something else: It's generally about 2 things: a) In order to use FFv1 among video "professionals", there needs to be more than sustainable code that "just works". b) What about possible changes of FFv1's bitstream (for any reason), which cause incompatibilities. Stuff like that. If there'd be any kind of tech-specs to refer to, it would make it easier to propose FFv1's usage. Thanks, Pb From ben.halhead at quvex.com Thu Mar 8 11:23:38 2012 From: ben.halhead at quvex.com (Ben Halhead) Date: Thu, 08 Mar 2012 10:23:38 +0000 Subject: [FFmpeg-user] FLV to MP4 conversion Message-ID: <4F5888AA.7020102@quvex.com> Hi We are using HDFVR - http://hdfvr.com/ - to record flv files from the user's webcam. The .flv files appear fine and playback at a good quality, however we are then trying to convert the to .mp4 files with ffmpeg and this is where we are hitting problems. At first we used a custom build installed by a Rackspace technician with ffmpeg experience, I believe thsi was version 0.5.2. This worked intermittently but on some larger files we would get the error "error [libx264 @ xxxxxxxx]error, non monotone timestamps xxxxx >= xxxxx" and the encoding would fail. The ffmpeg command used for this conversion was "ffmpeg -i PATH_TO_CURRENT_FILE.flv -acodec libfaac -ab 96k -ac 2 -vcodec libx264 -vpre hq -vpre ipod320 -threads 0 -crf 22 PATH_TO_NEW_FILE.mp4". Unsure of whether this was a problem with the input file or the ffmpeg installation we then fired up a cloud server and installed ffmpeg ourselves via this link - http://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide. Now we can encode the file fully to .mp4 however the audio plays back fine whilst the video stutters as if buffering on a dial up connection. Can anyone give any advice either on the first issue with the 'non monotone timestamps' or on the second issue and maintaining a good sync with the video and audio? Any advice/instruction is greatly appreciated. Many thanks Ben Halhead From h.reindl at thelounge.net Thu Mar 8 11:30:14 2012 From: h.reindl at thelounge.net (Reindl Harald) Date: Thu, 08 Mar 2012 11:30:14 +0100 Subject: [FFmpeg-user] FLV to MP4 conversion In-Reply-To: <4F5888AA.7020102@quvex.com> References: <4F5888AA.7020102@quvex.com> Message-ID: <4F588A36.4020701@thelounge.net> Am 08.03.2012 11:23, schrieb Ben Halhead: > At first we used a custom build installed by a Rackspace technician with > ffmpeg experience, I believe this was version 0.5.2 if you are using really 0.5.2 dianostic has no value because this version is outdated for many years now, there where thousands of bugfixes and imporvements from 0.5 to currently supported 0.7-0.10 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From cehoyos at ag.or.at Thu Mar 8 11:45:45 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 8 Mar 2012 10:45:45 +0000 (UTC) Subject: [FFmpeg-user] FLV to MP4 conversion References: <4F5888AA.7020102@quvex.com> Message-ID: Ben Halhead quvex.com> writes: > Now we can encode the file fully to .mp4 however the audio > plays back fine whilst the video stutters as if buffering > on a dial up connection. Command line and complete, uncut console output missing. Which application plays the video stuttering? vlc, mplayer, ffplay? Carl Eugen From cehoyos at ag.or.at Thu Mar 8 11:50:54 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 8 Mar 2012 10:50:54 +0000 (UTC) Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <4F3C161A.10701@alcatel-lucent.com> <20120222044418.GI1998@mrvideo.vidiot.com> <4F44FB46.1070506@alcatel-lucent.com> Message-ID: Mike Scheutzow alcatel-lucent.com> writes: > As far as I know, MPEG stream type 0x83 is not unique. Different systems > use it for different purposes. In some situations, it indicates a > Meridian Lossless Packing (MLP) audio stream. Could you provide such a sample? Carl Eugen From cehoyos at ag.or.at Thu Mar 8 11:53:22 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 8 Mar 2012 10:53:22 +0000 (UTC) Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> Message-ID: Mike Brown mrvideo.vidiot.com> writes: > I have a D-VHS deck that will firewire the analog video to > MPEG-2 video conversion to my computer Could you add some information about the D-VHS deck? (If we add decoding support for this supposedly proprietary way of saving S302M in mpeg-ts, we at least need to add information about which hardware exactly is producing the stream.) Carl Eugen From ben.halhead at quvex.com Thu Mar 8 12:06:28 2012 From: ben.halhead at quvex.com (Ben Halhead) Date: Thu, 08 Mar 2012 11:06:28 +0000 Subject: [FFmpeg-user] FLV to MP4 conversion In-Reply-To: <4F588A36.4020701@thelounge.net> References: <4F5888AA.7020102@quvex.com> <4F588A36.4020701@thelounge.net> Message-ID: <4F5892B4.2020001@quvex.com> Hi, yes this is why we followed the tutorial here - http://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide - to install the latest version from git and it is here that we are getting the conversion to work but getting terribly out of sync video and audio. Cheers On 08/03/2012 10:30, Reindl Harald wrote: > > Am 08.03.2012 11:23, schrieb Ben Halhead: >> At first we used a custom build installed by a Rackspace technician with >> ffmpeg experience, I believe this was version 0.5.2 > if you are using really 0.5.2 dianostic has no value because this > version is outdated for many years now, there where thousands > of bugfixes and imporvements from 0.5 to currently supported > 0.7-0.10 > > > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From ben.halhead at quvex.com Thu Mar 8 12:21:28 2012 From: ben.halhead at quvex.com (Ben Halhead) Date: Thu, 08 Mar 2012 11:21:28 +0000 Subject: [FFmpeg-user] FLV to MP4 conversion In-Reply-To: References: <4F5888AA.7020102@quvex.com> Message-ID: <4F589638.1030902@quvex.com> Hi Yes VLC player is the application I have viewed the video in Thanks Ben On 08/03/2012 10:45, Carl Eugen Hoyos wrote: > Ben Halhead quvex.com> writes: > >> Now we can encode the file fully to .mp4 however the audio >> plays back fine whilst the video stutters as if buffering >> on a dial up connection. > Command line and complete, uncut console output missing. > > Which application plays the video stuttering? > vlc, mplayer, ffplay? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From nichot20 at yahoo.com Thu Mar 8 12:36:58 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Thu, 08 Mar 2012 11:36:58 +0000 Subject: [FFmpeg-user] Technical specification for FFv1? In-Reply-To: <20120308105120.11554yn483yy7xug@webmail.tuwien.ac.at> References: <20120307095223.1929961ws9nn0s7b@webmail.tuwien.ac.at> <20120308041815.GA22962@kiste2> <20120308105120.11554yn483yy7xug@webmail.tuwien.ac.at> Message-ID: <4F5899DA.1080207@yahoo.com> On 08/03/12 09:51, Peter B. wrote: > Zitat von Michael Niedermayer : > >> That said, FFmpeg is open source and free software, theres no risk >> of being locked into a format that cant be converted all of a sudden >> everyone has the code to decode it ... > > ...and that's exactly one of my top-5 arguments *why* I'm supporting and > promoting Free Software and Open Formats within the archive domain! :) > > However, I'm *not* afraid of you dropping support of FFv1, or that I'm not able > to keep the code alive to open the videos. It's something else: > > It's generally about 2 things: > > a) In order to use FFv1 among video "professionals", there needs to be more than > sustainable code that "just works". > I have never tried FFv1, in the past I have used Huffman YUV, for lossless. However I am always open to new ideas. Having not tried FFV1, because I new nothing about it,I am interested to know what wrapper/format you use around the codec, and therefore what other metadata can be kept. (comments, timecode etc) -- Tim From ajaxhe at gmail.com Thu Mar 8 12:45:56 2012 From: ajaxhe at gmail.com (ajaxhe) Date: Thu, 8 Mar 2012 03:45:56 -0800 (PST) Subject: [FFmpeg-user] How to the use vf_drawtext.c in my program In-Reply-To: References: <1331035056939-4449601.post@n4.nabble.com> Message-ID: <1331207156819-4456207.post@n4.nabble.com> Thanks very much for your reply! I spend hold day to compile the ffmpeg-0.10 on windows, and still have some compile problem. I generally view the filtering_video.c file, and I am still cann't get the idea to solve my problem. I am new guy in FFmpeg field, maybe I should spend more time to learn avfilter to solve the problem, If you could provide some more specific document, I will very appreciate! sorry for my poor English. thanks again! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-to-the-use-vf-drawtext-c-in-my-program-tp4449601p4456207.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From maurice at cmdrkey.com Thu Mar 8 14:28:35 2012 From: maurice at cmdrkey.com (Maurice Randall) Date: Thu, 08 Mar 2012 08:28:35 -0500 Subject: [FFmpeg-user] FLV to MP4 conversion In-Reply-To: <4F588A36.4020701@thelounge.net> References: <4F5888AA.7020102@quvex.com> <4F588A36.4020701@thelounge.net> Message-ID: On 2012-03-08 05:30, Reindl Harald wrote: > if you are using really 0.5.2 dianostic has no value because this > version is outdated for many years now, there where thousands > of bugfixes and imporvements from 0.5 to currently supported > 0.7-0.10 This is on the ffmpeg.org site: "May 25, 2010 We have just pushed out another point release from our 0.5 release branch: FFmpeg 0.5.2." Many years is less than 2 years? If it is so outdated, why is it still being worked on? It appears that it is still supported and is at 0.5.8 now, which was updated January 12, 2012. Not to worry, though. I agree that it is outdated. But apparently there are many people out there still using it, so it must have some value. -Maurice From ch.sureshkumar.24 at gmail.com Thu Mar 8 14:46:46 2012 From: ch.sureshkumar.24 at gmail.com (suresh kumar) Date: Thu, 8 Mar 2012 19:16:46 +0530 Subject: [FFmpeg-user] FLV to MP4 conversion In-Reply-To: References: <4F5888AA.7020102@quvex.com> <4F588A36.4020701@thelounge.net> Message-ID: thanks for your reply.But i am using ffmpeg-0.9.1 version on windows xp. On Thu, Mar 8, 2012 at 6:58 PM, Maurice Randall wrote: > On 2012-03-08 05:30, Reindl Harald wrote: > >> if you are using really 0.5.2 dianostic has no value because this >> version is outdated for many years now, there where thousands >> of bugfixes and imporvements from 0.5 to currently supported >> 0.7-0.10 >> > > This is on the ffmpeg.org site: > > "May 25, 2010 We have just pushed out another point release from our 0.5 > release branch: FFmpeg 0.5.2." > > Many years is less than 2 years? If it is so outdated, why is it still > being worked on? It appears that it is still supported and is at 0.5.8 > now, which was updated January 12, 2012. > > Not to worry, though. I agree that it is outdated. But apparently there > are many people out there still using it, so it must have some value. > > -Maurice > > > > ______________________________**_________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/**listinfo/ffmpeg-user > From ch.sureshkumar.24 at gmail.com Thu Mar 8 14:48:28 2012 From: ch.sureshkumar.24 at gmail.com (suresh kumar) Date: Thu, 8 Mar 2012 19:18:28 +0530 Subject: [FFmpeg-user] FLV to MP4 conversion In-Reply-To: References: <4F5888AA.7020102@quvex.com> <4F588A36.4020701@thelounge.net> Message-ID: Just i am trying to build H264. Actually i am new bee to this. I am eager to know on this. On Thu, Mar 8, 2012 at 7:16 PM, suresh kumar wrote: > thanks for your reply.But i am using ffmpeg-0.9.1 version on windows xp. > > > > > On Thu, Mar 8, 2012 at 6:58 PM, Maurice Randall wrote: > >> On 2012-03-08 05:30, Reindl Harald wrote: >> >>> if you are using really 0.5.2 dianostic has no value because this >>> version is outdated for many years now, there where thousands >>> of bugfixes and imporvements from 0.5 to currently supported >>> 0.7-0.10 >>> >> >> This is on the ffmpeg.org site: >> >> "May 25, 2010 We have just pushed out another point release from our 0.5 >> release branch: FFmpeg 0.5.2." >> >> Many years is less than 2 years? If it is so outdated, why is it still >> being worked on? It appears that it is still supported and is at 0.5.8 >> now, which was updated January 12, 2012. >> >> Not to worry, though. I agree that it is outdated. But apparently there >> are many people out there still using it, so it must have some value. >> >> -Maurice >> >> >> >> ______________________________**_________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/**listinfo/ffmpeg-user >> > > From h.reindl at thelounge.net Thu Mar 8 14:49:12 2012 From: h.reindl at thelounge.net (Reindl Harald) Date: Thu, 08 Mar 2012 14:49:12 +0100 Subject: [FFmpeg-user] FLV to MP4 conversion In-Reply-To: References: <4F5888AA.7020102@quvex.com> <4F588A36.4020701@thelounge.net> Message-ID: <4F58B8D8.2080309@thelounge.net> Am 08.03.2012 14:28, schrieb Maurice Randall: > On 2012-03-08 05:30, Reindl Harald wrote: >> if you are using really 0.5.2 dianostic has no value because this >> version is outdated for many years now, there where thousands >> of bugfixes and imporvements from 0.5 to currently supported >> 0.7-0.10 > > This is on the ffmpeg.org site: > "May 25, 2010 We have just pushed out another point release from our 0.5 release branch: FFmpeg 0.5.2." i know that > Many years is less than 2 years? if you look how fast ffmpeg is developed yes > If it is so outdated, why is it still being worked on? It appears that it is > still supported and is at 0.5.8 now, which was updated January 12, 2012. it is not really worked on this security bugs are fixed - that was it > Not to worry, though. I agree that it is outdated. > But apparently there are many people out there still using it, > so it must have some value to support people which will not update or using things like debian with security updates - if it works for you fine, you are secure and it does what you want if it does not work for you on a functional base you are strongly recommended to update because there are reasons making new releases, most of them are new features, changes in command-lines and any changes which would break BC for legacy releases this is common sense: if you are running any outdated software and it does not work the way you want use the current version because you will find much more support from other people because most of them are using recent versions -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From pb at das-werkstatt.com Thu Mar 8 14:52:38 2012 From: pb at das-werkstatt.com (Peter B.) Date: Thu, 08 Mar 2012 14:52:38 +0100 Subject: [FFmpeg-user] Technical specification for FFv1? In-Reply-To: <4F5899DA.1080207@yahoo.com> References: <20120307095223.1929961ws9nn0s7b@webmail.tuwien.ac.at> <20120308041815.GA22962@kiste2> <20120308105120.11554yn483yy7xug@webmail.tuwien.ac.at> <4F5899DA.1080207@yahoo.com> Message-ID: <20120308145238.13865h0xycoycuxy@webmail.tuwien.ac.at> Zitat von Tim Nicholson : > I have never tried FFv1, in the past I have used Huffman YUV, for lossless. > However I am always open to new ideas. I think it greatly depends on the use case of "why lossless": For mass digitization and serious long-term archival purposes, it makes a difference whether you can: a) encode in realtime during capture b) work with your lossless format c) preserve the original video data as good as possible: preserve colorspace, use 10bit, ... Since about 2 years, computers are fast enough to handle FFv1 for all these requirements. Huffyuv does not. It only supports yuv422/rgb - and it's waaaaay bigger in filesize than FFv1. We (and NOA Audio Solutions) have done some tests while evaluating FFv1, and compared its performance and filesize to JPEG2k, as well as Huffyuv - and in most cases FFv1's filesize is equal to JPEG2k - but with amazing performance: It's currently still single-threading, but outperforms multithreading-JPEG2k implementations. Additionally, thanks to the widespread usage of FFmpeg's libav, FFv1 can be encoded/decoded with many tools out-of-the-box. On a regular office computer! Show me an archive that is actually able to work with its *lossless* (!) archive copy for production purposes - as smooth as that? ;) > Having not tried FFV1, because I new nothing about it,I am interested to know > what wrapper/format you use around the codec, and therefore what > other metadata > can be kept. (comments, timecode etc) == About the container: We're currently using AVI as container (unfortunately), due to its widespread support within applications (across operating systems and vendors). This is not the case for MOV. Let alone MXF: Ask BBC, why they had to write tools to convert MXF-to-MXF [1]. Cross vendor/tool support: in your dreams! I'd love to go for MKV (which seems like a very good container for the job, regarding its widespread, and increasing support among applications and hardware devices) - but we're not there, yet. Additionally, among "professional" video people, it's tainted with "That's what those DVD-pirates use! Bah." Sad :( == About the metadata: I know that within the archival domain, "one container to rule them all" is often propagated as a good solution - for example: MXF However, as I've been dealing with long-term archival for over 7 years now, I can say that in practice it's better to keep it "slightly" separated. Why? a) Some metadata is bound to change over time (coding history, descriptive metadata, etc) - but if its stored together with the content in one file, it's difficult to do checksum-validation of those files. b) Having to extract/embed the metadata from/into the container adds additional effort. With data quantities of medium-to-large archives, this adds a tremendous I/O effort for handling metadata. c) In practice, having a small (few kB), well-documented-and-based-on-standards XML (e.g. METS [2]) next to your actual A/V content, makes life easier. d) Long-term archival means infinite migration. That's a hard one, but it's necessary. Stuffing everything in one container sounds nice, but it's quite a gamble to find tools that will actually be able to extract-and-migrate *every* bit of data you've stored in your container. There's way more about this topic - and it's a hot topic among archivists - but that's at least an overview. Regards, Pb == References: [1] http://ingex.sourceforge.net/ [2] http://www.loc.gov/standards/mets/mets-schemadocs.html From ch.sureshkumar.24 at gmail.com Thu Mar 8 15:01:36 2012 From: ch.sureshkumar.24 at gmail.com (suresh kumar) Date: Thu, 8 Mar 2012 19:31:36 +0530 Subject: [FFmpeg-user] where to get mspdb100.dll file. Message-ID: I am tried to compile FFMPEG-0.9.1 and while compiling i get this message "This application has failed to start because mspdb100.dll was not found.Re-installing the application may fix this problem". Where i can get mspdb100.dll file. I am using in windows. From ben.halhead at quvex.com Thu Mar 8 15:05:27 2012 From: ben.halhead at quvex.com (Ben Halhead) Date: Thu, 08 Mar 2012 14:05:27 +0000 Subject: [FFmpeg-user] FLV to MP4 conversion In-Reply-To: <4F58B8D8.2080309@thelounge.net> References: <4F5888AA.7020102@quvex.com> <4F588A36.4020701@thelounge.net> <4F58B8D8.2080309@thelounge.net> Message-ID: <4F58BCA7.3090004@quvex.com> Hi Sorry to try and get this back on track but the post was originally about converting flv to mp4! We were indeed originally using 0.5.2 but I have now installed the latest version from git and still have issues, albeit different issues than those experienced with 0.5.2. The issue now is that video and audio are terribly out of sync and any help in getting them back in sync is greatly appreciated. Many thanks Ben Halhead On 08/03/2012 13:49, Reindl Harald wrote: > > Am 08.03.2012 14:28, schrieb Maurice Randall: >> On 2012-03-08 05:30, Reindl Harald wrote: >>> if you are using really 0.5.2 dianostic has no value because this >>> version is outdated for many years now, there where thousands >>> of bugfixes and imporvements from 0.5 to currently supported >>> 0.7-0.10 >> This is on the ffmpeg.org site: >> "May 25, 2010 We have just pushed out another point release from our 0.5 release branch: FFmpeg 0.5.2." > i know that > >> Many years is less than 2 years? > if you look how fast ffmpeg is developed yes > >> If it is so outdated, why is it still being worked on? It appears that it is >> still supported and is at 0.5.8 now, which was updated January 12, 2012. > it is not really worked on this > security bugs are fixed - that was it > >> Not to worry, though. I agree that it is outdated. >> But apparently there are many people out there still using it, >> so it must have some value > to support people which will not update or using things > like debian with security updates - if it works for you > fine, you are secure and it does what you want > > if it does not work for you on a functional base you > are strongly recommended to update because there are > reasons making new releases, most of them are new > features, changes in command-lines and any changes > which would break BC for legacy releases > > this is common sense: > if you are running any outdated software and it does not > work the way you want use the current version because > you will find much more support from other people because > most of them are using recent versions > > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From onemda at gmail.com Thu Mar 8 15:11:28 2012 From: onemda at gmail.com (Paul B Mahol) Date: Thu, 8 Mar 2012 14:11:28 +0000 Subject: [FFmpeg-user] Technical specification for FFv1? In-Reply-To: <20120308145238.13865h0xycoycuxy@webmail.tuwien.ac.at> References: <20120307095223.1929961ws9nn0s7b@webmail.tuwien.ac.at> <20120308041815.GA22962@kiste2> <20120308105120.11554yn483yy7xug@webmail.tuwien.ac.at> <4F5899DA.1080207@yahoo.com> <20120308145238.13865h0xycoycuxy@webmail.tuwien.ac.at> Message-ID: On 3/8/12, Peter B. wrote: > Zitat von Tim Nicholson : > >> I have never tried FFv1, in the past I have used Huffman YUV, for >> lossless. >> However I am always open to new ideas. > > I think it greatly depends on the use case of "why lossless": > For mass digitization and serious long-term archival purposes, it > makes a difference whether you can: > a) encode in realtime during capture > b) work with your lossless format > c) preserve the original video data as good as possible: preserve > colorspace, use 10bit, ... > > Since about 2 years, computers are fast enough to handle FFv1 for all > these requirements. Huffyuv does not. It only supports yuv422/rgb - > and it's waaaaay bigger in filesize than FFv1. > > We (and NOA Audio Solutions) have done some tests while evaluating > FFv1, and compared its performance and filesize to JPEG2k, as well as > Huffyuv - and in most cases FFv1's filesize is equal to JPEG2k - but > with amazing performance: It's currently still single-threading, but > outperforms multithreading-JPEG2k implementations. FFv1 support multi-threaded slice decoding/encoding for more than one year. From deepesh.basu at gmail.com Thu Mar 8 15:34:39 2012 From: deepesh.basu at gmail.com (Deepesh Basu) Date: Thu, 8 Mar 2012 20:04:39 +0530 Subject: [FFmpeg-user] where to get mspdb100.dll file. In-Reply-To: References: Message-ID: Hi, You need to install MS Visual Studio C++ Express 10/2010 along with MS Dot Net Framework 4.0 to get the file! After that, alter the values of your windows %PATH% and %CLASSPATH% variables (Environment) to include then in the -I/-L path (include-dir/path). Please note that, Windows (i.e. WINNT v5.1x and above) provides a Native MSCORLIB.DLL which sud suffice for the needs of FFMPEG. Also, while including specific combinations, we sometimes do come-across Direct-X, Windows SDK related misc dependencies, but more often than NOT, its OUR MANUAL Error in specifying something incorrectly! Hope it helps! Thnx/BR, Deepesh Basu DVB-CP/CM Solution Architect, 81671 M?nchen, Deutschland *R: 0049-(0)-89-2351-4997* On Thu, Mar 8, 2012 at 7:31 PM, suresh kumar wrote: > I am tried to compile FFMPEG-0.9.1 and while compiling i get this > message "This application has failed to start because mspdb100.dll was not > found.Re-installing the application may fix this problem". Where i can get > mspdb100.dll file. I am using in windows. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From pb at das-werkstatt.com Thu Mar 8 15:39:27 2012 From: pb at das-werkstatt.com (Peter B.) Date: Thu, 08 Mar 2012 15:39:27 +0100 Subject: [FFmpeg-user] Technical specification for FFv1? In-Reply-To: References: <20120307095223.1929961ws9nn0s7b@webmail.tuwien.ac.at> <20120308041815.GA22962@kiste2> <20120308105120.11554yn483yy7xug@webmail.tuwien.ac.at> <4F5899DA.1080207@yahoo.com> <20120308145238.13865h0xycoycuxy@webmail.tuwien.ac.at> Message-ID: <20120308153927.15043prmz1agghi7@webmail.tuwien.ac.at> Zitat von Paul B Mahol : > FFv1 support multi-threaded slice decoding/encoding for more than one year. I know. I've been the one asking Michael to implement it ;) (Thanks Michael, btw!) The last status known to me about this is, that it's "in position", but not enabled by default in the current codebase. It's on my TODO-list to thoroughly test it, though and report feedback. Unfortunately, my work schedule has caused this testing to be postponed... :( Since it's not enabled by default, and we're using ffdshow-tryouts to have it as capture codec on the clients, I'm not yet able to profit from this improvement. The benchmarks I've done with it (last year, right after it has been implemented), were really promising! :) Pb From cehoyos at ag.or.at Thu Mar 8 16:09:30 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 8 Mar 2012 15:09:30 +0000 (UTC) Subject: [FFmpeg-user] FLV to MP4 conversion References: <4F5888AA.7020102@quvex.com> <4F588A36.4020701@thelounge.net> <4F58B8D8.2080309@thelounge.net> <4F58BCA7.3090004@quvex.com> Message-ID: Ben Halhead quvex.com> writes: > The issue now is that video and audio are terribly out of > sync and any help in getting them back in sync is greatly > appreciated. Command line and complete, uncut console output missing. Does the resulting file play in-sync with MPlayer (or ffplay)? Carl Eugen From brown at mrvideo.vidiot.com Thu Mar 8 16:50:20 2012 From: brown at mrvideo.vidiot.com (Mike Brown) Date: Thu, 8 Mar 2012 09:50:20 -0600 Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck In-Reply-To: References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> Message-ID: <20120308155020.GA18290@mrvideo.vidiot.com> On Thu, Mar 08, 2012 at 10:53:22AM +0000, Carl Eugen Hoyos wrote: > Mike Brown mrvideo.vidiot.com> writes: > > > I have a D-VHS deck that will firewire the analog video to > > MPEG-2 video conversion to my computer > > Could you add some information about the D-VHS deck? > (If we add decoding support for this supposedly proprietary way > of saving S302M in mpeg-ts, we at least need to add information > about which hardware exactly is producing the stream.) Yep, there are two models that I have, one that is basically working. They are both JVC models: HM-DH30000U & HM-DH40000U It is the 40K that I am using. MB -- e-mail: vidiot at vidiot.com | vidiot at vidiot.net /~\ The ASCII 6082066843 at email.uscc.net (140 char limit) \ / Ribbon Campaign Visit - URL: http://vidiot.com/ X Against http://vidiot.net/ / \ HTML Email From phil_rhodes at rocketmail.com Thu Mar 8 17:31:51 2012 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Thu, 08 Mar 2012 16:31:51 -0000 Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck In-Reply-To: <20120308075046.GE24977@mrvideo.vidiot.com> References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <4F3C161A.10701@alcatel-lucent.com> <20120307223230.GX19913@mrvideo.vidiot.com> <20120308075046.GE24977@mrvideo.vidiot.com> Message-ID: > Please do not top-post here, it is considered rude, Carl Eugen What's rude, Carl, and completely unnecessary, is your habit of endlessly picking people up on this, even when it's obvious why it was done. Please stop. P From tevans.uk at googlemail.com Thu Mar 8 17:41:18 2012 From: tevans.uk at googlemail.com (Tom Evans) Date: Thu, 8 Mar 2012 16:41:18 +0000 Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck In-Reply-To: References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <4F3C161A.10701@alcatel-lucent.com> <20120307223230.GX19913@mrvideo.vidiot.com> <20120308075046.GE24977@mrvideo.vidiot.com> Message-ID: On Thu, Mar 8, 2012 at 4:31 PM, Phil Rhodes wrote: > >> Please do not top-post here, it is considered rude, Carl Eugen > > > What's rude, Carl, and completely unnecessary, is your habit of endlessly > picking people up on this, even when it's obvious why it was done. Please > stop. > Er, not really. If the quoted content was irrelevant, it should have been trimmed, and the reply gone below. The top post in question actually interspersed the message in between quoted sections, for no reason other than "cba to trim the post and move the cursor below". There is never an appropriate reason to top post on a list that is strictly bottom post only. Cheers Tom From nichot20 at yahoo.com Thu Mar 8 17:42:21 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Thu, 08 Mar 2012 16:42:21 +0000 Subject: [FFmpeg-user] Technical specification for FFv1? In-Reply-To: <20120308145238.13865h0xycoycuxy@webmail.tuwien.ac.at> References: <20120307095223.1929961ws9nn0s7b@webmail.tuwien.ac.at> <20120308041815.GA22962@kiste2> <20120308105120.11554yn483yy7xug@webmail.tuwien.ac.at> <4F5899DA.1080207@yahoo.com> <20120308145238.13865h0xycoycuxy@webmail.tuwien.ac.at> Message-ID: <4F58E16D.1030401@yahoo.com> On 08/03/12 13:52, Peter B. wrote: > Zitat von Tim Nicholson : > >> I have never tried FFv1, in the past I have used Huffman YUV, for lossless. >> However I am always open to new ideas. > > I think it greatly depends on the use case of "why lossless": > For mass digitization and serious long-term archival purposes, it makes a > difference whether you can: > a) encode in realtime during capture > b) work with your lossless format > c) preserve the original video data as good as possible: preserve colorspace, > use 10bit, ... > > Since about 2 years, computers are fast enough to handle FFv1 for all these > requirements. Huffyuv does not. It only supports yuv422/rgb - and it's waaaaay > bigger in filesize than FFv1. > > We (and NOA Audio Solutions) have done some tests while evaluating FFv1, and > compared its performance and filesize to JPEG2k, as well as Huffyuv - and in > most cases FFv1's filesize is equal to JPEG2k - but with amazing performance: > It's currently still single-threading, but outperforms multithreading-JPEG2k > implementations. > Very interesting. So presumably to use it you just set the codec, and pix_fmt to get the bit depth, bit rate ends up being what it is.... I'll have a try with it sometime soon, I need a good intermediate format. >[...] -- Tim From brown at mrvideo.vidiot.com Thu Mar 8 17:47:54 2012 From: brown at mrvideo.vidiot.com (Mike Brown) Date: Thu, 8 Mar 2012 10:47:54 -0600 Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck In-Reply-To: References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <4F3C161A.10701@alcatel-lucent.com> <20120307223230.GX19913@mrvideo.vidiot.com> <20120308075046.GE24977@mrvideo.vidiot.com> Message-ID: <20120308164754.GB18290@mrvideo.vidiot.com> On Thu, Mar 08, 2012 at 04:41:18PM +0000, Tom Evans wrote: > Er, not really. If the quoted content was irrelevant, it should have > been trimmed, and the reply gone below. The top post in question > actually interspersed the message in between quoted sections, for no > reason other than "cba to trim the post and move the cursor below". I was thinking of putting what I wrote above the quote line. The idea was to make it easy for readers to see the new text asking about an update to the material below. Putting it at the bottom would have required the reader to search for the new text. I was trying to do the KISS principle. Guess it didn't come across that way. MB -- e-mail: vidiot at vidiot.com | vidiot at vidiot.net /~\ The ASCII 6082066843 at email.uscc.net (140 char limit) \ / Ribbon Campaign Visit - URL: http://vidiot.com/ X Against http://vidiot.net/ / \ HTML Email From bcardarella at gmail.com Thu Mar 8 17:48:56 2012 From: bcardarella at gmail.com (Brian Cardarella) Date: Thu, 8 Mar 2012 11:48:56 -0500 Subject: [FFmpeg-user] ffmpeg and rmvb conversions In-Reply-To: References: Message-ID: So I'm just using something like: ffmpeg -i video.rmvb -qscale 5 out.mp4 - Brian On Thu, Mar 8, 2012 at 1:55 AM, Carl Eugen Hoyos wrote: > Brian Cardarella gmail.com> writes: > >> I have a rmvb (RealMedia Variable Bitrate) file that I'd like to >> convert. All of my attempts has resulted in a jittery video. > > Command line and complete, uncut ouput missing. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From cehoyos at ag.or.at Thu Mar 8 19:09:40 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 8 Mar 2012 18:09:40 +0000 (UTC) Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <20120308155020.GA18290@mrvideo.vidiot.com> Message-ID: Mike Brown mrvideo.vidiot.com> writes: > > > I have a D-VHS deck that will firewire the analog video to > > > MPEG-2 video conversion to my computer > > > > Could you add some information about the D-VHS deck? > Yep, there are two models that I have, one that is basically working. > They are both JVC models: HM-DH30000U & HM-DH40000U > > It is the 40K that I am using. Sorry, I am not sure I understand correctly: Do you mean that sound is working fine for the 30k, but currently fails for the 40k, or something else? Thank you, Carl Eugen From brown at mrvideo.vidiot.com Thu Mar 8 20:07:30 2012 From: brown at mrvideo.vidiot.com (Mike Brown) Date: Thu, 8 Mar 2012 13:07:30 -0600 Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck In-Reply-To: References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <20120308155020.GA18290@mrvideo.vidiot.com> Message-ID: <20120308190730.GC18290@mrvideo.vidiot.com> On Thu, Mar 08, 2012 at 06:09:40PM +0000, Carl Eugen Hoyos wrote: > Mike Brown mrvideo.vidiot.com> writes: > > > > > I have a D-VHS deck that will firewire the analog video to > > > > MPEG-2 video conversion to my computer > > > > > > Could you add some information about the D-VHS deck? > > > Yep, there are two models that I have, one that is basically working. > > They are both JVC models: HM-DH30000U & HM-DH40000U > > > > It is the 40K that I am using. > > Sorry, I am not sure I understand correctly: > Do you mean that sound is working fine for the 30k, but currently > fails for the 40k, or something else? No. What I meant, and should have been more clear, is that the 30K has other technical issues. AFAIK, the LPCM from both decks should be the same, as the 40K is not a major change from the 30K. MB -- e-mail: vidiot at vidiot.com | vidiot at vidiot.net /~\ The ASCII 6082066843 at email.uscc.net (140 char limit) \ / Ribbon Campaign Visit - URL: http://vidiot.com/ X Against http://vidiot.net/ / \ HTML Email From phil_rhodes at rocketmail.com Thu Mar 8 20:09:38 2012 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Thu, 08 Mar 2012 19:09:38 -0000 Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck In-Reply-To: References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <4F3C161A.10701@alcatel-lucent.com> <20120307223230.GX19913@mrvideo.vidiot.com> <20120308075046.GE24977@mrvideo.vidiot.com> Message-ID: > There is never an appropriate reason to top post on a list that is > strictly bottom post only. Says who? And that's a sincere question; one of the problems with opensource projects is finding out who's actually in charge. And I disagree anyway. P From phil_rhodes at rocketmail.com Thu Mar 8 20:10:09 2012 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Thu, 08 Mar 2012 19:10:09 -0000 Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck In-Reply-To: <20120308164754.GB18290@mrvideo.vidiot.com> References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <4F3C161A.10701@alcatel-lucent.com> <20120307223230.GX19913@mrvideo.vidiot.com> <20120308075046.GE24977@mrvideo.vidiot.com> <20120308164754.GB18290@mrvideo.vidiot.com> Message-ID: > I was trying to do the KISS principle. > > Guess it didn't come across that way. It did, you're just dealing with... well, a certain type of person. P From rhodri at kynesim.co.uk Thu Mar 8 20:18:44 2012 From: rhodri at kynesim.co.uk (Rhodri James) Date: Thu, 08 Mar 2012 19:18:44 -0000 Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck In-Reply-To: References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <4F3C161A.10701@alcatel-lucent.com> <20120307223230.GX19913@mrvideo.vidiot.com> <20120308075046.GE24977@mrvideo.vidiot.com> Message-ID: On Thu, 08 Mar 2012 19:09:38 -0000, Phil Rhodes wrote: > >> There is never an appropriate reason to top post on a list that is >> strictly bottom post only. s/never/rarely/ and I'll happily agree. > Says who? > > And that's a sincere question; one of the problems with opensource > projects is finding out who's actually in charge. In rather a lot of open source projects, "Who's in charge?" isn't a well-formed question. Open source projects don't have to have anyone in charge, so just asking the question can get you into entirely the wrong mind-set. > And I disagree anyway. A: Because it breaks the logical flow of conversation. Q: Why is top-posting less than ideal? -- Rhodri James Kynesim Ltd From wenbo828 at gmail.com Thu Mar 8 11:38:57 2012 From: wenbo828 at gmail.com (wangwenbo) Date: Thu, 8 Mar 2012 18:38:57 +0800 Subject: [FFmpeg-user] ffmpeg configure issue Message-ID: Hi all, I'm compiling the latest ffmpeg from git, when I run "./configure <--some-parameters> --enable-libass", I always got the message "ERROR: libass not found", but the lib files are in the "/usr/local/lib" for sure. I also add the lib path to "/etc/ld.so.conf" and run ldconfig many times, this error still remains. Can anyone give me some helps? -- >From *Google Mail* -------------- next part -------------- A non-text attachment was scrubbed... Name: config.log Type: application/octet-stream Size: 151693 bytes Desc: not available URL: From ch.sureshkumar.24 at gmail.com Fri Mar 9 05:19:15 2012 From: ch.sureshkumar.24 at gmail.com (suresh kumar) Date: Fri, 9 Mar 2012 09:49:15 +0530 Subject: [FFmpeg-user] where to get mspdb100.dll file. In-Reply-To: References: Message-ID: Thank you for your kind help. Hope to continue further more. On Thu, Mar 8, 2012 at 8:04 PM, Deepesh Basu wrote: > Hi, > > You need to install MS Visual Studio C++ Express 10/2010 along with MS Dot > Net Framework 4.0 to get the file! After that, alter the values of your > windows %PATH% and %CLASSPATH% variables (Environment) to include then in > the -I/-L path (include-dir/path). > > Please note that, Windows (i.e. WINNT v5.1x and above) provides a Native > MSCORLIB.DLL which sud suffice for the needs of FFMPEG. Also, while > including specific combinations, we sometimes do come-across Direct-X, > Windows SDK related misc dependencies, but more often than NOT, its OUR > MANUAL Error in specifying something incorrectly! > > Hope it helps! > > Thnx/BR, > Deepesh Basu > DVB-CP/CM Solution Architect, 81671 M?nchen, Deutschland > *R: 0049-(0)-89-2351-4997* > > On Thu, Mar 8, 2012 at 7:31 PM, suresh kumar >wrote: > > > I am tried to compile FFMPEG-0.9.1 and while compiling i get this > > message "This application has failed to start because mspdb100.dll was > not > > found.Re-installing the application may fix this problem". Where i can > get > > mspdb100.dll file. I am using in windows. > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ch.sureshkumar.24 at gmail.com Fri Mar 9 05:33:04 2012 From: ch.sureshkumar.24 at gmail.com (suresh kumar) Date: Fri, 9 Mar 2012 10:03:04 +0530 Subject: [FFmpeg-user] I would like to get library files and dll files under prefix/bin Message-ID: Hi, I am configuring for libx264 using FFMPEG on windows-xp. I got all related files. I made configuration has --enable-shared --enable-static. But I am getting avdevice-53.dll file alone.May know reason for why I am not getting my files under prefix/bin. While configure getting has "WARNING:Package-config not found,library detection may file". let me know reason for getting like this. "*LD libswresample/swresample-0.dll* *Creating library file: libswresample/libswresample.dll.a* *lib.exe /machine:i386 /def:libswresample/swresample-0.def /out:libswresample/swr* *make: execvp: lib.exe: Bad file number* *make: [libswresample/swresample-0.dll] Error 127 (ignored)"* From de.techno at gmail.com Fri Mar 9 07:14:42 2012 From: de.techno at gmail.com (dE .) Date: Fri, 09 Mar 2012 11:44:42 +0530 Subject: [FFmpeg-user] ffmpeg and rmvb conversions In-Reply-To: References: Message-ID: <4F599FD2.6030408@gmail.com> On 03/08/12 22:18, Brian Cardarella wrote: > So I'm just using something like: > > ffmpeg -i video.rmvb -qscale 5 out.mp4 > > - Brian > > On Thu, Mar 8, 2012 at 1:55 AM, Carl Eugen Hoyos wrote: >> Brian Cardarella gmail.com> writes: >> >>> I have a rmvb (RealMedia Variable Bitrate) file that I'd like to >>> convert. All of my attempts has resulted in a jittery video. >> Command line and complete, uncut ouput missing. >> >> Carl Eugen >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user ffmpeg output is missing. How about ffmpeg -i video.rmvb -r 25 -qscale 5 out.mp4 From nichot20 at yahoo.com Fri Mar 9 08:34:48 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Fri, 09 Mar 2012 07:34:48 +0000 Subject: [FFmpeg-user] ffmpeg configure issue In-Reply-To: References: Message-ID: <4F59B298.7040501@yahoo.com> On 08/03/12 10:38, wangwenbo wrote: > Hi all, > > I'm compiling the latest ffmpeg from git, when I run "./configure > <--some-parameters> --enable-libass", I always got the message "ERROR: > libass not found", but the lib files are in the "/usr/local/lib" for sure. > > I also add the lib path to "/etc/ld.so.conf" and run ldconfig many times, > this error still remains. > > Can anyone give me some helps? > Are the header files also in /usr/local/include ? -- Tim From r.harakaly at gmail.com Fri Mar 9 10:08:00 2012 From: r.harakaly at gmail.com (Robert Harakaly) Date: Fri, 09 Mar 2012 10:08:00 +0100 Subject: [FFmpeg-user] live stream transcoding problem Message-ID: <1331284080.2913.17.camel@maserati> Hello everybody, I'm new to this mailinglist. I try to transcode my IPTV live stream to VP8/webm. I use following command line: ---- $ ffmpeg-0.9/ffmpeg -i udp://@225.1.1.62:1111 -r 26 -g 26 -acodec libvorbis -ab 128k -channels 2 -threads 4 -vcodec libvpx -b:v 1200k -bt 200k -quality good -f webm http://localhost:8081/publish/bbc-hi?password=secret or $ ffmpeg-0.9/ffmpeg -i http://localhost:12333 -r 26 -g 26 -acodec libvorbis -ab 128k -channels 2 -threads 4 -vcodec libvpx -b:v 1200k -bt 200k -quality good -f webm http://localhost:8081/publish/bbc-hi?password=secret where I use vlc to "prepare" the stream: vlc -I dummy udp://@225.1.1.62:1111 :sout=#http{mux=ts,dst=:12333} My goal is to transcode the input stream into (preferably VBR) approx 1.2Mbps WBEM stream. In both cases the transcoding fails either practically immediately (after 40-80 frames) or "randomly" within 4 hours. I've never got longer functioning than 4 hours. The error message I see is (UDP source): [webm @ 0x2b72a00] Writing block at offset 21523, size 206, pts 2927, dts 2927, duration 0, flags 128 [webm @ 0x2b72a00] Starting new cluster at offset 21736 bytes, pts 2962 [udp @ 0x2aded40] circular_buffer: OVERRUN or [webm @ 0x151e900] Writing block at offset 31196, size 8423, pts 2462, dts 2462, duration 38, flags 0 [webm @ 0x151e900] Starting new cluster at offset 39626 bytes, pts 2473 in case I take http stream from vlc. In both cases, the common message is: "Starting new cluster" I use Ubuntu 11.4 with different versions of ffmpeg (0.9, 0.9.1, 0.10) downloaded from ffmpeg.org and compiled manually. I attach below the debug output from ffmpeg Thanks everybody for help Robert The debug information from ffmpeg: # ffmpeg-0.9/ffmpeg -i http://localhost:12333 -r 26 -g 26 -acodec libvorbis -ab 128k -channels 2 -threads 4 -vcodec libvpx -b:v 1200k -bt 200k -quality good -f webm -loglevel debug http://localhost:8081/publish/bbc-hi?password=secret ffmpeg version 0.9, Copyright (c) 2000-2011 the FFmpeg developers built on Jan 6 2012 17:32:08 with gcc 4.5.2 configuration: --enable-libvpx --enable-libvorbis --enable-libx264 --enable-gpl libavutil 51. 32. 0 / 51. 32. 0 libavcodec 53. 42. 0 / 53. 42. 0 libavformat 53. 24. 0 / 53. 24. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 53. 0 / 2. 53. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpegts @ 0x1e3a7a0] Format mpegts probed with size=2048 and score=100 [mpegts @ 0x1e3a7a0] stream=0 stream_type=3 pid=200 prog_reg_desc= [mpegts @ 0x1e3a7a0] stream=1 stream_type=2 pid=201 prog_reg_desc= [mp3 @ 0x1e40fe0] err{or,}_recognition separate: 1; 1 [mp3 @ 0x1e40fe0] err{or,}_recognition combined: 1; 65537 [mpeg2video @ 0x1e41ac0] err{or,}_recognition separate: 1; 1 [mpeg2video @ 0x1e41ac0] err{or,}_recognition combined: 1; 65537 [mpeg2video @ 0x1e41ac0] mpeg_decode_postinit() failure Last message repeated 2 times [mpeg2video @ 0x1e41ac0] Unsupported bit depth: 0 [mpegts @ 0x1e3a7a0] max_analyze_duration 5000000 reached at 5016000 rfps: 24.916667 0.015015 rfps: 25.000000 0.000009 Last message repeated 1 times rfps: 25.083333 0.013932 rfps: 49.916667 0.015583 rfps: 50.000000 0.000035 Last message repeated 1 times rfps: 50.083333 0.013416 [mpegts @ 0x1e3a7a0] Estimating duration from bitrate, this may be inaccurate Input #0, mpegts, from 'http://localhost:12333': Duration: N/A, start: 22274.653789, bitrate: 15256 kb/s Program 1 Stream #0:0[0x200](eng), 211, 1/90000: Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16, 256 kb/s Stream #0:1[0x201], 126, 1/90000: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 1/50, 15000 kb/s, 25.20 fps, 25 tbr, 90k tbn, 50 tbc [buffer @ 0x20f4f40] w:720 h:576 pixfmt:yuv420p tb:1/1000000 sar:64/45 sws_param: [libvpx @ 0x1e44400] err{or,}_recognition separate: 1; 1 [libvpx @ 0x1e44400] err{or,}_recognition combined: 1; 65537 [libvpx @ 0x1e44400] v0.9.7-p1 [libvpx @ 0x1e44400] [libvpx @ 0x1e44400] vpx_codec_enc_cfg [libvpx @ 0x1e44400] generic settings g_usage: 0 g_threads: 0 g_profile: 0 g_w: 320 g_h: 240 g_timebase: {1/30} g_error_resilient: 0 g_pass: 0 g_lag_in_frames: 0 [libvpx @ 0x1e44400] rate control settings rc_dropframe_thresh: 0 rc_resize_allowed: 0 rc_resize_up_thresh: 60 rc_resize_down_thresh: 30 rc_end_usage: 0 rc_twopass_stats_in: (nil)(0) rc_target_bitrate: 256 [libvpx @ 0x1e44400] quantizer settings rc_min_quantizer: 4 rc_max_quantizer: 63 [libvpx @ 0x1e44400] bitrate tolerance rc_undershoot_pct: 100 rc_overshoot_pct: 100 [libvpx @ 0x1e44400] decoder buffer model rc_buf_sz: 6000 rc_buf_initial_sz: 4000 rc_buf_optimal_sz: 5000 [libvpx @ 0x1e44400] 2 pass rate control settings rc_2pass_vbr_bias_pct: 50 rc_2pass_vbr_minsection_pct: 0 rc_2pass_vbr_maxsection_pct: 400 [libvpx @ 0x1e44400] keyframing settings kf_mode: 1 kf_min_dist: 0 kf_max_dist: 9999 [libvpx @ 0x1e44400] [libvpx @ 0x1e44400] vpx_codec_enc_cfg [libvpx @ 0x1e44400] generic settings g_usage: 0 g_threads: 4 g_profile: 0 g_w: 720 g_h: 576 g_timebase: {1/26} g_error_resilient: 0 g_pass: 0 g_lag_in_frames: 0 [libvpx @ 0x1e44400] rate control settings rc_dropframe_thresh: 0 rc_resize_allowed: 0 rc_resize_up_thresh: 60 rc_resize_down_thresh: 30 rc_end_usage: 0 rc_twopass_stats_in: (nil)(0) rc_target_bitrate: 1200 [libvpx @ 0x1e44400] quantizer settings rc_min_quantizer: 4 rc_max_quantizer: 63 [libvpx @ 0x1e44400] bitrate tolerance rc_undershoot_pct: 100 rc_overshoot_pct: 100 [libvpx @ 0x1e44400] decoder buffer model rc_buf_sz: 6000 rc_buf_initial_sz: 4000 rc_buf_optimal_sz: 5000 [libvpx @ 0x1e44400] 2 pass rate control settings rc_2pass_vbr_bias_pct: 50 rc_2pass_vbr_minsection_pct: 0 rc_2pass_vbr_maxsection_pct: 400 [libvpx @ 0x1e44400] keyframing settings kf_mode: 1 kf_min_dist: 0 kf_max_dist: 26 [libvpx @ 0x1e44400] [libvpx @ 0x1e44400] vpx_codec_control [libvpx @ 0x1e44400] VP8E_SET_CPUUSED: 3 [libvpx @ 0x1e44400] VP8E_SET_ARNR_MAXFRAMES: 0 [libvpx @ 0x1e44400] VP8E_SET_ARNR_STRENGTH: 3 [libvpx @ 0x1e44400] VP8E_SET_ARNR_TYPE: 3 [libvpx @ 0x1e44400] VP8E_SET_NOISE_SENSITIVITY: 0 [libvpx @ 0x1e44400] VP8E_SET_TOKEN_PARTITIONS: 0 [libvpx @ 0x1e44400] VP8E_SET_STATIC_THRESHOLD: 0 [libvpx @ 0x1e44400] VP8E_SET_CQ_LEVEL: 0 [libvpx @ 0x1e44400] Using deadline: 1000000 [libvorbis @ 0x214ade0] err{or,}_recognition separate: 1; 1 [libvorbis @ 0x214ade0] err{or,}_recognition combined: 1; 65537 [mp2 @ 0x1e40fe0] err{or,}_recognition separate: 1; 65537 [mp2 @ 0x1e40fe0] err{or,}_recognition combined: 1; 65537 [mpeg2video @ 0x1e41ac0] err{or,}_recognition separate: 1; 65537 [mpeg2video @ 0x1e41ac0] err{or,}_recognition combined: 1; 65537 Output #0, webm, to 'http://localhost:8081/publish/bbc-hi?password=secret': Metadata: encoder : Lavf53.24.0 Stream #0:0, 0, 1/1000: Video: vp8, yuv420p, 720x576 [SAR 64:45 DAR 16:9], 1/26, q=-1--1, 1200 kb/s, 1k tbn, 26 tbc Stream #0:1(eng), 0, 1/1000: Audio: vorbis, 48000 Hz, stereo, s16, 128 kb/s Stream mapping: Stream #0:1 -> #0:0 (mpeg2video -> libvpx) Stream #0:0 -> #0:1 (mp2 -> libvorbis) Press [q] to stop, [?] for help [mpeg2video @ 0x1e41ac0] Unsupported bit depth: 0 .... From pb at das-werkstatt.com Fri Mar 9 12:03:54 2012 From: pb at das-werkstatt.com (Peter B.) Date: Fri, 09 Mar 2012 12:03:54 +0100 Subject: [FFmpeg-user] Technical specification for FFv1? In-Reply-To: <4F58E16D.1030401@yahoo.com> References: <20120307095223.1929961ws9nn0s7b@webmail.tuwien.ac.at> <20120308041815.GA22962@kiste2> <20120308105120.11554yn483yy7xug@webmail.tuwien.ac.at> <4F5899DA.1080207@yahoo.com> <20120308145238.13865h0xycoycuxy@webmail.tuwien.ac.at> <4F58E16D.1030401@yahoo.com> Message-ID: <20120309120354.110071kqgiva6nq2@webmail.tuwien.ac.at> Zitat von Tim Nicholson : > I'll have a try with it sometime soon, I need a good intermediate format. :) Are you planning to capture directly in FFv1? If so, which OS do you use? Pb From cehoyos at ag.or.at Fri Mar 9 13:37:11 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 9 Mar 2012 12:37:11 +0000 (UTC) Subject: [FFmpeg-user] live stream transcoding problem References: <1331284080.2913.17.camel@maserati> Message-ID: Robert Harakaly gmail.com> writes: > ffmpeg version 0.9, Copyright (c) 2000-2011 the FFmpeg developers Please test current git head, it always contains less bugs and more features than any released version. (And there have been changes in the network code iirc.) Carl Eugen From cehoyos at ag.or.at Fri Mar 9 13:41:20 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 9 Mar 2012 12:41:20 +0000 (UTC) Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <4F3C161A.10701@alcatel-lucent.com> <20120222044418.GI1998@mrvideo.vidiot.com> <4F44FB46.1070506@alcatel-lucent.com> <20120222162233.GB9332@mrvideo.vidiot.com> Message-ID: Mike Brown mrvideo.vidiot.com> writes: > > With your patch, the resulting .wav file waveforms look really bad (severe > > phase reversals), so something is not right. > > > > I can't say for certain it is wrong, since I have no way to know what the > > audio is supposed to look/sound like. > > Should be 1kHz tone. It may make sense if you upload a sample with "real" audio, two different patches that both allow playback are discussed, but it is true that the 1kHz sample sounds bad in both cases. Carl Eugen From nichot20 at yahoo.com Fri Mar 9 14:49:36 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Fri, 09 Mar 2012 13:49:36 +0000 Subject: [FFmpeg-user] Technical specification for FFv1? In-Reply-To: <20120309120354.110071kqgiva6nq2@webmail.tuwien.ac.at> References: <20120307095223.1929961ws9nn0s7b@webmail.tuwien.ac.at> <20120308041815.GA22962@kiste2> <20120308105120.11554yn483yy7xug@webmail.tuwien.ac.at> <4F5899DA.1080207@yahoo.com> <20120308145238.13865h0xycoycuxy@webmail.tuwien.ac.at> <4F58E16D.1030401@yahoo.com> <20120309120354.110071kqgiva6nq2@webmail.tuwien.ac.at> Message-ID: <4F5A0A70.3020101@yahoo.com> On 09/03/12 11:03, Peter B. wrote: > Zitat von Tim Nicholson : > >> I'll have a try with it sometime soon, I need a good intermediate format. > > :) > Are you planning to capture directly in FFv1? If so, which OS do you use? No, I have some uncompressed files that I could do with an intermediate to save some of the heavy lifting... -- Tim From andrey.krieger.utkin at gmail.com Fri Mar 9 15:05:52 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Fri, 9 Mar 2012 16:05:52 +0200 Subject: [FFmpeg-user] live stream transcoding problem In-Reply-To: References: <1331284080.2913.17.camel@maserati> Message-ID: 2012/3/9 Carl Eugen Hoyos : > Robert Harakaly gmail.com> writes: > >> ffmpeg version 0.9, Copyright (c) 2000-2011 the FFmpeg developers > > Please test current git head, it always contains less bugs and > more features than any released version. > (And there have been changes in the network code iirc.) > > Carl Eugen +1. If still doesn't work - then this known bug is not fixed yet. Try appending this to your udp input URL: ?fifo_size=0 If you get bad quality video after this, increase wmem_max sysctl variable, and append sth like '&buffer_size=10000000' to input URL. -- Andrey Utkin From Donald.McLachlan at crc.ca Fri Mar 9 15:15:12 2012 From: Donald.McLachlan at crc.ca (Donald McLachlan) Date: Fri, 09 Mar 2012 09:15:12 -0500 Subject: [FFmpeg-user] ffmpeg configure issue In-Reply-To: References: Message-ID: <4F5A1070.2080004@crc.ca> It might not be able to find the include files. Maybe try using the config option "--extra-cflags=-I/usr/local/include" like I had to do to get libopenjpeg to work: crc at crc-fsmanager:~> ffmpeg ffmpeg version N-37669-gf2b20b7 Copyright (c) 2000-2012 the FFmpeg developers built on Feb 9 2012 09:03:32 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] configuration: --enable-libopenjpeg --extra-cflags=-I/usr/local/include --extra-cflags=-I/usr/local/include/openjpeg-1.5 If it really is the library it is complaining out, maybe something like "--extra-cflags=-L/usr/local/lib" will do it. On 08/03/2012 5:38 AM, wangwenbo wrote: > Hi all, > > I'm compiling the latest ffmpeg from git, when I run "./configure > <--some-parameters> --enable-libass", I always got the message "ERROR: > libass not found", but the lib files are in the "/usr/local/lib" for sure. > > I also add the lib path to "/etc/ld.so.conf" and run ldconfig many times, > this error still remains. > > Can anyone give me some helps? > > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From tevans.uk at googlemail.com Fri Mar 9 15:31:53 2012 From: tevans.uk at googlemail.com (Tom Evans) Date: Fri, 9 Mar 2012 14:31:53 +0000 Subject: [FFmpeg-user] ffmpeg configure issue In-Reply-To: <4F5A1070.2080004@crc.ca> References: <4F5A1070.2080004@crc.ca> Message-ID: On Fri, Mar 9, 2012 at 2:15 PM, Donald McLachlan wrote: > > It might not be able to find the include files. ?Maybe try using the config > option "--extra-cflags=-I/usr/local/include" like I had to do to get > libopenjpeg to work: > > ? ?crc at crc-fsmanager:~> ffmpeg > ? ?ffmpeg version N-37669-gf2b20b7 Copyright (c) 2000-2012 the FFmpeg > developers > ? ? ?built on Feb ?9 2012 09:03:32 with gcc 4.5.1 20101208 [gcc-4_5-branch > revision 167585] > ? ? ?configuration: --enable-libopenjpeg --extra-cflags=-I/usr/local/include > --extra-cflags=-I/usr/local/include/openjpeg-1.5 > > If it really is the library it is complaining out, maybe something like > "--extra-cflags=-L/usr/local/lib" will do it. > Linker options like -L or -l should go in --extra-ldflags, not --extra-cflags, which is where compilation flags (like -I) should go. Cheers Tom From cehoyos at ag.or.at Fri Mar 9 16:46:20 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 9 Mar 2012 15:46:20 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg configure issue References: <4F5A1070.2080004@crc.ca> Message-ID: Donald McLachlan crc.ca> writes: > It might not be able to find the include files. Maybe try using > the config option "--extra-cflags=-I/usr/local/include" like I > had to do to get libopenjpeg to work: As said before, the problem is that it is not clear where the libopenjpeg headers should be installed (there were two different Makefiles with two different locations in the source when I last looked), I was not successful reporting this to the libopenjpeg developers, please try! Carl Eugen From madzhugin at yandex.ru Fri Mar 9 17:16:57 2012 From: madzhugin at yandex.ru (=?utf-8?B?0JDQu9C10LrRgdCw0L3QtNGAINCc0LDQtNC20YPQs9C40L0=?=) Date: Fri, 09 Mar 2012 20:16:57 +0400 Subject: [FFmpeg-user] Fwd: problem with analyzeduration In-Reply-To: References: <5431331074978@web73.yandex.ru> Message-ID: Andrey Utkin ?????(?) ? ????? ?????? Wed, 07 Mar 2012 14:14:21 +0400: > 7 ????? 2012 ?. 1:02 ???????????? ???????? ????????? > ???????: >> Hi. >> I have faced the following problem - i have to make a photo with webcam >> and i have to do it quickly. >> But while running ffmpeg analizes input stream too long - about 2-3 sec. >> I tried to use the option "-analyzeduration 0", however, it didn't give >> any result - ffmpeg analizes input stream anyway. >> I also tried to describe input stream totally, but it didn't give any >> result again. >> >> I use the following command: >> "ffmpeg -f video4linux2,v4l2 -analyzeduration 0 -s 640x480 -qscale 0 -r >> 30 -pix_fmt yuyv422 -i /dev/video0 -f image2 -c:v mjpeg -pix_fmt >> yuvj422p -vframes 1 out.jpg" > > Try adding > -probesize 0 -fpsprobesize 0 > If i add -probesize 0 and -fpsprobesize 0, i get: ???? ?????? ???????? -probesize 0 ? -fpsprobesize 0, ?? ?? ???????: ---------------------------------------------------------------- $ ffmpeg -f video4linux2,v4l2 -analyzeduration 0 -probesize 0 -fpsprobesize 0 -s 640x480 -qscale 0 -r 30 -pix_fmt yuyv422 -i /dev/video0 -f image2 -c:v mjpeg -pix_fmt yuvj422p -vframes 1 out.jpg ... [video4linux2,v4l2 @ 0x9eb8680] Value 0.000000 for parameter 'probesize' out of range [video4linux2,v4l2 @ 0x9eb8680] Error setting option probesize to value 0. /dev/video0: Numerical result out of range ---------------------------------------------------------------- Minimal value for -probesize, which doesn't cause an error - 32. But it does not have any effect: ??????????? ???????? ??? -probesize, ??????? ?? ???????? ?????? - 32. ?? ??? ?? ???? ???????: ---------------------------------------------------------------- $ time ffmpeg -f video4linux2,v4l2 -analyzeduration 0 -probesize 32 -fpsprobesize 0 -s 640x480 -qscale 0 -r 30 -pix_fmt yuyv422 -i /dev/video0 -f image2 -c:v mjpeg -pix_fmt yuvj422p -vframes 1 out.jpg ... real 0m2.328s user 0m0.108s sys 0m0.176s ---------------------------------------------------------------- The situation is the same without -probesize: ??? -probesize ??????? ???????? ????? ??: ---------------------------------------------------------------- $ time ffmpeg -f video4linux2,v4l2 -analyzeduration 0 -fpsprobesize 0 -s 640x480 -qscale 0 -r 30 -pix_fmt yuyv422 -i /dev/video0 -f image2 -c:v mjpeg -pix_fmt yuvj422p -vframes 1 out.jpg ... real 0m2.332s user 0m0.112s sys 0m0.172s ---------------------------------------------------------------- From brown at mrvideo.vidiot.com Fri Mar 9 17:19:05 2012 From: brown at mrvideo.vidiot.com (Mike Brown) Date: Fri, 9 Mar 2012 10:19:05 -0600 Subject: [FFmpeg-user] Trouble with LPCM audio from D-VHS deck In-Reply-To: References: <20120214205140.GC5816@mrvideo.vidiot.com> <20120215013737.GK5816@mrvideo.vidiot.com> <20120215020621.GN5816@mrvideo.vidiot.com> <4F3C161A.10701@alcatel-lucent.com> <20120222044418.GI1998@mrvideo.vidiot.com> <4F44FB46.1070506@alcatel-lucent.com> <20120222162233.GB9332@mrvideo.vidiot.com> Message-ID: <20120309161905.GA28245@mrvideo.vidiot.com> On Fri, Mar 09, 2012 at 12:41:20PM +0000, Carl Eugen Hoyos wrote: > Mike Brown mrvideo.vidiot.com> writes: > > > > With your patch, the resulting .wav file waveforms look really bad (severe > > > phase reversals), so something is not right. > > > > > > I can't say for certain it is wrong, since I have no way to know what the > > > audio is supposed to look/sound like. > > > > Should be 1kHz tone. > > It may make sense if you upload a sample with "real" audio, > two different patches that both allow playback are discussed, > but it is true that the 1kHz sample sounds bad in both cases. Didn't know that 1 kHz tone wasn't real audio. Guess I'd better tell the broadcasters that :-) If you put the audio in an audio editing program and look at the waveform, you should see pure 1 kHz sinewaves (one for each channel). But, that said, I will capture some audio of an actual program and make it available for downloading. Thanks. MB -- e-mail: vidiot at vidiot.com | vidiot at vidiot.net /~\ The ASCII 6082066843 at email.uscc.net (140 char limit) \ / Ribbon Campaign Visit - URL: http://vidiot.com/ X Against http://vidiot.net/ / \ HTML Email From andrey.krieger.utkin at gmail.com Fri Mar 9 17:24:44 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Fri, 9 Mar 2012 18:24:44 +0200 Subject: [FFmpeg-user] Fwd: problem with analyzeduration In-Reply-To: References: <5431331074978@web73.yandex.ru> Message-ID: Hey devs, how do you think, should this problem be solved by writing an application that does not call avformat_find_stream_info()? -- Andrey Utkin From twilson7755 at rogers.com Sat Mar 10 16:49:14 2012 From: twilson7755 at rogers.com (TERRY WILSON) Date: Sat, 10 Mar 2012 07:49:14 -0800 (PST) Subject: [FFmpeg-user] Live streaming using ffmpeg Message-ID: <1331394554.25435.YahooMailNeo@web88606.mail.bf1.yahoo.com> I want to input a continous stream of images into FFmpeg (using a NamedPipe) and send the resulting continuous video stream out through a NamedPipe and then subsequently to an HTML5 based client application utilizing the video tag. I have the input and output mechanisms working but I am not sure about the format I should be using for the output video stream. I was going to use MP4 but I have read a couple of posts that suggest MP4 is not an appropriate format for a continuous video stream. I was hoping that someone here could clarify this for me and suggest what video format could be used for a continuous output stream that can be generated by ffmpeg and subsequently displayed by the HTML5 video tag. Note I tried an intermediate step where I simply write the output stream I received through the NamedPipe to a mp4 file. The resultant file is not recognized as a valid mp4 file. If I change my ffmpeg command to write the output directly to an mp4 file, then the resultant file does display correctly. The two files appear almost identical except that the one I wrote based on the output from the NamedPipe is about 68 bytes longer then the valid one (also when viewed in a binary editor, the files are identical except near the end)? If I try to use ffplay to display the video, it says "moov atom not found". Perhaps this is related to my first question about trying to stream mp4.? Just to be clear here are the two different ffmpeg commands I was trying: ? ffmpeg -re -f mjpeg -r 20 -sameq -i //./pipe/InputPipe -an mytest.mp4????? (resultant mp4 file works) ffmpeg -re -f mjpeg -r 20 -sameq -i //./pipe/InputPipe -an //./pipe/OutputPipe.mp4? (file generated by writing output from pipe does not work) Any suggestions on what output video format should be used and why my intermediate test doesn't work would be appreciated. ? Thanks, ? Terry From r.harakaly at gmail.com Sat Mar 10 17:09:00 2012 From: r.harakaly at gmail.com (Robert Harakaly) Date: Sat, 10 Mar 2012 17:09:00 +0100 Subject: [FFmpeg-user] live stream transcoding problem In-Reply-To: References: <1331284080.2913.17.camel@maserati> Message-ID: <1331395740.5081.35.camel@maserati> Hello Carl, Andrey Thanks both of you for the feedback. I took the latest published snapshot from the web page and compiled it. After more tests I worry the bug is not fixed yet (at least not completely) I tried all combinations with or without options sent by Andrey. While without "threads" option the command seems more stable, but when I added option threads=2 the problems restarted. In general (regardless of options used) there is a "critical" phase first 60-90 frames where many command invocation fails, always on: [webm @ 0x15fc0c0] Starting new cluster at offset 37796 bytes, pts 3564 After passing the critical phase the stream is up for ~10 minutes. Then the failures has 2 main reasons. Either: [webm @ 0x15fc0c0] Starting new cluster at offset 37796 bytes, pts 3564 or circular_buffer: OVERRUN0 size= 15103kB time=00:01:01.65 bitrate=2006.8kbits/s PES packet size mismatch0 size= 17666kB time=00:01:12.30 bitrate=2001.5kbits/s [mp2 @ 0x1953d80] incomplete frame Error while decoding stream #0:1 [mpeg2video @ 0x195a200] ac-tex damaged at 6 29 [mpeg2video @ 0x195a200] Warning MVs not available [mpeg2video @ 0x195a200] concealing 315 DC, 315 AC, 315 MV errors frame= 1805 fps= 22 q=0.0 Lsize= 17933kB time=00:01:12.77 bitrate=2018.8kbits/s video:17061kB audio:809kB global headers:4kB muxing overhead 0.329293% The second reason seems is more frequent when command running without threads option and without Andrey's options. Any idea ? Robert From cehoyos at ag.or.at Sat Mar 10 17:18:40 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 10 Mar 2012 16:18:40 +0000 (UTC) Subject: [FFmpeg-user] Live streaming using ffmpeg References: <1331394554.25435.YahooMailNeo@web88606.mail.bf1.yahoo.com> Message-ID: TERRY WILSON rogers.com> writes: > I have the input and output mechanisms working but I am > not sure about the format I should be using for the output > video stream. I was going to use MP4 but I have read a > couple of posts that suggest MP4 is not an > appropriate format for a continuous video stream. That is correct, I suspect isom (mp4) is the only container that does not work for your purpose. MPEG transport stream (ts) is the first container that comes to mind for continuous streams (it can be cut anywhere), but it is limited wrt codecs it allows. Carl Eugen From andrey.krieger.utkin at gmail.com Sat Mar 10 17:36:42 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Sat, 10 Mar 2012 18:36:42 +0200 Subject: [FFmpeg-user] Live streaming using ffmpeg In-Reply-To: References: <1331394554.25435.YahooMailNeo@web88606.mail.bf1.yahoo.com> Message-ID: 2012/3/10 Carl Eugen Hoyos : > That is correct, I suspect isom (mp4) is the only container > that does not work for your purpose. > MPEG transport stream (ts) is the first container that comes > to mind for continuous streams (it can be cut anywhere), but > it is limited wrt codecs it allows. AFAIK, MPEG TS is the most popular container used with HTTP Live Streaming protocol. It is ok for it to carry h264+aac. And HLS proto is usable with HTML5 video tag. -- Andrey Utkin From robert at theMakers.com Sat Mar 10 17:06:04 2012 From: robert at theMakers.com (Robert Reinhardt) Date: Sat, 10 Mar 2012 16:06:04 +0000 Subject: [FFmpeg-user] Live streaming using ffmpeg In-Reply-To: <1331394554.25435.YahooMailNeo@web88606.mail.bf1.yahoo.com> References: <1331394554.25435.YahooMailNeo@web88606.mail.bf1.yahoo.com> Message-ID: <2D405CD275952E49B92B7F48B3A0308A2B84BC41@nakedex.flaction.com> HTML5 video is not a ubiquitous specification. Most of my clients mean "Apple iOS" when they say "HTML5", especially with respect to video implementations. For a live stream, non-H.264 HTML5 browsers don't even have a live streaming specification (!!). Firefox, Opera only play WebM (VP8) or the older Theora codec, and even then, only progressive download/HTTP range requests work with these codecs/formats. My recommendation is to stick with H.264, using Flash Player where Flash Player is supported (which covers all of your desktop browsers), and fall back to HTML5 on iOS where HLS is supported with H.264. I've written two articles on the topic here: "The World of Pain that is HTML5 Video" http://transitioning.to/2012/01/the-world-of-pain-that-is-html5-video/ "Solving HTML5 Video Problems with Adaptive Streaming" http://transitioning.to/2012/03/solving-html5-video-problems-with-adaptive-streaming/ HTH. -Robert Robert Reinhardt The difference knowledge + experience makes | Consultant @ [theMAKERS] { work: http://www.theMakers.com } { video: http://videoRx.com } { blog: http://probablyjustme.com } ________________________________________ From: ffmpeg-user-bounces at ffmpeg.org [ffmpeg-user-bounces at ffmpeg.org] on behalf of TERRY WILSON [twilson7755 at rogers.com] Sent: Saturday, March 10, 2012 7:49 AM To: ffmpeg-user at ffmpeg.org Subject: [FFmpeg-user] Live streaming using ffmpeg I want to input a continous stream of images into FFmpeg (using a NamedPipe) and send the resulting continuous video stream out through a NamedPipe and then subsequently to an HTML5 based client application utilizing the video tag. I have the input and output mechanisms working but I am not sure about the format I should be using for the output video stream. I was going to use MP4 but I have read a couple of posts that suggest MP4 is not an appropriate format for a continuous video stream. I was hoping that someone here could clarify this for me and suggest what video format could be used for a continuous output stream that can be generated by ffmpeg and subsequently displayed by the HTML5 video tag. Note I tried an intermediate step where I simply write the output stream I received through the NamedPipe to a mp4 file. The resultant file is not recognized as a valid mp4 file. If I change my ffmpeg command to write the output directly to an mp4 file, then the resultant file does display correctly. The two files appear almost identical except that the one I wrote based on the output from the NamedPipe is about 68 bytes longer then the valid one (also when viewed in a binary editor, the files are identical except near the end) If I try to use ffplay to display the video, it says "moov atom not found". Perhaps this is related to my first question about trying to stream mp4. Just to be clear here are the two different ffmpeg commands I was trying: ffmpeg -re -f mjpeg -r 20 -sameq -i //./pipe/InputPipe -an mytest.mp4 (resultant mp4 file works) ffmpeg -re -f mjpeg -r 20 -sameq -i //./pipe/InputPipe -an //./pipe/OutputPipe.mp4 (file generated by writing output from pipe does not work) Any suggestions on what output video format should be used and why my intermediate test doesn't work would be appreciated. Thanks, Terry _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From andreas.gumm at gmx.de Sat Mar 10 17:42:55 2012 From: andreas.gumm at gmx.de (Andreas Gumm) Date: Sat, 10 Mar 2012 17:42:55 +0100 Subject: [FFmpeg-user] Live streaming using ffmpeg In-Reply-To: <2D405CD275952E49B92B7F48B3A0308A2B84BC41@nakedex.flaction.com> References: <1331394554.25435.YahooMailNeo@web88606.mail.bf1.yahoo.com> <2D405CD275952E49B92B7F48B3A0308A2B84BC41@nakedex.flaction.com> Message-ID: Just a short question, is ist possible to run ProRes encoding multithreaded? Andreas Gumm From andreas.gumm at gmx.de Sun Mar 11 02:58:21 2012 From: andreas.gumm at gmx.de (Andreas Gumm) Date: Sun, 11 Mar 2012 02:58:21 +0100 Subject: [FFmpeg-user] multi threaded ProRes encoding possible? In-Reply-To: References: Message-ID: Just a short question, is ist possible to run ProRes encoding multithreaded? Andreas Gumm From gerilsenem at hotmail.com Sun Mar 11 11:52:12 2012 From: gerilsenem at hotmail.com (Geril Senem) Date: Sun, 11 Mar 2012 12:52:12 +0200 Subject: [FFmpeg-user] =?windows-1256?q?Real_System_Make_Money_Easy_2012_T?= =?windows-1256?b?cmVuZP7+/v7+/g==?= Message-ID: Dear Friend; Thousands of people from all over the world are earning substantial incomes from home just by completing simple online surveys. This new system realy work!!! Your First Requirements: 1-) A computer or laptop. 2-) Access to the internet. 3-) A few extra minutes of your time. If you have this go to url and star win now. This is first section. http://tinyurl.com/6u2dyuw So get really excited... From oussama.stiti at gmail.com Sun Mar 11 14:29:46 2012 From: oussama.stiti at gmail.com (Oussama Stiti) Date: Sun, 11 Mar 2012 22:29:46 +0900 Subject: [FFmpeg-user] How to detect distorded frames with ffmpeg In-Reply-To: References: Message-ID: Hello, I'm beginner with ffmpeg, and i'm trying to detect defective frames from a video, and to extract from them important informations like slice #, frame ID, frame type, referenced frames... Please if someone knows how to do this, it would be very helpful for me. Kind Regards Ps: I'm running on Ubuntu 12.04 LTS -- *Oussama Stiti* From cehoyos at ag.or.at Sun Mar 11 16:41:06 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 11 Mar 2012 15:41:06 +0000 (UTC) Subject: [FFmpeg-user] How to detect distorded frames with ffmpeg References: Message-ID: Oussama Stiti gmail.com> writes: > Thank you for responding, actually, what i mean by "distorded frames" is a > corrupted, or a missing frame. I'm working on an analyzer of video quality > in real time, and i have to know exactly which frames are defective. Could > ffmpeg, provide me with such information ? Afaict, FFmpeg warns loudly when it encounters corrupted frames. If a frame is missing and this can be detected (there are intra-only codecs in raw containers where I believe this is not easily possible), FFmpeg also outputs a warning, often more than one warning for a missing frame. Do you have an example where you see no warnings although you know frames are corrupted / missing? Carl Eugen From juvalgot86 at gmail.com Sun Mar 11 13:46:48 2012 From: juvalgot86 at gmail.com (Junior Andre Valdivia Ramos) Date: Sun, 11 Mar 2012 12:46:48 +0000 Subject: [FFmpeg-user] error compile ffmpeg in linux - wifislax Message-ID: wifislax ffmpeg # git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg Cloning into 'ffmpeg'... remote: Counting objects: 210869, done. remote: Compressing objects: 100% (46415/46415), done. remote: Total 210869 (delta 165591), reused 209012 (delta 164113) Receiving objects: 100% (210869/210869), 54.85 MiB | 195 KiB/s, done. Resolving deltas: 100% (165591/165591), done. wifislax ffmpeg # ./configure --enable-static --enable-gpl --enable-version3 --enable-nonfree --enable-avisynth --enable-bzlib --enable-frei0r --enable-gnutls --enable-libaacplus --enable-libass --enable-libbluray --enable-libcelt --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopencv --enable-libcdio --enable-libdc1394 --enable-libdirac --enable-libfaac --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libnut --enable-libopenjpeg --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libstagefright-h264 --enable-libtheora --enable-libutvideo --enable-libv4l2 --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-openal --enable-openssl --enable-zlib ERROR: vfw32 not found If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solving the problem. and wifislax ffmpeg # ./configure --enable-static --enable-gpl --enable-libxvid ERROR: libxvid not found If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solving the problem. -------------- next part -------------- wifislax ffmpeg # dir -r * version.sh* README MAINTAINERS library.mak ffserver.c ffplay.c Doxyfile COPYING.LGPLv3 COPYING.GPLv3 configure* config.fate cmdutils.h cmdutils.c RELEASE Makefile LICENSE INSTALL ffprobe.c ffmpeg.c CREDITS COPYING.LGPLv2.1 COPYING.GPLv2 config.log common.mak cmdutils_common_opts.h Changelog tools: unwrap-diff* qt-faststart.c pktdumper.c lavfi-showfiltfmts.c graph2dot.c enum_options.c clean-diff* aviocat.c trasher.c probetest.c patcheck* ismindex.c ffeval.c cws2fws.c build_libstagefright tests: videogen.c regression-funcs.sh* Makefile lavfi-regression.sh* ffserver.conf fate.sh* fate/ base64.c tiny_psnr.c ref/ lena.pnm ffserver-regression.sh* fate-valgrind.supp fate-run.sh* copycooker.sh* audiogen.c rotozoom.c md5.sh lavf-regression.sh* ffserver.regression.ref fate-update.sh* fate_config.sh.template codec-regression.sh* asynth1.sw presets: libx264-ipod640.ffpreset libx264-ipod320.ffpreset libvpx-720p.ffpreset libvpx-720p50_60.ffpreset libvpx-360p.ffpreset libvpx-1080p.ffpreset libvpx-1080p50_60.ffpreset mt-work: yuvcmp.c valgrind-check.sh todo.txt test.sh raw.sh mplayer.diff email.sh libswscale: yuv2rgb.c utils.c swscale-test.c swscale.h sparc/ rgb2rgb.h ppc/ options.c libswscale.v colorspace-test.c x86/ swscale_unscaled.c swscale_internal.h swscale.c rgb2rgb_template.c rgb2rgb.c output.c Makefile input.c bfin/ libswresample: swresample_test.c swresample_internal.h swresample.h swresample.c resample.c rematrix_template.c rematrix.c Makefile libswresample.v audioconvert.h audioconvert.c libpostproc: postprocess_template.c postprocess_internal.h postprocess.h postprocess.c postprocess_altivec_template.c Makefile libpostproc.v libavutil: x86_cpu.h timecode.h samplefmt.c pixfmt.h opt.c Makefile libavutil.v intfloat.h fifo.h des.h colorspace.h avr32/ adler32.h x86/ timecode.c rc4.h pixdesc.h mips/ lzo.h lfg.h internal.h fifo.c des.c bswap.h avassert.h adler32.c utils.c softfloat.h rc4.c pixdesc.c mem.h lzo.c lfg.c integer.h eval.h crc.h bfin/ audioconvert.h tree.h softfloat.c rational.h pca.h mem.c log.h inverse.c integer.c eval.c crc_data.h base64.h audioconvert.c tree.c sha.h rational.c pca.c md5.h log.c intreadwrite.h imgutils.h error.h crc.c base64.c attributes.h tomi/ sha.c random_seed.h parseutils.h md5.c lls.h intmath.h imgutils.c error.c cpu.h avutil.h arm/ timestamp.h sh4/ random_seed.c parseutils.c mathematics.h lls.c intfloat_readwrite.h file.h dict.h cpu.c avstring.h aes.h timer.h samplefmt.h ppc/ opt.h mathematics.c libm.h intfloat_readwrite.c file.c dict.c common.h avstring.c aes.c libavformat: yuv4mpeg.c swfenc.c rtpenc_xiph.c rsoenc.c oggparsetheora.c mpc.c latmenc.c gsmdec.c cutils.c au.c yop.c swfdec.c rtpenc_vp8.c rsodec.c oggparsespeex.c mpc8.c jvdec.c gopher.c crypto.c assenc.c xwma.c srtdec.c rtpenc_mpv.c rso.c oggparseskeleton.c mp3enc.c ivfenc.c gif.c crcenc.c assdec.c xmv.c spdif.h rtpenc_latm.c rpl.c oggparseogm.c mp3dec.c ivfdec.c g729dec.c concat.c asf.h xa.c spdifenc.c rtpenc_h264.c rm.h oggparseflac.c movenchint.c iv8.c g723_1.c cdxl.c asfenc.c wv.c spdifdec.c rtpenc_h263_rfc2190.c rmenc.c oggparsedirac.c movenc.h iss.c framehash.c cdg.c asfdec.c wtv.h spdif.c rtpenc_h263.c rmdec.c oggparsecelt.c movenc.c isom.h framecrcenc.c cavsvideodec.c asfcrypt.h wtvenc.c sox.h rtpenc.h rm.c oggenc.c mov_chan.h isom.c flv.h caf.h asfcrypt.c wtvdec.c soxenc.c rtpenc_chain.h rl2.c oggdec.h mov_chan.c ipmovie.c flvenc.c cafenc.c asf.c wtv.c soxdec.c rtpenc_chain.c riff.h oggdec.c mov.c internal.h flvdec.c cafdec.c apetag.h westwood_vqa.c sol.c rtpenc.c riff.c nuv.c mmst.c ingenientdec.c flic.c caf.c apetag.c westwood_aud.c smjpeg.h rtpenc_amr.c rdt.h nut.h mmsh.c img2enc.c flacenc_header.c cache.c ape.c wc3movie.c smjpegenc.c rtpenc_aac.c rdt.c nutenc.c mms.h img2dec.c flacenc.h c93.c apc.c wav.c smjpegdec.c rtpdec_xiph.c rawvideodec.c nutdec.c mms.c img2.c flacenc.c bmv.c anm.c vqf.c smjpeg.c rtpdec_vp8.c rawenc.h nut.c mmf.c iff.c flacdec.c bluray.c amr.c vorbiscomment.h smacker.c rtpdec_svq3.c rawenc.c nullenc.c mm.c idroqenc.c filmstripenc.c bit.c allformats.c vorbiscomment.c siff.c rtpdec_qt.c rawdec.h nsvdec.c mkvtimestamp_v2.c idroqdec.c filmstripdec.c bintext.c aiff.h voc.h sierravmd.c rtpdec_qdm2.c rawdec.c network.h microdvdenc.c idcin.c file.c bink.c aiffenc.c vocenc.c segment.c rtpdec_qcelp.c r3d.c network.c microdvddec.c id3v2.h ffm.h bfi.c aiffdec.c vocdec.c segafilm.c rtpdec_mpeg4.c qtpalette.h ncdec.c metadata.h id3v2enc.c ffmeta.h bethsoftvid.c aea.c voc.c seek-test.c rtpdec_latm.c qcp.c mxg.c metadata.c id3v2.c ffmetaenc.c avs.c adxdec.c version.h seek.h rtpdec_h264.c pva.c mxf.h md5proto.c id3v1.h ffmetadec.c avlanguage.h adts.h vc1testenc.c seek.c rtpdec_h263_rfc2190.c psxstr.c mxfenc.c md5enc.c id3v1.c ffmenc.c avlanguage.c adtsenc.c vc1test.c sdp.c rtpdec_h263.c pmpdec.c mxfdec.c matroska.h icodec.c ffmdec.c avisynth.c act.c utils.c sbgdec.c rtpdec.h pcm.h mxf.c matroskaenc.c http.h electronicarts.c avio_internal.h ac3dec.c url.h sauce.h rtpdec_g726.c pcmenc.c mvi.c matroskadec.c http.c eacdata.c avio.h aacdec.c udp.c sauce.c rtpdec_formats.h pcmdec.c mtv.c matroska.c httpauth.h dxa.c avio.c a64.c txd.c sapenc.c rtpdec.c pcm.c msnwc_tcp.c Makefile httpauth.c dv.h aviobuf.c 4xm.c tty.c sapdec.c rtpdec_asf.c os_support.h mpjpeg.c m4vdec.c hlsproto.c dvenc.c avi.h tta.c rtsp.h rtpdec_amr.c os_support.c mpegvideodec.c lxfdec.c hls.c dv.c avienc.c tmv.c rtspenc.c rtp.c options.c mpegts.h loasdec.c h264dec.c dtsdec.c avidec.c tls.c rtspdec.c rtmpproto.c oma.h mpegtsenc.c lmlm4.c h263dec.c dsicin.c avformat.h tiertexseq.c rtspcodes.h rtmppkt.h omaenc.c mpegts.c librtmp.c h261dec.c dnxhddec.c avc.h thp.c rtsp.c rtmppkt.c omadec.c mpeg.h libnut.c gxf.h diracdec.c avc.c tcp.c rtpproto.c rtmp.h oma.c mpegenc.c libmodplug.c gxfenc.c dfa.c audiointerleave.h swf.h rtp.h rso.h oggparsevorbis.c mpeg.c libavformat.v gxf.c daud.c audiointerleave.c libavfilter: yadif.h vsrc_buffer.c vf_split.c vf_pad.c vf_frei0r.c vf_cropdetect.c transform.h graphdump.c avfiltergraph.h asink_anullsink.c af_ashowinfo.c x86/ vsink_nullsink.c vf_slicify.c vf_overlay.c vf_format.c vf_crop.c transform.c gradfun.h avfiltergraph.c allfilters.c af_aresample.c vsrc_testsrc.c vf_yadif.c vf_showinfo.c vf_null.c vf_fifo.c vf_copy.c src_movie.c formats.c avfilter.c all_channel_layouts.h af_anull.c vsrc_mptestsrc.c vf_vflip.c vf_settb.c vf_mp.c vf_fieldorder.c vf_boxblur.c sink_buffer.c drawutils.h avcodec.h af_volume.c af_amerge.c vsrc_mandelbrot.c vf_unsharp.c vf_setpts.c vf_lut.c vf_fade.c vf_blackframe.c Makefile drawutils.c avcodec.c af_silencedetect.c af_aformat.c vsrc_life.c vf_transpose.c vf_setfield.c vf_libopencv.c vf_drawtext.c vf_blackdetect.c libmpcodecs/ defaults.c asrc_anullsrc.c af_pan.c af_aconvert.c vsrc_color.c vf_tinterlace.c vf_select.c vf_hqdn3d.c vf_drawbox.c vf_ass.c libavfilter.v buffersrc.h asrc_aevalsrc.c af_earwax.c vsrc_cellauto.c vf_thumbnail.c vf_scale.c vf_hflip.c vf_deshake.c vf_aspect.c internal.h buffersink.h asrc_abuffer.h af_astreamsync.c vsrc_buffer.h vf_swapuv.c vf_pixdesctest.c vf_gradfun.c vf_delogo.c version.h graphparser.c avfilter.h asrc_abuffer.c af_asplit.c libavdevice: x11grab.c timefilter.h sndio_common.h oss_audio.c libcdio.c fbdev.c dshow.h dshow_common.c avdevice.c alsa-audio-common.c vfwcap.c timefilter.c sndio_common.c openal-dec.c libavdevice.v dv1394.h dshow_filter.c dshow.c alsa-audio.h alldevices.c v4l.c sndio_enc.c sdl.c Makefile lavfi.c dv1394.c dshow_enumpins.c bktr.c alsa-audio-enc.c v4l2.c sndio_dec.c pulse.c libdc1394.c jack_audio.c dshow_pin.c dshow_enummediatypes.c avdevice.h alsa-audio-dec.c libavcodec: zmbvenc.c v308enc.c roqvideodec.c mpegvideo_enc.c libspeexenc.c h263dec.c dvbsub.c atrac.h zmbv.c v308dec.c roqvideo.c mpegvideo_common.h libspeexdec.c h263data.h dump_extradata_bsf.c atrac.c yuv4enc.c v210x.c roqaudioenc.c mpegvideo.c libschroedinger.h h263.c dsputil_template.c atrac3data.h yuv4dec.c v210enc.c rl.h mpegaudio_tablegen.h libschroedingerenc.c h261_parser.c dsputil.h atrac3.c yop.c v210dec.h rle.h mpegaudio_tablegen.c libschroedingerdec.c h261.h dsputil.c atrac1data.h y41penc.c v210dec.c rle.c mpegaudiotab.h libschroedinger.c h261enc.c dsicinav.c atrac1.c y41pdec.c utvideo.c rl2.c mpegaudio_parser.c libopenjpegenc.c h261dec.c dpxenc.c asv1.c xxan.c utils.c resample.c mpegaudio.h libopenjpegdec.c h261data.h dpx.c ass_split.h xwd.h unary.h resample2.c mpegaudioenc.c libopencore-amr.c h261.c dpcm.c ass_split.c xwdenc.c ulti_cb.h remove_extradata_bsf.c mpegaudiodsp_template.c libmp3lame.c gsm_parser.c dnxhd_parser.c ass.h xwddec.c ulti.c rectangle.h mpegaudiodsp.h libgsm.c gsm.h dnxhdenc.h assenc.c xvmc_internal.h txd.c rdft.h mpegaudiodsp_float.c libfaac.c gsmdec_template.c dnxhdenc.c assdec.c xvmc.h twinvq_data.h rdft.c mpegaudiodsp_fixed.c libdirac_libschro.h gsmdec_data.h dnxhddec.c ass.c xsubenc.c twinvq.c raw.h mpegaudiodsp.c libdirac_libschro.c gsmdec_data.c dnxhddata.h arm/ xsubdec.c tta.c rawenc.c mpegaudiodectab.h libdirac.h gsmdec.c dnxhddata.c apedec.c xl.c tscc.c rawdec.c mpegaudiodecheader.h libdiracdec.c golomb-test.c dirac_parser.c ansi.c xiph.h truespeech_data.h raw.c mpegaudiodecheader.c libcelt_dec.c golomb.h dirac.h anm.c xiph.c truespeech.c ratecontrol.h mpegaudiodec_float.c libavcodec.v golomb.c diracdsp.h amrwbdec.c xan.c truemotion2.c ratecontrol.c mpegaudiodec.c libaacplus.c gifdec.c diracdsp.c amrwbdata.h x86/ truemotion1data.h rangecoder.h mpegaudiodata.h lcl.h gif.c diracdec.c amrnbdec.c ws-snd1.c truemotion1.c rangecoder.c mpegaudiodata.c lclenc.c get_bits.h dirac.c amrnbdata.h wnv1.c tmv.c ra288.h mpegaudio.c lcldec.c g729postfilter.h dirac_arith.h amr.h wmv2.h timecode.h ra288.c mpeg4video_parser.h latm_parser.c g729postfilter.c dirac_arith.c alsdec.c wmv2enc.c timecode.c ra144.h mpeg4video_parser.c lagarithrac.h g729.h dfa.c alpha/ wmv2dec.c tiff.h ra144enc.c mpeg4video.h lagarithrac.c g729dec.c dct-test.c allcodecs.c wmv2.c tiffenc.c ra144dec.c mpeg4videoenc.c lagarith.c g729data.h dctref.h alacenc.c wmavoice_data.h tiff.c ra144.c mpeg4videodec.c kmvc.c g726.c dctref.c alac.c wmavoice.c tiertexseqv.c r210enc.c mpeg4video.c kgv1dec.c g723_1_data.h dct.h adx_parser.c wmaprodec.c thread.h r210dec.c mpeg4data.h kbdwin.h g723_1.c dct.c adx.h wmaprodata.h targa.h qtrleenc.c mpeg4audio.h kbdwin.c g722.h dct32.h adxenc.c wmalosslessdec.c targaenc.c qtrle.c mpeg4audio.c jvdec.c g722enc.c dct32_float.c adxdec.c wma.h targa.c qpeg.c mpeg12.h jrevdct.c g722dec.c dct32_fixed.c adx.c wmaenc.c tableprint.h qdrw.c mpeg12enc.c jpegls.h g722.c dct32.c adpcm.h wmadec.c synth_filter.h qdm2_tablegen.h mpeg12decdata.h jpeglsenc.c frwu.c dca_parser.h adpcmenc.c wmadata.h synth_filter.c qdm2_tablegen.c mpeg12data.h jpeglsdec.h fraps.c dca_parser.c adpcm_data.h wma_common.h svq3.c qdm2data.h mpeg12data.c jpeglsdec.c fmtconvert.h dcahuff.h adpcm_data.c wma_common.c svq1_vlc.h qdm2.c mpeg12.c jpegls.c fmtconvert.c dca.h adpcm.c wma.c svq1.h qcelpdec.c mpc.h jfdctint_template.c flv.h dcaenc.h acelp_vectors.h wavpack.c svq1enc_cb.h qcelpdata.h mpcdata.h jfdctint.c flvenc.c dcaenc.c acelp_vectors.c w32pthreads.h svq1enc.c put_bits.h mpc.c jfdctfst.c flvdec.c dcadsp.h acelp_pitch_delay.h vqavideo.c svq1dec.c ptx.c mpc8huff.h j2k.h flicvideo.c dcadsp.c acelp_pitch_delay.c vp8_parser.c svq1_cb.h pthread.c mpc8data.h j2kenc.c flashsvenc.c dcadata.h acelp_filters.h vp8.h svq1.c psymodel.h mpc8.c j2k_dwt.h flashsv.c dca.c acelp_filters.c vp8dsp.h sunrast.h psymodel.c mpc7data.h j2k_dwt.c flashsv2enc.c cyuv.c ac3tab.h vp8dsp.c sunrastenc.c proresenc_kostya.c mpc7.c j2kdec.c flac_parser.c cscd.c ac3tab.c vp8data.h sunrast.c proresenc_anatoliy.c mp3_header_decompress_bsf.c j2k.c flac.h crystalhd.c ac3_parser.h vp8.c srtenc.c proresdsp.h mp3_header_compress_bsf.c ivi_dsp.h flacenc.c cos_tablegen.c ac3_parser.c vp6dsp.c srtdec.c proresdsp.c movsub_bsf.c ivi_dsp.c flacdec.c cookdata.h ac3.h vp6data.h sparc/ proresdec_lgpl.c motion-test.c ivi_common.h flacdata.h cook.c ac3enc_template.c vp6.c sp5x.h proresdec.h motionpixels_tablegen.h ivi_common.c flacdata.c codec_names.sh* ac3enc_opts_template.c vp5data.h sp5xdec.c proresdec2.c motionpixels_tablegen.c ituh263enc.c flac.c cljr.c ac3enc.h vp5.c sonic.c proresdata.h motionpixels.c ituh263dec.c ffwavesynth.c cinepak.c ac3enc_float.c vp56rac.c snow.h proresdata.c motion_est_template.c inverse.c ffv1.c chomp_bsf.c ac3enc_fixed.c vp56.h snowenc.c ppc/ motion_est.c intrax8huf.h fft-test.c cga_data.h ac3enc.c vp56dsp.h snowdec.c pnm_parser.c mmvideo.c intrax8.h fft-internal.h cga_data.c ac3dsp.h vp56dsp.c snowdata.h pnm.h mlp_parser.h intrax8dsp.c fft.h celp_math.h ac3dsp.c vp56data.h snow.c pnmenc.c mlp_parser.c intrax8.c fft_float.c celp_math.c ac3dec.h vp56data.c smc.c pnmdec.c mlp.h interplayvideo.c fft-fixed-test.c celp_filters.h ac3dec_data.h vp56.c smacker.c pnm.c mlpdsp.c internal.h fft_fixed.c celp_filters.c ac3dec_data.c vp3_parser.c sipr.h png.h mlpdec.c intelh263dec.c fft.c cdxl.c ac3dec.c vp3dsp.c siprdata.h pngenc.c mlp.c indeo5data.h faxcompr.h cdgraphics.c ac3.c vp3data.h sipr.c pngdsp.h mjpeg_parser.c indeo5.c faxcompr.c cbrt_tablegen.h aasc.c vp3.c sipr16kdata.h pngdsp.c mjpeg.h indeo4data.h faanidct.h cbrt_tablegen.c aandcttab.h vorbis_parser.h sipr16k.c pngdec.c mjpegenc.h indeo4.c faanidct.c cavs_parser.c aandcttab.c vorbis_parser.c sinewin_tablegen.h png.c mjpegenc.c indeo3data.h faandct.h cavs.h aac_tablegen.h vorbis.h sinewin_tablegen.c pictordec.c mjpegdec.h indeo3.c faandct.c cavsdsp.h aac_tablegen_decl.h vorbis_enc_data.h sinewin.h pgssubdec.c mjpegdec.c indeo2data.h escape130.c cavsdsp.c aac_tablegen.c vorbisenc.c sinewin.c pcxenc.c mjpeg.c indeo2.c escape124.c cavsdec.c aactab.h vorbisdec.c simple_idct_template.c pcx.c mjpegbdec.c imx_dump_header_bsf.c error_resilience.c cavsdata.h aactab.c vorbis_data.c simple_idct.h pcm_tablegen.h mjpega_dump_header_bsf.c imgconvert.h elbg.h cavs.c aacsbr.h vorbis.c simple_idct.c pcm_tablegen.c mjpeg2jpeg_bsf.c imgconvert.c elbg.c cabac.h aacsbrdata.h vmnc.c shorten.c pcm-mpeg.c mips/ imcdata.h eatqi.c cabac_functions.h aacsbr.c vmdav.c sh4/ pcm.c mimic.c imc.c eatgv.c cabac.c aacpsy.h version.h sgi.h parser.h mdec.c iirfilter.h eatgq.c c93.c aacpsy.c vdpau_internal.h sgienc.c parser.c mdct_float.c iirfilter.c eamad.c bytestream.h aacps_tablegen.h vdpau.h sgidec.c pamenc.c mdct_fixed.c iff.c eaidct.c bmv.c aacps_tablegen.c vdpau.c sbr.h os2threads.h mdct.c idcinvideo.c eacmv.c bmp.h aacps.h vda_internal.h sbrdsp.h options.c mathops.h huffyuv.c eac3enc.h bmpenc.c aacpsdata.c vda_h264.c sbrdsp.c nuv.c Makefile huffman.h eac3enc.c bmp.c aacps.c vda.h s3tc.h noise_bsf.c mace.c huffman.c eac3dec.c bitstream_filter.c aac_parser.c vda.c s3tc.c nellymoser.h lzw.h h264_sei.c eac3_data.h bitstream.c aac.h vcr1.c s302m.c nellymoserenc.c lzwenc.c h264_refs.c eac3_data.c bit_depth_template.c aacenc.h vc1_parser.c rv40vlc2.h nellymoserdec.c lzw.c h264_ps.c dxva2_vc1.c bintext.h aacenc.c vc1.h rv40dsp.c nellymoser.c lsp.h h264pred_template.c dxva2_mpeg2.c bintext.c aacdectab.h vc1dsp.h rv40data.h mxpegdec.c lsp.c h264pred.h dxva2_internal.h binkdsp.h aacdec.c vc1dsp.c rv40.c msvideo1enc.c lpc.h h264pred.c dxva2_h264.c binkdsp.c aaccoder.c vc1dec.c rv34vlc.h msvideo1.c lpc.c h264_parser.c dxva2.h binkdata.h aac_adtstoasc_bsf.c vc1data.h rv34_parser.c msrledec.h loco.c h264_mvpred.h dxva2.c bink.c aacadtsdec.h vc1data.c rv34.h msrledec.c ljpegenc.c h264_mp4toannexb_bsf.c dxtory.c binkaudio.c aacadtsdec.c vc1.c rv34dsp.h msrle.c libxvid_rc.c h264_loopfilter.c dxa.c bgmc.h aac_ac3_parser.h vc1acdata.h rv34dsp.c msmpeg4.h libxvid_internal.h h264idct_template.c dwt.h bgmc.c aac_ac3_parser.c vble.c rv34data.h msmpeg4enc.c libxvidff.c h264idct.c dwt.c bfin/ a64tables.h vb.c rv34.c msmpeg4data.h libxavs.c h264.h dv_vlc_data.h bfi.c a64multienc.c vaapi_vc1.c rv30dsp.c msmpeg4data.c libx264.c h264dsp_template.c dv_tablegen.h bethsoftvideo.h a64enc.h vaapi_mpeg4.c rv30data.h msmpeg4.c libvpxenc.c h264dsp.h dv_tablegen.c bethsoftvideo.c a64colors.h vaapi_mpeg2.c rv30.c msgsmdec.h libvpxdec.c h264dsp.c dvquant.h avs.c 8svx.c vaapi_internal.h rv20enc.c msgsmdec.c libvorbis.c h264_direct.c dvdsub_parser.c avr32/ 8bps.c vaapi_h264.c rv10enc.c mqc.h libvo-amrwbenc.c h264data.h dvdsubenc.c avpacket.c 4xm.c vaapi.h rv10.c mqcenc.c libvo-aacenc.c h264_cavlc.c dvdsubdec.c avfft.h vaapi.c rtjpeg.h mqcdec.c libutvideo.h h264_cabac.c dvdata.h avfft.c v410enc.c rtjpeg.c mqc.c libutvideoenc.cpp h264.c dvdata.c avcodec.h v410dec.c rpza.c mpegvideo_xvmc.c libutvideodec.cpp h263_parser.h dv.c aura.c v408enc.c roqvideo.h mpegvideo_parser.c libtheoraenc.c h263_parser.c dvbsub_parser.c audioconvert.h v408dec.c roqvideoenc.c mpegvideo.h libstagefright.cpp h263.h dvbsubdec.c audioconvert.c ffpresets: libvpx-720p.ffpreset libvpx-720p50_60.ffpreset libvpx-360p.ffpreset libvpx-1080p.ffpreset libvpx-1080p50_60.ffpreset doc: viterbi.txt soc.txt outdevs.texi libavfilter.texi filters.texi ffmpeg.txt eval.texi decoders.texi texi2pod.pl* snow.txt optimization.txt issue_tracker.txt ffserver.texi ffmpeg.texi errno.txt build_system.txt tablegen.txt RELEASE_NOTES muxers.texi indevs.texi ffserver.conf ffmpeg-mt-authorship.txt encoders.texi bitstream_filters.texi t2h.init rate_distortion.txt multithreading.txt git-howto.txt ffprobe.xsd fate.texi doxy/ avutil.txt swscale.txt protocols.texi metadata.texi git-howto.texi ffprobe.texi faq.texi developer.texi avtools-common-opts.texi swresample.txt platform.texi Makefile general.texi ffplay.texi examples/ demuxers.texi APIchanges wifislax ffmpeg # clear wifislax ffmpeg # dir -r * version.sh* README MAINTAINERS library.mak ffserver.c ffplay.c Doxyfile COPYING.LGPLv3 COPYING.GPLv3 configure* config.fate cmdutils.h cmdutils.c RELEASE Makefile LICENSE INSTALL ffprobe.c ffmpeg.c CREDITS COPYING.LGPLv2.1 COPYING.GPLv2 config.log common.mak cmdutils_common_opts.h Changelog tools: unwrap-diff* qt-faststart.c pktdumper.c lavfi-showfiltfmts.c graph2dot.c enum_options.c clean-diff* aviocat.c trasher.c probetest.c patcheck* ismindex.c ffeval.c cws2fws.c build_libstagefright tests: videogen.c regression-funcs.sh* Makefile lavfi-regression.sh* ffserver.conf fate.sh* fate/ base64.c tiny_psnr.c ref/ lena.pnm ffserver-regression.sh* fate-valgrind.supp fate-run.sh* copycooker.sh* audiogen.c rotozoom.c md5.sh lavf-regression.sh* ffserver.regression.ref fate-update.sh* fate_config.sh.template codec-regression.sh* asynth1.sw presets: libx264-ipod640.ffpreset libx264-ipod320.ffpreset libvpx-720p.ffpreset libvpx-720p50_60.ffpreset libvpx-360p.ffpreset libvpx-1080p.ffpreset libvpx-1080p50_60.ffpreset mt-work: yuvcmp.c valgrind-check.sh todo.txt test.sh raw.sh mplayer.diff email.sh libswscale: yuv2rgb.c utils.c swscale-test.c swscale.h sparc/ rgb2rgb.h ppc/ options.c libswscale.v colorspace-test.c x86/ swscale_unscaled.c swscale_internal.h swscale.c rgb2rgb_template.c rgb2rgb.c output.c Makefile input.c bfin/ libswresample: swresample_test.c swresample_internal.h swresample.h swresample.c resample.c rematrix_template.c rematrix.c Makefile libswresample.v audioconvert.h audioconvert.c libpostproc: postprocess_template.c postprocess_internal.h postprocess.h postprocess.c postprocess_altivec_template.c Makefile libpostproc.v libavutil: x86_cpu.h timecode.h samplefmt.c pixfmt.h opt.c Makefile libavutil.v intfloat.h fifo.h des.h colorspace.h avr32/ adler32.h x86/ timecode.c rc4.h pixdesc.h mips/ lzo.h lfg.h internal.h fifo.c des.c bswap.h avassert.h adler32.c utils.c softfloat.h rc4.c pixdesc.c mem.h lzo.c lfg.c integer.h eval.h crc.h bfin/ audioconvert.h tree.h softfloat.c rational.h pca.h mem.c log.h inverse.c integer.c eval.c crc_data.h base64.h audioconvert.c tree.c sha.h rational.c pca.c md5.h log.c intreadwrite.h imgutils.h error.h crc.c base64.c attributes.h tomi/ sha.c random_seed.h parseutils.h md5.c lls.h intmath.h imgutils.c error.c cpu.h avutil.h arm/ timestamp.h sh4/ random_seed.c parseutils.c mathematics.h lls.c intfloat_readwrite.h file.h dict.h cpu.c avstring.h aes.h timer.h samplefmt.h ppc/ opt.h mathematics.c libm.h intfloat_readwrite.c file.c dict.c common.h avstring.c aes.c libavformat: yuv4mpeg.c swfenc.c rtpenc_xiph.c rsoenc.c oggparsetheora.c mpc.c latmenc.c gsmdec.c cutils.c au.c yop.c swfdec.c rtpenc_vp8.c rsodec.c oggparsespeex.c mpc8.c jvdec.c gopher.c crypto.c assenc.c xwma.c srtdec.c rtpenc_mpv.c rso.c oggparseskeleton.c mp3enc.c ivfenc.c gif.c crcenc.c assdec.c xmv.c spdif.h rtpenc_latm.c rpl.c oggparseogm.c mp3dec.c ivfdec.c g729dec.c concat.c asf.h xa.c spdifenc.c rtpenc_h264.c rm.h oggparseflac.c movenchint.c iv8.c g723_1.c cdxl.c asfenc.c wv.c spdifdec.c rtpenc_h263_rfc2190.c rmenc.c oggparsedirac.c movenc.h iss.c framehash.c cdg.c asfdec.c wtv.h spdif.c rtpenc_h263.c rmdec.c oggparsecelt.c movenc.c isom.h framecrcenc.c cavsvideodec.c asfcrypt.h wtvenc.c sox.h rtpenc.h rm.c oggenc.c mov_chan.h isom.c flv.h caf.h asfcrypt.c wtvdec.c soxenc.c rtpenc_chain.h rl2.c oggdec.h mov_chan.c ipmovie.c flvenc.c cafenc.c asf.c wtv.c soxdec.c rtpenc_chain.c riff.h oggdec.c mov.c internal.h flvdec.c cafdec.c apetag.h westwood_vqa.c sol.c rtpenc.c riff.c nuv.c mmst.c ingenientdec.c flic.c caf.c apetag.c westwood_aud.c smjpeg.h rtpenc_amr.c rdt.h nut.h mmsh.c img2enc.c flacenc_header.c cache.c ape.c wc3movie.c smjpegenc.c rtpenc_aac.c rdt.c nutenc.c mms.h img2dec.c flacenc.h c93.c apc.c wav.c smjpegdec.c rtpdec_xiph.c rawvideodec.c nutdec.c mms.c img2.c flacenc.c bmv.c anm.c vqf.c smjpeg.c rtpdec_vp8.c rawenc.h nut.c mmf.c iff.c flacdec.c bluray.c amr.c vorbiscomment.h smacker.c rtpdec_svq3.c rawenc.c nullenc.c mm.c idroqenc.c filmstripenc.c bit.c allformats.c vorbiscomment.c siff.c rtpdec_qt.c rawdec.h nsvdec.c mkvtimestamp_v2.c idroqdec.c filmstripdec.c bintext.c aiff.h voc.h sierravmd.c rtpdec_qdm2.c rawdec.c network.h microdvdenc.c idcin.c file.c bink.c aiffenc.c vocenc.c segment.c rtpdec_qcelp.c r3d.c network.c microdvddec.c id3v2.h ffm.h bfi.c aiffdec.c vocdec.c segafilm.c rtpdec_mpeg4.c qtpalette.h ncdec.c metadata.h id3v2enc.c ffmeta.h bethsoftvid.c aea.c voc.c seek-test.c rtpdec_latm.c qcp.c mxg.c metadata.c id3v2.c ffmetaenc.c avs.c adxdec.c version.h seek.h rtpdec_h264.c pva.c mxf.h md5proto.c id3v1.h ffmetadec.c avlanguage.h adts.h vc1testenc.c seek.c rtpdec_h263_rfc2190.c psxstr.c mxfenc.c md5enc.c id3v1.c ffmenc.c avlanguage.c adtsenc.c vc1test.c sdp.c rtpdec_h263.c pmpdec.c mxfdec.c matroska.h icodec.c ffmdec.c avisynth.c act.c utils.c sbgdec.c rtpdec.h pcm.h mxf.c matroskaenc.c http.h electronicarts.c avio_internal.h ac3dec.c url.h sauce.h rtpdec_g726.c pcmenc.c mvi.c matroskadec.c http.c eacdata.c avio.h aacdec.c udp.c sauce.c rtpdec_formats.h pcmdec.c mtv.c matroska.c httpauth.h dxa.c avio.c a64.c txd.c sapenc.c rtpdec.c pcm.c msnwc_tcp.c Makefile httpauth.c dv.h aviobuf.c 4xm.c tty.c sapdec.c rtpdec_asf.c os_support.h mpjpeg.c m4vdec.c hlsproto.c dvenc.c avi.h tta.c rtsp.h rtpdec_amr.c os_support.c mpegvideodec.c lxfdec.c hls.c dv.c avienc.c tmv.c rtspenc.c rtp.c options.c mpegts.h loasdec.c h264dec.c dtsdec.c avidec.c tls.c rtspdec.c rtmpproto.c oma.h mpegtsenc.c lmlm4.c h263dec.c dsicin.c avformat.h tiertexseq.c rtspcodes.h rtmppkt.h omaenc.c mpegts.c librtmp.c h261dec.c dnxhddec.c avc.h thp.c rtsp.c rtmppkt.c omadec.c mpeg.h libnut.c gxf.h diracdec.c avc.c tcp.c rtpproto.c rtmp.h oma.c mpegenc.c libmodplug.c gxfenc.c dfa.c audiointerleave.h swf.h rtp.h rso.h oggparsevorbis.c mpeg.c libavformat.v gxf.c daud.c audiointerleave.c libavfilter: yadif.h vsrc_buffer.c vf_split.c vf_pad.c vf_frei0r.c vf_cropdetect.c transform.h graphdump.c avfiltergraph.h asink_anullsink.c af_ashowinfo.c x86/ vsink_nullsink.c vf_slicify.c vf_overlay.c vf_format.c vf_crop.c transform.c gradfun.h avfiltergraph.c allfilters.c af_aresample.c vsrc_testsrc.c vf_yadif.c vf_showinfo.c vf_null.c vf_fifo.c vf_copy.c src_movie.c formats.c avfilter.c all_channel_layouts.h af_anull.c vsrc_mptestsrc.c vf_vflip.c vf_settb.c vf_mp.c vf_fieldorder.c vf_boxblur.c sink_buffer.c drawutils.h avcodec.h af_volume.c af_amerge.c vsrc_mandelbrot.c vf_unsharp.c vf_setpts.c vf_lut.c vf_fade.c vf_blackframe.c Makefile drawutils.c avcodec.c af_silencedetect.c af_aformat.c vsrc_life.c vf_transpose.c vf_setfield.c vf_libopencv.c vf_drawtext.c vf_blackdetect.c libmpcodecs/ defaults.c asrc_anullsrc.c af_pan.c af_aconvert.c vsrc_color.c vf_tinterlace.c vf_select.c vf_hqdn3d.c vf_drawbox.c vf_ass.c libavfilter.v buffersrc.h asrc_aevalsrc.c af_earwax.c vsrc_cellauto.c vf_thumbnail.c vf_scale.c vf_hflip.c vf_deshake.c vf_aspect.c internal.h buffersink.h asrc_abuffer.h af_astreamsync.c vsrc_buffer.h vf_swapuv.c vf_pixdesctest.c vf_gradfun.c vf_delogo.c version.h graphparser.c avfilter.h asrc_abuffer.c af_asplit.c libavdevice: x11grab.c timefilter.h sndio_common.h oss_audio.c libcdio.c fbdev.c dshow.h dshow_common.c avdevice.c alsa-audio-common.c vfwcap.c timefilter.c sndio_common.c openal-dec.c libavdevice.v dv1394.h dshow_filter.c dshow.c alsa-audio.h alldevices.c v4l.c sndio_enc.c sdl.c Makefile lavfi.c dv1394.c dshow_enumpins.c bktr.c alsa-audio-enc.c v4l2.c sndio_dec.c pulse.c libdc1394.c jack_audio.c dshow_pin.c dshow_enummediatypes.c avdevice.h alsa-audio-dec.c libavcodec: zmbvenc.c v308enc.c roqvideodec.c mpegvideo_enc.c libspeexenc.c h263dec.c dvbsub.c atrac.h zmbv.c v308dec.c roqvideo.c mpegvideo_common.h libspeexdec.c h263data.h dump_extradata_bsf.c atrac.c yuv4enc.c v210x.c roqaudioenc.c mpegvideo.c libschroedinger.h h263.c dsputil_template.c atrac3data.h yuv4dec.c v210enc.c rl.h mpegaudio_tablegen.h libschroedingerenc.c h261_parser.c dsputil.h atrac3.c yop.c v210dec.h rle.h mpegaudio_tablegen.c libschroedingerdec.c h261.h dsputil.c atrac1data.h y41penc.c v210dec.c rle.c mpegaudiotab.h libschroedinger.c h261enc.c dsicinav.c atrac1.c y41pdec.c utvideo.c rl2.c mpegaudio_parser.c libopenjpegenc.c h261dec.c dpxenc.c asv1.c xxan.c utils.c resample.c mpegaudio.h libopenjpegdec.c h261data.h dpx.c ass_split.h xwd.h unary.h resample2.c mpegaudioenc.c libopencore-amr.c h261.c dpcm.c ass_split.c xwdenc.c ulti_cb.h remove_extradata_bsf.c mpegaudiodsp_template.c libmp3lame.c gsm_parser.c dnxhd_parser.c ass.h xwddec.c ulti.c rectangle.h mpegaudiodsp.h libgsm.c gsm.h dnxhdenc.h assenc.c xvmc_internal.h txd.c rdft.h mpegaudiodsp_float.c libfaac.c gsmdec_template.c dnxhdenc.c assdec.c xvmc.h twinvq_data.h rdft.c mpegaudiodsp_fixed.c libdirac_libschro.h gsmdec_data.h dnxhddec.c ass.c xsubenc.c twinvq.c raw.h mpegaudiodsp.c libdirac_libschro.c gsmdec_data.c dnxhddata.h arm/ xsubdec.c tta.c rawenc.c mpegaudiodectab.h libdirac.h gsmdec.c dnxhddata.c apedec.c xl.c tscc.c rawdec.c mpegaudiodecheader.h libdiracdec.c golomb-test.c dirac_parser.c ansi.c xiph.h truespeech_data.h raw.c mpegaudiodecheader.c libcelt_dec.c golomb.h dirac.h anm.c xiph.c truespeech.c ratecontrol.h mpegaudiodec_float.c libavcodec.v golomb.c diracdsp.h amrwbdec.c xan.c truemotion2.c ratecontrol.c mpegaudiodec.c libaacplus.c gifdec.c diracdsp.c amrwbdata.h x86/ truemotion1data.h rangecoder.h mpegaudiodata.h lcl.h gif.c diracdec.c amrnbdec.c ws-snd1.c truemotion1.c rangecoder.c mpegaudiodata.c lclenc.c get_bits.h dirac.c amrnbdata.h wnv1.c tmv.c ra288.h mpegaudio.c lcldec.c g729postfilter.h dirac_arith.h amr.h wmv2.h timecode.h ra288.c mpeg4video_parser.h latm_parser.c g729postfilter.c dirac_arith.c alsdec.c wmv2enc.c timecode.c ra144.h mpeg4video_parser.c lagarithrac.h g729.h dfa.c alpha/ wmv2dec.c tiff.h ra144enc.c mpeg4video.h lagarithrac.c g729dec.c dct-test.c allcodecs.c wmv2.c tiffenc.c ra144dec.c mpeg4videoenc.c lagarith.c g729data.h dctref.h alacenc.c wmavoice_data.h tiff.c ra144.c mpeg4videodec.c kmvc.c g726.c dctref.c alac.c wmavoice.c tiertexseqv.c r210enc.c mpeg4video.c kgv1dec.c g723_1_data.h dct.h adx_parser.c wmaprodec.c thread.h r210dec.c mpeg4data.h kbdwin.h g723_1.c dct.c adx.h wmaprodata.h targa.h qtrleenc.c mpeg4audio.h kbdwin.c g722.h dct32.h adxenc.c wmalosslessdec.c targaenc.c qtrle.c mpeg4audio.c jvdec.c g722enc.c dct32_float.c adxdec.c wma.h targa.c qpeg.c mpeg12.h jrevdct.c g722dec.c dct32_fixed.c adx.c wmaenc.c tableprint.h qdrw.c mpeg12enc.c jpegls.h g722.c dct32.c adpcm.h wmadec.c synth_filter.h qdm2_tablegen.h mpeg12decdata.h jpeglsenc.c frwu.c dca_parser.h adpcmenc.c wmadata.h synth_filter.c qdm2_tablegen.c mpeg12data.h jpeglsdec.h fraps.c dca_parser.c adpcm_data.h wma_common.h svq3.c qdm2data.h mpeg12data.c jpeglsdec.c fmtconvert.h dcahuff.h adpcm_data.c wma_common.c svq1_vlc.h qdm2.c mpeg12.c jpegls.c fmtconvert.c dca.h adpcm.c wma.c svq1.h qcelpdec.c mpc.h jfdctint_template.c flv.h dcaenc.h acelp_vectors.h wavpack.c svq1enc_cb.h qcelpdata.h mpcdata.h jfdctint.c flvenc.c dcaenc.c acelp_vectors.c w32pthreads.h svq1enc.c put_bits.h mpc.c jfdctfst.c flvdec.c dcadsp.h acelp_pitch_delay.h vqavideo.c svq1dec.c ptx.c mpc8huff.h j2k.h flicvideo.c dcadsp.c acelp_pitch_delay.c vp8_parser.c svq1_cb.h pthread.c mpc8data.h j2kenc.c flashsvenc.c dcadata.h acelp_filters.h vp8.h svq1.c psymodel.h mpc8.c j2k_dwt.h flashsv.c dca.c acelp_filters.c vp8dsp.h sunrast.h psymodel.c mpc7data.h j2k_dwt.c flashsv2enc.c cyuv.c ac3tab.h vp8dsp.c sunrastenc.c proresenc_kostya.c mpc7.c j2kdec.c flac_parser.c cscd.c ac3tab.c vp8data.h sunrast.c proresenc_anatoliy.c mp3_header_decompress_bsf.c j2k.c flac.h crystalhd.c ac3_parser.h vp8.c srtenc.c proresdsp.h mp3_header_compress_bsf.c ivi_dsp.h flacenc.c cos_tablegen.c ac3_parser.c vp6dsp.c srtdec.c proresdsp.c movsub_bsf.c ivi_dsp.c flacdec.c cookdata.h ac3.h vp6data.h sparc/ proresdec_lgpl.c motion-test.c ivi_common.h flacdata.h cook.c ac3enc_template.c vp6.c sp5x.h proresdec.h motionpixels_tablegen.h ivi_common.c flacdata.c codec_names.sh* ac3enc_opts_template.c vp5data.h sp5xdec.c proresdec2.c motionpixels_tablegen.c ituh263enc.c flac.c cljr.c ac3enc.h vp5.c sonic.c proresdata.h motionpixels.c ituh263dec.c ffwavesynth.c cinepak.c ac3enc_float.c vp56rac.c snow.h proresdata.c motion_est_template.c inverse.c ffv1.c chomp_bsf.c ac3enc_fixed.c vp56.h snowenc.c ppc/ motion_est.c intrax8huf.h fft-test.c cga_data.h ac3enc.c vp56dsp.h snowdec.c pnm_parser.c mmvideo.c intrax8.h fft-internal.h cga_data.c ac3dsp.h vp56dsp.c snowdata.h pnm.h mlp_parser.h intrax8dsp.c fft.h celp_math.h ac3dsp.c vp56data.h snow.c pnmenc.c mlp_parser.c intrax8.c fft_float.c celp_math.c ac3dec.h vp56data.c smc.c pnmdec.c mlp.h interplayvideo.c fft-fixed-test.c celp_filters.h ac3dec_data.h vp56.c smacker.c pnm.c mlpdsp.c internal.h fft_fixed.c celp_filters.c ac3dec_data.c vp3_parser.c sipr.h png.h mlpdec.c intelh263dec.c fft.c cdxl.c ac3dec.c vp3dsp.c siprdata.h pngenc.c mlp.c indeo5data.h faxcompr.h cdgraphics.c ac3.c vp3data.h sipr.c pngdsp.h mjpeg_parser.c indeo5.c faxcompr.c cbrt_tablegen.h aasc.c vp3.c sipr16kdata.h pngdsp.c mjpeg.h indeo4data.h faanidct.h cbrt_tablegen.c aandcttab.h vorbis_parser.h sipr16k.c pngdec.c mjpegenc.h indeo4.c faanidct.c cavs_parser.c aandcttab.c vorbis_parser.c sinewin_tablegen.h png.c mjpegenc.c indeo3data.h faandct.h cavs.h aac_tablegen.h vorbis.h sinewin_tablegen.c pictordec.c mjpegdec.h indeo3.c faandct.c cavsdsp.h aac_tablegen_decl.h vorbis_enc_data.h sinewin.h pgssubdec.c mjpegdec.c indeo2data.h escape130.c cavsdsp.c aac_tablegen.c vorbisenc.c sinewin.c pcxenc.c mjpeg.c indeo2.c escape124.c cavsdec.c aactab.h vorbisdec.c simple_idct_template.c pcx.c mjpegbdec.c imx_dump_header_bsf.c error_resilience.c cavsdata.h aactab.c vorbis_data.c simple_idct.h pcm_tablegen.h mjpega_dump_header_bsf.c imgconvert.h elbg.h cavs.c aacsbr.h vorbis.c simple_idct.c pcm_tablegen.c mjpeg2jpeg_bsf.c imgconvert.c elbg.c cabac.h aacsbrdata.h vmnc.c shorten.c pcm-mpeg.c mips/ imcdata.h eatqi.c cabac_functions.h aacsbr.c vmdav.c sh4/ pcm.c mimic.c imc.c eatgv.c cabac.c aacpsy.h version.h sgi.h parser.h mdec.c iirfilter.h eatgq.c c93.c aacpsy.c vdpau_internal.h sgienc.c parser.c mdct_float.c iirfilter.c eamad.c bytestream.h aacps_tablegen.h vdpau.h sgidec.c pamenc.c mdct_fixed.c iff.c eaidct.c bmv.c aacps_tablegen.c vdpau.c sbr.h os2threads.h mdct.c idcinvideo.c eacmv.c bmp.h aacps.h vda_internal.h sbrdsp.h options.c mathops.h huffyuv.c eac3enc.h bmpenc.c aacpsdata.c vda_h264.c sbrdsp.c nuv.c Makefile huffman.h eac3enc.c bmp.c aacps.c vda.h s3tc.h noise_bsf.c mace.c huffman.c eac3dec.c bitstream_filter.c aac_parser.c vda.c s3tc.c nellymoser.h lzw.h h264_sei.c eac3_data.h bitstream.c aac.h vcr1.c s302m.c nellymoserenc.c lzwenc.c h264_refs.c eac3_data.c bit_depth_template.c aacenc.h vc1_parser.c rv40vlc2.h nellymoserdec.c lzw.c h264_ps.c dxva2_vc1.c bintext.h aacenc.c vc1.h rv40dsp.c nellymoser.c lsp.h h264pred_template.c dxva2_mpeg2.c bintext.c aacdectab.h vc1dsp.h rv40data.h mxpegdec.c lsp.c h264pred.h dxva2_internal.h binkdsp.h aacdec.c vc1dsp.c rv40.c msvideo1enc.c lpc.h h264pred.c dxva2_h264.c binkdsp.c aaccoder.c vc1dec.c rv34vlc.h msvideo1.c lpc.c h264_parser.c dxva2.h binkdata.h aac_adtstoasc_bsf.c vc1data.h rv34_parser.c msrledec.h loco.c h264_mvpred.h dxva2.c bink.c aacadtsdec.h vc1data.c rv34.h msrledec.c ljpegenc.c h264_mp4toannexb_bsf.c dxtory.c binkaudio.c aacadtsdec.c vc1.c rv34dsp.h msrle.c libxvid_rc.c h264_loopfilter.c dxa.c bgmc.h aac_ac3_parser.h vc1acdata.h rv34dsp.c msmpeg4.h libxvid_internal.h h264idct_template.c dwt.h bgmc.c aac_ac3_parser.c vble.c rv34data.h msmpeg4enc.c libxvidff.c h264idct.c dwt.c bfin/ a64tables.h vb.c rv34.c msmpeg4data.h libxavs.c h264.h dv_vlc_data.h bfi.c a64multienc.c vaapi_vc1.c rv30dsp.c msmpeg4data.c libx264.c h264dsp_template.c dv_tablegen.h bethsoftvideo.h a64enc.h vaapi_mpeg4.c rv30data.h msmpeg4.c libvpxenc.c h264dsp.h dv_tablegen.c bethsoftvideo.c a64colors.h vaapi_mpeg2.c rv30.c msgsmdec.h libvpxdec.c h264dsp.c dvquant.h avs.c 8svx.c vaapi_internal.h rv20enc.c msgsmdec.c libvorbis.c h264_direct.c dvdsub_parser.c avr32/ 8bps.c vaapi_h264.c rv10enc.c mqc.h libvo-amrwbenc.c h264data.h dvdsubenc.c avpacket.c 4xm.c vaapi.h rv10.c mqcenc.c libvo-aacenc.c h264_cavlc.c dvdsubdec.c avfft.h vaapi.c rtjpeg.h mqcdec.c libutvideo.h h264_cabac.c dvdata.h avfft.c v410enc.c rtjpeg.c mqc.c libutvideoenc.cpp h264.c dvdata.c avcodec.h v410dec.c rpza.c mpegvideo_xvmc.c libutvideodec.cpp h263_parser.h dv.c aura.c v408enc.c roqvideo.h mpegvideo_parser.c libtheoraenc.c h263_parser.c dvbsub_parser.c audioconvert.h v408dec.c roqvideoenc.c mpegvideo.h libstagefright.cpp h263.h dvbsubdec.c audioconvert.c ffpresets: libvpx-720p.ffpreset libvpx-720p50_60.ffpreset libvpx-360p.ffpreset libvpx-1080p.ffpreset libvpx-1080p50_60.ffpreset doc: viterbi.txt soc.txt outdevs.texi libavfilter.texi filters.texi ffmpeg.txt eval.texi decoders.texi texi2pod.pl* snow.txt optimization.txt issue_tracker.txt ffserver.texi ffmpeg.texi errno.txt build_system.txt tablegen.txt RELEASE_NOTES muxers.texi indevs.texi ffserver.conf ffmpeg-mt-authorship.txt encoders.texi bitstream_filters.texi t2h.init rate_distortion.txt multithreading.txt git-howto.txt ffprobe.xsd fate.texi doxy/ avutil.txt swscale.txt protocols.texi metadata.texi git-howto.texi ffprobe.texi faq.texi developer.texi avtools-common-opts.texi swresample.txt platform.texi Makefile general.texi ffplay.texi examples/ demuxers.texi APIchanges wifislax ffmpeg # -------------- next part -------------- A non-text attachment was scrubbed... Name: ffmpeg.png Type: image/png Size: 51449 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: config.log Type: application/octet-stream Size: 151128 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: config.log Type: application/octet-stream Size: 147109 bytes Desc: not available URL: From james.darnley at gmail.com Mon Mar 12 10:29:22 2012 From: james.darnley at gmail.com (James Darnley) Date: Mon, 12 Mar 2012 10:29:22 +0100 Subject: [FFmpeg-user] error compile ffmpeg in linux - wifislax In-Reply-To: References: Message-ID: <4F5DC1F2.5040009@gmail.com> On 2012-03-11 13:46, Junior Andre Valdivia Ramos wrote: > --enable-avisynth > ERROR: vfw32 not found Unsurprising since avisynth is only available on Windows. I thought configure knew that. > wifislax ffmpeg # ./configure --enable-static --enable-gpl --enable-libxvid > ERROR: libxvid not found Did you install xvid? From bas at bushbaby.nl Mon Mar 12 12:45:06 2012 From: bas at bushbaby.nl (Bas Kamer) Date: Mon, 12 Mar 2012 12:45:06 +0100 Subject: [FFmpeg-user] loop Message-ID: <7C752A37-27EC-4C6F-825C-D49B0CD565FA@bushbaby.nl> Hi, for the past few weeks I have been trying to generate a MOV container containing a short MJPEG sequence that has it's looping option set. So when the end user doubleclicks the file it will have it loop menu option (Quicktime) selected. Unfortunately I have only been successful in generating a looping gif animation and are about to give up... The manual states "Repeatedly loop output for formats that support looping such as animated GIF (0 will loop the output infinitely). This option is deprecated, use -loop." Question: What formats are those? Or more precise. The .mov container should be able to do this. Can someone confirm that it is? Question: Does someone have some examples that do this? JPEG to QT with looping enabled thanks a million From bouke at editb.nl Mon Mar 12 15:05:27 2012 From: bouke at editb.nl (bouke) Date: Mon, 12 Mar 2012 15:05:27 +0100 Subject: [FFmpeg-user] loop References: <7C752A37-27EC-4C6F-825C-D49B0CD565FA@bushbaby.nl> Message-ID: <01d801cd0059$307c8a50$4301a8c0@hpkantoor> ----- Original Message ----- From: "Bas Kamer" > Hi, > > > for the past few weeks I have been trying to generate a MOV container > containing a short MJPEG sequence that has it's looping option set. So > when the end user doubleclicks the file it will have it loop menu option > (Quicktime) selected. Unfortunately I have only been successful in > generating a looping gif animation and are about to give up... > > The manual states "Repeatedly loop output for formats that support looping > such as animated GIF (0 will loop the output infinitely). This option is > deprecated, use -loop." > > Question: What formats are those? Or more precise. The .mov container > should be able to do this. Can someone confirm that it is? > > Question: Does someone have some examples that do this? JPEG to QT with > looping enabled Bas, No idea if / how / why FFmpeg does this, but if it is a one-off, use QTpro. Choose the loop function and save (no need for a save as) Nothing more to it.... hth, Bouke > > thanks a million > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From bas at bushbaby.nl Mon Mar 12 15:10:44 2012 From: bas at bushbaby.nl (Bas Kamer) Date: Mon, 12 Mar 2012 15:10:44 +0100 Subject: [FFmpeg-user] loop In-Reply-To: <01d801cd0059$307c8a50$4301a8c0@hpkantoor> References: <7C752A37-27EC-4C6F-825C-D49B0CD565FA@bushbaby.nl> <01d801cd0059$307c8a50$4301a8c0@hpkantoor> Message-ID: On 12 mrt. 2012, at 15:05, bouke wrote: > Bas, > No idea if / how / why FFmpeg does this, but if it is a one-off, use QTpro. > Choose the loop function and save (no need for a save as) > Nothing more to it.... Hi, Thanks for your suggestion of which I am aware. Unfortunately I need this process automated on the server. bas From bouke at editb.nl Mon Mar 12 15:25:42 2012 From: bouke at editb.nl (bouke) Date: Mon, 12 Mar 2012 15:25:42 +0100 Subject: [FFmpeg-user] loop References: <7C752A37-27EC-4C6F-825C-D49B0CD565FA@bushbaby.nl><01d801cd0059$307c8a50$4301a8c0@hpkantoor> Message-ID: <01f001cd005c$04ff16b0$4301a8c0@hpkantoor> From: "Bas Kamer" To: "FFmpeg user questions and RTFMs" Sent: Monday, March 12, 2012 3:10 PM Subject: Re: [FFmpeg-user] loop > On 12 mrt. 2012, at 15:05, bouke wrote: > >> Bas, >> No idea if / how / why FFmpeg does this, but if it is a one-off, use >> QTpro. >> Choose the loop function and save (no need for a save as) >> Nothing more to it.... > > Hi, > > Thanks for your suggestion of which I am aware. > Unfortunately I need this process automated on the server. Another workaround, make an empty QT with the loop attribute set, paste in your new movie and save. Or applescript the process (well, you probably don't run Mac...) Or wait untill someone tells you how to do it with FFmpeg :- Bouke > bas > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From jacobhameiri at gmail.com Mon Mar 12 18:36:30 2012 From: jacobhameiri at gmail.com (jacob s) Date: Mon, 12 Mar 2012 19:36:30 +0200 Subject: [FFmpeg-user] scale input video before transcoding Message-ID: Hello, I am using ffmpeg to stream windows XP desktop and sound ( directshow input ), but I am facing performance issues. this is the command I am using: -f dshow -i video='screen-capture-recorder':audio='Stereo Mix (IDT High Definition' -vcodec libx264 -preset ultrafast -r 10 -async 1 -ab 32k -ar 22050 -f mpegts udp://192.168.2.100:1234 When using this command ffmpeg uses about 65% CPU ( Pentium4 2.8 GHz ), and after a short while I am seeing glitches in the viewing machine. I thought of reducing CPU usage by scaling the input video to 50% - 75%, but even after scaling to 50% CPU usage doesn't change, I suspect this is because the scaling is performed after the transcoding process, is there a way I can tell ffmpeg to scale the input coming from the screen capturer ? What else can I do to reduce the glitches and CPU usage ( I already reduced to 10 fps, I cant go any lower than that ) ? Thanks, Jacob From cehoyos at ag.or.at Mon Mar 12 19:22:55 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 12 Mar 2012 18:22:55 +0000 (UTC) Subject: [FFmpeg-user] scale input video before transcoding References: Message-ID: jacob s gmail.com> writes: > -f dshow -i video='screen-capture-recorder':audio='Stereo > Mix (IDT High Definition' -vcodec libx264 -preset ultrafast > -r 10 -async 1 -ab 32k -ar 22050 -f mpegts udp://192.168.2.100:1234 Complete, uncut console output missing. > When using this command ffmpeg uses about 65% CPU > ( Pentium4 2.8 GHz ), and after a short while I am > seeing glitches in the viewing machine. Note that the Pentium4 has known limitations (that may or may not be the reason for your problems). > I thought of reducing CPU usage by scaling the input video > to 50% - 75%, but even after scaling to 50% CPU usage doesn't > change, I suspect this is Command line and complete, uncut console output missing. > because the scaling is performed after the transcoding process It is not possible to scale after transcoding (all scaling has to be done after decoding and before encoding). Did you test encoding to a file instead of udp? What speed / CPU usage do you see? Carl Eugen From cehoyos at ag.or.at Mon Mar 12 19:27:11 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 12 Mar 2012 18:27:11 +0000 (UTC) Subject: [FFmpeg-user] loop References: <7C752A37-27EC-4C6F-825C-D49B0CD565FA@bushbaby.nl> Message-ID: Bas Kamer bushbaby.nl> writes: > Question: Does someone have some examples that do this? > JPEG to QT with looping enabled $ ./ffmpeg -loop 1 -i tests/lena.pnm -vcodec mjpeg -vframes 100 out.mov Note that you are usually expected here to post the command line that you tried together with the complete, uncut console output and - if necessary - an explanation what went wrong. Carl Eugen From bas at bushbaby.nl Mon Mar 12 21:55:58 2012 From: bas at bushbaby.nl (Bas Kamer) Date: Mon, 12 Mar 2012 21:55:58 +0100 Subject: [FFmpeg-user] loop In-Reply-To: References: <7C752A37-27EC-4C6F-825C-D49B0CD565FA@bushbaby.nl> Message-ID: <1543198C-57A5-45C7-8A82-B3F7D4F57134@bushbaby.nl> On 12 mrt. 2012, at 19:27, Carl Eugen Hoyos wrote: > Bas Kamer bushbaby.nl> writes: > >> Question: Does someone have some examples that do this? >> JPEG to QT with looping enabled > > $ ./ffmpeg -loop 1 -i tests/lena.pnm -vcodec mjpeg -vframes 100 out.mov > short answer -loop 1 gives 'invalid value '1' for option loop -loop_output 0 does finish but does not loop... note I tried two version, unfortunatly it seems i am stuck = AS SUGGESTED ====================== Rattletrap:1330031941993 bas$ ffmpeg -loop 1 -i tests/lena.pnm -vcodec mjpeg -vframes 100 out.mov ffmpeg version 0.7.11, Copyright (c) 2000-2011 the FFmpeg developers built on Mar 12 2012 13:27:22 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libopenjpeg --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/opt/local/bin/gcc-apple-4.2 --arch=x86_64 --enable-yasm libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.123. 0 / 52.123. 0 libavformat 52.111. 0 / 52.111. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 Invalid value '1' for option 'loop' ffmpeg -loop 1 -i lena.pnm -vcodec mjpeg -vframes 100 out.mov FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers built on Jan 29 2012 23:56:18 with gcc 4.1.2 20080704 (Red Hat 4.1.2-51) configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m32 -march=i386 -mtune=generic -fasynchronous-unwind-tables' --enable-avfilter --enable-avfilter-lavf --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.19. 0 / 1.19. 0 libswscale 0.11. 0 / 0.11. 0 libpostproc 51. 2. 0 / 51. 2. 0 Invalid value '1' for option 'loop' = TRIED ====================== Rattletrap:1330031941993 bas$ ffmpeg -loop_output 0 -i tests/lena.pnm -vcodec mjpeg -vframes 100 out.mov ffmpeg version 0.7.11, Copyright (c) 2000-2011 the FFmpeg developers built on Mar 12 2012 13:27:22 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libopenjpeg --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/opt/local/bin/gcc-apple-4.2 --arch=x86_64 --enable-yasm libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.123. 0 / 52.123. 0 libavformat 52.111. 0 / 52.111. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, image2, from 'tests/lena.pnm': Duration: 00:00:00.04, start: 0.000000, bitrate: N/A Stream #0.0: Video: ppm, rgb24, 256x256, 25 tbr, 25 tbn, 25 tbc File 'out.mov' already exists. Overwrite ? [y/N] y Incompatible pixel format 'rgb24' for codec 'mjpeg', auto-selecting format 'yuvj420p' [buffer @ 0x7ffddac31da0] w:256 h:256 pixfmt:rgb24 tb:1/1000000 sar:0/1 sws_param: [buffersink @ 0x7ffddac32620] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' [scale @ 0x7ffddac32960] w:256 h:256 fmt:rgb24 -> w:256 h:256 fmt:yuvj420p flags:0x4 Output #0, mov, to 'out.mov': Metadata: encoder : Lavf52.111.0 Stream #0.0: Video: mjpeg, yuvj420p, 256x256, q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop, [?] for help frame= 1 fps= 0 q=4.1 Lsize= 13kB time=00:00:00.04 bitrate=2618.0kbits/s video:12kB audio:0kB global headers:0kB muxing overhead 5.470953% [lumasol at srv1 ~]$ ffmpeg -loop_output 1 -i lena.pnm -vcodec mjpeg -vframes 100 out.mov FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers built on Jan 29 2012 23:56:18 with gcc 4.1.2 20080704 (Red Hat 4.1.2-51) configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m32 -march=i386 -mtune=generic -fasynchronous-unwind-tables' --enable-avfilter --enable-avfilter-lavf --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.19. 0 / 1.19. 0 libswscale 0.11. 0 / 0.11. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, image2, from 'lena.pnm': Duration: 00:00:00.04, start: 0.000000, bitrate: N/A Stream #0.0: Video: ppm, rgb24, 256x256, 25 tbr, 25 tbn, 25 tbc File 'out.mov' already exists. Overwrite ? [y/N] y Output #0, mov, to 'out.mov': Metadata: encoder : Lavf52.64.2 Stream #0.0: Video: mjpeg, yuvj420p, 256x256, q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 1 fps= 0 q=4.1 Lsize= 13kB time=0.04 bitrate=2621.6kbits/s video:12kB audio:0kB global headers:0kB muxing overhead 5.454545% From mailer.tovis at freemail.hu Mon Mar 12 21:23:23 2012 From: mailer.tovis at freemail.hu (tovis) Date: Mon, 12 Mar 2012 21:23:23 +0100 Subject: [FFmpeg-user] Grabbing quality from TV card Message-ID: <919bb625a40b5bef37c86d3674bc288a.squirrel@nusi> Hi! I'm trying to grab video (at first) from a Pinacle PCTV Pro (TV + FM stereo receiver) as lspci shows. I have some success, but the quality is really wrong :( I have tested TV using fbtv and it was good enough, but grabbing using command: ffmpeg -v 10 -s 640x480 -r 25 -f video4linux2 -i /dev/video0 out0.avi gives me really low quality result. I'm not sure what is wrong but the result contains, scratches and digital noise, also some rude "combined" pixels. I have try to lower resolution such as 320x240 but the result is even worth. I'm using Debian squeeze, with binary packaged ffmpeg version SVN-r0.5.6-4:0.5.6-3 Any suggestion? Sincerely tovis From andreas.gumm at gmx.de Mon Mar 12 23:23:34 2012 From: andreas.gumm at gmx.de (Andreas Gumm) Date: Mon, 12 Mar 2012 23:23:34 +0100 Subject: [FFmpeg-user] ProRes multithreaded encoding Message-ID: <4F5E7766.90104@gmx.de> Hello to the list, I have just a short question, is it possible to encode ProRes multithreaded? Thanks a lot in advance! Andreas From rodney.baker at iinet.net.au Mon Mar 12 23:51:08 2012 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Tue, 13 Mar 2012 09:21:08 +1030 Subject: [FFmpeg-user] Grabbing quality from TV card In-Reply-To: <919bb625a40b5bef37c86d3674bc288a.squirrel@nusi> References: <919bb625a40b5bef37c86d3674bc288a.squirrel@nusi> Message-ID: <201203130921.09027.rodney.baker@iinet.net.au> On Tue, 13 Mar 2012 06:53:23 tovis wrote: > Hi! > I'm trying to grab video (at first) from a Pinacle PCTV Pro (TV + FM > stereo receiver) as lspci shows. I have some success, but the quality is > really wrong :( I have tested TV using fbtv and it was good enough, but > grabbing using command: > > ffmpeg -v 10 -s 640x480 -r 25 -f video4linux2 -i /dev/video0 out0.avi > > gives me really low quality result. I'm not sure what is wrong but the > result contains, scratches and digital noise, also some rude "combined" > pixels. I have try to lower resolution such as 320x240 but the result is > even worth. > > I'm using Debian squeeze, with binary packaged ffmpeg version > SVN-r0.5.6-4:0.5.6-3 > Any suggestion? > > Sincerely > tovis > Order of options is important. Move -i to the first command in your list. At the moment you're setting parameters for the input file, then specifying the input file, then the output file with no parameters (thus using defaults). Remember, when asking for help on this list *always* post your command line and *full uncut console output* from ffmpeg, otherwise you won't get the help you need (because unless it is really obvious, as in this case, without the console output no-one can tell what is going on. -- =================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au =================================================== From rodney.baker at iinet.net.au Mon Mar 12 23:54:13 2012 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Tue, 13 Mar 2012 09:24:13 +1030 Subject: [FFmpeg-user] Grabbing quality from TV card In-Reply-To: <919bb625a40b5bef37c86d3674bc288a.squirrel@nusi> References: <919bb625a40b5bef37c86d3674bc288a.squirrel@nusi> Message-ID: <201203130924.13604.rodney.baker@iinet.net.au> Oh, yes, I nearly forgot. On Tue, 13 Mar 2012 06:53:23 tovis wrote: > [..] > > I'm using Debian squeeze, with binary packaged ffmpeg version > SVN-r0.5.6-4:0.5.6-3 > Any suggestion? Update to the latest git-head - there have been *many* improvements and security fixes since that version. Old versions (especially that old) are unsupported. -- =================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au =================================================== From jacobhameiri at gmail.com Mon Mar 12 23:56:28 2012 From: jacobhameiri at gmail.com (jacob s) Date: Tue, 13 Mar 2012 00:56:28 +0200 Subject: [FFmpeg-user] play only left/right channel from an audio input Message-ID: Hello, Given a stereo audio input, is it possible to filter only left/right channel ? For example if I have a file with 2 audio channels can I create a new file that has the left channel audio only ? From mailer.tovis at freemail.hu Tue Mar 13 00:12:03 2012 From: mailer.tovis at freemail.hu (tovis) Date: Tue, 13 Mar 2012 00:12:03 +0100 Subject: [FFmpeg-user] Grabbing quality from TV card In-Reply-To: <201203130924.13604.rodney.baker@iinet.net.au> References: <919bb625a40b5bef37c86d3674bc288a.squirrel@nusi> <201203130924.13604.rodney.baker@iinet.net.au> Message-ID: <4b2f24783f90e3d4b90c8469cce6542e.squirrel@nusi> Hi Rodney! Thanks for your quick answer. I have built another system install development packages (headers gcc make ...) get latest git version. But not even worth :( This is the command: $ ffmpeg -v verbose -i /dev/video0 -f video4linux2 -s 640x480 -r 25 out10.avi 2>&1 | tee -a ffmpeg-grab.log and this is the console output: ffmpeg version N-38750-g599888a Copyright (c) 2000-2012 the FFmpeg developers built on Mar 12 2012 23:50:31 with gcc 4.4.5 configuration: --enable-nonfree --enable-libv4l2 libavutil 51. 42.100 / 51. 42.100 libavcodec 54. 10.100 / 54. 10.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 64.101 / 2. 64.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 /dev/video0: Invalid data found when processing input What could be missed? Sincerely tovis > Oh, yes, I nearly forgot. > > On Tue, 13 Mar 2012 06:53:23 tovis wrote: >> [..] >> >> I'm using Debian squeeze, with binary packaged ffmpeg version >> SVN-r0.5.6-4:0.5.6-3 >> Any suggestion? > > Update to the latest git-head - there have been *many* improvements and > security fixes since that version. Old versions (especially that old) are > unsupported. > > -- > =================================================== > Rodney Baker VK5ZTV > rodney.baker at iinet.net.au > =================================================== > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From onemda at gmail.com Tue Mar 13 00:12:45 2012 From: onemda at gmail.com (Paul B Mahol) Date: Mon, 12 Mar 2012 23:12:45 +0000 Subject: [FFmpeg-user] ProRes multithreaded encoding In-Reply-To: <4F5E7766.90104@gmx.de> References: <4F5E7766.90104@gmx.de> Message-ID: On 3/12/12, Andreas Gumm wrote: > Hello to the list, > > I have just a short question, > is it possible to encode ProRes multithreaded? It is possible, but currently not implemented. From lou at lrcd.com Tue Mar 13 00:17:34 2012 From: lou at lrcd.com (Lou) Date: Mon, 12 Mar 2012 15:17:34 -0800 Subject: [FFmpeg-user] play only left/right channel from an audio input In-Reply-To: References: Message-ID: <20120312151734.4693268d@lrcd.com> On Tue, 13 Mar 2012 00:56:28 +0200 jacob s wrote: > Hello, > > Given a stereo audio input, is it possible to filter only left/right > channel ? > For example if I have a file with 2 audio channels can I create a new file > that has the left channel audio only ? We need more information; "a new file that has the left channel audio only" can be interpreted as: * A mono output. * A stereo output with input left audio in both channels of the output. * A stereo output with input left audio in the left channel and silence in the right channel of the output. From rodney.baker at iinet.net.au Tue Mar 13 00:25:36 2012 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Tue, 13 Mar 2012 09:55:36 +1030 Subject: [FFmpeg-user] Grabbing quality from TV card In-Reply-To: <4b2f24783f90e3d4b90c8469cce6542e.squirrel@nusi> References: <919bb625a40b5bef37c86d3674bc288a.squirrel@nusi> <201203130924.13604.rodney.baker@iinet.net.au> <4b2f24783f90e3d4b90c8469cce6542e.squirrel@nusi> Message-ID: <201203130955.36230.rodney.baker@iinet.net.au> On Tue, 13 Mar 2012 09:42:03 tovis wrote: > Hi Rodney! > Thanks for your quick answer. > I have built another system install development packages (headers gcc make > ...) get latest git version. But not even worth :( > This is the command: > > $ ffmpeg -v verbose -i /dev/video0 -f video4linux2 -s 640x480 -r 25 > out10.avi 2>&1 | tee -a ffmpeg-grab.log > > and this is the console output: > ffmpeg version N-38750-g599888a Copyright (c) 2000-2012 the FFmpeg > developers built on Mar 12 2012 23:50:31 with gcc 4.4.5 > configuration: --enable-nonfree --enable-libv4l2 > libavutil 51. 42.100 / 51. 42.100 > libavcodec 54. 10.100 / 54. 10.100 > libavformat 54. 2.100 / 54. 2.100 > libavdevice 53. 4.100 / 53. 4.100 > libavfilter 2. 64.101 / 2. 64.101 > libswscale 2. 1.100 / 2. 1.100 > libswresample 0. 7.100 / 0. 7.100 > /dev/video0: Invalid data found when processing input > > What could be missed? > Sincerely Ok, I missed something. Try -f video4linux2 -i /dev/video0... -- =================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au =================================================== From robert at theMakers.com Mon Mar 12 23:57:35 2012 From: robert at theMakers.com (Robert Reinhardt) Date: Mon, 12 Mar 2012 22:57:35 +0000 Subject: [FFmpeg-user] play only left/right channel from an audio input In-Reply-To: References: Message-ID: <2D405CD275952E49B92B7F48B3A0308A2B84E306@nakedex.flaction.com> I just had to do this with a client in Canada. Do you want the left channel to be a mono track, or do you want it to play only in the left channel and silence in the right channel in the new input? -Robert -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of jacob s Sent: Monday, March 12, 2012 3:56 PM To: ffmpeg-user at ffmpeg.org Subject: [FFmpeg-user] play only left/right channel from an audio input Hello, Given a stereo audio input, is it possible to filter only left/right channel ? For example if I have a file with 2 audio channels can I create a new file that has the left channel audio only ? _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From jacobhameiri at gmail.com Tue Mar 13 00:30:01 2012 From: jacobhameiri at gmail.com (jacob s) Date: Tue, 13 Mar 2012 01:30:01 +0200 Subject: [FFmpeg-user] play only left/right channel from an audio input In-Reply-To: <20120312151734.4693268d@lrcd.com> References: <20120312151734.4693268d@lrcd.com> Message-ID: - A stereo output with input left audio in the left channel and silence in the right channel of the output. 2012/3/13 Lou > On Tue, 13 Mar 2012 00:56:28 +0200 > jacob s wrote: > > > Hello, > > > > Given a stereo audio input, is it possible to filter only left/right > > channel ? > > For example if I have a file with 2 audio channels can I create a new > file > > that has the left channel audio only ? > > We need more information; "a new file that has the left channel audio > only" can be interpreted as: > > * A mono output. > * A stereo output with input left audio in both channels of the output. > * A stereo output with input left audio in the left channel and silence > in the right channel of the output. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From lou at lrcd.com Tue Mar 13 01:06:04 2012 From: lou at lrcd.com (Lou) Date: Mon, 12 Mar 2012 16:06:04 -0800 Subject: [FFmpeg-user] play only left/right channel from an audio input In-Reply-To: References: <20120312151734.4693268d@lrcd.com> Message-ID: <20120312160604.0b760440@lrcd.com> On Tue, 13 Mar 2012 01:30:01 +0200 jacob s wrote: > 2012/3/13 Lou > > > On Tue, 13 Mar 2012 00:56:28 +0200 > > jacob s wrote: > > > > > Hello, > > > > > > Given a stereo audio input, is it possible to filter only left/right > > > channel ? > > > For example if I have a file with 2 audio channels can I create a new > > file > > > that has the left channel audio only ? > > > > We need more information; "a new file that has the left channel audio > > only" can be interpreted as: > > > > * A mono output. > > * A stereo output with input left audio in both channels of the output. > > * A stereo output with input left audio in the left channel and silence > > in the right channel of the output. > > - A stereo output with input left audio in the left channel and silence in > the right channel of the output. ffmpeg -i input.wav -map_channel 0.0.0 -map_channel -1 output.wav This will select the first channel, which I assume is your left channel. Adapted from http://ffmpeg.org/ffmpeg.html#Advanced-options From andreas.gumm at gmx.de Tue Mar 13 01:10:56 2012 From: andreas.gumm at gmx.de (Andreas Gumm) Date: Tue, 13 Mar 2012 01:10:56 +0100 Subject: [FFmpeg-user] ProRes multithreaded encoding In-Reply-To: References: <4F5E7766.90104@gmx.de> Message-ID: <4F5E9090.1050005@gmx.de> Thank you for answering, Paul! Hopefully it will be implemented in one of the next releases! Cheers Andreas Am 13.03.2012 00:12, schrieb Paul B Mahol: > On 3/12/12, Andreas Gumm wrote: >> Hello to the list, >> >> I have just a short question, >> is it possible to encode ProRes multithreaded? > It is possible, but currently not implemented. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From p.wallis at dcs.shef.ac.uk Tue Mar 13 04:49:16 2012 From: p.wallis at dcs.shef.ac.uk (Peter Wallis) Date: Tue, 13 Mar 2012 03:49:16 +0000 Subject: [FFmpeg-user] header information in generated avi file? Message-ID: Hi All, I'm trying to generate a video (mjpeg) from a jpeg image with sound. ffmpeg -loop_input -shortest -r 5 -i f01.jpg -i f01.wav -acodec pcm_s16le -ar 44100 -ac 1 -vcodec mjpeg -y f01.avi The resulting avi file works on most players (although there is in some cases confusion about when the stop playing) but, when I play it with an application I have written using JMF (I know ...) I get no sound and no image - although the duration is correct. The JMF application works on other avi files. I am assuming the above command is not producing some of the header information? Should I be using the -type option? If so what type do I want? I see there is an option for reading the data, producing a header and then adding it back in, but the manual does not say how to do it; just that it is possible. Any help much appreciated, peter From cehoyos at ag.or.at Tue Mar 13 08:06:47 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 13 Mar 2012 07:06:47 +0000 (UTC) Subject: [FFmpeg-user] loop References: <7C752A37-27EC-4C6F-825C-D49B0CD565FA@bushbaby.nl> <1543198C-57A5-45C7-8A82-B3F7D4F57134@bushbaby.nl> Message-ID: Bas Kamer bushbaby.nl> writes: > ffmpeg version 0.7.11, Copyright (c) 2000-2011 the FFmpeg developers old > FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers Ancient. (Additionally, all 0.6 releases contain a very high number of known bugs, and really should not be used at all.) Please try current git head, it contains more features and less bugs than any released version, Carl Eugen From cehoyos at ag.or.at Tue Mar 13 08:09:36 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 13 Mar 2012 07:09:36 +0000 (UTC) Subject: [FFmpeg-user] header information in generated avi file? References: Message-ID: Peter Wallis dcs.shef.ac.uk> writes: > ffmpeg -loop_input -shortest -r 5 -i f01.jpg -i f01.wav -acodec pcm_s16le > -ar 44100 -ac 1 -vcodec mjpeg -y f01.avi Complete, uncut output missing. Does WMP play the resulting file? > The resulting avi file works on most players (although there is in some > cases confusion about when the stop playing) but, when I play it with an > application I have written using JMF (I know ...) I get no sound and no > image - although the duration is correct. The JMF application works on > other avi files. Does it play the result of ffmpeg -loop_input -r 5 -i f01.jpg -vcodec mjpeg -vframes 100 video.avi ffmpeg -i f01.wav -acodec pcm_s16le audio.avi ? Carl Eugen From cehoyos at ag.or.at Tue Mar 13 08:12:14 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 13 Mar 2012 07:12:14 +0000 (UTC) Subject: [FFmpeg-user] Grabbing quality from TV card References: <919bb625a40b5bef37c86d3674bc288a.squirrel@nusi> Message-ID: tovis freemail.hu> writes: > I'm trying to grab video (at first) from a Pinacle PCTV Pro (TV + FM > stereo receiver) as lspci shows. I have some success, but the quality is > really wrong :( I have tested TV using fbtv and it was good enough, but > grabbing using command: > > ffmpeg -v 10 -s 640x480 -r 25 -f video4linux2 -i /dev/video0 out0.avi Since it is likely that the input is interlaced, please first try -qscale 2 out0.avi to test if this improves quality, then use yadif. Carl Eugen From bostjan.strojan at gmail.com Tue Mar 13 09:57:55 2012 From: bostjan.strojan at gmail.com (=?UTF-8?Q?Bo=C5=A1tjan_Strojan?=) Date: Tue, 13 Mar 2012 09:57:55 +0100 Subject: [FFmpeg-user] jpg quality? Message-ID: hello, a. What would a command line to convert a movie to jpg sequence with quality 50 look like? b. What would a command line to convert a movie to jpg sequence with quality 85 look like? p.s. If i have to post my full command like with full cli output, i will cry. Thanks, Bo?tjan From p.wallis at dcs.shef.ac.uk Tue Mar 13 10:44:39 2012 From: p.wallis at dcs.shef.ac.uk (Peter Wallis) Date: Tue, 13 Mar 2012 09:44:39 +0000 Subject: [FFmpeg-user] header information in generated avi file? In-Reply-To: References: Message-ID: Hi Carl, There does seem to be output using the ffmpeg command I described (I can look at it with a HEX editor) but it is a bit hard to use WMP because I'm on a linux box. Annoyingly, JMStudio plays it (bad timing problem mentioned) but the Java source for JMStudio is no longer supplied officially and the version I can get use beans technology rather than implementing things as described in the guide. producing video only and a fixed number of frames (15 at -r 5) produces no video. It is a little difficult to run the audio only through the code because it is designed to run avi files converted using ffmpeg - that is why I am producing video from stills actually. P On 13 March 2012 07:09, Carl Eugen Hoyos wrote: > Peter Wallis dcs.shef.ac.uk> writes: > > > ffmpeg -loop_input -shortest -r 5 -i f01.jpg -i f01.wav -acodec pcm_s16le > > -ar 44100 -ac 1 -vcodec mjpeg -y f01.avi > > Complete, uncut output missing. > Does WMP play the resulting file? > > > The resulting avi file works on most players (although there is in some > > cases confusion about when the stop playing) but, when I play it with an > > application I have written using JMF (I know ...) I get no sound and no > > image - although the duration is correct. The JMF application works on > > other avi files. > > Does it play the result of > ffmpeg -loop_input -r 5 -i f01.jpg -vcodec mjpeg -vframes 100 video.avi > ffmpeg -i f01.wav -acodec pcm_s16le audio.avi > ? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Tue Mar 13 10:59:49 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 13 Mar 2012 09:59:49 +0000 (UTC) Subject: [FFmpeg-user] jpg quality? References: Message-ID: Bo?tjan Strojan gmail.com> writes: > p.s. If i have to post my full command like with full cli output, > i will cry. Please do so;-) Carl Eugen From pavel at sokolov.me Tue Mar 13 11:14:52 2012 From: pavel at sokolov.me (Pavel Sokolov) Date: Tue, 13 Mar 2012 14:14:52 +0400 Subject: [FFmpeg-user] need fix: correct muxing to VOB with LPCM audio Message-ID: <4F5F1E1C.7080407@sokolov.me> Hi All! I got errors when I try to remux any file to VOB with LPCM by the following command: ffmpeg -i video-mpeg4_720x544-audio_ac3_48000_stereo.avi -vcodec copy -acodec pcm_s16be -f vob out.vob --------- vob @ 0x1745b80] buffer underflow i=1 bufi=4026 size=6144 Last message repeated 34 times [vob @ 0x1745b80] packet too large, ignoring buffer limits to mux it [vob @ 0x1745b80] buffer underflow i=1 bufi=4026 size=6144 [vob @ 0x1745b80] buffer underflow i=1 bufi=6043 size=6144 ... ... --------- VLC does not play the resulting file. Sample: http://sokolov.me/tmp/video-mpeg4_720x544-audio_ac3_48000_stereo.avi Examples with correct LPCM: (VLC play it without any problems) http://streams.videolan.org/samples/MPEG-VOB/LPCM/Fever.vob http://streams.videolan.org/samples/MPEG-VOB/LPCM/Kylie%20Minogue%20%5bGreatest%20Hits%20%2787-%2797%2021%5d%20-%20Confide%20In%20Me%20(5m56s%20VOB%20720x480%20LPCM).vob Can somebody fix this issue? I will pay for this work. -- With best regards, Pavel A. Sokolov mobile: +7(921)419-1819 skype: pavel_a_sokolov _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel From james.darnley at gmail.com Tue Mar 13 11:27:45 2012 From: james.darnley at gmail.com (James Darnley) Date: Tue, 13 Mar 2012 11:27:45 +0100 Subject: [FFmpeg-user] jpg quality? In-Reply-To: References: Message-ID: <4F5F2121.3010009@gmail.com> On 2012-03-13 09:57, Bo?tjan Strojan wrote: > hello, > > a. What would a command line to convert a movie to jpg sequence with > quality 50 look like? > b. What would a command line to convert a movie to jpg sequence with > quality 85 look like? What is "quality 50"? I think the quality option (qscale) for the mpeg encoder goes from 1 to 31. From tevans.uk at googlemail.com Tue Mar 13 11:33:18 2012 From: tevans.uk at googlemail.com (Tom Evans) Date: Tue, 13 Mar 2012 10:33:18 +0000 Subject: [FFmpeg-user] jpg quality? In-Reply-To: <4F5F2121.3010009@gmail.com> References: <4F5F2121.3010009@gmail.com> Message-ID: On Tue, Mar 13, 2012 at 10:27 AM, James Darnley wrote: > On 2012-03-13 09:57, Bo?tjan Strojan wrote: >> hello, >> >> a. What would a command line to convert a movie to jpg sequence with >> quality 50 look like? >> b. What would a command line to convert a movie to jpg sequence with >> quality 85 look like? > > What is "quality 50"? ?I think the quality option (qscale) for the mpeg > encoder goes from 1 to 31. He is talking about the output. JPEG pictures can be encoded with various degrees of quality, from 0-100. Cheers Tom From jacobhameiri at gmail.com Tue Mar 13 12:02:31 2012 From: jacobhameiri at gmail.com (jacob s) Date: Tue, 13 Mar 2012 13:02:31 +0200 Subject: [FFmpeg-user] scale input video before transcoding In-Reply-To: References: Message-ID: command: *ffmpeg -f dshow -i video="screen-capture-recorder":audio="SoundMAX Digital Audio" -vcodec libx264 -preset ultrafast -tune film -r 10 -async 1 -ab 32k -ar 22050 -f mpegts udp://192.168.5.215:48550* console output: http://pastebin.com/1zQKKZe4 command: *ffmpeg -f dshow -i video="screen-capture-recorder":audio="SoundMAX Digital Audio" -vf "scale=iw/2:ih/2" -vcodec libx264 -preset ultrafast -tune film -r 10 -async 1 -ab 32k -ar 22050 -f mpegts udp://192.168.5.215:48550* console output: http://pastebin.com/AG8juKkU > Did you test encoding to a file instead of udp? > What speed / CPU usage do you see? I tried encoding to file and cpu was the same ( around 65%) Can you help me reduce CPU usage, these Pentium 4 machines are weak in CPU power and it causes the stream to glitch and freeze, this is why I am trying to scale the input and hopefully it will reduce CPU usage ( no luck so far ) . 2012/3/12 Carl Eugen Hoyos > jacob s gmail.com> writes: > > > -f dshow -i video='screen-capture-recorder':audio='Stereo > > Mix (IDT High Definition' -vcodec libx264 -preset ultrafast > > -r 10 -async 1 -ab 32k -ar 22050 -f mpegts udp://192.168.2.100:1234 > > Complete, uncut console output missing. > > > When using this command ffmpeg uses about 65% CPU > > ( Pentium4 2.8 GHz ), and after a short while I am > > seeing glitches in the viewing machine. > > Note that the Pentium4 has known limitations > (that may or may not be the reason for your problems). > > > I thought of reducing CPU usage by scaling the input video > > to 50% - 75%, but even after scaling to 50% CPU usage doesn't > > change, I suspect this is > > Command line and complete, uncut console output missing. > > > because the scaling is performed after the transcoding process > > It is not possible to scale after transcoding > (all scaling has to be done after decoding and before encoding). > > Did you test encoding to a file instead of udp? > What speed / CPU usage do you see? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From sven at nablet.com Tue Mar 13 12:12:49 2012 From: sven at nablet.com (Sven Dueking) Date: Tue, 13 Mar 2012 12:12:49 +0100 Subject: [FFmpeg-user] VSYNC 1 Message-ID: <004e01cd010a$3a51c2f0$aef548d0$@com> Hi everybody, I have a simple question regarding the VSYNC=1 option. Is the assumption correct that VSYNC=1 results in dropped and duped frames, because ffmpeg cannot handle the incoming data according to the specified frame rate in realtime ? My workflow is quite simple, I have a ffmpeg instance running on a EC, input is X11 and ALSA and the task is to encode into h.264 in realtime (the data will be send as rtmp). I noticed a huge number of duped frames and want to be sure that my understanding is correct. Thanks ! Best, Sven From cehoyos at ag.or.at Tue Mar 13 13:29:14 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 13 Mar 2012 12:29:14 +0000 (UTC) Subject: [FFmpeg-user] VSYNC 1 References: <004e01cd010a$3a51c2f0$aef548d0$@com> Message-ID: Sven Dueking nablet.com> writes: > Is the assumption correct that VSYNC=1 results in dropped > and duped frames, Yes > because ffmpeg cannot handle the incoming data according to > the specified frame rate in realtime ? I don't think this is correct. Frames are duplicated and dropped to match the output stream's fps rate, this has not necessarily anything to do with realtime. Did you try to reduce the resolution significantly to test if it is performance issue? Please provide your command line together with complete, uncut console output. Carl Eugen From r.harakaly at gmail.com Tue Mar 13 13:36:34 2012 From: r.harakaly at gmail.com (Robert Harakaly) Date: Tue, 13 Mar 2012 13:36:34 +0100 Subject: [FFmpeg-user] live stream transcoding problem In-Reply-To: References: <1331284080.2913.17.camel@maserati> Message-ID: <1331642194.2301.48.camel@xfr> Carl, Andrey, I'm not sure if my last mail from Saturday passed. In fact it seems the bug still (partially) persist. I took the latest snapshot from you web site and I tried. There is critical period (first 60-90 frames) where it still fails as I described in my first mail. It looks like using multiple threads (option threads=2 or 4) makes this behaviour worse. When it passes this period it fails later but there the error messages vary. The fifo_size seems helps but after I get a huge amount of error mpeg2 messages. Any idea? Robert On Fri, 2012-03-09 at 16:05 +0200, Andrey Utkin wrote: > 2012/3/9 Carl Eugen Hoyos : > > Robert Harakaly gmail.com> writes: > > > >> ffmpeg version 0.9, Copyright (c) 2000-2011 the FFmpeg developers > > > > Please test current git head, it always contains less bugs and > > more features than any released version. > > (And there have been changes in the network code iirc.) > > > > Carl Eugen > > +1. > If still doesn't work - then this known bug is not fixed yet. Try > appending this to your udp input URL: > ?fifo_size=0 > If you get bad quality video after this, increase wmem_max sysctl > variable, and append sth like '&buffer_size=10000000' to input URL. > From sven at nablet.com Tue Mar 13 14:13:37 2012 From: sven at nablet.com (Sven Dueking) Date: Tue, 13 Mar 2012 14:13:37 +0100 Subject: [FFmpeg-user] VSYNC 1 In-Reply-To: References: <004e01cd010a$3a51c2f0$aef548d0$@com> Message-ID: <006101cd011b$1ad19520$5074bf60$@com> Hi Carl, Many thanks for your fast feedback. My assumption based on the different behavior if I use one or two transcodes. Same behavior if the instances are started parallel. And I also noticed a reduced number of dropped / duped frames with subme = 0 for instance. But maybe I made just a simple mistake .. Anyway, thanks again for your help and time. Best, Sven Command line is : ffmpeg \ -f alsa -ac 2 -i hw:0,1 \ -f x11grab -s 1280x720 -r 30 -g 30 -i :1.0 -vsync 1 -vcodec libx264 -aspect 16:9 -b:v 3000k -s 1280x720 -g 30 -r 30 \ -acodec libfaac -ac 2 -ar 44100 -ab 96k \ -x264opts no-mbtree:bframes=0:rc_lookahead=0:ref=2:subme=5 \ -f flv -y rtmp://178.63.185.152:1936/live/channel1 \ -vcodec libx264 -aspect 16:9 -b:v 2000k -s 852x480 -g 30 -r 30 \ -x264opts no-mbtree:bframes=0:rc_lookahead=0:ref=2:subme=5 \ -f flv -y rtmp://178.63.185.152:1936/live/channel2 Console output : root at ip-10-248-105-212:~# ./multi.sh ffmpeg version 0.10 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 9 2012 15:14:17 with gcc 4.6.1 configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-pthreads --enable-libfaac --enable-libx264 --enable-x11grab --enable-librtmp --enable-postproc --enable-bzlib --enable-zlib --enable-swscale libavutil 51. 34.101 / 51. 34.101 libavcodec 53. 60.100 / 53. 60.100 libavformat 53. 31.100 / 53. 31.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 60.100 / 2. 60.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 52. 0.100 / 52. 0.100 [alsa @ 0x1841420] Estimating duration from bitrate, this may be inaccurate Input #0, alsa, from 'hw:0,1': Duration: N/A, start: 1331644039.931867, bitrate: N/A Stream #0:0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s [x11grab @ 0x18581a0] device: :1.0 -> display: :1.0 x: 0 y: 0 width: 1280 height: 720 [x11grab @ 0x18581a0] shared memory extension found [x11grab @ 0x18581a0] Estimating duration from bitrate, this may be inaccurate Input #1, x11grab, from ':1.0': Duration: N/A, start: 1331644039.971963, bitrate: 884736 kb/s Stream #1:0: Video: rawvideo (BGRA / 0x41524742), bgra, 1280x720, 884736 kb/s, 30 tbr, 1000k tbn, 30 tbc HandShake: client signature does not match! HandShake: client signature does not match! Incompatible pixel format 'bgra' for codec 'libx264', auto-selecting format 'yuv420p' [buffer @ 0x1865640] w:1280 h:720 pixfmt:bgra tb:1/1000000 sar:0/1 sws_param: [buffersink @ 0x1896620] auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'out' [scale @ 0x1897f00] w:1280 h:720 fmt:bgra -> w:1280 h:720 fmt:yuv420p flags:0x4 Incompatible pixel format 'bgra' for codec 'libx264', auto-selecting format 'yuv420p' [buffer @ 0x1896c60] w:1280 h:720 pixfmt:bgra tb:1/1000000 sar:0/1 sws_param: [scale @ 0x1901780] w:1280 h:720 fmt:bgra -> w:852 h:480 fmt:yuv420p flags:0x4 [libx264 @ 0x183fb40] using SAR=1/1 [libx264 @ 0x183fb40] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x183fb40] profile High, level 3.1 [libx264 @ 0x183fb40] 264 - core 120 r2164 da19765 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=2 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=5 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=12 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=2 keyint=30 keyint_min=3 scenecut=40 intra_refresh=0 rc=abr mbtree=0 bitrate=3000 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 [libx264 @ 0x186be20] using SAR=1/1 [libx264 @ 0x186be20] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x186be20] profile High, level 3.1 [libx264 @ 0x186be20] 264 - core 120 r2164 da19765 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=2 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=5 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=12 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=2 keyint=30 keyint_min=3 scenecut=40 intra_refresh=0 rc=abr mbtree=0 bitrate=2000 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, flv, to 'rtmp://178.63.185.152:1936/live/channel1': Metadata: encoder : Lavf53.31.100 Stream #0:0: Video: h264 ([7][0][0][0] / 0x0007), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 3000 kb/s, 1k tbn, 30 tbc Stream #0:1: Audio: aac ([10][0][0][0] / 0x000A), 44100 Hz, 2 channels, s16, 96 kb/s Output #1, flv, to 'rtmp://178.63.185.152:1936/live/channel2': Metadata: encoder : Lavf53.31.100 Stream #1:0: Video: h264 ([7][0][0][0] / 0x0007), yuv420p, 852x480 [SAR 1:1 DAR 71:40], q=-1--1, 2000 kb/s, 1k tbn, 30 tbc Stream #1:1: Audio: adpcm_swf ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 352 kb/s Stream mapping: Stream #1:0 -> #0:0 (rawvideo -> libx264) Stream #0:0 -> #0:1 (pcm_s16le -> libfaac) Stream #1:0 -> #1:0 (rawvideo -> libx264) Stream #0:0 -> #1:1 (pcm_s16le -> adpcm_swf) Press [q] to stop, [?] for help ALSA buffer xrun.31 q=11.0 q=13.0 size= 2501kB time=00:00:03.64 bitrate=5620.4kbits/s dup=162 drop=0 frame= 4419 fps= 30 q=29.0 Lq=28.0 size= 57394kB time=00:02:25.82 bitrate=3224.3kbits/s dup=2430 drop=0 video:95377kB audio:6373kB global headers:0kB muxing overhead -43.593247% [libx264 @ 0x183fb40] frame I:153 Avg QP:22.20 size: 99281 [libx264 @ 0x183fb40] frame P:4266 Avg QP:27.03 size: 10151 [libx264 @ 0x183fb40] mb I I16..4: 27.8% 14.7% 57.5% [libx264 @ 0x183fb40] mb P I16..4: 1.5% 1.3% 1.0% P16..4: 16.4% 11.4% 5.6% 0.0% 0.0% skip:62.7% [libx264 @ 0x183fb40] final ratefactor: 23.03 [libx264 @ 0x183fb40] 8x8 transform intra:24.8% inter:23.1% [libx264 @ 0x183fb40] coded y,uvDC,uvAC intra: 59.0% 53.1% 29.1% inter: 11.7% 7.8% 0.7% [libx264 @ 0x183fb40] i16 v,h,dc,p: 69% 19% 8% 3% [libx264 @ 0x183fb40] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 19% 21% 17% 8% 5% 7% 5% 10% 8% [libx264 @ 0x183fb40] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 25% 18% 10% 8% 8% 9% 7% 8% 8% [libx264 @ 0x183fb40] i8c dc,h,v,p: 53% 21% 19% 7% [libx264 @ 0x183fb40] Weighted P-Frames: Y:0.5% UV:0.3% [libx264 @ 0x183fb40] ref P L0: 78.2% 15.1% 6.8% 0.0% [libx264 @ 0x183fb40] kb/s:3176.96 [libx264 @ 0x186be20] frame I:153 Avg QP:21.38 size: 64168 [libx264 @ 0x186be20] frame P:4266 Avg QP:25.79 size: 6880 [libx264 @ 0x186be20] mb I I16..4: 24.1% 8.5% 67.3% [libx264 @ 0x186be20] mb P I16..4: 0.4% 0.6% 1.1% P16..4: 18.1% 13.3% 9.2% 0.0% 0.0% skip:57.4% [libx264 @ 0x186be20] final ratefactor: 21.19 [libx264 @ 0x186be20] 8x8 transform intra:15.6% inter:20.1% [libx264 @ 0x186be20] coded y,uvDC,uvAC intra: 72.3% 70.7% 47.8% inter: 17.0% 9.9% 1.7% [libx264 @ 0x186be20] i16 v,h,dc,p: 57% 23% 16% 4% [libx264 @ 0x186be20] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 19% 19% 19% 7% 5% 6% 5% 10% 9% [libx264 @ 0x186be20] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 17% 11% 8% 8% 9% 7% 9% 8% [libx264 @ 0x186be20] i8c dc,h,v,p: 48% 24% 19% 9% [libx264 @ 0x186be20] Weighted P-Frames: Y:0.5% UV:0.3% [libx264 @ 0x186be20] ref P L0: 79.2% 13.6% 7.2% 0.0% [libx264 @ 0x186be20] kb/s:2127.33 root at ip-10-248-105-212:~# -----Urspr?ngliche Nachricht----- Von: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] Im Auftrag von Carl Eugen Hoyos Gesendet: Dienstag, 13. M?rz 2012 13:29 An: ffmpeg-user at ffmpeg.org Betreff: Re: [FFmpeg-user] VSYNC 1 Sven Dueking nablet.com> writes: > Is the assumption correct that VSYNC=1 results in dropped and duped > frames, Yes > because ffmpeg cannot handle the incoming data according to the > specified frame rate in realtime ? I don't think this is correct. Frames are duplicated and dropped to match the output stream's fps rate, this has not necessarily anything to do with realtime. Did you try to reduce the resolution significantly to test if it is performance issue? Please provide your command line together with complete, uncut console output. Carl Eugen _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From jacobhameiri at gmail.com Tue Mar 13 14:39:19 2012 From: jacobhameiri at gmail.com (jacob s) Date: Tue, 13 Mar 2012 15:39:19 +0200 Subject: [FFmpeg-user] play only left/right channel from an audio input In-Reply-To: <20120312160604.0b760440@lrcd.com> References: <20120312151734.4693268d@lrcd.com> <20120312160604.0b760440@lrcd.com> Message-ID: I have 2 steams, one video and one audio ( stereo ) and I want the output to have 2 streams one video and one stereo with the left ( or right ) channel muted, what is the correct command. This is my original command: -f dshow -i video="screen-capture-recorder":audio="Stereo Mix (IDT High Definition" -vcodec libx264 -preset ultrafast -tune film -r 10 -async 1 -ab 32k -ar 22050 -f mpegts udp://192.168.5.230:48551 ffmpeg version N-38292-ga4c22e3 Copyright (c) 2000-2012 the FFmpeg developers built on Feb 27 2012 14:50:39 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 41.100 / 51. 41.100 libavcodec 54. 4.100 / 54. 4.100 libavformat 54. 1.100 / 54. 1.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 62.101 / 2. 62.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 [dshow @ 01BE8AA0] Estimating duration from bitrate, this may be inaccurate Input #0, dshow, from 'video=screen-capture-recorder:audio=Stereo Mix (IDT High Definition': Duration: N/A, start: 20641.769000, bitrate: 1411 kb/s Stream #0:0: Video: rawvideo, bgr24, 1680x1050, 25 tbr, 10000k tbn, 25 tbc Stream #0:1: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s Incompatible pixel format 'bgr24' for codec 'libx264', auto-selecting format 'yuv420p' [buffer @ 01C509A0] w:1680 h:1050 pixfmt:bgr24 tb:1/1000000 sar:0/1 sws_param: [buffersink @ 01C6E960] auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'out' [scale @ 01C6EBE0] w:1680 h:1050 fmt:bgr24 -> w:1680 h:1050 fmt:yuv420p flags:0x4 [libx264 @ 01C51180] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 01C51180] profile Constrained Baseline, level 4.0 [mpegts @ 01C50A00] muxrate VBR, pcr every 1 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to 'udp://192.168.5.230:48551': Metadata: encoder : Lavf54.1.100 Stream #0:0: Video: h264, yuv420p, 1680x1050, q=-1--1, 90k tbn, 10 tbc Stream #0:1: Audio: mp2, 22050 Hz, 2 channels, s16, 32 kb/s Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> libx264) Stream #0:1 -> #0:1 (pcm_s16le -> mp2) Press [q] to stop, [?] for help frame= 8 fps= 0 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s dup=0 drop=7 frame= 13 fps= 12 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s dup=0 drop=15 frame= 18 fps= 11 q=10.0 size= 351kB time=00:00:00.50 bitrate=5751.3kbits/s dup=0 drop=23 frame= 23 fps= 11 q=0.0 size= 577kB time=00:00:01.00 bitrate=4728.6kbits/s dup=0 drop=31 frame= 28 fps= 11 q=8.0 size= 759kB time=00:00:01.50 bitrate=4144.0kbits/s dup=0 drop=39 frame= 34 fps= 11 q=0.0 size= 813kB time=00:00:02.10 bitrate=3169.9kbits/s dup=0 drop=46 frame= 34 fps= 10 q=-1.0 Lsize= 881kB time=00:00:03.03 bitrate=2382.4kbits/s dup=0 drop=48 video:801kB audio:12kB global headers:0kB muxing overhead 8.460626% [libx264 @ 01C51180] frame I:1 Avg QP:20.00 size:332259 [libx264 @ 01C51180] frame P:33 Avg QP: 8.12 size: 14777 [libx264 @ 01C51180] mb I I16..4: 100.0% 0.0% 0.0% [libx264 @ 01C51180] mb P I16..4: 1.2% 0.0% 0.0% P16..4: 9.1% 0.0% 0.0% 0.0% 0.0% skip:89.7% [libx264 @ 01C51180] coded y,uvDC,uvAC intra: 34.1% 14.3% 9.9% inter: 4.4% 3.4% 3.4% [libx264 @ 01C51180] i16 v,h,dc,p: 60% 38% 1% 1% [libx264 @ 01C51180] i8c dc,h,v,p: 67% 22% 12% 0% [libx264 @ 01C51180] kb/s:1929.15 [dshow @ 01BE8AA0] real-time buffer 348% full! frame dropped! Last message repeated 1 times Received signal 2: terminating. 2012/3/13 Lou > On Tue, 13 Mar 2012 01:30:01 +0200 > jacob s wrote: > > > 2012/3/13 Lou > > > > > On Tue, 13 Mar 2012 00:56:28 +0200 > > > jacob s wrote: > > > > > > > Hello, > > > > > > > > Given a stereo audio input, is it possible to filter only left/right > > > > channel ? > > > > For example if I have a file with 2 audio channels can I create a new > > > file > > > > that has the left channel audio only ? > > > > > > We need more information; "a new file that has the left channel audio > > > only" can be interpreted as: > > > > > > * A mono output. > > > * A stereo output with input left audio in both channels of the output. > > > * A stereo output with input left audio in the left channel and silence > > > in the right channel of the output. > > > > - A stereo output with input left audio in the left channel and silence > in > > the right channel of the output. > > ffmpeg -i input.wav -map_channel 0.0.0 -map_channel -1 output.wav > > This will select the first channel, which I assume is your left > channel. Adapted from http://ffmpeg.org/ffmpeg.html#Advanced-options > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From osoleil at ubisoft.fr Tue Mar 13 17:01:07 2012 From: osoleil at ubisoft.fr (Olivier Soleil) Date: Tue, 13 Mar 2012 17:01:07 +0100 Subject: [FFmpeg-user] How can I replace numbered sound tracks in my AVI ? Message-ID: <524EA12FEC145E4E86154A23DA429D2D5DE588AFD3@PDC-MAIL-CMS01.ubisoft.org> Hello everyone, this is my first post here, hope it's gonna be interesting and fruitful. Let's say that I have to produce CGI movies for a game, and voiced tracks (ENG, FRA, ...) are located above stream 100. So, I have a video track on the 0 stream, surround ambient sounds located between 1-5 streams, and voiced tracks above 100th stream. Well, it may look crazy, but why not ? Every time the soundmix for a language is updated, I would like to replace the old one with the new one, without changing the rest of the data in the AVI. Working this way is very easy with Bink, but I don't see how the available options of FFMPEG would allow this kind of manipulations. I still hope someone here can answer me. Please ? Thank you for your attention. From mlefe74 at gmail.com Tue Mar 13 16:42:00 2012 From: mlefe74 at gmail.com (LsBender) Date: Tue, 13 Mar 2012 08:42:00 -0700 (PDT) Subject: [FFmpeg-user] blackframe and blackdetect Message-ID: <1331653320238-4469396.post@n4.nabble.com> Hello, I'm new with ffmpeg and your help is welcome I would like to have a line command that allows me to detect black-frames in my avi files. And, if it's possible, I would like to be inform (with timestamps?) when black-frames occurr in the video and their duration. Thanks a lot -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/blackframe-and-blackdetect-tp4469396p4469396.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From mailer.tovis at freemail.hu Tue Mar 13 20:55:12 2012 From: mailer.tovis at freemail.hu (tovis) Date: Tue, 13 Mar 2012 20:55:12 +0100 Subject: [FFmpeg-user] Grabbing quality from TV card In-Reply-To: <201203130955.36230.rodney.baker@iinet.net.au> References: <919bb625a40b5bef37c86d3674bc288a.squirrel@nusi> <201203130924.13604.rodney.baker@iinet.net.au> <4b2f24783f90e3d4b90c8469cce6542e.squirrel@nusi> <201203130955.36230.rodney.baker@iinet.net.au> Message-ID: Hi Rodney! Much better, but not good enough :( Quality is still wrong, does not change against the very old Debianized version. On time I have a lot of error messages: $ ffmpeg -v verbose -f video4linux2 -i /dev/video0 -s 640x480 -r 25 out10.avi 2>&1 | tee ffmpeg-grab.log The log looks like this: ffmpeg version N-38750-g599888a Copyright (c) 2000-2012 the FFmpeg developers built on Mar 12 2012 23:50:31 with gcc 4.4.5 configuration: --enable-nonfree --enable-libv4l2 libavutil 51. 42.100 / 51. 42.100 libavcodec 54. 10.100 / 54. 10.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 64.101 / 2. 64.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 [video4linux2,v4l2 @ 0x30f13e0] [3]Capabilities: 5010015 [video4linux2,v4l2 @ 0x30f13e0] Querying the device for the current frame size [video4linux2,v4l2 @ 0x30f13e0] Setting frame size to 320x240 libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 1174 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable [video4linux2,v4l2 @ 0x30f13e0] Estimating duration from bitrate, this may be inaccurate Input #0, video4linux2,v4l2, from '/dev/video0': Duration: N/A, start: 1331666886.562734, bitrate: 23040 kb/s Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 320x240, 23040 kb/s, 25 tbr, 1000k tbn, 25 tbc [buffer @ 0x30f9020] w:320 h:240 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param: [scale @ 0x30f6e40] w:320 h:240 fmt:yuv420p -> w:640 h:480 fmt:yuv420p flags:0x4 Output #0, avi, to 'out10.avi': Metadata: ISFT : Lavf54.2.100 Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 640x480, q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> mpeg4) Press [q] to stop, [?] for help libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 18 fps=0.0 q=31.0 size= 183kB time=00:00:00.72 bitrate=2081.3kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 31 fps= 29 q=31.0 size= 202kB time=00:00:01.24 bitrate=1333.2kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 44 fps= 28 q=31.0 size= 220kB time=00:00:01.76 bitrate=1025.8kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 57 fps= 27 q=31.0 size= 239kB time=00:00:02.28 bitrate= 858.0kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 70 fps= 27 q=31.0 size= 255kB time=00:00:02.80 bitrate= 745.1kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 83 fps= 26 q=31.0 size= 272kB time=00:00:03.32 bitrate= 672.0kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 96 fps= 26 q=31.0 size= 290kB time=00:00:03.84 bitrate= 619.7kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 109 fps= 26 q=24.8 size= 312kB time=00:00:04.36 bitrate= 586.2kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 122 fps= 26 q=31.0 size= 331kB time=00:00:04.88 bitrate= 555.1kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 135 fps= 26 q=31.0 size= 349kB time=00:00:05.40 bitrate= 530.0kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 148 fps= 26 q=31.0 size= 366kB time=00:00:05.92 bitrate= 506.9kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 161 fps= 26 q=31.0 size= 386kB time=00:00:06.44 bitrate= 490.5kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 174 fps= 26 q=31.0 size= 403kB time=00:00:06.96 bitrate= 474.8kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 187 fps= 26 q=31.0 size= 419kB time=00:00:07.48 bitrate= 458.5kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 200 fps= 26 q=31.0 size= 438kB time=00:00:08.00 bitrate= 448.8kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 213 fps= 26 q=31.0 size= 454kB time=00:00:08.52 bitrate= 436.7kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 44 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 226 fps= 26 q=31.0 size= 468kB time=00:00:09.04 bitrate= 424.0kbits/s libv4l2: error dequeuing buf: Resource temporarily unavailable -- repeated 15 times -- libv4l2: error dequeuing buf: Resource temporarily unavailable frame= 230 fps= 25 q=31.0 Lsize= 480kB time=00:00:09.20 bitrate= 427.2kbits/s video:469kB audio:0kB global headers:0kB muxing overhead 2.358736% May be I should not have to compile with libv4l2? What is bother me that may be I should have to set another input? Any farther suggestions? Sincerely tovis > On Tue, 13 Mar 2012 09:42:03 tovis wrote: >> Hi Rodney! >> Thanks for your quick answer. >> I have built another system install development packages (headers gcc >> make >> ...) get latest git version. But not even worth :( >> This is the command: >> >> $ ffmpeg -v verbose -i /dev/video0 -f video4linux2 -s 640x480 -r 25 >> out10.avi 2>&1 | tee -a ffmpeg-grab.log >> >> and this is the console output: >> ffmpeg version N-38750-g599888a Copyright (c) 2000-2012 the FFmpeg >> developers built on Mar 12 2012 23:50:31 with gcc 4.4.5 >> configuration: --enable-nonfree --enable-libv4l2 >> libavutil 51. 42.100 / 51. 42.100 >> libavcodec 54. 10.100 / 54. 10.100 >> libavformat 54. 2.100 / 54. 2.100 >> libavdevice 53. 4.100 / 53. 4.100 >> libavfilter 2. 64.101 / 2. 64.101 >> libswscale 2. 1.100 / 2. 1.100 >> libswresample 0. 7.100 / 0. 7.100 >> /dev/video0: Invalid data found when processing input >> >> What could be missed? >> Sincerely > > Ok, I missed something. Try -f video4linux2 -i /dev/video0... > > -- > =================================================== > Rodney Baker VK5ZTV > rodney.baker at iinet.net.au > =================================================== > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From bas at bushbaby.nl Tue Mar 13 22:33:15 2012 From: bas at bushbaby.nl (Bas Kamer) Date: Tue, 13 Mar 2012 22:33:15 +0100 Subject: [FFmpeg-user] loop In-Reply-To: References: <7C752A37-27EC-4C6F-825C-D49B0CD565FA@bushbaby.nl> <1543198C-57A5-45C7-8A82-B3F7D4F57134@bushbaby.nl> Message-ID: On 13 mrt. 2012, at 08:06, Carl Eugen Hoyos wrote: > Bas Kamer bushbaby.nl> writes: > >> ffmpeg version 0.7.11, Copyright (c) 2000-2011 the FFmpeg developers > > old > >> FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers > > Ancient. > (Additionally, all 0.6 releases contain a very high number of known bugs, > and really should not be used at all.) > > Please try current git head, it contains more features and less bugs > than any released version, Carl Eugen Agreed too old. Asked my provider to install from source 0.10 but still not looping :-( the example command does give me a .mov with 100 frames contained in it (that seems looped input?) but these 100 frames are not looped when opened in QT? http://pastebin.com/Z2SJ1mYt I still wonder if is possible at all what I try to accomplish here. Can ffmpeg generate QT mov containers with the loop option set? Bas From p.wallis at dcs.shef.ac.uk Wed Mar 14 02:04:34 2012 From: p.wallis at dcs.shef.ac.uk (Peter Wallis) Date: Wed, 14 Mar 2012 01:04:34 +0000 Subject: [FFmpeg-user] header information in generated avi file? In-Reply-To: References: Message-ID: Should I expect an avi file to have "global headers:0kB" ? Lets assume it works on WMP, just not on my flakey code ... just can't figure out why. P On 13 March 2012 03:49, Peter Wallis wrote: > Hi All, > I'm trying to generate a video (mjpeg) from a jpeg image with sound. > > ffmpeg -loop_input -shortest -r 5 -i f01.jpg -i f01.wav -acodec pcm_s16le > -ar 44100 -ac 1 -vcodec mjpeg -y f01.avi > > The resulting avi file works on most players (although there is in some > cases confusion about when the stop playing) but, when I play it with an > application I have written using JMF (I know ...) I get no sound and no > image - although the duration is correct. The JMF application works on > other avi files. > > I am assuming the above command is not producing some of the header > information? Should I be using the -type option? If so what type do I > want? I see there is an option for reading the data, producing a header > and then adding it back in, but the manual does not say how to do it; just > that it is possible. > > Any help much appreciated, > > peter > From p.wallis at dcs.shef.ac.uk Wed Mar 14 02:12:56 2012 From: p.wallis at dcs.shef.ac.uk (Peter Wallis) Date: Wed, 14 Mar 2012 01:12:56 +0000 Subject: [FFmpeg-user] header information in generated avi file? In-Reply-To: References: Message-ID: Solved the frame rate was too low. using -r 25 rather than -r 5 works I am going to assume that, indeed, avi files do not have a header (!) and my Java JMF code is not prefetching properly. cheers, P On 14 March 2012 01:04, Peter Wallis wrote: > Should I expect an avi file to have "global headers:0kB" ? > > Lets assume it works on WMP, just not on my flakey code ... just can't > figure out why. > > P > > > On 13 March 2012 03:49, Peter Wallis wrote: > >> Hi All, >> I'm trying to generate a video (mjpeg) from a jpeg image with sound. >> >> ffmpeg -loop_input -shortest -r 5 -i f01.jpg -i f01.wav -acodec pcm_s16le >> -ar 44100 -ac 1 -vcodec mjpeg -y f01.avi >> >> The resulting avi file works on most players (although there is in some >> cases confusion about when the stop playing) but, when I play it with an >> application I have written using JMF (I know ...) I get no sound and no >> image - although the duration is correct. The JMF application works on >> other avi files. >> >> I am assuming the above command is not producing some of the header >> information? Should I be using the -type option? If so what type do I >> want? I see there is an option for reading the data, producing a header >> and then adding it back in, but the manual does not say how to do it; just >> that it is possible. >> >> Any help much appreciated, >> >> peter >> > > From funkyirish at gmail.com Wed Mar 14 04:54:02 2012 From: funkyirish at gmail.com (Josh long) Date: Tue, 13 Mar 2012 22:54:02 -0500 Subject: [FFmpeg-user] bgra to yuv In-Reply-To: References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: On Sat, Mar 3, 2012 at 1:11 AM, Carl Eugen Hoyos wrote: > Josh long gmail.com> writes: > > > I've been given raw bgra files from a client in a school project > > and want to compress them with dirac, so that I can later > > parallelize the code. > > ffmpeg -i input -vcodec libschroedinger out.avi > > But please post the command line you tried together with complete, > uncut console output and explain what goes wrong. > > Please do not top-post here, it is considered rude, Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Sorry about the top post. I think it may shed some light in this matter if I briefly describe my setup and goals. I'm trying to compress bgra format avi files with dirac/schodinger. Some things, like the suggested command above. However, when I try to play any of the resulting files in VLC I get garbage. Here is my ffmpeg version and info: ffmpeg version 0.7.3-4:0.7.3-0ubuntu0.11.10.1, Copyright (c) 2000-2011 the Libav developers built on Jan 4 2012 16:21:50 with gcc 4.6.1 configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static WARNING: library configuration mismatch avutil configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avcodec configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avformat configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avdevice configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avfilter configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay swscale configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay postproc configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libavutil 51. 7. 0 / 51. 7. 0 libavcodec 53. 6. 0 / 53. 6. 0 libavformat 53. 3. 0 / 53. 3. 0 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 4. 0 / 2. 4. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 52. 0. 0 / 52. 0. 0 ffmpeg 0.7.3-4:0.7.3-0ubuntu0.11.10.1 libavutil 51. 7. 0 / 51. 7. 0 libavcodec 53. 6. 0 / 53. 6. 0 libavformat 53. 3. 0 / 53. 3. 0 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 4. 0 / 2. 4. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 52. 0. 0 / 52. 0. 0 I seem to have no problem using dirac_encoder to compress yuv files, so I tried to convert my files first using this command: bn408-3-2 at bn40832-Dimension-8200:~$ ffmpeg -i 0_10_sec.avi -vcodec rawvideo -pix_fmt yuv444p test.nut ffmpeg version 0.7.3-4:0.7.3-0ubuntu0.11.10.1, Copyright (c) 2000-2011 the Libav developers built on Jan 4 2012 16:21:50 with gcc 4.6.1 configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static WARNING: library configuration mismatch avutil configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avcodec configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avformat configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avdevice configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avfilter configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay swscale configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay postproc configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libavutil 51. 7. 0 / 51. 7. 0 libavcodec 53. 6. 0 / 53. 6. 0 libavformat 53. 3. 0 / 53. 3. 0 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 4. 0 / 2. 4. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 52. 0. 0 / 52. 0. 0 Input #0, avi, from '0_10_sec.avi': Duration: 00:00:10.00, start: 0.000000, bitrate: 1990665 kb/s Stream #0.0: Video: rawvideo, bgra, 1920x1080, 30 tbr, 30 tbn, 30 tbc File 'test.nut' already exists. Overwrite ? [y/N] y [buffer @ 0x8d7b080] w:1920 h:1080 pixfmt:bgra [ffsink @ 0x8d7a520] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' [scale @ 0x8d7a980] w:1920 h:1080 fmt:bgra -> w:1920 h:1080 fmt:yuv444p flags:0x4 Output #0, nut, to 'test.nut': Metadata: encoder : Lavf53.3.0 Stream #0.0: Video: rawvideo, yuv444p, 1920x1080, q=2-31, 200 kb/s, 30 tbn, 30 tbc Stream mapping: Stream #0.0 -> #0.0 Press ctrl-c to stop encoding frame= 300 fps= 2 q=0.0 Lsize= 1822509kB time=10.00 bitrate=1492999.0kbits/s video:1822500kB audio:0kB global headers:0kB muxing overhead 0.000469% this seems to work fine, but again VLC gives me garbage. Now to be truthful, the computer this has been run on so far is a pentium 4 with 1 gig of ram, so I'm not convinced that the encoding isn't working properly. Any direction at this point would be wonderful, thanks in advance. From joolzg at btinternet.com Wed Mar 14 07:11:00 2012 From: joolzg at btinternet.com (JULIAN GARDNER) Date: Wed, 14 Mar 2012 06:11:00 +0000 (GMT) Subject: [FFmpeg-user] Playlist input and Mosaic output Message-ID: <1331705460.5440.YahooMailNeo@web87706.mail.ir2.yahoo.com> I was wondering if its possible to have ffmpeg encode a playlist to produce a single output, either file or udp stream? would like to use it for building a "barker" channel. Also ive been trying to get VLC mosaic working and all i have managed to do is get a black screen, so is there any way of using ffmpeg for creating a mosaic screen from live sources? joolz From mlefe74 at gmail.com Wed Mar 14 10:11:50 2012 From: mlefe74 at gmail.com (LsBender) Date: Wed, 14 Mar 2012 02:11:50 -0700 (PDT) Subject: [FFmpeg-user] blackframe and blackdetect In-Reply-To: <1331653320238-4469396.post@n4.nabble.com> References: <1331653320238-4469396.post@n4.nabble.com> Message-ID: <1331716310370-4471324.post@n4.nabble.com> I had a look at: http://ffmpeg.org/ffmpeg.html#blackframe http://ffmpeg.org/ffmpeg.html#blackframe That's why I tried something like that: /ffmpeg blackframe=98:32 test.avi/ But I have this error: / lefebvre at lefebvre-laptop:~/Bureau$ ffmpeg blackframe=98:32 test.avi FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1.3, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.1-1ubuntu1.3 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Dec 21 2011 18:37:21, gcc: 4.4.3 Unable to find a suitable output format for 'blackframe=98:32'/ I didn't understand this line of the ffmpeg documentation : "In order to display the output lines, you need to set the loglevel at least to the AV_LOG_INFO value" Thanks for your time -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/blackframe-and-blackdetect-tp4469396p4471324.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From ubitux at gmail.com Wed Mar 14 10:38:03 2012 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Wed, 14 Mar 2012 10:38:03 +0100 Subject: [FFmpeg-user] blackframe and blackdetect In-Reply-To: <1331716310370-4471324.post@n4.nabble.com> References: <1331653320238-4469396.post@n4.nabble.com> <1331716310370-4471324.post@n4.nabble.com> Message-ID: <20120314093803.GB26897@leki> On Wed, Mar 14, 2012 at 02:11:50AM -0700, LsBender wrote: > I had a look at: > http://ffmpeg.org/ffmpeg.html#blackframe > http://ffmpeg.org/ffmpeg.html#blackframe > That's why I tried something like that: > /ffmpeg blackframe=98:32 test.avi/ > You may try ffmpeg -i test.avi -vf blackframe=... -f null - [...] -- Cl?ment B. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 490 bytes Desc: not available URL: From mlefe74 at gmail.com Wed Mar 14 11:07:31 2012 From: mlefe74 at gmail.com (LsBender) Date: Wed, 14 Mar 2012 03:07:31 -0700 (PDT) Subject: [FFmpeg-user] blackframe and blackdetect In-Reply-To: <20120314093803.GB26897@leki> References: <1331653320238-4469396.post@n4.nabble.com> <1331716310370-4471324.post@n4.nabble.com> <20120314093803.GB26897@leki> Message-ID: <1331719651679-4471452.post@n4.nabble.com> Thank for your help Clement, I tried your command but i have this: "unrecognized option '-vf'": / lefebvre at lefebvre-laptop:~/Bureau$ ffmpeg -i test.avi -vf blackframe=98:32 -f null- FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1.3, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.1-1ubuntu1.3 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Dec 21 2011 18:37:21, gcc: 4.4.3 Input #0, avi, from 'test.avi': Duration: 01:02:12.22, start: 0.000000, bitrate: 1012 kb/s Stream #0.0: Video: mpeg4, yuv420p, 704x576 [PAR 1:1 DAR 11:9], 12.50 tbr, 12.50 tbn, 12.50 tbc Stream #0.1: Audio: mp2, 32000 Hz, stereo, s16, 64 kb/s ffmpeg: unrecognized option '-vf'/ I already saw some post about blackframe and how to list them, but there are listing blackframe while ffmpeg is encoding. And in my case, I don't want to encode. I just want to collect blackframe informations about my video without encoding. Is it possible?! Thanks, Marc -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/blackframe-and-blackdetect-tp4469396p4471452.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From ubitux at gmail.com Wed Mar 14 11:58:07 2012 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Wed, 14 Mar 2012 11:58:07 +0100 Subject: [FFmpeg-user] blackframe and blackdetect In-Reply-To: <1331719651679-4471452.post@n4.nabble.com> References: <1331653320238-4469396.post@n4.nabble.com> <1331716310370-4471324.post@n4.nabble.com> <20120314093803.GB26897@leki> <1331719651679-4471452.post@n4.nabble.com> Message-ID: <20120314105807.GC26897@leki> On Wed, Mar 14, 2012 at 03:07:31AM -0700, LsBender wrote: > Thank for your help Clement, > I tried your command but i have this: "unrecognized option '-vf'": FFmpeg 0.5 was release years ago, you need a recent version for the video filters, and even more recent for the blackframe filter. Check http://ffmpeg.org/download.html and https://launchpad.net/~jon-severinsson/+archive/ffmpeg [...] -- Cl?ment B. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 490 bytes Desc: not available URL: From auguste at gmail.com Wed Mar 14 14:18:11 2012 From: auguste at gmail.com (Auguste Pop) Date: Wed, 14 Mar 2012 21:18:11 +0800 Subject: [FFmpeg-user] how do i strip away all metadata from input source? Message-ID: hi, i was using -map_metadata -1 -map_metadata:s:0 -1 -map_metadata:s:1 -1, but it does not work now. every metadata from the source is copied over to the output file. i have tried to google it, but it only came up with some answer with older syntax that ffmpeg no long use. i am using archlinux $ ffmpeg -version ffmpeg version N-37208-g01fcbdf built on Feb 7 2012 10:36:16 with gcc 4.6.2 20120120 (prerelease) configuration: --prefix=/usr --enable-libmp3lame --enable-libvorbis --enable-libxvid --enable-libx264 --enable-libvpx --enable-libtheora --enable-libgsm --enable-libspeex --enable-postproc --enable-shared --enable-x11grab --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libschroedinger --enable-libopenjpeg --enable-librtmp --enable-libpulse --enable-gpl --enable-version3 --enable-runtime-cpudetect --disable-debug --disable-static libavutil 51. 34.101 / 51. 34.101 libavcodec 53. 60.100 / 53. 60.100 libavformat 53. 31.100 / 53. 31.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 60.100 / 2. 60.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 52. 0.100 / 52. 0.100 thank you for your kind attention. best regards, From rodney.baker at iinet.net.au Wed Mar 14 15:24:06 2012 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Thu, 15 Mar 2012 00:54:06 +1030 Subject: [FFmpeg-user] Grabbing quality from TV card In-Reply-To: References: <919bb625a40b5bef37c86d3674bc288a.squirrel@nusi> <201203130955.36230.rodney.baker@iinet.net.au> Message-ID: <201203150054.07076.rodney.baker@iinet.net.au> On Wed, 14 Mar 2012 06:25:12 tovis wrote: > Hi Rodney! > Much better, but not good enough :( At least we're making progress. :-) > Quality is still wrong, does not change against the very old Debianized > version. On time I have a lot of error messages: > $ ffmpeg -v verbose -f video4linux2 -i /dev/video0 -s 640x480 -r 25 > out10.avi 2>&1 | tee ffmpeg-grab.log OK - you're setting the output size to 640x480 but... > > The log looks like this: > ffmpeg version N-38750-g599888a Copyright (c) 2000-2012 the FFmpeg > developers built on Mar 12 2012 23:50:31 with gcc 4.4.5 > configuration: --enable-nonfree --enable-libv4l2 > libavutil 51. 42.100 / 51. 42.100 > libavcodec 54. 10.100 / 54. 10.100 > libavformat 54. 2.100 / 54. 2.100 > libavdevice 53. 4.100 / 53. 4.100 > libavfilter 2. 64.101 / 2. 64.101 > libswscale 2. 1.100 / 2. 1.100 > libswresample 0. 7.100 / 0. 7.100 > [video4linux2,v4l2 @ 0x30f13e0] [3]Capabilities: 5010015 > [video4linux2,v4l2 @ 0x30f13e0] Querying the device for the current frame > size > [video4linux2,v4l2 @ 0x30f13e0] Setting frame size to 320x240 ...the input is recognised as 320x240. Is this correct? If so, you're scaling up the input which means you're interpolating a lot of pixels which will, in turn, reduce the quality considerably. I don't know about the "Resource temporarily unavailable" errors - perhaps Carl Eugen (or one of the other devs) may have an idea. I'd try Carl's suggestion re yadif too - if you have an interlaced source you probably want to run it through the deinterlace filter. > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 1174 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > [video4linux2,v4l2 @ 0x30f13e0] Estimating duration from bitrate, this may > be inaccurate > Input #0, video4linux2,v4l2, from '/dev/video0': > Duration: N/A, start: 1331666886.562734, bitrate: 23040 kb/s > Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 320x240, > 23040 kb/s, 25 tbr, 1000k tbn, 25 tbc > [buffer @ 0x30f9020] w:320 h:240 pixfmt:yuv420p tb:1/1000000 sar:0/1 > sws_param: > [scale @ 0x30f6e40] w:320 h:240 fmt:yuv420p -> w:640 h:480 fmt:yuv420p > flags:0x4 > Output #0, avi, to 'out10.avi': > Metadata: > ISFT : Lavf54.2.100 > Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 640x480, > q=2-31, 200 kb/s, 25 tbn, 25 tbc > Stream mapping: > Stream #0:0 -> #0:0 (rawvideo -> mpeg4) > Press [q] to stop, [?] for help > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 18 fps=0.0 q=31.0 size= 183kB time=00:00:00.72 > bitrate=2081.3kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 31 fps= 29 q=31.0 size= 202kB time=00:00:01.24 > bitrate=1333.2kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 44 fps= 28 q=31.0 size= 220kB time=00:00:01.76 > bitrate=1025.8kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 57 fps= 27 q=31.0 size= 239kB time=00:00:02.28 bitrate= > 858.0kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 70 fps= 27 q=31.0 size= 255kB time=00:00:02.80 bitrate= > 745.1kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 83 fps= 26 q=31.0 size= 272kB time=00:00:03.32 bitrate= > 672.0kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 96 fps= 26 q=31.0 size= 290kB time=00:00:03.84 bitrate= > 619.7kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 109 fps= 26 q=24.8 size= 312kB time=00:00:04.36 bitrate= > 586.2kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 122 fps= 26 q=31.0 size= 331kB time=00:00:04.88 bitrate= > 555.1kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 135 fps= 26 q=31.0 size= 349kB time=00:00:05.40 bitrate= > 530.0kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 148 fps= 26 q=31.0 size= 366kB time=00:00:05.92 bitrate= > 506.9kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 161 fps= 26 q=31.0 size= 386kB time=00:00:06.44 bitrate= > 490.5kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 174 fps= 26 q=31.0 size= 403kB time=00:00:06.96 bitrate= > 474.8kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 187 fps= 26 q=31.0 size= 419kB time=00:00:07.48 bitrate= > 458.5kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 200 fps= 26 q=31.0 size= 438kB time=00:00:08.00 bitrate= > 448.8kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 213 fps= 26 q=31.0 size= 454kB time=00:00:08.52 bitrate= > 436.7kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 44 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 226 fps= 26 q=31.0 size= 468kB time=00:00:09.04 bitrate= > 424.0kbits/s > libv4l2: error dequeuing buf: Resource temporarily unavailable > -- repeated 15 times -- > libv4l2: error dequeuing buf: Resource temporarily unavailable > frame= 230 fps= 25 q=31.0 Lsize= 480kB time=00:00:09.20 bitrate= > 427.2kbits/s > video:469kB audio:0kB global headers:0kB muxing overhead 2.358736% > > May be I should not have to compile with libv4l2? What is bother me that > may be I should have to set another input? Any farther suggestions? > > Sincerely > tovis Only what Carl Eugen suggested: "Since it is likely that the input is interlaced, please first try -qscale 2 out0.avi to test if this improves quality, then use yadif." -- =================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au =================================================== From cehoyos at ag.or.at Wed Mar 14 16:30:11 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 14 Mar 2012 15:30:11 +0000 (UTC) Subject: [FFmpeg-user] bgra to yuv References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: Josh long gmail.com> writes: > ffmpeg version 0.7.3-4:0.7.3-0ubuntu0.11.10.1, This is an intentionally broken version of ffmpeg that contains several hundred regressions, many of them security relevant, it is therefore unsupported here. Please test current git head (you do not have to install, the binary in the build directory works fine). You have to configure with --enable-libschroedinger to encode to dirac (the "dirac" encoder is deprecated afaik). Alternatively, there should be a launchpad version: https://launchpad.net/~jon-severinsson/+archive/ffmpeg Carl Eugen From mike.scheutzow at alcatel-lucent.com Wed Mar 14 16:45:26 2012 From: mike.scheutzow at alcatel-lucent.com (Mike Scheutzow) Date: Wed, 14 Mar 2012 11:45:26 -0400 Subject: [FFmpeg-user] Grabbing quality from TV card In-Reply-To: <201203150054.07076.rodney.baker@iinet.net.au> References: <919bb625a40b5bef37c86d3674bc288a.squirrel@nusi> <201203130955.36230.rodney.baker@iinet.net.au> <201203150054.07076.rodney.baker@iinet.net.au> Message-ID: <4F60BD16.9050009@alcatel-lucent.com> Rodney Baker wrote: > On Wed, 14 Mar 2012 06:25:12 tovis wrote: >> Quality is still wrong, does not change against the very old Debianized >> version. On time I have a lot of error messages: >> $ ffmpeg -v verbose -f video4linux2 -i /dev/video0 -s 640x480 -r 25 >> out10.avi 2>&1 | tee ffmpeg-grab.log > ... >> [video4linux2,v4l2 @ 0x30f13e0] Setting frame size to 320x240 > > ...the input is recognised as 320x240. Is this correct? If so, you're scaling > up the input which means you're interpolating a lot of pixels which will, in > turn, reduce the quality considerably. > Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 640x480, > q=2-31, 200 kb/s, 25 tbn, 25 tbc > "200 kb/s" -> this is your output bitrate. You are still not specifying a quality level or bitrate, so you're getting the default value of 200 kbits per second. Most 640x480 material is going to look poor at this bitrate. Mike Scheutzow From jeff.mihalik at gmail.com Wed Mar 14 16:48:01 2012 From: jeff.mihalik at gmail.com (Jeff Mihalik) Date: Wed, 14 Mar 2012 09:48:01 -0600 Subject: [FFmpeg-user] Runtime conversion from byte array Message-ID: Hello, I am creating a java project where the goal is to receive byte arrays of mpeg-2 videos from the database and convert them at runtime to mpeg-4 videos (also as byte arrays, to pass to another application for display). ffmpeg seems like the right tool for the conversion, but every command line example I have seen takes a file as an input and saves off the output as a file. I do not have the option of storing the videos on the file structure, hence I have an input of a byte array mpeg-2, and an output of a byte array mpeg-4. Is it possible to convert from a byte array of an MPEG-2 to a byte array of an MPEG-4 (using java) through ffmpeg, or are real files necessary? I would try this out for myself, but my company takes weeks to approve new software so I want to determine if it is possible before requesting the software. Thank you. -Jeff From jacobhameiri at gmail.com Wed Mar 14 16:52:57 2012 From: jacobhameiri at gmail.com (jacob s) Date: Wed, 14 Mar 2012 17:52:57 +0200 Subject: [FFmpeg-user] what does "real-time buffer 155% full! frame dropped!" mean ? Message-ID: hello all, what does this message mean?: real-time buffer 155% full! frame dropped! , I am getting them alot this is my command: -f dshow -i video="screen-capture-recorder":audio="Realtek HD Audio Input" -vcodec libx264 -preset ultrafast -tune film -r 10 -async 1 -ab 16k -ar 11025 -f mpegts udp://172.21.120.185:48552 this is the full output: http://pastebin.com/YWtZSnWi From cehoyos at ag.or.at Wed Mar 14 16:57:47 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 14 Mar 2012 15:57:47 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?what_does_=22real-time_buffer_155=25_full?= =?utf-8?q?!_frame=09dropped!=22_mean_=3F?= References: Message-ID: jacob s gmail.com> writes: > this is my command: -f dshow -i > video="screen-capture-recorder":audio="Realtek HD Audio Input" -vcodec > libx264 -preset ultrafast -tune film -r 10 -async 1 -ab 16k -ar 11025 -f > mpegts udp://172.21.120.185:48552 > this is the full output: http://pastebin.com/YWtZSnWi External references tend to disappear, please post complete, uncut console output here. Did you test "cheaper" encoders or lower resulutions to make sure this is not a performance issue? Carl Eugen From mailer.tovis at freemail.hu Wed Mar 14 17:08:51 2012 From: mailer.tovis at freemail.hu (tovis) Date: Wed, 14 Mar 2012 17:08:51 +0100 Subject: [FFmpeg-user] Grabbing quality from TV card Message-ID: <4baa3807f82725a943f82dfdcb0db892.squirrel@nusi> Hi Rodney! It is much better with "-qscale 2" even if I use 640x480 resolution :D I steel have errors: libv4l2: error dequeuing buf: Resource temporarily unavailable > > I don't know about the "Resource temporarily unavailable" errors - perhaps Carl Eugen (or one of the other devs) may have an idea. I'd try Carl's suggestion re yadif too - if you have an interlaced source you probably want > to run it through the deinterlace filter. > Definitely I have interlaced input. I have put v4l2-info's output into the pastebin: http://pastebin.com/SWb1u94B The VIDIOC_G_FMT(VIDEO_CAPTURE) shows that is it has resolution 320x240 and it is interlaced and pixelformat 0x33524742 [BGR3]. I'm not so much understand v4l2 interface, there is such an interface video overlay which have resolution 768x576, also interlaced. ffmpeg can not use this interface? Also how deinterlace/filter command parameter should be look like? I know I ask too much, but I do not realy understand "-qscale 2" parameter too :( Also yadif and -qscale should be used together? Sincerely tovis > > Only what Carl Eugen suggested: > > "Since it is likely that the input is interlaced, please first try -qscale 2 out0.avi to test if this improves quality, then use yadif." > From mike.scheutzow at alcatel-lucent.com Wed Mar 14 17:30:23 2012 From: mike.scheutzow at alcatel-lucent.com (Mike Scheutzow) Date: Wed, 14 Mar 2012 12:30:23 -0400 Subject: [FFmpeg-user] Runtime conversion from byte array In-Reply-To: References: Message-ID: <4F60C79F.7010805@alcatel-lucent.com> Jeff Mihalik wrote: > Hello, > > I am creating a java project where the goal is to receive byte arrays of > mpeg-2 videos from the database and convert them at runtime to mpeg-4 > videos (also as byte arrays, to pass to another application for display). > ffmpeg seems like the right tool for the conversion, but every command line > example I have seen takes a file as an input and saves off the output as a > file. I do not have the option of storing the videos on the file > structure, hence I have an input of a byte array mpeg-2, and an output of a > byte array mpeg-4. Is it possible to convert from a byte array of an > MPEG-2 to a byte array of an MPEG-4 (using java) through ffmpeg, or are > real files necessary? I would try this out for myself, but my company > takes weeks to approve new software so I want to determine if it is > possible before requesting the software. Thank you. > > -Jeff FFmpeg is a big, complex library written in the C language. It includes an application, also called ffmpeg, which makes use of this library. Your question is not very clear about whether you would execute the ffmpeg app using java's Runtime.exec(), or access the FFmpeg library at a lower level using the C api. What you ask for can be done through the API, but I wouldn't consider it a simple project. It is going to be much more work than simply calling a function from a java class. You do not say how much experience you have with the C language, or with the Java Native Interface. Also, if you use the API for an application that you distribute/sell, you'll need to obey the licensing requirements (some parts of FFmpeg are LGPL, some are GPL). Mike Scheutzow From tevans.uk at googlemail.com Wed Mar 14 18:19:35 2012 From: tevans.uk at googlemail.com (Tom Evans) Date: Wed, 14 Mar 2012 17:19:35 +0000 Subject: [FFmpeg-user] Runtime conversion from byte array In-Reply-To: References: Message-ID: On Wed, Mar 14, 2012 at 3:48 PM, Jeff Mihalik wrote: > Hello, > > I am creating a java project where the goal is to receive byte arrays of > mpeg-2 videos from the database and convert them at runtime to mpeg-4 > videos (also as byte arrays, to pass to another application for display). > ffmpeg seems like the right tool for the conversion, but every command line > example I have seen takes a file as an input and saves off the output as a > file. ?I do not have the option of storing the videos on the file > structure, hence I have an input of a byte array mpeg-2, and an output of a > byte array mpeg-4. ?Is it possible to convert from a byte array of an > MPEG-2 to a byte array of an MPEG-4 (using java) through ffmpeg, or are > real files necessary? ?I would try this out for myself, but my company > takes weeks to approve new software so I want to determine if it is > possible before requesting the software. ?Thank you. > > -Jeff ffmpeg can read from stdin and write to stdout*, so can be inserted into any application trivially. Embedding it into your application is not necessary. Cheers Tom * Assuming containers that can be read/written in that manner, eg mpeg-ts From lou at lrcd.com Wed Mar 14 18:58:35 2012 From: lou at lrcd.com (Lou) Date: Wed, 14 Mar 2012 09:58:35 -0800 Subject: [FFmpeg-user] blackframe and blackdetect In-Reply-To: <1331719651679-4471452.post@n4.nabble.com> References: <1331653320238-4469396.post@n4.nabble.com> <1331716310370-4471324.post@n4.nabble.com> <20120314093803.GB26897@leki> <1331719651679-4471452.post@n4.nabble.com> Message-ID: <20120314095835.5eb6f895@lrcd.com> On Wed, 14 Mar 2012 03:07:31 -0700 (PDT) LsBender wrote: > Thank for your help Clement, > I tried your command but i have this: "unrecognized option '-vf'": > / > lefebvre at lefebvre-laptop:~/Bureau$ ffmpeg -i test.avi -vf blackframe=98:32 > -f null- > FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1.3, Copyright (c) 2000-2009 > Fabrice Bellard, et al. > configuration: --extra-version=4:0.5.1-1ubuntu1.3 --prefix=/usr > --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib > --enable-libgsm --enable-libschroedinger --enable-libspeex > --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib > --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl > --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 > --enable-shared --disable-static > libavutil 49.15. 0 / 49.15. 0 > libavcodec 52.20. 1 / 52.20. 1 > libavformat 52.31. 0 / 52.31. 0 > libavdevice 52. 1. 0 / 52. 1. 0 > libavfilter 0. 4. 0 / 0. 4. 0 > libswscale 0. 7. 1 / 0. 7. 1 > libpostproc 51. 2. 0 / 51. 2. 0 > built on Dec 21 2011 18:37:21, gcc: 4.4.3 > Input #0, avi, from 'test.avi': > Duration: 01:02:12.22, start: 0.000000, bitrate: 1012 kb/s > Stream #0.0: Video: mpeg4, yuv420p, 704x576 [PAR 1:1 DAR 11:9], 12.50 > tbr, 12.50 tbn, 12.50 tbc > Stream #0.1: Audio: mp2, 32000 Hz, stereo, s16, 64 kb/s > ffmpeg: unrecognized option '-vf'/ > > I already saw some post about blackframe and how to list them, but there are > listing blackframe while ffmpeg is encoding. And in my case, I don't want to > encode. I just want to collect blackframe informations about my video > without encoding. Is it possible?! > > Thanks, > Marc Your version of ffmpeg from the Ubuntu repository does not have filtering capability. You can compile the most recent FFmpeg to gain filtering, bug fixes, and other features. For step-by-step compiling instructions see: HOWTO: Install and use the latest FFmpeg and x264 on Ubuntu http://ubuntuforums.org/showthread.php?t=786095 From cehoyos at ag.or.at Wed Mar 14 19:37:53 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 14 Mar 2012 18:37:53 +0000 (UTC) Subject: [FFmpeg-user] Grabbing quality from TV card References: <4baa3807f82725a943f82dfdcb0db892.squirrel@nusi> Message-ID: tovis freemail.hu> writes: > Definitely I have interlaced input. I have put v4l2-info's > output into the pastebin: http://pastebin.com/SWb1u94B > The VIDIOC_G_FMT(VIDEO_CAPTURE) shows that is it has > resolution 320x240 and it is interlaced and pixelformat > 0x33524742 [BGR3]. Note that it is possible to encode progressive material interlaced and interlaced material progressive, afaik only the visual inspection of the material is relevant. (You only need a deinterlacer if the input material is really interlaced, no matter how it was encoded.) > I'm not so much understand v4l2 interface, there is such an > interface video overlay which have resolution 768x576, also > interlaced. ffmpeg can not use this interface? ? > Also how deinterlace/filter command parameter should be > look like? I know I ask too much, but I do not realy > understand "-qscale 2" parameter too :( -qscale means "constant quality" (and variable bitrate as opposed to constant or adaptive bitrate), "2" is very high quality, it is normally only used for testing (or high quality lossy backup). Try -vf yadif=0 and -vf yadif=1 - 1 doubles the framerate of the output file. Please try to avoid top-posting, it is considered rude here. Carl Eugen From jeff.mihalik at gmail.com Wed Mar 14 19:45:11 2012 From: jeff.mihalik at gmail.com (Jeff) Date: Wed, 14 Mar 2012 11:45:11 -0700 (PDT) Subject: [FFmpeg-user] Runtime conversion from byte array In-Reply-To: <4F60C79F.7010805@alcatel-lucent.com> References: <4F60C79F.7010805@alcatel-lucent.com> Message-ID: <1331750711155-4472791.post@n4.nabble.com> Mike Scheutzow-3 wrote > > Jeff Mihalik wrote: >> Hello, >> >> I am creating a java project where the goal is to receive byte arrays of >> mpeg-2 videos from the database and convert them at runtime to mpeg-4 >> videos (also as byte arrays, to pass to another application for display). >> ffmpeg seems like the right tool for the conversion, but every command >> line >> example I have seen takes a file as an input and saves off the output as >> a >> file. I do not have the option of storing the videos on the file >> structure, hence I have an input of a byte array mpeg-2, and an output of >> a >> byte array mpeg-4. Is it possible to convert from a byte array of an >> MPEG-2 to a byte array of an MPEG-4 (using java) through ffmpeg, or are >> real files necessary? I would try this out for myself, but my company >> takes weeks to approve new software so I want to determine if it is >> possible before requesting the software. Thank you. >> >> -Jeff > > FFmpeg is a big, complex library written in the C language. It includes > an application, also called ffmpeg, which makes use of this library. > > Your question is not very clear about whether you would execute the > ffmpeg app using java's Runtime.exec(), or access the FFmpeg library at > a lower level using the C api. > > What you ask for can be done through the API, but I wouldn't consider it > a simple project. It is going to be much more work than simply calling a > function from a java class. > > You do not say how much experience you have with the C language, or with > the Java Native Interface. > > Also, if you use the API for an application that you distribute/sell, > you'll need to obey the licensing requirements (some parts of FFmpeg are > LGPL, some are GPL). > > > Mike Scheutzow > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user@ > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > I have two years experience with C, and I have experience with JNI as well so that is a viable option (as well as java's Runtime.exec()). Whichever option has the simplest implementation works for me. I will look through the C API - are there certain files you could point me to to work with byte arrays? Otherwise, if java's Runtime.exec() is simpler, how would it work with byte arrays? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Runtime-conversion-from-byte-array-tp4472216p4472791.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From mailer.tovis at freemail.hu Wed Mar 14 20:20:58 2012 From: mailer.tovis at freemail.hu (tovis) Date: Wed, 14 Mar 2012 20:20:58 +0100 Subject: [FFmpeg-user] Grabbing quality from TV card In-Reply-To: References: <4baa3807f82725a943f82dfdcb0db892.squirrel@nusi> Message-ID: <9e46fb78816217205cf6aef85ee7f481.squirrel@nusi> > tovis freemail.hu> writes: > >> Definitely I have interlaced input. I have put v4l2-info's >> output into the pastebin: http://pastebin.com/SWb1u94B >> The VIDIOC_G_FMT(VIDEO_CAPTURE) shows that is it has >> resolution 320x240 and it is interlaced and pixelformat >> 0x33524742 [BGR3]. > > Note that it is possible to encode progressive material > interlaced and interlaced material progressive, afaik only > the visual inspection of the material is relevant. > (You only need a deinterlacer if the input material is > really interlaced, no matter how it was encoded.) > >> I'm not so much understand v4l2 interface, there is such an >> interface video overlay which have resolution 768x576, also >> interlaced. ffmpeg can not use this interface? > > ? > >> Also how deinterlace/filter command parameter should be >> look like? I know I ask too much, but I do not realy >> understand "-qscale 2" parameter too :( > > -qscale means "constant quality" (and variable bitrate as > opposed to constant or adaptive bitrate), "2" is very high > quality, it is normally only used for testing > (or high quality lossy backup). > Try -vf yadif=0 and -vf yadif=1 - 1 doubles the framerate > of the output file. > > Please try to avoid top-posting, it is considered rude here. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > Thanks for respond Carl! Test was really quick :( "No such a filter: 'yadif' Error opening filters! I have check possible configuration parameters, and does not found "--enable_filters"? It's real, do I need to enable every filters one-by-one? Sincerelly tovis OK! I will avoid top-posting (it means do not post between citated answers and continue post at bottom?). In the process it would generate very huge posts, may be delete answer at all, and create a new document? I have seen our conversation on gmane (it was faster delivered then my postmaster), it will be very messy, I think. From mailer.tovis at freemail.hu Wed Mar 14 20:40:59 2012 From: mailer.tovis at freemail.hu (tovis) Date: Wed, 14 Mar 2012 20:40:59 +0100 Subject: [FFmpeg-user] Grabbing quality from TV card In-Reply-To: References: <4baa3807f82725a943f82dfdcb0db892.squirrel@nusi> Message-ID: <8639f1bfff3df8d1538787da5eb08526.squirrel@nusi> Hi! Make/configure problem? I have rebuild ffmpeg, make clean, ./configure --enable-nonfree --enable-libv4l2 --enable-filter=yadif, make, make install but it is still "No such filter: 'yadif' Error opening filters!" What is wrong? Sincerely tovis From de.techno at gmail.com Thu Mar 15 04:37:54 2012 From: de.techno at gmail.com (dE .) Date: Thu, 15 Mar 2012 09:07:54 +0530 Subject: [FFmpeg-user] Playlist input and Mosaic output In-Reply-To: <1331705460.5440.YahooMailNeo@web87706.mail.ir2.yahoo.com> References: <1331705460.5440.YahooMailNeo@web87706.mail.ir2.yahoo.com> Message-ID: sounds like a good idea Also an option to convert to indivisual tracks also. Maybe you should open a ticker. On Mar 14, 2012 11:41 AM, "JULIAN GARDNER" wrote: > I was wondering if its possible to have ffmpeg encode a playlist to > produce a single output, either file or udp stream? > > would like to use it for building a "barker" channel. > > Also ive been trying to get VLC mosaic working and all i have managed to > do is get a black screen, so is there any way of using ffmpeg for creating > a mosaic screen from live sources? > > joolz > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From de.techno at gmail.com Thu Mar 15 06:26:14 2012 From: de.techno at gmail.com (dE .) Date: Thu, 15 Mar 2012 10:56:14 +0530 Subject: [FFmpeg-user] what does "real-time buffer 155% full! frame dropped!" mean ? In-Reply-To: References: Message-ID: I think raw video is buffered in memory until the processor gets enough time to encode it. Just in case this buffer gets full, some of the frames will get dropped to make up for the slow or busy processor. On Mar 14, 2012 9:28 PM, "Carl Eugen Hoyos" wrote: > jacob s gmail.com> writes: > > > this is my command: -f dshow -i > > video="screen-capture-recorder":audio="Realtek HD Audio Input" -vcodec > > libx264 -preset ultrafast -tune film -r 10 -async 1 -ab 16k -ar 11025 -f > > mpegts udp://172.21.120.185:48552 > > this is the full output: http://pastebin.com/YWtZSnWi > > External references tend to disappear, please post complete, uncut > console output here. > > Did you test "cheaper" encoders or lower resulutions to make sure > this is not a performance issue? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From de.techno at gmail.com Thu Mar 15 07:08:57 2012 From: de.techno at gmail.com (dE .) Date: Thu, 15 Mar 2012 11:38:57 +0530 Subject: [FFmpeg-user] Grabbing quality from TV card In-Reply-To: <4baa3807f82725a943f82dfdcb0db892.squirrel@nusi> References: <4baa3807f82725a943f82dfdcb0db892.squirrel@nusi> Message-ID: I'll also suggest using x264. On Mar 14, 2012 9:39 PM, "tovis" wrote: > Hi Rodney! > It is much better with "-qscale 2" even if I use 640x480 resolution :D I > steel have errors: > libv4l2: error dequeuing buf: Resource temporarily unavailable > > > > I don't know about the "Resource temporarily unavailable" errors - > perhaps Carl Eugen (or one of the other devs) may have an idea. I'd try > Carl's suggestion re yadif too - if you have an interlaced source you > probably want > > to run it through the deinterlace filter. > > > Definitely I have interlaced input. I have put v4l2-info's output into the > pastebin: http://pastebin.com/SWb1u94B > The VIDIOC_G_FMT(VIDEO_CAPTURE) shows that is it has resolution 320x240 > and it is interlaced and pixelformat 0x33524742 [BGR3]. > I'm not so much understand v4l2 interface, there is such an interface > video overlay which have resolution 768x576, also interlaced. ffmpeg can > not use this interface? > Also how deinterlace/filter command parameter should be look like? I know > I ask too much, but I do not realy understand "-qscale 2" parameter too :( > Also yadif and -qscale should be used together? > > Sincerely > tovis > > > > Only what Carl Eugen suggested: > > > > "Since it is likely that the input is interlaced, please first try > -qscale 2 out0.avi to test if this improves quality, then use yadif." > > > > > > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From jacobhameiri at gmail.com Thu Mar 15 10:27:38 2012 From: jacobhameiri at gmail.com (jacob s) Date: Thu, 15 Mar 2012 11:27:38 +0200 Subject: [FFmpeg-user] what does "real-time buffer 155% full! frame dropped!" mean ? In-Reply-To: References: Message-ID: thanks, that helped me find the problem 2012/3/15 dE . > I think raw video is buffered in memory until the processor gets enough > time to encode it. > > Just in case this buffer gets full, some of the frames will get dropped to > make up for the slow or busy processor. > On Mar 14, 2012 9:28 PM, "Carl Eugen Hoyos" wrote: > > > jacob s gmail.com> writes: > > > > > this is my command: -f dshow -i > > > video="screen-capture-recorder":audio="Realtek HD Audio Input" -vcodec > > > libx264 -preset ultrafast -tune film -r 10 -async 1 -ab 16k -ar 11025 > -f > > > mpegts udp://172.21.120.185:48552 > > > this is the full output: http://pastebin.com/YWtZSnWi > > > > External references tend to disappear, please post complete, uncut > > console output here. > > > > Did you test "cheaper" encoders or lower resulutions to make sure > > this is not a performance issue? > > > > Carl Eugen > > > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From mike.scheutzow at alcatel-lucent.com Thu Mar 15 13:48:47 2012 From: mike.scheutzow at alcatel-lucent.com (Mike Scheutzow) Date: Thu, 15 Mar 2012 08:48:47 -0400 Subject: [FFmpeg-user] Runtime conversion from byte array In-Reply-To: <1331750711155-4472791.post@n4.nabble.com> References: <4F60C79F.7010805@alcatel-lucent.com> <1331750711155-4472791.post@n4.nabble.com> Message-ID: <4F61E52F.6090303@alcatel-lucent.com> Jeff wrote: >> Jeff Mihalik wrote: >>> I am creating a java project where the goal is to receive byte arrays of >>> mpeg-2 videos from the database and convert them at runtime to mpeg-4 >>> videos (also as byte arrays, to pass to another application for display). > > I have two years experience with C, and I have experience with JNI as well > so that is a viable option (as well as java's Runtime.exec()). Whichever > option has the simplest implementation works for me. I will look through > the C API - are there certain files you could point me to to work with byte > arrays? Otherwise, if java's Runtime.exec() is simpler, how would it work > with byte arrays? The simplest method, by far, is to write the byte array to a temp file, create the correct command line, call Runtime.exec(), then read the result from the output temp file. A slightly more elegant approach is to connect to the stdin and stdout of the exec'ed process. This avoids the need for temp files. The ffmpeg app can be told to read stdin and write stdout. Using Runtime.exec() is also the easiest to comply with, license-wise. Mike Scheutzow From adam at the-adam.com Thu Mar 15 14:14:34 2012 From: adam at the-adam.com (Adam N. Rosenberg) Date: Thu, 15 Mar 2012 06:14:34 -0700 (MST) Subject: [FFmpeg-user] Vimeo uploading - may be off topic In-Reply-To: References: Message-ID: On Thu, 1 Mar 2012, Tom Evans wrote: > On Thu, Mar 1, 2012 at 2:41 PM, Adam N. Rosenberg wrote: >> Hi Dave, >> >> I reckon it does have everything, except the basic stuff of how >> to setup and to run the programs. They figure you already know that. >> I'm looking for real baby steps for a first-time Vimeo programmer. >> > > Might I humbly suggest if you are looking for vimeo support, the > ffmpeg users mailing list is not the best place to look. If you were > looking for baby steps for a first time ffmpeg programmer, you would > have been exactly right... > > Cheers > > Tom Hi Tom, You're right, but I figured people who mess around with video files might also mess around with YouTube and Vimeo, and maybe one of those solved my problem. Thanks. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- Adam N. Rosenberg mailto:adam at the-adam.com http://www.the-adam.com From cehoyos at ag.or.at Thu Mar 15 14:19:52 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 15 Mar 2012 13:19:52 +0000 (UTC) Subject: [FFmpeg-user] what does "real-time buffer 155% full! frame dropped!" mean ? References: Message-ID: dE . gmail.com> writes: [...] Please consider cutting your quotes as this makes reading emails easier and please do not top-post here, it is considered rude. Carl Eugen From cehoyos at ag.or.at Thu Mar 15 14:22:39 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 15 Mar 2012 13:22:39 +0000 (UTC) Subject: [FFmpeg-user] Grabbing quality from TV card References: <4baa3807f82725a943f82dfdcb0db892.squirrel@nusi> <9e46fb78816217205cf6aef85ee7f481.squirrel@nusi> Message-ID: tovis freemail.hu> writes: > Test was really quick :( "No such a filter: 'yadif' (Command line and complete, uncut console output missing.) You have to configure with --enable-gpl Carl Eugen From mlefe74 at gmail.com Thu Mar 15 16:51:04 2012 From: mlefe74 at gmail.com (LsBender) Date: Thu, 15 Mar 2012 08:51:04 -0700 (PDT) Subject: [FFmpeg-user] blackframe and blackdetect In-Reply-To: <20120314095835.5eb6f895@lrcd.com> References: <1331653320238-4469396.post@n4.nabble.com> <1331716310370-4471324.post@n4.nabble.com> <20120314093803.GB26897@leki> <1331719651679-4471452.post@n4.nabble.com> <20120314095835.5eb6f895@lrcd.com> Message-ID: <1331826664285-4475432.post@n4.nabble.com> Thanks Clement and Lou-2. Effectively you were right, my version was out of date. As you recommended me, I upgraded my ubuntu version from 10.04 to 11.10 and my ffmpeg version from 0.5 to 0.9.1 I now have the /blackdetect/ filter and I think this one is better than the blackframe filter for what I want to do. But I still have to understand how to use this filter, in order to have a list of when blackframes occurs in my avi files (if it's possible without encoding). As soon as I manage to get it work, I tell you. Once again, thanks for your help. Marc -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/blackframe-and-blackdetect-tp4469396p4475432.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From mailer.tovis at freemail.hu Thu Mar 15 17:17:27 2012 From: mailer.tovis at freemail.hu (tovis) Date: Thu, 15 Mar 2012 17:17:27 +0100 Subject: [FFmpeg-user] Grabbing quality from TV card In-Reply-To: References: <4baa3807f82725a943f82dfdcb0db892.squirrel@nusi> <9e46fb78816217205cf6aef85ee7f481.squirrel@nusi> Message-ID: <465798b6b8eeb0ee0a148a84f773b315.squirrel@nusi> > (Command line and complete, uncut console output missing.) > You have to configure with --enable-gpl > > Carl Eugen > Hi Carl! That is solve th making problem :) But using filter yadif does not improve quality :( Option '-qscale 2' was much better. I'm afraid, my problem is not (or not only) about ffmpeg, but the video source. Is it possible to use overlay for grabbing? Why the quality is so much differ from quality of fbtv? I was tried out using camera (a simple 640x480 resolution) utilize composite video input, quality was good (yes, the source I was switch to "CAMERA" using v4l2-ctl after ffmpeg was started). What I can do more, to have good quality. Sincerely tovis From vsethi at iglou.com Thu Mar 15 17:55:57 2012 From: vsethi at iglou.com (vsethi at iglou.com) Date: Thu, 15 Mar 2012 12:55:57 -0400 Subject: [FFmpeg-user] Retrieving deleted segments Message-ID: Is there any way to retrieve an accidentally deleted segment from a non-finalized dvd-r? Thanks. From cehoyos at ag.or.at Thu Mar 15 17:59:13 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 15 Mar 2012 16:59:13 +0000 (UTC) Subject: [FFmpeg-user] blackframe and blackdetect References: <1331653320238-4469396.post@n4.nabble.com> <1331716310370-4471324.post@n4.nabble.com> <20120314093803.GB26897@leki> <1331719651679-4471452.post@n4.nabble.com> <20120314095835.5eb6f895@lrcd.com> <1331826664285-4475432.post@n4.nabble.com> Message-ID: LsBender gmail.com> writes: > As you recommended me, I upgraded my ubuntu version from 10.04 > to 11.10 and my ffmpeg version from 0.5 to 0.9.1 0.9.1 is quite outdated, if you are a user (not a distributor of FFmpeg), please use current git head, it contains more features and less bugs than any released version. Carl Eugen From cehoyos at ag.or.at Thu Mar 15 18:04:18 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 15 Mar 2012 17:04:18 +0000 (UTC) Subject: [FFmpeg-user] Grabbing quality from TV card References: <4baa3807f82725a943f82dfdcb0db892.squirrel@nusi> <9e46fb78816217205cf6aef85ee7f481.squirrel@nusi> <465798b6b8eeb0ee0a148a84f773b315.squirrel@nusi> Message-ID: tovis freemail.hu> writes: > That is solve th making problem :) But using filter yadif does > not improve quality :( Option '-qscale 2' was much better. Please understand that yadif is only useful if your content is interlaced, and that only you can say if the content is interlaced (after a visual inspection). If the sample is interlaced you should definitely use yadif; using -qscale (or a high adaptive bitrate) is completely orthogonal. Carl Eugen From anders.branderud at gmail.com Thu Mar 15 16:12:34 2012 From: anders.branderud at gmail.com (Anders Branderud) Date: Thu, 15 Mar 2012 17:12:34 +0200 Subject: [FFmpeg-user] =?windows-1252?q?libavutil/common=2Eh=3A173=3A47=3A?= =?windows-1252?q?_error=3A_=91UINT64=5FC=92_was_not_declared_in_th?= =?windows-1252?q?is_scope?= Message-ID: Hello! I get a problem when compiling, which I will describe below: I am using the latest Ubuntu--version in VirtualBox. I am running parts of the encoding-example avaiable in ffmpeg/doc/examples. When I run it I get the following error: In file included from /usr/local/include/libavutil/avutil.h:327:0, from /usr/local/include/libavutil/imgutils.h:30, from videofecencoder.cc:31: /usr/local/include/libavutil/common.h: In function ?int32_t av_clipl_int32_c(int64_t)?: /usr/local/include/libavutil/common.h:173:47: error: ?UINT64_C? was not declared in this scope How is this resolved? Thanks in advance! --*Anders Brandeud *[Personal blog] Will of the Creator : Logical reasons - based on scientific premises - for the existence of a Super intelligent and Orderly Creator and that He hasn't left His sapient creatures without an Instruction Manual - Torah ['books of Moses'] - to ascertain, and aspire to, His purpose. [Company] Anders Branderud IT Solutions - www.abitsolutions.org From gamekings at hotmail.com Thu Mar 15 16:59:52 2012 From: gamekings at hotmail.com (lf2) Date: Thu, 15 Mar 2012 08:59:52 -0700 (PDT) Subject: [FFmpeg-user] ffmpeg extract metadata to text or php file ? Message-ID: <1331827192659-4475468.post@n4.nabble.com> ffmpeg extract metadata to text or php file ? how can i do this -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-extract-metadata-to-text-or-php-file-tp4475468p4475468.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From jacobhameiri at gmail.com Thu Mar 15 22:40:21 2012 From: jacobhameiri at gmail.com (jacob s) Date: Thu, 15 Mar 2012 23:40:21 +0200 Subject: [FFmpeg-user] increase the size of the real-time buffer. Message-ID: Hi, I am streaming windows directshow desktop capture via udp and I am getting occasional "real-time buffer 155% full! frame dropped!" errors, which are causing audio ( and rarely video ) glitches. Is there a way to increase the size of the real-time buffer ? I am willing to get higher latency if it will solve the frame drops. This is my command: -f dshow -i video="screen-capture-recorder":audio="Stereo Mix (IDT High Definition" -vcodec libx264 -preset ultrafast -tune zerolatency -r 10 -async 1 -acodec libmp3lame -ab 24k -ar 22050 -bsf:v h264_mp4toannexb -maxrate 750k -bufsize 3000k -f mpegts udp://192.168.5.215:48550 From jingke2000 at gmail.com Fri Mar 16 05:45:21 2012 From: jingke2000 at gmail.com (Ke (Kevin) Yu) Date: Thu, 15 Mar 2012 23:45:21 -0500 Subject: [FFmpeg-user] Select network interface card Message-ID: I have a dual NIC server. I need to receive RTP streams (with A/V elementary streams) from one NIC and demux/mux them into a transport stream then send it out from another NIC. How can I do that with FFMpeg? Thanks. From andrey.krieger.utkin at gmail.com Fri Mar 16 10:41:10 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Fri, 16 Mar 2012 11:41:10 +0200 Subject: [FFmpeg-user] =?windows-1252?q?libavutil/common=2Eh=3A173=3A47=3A?= =?windows-1252?q?_error=3A_=91UINT64=5FC=92_was_not_declared_in_th?= =?windows-1252?q?is_scope?= In-Reply-To: References: Message-ID: 2012/3/15 Anders Branderud : > Hello! > > I get a problem when compiling, which I will describe below: > I am using the latest Ubuntu--version in VirtualBox. > > I am running parts of the encoding-example avaiable in ffmpeg/doc/examples. > When I run it I get the following error: > In file included from /usr/local/include/libavutil/avutil.h:327:0, > ? ? ? ? ? ? ? ? from /usr/local/include/libavutil/imgutils.h:30, > ? ? ? ? ? ? ? ? from videofecencoder.cc:31: > /usr/local/include/libavutil/common.h: In function ?int32_t > av_clipl_int32_c(int64_t)?: > /usr/local/include/libavutil/common.h:173:47: error: ?UINT64_C? was not > declared in this scope Please consider posting all related info. Missing your compilation command line. Next time, use libav-user maillist for ffmpeg _libraries_ usage. -- Andrey Utkin From cehoyos at ag.or.at Fri Mar 16 14:36:56 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 16 Mar 2012 13:36:56 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?libavutil/common=2Eh=3A173=3A47=3A_error?= =?utf-8?b?OiDigJhVSU5UNjRfQ+KAmSB3YXMgbm90IGRlY2xhcmVkIGluIHRoaXMg?= =?utf-8?q?scope?= References: Message-ID: Anders Branderud gmail.com> writes: > I am running parts of the encoding-example avaiable in ffmpeg/doc/examples. > When I run it I get the following error: > In file included from /usr/local/include/libavutil/avutil.h:327:0, > from /usr/local/include/libavutil/imgutils.h:30, > from videofecencoder.cc:31: > /usr/local/include/libavutil/common.h: In function ?int32_t > av_clipl_int32_c(int64_t)?: > /usr/local/include/libavutil/common.h:173:47: error: ?UINT64_C? > was not declared in this scope Could you confirm that you are using a C compiler (not a C++ compiler) to compile files from a C library (or that you took appropriate action like defining all necessary constants to use a C library from a C++ project)? Carl Eugen From diego.cdomingos2010 at gmail.com Fri Mar 16 18:09:46 2012 From: diego.cdomingos2010 at gmail.com (Diego Carvalho) Date: Fri, 16 Mar 2012 14:09:46 -0300 Subject: [FFmpeg-user] What's the best way to send a single static image through a RTP connection? Message-ID: Hi all, I'm working on a VoIP application that in some moment needs to send to the other endpoint a static image. I can read the image, decode, scale (if needed), encode to the correct format (h264) and generate the RTP packets that will be sent. However, as I'm sending a static image there is no need to do this whole process all the time. I believe that I should generate the first IDR frame and the first non-IDR and then just send this same non-IDR every time or something like that but I don't know how to do this. Or maybe send the first IDR and then keep sending a packet that says "there aren't changes between this frame and the previous one" but I don't know if such packet exists. I appreciate any help. Thanks. From anders.branderud at gmail.com Thu Mar 15 19:54:14 2012 From: anders.branderud at gmail.com (Anders Branderud) Date: Thu, 15 Mar 2012 18:54:14 +0000 Subject: [FFmpeg-user] 'Undefined reference'-error when compiling FFMpeg Message-ID: Hello! Here follows a problem that I would need your help with: I have installed the latest FFMpeg-library. I have tried to link to the library using two different Ubuntu-installations without any success. When I run the command 'ubuntu at ubuntu:/media/6982d030-e492-4bc9-a1e4-b674ef78cff0/home/david/Programs$ sudo g++ -Iffmpeg/ -L/usr/lib videofecencoder.cc /media/6982d030-e492-4bc9-a1e4-b674ef78cff0/home/david/Programs/ffmpeg/libavformat/libavformat.a libavformat.so.53 -pthread', I get this error: ubuntu at ubuntu:/media/6982d030-e492-4bc9-a1e4-b674ef78cff0/home/david/Programs$ sudo g++ -Iffmpeg/ -L/usr/lib videofecencoder.cc /media/6982d030-e492-4bc9-a1e4-b674ef78cff0/home/david/Programs/ffmpeg/libavformat/libavformat.a /media/6982d030-e492-4bc9-a1e4-b674ef78cff0/home/david/Programs/ffmpeg/libavutil/libavutil.a libavformat.so.53 -pthread /tmp/ccpX60ph.o: In function `video_encode_example(char const*, int)': videofecencoder.cc:(.text+0x30): undefined reference to `avcodec_find_encoder(CodecID)' videofecencoder.cc:(.text+0x75): undefined reference to `avcodec_alloc_context3(AVCodec*)' videofecencoder.cc:(.text+0x7d): undefined reference to `avcodec_alloc_frame()' videofecencoder.cc:(.text+0x108): undefined reference to `av_opt_set(void*, char const*, char const*, int)' videofecencoder.cc:(.text+0x122): undefined reference to `avcodec_open2(AVCodecContext*, AVCodec*, AVDictionary**)' videofecencoder.cc:(.text+0x1f4): undefined reference to `av_image_alloc(unsigned char**, int*, int, int, PixelFormat, int)' videofecencoder.cc:(.text+0x348): undefined reference to `avcodec_encode_video(AVCodecContext*, unsigned char*, int, AVFrame const*)' videofecencoder.cc:(.text+0x3cb): undefined reference to `avcodec_encode_video(AVCodecContext*, unsigned char*, int, AVFrame const*)' videofecencoder.cc:(.text+0x47c): undefined reference to `avcodec_close(AVCodecContext*)' videofecencoder.cc:(.text+0x487): undefined reference to `av_free(void*)' videofecencoder.cc:(.text+0x494): undefined reference to `av_free(void*)' videofecencoder.cc:(.text+0x49f): undefined reference to `av_free(void*)' /tmp/ccpX60ph.o: In function `main': videofecencoder.cc:(.text+0x4c0): undefined reference to `avcodec_register_all()' collect2: ld returned 1 exit status ubuntu at ubuntu :/media/6982d030-e492-4bc9-a1e4-b674ef78cff0/home/david/Programs What should I do to resolve the above? Thanks! -- Anders Branderud [Personal blog] Will of the Creator : Logical reasons - based on scientific premises - for the existence of a Super intelligent and Orderly Creator and that He hasn't left His sapient creatures without an Instruction Manual - Torah ['books of Moses'] - to ascertain, and aspire to, His purpose. [Company] Anders Branderud IT Solutions - www.abitsolutions.org From tevans.uk at googlemail.com Fri Mar 16 20:01:54 2012 From: tevans.uk at googlemail.com (Tom Evans) Date: Fri, 16 Mar 2012 19:01:54 +0000 Subject: [FFmpeg-user] 'Undefined reference'-error when compiling FFMpeg In-Reply-To: References: Message-ID: On Thu, Mar 15, 2012 at 6:54 PM, Anders Branderud wrote: > Hello! > > Here follows a problem that I would need your help with: > I have installed the latest FFMpeg-library. I have tried to link to the > library using two different Ubuntu-installations without any success. > > When I run the command > 'ubuntu at ubuntu:/media/6982d030-e492-4bc9-a1e4-b674ef78cff0/home/david/Programs$ > sudo g++ -Iffmpeg/ -L/usr/lib videofecencoder.cc > /media/6982d030-e492-4bc9-a1e4-b674ef78cff0/home/david/Programs/ffmpeg/libavformat/libavformat.a > libavformat.so.53 -pthread', I get this error: > ubuntu at ubuntu:/media/6982d030-e492-4bc9-a1e4-b674ef78cff0/home/david/Programs$ > sudo g++ -Iffmpeg/ -L/usr/lib videofecencoder.cc > /media/6982d030-e492-4bc9-a1e4-b674ef78cff0/home/david/Programs/ffmpeg/libavformat/libavformat.a > /media/6982d030-e492-4bc9-a1e4-b674ef78cff0/home/david/Programs/ffmpeg/libavutil/libavutil.a > libavformat.so.53 -pthread > /tmp/ccpX60ph.o: In function `video_encode_example(char const*, int)': > videofecencoder.cc:(.text+0x30): undefined reference to > `avcodec_find_encoder(CodecID)' ld isn't prescient, it expects you to give it the libraries to link against in the right order, or to tell it to loop through the libraries specified to resolve the symbols (much slower). The right order is the one where the required symbols are loaded before they are referred to, so: g++ -c -o videofecencoder.o videofecencoder.cc g++ -o videofecencoder -L/foo/bar -lfoo -lbar videofecencoder.o If you are still getting errors, use nm to examine the libraries you are linking in and ensure they do have the symbols it is expecting. Cheers Tom From mmulhaupt at gmail.com Fri Mar 16 21:39:44 2012 From: mmulhaupt at gmail.com (Michael Mulhaupt) Date: Fri, 16 Mar 2012 16:39:44 -0400 Subject: [FFmpeg-user] Unable to initialize H.264 codec with SPS/PPS from IP video camera... Message-ID: Hi All, I'm trying to view the video from a locally connected IP camera (a cheap Starcam H.264 camera from China), and I am not able to successfully initialize the H.264 video codec with the SPS/PPS parameters from the SDP. ?I tried removing the extradata from the codec initialization, and just passing through on the full frames (sending [7][8][5] H.264 NALUs) with the same behavior when decoding the frame. I would have guessed the camera is at fault, BUT VLC player is able to display my camera, and it should be using the same codec. So I'm not sure if VLC does some extra massaging of the SPS/PPS or what. Or am I missing something obvious - like extracting some info from the SPS/PPS myself and passing it explicitly to initialize the codec. Including initial ffprobe errors, SDP, and source code initialization snippet. Running ffprobe shows the same problems I'm seeing with my code: >>ffprobe.exe rtsp://192.168.0.20/H264 ffprobe version 0.9.1.git-88c76c7 Copyright (c) 2007-2012 the FFmpeg developers ? built on Mar 14 2012 19:14:58 with gcc 3.4.5 (mingw-vista special r3) ? configuration: --extra-cflags='-mno-cygwin -mms-bitfields' --extra-ldflags='-Wl,-add-stdcall-alias' --enable-memalign-hack --target-os=mingw32 ? libavutil ? ? ?51. 42.100 / 51. 42.100 ? libavcodec ? ? 54. 10.100 / 54. 10.100 ? libavformat ? ?54. ?2.100 / 54. ?2.100 ? libavdevice ? ?53. ?4.100 / 53. ?4.100 ? libavfilter ? ? 2. 64.101 / ?2. 64.101 ? libswscale ? ? ?2. ?1.100 / ?2. ?1.100 ? libswresample ? 0. ?7.100 / ?0. ?7.100 [h264 @ 0x2b65b40] Overread VUI by 6 bits [h264 @ 0x2b65b40] cpb_count 33 invalid [h264 @ 0x2b65b40] sps_id out of range [h264 @ 0x2b65b40] Overread VUI by 6 bits [h264 @ 0x2b65b40] cpb_count 33 invalid [h264 @ 0x2b65b40] sps_id out of range [h264 @ 0x2b65b40] non-existing PPS referenced [h264 @ 0x2b65b40] non-existing PPS 0 referenced [h264 @ 0x2b65b40] decode_slice_header error [h264 @ 0x2b65b40] no frame! [h264 @ 0x2b65b40] non-existing PPS referenced ... My SDP is v=0 o=- 12930000 1 IN IP4 192.168.0.20 s=Session streamed by "Object RTSPServer" i=H264 t=0 0 a=tool:LIVE555 Streaming Media v2011.07.08 a=type:broadcast a=control:* a=range:npt=0- a=x-qt-text-nam:Session streamed by "Object RTSPServer" a=x-qt-text-inf:H264 m=video 0 RTP/AVP 96 c=IN IP4 0.0.0.0 b=AS:500 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1;profile-level-id=42001E;sprop-parameter-sets=J0IAHqpAUB6TcCAgJAAAAwAEAAADAGeAC5zgCAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA,KM48gAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA== a=control:track1 char* sprops="J0IAHqpAUB6TcCAgJAAAAwAEAAADAGeAC5zgCAAAAAAAAAAAAAAA" "AAAAAAAAAAAAAAAAAAAAAAAAAAAA,KM48gAAAAAAAAAAAAAAAA" "AAAAAAAAAAAAAAAAA=="; codec = avcodec_find_decoder(CODEC_ID_H264); codecCtx = avcodec_alloc_context(); codecCtx->pix_fmt = PIX_FMT_YUV420P ; codecCtx->skip_frame = AVDISCARD_DEFAULT; codecCtx->error_concealment = 3; codecCtx->skip_loop_filter = AVDISCARD_DEFAULT; codecCtx->workaround_bugs = 1; codecCtx->codec_type = AVMEDIA_TYPE_VIDEO; codecCtx->codec_id = CODEC_ID_H264; codecCtx->pix_fmt = PIX_FMT_YUV420P; codecCtx->sample_fmt = AV_SAMPLE_FMT_NONE; codecCtx->codec_type = AVMEDIA_TYPE_VIDEO; codecCtx->skip_loop_filter = AVDISCARD_DEFAULT; codecCtx->skip_idct = AVDISCARD_DEFAULT; codecCtx->skip_frame = AVDISCARD_DEFAULT; codecCtx->color_primaries = AVCOL_PRI_UNSPECIFIED; codecCtx->color_trc = AVCOL_TRC_UNSPECIFIED; codecCtx->colorspace = AVCOL_SPC_UNSPECIFIED; codecCtx->color_range = AVCOL_RANGE_UNSPECIFIED; codecCtx->chroma_sample_location = AVCHROMA_LOC_LEFT; codecCtx->extradata = parseH264ConfigStr(sprops, &codecCtx->extradata_size); avcodec_open(codecCtx, codec) // spits out these errors parseH264ConfigStr() from vlc source code - modules/demux/live555.cpp From deron at pagestream.org Sat Mar 17 04:57:50 2012 From: deron at pagestream.org (Deron) Date: Fri, 16 Mar 2012 21:57:50 -0600 Subject: [FFmpeg-user] How to crop extra audio channels? Message-ID: <4F640BBE.50602@pagestream.org> I have a file I want to general a mono audio file from, but this file has 16 audio channels in a single stream (near as I can tell, only the first 2 channels have any data in them. Must have been captured from HD-SDI). Normally, this works: ffmpeg -i "file.mov" -f s8 -vn -ac 2 -ar 48000 -acodec pcm_s8 -y "file.audio" Works find for me, but this particular file belches out: [SWR @ 0x2fe68a0] Input channel layout isnt supported swr_init() failed Can not resample 16 channels @ 48000 Hz to 2 channels @ 48000 Hz What can I do to strip out the offending 14 channels? Thanks, Deron -------------- ffmpeg -i "file.mov" -f s8 -vn -ac 2 -ar 48000 -acodec pcm_s8 -y "file.audio" ffmpeg version N-37208-g01fcbdf Copyright (c) 2000-2012 the FFmpeg developers built on Jan 26 2012 21:23:21 with gcc 4.5.2 configuration: --enable-gpl --enable-nonfree --enable-libxvid --enable-libx264 --enable-libmp3lame --enable-libvorbis --enable-libfaac libavutil 51. 34.101 / 51. 34.101 libavcodec 53. 60.100 / 53. 60.100 libavformat 53. 31.100 / 53. 31.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 60.100 / 2. 60.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 52. 0.100 / 52. 0.100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x2fd53a0] decoding for stream 1 failed Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'file.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2012-03-11 02:48:44 Duration: 00:58:31.21, start: 0.000000, bitrate: 236019 kb/s Stream #0:0(eng): Video: v210 (v210 / 0x30313276), yuv422p10le, 720x486, 223724 kb/s, 29.97 fps, 29.97 tbr, 2997 tbn, 2997 tbc Metadata: creation_time : 2012-03-11 03:47:15 handler_name : ?Apple Alias Data Handler Stream #0:1(eng): Data: none (tmcd / 0x64636D74) Metadata: creation_time : 2012-03-11 03:47:15 handler_name : ?Apple Alias Data Handler timecode : 00:00:00;00 Stream #0:2(eng): Audio: pcm_s16be (lpcm / 0x6D63706C), 48000 Hz, 16 channels, s16, 12288 kb/s Metadata: creation_time : 2012-03-11 03:47:15 handler_name : ?Apple Alias Data Handler Incompatible sample format 's16' for codec 'pcm_s8', auto-selecting format 'u8' Output #0, s8, to 'file.audio': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2012-03-11 02:48:44 encoder : Lavf53.31.100 Stream #0:0(eng): Audio: pcm_s8, 48000 Hz, 2 channels, u8, 768 kb/s Metadata: creation_time : 2012-03-11 03:47:15 handler_name : ?Apple Alias Data Handler Stream mapping: Stream #0:2 -> #0:0 (pcm_s16be -> pcm_s8) Press [q] to stop, [?] for help [SWR @ 0x2fe68a0] Input channel layout isnt supported swr_init() failed Can not resample 16 channels @ 48000 Hz to 2 channels @ 48000 Hz From rickcorteza at gmail.com Sat Mar 17 07:51:32 2012 From: rickcorteza at gmail.com (Rick C.) Date: Sat, 17 Mar 2012 14:51:32 +0800 Subject: [FFmpeg-user] new ffmpeg version Message-ID: <0F5FBE5F-7E75-441C-AED5-C2028EE0284A@gmail.com> Hi, I'm getting this error when building the latest release: make: *** [libavcodec/libx264.o] Error 1 I have the latest stable release of x264. If I build with ffmpeg 0.10 it's fine, but when I use 0.10.1 it fails. What could the problem be thanks, rc From de.techno at gmail.com Sat Mar 17 09:46:13 2012 From: de.techno at gmail.com (dE .) Date: Sat, 17 Mar 2012 14:16:13 +0530 Subject: [FFmpeg-user] what does "real-time buffer 155% full! frame dropped!" mean ? In-Reply-To: References: Message-ID: <4F644F55.1030006@gmail.com> On 03/15/12 18:49, Carl Eugen Hoyos wrote: > dE . gmail.com> writes: > > [...] > Please consider cutting your quotes as this makes reading emails > easier and please do not top-post here, it is considered rude. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Sorry. That was done via my Android phone automatically. From de.techno at gmail.com Sat Mar 17 09:47:31 2012 From: de.techno at gmail.com (dE .) Date: Sat, 17 Mar 2012 14:17:31 +0530 Subject: [FFmpeg-user] Retrieving deleted segments In-Reply-To: References: Message-ID: <4F644FA3.6020904@gmail.com> On 03/15/12 22:25, vsethi at iglou.com wrote: > Is there any way to retrieve an accidentally deleted segment from a > non-finalized dvd-r? > > Thanks. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user You did a multisession DVD? From de.techno at gmail.com Sat Mar 17 09:52:56 2012 From: de.techno at gmail.com (dE .) Date: Sat, 17 Mar 2012 14:22:56 +0530 Subject: [FFmpeg-user] increase the size of the real-time buffer. In-Reply-To: References: Message-ID: <4F6450E8.6020103@gmail.com> On 03/16/12 03:10, jacob s wrote: > Hi, > I am streaming windows directshow desktop capture via udp and I am getting > occasional "real-time buffer 155% full! frame dropped!" errors, which are > causing audio ( and rarely video ) glitches. > Is there a way to increase the size of the real-time buffer ? I am willing > to get higher latency if it will solve the frame drops. > This is my command: > -f dshow -i video="screen-capture-recorder":audio="Stereo Mix (IDT High > Definition" -vcodec libx264 -preset ultrafast -tune zerolatency -r 10 > -async 1 -acodec libmp3lame -ab 24k -ar 22050 -bsf:v h264_mp4toannexb > -maxrate 750k -bufsize 3000k -f mpegts udp://192.168.5.215:48550 > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user buffer_size=size It's in the man page. From de.techno at gmail.com Sat Mar 17 09:53:52 2012 From: de.techno at gmail.com (dE .) Date: Sat, 17 Mar 2012 14:23:52 +0530 Subject: [FFmpeg-user] Select network interface card In-Reply-To: References: Message-ID: <4F645120.8030001@gmail.com> On 03/16/12 10:15, Ke (Kevin) Yu wrote: > I have a dual NIC server. I need to receive RTP streams (with A/V > elementary streams) from one NIC and demux/mux them into a transport stream > then send it out from another NIC. How can I do that with FFMpeg? > > Thanks. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user This depends on your routing tables. From cehoyos at ag.or.at Sat Mar 17 09:58:11 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 17 Mar 2012 08:58:11 +0000 (UTC) Subject: [FFmpeg-user] new ffmpeg version References: <0F5FBE5F-7E75-441C-AED5-C2028EE0284A@gmail.com> Message-ID: Rick C. gmail.com> writes: > I'm getting this error when building the latest release: (If this were not known: Complete, uncut output of a repeated call to "make" missing - please never post the complete, uncut output of the first call to make) A new release will be made soon. If you are a user, please always use current git head, it contains more features and less bugs than any release. Carl Eugen From cehoyos at ag.or.at Sat Mar 17 10:07:53 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 17 Mar 2012 09:07:53 +0000 (UTC) Subject: [FFmpeg-user] How to crop extra audio channels? References: <4F640BBE.50602@pagestream.org> Message-ID: Deron pagestream.org> writes: > [SWR @ 0x2fe68a0] Input channel layout isnt supported > swr_init() failed > Can not resample 16 channels @ 48000 Hz to 2 channels @ 48000 Hz This is currently unsupported, I opened ticket #1088: http://ffmpeg.org/trac/ffmpeg/ticket/1088 Carl Eugen From dimon.bigdog at gmail.com Sat Mar 17 11:48:56 2012 From: dimon.bigdog at gmail.com (bigdogydog) Date: Sat, 17 Mar 2012 03:48:56 -0700 (PDT) Subject: [FFmpeg-user] FFMPEG dshow option and libx264 video grab devices issue In-Reply-To: References: Message-ID: <1331981336770-4480387.post@n4.nabble.com> Well, when I use the latest windows ffmpeg build (ffmpeg-git-1eabd71-win32-static) I receive a lot of real-time buffer fullness errors and drops of frames too (aproximatly 50%), but when i run earlier build of ffmpeg from 2011, May (ffmpeg-git-01a73d6-win32-static), arter some errors of buffer and dropped frames at the beggining of capture, everything becomes good and runs smoothly without any errors -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/FFMPEG-dshow-option-and-libx264-video-grab-devices-issue-tp4405068p4480387.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From cehoyos at ag.or.at Sat Mar 17 12:46:18 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 17 Mar 2012 11:46:18 +0000 (UTC) Subject: [FFmpeg-user] How to crop extra audio channels? References: <4F640BBE.50602@pagestream.org> Message-ID: Deron pagestream.org> writes: > ffmpeg -i "file.mov" -f s8 -vn -ac 2 -ar 48000 -acodec pcm_s8 -y > "file.audio" Sorry for my first comment, please try the following and report back: ffmpeg -i file.mov -f s8 -map_channel 0.0.1 -map_channel 0.0.2 file.audio As explained in ticket #1088, -ac 2 would downmix from 16 to two channels, since the layout for 16 channels is unknown, this is generally impossible. Carl Eugen From greenythebeast at gmail.com Sat Mar 17 19:07:34 2012 From: greenythebeast at gmail.com (Eli Greenberg) Date: Sat, 17 Mar 2012 13:07:34 -0500 Subject: [FFmpeg-user] Different audio lengths when converting nellymoser to WAV Message-ID: <4F64D2E6.10208@gmail.com> Question for all you smart people. I'm trying to use FFmpeg to convert some nellymoser audio to a .wav file. However, when the .wav file is produced it is a different length than the nellymoser audio. What would cause this? Terminal output: Duration: 01:03:50.37, start: 0.000000, bitrate: 483 kb/s Stream #0:0: Video: h264 (Main), yuv420p, 1024x768 [SAR 1:1 DAR 4:3], 15 tbr, 1k tbn, 30 tbc Stream #0:1: Audio: nellymoser, 8000 Hz, mono, s16 Output #0, wav, to 'test.wav': Metadata: encoder : Lavf54.2.100 Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, mono, s16, 128 kb/s Stream mapping: Stream #0:1 -> #0:0 (nellymoser -> pcm_s16le) Press [q] to stop, [?] for help Input stream #0:1 frame changed from rate:8000 fmt:s16 ch:1 to rate:5512 fmt:s16 ch:1 Input stream #0:1 frame changed from rate:5512 fmt:s16 ch:1 to rate:8000 fmt:s16 ch:1 size= 52040kB time=00:55:30.57 bitrate= 128.0kbits/s video:0kB audio:52040kB global headers:0kB muxing overhead 0.000086% From cehoyos at ag.or.at Sat Mar 17 21:47:20 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 17 Mar 2012 20:47:20 +0000 (UTC) Subject: [FFmpeg-user] Different audio lengths when converting nellymoser to WAV References: <4F64D2E6.10208@gmail.com> Message-ID: Eli Greenberg gmail.com> writes: > Question for all you smart people. I'm trying to use FFmpeg to convert > some nellymoser audio to a .wav file. However, when the .wav file is > produced it is a different length than the nellymoser audio. What would > cause this? Command line and complete, uncut console output missing. Carl Eugen From rickcorteza at gmail.com Sun Mar 18 03:50:49 2012 From: rickcorteza at gmail.com (Rick C.) Date: Sun, 18 Mar 2012 10:50:49 +0800 Subject: [FFmpeg-user] new ffmpeg version In-Reply-To: References: <0F5FBE5F-7E75-441C-AED5-C2028EE0284A@gmail.com> Message-ID: On Mar 17, 2012, at 4:58 PM, Carl Eugen Hoyos wrote: > Rick C. gmail.com> writes: > >> I'm getting this error when building the latest release: > > (If this were not known: Complete, uncut output of a repeated > call to "make" missing - please never post the complete, uncut > output of the first call to make) > > A new release will be made soon. > > If you are a user, please always use current git head, it > contains more features and less bugs than any release. > > Carl Eugen > Thanks Carl the new version fixed everything! rc From de.techno at gmail.com Sun Mar 18 06:56:48 2012 From: de.techno at gmail.com (dE .) Date: Sun, 18 Mar 2012 11:26:48 +0530 Subject: [FFmpeg-user] Different audio lengths when converting nellymoser to WAV In-Reply-To: <4F64D2E6.10208@gmail.com> References: <4F64D2E6.10208@gmail.com> Message-ID: <4F657920.2010507@gmail.com> On 03/17/12 23:37, Eli Greenberg wrote: > Question for all you smart people. I'm trying to use FFmpeg to convert > some nellymoser audio to a .wav file. However, when the .wav file is > produced it is a different length than the nellymoser audio. What > would cause this? > > Terminal output: > Duration: 01:03:50.37, start: 0.000000, bitrate: 483 kb/s > Stream #0:0: Video: h264 (Main), yuv420p, 1024x768 [SAR 1:1 DAR > 4:3], 15 tbr, 1k tbn, 30 tbc > Stream #0:1: Audio: nellymoser, 8000 Hz, mono, s16 > Output #0, wav, to 'test.wav': > Metadata: > encoder : Lavf54.2.100 > Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, > mono, s16, 128 kb/s > Stream mapping: > Stream #0:1 -> #0:0 (nellymoser -> pcm_s16le) > Press [q] to stop, [?] for help > Input stream #0:1 frame changed from rate:8000 fmt:s16 ch:1 to > rate:5512 fmt:s16 ch:1 > Input stream #0:1 frame changed from rate:5512 fmt:s16 ch:1 to > rate:8000 fmt:s16 ch:1 > size= 52040kB time=00:55:30.57 bitrate= 128.0kbits/s > video:0kB audio:52040kB global headers:0kB muxing overhead 0.000086% > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Post ffprobe output of the input and output. From liuxianguo1982 at hotmail.com Sun Mar 18 04:07:24 2012 From: liuxianguo1982 at hotmail.com (=?gb2312?B?z9y5+iDB9Q==?=) Date: Sun, 18 Mar 2012 11:07:24 +0800 Subject: [FFmpeg-user] Question about compile ffmpeg for windows application on Ubuntu via cross-tools i686-w64-mingw32 Message-ID: Hi, I have a question about compile ffmpeg for windows application on Ubuntu via cross-tools i686-w64-mingw32. ../ffmpeg-0.8.7/configure --enable-cross-compile --cross-prefix=i686-w64-mingw32- --target-os=mingw32 --arch=x86 --prefix=/home/liuxianguo/opensdk/ffmpeg --disable-static --enable-shared --enable-version3 --enable-gpl --enable-nonfree --enable-w32threads --enable-runtime-cpudetect --enable-memalign-hack --enable-ffplay --extra-cflags="-I/home/liuxianguo/opensdk/include -I/home/liuxianguo/mingw/include" --extra-ldflags="-L/home/liuxianguo/opensdk/lib -L/home/liuxianguo/mingw/lib" --enable-libfaac --enable-libvo-aacenc --enable-libmp3lame --enable-libopenjpeg --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-zlib --extra-libs='-lx264 -lpthread' when I add --extra-libs='-lx264 -lpthread' root at ubuntu:/home/liuxianguo/studio/opensource/build-ffmpeg# ../ffmpeg-0.8.7/configure --enable-cross-compile --cross-prefix=i686-w64-mingw32- --pkg-config=pkg-config --target-os=mingw32 --arch=x86 --prefix=/home/liuxianguo/opensdk/ffmpeg --disable-static --enable-shared --enable-version3 --enable-gpl --enable-nonfree --enable-w32threads --enable-runtime-cpudetect --enable-memalign-hack --enable-ffplay --extra-cflags="-I/home/liuxianguo/opensdk/include -I/home/liuxianguo/studio/mingw-w64-v2.0.1/mingw-w64-headers/include" --extra-ldflags=-L/home/liuxianguo/opensdk/lib --enable-libfaac --enable-libvo-aacenc --enable-libmp3lame --enable-libopenjpeg --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --extra-libs='-lx264 -lpthread' --enable-libxvid --enable-zlib i686-w64-mingw32-gcc is unable to create an executable file. C compiler test failed. If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solving the problem. From wesleyacole84 at gmail.com Sun Mar 18 00:48:07 2012 From: wesleyacole84 at gmail.com (Wesley A Cole) Date: Sat, 17 Mar 2012 18:48:07 -0500 Subject: [FFmpeg-user] FFMPEG File Reading for conversion Message-ID: <003c01cd0498$68faf630$3af0e290$@gmail.com> Hi, I'm trying to make a .NET app that manages my media in all its forms (Shows, Music, Playlists, Movies, etc). I've got most of the work done for Shows and Music however I'm trying to develop a conversion attribute of the service that will actively read all my movie/show files and actively convert them to xvid format which is a pullable format from all my devices in my home. If possible I'd like to retain the 5.1 AAC format or at least convert the 5.1's to a usable 2CH MP3 or 2 CH AAC format so the audio isn't equal to the video's compression size. Can you help me d this possibly from the command line ffmpeg.exe binary. From andrey.krieger.utkin at gmail.com Sun Mar 18 07:26:56 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Sun, 18 Mar 2012 08:26:56 +0200 Subject: [FFmpeg-user] Question about compile ffmpeg for windows application on Ubuntu via cross-tools i686-w64-mingw32 In-Reply-To: References: Message-ID: 2012/3/18 ?? ? : > > Include the log file "config.log" produced by configure as this will help solving the problem. > \-- Andrey Utkin From mailer.tovis at freemail.hu Sun Mar 18 12:54:58 2012 From: mailer.tovis at freemail.hu (tovis) Date: Sun, 18 Mar 2012 12:54:58 +0100 Subject: [FFmpeg-user] Grabbing quality from TV card Message-ID: I'm still struggling to have good streaming quality for my TV card, driven by bttv driver module, and v4l2 subsystem. I have clear the support for libv4l2 (configuration option --enable-libv4l2) and error messages: "libv4l2: error dequeuing buf: Resource temporarily unavailable" are gone :) But the streaming quality still unacceptable :( Carl said that yadif filter is must be for interleaved video source, and only me could decide this by "visualization" OK! What I should have see? For now w/o yadif and qscale I can see big digital spots - really rude digitalization errors. I'm suspect that this is because of v4l2 settings. The ffmpeg, in case of "-f video4linux2 -i /dev/video0" changes any, previously set parameters for v4l2 subsystem, or simply use it, does not meter what is it set on? If ffmpeg leave v4l2 settings untouched, I can set them before or even change settings under active grabbing? Please give me a short description how ffmpeg handling the video4linux2 input. Sincerely tovis PS: I suspect that libv4l2 support have some problems, I have read several bug reports about ubuntu where they conclude that cause of this error is not in the driver, but the application. From anders.branderud at gmail.com Sun Mar 18 13:55:09 2012 From: anders.branderud at gmail.com (Anders Branderud) Date: Sun, 18 Mar 2012 14:55:09 +0200 Subject: [FFmpeg-user] Error message 'Could not open codec' when running FFMPEG-codec. Message-ID: Hello! I am running the following code: https://docs.google.com/document/d/1DbNpoy0MHNjS3ZV-6plaVdbmvW6g08riqtIz6fBO3ZE/edit Some excerpts: video_encode_example("/home/anders/grb_1.mpg", CODEC_ID_MPEG4); static void video_encode_example(const char *filename, int codec_id) { { //For full code see document attached above. /* open it */ if (avcodec_open2(c, codec, NULL) < 0) { fprintf(stderr, "could not open codec\n"); exit(1); } } The message 'could not open codec' is printed out. video_encode_example("/home/anders/grb_1.mpg", CODEC_ID_MPEG2VIDEO); generates the same error message. How do I resolve this? Thanks in advance! -- *Kind regards, Anders *[Personal blog] Will of the Creator : Logical reasons - based on scientific premises - for the existence of a Super intelligent and Orderly Creator and that He hasn't left His sapient creatures without an Instruction Manual - Torah ['books of Moses'] - to ascertain, and aspire to, His purpose. [Company] Anders Branderud IT Solutions - www.abitsolutions.org From deron at pagestream.org Sun Mar 18 16:50:24 2012 From: deron at pagestream.org (Deron) Date: Sun, 18 Mar 2012 09:50:24 -0600 Subject: [FFmpeg-user] How to crop extra audio channels? In-Reply-To: References: <4F640BBE.50602@pagestream.org> Message-ID: <4F660440.3000608@pagestream.org> On 3/17/12 5:46 AM, Carl Eugen Hoyos wrote: > Deron pagestream.org> writes: > >> ffmpeg -i "file.mov" -f s8 -vn -ac 2 -ar 48000 -acodec pcm_s8 -y >> "file.audio" > Sorry for my first comment, please try the following and report back: > ffmpeg -i file.mov -f s8 -map_channel 0.0.1 -map_channel 0.0.2 file.audio > > As explained in ticket #1088, -ac 2 would downmix from 16 to two channels, > since the layout for 16 channels is unknown, this is generally impossible. > > Carl Eugen > > Thanks for the suggestion, but it still output the full 16 channels! I guess I need to simply(?) export the full 16 channel to raw audio, write a little program to strip out the 2 channels, and then mux it back in? Deron From vsethi at iglou.com Sun Mar 18 18:34:46 2012 From: vsethi at iglou.com (vsethi at iglou.com) Date: Sun, 18 Mar 2012 13:34:46 -0400 Subject: [FFmpeg-user] Retrieving deleted segments oweQ== Message-ID: "dE ." wrote: >On 03/15/12 22:25, I wrote: >> Is there any way to retrieve an accidentally deleted segment from a >> non-finalized dvd-r? >> >> Thanks. > > You did a multisession DVD? Effectively, yes. From nicolas.george at normalesup.org Sun Mar 18 19:04:07 2012 From: nicolas.george at normalesup.org (Nicolas George) Date: Sun, 18 Mar 2012 19:04:07 +0100 Subject: [FFmpeg-user] How to crop extra audio channels? In-Reply-To: <4F660440.3000608@pagestream.org> References: <4F640BBE.50602@pagestream.org> <4F660440.3000608@pagestream.org> Message-ID: <20120318180407.GA12310@phare.normalesup.org> Le nonidi 29 vent?se, an CCXX, Deron a ?crit?: > Thanks for the suggestion, but it still output the full 16 channels! Please show your complete command line and console output. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From davidkempers at gmail.com Sun Mar 18 19:28:49 2012 From: davidkempers at gmail.com (David Kempers) Date: Sun, 18 Mar 2012 18:28:49 +0000 Subject: [FFmpeg-user] Error message 'Could not open codec' when running FFMPEG-codec. In-Reply-To: References: Message-ID: On 18 March 2012 12:55, Anders Branderud wrote: > Hello! > > I am running the following code: > > https://docs.google.com/document/d/1DbNpoy0MHNjS3ZV-6plaVdbmvW6g08riqtIz6fBO3ZE/edit > > Some excerpts: > video_encode_example("/home/anders/grb_1.mpg", CODEC_ID_MPEG4); > > static void video_encode_example(const char *filename, int codec_id) > { > { > //For full code see document attached above. > /* open it */ > if (avcodec_open2(c, codec, NULL) < 0) { > fprintf(stderr, "could not open codec\n"); > exit(1); > } > > } > > The message 'could not open codec' is printed out. > video_encode_example("/home/anders/grb_1.mpg", CODEC_ID_MPEG2VIDEO); > generates the same error message. > > How do I resolve this? > Thanks in advance! > > There's probably a wrong parameter with AVCodecContext. Off the top of my head I can't remember if CODEC_ID_MPEG2VIDEO and CODEC_ID_MPEG4 support b frames but I would try to set max_b_frames to 0. Are all the other parameters the same as the example? From anders.branderud at gmail.com Sun Mar 18 19:52:53 2012 From: anders.branderud at gmail.com (Anders Branderud) Date: Sun, 18 Mar 2012 20:52:53 +0200 Subject: [FFmpeg-user] Error message 'Could not open codec' when running FFMPEG-codec. In-Reply-To: References: Message-ID: Thanks for the reply! Yes, the other parameters are the same. I did set max_b_frames= 0. However the error message remains after compiling again. > > > I am running the following code: > > > > > https://docs.google.com/document/d/1DbNpoy0MHNjS3ZV-6plaVdbmvW6g08riqtIz6fBO3ZE/edit > > > > Some excerpts: > > video_encode_example("/home/anders/grb_1.mpg", CODEC_ID_MPEG4); > > > > static void video_encode_example(const char *filename, int codec_id) > > { > > { > > //For full code see document attached above. > > /* open it */ > > if (avcodec_open2(c, codec, NULL) < 0) { > > fprintf(stderr, "could not open codec\n"); > > exit(1); > > } > > > > } > > > > The message 'could not open codec' is printed out. > > video_encode_example("/home/anders/grb_1.mpg", CODEC_ID_MPEG2VIDEO); > > generates the same error message. > > > > How do I resolve this? > > Thanks in advance! > > > > > There's probably a wrong parameter with AVCodecContext. > > Off the top of my head I can't remember if CODEC_ID_MPEG2VIDEO and > CODEC_ID_MPEG4 support b frames but I would try to set max_b_frames to 0. > > Are all the other parameters the same as the example? > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- *Kind regards,* [Personal blog] Will of the Creator : Logical reasons - based on scientific premises - for the existence of a Super intelligent and Orderly Creator and that He hasn't left His sapient creatures without an Instruction Manual - Torah ['books of Moses'] - to ascertain, and aspire to, His purpose. [Company] Anders Branderud IT Solutions - www.abitsolutions.org From davidkempers at gmail.com Sun Mar 18 20:18:31 2012 From: davidkempers at gmail.com (David Kempers) Date: Sun, 18 Mar 2012 19:18:31 +0000 Subject: [FFmpeg-user] Error message 'Could not open codec' when running FFMPEG-codec. In-Reply-To: References: Message-ID: On 18 March 2012 18:52, Anders Branderud wrote: > Thanks for the reply! > Yes, the other parameters are the same. > I did set max_b_frames= 0. However the error message remains after > compiling again. > > Looking through the example I can't see if it sets codec_id parameter of AVCodecContext. I would add the line: c->codec_id = CODEC_ID_MPEG2VIDEO; From anders.branderud at gmail.com Sun Mar 18 20:34:45 2012 From: anders.branderud at gmail.com (Anders Branderud) Date: Sun, 18 Mar 2012 21:34:45 +0200 Subject: [FFmpeg-user] Error message 'Could not open codec' when running FFMPEG-codec. In-Reply-To: References: Message-ID: I inserted ' c->codec_id = CODEC_ID_MPEG2VIDEO' into the code and the outcome is the same. 2012/3/18 David Kempers > On 18 March 2012 18:52, Anders Branderud > wrote: > > > Thanks for the reply! > > Yes, the other parameters are the same. > > I did set max_b_frames= 0. However the error message remains after > > compiling again. > > > > > Looking through the example I can't see if it sets codec_id parameter of > AVCodecContext. I would add the line: > > c->codec_id = CODEC_ID_MPEG2VIDEO; > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- *Kind regards,* [Personal blog] Will of the Creator : Logical reasons - based on scientific premises - for the existence of a Super intelligent and Orderly Creator and that He hasn't left His sapient creatures without an Instruction Manual - Torah ['books of Moses'] - to ascertain, and aspire to, His purpose. [Company] Anders Branderud IT Solutions - www.abitsolutions.org From mailer.tovis at freemail.hu Sun Mar 18 21:22:00 2012 From: mailer.tovis at freemail.hu (tovis) Date: Sun, 18 Mar 2012 21:22:00 +0100 Subject: [FFmpeg-user] Grabbing quality from TV card In-Reply-To: References: <4baa3807f82725a943f82dfdcb0db892.squirrel@nusi> <9e46fb78816217205cf6aef85ee7f481.squirrel@nusi> <465798b6b8eeb0ee0a148a84f773b315.squirrel@nusi> Message-ID: Hi! I have reach some progress :D Some one later suggest to use h264 codec (I think dE.) - I've decided to give a chance. I've downloaded and compile x264 library from git (I have several pain, because I have to get also latest yasm to compile it with appropriate assembler, and also install Debian package autoconf). But the result was prove this. I have using command: $ ffmpeg -v verbose -f video4linux2 -i /dev/video0 -r 29.97 -s 640x480 -vcodec libx264 \ -t 10 -f avi -y /tmp/test01.avi In result I have resonable good quality of the video stream :D Next I have tried to get sound. As it suggested in ffmpeg documentation I have checked available options for alsa: $ cat /proc/asound/cards 0 [SB ]: HDA-Intel - HDA ATI SB HDA ATI SB at 0xfe024000 irq 16 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfdffc000 irq 19 2 [Bt878 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xfdbfe000, irq 20 $ cat /proc/asound/devs 2: : timer 3: : sequencer 4: [ 0- 2]: digital audio capture 5: [ 0- 1]: digital audio playback 6: [ 0- 1]: digital audio capture 7: [ 0- 0]: digital audio playback 8: [ 0- 0]: digital audio capture 9: [ 0- 0]: hardware dependent 10: [ 0] : control 11: [ 1- 3]: digital audio playback 12: [ 1- 0]: hardware dependent 13: [ 1] : control 14: [ 2- 1]: digital audio capture 15: [ 2- 0]: digital audio capture 16: [ 2] : control I have using command for grabbing: $ ffmpeg -v verbose -f alsa -ac 2 -i hw:0 -f video4linux2 -i /dev/video0 \ -r 29.97 -s 640x480 -vcodec libx264 \ -f avi -y test02.avi No sound :( I have tried hw:0, hw:1 and hw:2 as input, but nothing. I have compile latest mp3lame from sourceforge. No result. But in video stream, mp3 track is exist (I have checked on windows using mediainfo)!? I goes back to check v4l2 facilities. I have list available controls for /dev/video0, using command v4l2-ctl (getting from Squeeze repository), here is the result: brightness (int) : min=0 max=65535 step=256 default=32768 value=32768 contrast (int) : min=0 max=65535 step=128 default=32768 value=32768 saturation (int) : min=0 max=65535 step=128 default=32768 value=32768 hue (int) : min=0 max=65535 step=256 default=32768 value=32768 balance (int) : min=0 max=65535 step=655 default=32768 value=32768 bass (int) : min=0 max=65535 step=655 default=32768 value=32768 treble (int) : min=0 max=65535 step=655 default=32768 value=32768 mute (bool) : default=0 value=1 chroma_agc (bool) : default=0 value=0 combfilter (bool) : default=0 value=0 automute (bool) : default=0 value=1 luma_decimation_filter (bool) : default=0 value=0 agc_crush (bool) : default=0 value=1 vcr_hack (bool) : default=0 value=0 whitecrush_upper (int) : min=0 max=255 step=1 default=207 value=207 whitecrush_lower (int) : min=0 max=255 step=1 default=127 value=127 uv_ratio (int) : min=0 max=100 step=1 default=50 value=50 full_luma_range (bool) : default=0 value=0 coring (int) : min=0 max=3 step=1 default=0 value=0 Tuner card internal sound output attached to CD input on mainboard for onboard audio. I've goes back to "-i hw:0" and start grabbing w/o time limit, in other console I have set mute to '0' : $ v4l2-ctl --set-ctrl=mute=0 Sound from TV tuner is up, after several seconds, I have stopped grabbing, when ffmpeg step out to shell sound is goes down - I have checked the sort clipp - I've got sound :D Happy? Not yet. I need NO sound on the card's host box - I suppose to use it as a stream server. The best (for me) if I could get sound from tuner card directly, avoid sound card. I have tried similar procedure using grabbing command: $ ffmpeg -v verbose -f alsa -ac 2 -i hw:2 -f video4linux2 -i /dev/video0 \ -r 29.97 -s 640x480 -vcodec libx264 \ -f avi -y test02.avi But no success. Is it possible to get sound avoid the sound card? Any suggestion? Sincerely tovis From davidkempers at gmail.com Sun Mar 18 21:32:23 2012 From: davidkempers at gmail.com (David Kempers) Date: Sun, 18 Mar 2012 20:32:23 +0000 Subject: [FFmpeg-user] Error message 'Could not open codec' when running FFMPEG-codec. In-Reply-To: References: Message-ID: On 18 March 2012 19:34, Anders Branderud wrote: > I inserted ' c->codec_id = CODEC_ID_MPEG2VIDEO' into the code and the > outcome is the same. > > You need to change "/home/anders/grb_1.mpg" to a directory that exists and is writeable. From cehoyos at ag.or.at Sun Mar 18 22:05:37 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 18 Mar 2012 21:05:37 +0000 (UTC) Subject: [FFmpeg-user] Grabbing quality from TV card References: Message-ID: tovis freemail.hu> writes: > Carl said that yadif filter is must be for interleaved video source, and No, I wrote that you need yadif if your input is interlaced. > only me could decide this by "visualization" OK! What I should have see? What did Google tell you when your searched for "interlaced"? > For now w/o yadif and qscale I can see big digital spots - really rude > digitalization errors. Command line and complete, uncut console output missing. I use the video4linux input often and it works fine, please note that the FFmpeg version Ubuntu includes is intentionally broken, so it is not surprising that there are reports it is buggy. Carl Eugen From chenct1976 at sina.com Mon Mar 19 04:38:25 2012 From: chenct1976 at sina.com (chenct1976 at sina.com) Date: Mon, 19 Mar 2012 11:38:25 +0800 Subject: [FFmpeg-user] Mkv convert avi problem: Output file duration is not right: Message-ID: <20120319033825.95117DD8001@webmail.sinamail.sina.com.cn> Dear all, I have met a problem when I convert a mkv file to avi. Problem: input file duration is 25s, output file is 3'41s. v0.8 is ok, but v0.10 is wrong. OS : MINGW32_NT-6.1 DEV-TOM 1.0.17(0.48/3/2) 2011-04-24 23:39 i686 Msys Command line: ./ffmpeg_g.exe -i intput1.mkv -vcodec mjpeg -b 2500000 -r 30 -s 640x480 -acodec pcm_s16le -ac 2 -ab 160000 -ar 44100 -y intput1.avi Test input file link: http://www.mediafire.com/?bb1x23c4wzwff95 Test output log: 2012/03/19-11:16: ffmpeg version 0.10 2012/03/19-11:16: Copyright (c) 2000-2012 the FFmpeg developers 2012/03/19-11:16: built on Mar 19 2012 10:33:03 with gcc 4.5.2 2012/03/19-11:16: configuration: --prefix=/ffmpeg/release10 --enable-debug --disable-static --enable-shared --enable-gpl --enable-version3 --enable-avfilter --enable-memalign-hack --enable-avisynth --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libfaac --enable-nonfree --enable-libspeex --enable-libopenjpeg --enable-libxavs --enable-libvpx --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libschroedinger --enable-libcelt --enable-frei0r --enable-libdirac --disable-decoder=libdirac --extra-cflags=-I/ffmpeg/olibs9/include --extra-ldflags=-L/ffmpeg/olibs9/lib --extra-libs=/mingw/lib/libdl.a 2012/03/19-11:16: libavutil 51. 34.101 / 51. 34.101 2012/03/19-11:16: libavcodec 53. 60.100 / 53. 60.100 2012/03/19-11:16: libavformat 53. 31.100 / 53. 31.100 2012/03/19-11:16: libavdevice 53. 4.100 / 53. 4.100 2012/03/19-11:16: libavfilter 2. 60.100 / 2. 60.100 2012/03/19-11:16: libswscale 2. 1.100 / 2. 1.100 2012/03/19-11:16: libswresample 0. 6.100 / 0. 6.100 2012/03/19-11:16: libpostproc 52. 0.100 / 52. 0.100 2012/03/19-11:16: Input #0, matroska,webm, from 'intput1.mkv': 2012/03/19-11:16: Duration: 2012/03/19-11:16: 00:00:25.39 2012/03/19-11:16: start: 2012/03/19-11:16: 0.000000 2012/03/19-11:16: bitrate: 2012/03/19-11:16: 966 kb/s 2012/03/19-11:16: Stream #0:0 2012/03/19-11:16: Video: h264 (Constrained Baseline), yuv420p, 640x480 2012/03/19-11:16: SAR 1:1 DAR 4:3 2012/03/19-11:16: 30 tbr 2012/03/19-11:16: 1k tbn 2012/03/19-11:16: 60 tbc 2012/03/19-11:16: (default) 2012/03/19-11:16: Stream #0:1 2012/03/19-11:16: Audio: aac, 11025 Hz, mono, s16 2012/03/19-11:16: (default) 2012/03/19-11:16: Please use -b:a or -b:v, -b is ambiguous 2012/03/19-11:16: Incompatible pixel format 'yuv420p' for codec 'mjpeg', auto-selecting format 'yuvj420p' 2012/03/19-11:16: [buffer @ 02625fa0] w:640 h:480 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param: 2012/03/19-11:16: [buffersink @ 025c3ea0] auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'out' 2012/03/19-11:16: [scale @ 0256b820] w:640 h:480 fmt:yuv420p -> w:640 h:480 fmt:yuvj420p flags:0x4 2012/03/19-11:16: Output #0, avi, to 'intput1.avi': 2012/03/19-11:16: Metadata: 2012/03/19-11:16: ISFT : Lavf53.31.100 2012/03/19-11:16: Stream #0:0 2012/03/19-11:16: Video: mjpeg (MJPG / 0x47504A4D), yuvj420p, 640x480 [SAR 1:1 DAR 4:3], q=2-31, 2500 kb/s 2012/03/19-11:16: 30 tbn 2012/03/19-11:16: 30 tbc 2012/03/19-11:16: (default) 2012/03/19-11:16: Stream #0:1 2012/03/19-11:16: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s 2012/03/19-11:16: (default) 2012/03/19-11:16: Stream mapping: 2012/03/19-11:16: Stream #0:0 -> #0:0 2012/03/19-11:16: (h264 -> mjpeg) 2012/03/19-11:16: Stream #0:1 -> #0:1 2012/03/19-11:16: (aac -> pcm_s16le) 2012/03/19-11:16: Press [q] to stop, [?] for help 2012/03/19-11:16: frame= 79 fps= 0 q=24.8 size= 1562kB time=00:00:02.63 bitrate=4859.2kbits/s 2012/03/19-11:16: frame= 149 fps=148 q=24.8 size= 2731kB time=00:00:04.96 bitrate=4504.8kbits/s 2012/03/19-11:16: frame= 235 fps=155 q=24.8 size= 4170kB time=00:00:07.83 bitrate=4360.4kbits/s 2012/03/19-11:16: frame= 325 fps=161 q=24.8 size= 5686kB time=00:00:10.83 bitrate=4300.0kbits/s 2012/03/19-11:16: frame= 411 fps=163 q=24.8 size= 7154kB time=00:00:13.70 bitrate=4277.8kbits/s 2012/03/19-11:16: frame= 500 fps=165 q=24.8 size= 8646kB time=00:00:16.66 bitrate=4249.9kbits/s 2012/03/19-11:16: frame= 589 fps=167 q=24.8 size= 10114kB time=00:00:19.63 bitrate=4220.2kbits/s 2012/03/19-11:16: frame= 676 fps=168 q=24.8 size= 11562kB time=00:00:22.53 bitrate=4203.5kbits/s 2012/03/19-11:16: frame= 762 fps=169 q=24.8 Lsize= 12953kB time=00:00:25.07 bitrate=4231.6kbits/s 2012/03/19-11:16: video:8599kB audio:4320kB global headers:0kB muxing overhead 0.265536%. Can yout tell me why? Any reply is appreciated! Thank you in advance! Tom.chen. From j.baumgarten at netvisio.com Mon Mar 19 09:44:41 2012 From: j.baumgarten at netvisio.com (Julien Baumgarten) Date: Mon, 19 Mar 2012 09:44:41 +0100 Subject: [FFmpeg-user] Problem with ffmpeg and RTMP Message-ID: Hi everyone, I developed that stream and get webcam streams. I developed it using xuggler on a mac and my software works very well. I tried to use it on a windows laptop. I succeed to compile the different libraries but I got an error coming from ffmpeg. Apparently, the RTMP stream is not found but on a Mac the stream is found. Is there an issue on RTMP stream and ffmpeg? Yours sincerely, Julien BAUMGARTEN From anders.branderud at gmail.com Mon Mar 19 10:36:21 2012 From: anders.branderud at gmail.com (Anders Branderud) Date: Mon, 19 Mar 2012 11:36:21 +0200 Subject: [FFmpeg-user] Error message 'Could not open codec' when running FFMPEG-codec. In-Reply-To: References: Message-ID: The directory exists and is writable. I also tested to open the file by a function-call in the code and it works. 2012/3/18 David Kempers > On 18 March 2012 19:34, Anders Branderud > wrote: > > > I inserted ' c->codec_id = CODEC_ID_MPEG2VIDEO' into the code and the > > outcome is the same. > > > > > You need to change "/home/anders/grb_1.mpg" to a directory that exists and > is writeable. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- *Kind regards,* [Personal blog] Will of the Creator : Logical reasons - based on scientific premises - for the existence of a Super intelligent and Orderly Creator and that He hasn't left His sapient creatures without an Instruction Manual - Torah ['books of Moses'] - to ascertain, and aspire to, His purpose. [Company] Anders Branderud IT Solutions - www.abitsolutions.org From cehoyos at ag.or.at Mon Mar 19 11:30:56 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 19 Mar 2012 10:30:56 +0000 (UTC) Subject: [FFmpeg-user] Problem with ffmpeg and RTMP References: Message-ID: Julien Baumgarten netvisio.com> writes: > I tried to use it on a windows laptop. I succeed to compile > the different libraries but I got an error coming from ffmpeg. Unfortunately, this is not enough information to understand your problem. Carl Eugen From j.baumgarten at netvisio.com Mon Mar 19 11:38:04 2012 From: j.baumgarten at netvisio.com (Julien BAUMGARTEN) Date: Mon, 19 Mar 2012 11:38:04 +0100 Subject: [FFmpeg-user] Problem with ffmpeg and RTMP In-Reply-To: References: Message-ID: <41732573-C111-437B-964A-B9C7374CD142@netvisio.com> Hi Carl, I am sorry if I did not explain enough my problem. On my Mac, my software succeed to connect and get the stream from a RTMP server. Unfortunately, when I try on a Windows, I got the error that the stream is not found. The RTMP URL is like: rtmp:// [IP] /confcam/myStream Yours sincerely, Julien BAUMGARTEN Le 19 mars 2012 ? 11:30, Carl Eugen Hoyos a ?crit : > Julien Baumgarten netvisio.com> writes: > >> I tried to use it on a windows laptop. I succeed to compile >> the different libraries but I got an error coming from ffmpeg. > > Unfortunately, this is not enough information to understand your problem. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From cehoyos at ag.or.at Mon Mar 19 11:43:16 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 19 Mar 2012 10:43:16 +0000 (UTC) Subject: [FFmpeg-user] Problem with ffmpeg and RTMP References: <41732573-C111-437B-964A-B9C7374CD142@netvisio.com> Message-ID: Julien BAUMGARTEN netvisio.com> writes: > Unfortunately, when I try on a Windows, I got the error that the > stream is not found. Please post the command line together with complete, uncut console output. Consider also posting command line and output for the working case (Mac). Please try to avoid top-posting here, Carl Eugen From mailer.tovis at freemail.hu Mon Mar 19 13:36:01 2012 From: mailer.tovis at freemail.hu (tovis) Date: Mon, 19 Mar 2012 13:36:01 +0100 Subject: [FFmpeg-user] Grabbing quality from TV card Message-ID: REMARK: This is a secondary sent email, first seem to be spinning some where :( Hi! I have reach some progress :D Some one later suggest to use h264 codec (I think dE.) - I've decided to give a chance. I've downloaded and compile x264 library from git (I have several pain, because I have to get also latest yasm to compile it with appropriate assembler, and also install Debian package autoconf). But the result was prove this. I have using command: $ ffmpeg -v verbose -f video4linux2 -i /dev/video0 -r 29.97 -s 640x480 -vcodec libx264 \ -t 10 -f avi -y /tmp/test01.avi In result I have resonable good quality of the video stream :D Next I have tried to get sound. As it suggested in ffmpeg documentation I have checked available options for alsa: $ cat /proc/asound/cards 0 [SB ]: HDA-Intel - HDA ATI SB HDA ATI SB at 0xfe024000 irq 16 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfdffc000 irq 19 2 [Bt878 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xfdbfe000, irq 20 $ cat /proc/asound/devs 2: : timer 3: : sequencer 4: [ 0- 2]: digital audio capture 5: [ 0- 1]: digital audio playback 6: [ 0- 1]: digital audio capture 7: [ 0- 0]: digital audio playback 8: [ 0- 0]: digital audio capture 9: [ 0- 0]: hardware dependent 10: [ 0] : control 11: [ 1- 3]: digital audio playback 12: [ 1- 0]: hardware dependent 13: [ 1] : control 14: [ 2- 1]: digital audio capture 15: [ 2- 0]: digital audio capture 16: [ 2] : control I have using command for grabbing: $ ffmpeg -v verbose -f alsa -ac 2 -i hw:0 -f video4linux2 -i /dev/video0 \ -r 29.97 -s 640x480 -vcodec libx264 \ -f avi -y test02.avi No sound :( I have tried hw:0, hw:1 and hw:2 as input, but nothing. I have compile latest mp3lame from sourceforge. No result. But in video stream, mp3 track is exist (I have checked on windows using mediainfo)!? I goes back to check v4l2 facilities. I have list available controls for /dev/video0, using command v4l2-ctl (getting from Squeeze repository), here is the result: brightness (int) : min=0 max=65535 step=256 default=32768 value=32768 contrast (int) : min=0 max=65535 step=128 default=32768 value=32768 saturation (int) : min=0 max=65535 step=128 default=32768 value=32768 hue (int) : min=0 max=65535 step=256 default=32768 value=32768 balance (int) : min=0 max=65535 step=655 default=32768 value=32768 bass (int) : min=0 max=65535 step=655 default=32768 value=32768 treble (int) : min=0 max=65535 step=655 default=32768 value=32768 mute (bool) : default=0 value=1 chroma_agc (bool) : default=0 value=0 combfilter (bool) : default=0 value=0 automute (bool) : default=0 value=1 luma_decimation_filter (bool) : default=0 value=0 agc_crush (bool) : default=0 value=1 vcr_hack (bool) : default=0 value=0 whitecrush_upper (int) : min=0 max=255 step=1 default=207 value=207 whitecrush_lower (int) : min=0 max=255 step=1 default=127 value=127 uv_ratio (int) : min=0 max=100 step=1 default=50 value=50 full_luma_range (bool) : default=0 value=0 coring (int) : min=0 max=3 step=1 default=0 value=0 Tuner card internal sound output attached to CD input on mainboard for onboard audio. I've goes back to "-i hw:0" and start grabbing w/o time limit, in other console I have set mute to '0' : $ v4l2-ctl --set-ctrl=mute=0 Sound from TV tuner is up, after several seconds, I have stopped grabbing, when ffmpeg step out to shell sound is goes down - I have checked the sort clipp - I've got sound :D Happy? Not yet. I need NO sound on the card's host box - I suppose to use it as a stream server. The best (for me) if I could get sound from tuner card directly, avoid sound card. I have tried similar procedure using grabbing command: $ ffmpeg -v verbose -f alsa -ac 2 -i hw:2 -f video4linux2 -i /dev/video0 \ -r 29.97 -s 640x480 -vcodec libx264 \ -f avi -y test02.avi But no success. Is it possible to get sound avoid the sound card? Any suggestion? Sincerely tovis From jacobhameiri at gmail.com Mon Mar 19 17:43:22 2012 From: jacobhameiri at gmail.com (jacob s) Date: Mon, 19 Mar 2012 18:43:22 +0200 Subject: [FFmpeg-user] increase the size of the real-time buffer. In-Reply-To: <4F6450E8.6020103@gmail.com> References: <4F6450E8.6020103@gmail.com> Message-ID: found it, the option I was looking for is rtbufsize 2012/3/17 dE . > On 03/16/12 03:10, jacob s wrote: > >> Hi, >> I am streaming windows directshow desktop capture via udp and I am getting >> occasional "real-time buffer 155% full! frame dropped!" errors, which are >> causing audio ( and rarely video ) glitches. >> Is there a way to increase the size of the real-time buffer ? I am willing >> to get higher latency if it will solve the frame drops. >> This is my command: >> -f dshow -i video="screen-capture-**recorder":audio="Stereo Mix (IDT High >> Definition" -vcodec libx264 -preset ultrafast -tune zerolatency -r 10 >> -async 1 -acodec libmp3lame -ab 24k -ar 22050 -bsf:v h264_mp4toannexb >> -maxrate 750k -bufsize 3000k -f mpegts udp://192.168.5.215:48550 >> ______________________________**_________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/**listinfo/ffmpeg-user >> > > buffer_size=size > > It's in the man page. > ______________________________**_________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/**listinfo/ffmpeg-user > From jacobhameiri at gmail.com Mon Mar 19 17:51:10 2012 From: jacobhameiri at gmail.com (jacob s) Date: Mon, 19 Mar 2012 18:51:10 +0200 Subject: [FFmpeg-user] correct syntax for using the pan filter Message-ID: anyone knows the proper way to use the pan filter ? I am trying to take a stereo input and mute one channel and get a stereo output with one channel muted. This is my original command: -rtbufsize 100000000 -f dshow -i video="screen-capture-recorder":audio="SoundMAX Digital Audio" -vcodec libx264 -preset ultrafast -tune zerolatency -r 10 -async 1 -ab 32k -ar 22050 -bsf:v h264_mp4toannexb -b 614400 -f mpegts udp://192.168.5.215:48550 According to the manual this is possible ( "If the input is a stereo audio stream, you can mute the front left channel (and still keep the stereo channel layout) with: pan="stereo:c1=c1" ") I tried this but it didn't work, I guess I am not using the correct cli syntax. Please reply with the correct way to use pan ( within my original command ) Thanks From arty.net2 at gmail.com Mon Mar 19 19:36:18 2012 From: arty.net2 at gmail.com (Arturo Rinaldi) Date: Mon, 19 Mar 2012 19:36:18 +0100 Subject: [FFmpeg-user] configuring source for x86_64 architectures Message-ID: <4F677CA2.2070404@gmail.com> i put the switch */--arch=x86_64/* in the */./configure/* step, however the configure output returns always */x86 (generic)/* as architecture. I also tried to put */--arch=amd64/* but it didn't solve the problem. Can you help me ? Regards, Arturo From cehoyos at ag.or.at Mon Mar 19 20:13:23 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 19 Mar 2012 19:13:23 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?configuring_source_for_x86=5F64_architect?= =?utf-8?q?ures?= References: <4F677CA2.2070404@gmail.com> Message-ID: Arturo Rinaldi gmail.com> writes: > i put the switch */--arch=x86_64/* in the */./configure/* step, however > the configure output returns always */x86 (generic)/* as architecture. I > also tried to put */--arch=amd64/* but it didn't solve the problem. (Missing configure line and "cc -v" output.) configure will select the correct architecture for your system's default compiler if you do not explicitly set a compiler with "--cc=". To compile for 32bit on a 64 bit system, you can for example use: ./configure --cc='cc -m32' Afaict, --arch is only useful if you cross-compile. Carl Eugen From cehoyos at ag.or.at Mon Mar 19 20:14:53 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 19 Mar 2012 19:14:53 +0000 (UTC) Subject: [FFmpeg-user] correct syntax for using the pan filter References: Message-ID: jacob s gmail.com> writes: > According to the manual this is possible ( "If the input is a stereo audio > stream, you can mute the front left channel (and still keep the stereo > channel layout) with: pan="stereo:c1=c1" ") > I tried this but it didn't work, I guess I am not using the correct cli > syntax. Command line (with -vf pan) and complete, uncut console output missing. Carl Eugen From jacobhameiri at gmail.com Mon Mar 19 20:42:09 2012 From: jacobhameiri at gmail.com (jacob s) Date: Mon, 19 Mar 2012 21:42:09 +0200 Subject: [FFmpeg-user] correct syntax for using the pan filter In-Reply-To: References: Message-ID: ffmpeg started on 2012-03-19 at 21:36:53 Report written to "ffmpeg-20120319-213653.log" Command line: ffmpeg.exe -rtbufsize 100000000 -f dshow -i "video=screen-capture-recorder:audio=Stereo Mix (IDT High Definition" -vcodec libx264 -preset ultrafast -tune zerolatency -r 10 -async 1 -ab 32k -ar 22050 -bsf:v h264_mp4toannexb -b 614400 -vf "pan=stereo:c1=c1" -f mpegts udp://192.168.5.215:48551 -report ffmpeg version N-38292-ga4c22e3 Copyright (c) 2000-2012 the FFmpeg developers built on Feb 27 2012 14:50:39 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 41.100 / 51. 41.100 libavcodec 54. 4.100 / 54. 4.100 libavformat 54. 1.100 / 54. 1.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 62.101 / 2. 62.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 [dshow @ 01F4E4E0] Probe buffer size limit 5000000 reached [dshow @ 01F4E4E0] Estimating duration from bitrate, this may be inaccurate Input #0, dshow, from 'video=screen-capture-recorder:audio=Stereo Mix (IDT High Definition': Duration: N/A, start: 560775.791000, bitrate: 1411 kb/s Stream #0:0, 1, 1/10000000: Video: rawvideo, bgr24, 1600x900, 10 tbr, 10000k tbn, 10 tbc Stream #0:1, 0, 1/10000000: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s Please use -b:a or -b:v, -b is ambiguous Incompatible pixel format 'bgr24' for codec 'libx264', auto-selecting format 'yuv420p' [buffer @ 03581300] w:1600 h:900 pixfmt:bgr24 tb:1/1000000 sar:0/1 sws_param: [buffer @ 03581300] Media type mismatch between the 'src' filter output pad 0 and the 'Parsed_pan_0' filter input pad 0 Cannot create the link buffer:0 -> pan:0 Error opening filters! 2012/3/19 Carl Eugen Hoyos > jacob s gmail.com> writes: > > > According to the manual this is possible ( "If the input is a stereo > audio > > stream, you can mute the front left channel (and still keep the stereo > > channel layout) with: pan="stereo:c1=c1" ") > > I tried this but it didn't work, I guess I am not using the correct cli > > syntax. > > Command line (with -vf pan) and complete, uncut console output missing. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From chenct1976 at sina.com Tue Mar 20 01:57:06 2012 From: chenct1976 at sina.com (chenct1976 at sina.com) Date: Tue, 20 Mar 2012 08:57:06 +0800 Subject: [FFmpeg-user] Could you please give some advice ? // Mkv convert avi problem: Output file duration is not right: Message-ID: <20120320005706.285D5E8C5C1@webmail.sinamail.sina.com.cn> Dear all, I have met a problem when I convert a mkv file to avi. Problem: input file duration is 25s, output file is 3'41s. v0.8 is ok, but v0.10 is wrong. OS : MINGW32_NT-6.1 DEV-TOM 1.0.17(0.48/3/2) 2011-04-24 23:39 i686 Msys Command line: ./ffmpeg_g.exe -i intput1.mkv -vcodec mjpeg -b 2500000 -r 30 -s 640x480 -acodec pcm_s16le -ac 2 -ab 160000 -ar 44100 -y intput1.avi Test input file link: http://www.mediafire.com/?bb1x23c4wzwff95 Test output log: 2012/03/19-11:16: ffmpeg version 0.10 2012/03/19-11:16: Copyright (c) 2000-2012 the FFmpeg developers 2012/03/19-11:16: built on Mar 19 2012 10:33:03 with gcc 4.5.2 2012/03/19-11:16: configuration: --prefix=/ffmpeg/release10 --enable-debug --disable-static --enable-shared --enable-gpl --enable-version3 --enable-avfilter --enable-memalign-hack --enable-avisynth --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libfaac --enable-nonfree --enable-libspeex --enable-libopenjpeg --enable-libxavs --enable-libvpx --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libschroedinger --enable-libcelt --enable-frei0r --enable-libdirac --disable-decoder=libdirac --extra-cflags=-I/ffmpeg/olibs9/include --extra-ldflags=-L/ffmpeg/olibs9/lib --extra-libs=/mingw/lib/libdl.a 2012/03/19-11:16: libavutil 51. 34.101 / 51. 34.101 2012/03/19-11:16: libavcodec 53. 60.100 / 53. 60.100 2012/03/19-11:16: libavformat 53. 31.100 / 53. 31.100 2012/03/19-11:16: libavdevice 53. 4.100 / 53. 4.100 2012/03/19-11:16: libavfilter 2. 60.100 / 2. 60.100 2012/03/19-11:16: libswscale 2. 1.100 / 2. 1.100 2012/03/19-11:16: libswresample 0. 6.100 / 0. 6.100 2012/03/19-11:16: libpostproc 52. 0.100 / 52. 0.100 2012/03/19-11:16: Input #0, matroska,webm, from 'intput1.mkv': 2012/03/19-11:16: Duration: 2012/03/19-11:16: 00:00:25.39 2012/03/19-11:16: start: 2012/03/19-11:16: 0.000000 2012/03/19-11:16: bitrate: 2012/03/19-11:16: 966 kb/s 2012/03/19-11:16: Stream #0:0 2012/03/19-11:16: Video: h264 (Constrained Baseline), yuv420p, 640x480 2012/03/19-11:16: SAR 1:1 DAR 4:3 2012/03/19-11:16: 30 tbr 2012/03/19-11:16: 1k tbn 2012/03/19-11:16: 60 tbc 2012/03/19-11:16: (default) 2012/03/19-11:16: Stream #0:1 2012/03/19-11:16: Audio: aac, 11025 Hz, mono, s16 2012/03/19-11:16: (default) 2012/03/19-11:16: Please use -b:a or -b:v, -b is ambiguous 2012/03/19-11:16: Incompatible pixel format 'yuv420p' for codec 'mjpeg', auto-selecting format 'yuvj420p' 2012/03/19-11:16: [buffer @ 02625fa0] w:640 h:480 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param: 2012/03/19-11:16: [buffersink @ 025c3ea0] auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'out' 2012/03/19-11:16: [scale @ 0256b820] w:640 h:480 fmt:yuv420p -> w:640 h:480 fmt:yuvj420p flags:0x4 2012/03/19-11:16: Output #0, avi, to 'intput1.avi': 2012/03/19-11:16: Metadata: 2012/03/19-11:16: ISFT : Lavf53.31.100 2012/03/19-11:16: Stream #0:0 2012/03/19-11:16: Video: mjpeg (MJPG / 0x47504A4D), yuvj420p, 640x480 [SAR 1:1 DAR 4:3], q=2-31, 2500 kb/s 2012/03/19-11:16: 30 tbn 2012/03/19-11:16: 30 tbc 2012/03/19-11:16: (default) 2012/03/19-11:16: Stream #0:1 2012/03/19-11:16: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s 2012/03/19-11:16: (default) 2012/03/19-11:16: Stream mapping: 2012/03/19-11:16: Stream #0:0 -> #0:0 2012/03/19-11:16: (h264 -> mjpeg) 2012/03/19-11:16: Stream #0:1 -> #0:1 2012/03/19-11:16: (aac -> pcm_s16le) 2012/03/19-11:16: Press [q] to stop, [?] for help 2012/03/19-11:16: frame= 79 fps= 0 q=24.8 size= 1562kB time=00:00:02.63 bitrate=4859.2kbits/s 2012/03/19-11:16: frame= 149 fps=148 q=24.8 size= 2731kB time=00:00:04.96 bitrate=4504.8kbits/s 2012/03/19-11:16: frame= 235 fps=155 q=24.8 size= 4170kB time=00:00:07.83 bitrate=4360.4kbits/s 2012/03/19-11:16: frame= 325 fps=161 q=24.8 size= 5686kB time=00:00:10.83 bitrate=4300.0kbits/s 2012/03/19-11:16: frame= 411 fps=163 q=24.8 size= 7154kB time=00:00:13.70 bitrate=4277.8kbits/s 2012/03/19-11:16: frame= 500 fps=165 q=24.8 size= 8646kB time=00:00:16.66 bitrate=4249.9kbits/s 2012/03/19-11:16: frame= 589 fps=167 q=24.8 size= 10114kB time=00:00:19.63 bitrate=4220.2kbits/s 2012/03/19-11:16: frame= 676 fps=168 q=24.8 size= 11562kB time=00:00:22.53 bitrate=4203.5kbits/s 2012/03/19-11:16: frame= 762 fps=169 q=24.8 Lsize= 12953kB time=00:00:25.07 bitrate=4231.6kbits/s 2012/03/19-11:16: video:8599kB audio:4320kB global headers:0kB muxing overhead 0.265536%. Can yout tell me why? Any reply is appreciated! Thank you in advance! Tom.chen. _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From way2012 at live.cn Mon Mar 19 17:50:28 2012 From: way2012 at live.cn (luchen) Date: Mon, 19 Mar 2012 16:50:28 +0000 Subject: [FFmpeg-user] I can't find any encoders Message-ID: Hello,I get a big trouble.When I the codes blew,I can't find any encoders.But when I configure,I did add the code ?--enable-encoders?. codec = avcodec_find_encoder(CODEC_ID_MPEG1VIDEO); if (!codec) { fprintf(stderr, "codec not found\n"); exit(1); }I just want to know have you ever met this problem and can you give me some advise.Thanks a lot.Best wishes. Thomas Lu China, Nanjing From cehoyos at ag.or.at Tue Mar 20 09:03:59 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 20 Mar 2012 08:03:59 +0000 (UTC) Subject: [FFmpeg-user] Mkv convert avi problem: Output file duration is not right: References: <20120319033825.95117DD8001@webmail.sinamail.sina.com.cn> Message-ID: sina.com> writes: > I have met a problem when I convert a mkv file to avi. > Problem: input file duration is 25s, output file is 3'41s. > v0.8 is ok, but v0.10 is wrong. This is a regression, it has not been fixed yet. Carl Eugen From de.techno at gmail.com Tue Mar 20 10:09:38 2012 From: de.techno at gmail.com (dE .) Date: Tue, 20 Mar 2012 14:39:38 +0530 Subject: [FFmpeg-user] Grabbing quality from TV card In-Reply-To: References: <4baa3807f82725a943f82dfdcb0db892.squirrel@nusi> <9e46fb78816217205cf6aef85ee7f481.squirrel@nusi> <465798b6b8eeb0ee0a148a84f773b315.squirrel@nusi> Message-ID: <4F684952.9050007@gmail.com> On 03/19/12 01:52, tovis wrote: > Hi! > I have reach some progress :D > Some one later suggest to use h264 codec (I think dE.) - I've decided to > give a chance. I've downloaded and compile x264 library from git (I have > several pain, because I have to get also latest yasm to compile it with > appropriate assembler, and also install Debian package autoconf). But the > result was prove this. I have using command: > > $ ffmpeg -v verbose -f video4linux2 -i /dev/video0 -r 29.97 -s 640x480 > -vcodec libx264 \ > -t 10 -f avi -y /tmp/test01.avi > > In result I have resonable good quality of the video stream :D > > Next I have tried to get sound. As it suggested in ffmpeg documentation I > have checked available options for alsa: > $ cat /proc/asound/cards > 0 [SB ]: HDA-Intel - HDA ATI SB > HDA ATI SB at 0xfe024000 irq 16 > 1 [HDMI ]: HDA-Intel - HDA ATI HDMI > HDA ATI HDMI at 0xfdffc000 irq 19 > 2 [Bt878 ]: Bt87x - Brooktree Bt878 > Brooktree Bt878 at 0xfdbfe000, irq 20 > > $ cat /proc/asound/devs > 2: : timer > 3: : sequencer > 4: [ 0- 2]: digital audio capture > 5: [ 0- 1]: digital audio playback > 6: [ 0- 1]: digital audio capture > 7: [ 0- 0]: digital audio playback > 8: [ 0- 0]: digital audio capture > 9: [ 0- 0]: hardware dependent > 10: [ 0] : control > 11: [ 1- 3]: digital audio playback > 12: [ 1- 0]: hardware dependent > 13: [ 1] : control > 14: [ 2- 1]: digital audio capture > 15: [ 2- 0]: digital audio capture > 16: [ 2] : control > > I have using command for grabbing: > > $ ffmpeg -v verbose -f alsa -ac 2 -i hw:0 -f video4linux2 -i /dev/video0 \ > -r 29.97 -s 640x480 -vcodec libx264 \ > -f avi -y test02.avi > > No sound :( I have tried hw:0, hw:1 and hw:2 as input, but nothing. > I have compile latest mp3lame from sourceforge. No result. But in video > stream, mp3 track is exist (I have checked on windows using mediainfo)!? > > I goes back to check v4l2 facilities. I have list available controls for > /dev/video0, using command v4l2-ctl (getting from Squeeze repository), > here is the result: > > brightness (int) : min=0 max=65535 step=256 default=32768 value=32768 > contrast (int) : min=0 max=65535 step=128 default=32768 value=32768 > saturation (int) : min=0 max=65535 step=128 default=32768 value=32768 > hue (int) : min=0 max=65535 step=256 default=32768 value=32768 > balance (int) : min=0 max=65535 step=655 default=32768 value=32768 > bass (int) : min=0 max=65535 step=655 default=32768 value=32768 > treble (int) : min=0 max=65535 step=655 default=32768 value=32768 > mute (bool) : default=0 value=1 > chroma_agc (bool) : default=0 value=0 > combfilter (bool) : default=0 value=0 > automute (bool) : default=0 value=1 > luma_decimation_filter (bool) : default=0 value=0 > agc_crush (bool) : default=0 value=1 > vcr_hack (bool) : default=0 value=0 > whitecrush_upper (int) : min=0 max=255 step=1 default=207 value=207 > whitecrush_lower (int) : min=0 max=255 step=1 default=127 value=127 > uv_ratio (int) : min=0 max=100 step=1 default=50 value=50 > full_luma_range (bool) : default=0 value=0 > coring (int) : min=0 max=3 step=1 default=0 value=0 > > Tuner card internal sound output attached to CD input on mainboard for > onboard audio. I've goes back to "-i hw:0" and start grabbing w/o time > limit, in other console I have set mute to '0' : > $ v4l2-ctl --set-ctrl=mute=0 > Sound from TV tuner is up, after several seconds, I have stopped grabbing, > when ffmpeg step out to shell sound is goes down - I have checked the sort > clipp - I've got sound :D > > Happy? Not yet. > I need NO sound on the card's host box - I suppose to use it as a stream > server. The best (for me) if I could get sound from tuner card directly, > avoid sound card. I have tried similar procedure using grabbing command: > > $ ffmpeg -v verbose -f alsa -ac 2 -i hw:2 -f video4linux2 -i /dev/video0 \ > -r 29.97 -s 640x480 -vcodec libx264 \ > -f avi -y test02.avi > > But no success. Is it possible to get sound avoid the sound card? Any > suggestion? > > Sincerely > tovis > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user I sort of don't understand your setup. You capture card connects via USB right? And there's an audio controller in the TV card, which alsa recognizes and makes the corresponding devices? From ffmpeg at csmiller.demon.co.uk Tue Mar 20 11:35:01 2012 From: ffmpeg at csmiller.demon.co.uk (ffmpeg at csmiller.demon.co.uk) Date: Tue, 20 Mar 2012 10:35:01 +0000 Subject: [FFmpeg-user] ffmpeg-0.5.5 on MinGW64 Message-ID: Hi, I'm trying to build ffmpeg-0.5.5 for MS Windows 7 64bit. I realise that it is an old version, but there are several API changes after 0.5.5 that will make it hard to integrate the latest version into my company's project. I downloaded MinGW32 from http://garr.dl.sourceforge.net/project/mingw/Installer/mingw-get-inst/mingw-get-inst-20111118/mingw-get-inst-20111118.exe and MingGW64 from http://switch.dl.sourceforge.net/project/mingw-w64/Toolchains%20targetting%20Win64/Automated%20Builds/mingw-w64-bin_i686-mingw_20111220.zip However, MingGW defines #define __MINGW32_MAJOR_VERSION 3 #define __MINGW32_MINOR_VERSION 20 and MinGW64 defines them as #define __MINGW32_MAJOR_VERSION 3 #define __MINGW32_MINOR_VERSION 11 According to its configure script ffmeg 0.5.5 for MinGW64 requires 3.15 or later. I asked yesterday on the MinGW64 list (mingw-w64-public AT lists.sourceforge.net) for advice; a user there said to increment the MinGW64 version number. I changed it to 3.20, and the w32api version from 3.12 to 3.13. However libavdevice\vfwcap.c has this code #include "libavformat/avformat.h" #include #include and the MinGW64 verison of vfw.h doesn't #define / typedef DWORD etc, which causes a compile failure. The configuration command was ./configure --enable-memalign-hack --cc=x86_64-w64-mingw32-gcc.exe Can anyone remember what version of MinGW64 was used to build ffmpeg-0.5.5? TIA, Colin S. Miller From cehoyos at ag.or.at Tue Mar 20 11:39:56 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 20 Mar 2012 10:39:56 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg-0.5.5 on MinGW64 References: Message-ID: csmiller.demon.co.uk> writes: > I'm trying to build ffmpeg-0.5.5 for MS Windows 7 64bit. > I realise that it is an old version, but there are several > API changes after 0.5.5 that will make it hard to integrate > the latest version into my company's project. Then use at least 0.7 or oldabi, they are only one year instead of three years old and API-compatible. Please note that the relevant version bump was several months ago, and we unfortunately do not have the manpower to support ancient versions of FFmpeg, you should better port your project! Carl Eugen From cehoyos at ag.or.at Tue Mar 20 11:42:57 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 20 Mar 2012 10:42:57 +0000 (UTC) Subject: [FFmpeg-user] how do i strip away all metadata from input source? References: Message-ID: Auguste Pop gmail.com> writes: > i was using -map_metadata -1 -map_metadata:s:0 -1 -map_metadata:s:1 > -1, but it does not work now. every metadata from the source is copied > over to the output file. Thank you for the report, -map_metadata -1 should now work with current git head. Please report if you still have problems, Carl Eugen From langdon at gmail.com Tue Mar 20 11:48:39 2012 From: langdon at gmail.com (Langdon) Date: Tue, 20 Mar 2012 06:48:39 -0400 Subject: [FFmpeg-user] forcing bitrate? In-Reply-To: References: Message-ID: is there a way to force the bitrate when extracting audio from a video? I tried -b:a 1536k, but ffmpeg will force it to 1536.2kbps, which seems to affect the duration of the sound clip I get back (seems to add silence to the beginning). After doing some research it appears that maybe the video wasn't clipped on a GOP boundary... if that's the case am I out of luck for any easy fix? http://ffmpeg-users.933282.n4.nabble.com/inaccurate-splitting-of-PCM-audio-td935314.html TIA From cehoyos at ag.or.at Tue Mar 20 13:36:41 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 20 Mar 2012 12:36:41 +0000 (UTC) Subject: [FFmpeg-user] forcing bitrate? References: Message-ID: Langdon gmail.com> writes: > is there a way to force the bitrate when extracting audio from a video? > > I tried -b:a 1536k, but ffmpeg will force it to 1536.2kbps, which seems to > affect the duration of the sound clip I get back (seems to add silence to > the beginning). Command line and complete, uncut console output missing. Carl Eugen From langdon at gmail.com Tue Mar 20 15:10:08 2012 From: langdon at gmail.com (Langdon) Date: Tue, 20 Mar 2012 10:10:08 -0400 Subject: [FFmpeg-user] forcing bitrate? In-Reply-To: References: Message-ID: My apologies, the command with output is below. >*ffmpeg -i source/e0b.mov -vn -b:a 1536k e0b.wav* ffmpeg version N-38622-g1eabd71 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 7 2012 00:21:47 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable -libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-lib schroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --ena ble-libxavs --enable-libxvid --enable-zlib libavutil 51. 42.100 / 51. 42.100 libavcodec 54. 10.100 / 54. 10.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 63.100 / 2. 63.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'source/e0b.mov': Metadata: creation_time : 2012-03-15 16:22:30 Duration: 00:00:01.66, start: 0.000000, bitrate: 85949 kb/s * Stream #0:0(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, 2 channels, s16, 1536 kb/s* Metadata: creation_time : 2012-03-15 16:22:30 handler_name : ?Apple Alias Data Handler Stream #0:1(eng): Video: prores (apch / 0x68637061), yuv422p10le, 1280x720, 83140 kb/s, 23.98 fps, 23.98 tbr, 23976 tbn, 23976 tbc Metadata: creation_time : 2012-03-15 16:22:30 handler_name : ?Apple Alias Data Handler Stream #0:2(eng): Data: none (tmcd / 0x64636D74) Metadata: creation_time : 2012-03-15 16:22:32 handler_name : ?Apple Alias Data Handler timecode : 01:00:18:01 Output #0, wav, to 'e0b.wav': Metadata: creation_time : 2012-03-15 16:22:30 encoder : Lavf54.2.100 * Stream #0:0(eng): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s* Metadata: creation_time : 2012-03-15 16:22:30 handler_name : ?Apple Alias Data Handler Stream mapping: Stream #0:0 -> #0:0 (pcm_s16be -> pcm_s16le) Press [q] to stop, [?] for help *size= 313kB time=00:00:01.66 bitrate=1536.2kbits/s* video:0kB audio:313kB global headers:0kB muxing overhead 0.014361% Is the "muxing overhead" bit causing it? I noticed when I do MP3, the bitrate changes to 320kbps (which I believe is the maximum for MP3), and the length of the clip noticeably changes to 1.07s (with a higher muxing overhead). Output #0, mp3, to 'e0b.mp3': Metadata: TDEN : 2012-03-15 16:22:30 TSSE : Lavf54.2.100 Stream #0:0(eng): Audio: mp3, 48000 Hz, 2 channels, s16, 1536 kb/s Metadata: creation_time : 2012-03-15 16:22:30 handler_name : ?Apple Alias Data Handler Stream mapping: Stream #0:0 -> #0:0 (pcm_s16be -> libmp3lame) Press [q] to stop, [?] for help *size= 68kB time=00:00:01.70 bitrate= 324.8kbits/s* video:0kB audio:67kB global headers:0kB muxing overhead 1.503815% Again, I'm wondering if this is because the video clip was made, that it wasn't cut on a GOP boundary (based on http://ffmpeg-users.933282.n4.nabble.com/inaccurate-splitting-of-PCM-audio-td935314.html ). I'm ultimately trying to take a sequence of clips (that will vary based on user input), and compile the variation into a single video. This is why I need the seams to be non-existent. Thanks! On Tue, Mar 20, 2012 at 8:36 AM, Carl Eugen Hoyos wrote: > Langdon gmail.com> writes: > > > is there a way to force the bitrate when extracting audio from a video? > > > > I tried -b:a 1536k, but ffmpeg will force it to 1536.2kbps, which seems > to > > affect the duration of the sound clip I get back (seems to add silence to > > the beginning). > > Command line and complete, uncut console output missing. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Tue Mar 20 15:15:09 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 20 Mar 2012 14:15:09 +0000 (UTC) Subject: [FFmpeg-user] forcing bitrate? References: Message-ID: Langdon gmail.com> writes: > *ffmpeg -i source/e0b.mov -vn -b:a 1536k e0b.wav* You cannot set the bitrate for pcm_s16le (which is the default audio codec for the wav file format), pcm is uncompressed and the bitrate is a property of the codec (not something you can set). Carl Eugen From langdon at gmail.com Tue Mar 20 15:25:19 2012 From: langdon at gmail.com (Langdon) Date: Tue, 20 Mar 2012 10:25:19 -0400 Subject: [FFmpeg-user] forcing bitrate? In-Reply-To: References: Message-ID: Do you know why it ends up at 1536.2? What's the significance of the .2? Is it possible to disable the muxing overhead, if that's the cause? Is there another audio codec+format I can use that (a) is able to be concatenated (I'm using the cat command per the ffmpeg FAQ for joining video), and (b) won't add any silence before/after the clip? Thanks for the quick reponses, Langdon On Tue, Mar 20, 2012 at 10:15 AM, Carl Eugen Hoyos wrote: > Langdon gmail.com> writes: > > > *ffmpeg -i source/e0b.mov -vn -b:a 1536k e0b.wav* > > You cannot set the bitrate for pcm_s16le (which is the default > audio codec for the wav file format), pcm is uncompressed and > the bitrate is a property of the codec (not something you can set). > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Tue Mar 20 15:44:27 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 20 Mar 2012 14:44:27 +0000 (UTC) Subject: [FFmpeg-user] forcing bitrate? References: Message-ID: Langdon gmail.com> writes: > Do you know why it ends up at 1536.2? What's the significance of the .2? > Is it possible to disable the muxing overhead, if that's the cause? > > Is there another audio codec+format I can use that (a) is able to be > concatenated I don't think wav can be concatenated. Perhaps you search for -f s16le ? > (I'm using the cat command per the ffmpeg FAQ for joining > video), and (b) won't add any silence before/after the clip? Please do not top-post, it is considered rude here, Carl Eugen From vrparekh at gmail.com Tue Mar 20 15:49:32 2012 From: vrparekh at gmail.com (vrparekh at gmail.com) Date: Tue, 20 Mar 2012 07:49:32 -0700 (PDT) Subject: [FFmpeg-user] aspect ratio is changed with codec copy and sameq Message-ID: <1332254972184-4489075.post@n4.nabble.com> Hello, i want to extract clip from one mp4 video, i tried with "-vcodec copy -acodec copy" and "-sameq" in both, aspect ratio of generated file is changed. source file is of aspect ratio sar=4:3 dar=4:3 new file is has aspect ratio sar=4:3 dar=1:1 please help me to solve this problem, thanks vishal parekh -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/aspect-ratio-is-changed-with-codec-copy-and-sameq-tp4489075p4489075.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From sjames at susanjamescompany.com Tue Mar 20 16:24:30 2012 From: sjames at susanjamescompany.com (Susan James) Date: Tue, 20 Mar 2012 08:24:30 -0700 Subject: [FFmpeg-user] ffmpeg processing in batch Message-ID: <4F68A12E.8090203@susanjamescompany.com> Hi All, I'm encoding videos with a batch script and my script will only process one file at a time even though I make the call to process all files listed in a file.list My command pipeline is below for both MAC and Linux. My batch script is attached. I can't figure out why ffmpeg refuses to process multiple files in an automated script. Can someone help me with this? Are my command pipelines below wanting an additional parameter to process in batch? I log to a logs.txt but the only output I get is for the one file that ffmpeg processes. I've also attached the logs.txt which includes output from both the MAC and LINUX processing. Any help is much appreciated. I've been working at this for days and days with no resolution for processing in batch. thanks! Susan ============ ffmpeg commands ================ MAC ffmpeg -i "file1.HiRes" -ab 320k -vol 1024 -strict experimental -threads 5 -vcodec libx264 -b:v 2000k -pix_fmt yuv420p "file1-as.mp4" LINUX ffmpeg -i "file1.HiRes" -ab 320k -vol 1024 -vcodec libx264 -b:v 2000k -threads 5 -pix_fmt yuv420p "file1-as.mp4" ============================================== -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: resume-08032012.sh.txt URL: -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: logs-mac-linux.txt URL: From langdon at gmail.com Tue Mar 20 16:35:27 2012 From: langdon at gmail.com (Langdon) Date: Tue, 20 Mar 2012 11:35:27 -0400 Subject: [FFmpeg-user] forcing bitrate? In-Reply-To: References: Message-ID: > Please do not top-post, it is considered rude here, Carl Eugen Ah, my apologies, Gmail has made me lazy about my replies. > I don't think wav can be concatenated. > Perhaps you search for -f s16le ? When I do "-f s16le", I do get an even 1536.0 kbps, but there's still silence at the beginning and end of the clip. Am I to assume that is just the nature of the video clip? I'm expecting that the editor who I got this clip from, should have added no pre/post-silence. Is it normal for muxing overhead to increase the length of the audio clip (from what I've read, I gather it does increase filesize)? From cehoyos at ag.or.at Tue Mar 20 16:51:05 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 20 Mar 2012 15:51:05 +0000 (UTC) Subject: [FFmpeg-user] aspect ratio is changed with codec copy and sameq References: <1332254972184-4489075.post@n4.nabble.com> Message-ID: vrparekh gmail.com gmail.com> writes: > i want to extract clip from one mp4 video, > > i tried with "-vcodec copy -acodec copy" and "-sameq" > > in both, aspect ratio of generated file is changed. Please provide command line together with complete, uncut console output. Carl Eugen From cehoyos at ag.or.at Tue Mar 20 16:59:00 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 20 Mar 2012 15:59:00 +0000 (UTC) Subject: [FFmpeg-user] forcing bitrate? References: Message-ID: Langdon gmail.com> writes: > When I do "-f s16le", I do get an even 1536.0 kbps, but there's still > silence at the beginning and end of the clip. Am I to assume that is just > the nature of the video clip? If not, this sounds like a bug (that at least needs a sample file). Carl Eugen From vrparekh at gmail.com Tue Mar 20 17:59:23 2012 From: vrparekh at gmail.com (vrparekh at gmail.com) Date: Tue, 20 Mar 2012 09:59:23 -0700 (PDT) Subject: [FFmpeg-user] aspect ratio is changed with codec copy and sameq In-Reply-To: References: <1332254972184-4489075.post@n4.nabble.com> Message-ID: <1332262763259-4489550.post@n4.nabble.com> Thanks Carl, one weird thing is when i see details in another video tool, it shows me sar=4:3 dar=4:3 of original video but when i use command ffpeg -i sourcefile, it shows me sar=300:400 dar=1:1 below is command line output of codec copy ffmpeg.exe -ss 5 -t 150 -i "d:\source.mp4" -acodec copy -vcodec copy "d:\output.mp4" ffmpeg version N-38938-ge01f478 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 19 2012 23:18:25 with gcc 4.6.2 configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzli b --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable -libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-l ibrtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-li bvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-l ibx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 42.100 / 51. 42.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 65.101 / 2. 65.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'd:\source.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42isom creation_time : 2012-01-26 21:00:03 Duration: 00:05:00.32, start: 0.000000, bitrate: 539 kb/s Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 400x300 [ SAR 300:400 DAR 1:1], 469 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 60 tbc Metadata: creation_time : 2012-01-26 21:00:03 handler_name : Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 32000 Hz, stereo, s16, 64 kb/s Metadata: creation_time : 2012-01-26 21:00:03 handler_name : strptime() unavailable on this system, cannot convert the date string. Output #0, mp4, to 'd:\output.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42isom creation_time : 2012-01-26 21:00:03 encoder : Lavf54.2.100 Stream #0:0(eng): Video: h264 (![0][0][0] / 0x0021), yuv420p, 400x300 [SAR 3 00:400 DAR 1:1], q=2-31, 469 kb/s, 29.97 fps, 90k tbn, 90k tbc Metadata: creation_time : 2012-01-26 21:00:03 handler_name : Stream #0:1(eng): Audio: aac (@[0][0][0] / 0x0040), 32000 Hz, stereo, 64 kb/ s Metadata: creation_time : 2012-01-26 21:00:03 handler_name : Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help frame= 9001 fps=0.0 q=-1.0 Lsize= 19815kB time=00:05:00.29 bitrate= 540.6kbits /s video:17212kB audio:2350kB global headers:0kB muxing overhead 1.294060% -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/aspect-ratio-is-changed-with-codec-copy-and-sameq-tp4489075p4489550.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From cehoyos at ag.or.at Tue Mar 20 18:50:18 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 20 Mar 2012 17:50:18 +0000 (UTC) Subject: [FFmpeg-user] aspect ratio is changed with codec copy and sameq References: <1332254972184-4489075.post@n4.nabble.com> <1332262763259-4489550.post@n4.nabble.com> Message-ID: vrparekh gmail.com gmail.com> writes: > one weird thing is when i see details in another video tool, it shows me > sar=4:3 > dar=4:3 > of original video > > but when i use command ffpeg -i sourcefile, it shows me > sar=300:400 > dar=1:1 If the input file is decoded incorrectly, please provide a sample. > > below is command line output of codec copy > > ffmpeg.exe -ss 5 -t 150 -i "d:\source.mp4" -acodec > copy -vcodec copy "d:\output.mp4" What does ffmpeg -i output.mp4 show? Carl Eugen From ffmpeg at csmiller.demon.co.uk Tue Mar 20 21:30:28 2012 From: ffmpeg at csmiller.demon.co.uk (Colin S. Miller) Date: Tue, 20 Mar 2012 20:30:28 +0000 Subject: [FFmpeg-user] ffmpeg-0.5.5 on MinGW64 In-Reply-To: References: Message-ID: <4F68E8E4.2050701@csmiller.demon.co.uk> On 20/03/12 10:39, Carl Eugen Hoyos wrote: > csmiller.demon.co.uk> writes: > >> I'm trying to build ffmpeg-0.5.5 for MS Windows 7 64bit. >> I realise that it is an old version, but there are several >> API changes after 0.5.5 that will make it hard to integrate >> the latest version into my company's project. > > Then use at least 0.7 or oldabi, they are only one year instead > of three years old and API-compatible. > > Please note that the relevant version bump was several months ago, > and we unfortunately do not have the manpower to support ancient > versions of FFmpeg, you should better port your project! > Carl, Is the version of the build environment used to test each release recorded? If so, I can download that version of MinGW64. The project uses libavcodec, and the functions img_convert, avcodec_decocde_video and avcodec_decode_audio2. I will look at the practicalities of upgrading to a later libavcodec. Colin S. Miller From joddo at jerfu.com Tue Mar 20 22:10:12 2012 From: joddo at jerfu.com (Jeremy Oddo) Date: Tue, 20 Mar 2012 14:10:12 -0700 Subject: [FFmpeg-user] ProRes (LT) Quicktimes: Good for Windows. Not so good for Mac. Message-ID: Has anyone else run into this issue: We generate some ProRes (LT) Quicktimes on a Windows box. We play the QT and the player says that it's a ProRes (LT). That's good. We play the SAME quicktime on a Mac and it DOES NOT claim to be an (LT). The quicktime needs to be re-rendered in Final Cut to make it a true ProRes (LT). That's not good. Is this a known issue? Is there a work-a-round? From andreas.gumm at gmx.de Tue Mar 20 22:22:01 2012 From: andreas.gumm at gmx.de (Andreas Gumm) Date: Tue, 20 Mar 2012 22:22:01 +0100 Subject: [FFmpeg-user] ProRes (LT) Quicktimes: Good for Windows. Not so good for Mac. In-Reply-To: References: Message-ID: <4F68F4F9.3040504@gmx.de> Yes, this is a known issue! Your files has been correct encoded! FCP just complains sometimes about external encoded ProRes footage! It seem to be that some missing header informations or other stupid missing informations let FCP think the file looks not compatible! Stupid, but real! Since FFMBC is more broadcast orientated try to encode with FFMBC fork! Maybe this has been solved there in the actual release! Andreas Am 20.03.2012 22:10, schrieb Jeremy Oddo: > Has anyone else run into this issue: > > We generate some ProRes (LT) Quicktimes on a Windows box. We play the > QT and the player says that it's a ProRes (LT). That's good. > > We play the SAME quicktime on a Mac and it DOES NOT claim to be an > (LT). The quicktime needs to be re-rendered in Final Cut to make it a > true ProRes (LT). That's not good. > > Is this a known issue? Is there a work-a-round? > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From dev at rarevision.com Tue Mar 20 22:46:45 2012 From: dev at rarevision.com (Thomas Worth) Date: Tue, 20 Mar 2012 14:46:45 -0700 Subject: [FFmpeg-user] ProRes (LT) Quicktimes: Good for Windows. Not so good for Mac. In-Reply-To: <4F68F4F9.3040504@gmx.de> References: <4F68F4F9.3040504@gmx.de> Message-ID: On Tue, Mar 20, 2012 at 2:22 PM, Andreas Gumm wrote: > Yes, this is a known issue! Your files has been correct encoded! FCP just > complains sometimes about external encoded ProRes footage! > It seem to be that some missing header informations or other stupid missing > informations let FCP think the file looks not compatible! About this, I think the Final Cut media warning may be related to the A/V interleaving, i.e. the way the chunk offset tables are written. I don't think it has as much to do with information in the QuickTime header. Files written without audio don't show the warning. Baptiste Coudurier might comment on this. If anyone knows why this is happening, is would be him. Also it should be noted that the media performance warning does not occur when QuickTime files are written with a co64 atom. So, Final Cut apparently likes FFmpeg MOV files written with 64 bit offsets. This is just more evidence to support the theory that offset tables for both video and audio that don't play well with each other may be responsible for triggering the media warning. This should be fixed because if the A/V offsets aren't occurring in contiguous order and/or aligned, it is potentially hindering disk performance. Again, this is only speculation but I am pretty sure this is where to look for the problem. Someone might take a gander at movenc.c and see if interleaving is done differently with 64 bit MOV offsets vs. 32 bit ones. From andreas.gumm at gmx.de Tue Mar 20 22:52:52 2012 From: andreas.gumm at gmx.de (Andreas Gumm) Date: Tue, 20 Mar 2012 22:52:52 +0100 Subject: [FFmpeg-user] ProRes (LT) Quicktimes: Good for Windows. Not so good for Mac. In-Reply-To: References: <4F68F4F9.3040504@gmx.de> Message-ID: <4F68FC34.4070401@gmx.de> Interesting! Your sounds really conclusive. Andreas Am 20.03.2012 22:46, schrieb Thomas Worth: > On Tue, Mar 20, 2012 at 2:22 PM, Andreas Gumm wrote: >> Yes, this is a known issue! Your files has been correct encoded! FCP just >> complains sometimes about external encoded ProRes footage! >> It seem to be that some missing header informations or other stupid missing >> informations let FCP think the file looks not compatible! > About this, I think the Final Cut media warning may be related to the > A/V interleaving, i.e. the way the chunk offset tables are written. I > don't think it has as much to do with information in the QuickTime > header. Files written without audio don't show the warning. Baptiste > Coudurier might comment on this. If anyone knows why this is > happening, is would be him. > > Also it should be noted that the media performance warning does not > occur when QuickTime files are written with a co64 atom. So, Final Cut > apparently likes FFmpeg MOV files written with 64 bit offsets. This is > just more evidence to support the theory that offset tables for both > video and audio that don't play well with each other may be > responsible for triggering the media warning. This should be fixed > because if the A/V offsets aren't occurring in contiguous order and/or > aligned, it is potentially hindering disk performance. Again, this is > only speculation but I am pretty sure this is where to look for the > problem. > > Someone might take a gander at movenc.c and see if interleaving is > done differently with 64 bit MOV offsets vs. 32 bit ones. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From maxim.levkov at gmail.com Tue Mar 20 23:09:02 2012 From: maxim.levkov at gmail.com (Maxim Levkov) Date: Tue, 20 Mar 2012 15:09:02 -0700 Subject: [FFmpeg-user] ffmpeg processing in batch In-Reply-To: <4F68A12E.8090203@susanjamescompany.com> References: <4F68A12E.8090203@susanjamescompany.com> Message-ID: Hi Susan, Without reading your script in detail, the one thing I can think of is that you probably need to instantiate one instance per processing file and not all files by one instance of ffmpeg. Have you tried pushing single instance of FFMPEG against single transcoding process? Hence, if you launch multiple instances of FFMPEG you would end up processing multiple files, just make sure 1:1 relationship (e.g. one executed instance of FFMPEG for one processed output). Once I will go over your script, I might think of something else. Regards, Maxim Levkov On Tue, Mar 20, 2012 at 8:24 AM, Susan James wrote: > Hi All, > > I'm encoding videos with a batch script and my script will only process > one file at a time even though I make the call to process all files listed > in a file.list > My command pipeline is below for both MAC and Linux. > My batch script is attached. > > I can't figure out why ffmpeg refuses to process multiple files in an > automated script. Can someone help me with this? Are my command pipelines > below wanting an additional parameter to process in batch? > > I log to a logs.txt but the only output I get is for the one file that > ffmpeg processes. I've also attached the logs.txt which includes output > from both the MAC and LINUX processing. > > Any help is much appreciated. I've been working at this for days and days > with no resolution for processing in batch. > > thanks! > Susan > > > > ============ ffmpeg commands ================ > MAC > ffmpeg -i "file1.HiRes" -ab 320k -vol 1024 -strict experimental -threads 5 > -vcodec libx264 -b:v 2000k -pix_fmt yuv420p "file1-as.mp4" > > > LINUX > ffmpeg -i "file1.HiRes" -ab 320k -vol 1024 -vcodec libx264 -b:v 2000k > -threads 5 -pix_fmt yuv420p "file1-as.mp4" > ==============================**================ > > #! /bin/bash > > # > # Arguments: > # "SourceFileDir" "TargetDir" "file.list" countToProcess > # > > function usage () > { > cat << __END_USAGE__ > Usage: > $0 "SourceFileDir" "TargetDir" "file.list" countToProcess > > Example: > > $0 "/source/ChandiHiResClasses" "/target" "file.list" 2 > __END_USAGE__ > > exit 123 > } > > [ $# == 4 ] || usage > > > srcFileDirName=`basename "$1"` > targFileDir="$2/${srcFileDirName}-as" > fileList="$3" > > # How many files to process in each invocation > declare -i count=$4 > > # Create the target directory if it does not exist. > > echo "" > echo -n "Checking for existence of target directory :: ${targFileDir} ... " > if [ ! -d "${targFileDir}" ] > then > echo "Creating it." > mkdir -p "${targFileDir}" > else > echo "It Exists." > fi > > cat << __EOM__ > > Will process maximum ${count} files in this invocation, lesser if file > list in > "${fileList}" contains less than ${count} entries. > > __EOM__ > > function updateFiles () > { > echo "" > echo "Processed \"$1\"." > echo "Updating \"${fileList}\" and \"${fileList}.done\"." > echo "" > > # Append the names of processed file to the list of already > processed files > # kept in "${fileList}.done". > echo "$1" >> "${fileList}.done" > > # Remove the processed files from list of files kept in > "${fileList}" > tempFile=`mktemp tmp.XXXXXXXXXX` > grep -v "^$1" "${fileList}" > "${tempFile}" > mv "${tempFile}" "${fileList}" > } > > # Make a copy of file containing list of file names to be processed. As we > will > # be updating the original file after each successful run of invoked ffmpeg > # command, we will use this copy in while loop below. > > listCopy=`mktemp tmp.XXXXXXXXXX` > cp "${fileList}" "${listCopy}" > > declare -i i=1 > while read fileName > do > if [ $i -le $count ] > then > echo "${i}. Starting with processing of \"${fileName}\" ..." > > # Add similar lines to remove other extensions if needed. > targetFileName=${fileName/%.mov} > > targetFileName="${targetFileName}-as.mp4" > > echo "Target filename : \"${targetFileName}\"" > echo "" > echo "Running command ..." > echo "ffmpeg -i \"$1/$fileName\" -ab 320k -vol 1024 -vcodec > libx264 -b:v 2000k -pix_fmt yuv420p \"${targFileDir}/${targetFileName}\" && > updateFiles \"${fileName}\"" > > # Uncomment following command, above echo line is for > verbose output. > > ffmpeg -i "$1/$fileName" -ab 320k -vol 1024 -vcodec libx264 > -b:v 2000k -pix_fmt yuv420p "${targFileDir}/${targetFileName}" && > updateFiles "${fileName}" > > echo "Completed processing of \"${fileName}\"." > echo "" > i=$[$i+1] # Processed one more file. > else > break > fi > done < "${listCopy}" > > # Remove the temporary copy of file list. > rm ${listCopy} > > echo "" > echo "Exiting this invocation of script." > > MAC > > cat /tmp/logs.txt > > Checking for existence of target directory :: > /Volumes/My_Book_2//From_Birth_to_Death.HiRes-as ... It Exists. > > Will process maximum 5 files in this invocation, lesser if file list in > "file.list" contains less than 5 entries. > > 1. Starting with processing of "From_Birth_to_Death_14.HiRes" ... > Target filename : "From_Birth_to_Death_14.HiRes-as.mp4" > > Running command ... > ffmpeg -i > "/Volumes/LaCie_Disk_2TB/From_Birth_to_Death.HiRes/From_Birth_to_Death_14.HiRes" > -ab 320k -vol 1024 -vcodec libx264 -b:v 2000k -pix_fmt yuv420p > "/Volumes/My_Book_2//From_Birth_to_Death.HiRes-as/From_Birth_to_Death_14.HiRes-as.mp4" > && updateFiles "From_Birth_to_Death_14.HiRes" > > > Linux > > cat /tmp/logs.txt > > Checking for existence of target directory :: > /source2//From_Birth_to_Death.HiRes-as ... It Exists. > > Will process maximum 2 files in this invocation, lesser if file list in > "file.list" contains less than 2 entries. > > 1. Starting with processing of "From_Birth_to_Death_12.HiRes" ... > Target filename : "From_Birth_to_Death_12.HiRes-as.mp4" > > Running command ... > ffmpeg -i > "/source/From_Birth_to_Death.HiRes//From_Birth_to_Death_12.HiRes" -ab 320k > -vol 1024 -vcodec libx264 -b:v 2000k -pix_fmt yuv420p > "/source2//From_Birth_to_Death.HiRes-as/From_Birth_to_Death_12.HiRes-as.mp4" > && updateFiles "From_Birth_to_Death_12.HiRes" > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From stefasab at gmail.com Wed Mar 21 00:16:02 2012 From: stefasab at gmail.com (Stefano Sabatini) Date: Wed, 21 Mar 2012 00:16:02 +0100 Subject: [FFmpeg-user] ffmpeg extract metadata to text or php file ? In-Reply-To: <1331827192659-4475468.post@n4.nabble.com> References: <1331827192659-4475468.post@n4.nabble.com> Message-ID: <20120320231602.GI27056@arborea> On date Thursday 2012-03-15 08:59:52 -0700, lf2 encoded: > ffmpeg extract metadata to text or php file ? > > how can i do this ffprobe will show you that. You can then parse the output for getting the required info (hint: JSON and XML output is supported). -- ffmpeg-user random tip #17 A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing in e-mail? From chris at osric.com Wed Mar 21 04:29:44 2012 From: chris at osric.com (Chris Herdt) Date: Tue, 20 Mar 2012 23:29:44 -0400 Subject: [FFmpeg-user] Using ffmpeg to excerpt and concatenate video segments Message-ID: I am trying to use ffmpeg to slice & splice video files (for example, from an 1800 second video I would like to create a video that contains just seconds 300-360, 600-660, and 900-960. The videos I am working with are raw h.264 videos from a Night Owl security system. The frame rate of the original videos is 7 fps. I am able to convert segments of the video to an mp4: ffmpeg -f h264 -r:v 7 -i infile.264 -qscale:v 1 -vsync cfr -ss 300 -t 60 outfile300-360.mp4 However, when I attempt to use one of the formats that the FAQ suggests can be concatenated (e.g. MPEG-2, DV), it tells me: [mpeg1video @ 0x101033200] MPEG1/2 does not support 5/1 fps If I remove the video frame rate option, ffmpeg creates the segment, but playback is too fast and it contains a different portion of the video than what I'm after: ffmpeg -f h264 -i infile.264 -qscale:v 1 -vsync cfr -ss 300 -t 60 outfile300-360.mpeg I'm wondering if anyone has suggestions as to how to accomplish slicing and splicing the video while maintaining the original frame rate/playback speed? Thanks, Chris -- Chris Herdt Web Applications Developer 267-603-1066 (home) 734-754-3585 (mobile) http://osric.com/chris/ From roko98 at yahoo.com Wed Mar 21 06:33:01 2012 From: roko98 at yahoo.com (roko) Date: Tue, 20 Mar 2012 22:33:01 -0700 (PDT) Subject: [FFmpeg-user] MPEG-TS trouble Message-ID: <1332307981.76400.YahooMailNeo@web162003.mail.bf1.yahoo.com> Hi all I'm doing some testing with and IP camera.? If I do this: ffmpeg -i rtsp://admin:admin at 192.168.99.146/11 -vcodec copy -y -r 25 algo.mp4 ... Input #0, rtsp, from 'rtsp://admin:admin at 192.168.99.146/11':??????????????????? ? Metadata: ??? title?????????? : \11 ? Duration: N/A, start: 0.039956, bitrate: N/A ??? Stream #0:0: Video: h264 (Constrained Baseline), yuv420p, 640x480, 25 tbr, 90k tbn, 180k tbc Output #0, mp4, to 'algo.mp4': ? Metadata: ??? title?????????? : \11 ??? encoder???????? : Lavf53.31.100 ??? Stream #0:0: Video: h264 (![0][0][0] / 0x0021), yuv420p, 640x480, q=2-31, 90k tbn, 90k tbc Stream mapping: ? Stream #0:0 -> #0:0 (copy) ... It works fine, I can play the file anywhere.? But, if I do this: ffmpeg -i rtsp://admin:admin at 192.168.99.146/11 -vcodec copy -y -r 25 algo.ts I get an error: [mpegts @ 0x666660] H.264 bitstream malformed, no startcode found, use the h264_mp4toannexb bitstream filter av_interleaved_write_frame(): Invalid data found when processing input If I include the filter, like this: fmpeg -i rtsp://admin:admin at 192.168.99.146/11 -vcodec copy -y -r 25 -vbsf h264_mp4toannexb algo.ts The error changes to: Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument [mpegts @ 0x666660] H.264 bitstream malformed, no startcode found, use the h264_mp4toannexb bitstream filter av_interleaved_write_frame(): Invalid data found when processing input Pls forgive me if I'm missing something very basic here.? I want to save the file to MPEG-TS without any transcoding.? ??? Thx. From vrparekh at gmail.com Wed Mar 21 06:23:34 2012 From: vrparekh at gmail.com (vrparekh at gmail.com) Date: Tue, 20 Mar 2012 22:23:34 -0700 (PDT) Subject: [FFmpeg-user] aspect ratio is changed with codec copy and sameq In-Reply-To: References: <1332254972184-4489075.post@n4.nabble.com> <1332262763259-4489550.post@n4.nabble.com> Message-ID: <1332307414992-4491324.post@n4.nabble.com> > ffmpeg.exe -i d:\output.mp4 ffmpeg version N-38938-ge01f478 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 19 2012 23:18:25 with gcc 4.6.2 configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzli b --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable -libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-l ibrtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-li bvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-l ibx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 42.100 / 51. 42.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 65.101 / 2. 65.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'd:\output.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf54.2.100 Duration: 00:05:00.32, start: 0.000000, bitrate: 540 kb/s Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 400x300 [ SAR 300:400 DAR 1:1], 469 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 60 tbc Metadata: handler_name : VideoHandler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 32000 Hz, stereo, s16, 64 kb/s Metadata: handler_name : At least one output file must be specified -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/aspect-ratio-is-changed-with-codec-copy-and-sameq-tp4489075p4491324.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From cehoyos at ag.or.at Wed Mar 21 08:13:47 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 21 Mar 2012 07:13:47 +0000 (UTC) Subject: [FFmpeg-user] MPEG-TS trouble References: <1332307981.76400.YahooMailNeo@web162003.mail.bf1.yahoo.com> Message-ID: roko yahoo.com> writes: > ffmpeg -i rtsp://admin:admin 192.168.99.146/11 -vcodec copy > -y -r 25 algo.mp4 > ... Always post complete, uncut console output. [...] > Pls forgive me if I'm missing something very basic here.? I want > to save the file to MPEG-TS without any transcoding. No transcoding takes place with above command line. Carl Eugen From cehoyos at ag.or.at Wed Mar 21 08:15:36 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 21 Mar 2012 07:15:36 +0000 (UTC) Subject: [FFmpeg-user] aspect ratio is changed with codec copy and sameq References: <1332254972184-4489075.post@n4.nabble.com> <1332262763259-4489550.post@n4.nabble.com> <1332307414992-4491324.post@n4.nabble.com> Message-ID: vrparekh gmail.com gmail.com> writes: > Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, > 400x300 [SAR 300:400 DAR 1:1] This is exactly the same as your input file shows. Where is the problem? Carl Eugen From oussama.stiti at gmail.com Wed Mar 21 09:05:41 2012 From: oussama.stiti at gmail.com (Oussama Stiti) Date: Wed, 21 Mar 2012 17:05:41 +0900 Subject: [FFmpeg-user] Switch "on" Reference Picture Selection (RPS) in the libavcodec/ituh263enc.c Source File Message-ID: Hello, An H.264 error resilience tool called Reference Picture Selection (RPS) is able to automatically identify and inform the encoder (server) which Group of Blocks (slices) are damaged. These decoding error notifications could refer to MacroBlocks or Slices. Various feedback error signalling modes have been also defined (ACK, NACK, ...). I just want to activate this tool on my ffmpeg source code, and i think that it's a modification in the file $HOME/ffmpeg/libavcodec/ituh263enc.c . Does anyone have any information about that ? How to activate it ? I need notifications of damaged frames when i receive a H264 bitstream. Here is, few lines of code in ituh263enc.c: put_bits(&s->pb,1, s->custom_pcf); put_bits(&s->pb,1, s->umvplus); /* Unrestricted Motion Vector */ put_bits(&s->pb,1,0); /* SAC: off */ put_bits(&s->pb,1,s->obmc); /* Advanced Prediction Mode */ put_bits(&s->pb,1,s->h263_aic); /* Advanced Intra Coding */ put_bits(&s->pb,1,s->loop_filter); /* Deblocking Filter */ put_bits(&s->pb,1,s->h263_slice_structured); /* Slice Structured */ *put_bits(&s->pb,1,0); /* Reference Picture Selection: off */* put_bits(&s->pb,1,0); /* Independent Segment Decoding: off */ put_bits(&s->pb,1,s->alt_inter_vlc); /* Alternative Inter VLC */ put_bits(&s->pb,1,s->modified_quant); /* Modified Quantization: */ put_bits(&s->pb,1,1); /* "1" to prevent start code emulation */ put_bits(&s->pb,3,0); /* Reserved */ Thank you. Regards -- *O. Stiti* From cehoyos at ag.or.at Wed Mar 21 09:44:34 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 21 Mar 2012 08:44:34 +0000 (UTC) Subject: [FFmpeg-user] Switch "on" Reference Picture Selection (RPS) in the libavcodec/ituh263enc.c Source File References: Message-ID: Oussama Stiti gmail.com> writes: > I need notifications of damaged frames when i receive > a H264 bitstream. I suspect you will have to edit the files that contain the H264 decoder to get those "notifications", see libavcodec/h264* What surprises me is that ffmpeg always outputs all kinds of messages whenever I feed a damaged H264 stream into the decoder: Do you have a damaged sample that produces no such messages? Carl Eugen From rodney.baker at iinet.net.au Wed Mar 21 10:01:51 2012 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Wed, 21 Mar 2012 19:31:51 +1030 Subject: [FFmpeg-user] aspect ratio is changed with codec copy and sameq In-Reply-To: References: <1332254972184-4489075.post@n4.nabble.com> <1332307414992-4491324.post@n4.nabble.com> Message-ID: <201203211931.52053.rodney.baker@iinet.net.au> On Wed, 21 Mar 2012 17:45:36 Carl Eugen Hoyos wrote: > vrparekh gmail.com gmail.com> writes: > > Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, > > > > 400x300 [SAR 300:400 DAR 1:1] > > This is exactly the same as your input file shows. > Where is the problem? > > Carl Eugen > Carl, I think the key is in this quote from a previous mail in this thread: >one weird thing is when i see details in another video tool, it shows me >sar=4:3 >dar=4:3 >of original video >but when i use command ffpeg -i sourcefile, it shows me >sar=300:400 >dar=1:1 In other words, ffmpeg appears to be detecting the SAR and DAR differently to "another video tool" and creating the output file the same as the alledgedly incorrectly detected input file aspect ratio (thus incorrectly altering the aspect ratio to something different from the source file. At least, that is how I read it. -- =================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au =================================================== From cehoyos at ag.or.at Wed Mar 21 10:05:34 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 21 Mar 2012 09:05:34 +0000 (UTC) Subject: [FFmpeg-user] aspect ratio is changed with codec copy and sameq References: <1332254972184-4489075.post@n4.nabble.com> <1332307414992-4491324.post@n4.nabble.com> <201203211931.52053.rodney.baker@iinet.net.au> Message-ID: Rodney Baker iinet.net.au> writes: > >one weird thing is when i see details in another video tool, it shows me > >sar=4:3 > >dar=4:3 > >of original video > > >but when i use command ffpeg -i sourcefile, it shows me > >sar=300:400 > >dar=1:1 > > In other words, ffmpeg appears to be detecting the SAR and DAR > differently to "another video tool" My answer was: "If the input file is decoded incorrectly, please provide a sample." I assumed the original report was about FFmpeg destroying some information when remuxing the stream. That some information is possibly read incorrectly sounds different to me. Thank you for your help, Carl Eugen From ml at access-dev.com Wed Mar 21 10:14:35 2012 From: ml at access-dev.com (Access-Dev) Date: Wed, 21 Mar 2012 10:14:35 +0100 Subject: [FFmpeg-user] [flv @ 0x6378a0] Error, Invalid timestamp=0, last=0 Message-ID: <4F699BFB.3080500@access-dev.com> Hello, I have trouble encoding a 3gp (iphone) video /usr/bin/ffmpeg -i "/home/mypath/uploads/407.3gp" -acodec libfaac -ar 44100 -ab 160k -s 488x286 -b 750k -y -f flv -r 24 -async 1 "/home/mypath/videos/407.flv" ffmpeg version N-38475-gc266eb1 Copyright (c) 2000-2012 the FFmpeg developers/da built on Mar 6 2012 14:43:12 with gcc 4.4.5 configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-shared libavutil 51. 41.100 / 51. 41.100 libavcodec 54. 6.100 / 54. 6.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 62.101 / 2. 62.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/home/mypath/uploads/407.3gp': Metadata: major_brand : 3gp4 minor_version : 0 compatible_brands: 3gp4emp creation_time : 2012-03-19 21:58:26 Duration: 00:01:40.48, start: 0.000000, bitrate: 493 kb/s Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 16000 Hz, mono, s16, 31 kb/s Metadata: creation_time : 2012-03-19 21:58:26 handler_name : Stream #0:1(und): Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D), yuv420p, 320x240 [SAR 1:1 DAR 4:3], 460 kb/s, 7.52 fps, 30 tbr, 1k tbn, 30 tbc Metadata: creation_time : 2012-03-19 21:58:26 handler_name : Please use -b:a or -b:v, -b is ambiguous [buffer @ 0x6489c0] w:320 h:240 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param: [scale @ 0x625ce0] w:320 h:240 fmt:yuv420p -> w:488 h:286 fmt:yuv420p flags:0x4 Output #0, flv, to '/home/mypath/medias/videos/407.flv': Metadata: major_brand : 3gp4 minor_version : 0 compatible_brands: 3gp4emp creation_time : 2012-03-19 21:58:26 encoder : Lavf54.2.100 Stream #0:0(und): Video: flv1 ([2][0][0][0] / 0x0002), yuv420p, 488x286 [SAR 143:183 DAR 4:3], q=2-31, 750 kb/s, 1k tbn, 24 tbc Metadata: creation_time : 2012-03-19 21:58:26 handler_name : Stream #0:1(und): Audio: aac ([10][0][0][0] / 0x000A), 44100 Hz, mono, s16, 160 kb/s Metadata: creation_time : 2012-03-19 21:58:26 handler_name : Stream mapping: Stream #0:1 -> #0:0 (mpeg4 -> flv) Stream #0:0 -> #0:1 (aac -> libfaac) Press [q] to stop, [?] for help [flv @ 0x6378a0] Error, Invalid timestamp=0, last=0 Video encoding failed Any suggestions ? Nicolas From oussama.stiti at gmail.com Wed Mar 21 10:44:44 2012 From: oussama.stiti at gmail.com (Oussama Stiti) Date: Wed, 21 Mar 2012 18:44:44 +0900 Subject: [FFmpeg-user] Switch "on" Reference Picture Selection (RPS) in the libavcodec/ituh263enc.c Source File In-Reply-To: References: Message-ID: > > What surprises me is that ffmpeg always outputs all kinds of > messages whenever I feed a damaged H264 stream into the decoder: > Do you have a damaged sample that produces no such messages? > > Coul you, give me an example, of an ffmpeg command which display errors messages, when you receive a video on a streaming source? In fact, i need a command which describe me the video frame by frame ( type of frame, frame ID, related frames) and if it's damager or not. Thank you Regards O. Stiti From andrey.krieger.utkin at gmail.com Wed Mar 21 10:54:58 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Wed, 21 Mar 2012 11:54:58 +0200 Subject: [FFmpeg-user] MPEG-TS trouble In-Reply-To: References: <1332307981.76400.YahooMailNeo@web162003.mail.bf1.yahoo.com> Message-ID: 2012/3/21 Carl Eugen Hoyos : > No transcoding takes place with above command line. Carl, i think you're wrong here, i use this bitstream filter with -c copy successfully. Pity that this bsf is not verbose on reason of fail. -- Andrey Utkin From cehoyos at ag.or.at Wed Mar 21 11:06:41 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 21 Mar 2012 10:06:41 +0000 (UTC) Subject: [FFmpeg-user] MPEG-TS trouble References: <1332307981.76400.YahooMailNeo@web162003.mail.bf1.yahoo.com> Message-ID: Andrey Utkin gmail.com> writes: > 2012/3/21 Carl Eugen Hoyos ag.or.at>: > > No transcoding takes place with above command line. > > Carl, i think you're wrong here, i use this bitstream filter with -c > copy successfully. But a bitstream filter does no transcoding afaict. Carl Eugen From cehoyos at ag.or.at Wed Mar 21 11:07:19 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 21 Mar 2012 10:07:19 +0000 (UTC) Subject: [FFmpeg-user] Switch "on" Reference Picture Selection (RPS) in the libavcodec/ituh263enc.c Source File References: Message-ID: Oussama Stiti gmail.com> writes: > Coul you, give me an example, of an ffmpeg command which display errors > messages, when you receive a video on a streaming source? ffmpeg -i input -f null - Carl Eugen From cehoyos at ag.or.at Wed Mar 21 11:16:26 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 21 Mar 2012 10:16:26 +0000 (UTC) Subject: [FFmpeg-user] [flv @ 0x6378a0] Error, Invalid timestamp=0, last=0 References: <4F699BFB.3080500@access-dev.com> Message-ID: Access-Dev access-dev.com> writes: > [flv @ 0x6378a0] Error, Invalid timestamp=0, last=0 > Video encoding failed > > Any suggestions ? Please provide the input sample. Carl Eugen From andrey.krieger.utkin at gmail.com Wed Mar 21 11:23:49 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Wed, 21 Mar 2012 12:23:49 +0200 Subject: [FFmpeg-user] MPEG-TS trouble In-Reply-To: References: <1332307981.76400.YahooMailNeo@web162003.mail.bf1.yahoo.com> Message-ID: 2012/3/21 Carl Eugen Hoyos : > Andrey Utkin gmail.com> writes: > >> 2012/3/21 Carl Eugen Hoyos ag.or.at>: >> > No transcoding takes place with above command line. >> >> Carl, i think you're wrong here, i use this bitstream filter with -c >> copy successfully. > > But a bitstream filter does no transcoding afaict. Ah, sorry, i misunderstood you. roko, First of all check that you use recent ffmpeg, at last 0.10 version. Then try doing ffmpeg -i rtsp://admin:admin at 192.168.99.146/11 -vcodec copy -y -r 25 algo.mp4 ffmpeg -loglevel debug -i algo.mp4 -c copy -y -r 25 -vbsf h264_mp4toannexb algo.ts And report here does error stays. If so, please share full log of second command execution, and algo.mp4 file. -- Andrey Utkin From oussama.stiti at gmail.com Wed Mar 21 11:57:19 2012 From: oussama.stiti at gmail.com (Oussama Stiti) Date: Wed, 21 Mar 2012 19:57:19 +0900 Subject: [FFmpeg-user] Switch "on" Reference Picture Selection (RPS) in the libavcodec/ituh263enc.c Source File In-Reply-To: References: Message-ID: Here is the result of the command line: ~$ ffmpeg -i $HOME/T?l?chargements/lazy.mp4 -f null - ffmpeg version N-38203-g388b7ac Copyright (c) 2000-2012 the FFmpeg developers built on Mar 2 2012 23:24:13 with gcc 4.6.2 configuration: --enable-libmp3lame --enable-libxvid --enable-libvorbis --enable-gpl --enable-libfaac --enable-libtheora --enable-zlib --disable-shared --enable-libx264 --enable-libdirac --enable-nonfree --enable-version3 --enable-libschroedinger --enable-avfilter --enable-libspeex --enable-libopenjpeg --enable-libgsm --enable-postproc --enable-pthreads --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-ffplay --enable-pthreads --prefix=/usr/local --enable-x11grab --enable-runtime-cpudetect --enable-bzlib --enable-libdc1394 --enable-libvpx libavutil 51. 40.100 / 51. 40.100 libavcodec 54. 4.100 / 54. 4.100 libavformat 54. 1.100 / 54. 1.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 62.101 / 2. 62.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/home/oussama/T?l?chargements/lazy.mp4': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42avc1 creation_time : 2011-08-30 21:43:14 Duration: 00:03:28.18, start: 0.000000, bitrate: 830 kb/s Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 120 kb/s Metadata: creation_time : 2011-08-30 21:43:14 handler_name : Apple Sound Media Handler Stream #0:1(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 640x480, 704 kb/s, 30 fps, 30 tbr, 600 tbn, 1200 tbc Metadata: creation_time : 2011-08-30 21:43:14 handler_name : Apple Video Media Handler [buffer @ 0xa516ea0] w:640 h:480 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param: Output #0, null, to 'pipe:': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42avc1 creation_time : 2011-08-30 21:43:14 encoder : Lavf54.1.100 Stream #0:0(eng): Video: rawvideo (I420 / 0x30323449), yuv420p, 640x480, q=2-31, 200 kb/s, 90k tbn, 30 tbc Metadata: creation_time : 2011-08-30 21:43:14 handler_name : Apple Video Media Handler Stream #0:1(eng): Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s Metadata: creation_time : 2011-08-30 21:43:14 handler_name : Apple Sound Media Handler Stream mapping: Stream #0:1 -> #0:0 (h264 -> rawvideo) Stream #0:0 -> #0:1 (aac -> pcm_s16le) Press [q] to stop, [?] for help *[h264 @ 0xa50ab80] Increasing reorder buffer to 1 frame= 6246 fps=658 q=0.0 Lsize= 0kB time=00:03:28.19 bitrate= 0.0kbits/s video:0kB audio:35864kB global headers:0kB muxing overhead -100.000000%* No more informations.. I want to analyse the video frame by frame. Is it possible ? O. Stiti From ml at access-dev.com Wed Mar 21 12:40:44 2012 From: ml at access-dev.com (Access-Dev) Date: Wed, 21 Mar 2012 12:40:44 +0100 Subject: [FFmpeg-user] [flv @ 0x6378a0] Error, Invalid timestamp=0, last=0 In-Reply-To: References: <4F699BFB.3080500@access-dev.com> Message-ID: <4F69BE3C.5030409@access-dev.com> Hi, http://accessdev.s3.amazonaws.com/temp/407.3gp Nicolas > Access-Dev access-dev.com> writes: > >> [flv @ 0x6378a0] Error, Invalid timestamp=0, last=0 >> Video encoding failed >> >> Any suggestions ? > Please provide the input sample. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From rodney.baker at iinet.net.au Wed Mar 21 13:12:04 2012 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Wed, 21 Mar 2012 22:42:04 +1030 Subject: [FFmpeg-user] Switch "on" Reference Picture Selection (RPS) in the libavcodec/ituh263enc.c Source File In-Reply-To: References: Message-ID: <201203212242.04483.rodney.baker@iinet.net.au> On Wed, 21 Mar 2012 21:27:19 Oussama Stiti wrote: > Here is the result of the command line: > > ~$ ffmpeg -i $HOME/T?l?chargements/lazy.mp4 -f null - > ffmpeg version N-38203-g388b7ac Copyright (c) 2000-2012 the FFmpeg > developers > built on Mar 2 2012 23:24:13 with gcc 4.6.2 > configuration: --enable-libmp3lame --enable-libxvid --enable-libvorbis > --enable-gpl --enable-libfaac --enable-libtheora --enable-zlib > --disable-shared --enable-libx264 --enable-libdirac --enable-nonfree > --enable-version3 --enable-libschroedinger --enable-avfilter > --enable-libspeex --enable-libopenjpeg --enable-libgsm --enable-postproc > --enable-pthreads --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-ffplay --enable-pthreads --prefix=/usr/local --enable-x11grab > --enable-runtime-cpudetect --enable-bzlib --enable-libdc1394 > --enable-libvpx libavutil 51. 40.100 / 51. 40.100 > libavcodec 54. 4.100 / 54. 4.100 > libavformat 54. 1.100 / 54. 1.100 > libavdevice 53. 4.100 / 53. 4.100 > libavfilter 2. 62.101 / 2. 62.101 > libswscale 2. 1.100 / 2. 1.100 > libswresample 0. 7.100 / 0. 7.100 > libpostproc 52. 0.100 / 52. 0.100 > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from > '/home/oussama/T?l?chargements/lazy.mp4': > Metadata: > major_brand : mp42 > minor_version : 1 > compatible_brands: mp42avc1 > creation_time : 2011-08-30 21:43:14 > Duration: 00:03:28.18, start: 0.000000, bitrate: 830 kb/s > Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, > s16, 120 kb/s > Metadata: > creation_time : 2011-08-30 21:43:14 > handler_name : Apple Sound Media Handler > Stream #0:1(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, > 640x480, 704 kb/s, 30 fps, 30 tbr, 600 tbn, 1200 tbc > Metadata: > creation_time : 2011-08-30 21:43:14 > handler_name : Apple Video Media Handler > [buffer @ 0xa516ea0] w:640 h:480 pixfmt:yuv420p tb:1/1000000 sar:0/1 > sws_param: > Output #0, null, to 'pipe:': > Metadata: > major_brand : mp42 > minor_version : 1 > compatible_brands: mp42avc1 > creation_time : 2011-08-30 21:43:14 > encoder : Lavf54.1.100 > Stream #0:0(eng): Video: rawvideo (I420 / 0x30323449), yuv420p, > 640x480, q=2-31, 200 kb/s, 90k tbn, 30 tbc > Metadata: > creation_time : 2011-08-30 21:43:14 > handler_name : Apple Video Media Handler > Stream #0:1(eng): Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s > Metadata: > creation_time : 2011-08-30 21:43:14 > handler_name : Apple Sound Media Handler > Stream mapping: > Stream #0:1 -> #0:0 (h264 -> rawvideo) > Stream #0:0 -> #0:1 (aac -> pcm_s16le) > Press [q] to stop, [?] for help > *[h264 @ 0xa50ab80] Increasing reorder buffer to 1 > frame= 6246 fps=658 q=0.0 Lsize= 0kB time=00:03:28.19 bitrate= > 0.0kbits/s > video:0kB audio:35864kB global headers:0kB muxing overhead -100.000000%* > > No more informations.. I want to analyse the video frame by frame. Is it > possible ? > > > O. Stiti Try ffprobe -show_frames -- =================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au =================================================== From nicolas.george at normalesup.org Wed Mar 21 14:11:55 2012 From: nicolas.george at normalesup.org (Nicolas George) Date: Wed, 21 Mar 2012 14:11:55 +0100 Subject: [FFmpeg-user] correct syntax for using the pan filter In-Reply-To: References: Message-ID: <20120321131155.GC18560@phare.normalesup.org> Le decadi 30 vent?se, an CCXX, jacob s a ?crit?: > anyone knows the proper way to use the pan filter ? I am trying to take a > stereo input and mute one channel and get a stereo output with one channel > muted. > This is my original command: > -rtbufsize 100000000 -f dshow -i > video="screen-capture-recorder":audio="SoundMAX Digital Audio" -vcodec > libx264 -preset ultrafast -tune zerolatency -r 10 -async 1 -ab 32k -ar > 22050 -bsf:v h264_mp4toannexb -b 614400 -f mpegts udp://192.168.5.215:48550 > > According to the manual this is possible ( "If the input is a stereo audio > stream, you can mute the front left channel (and still keep the stereo > channel layout) with: pan="stereo:c1=c1" ") > I tried this but it didn't work, I guess I am not using the correct cli > syntax. > Please reply with the correct way to use pan ( within my original command ) You should be able to write: ffmpeg -f dshow -i blablabla -af pan=stereo:c1=c1 ... Unfortunately, the -af option and corresponding infrastructure is not yet implemented in the ffmpeg command-line tool. On the other hand, you should be able to write: ffmpeg -f lavfi -i 'movie=video=blabla:f=dshow [out] ; amovie=audio=blabla:f=dshow, af=pan=stereo:c1=c1" ... or something like that. Unfortunately, you will have two demuxers instead of one, and I am not sure the rtbufsize option can be passed. The situation may improve progressively though. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From guisheng315 at gmail.com Wed Mar 21 15:17:10 2012 From: guisheng315 at gmail.com (gs_gail) Date: Wed, 21 Mar 2012 22:17:10 +0800 Subject: [FFmpeg-user] MPEG-TS trouble In-Reply-To: References: <1332307981.76400.YahooMailNeo@web162003.mail.bf1.yahoo.com> Message-ID: <1332339430.4061.14.camel@gw-laptop> there is a bug in libavcodec/h264_mp4toannexb_bsf.c when the ctx->length_size == 3 , the filter will report the error "Invalid argument" diff --git a/libavcodec/h264_mp4toannexb_bsf.c b/libavcodec/h264_mp4toannexb_bsf.c index 5085ecb..fa16f2a 100644 --- a/libavcodec/h264_mp4toannexb_bsf.c +++ b/libavcodec/h264_mp4toannexb_bsf.c @@ -82,8 +82,8 @@ static int h264_mp4toannexb_filter(AVBitStreamFilterContext *bsfc, /* retrieve length coded size */ ctx->length_size = (*extradata++ & 0x3) + 1; - if (ctx->length_size == 3) - return AVERROR(EINVAL); + // if (ctx->length_size == 3) + // return AVERROR(EINVAL); /* retrieve sps and pps unit(s) */ unit_nb = *extradata++ & 0x1f; /* number of sps unit(s) */ @@ -146,8 +146,10 @@ pps: nal_size = buf[0]; } else if (ctx->length_size == 2) { nal_size = AV_RB16(buf); - } else - nal_size = AV_RB32(buf); + } else { + for(nal_size = 0, unit_type = 0; unit_typelength_size; unit_type++) + nal_size = (nal_size << 8) | buf[unit_type]; + } buf += ctx->length_size; unit_type = *buf & 0x1f; ? 2012-03-21?? 12:23 +0200?Andrey Utkin??? > 2012/3/21 Carl Eugen Hoyos : > > Andrey Utkin gmail.com> writes: > > > >> 2012/3/21 Carl Eugen Hoyos ag.or.at>: > >> > No transcoding takes place with above command line. > >> > >> Carl, i think you're wrong here, i use this bitstream filter with -c > >> copy successfully. > > > > But a bitstream filter does no transcoding afaict. > > Ah, sorry, i misunderstood you. > > roko, > First of all check that you use recent ffmpeg, at last 0.10 version. > Then try doing > ffmpeg -i rtsp://admin:admin at 192.168.99.146/11 -vcodec copy -y -r 25 algo.mp4 > ffmpeg -loglevel debug -i algo.mp4 -c copy -y -r 25 -vbsf > h264_mp4toannexb algo.ts > And report here does error stays. If so, please share full log of > second command execution, and algo.mp4 file. > From joddo at jerfu.com Wed Mar 21 15:56:54 2012 From: joddo at jerfu.com (Jeremy Oddo) Date: Wed, 21 Mar 2012 07:56:54 -0700 Subject: [FFmpeg-user] ProRes (LT) Quicktimes: Good for Windows. Not so good for Mac. In-Reply-To: References: <4F68F4F9.3040504@gmx.de> Message-ID: This is very interesting, Thomas. Thanks for your insight. > About this, I think the Final Cut media warning may be related to the > A/V interleaving, i.e. the way the chunk offset tables are written. I > don't think it has as much to do with information in the QuickTime > header. Files written without audio don't show the warning. Baptiste > Coudurier might comment on this. If anyone knows why this is > happening, is would be him. I could be wrong (and I'll double-check this) but I believe our QTs DID NOT have an audio track. > Also it should be noted that the media performance warning does not > occur when QuickTime files are written with a co64 atom. So, Final Cut > apparently likes FFmpeg MOV files written with 64 bit offsets. Could you explain the "co64 atom" thing a bit? I don't know what co64 atom is. Is it possible for FFMPEG to write ProRes QTs with a co64 atom or is that related to some other codec? > Someone might take a gander at movenc.c and see if interleaving is > done differently with 64 bit MOV offsets vs. 32 bit ones. Does this 32/64-bit offsets related to the OS or can a 64-bit MOV offset be written from a 32-bit OS? Don't know if it matters but we're using the 64-bit FFMPEG. From pb at das-werkstatt.com Wed Mar 21 17:44:31 2012 From: pb at das-werkstatt.com (Peter B.) Date: Wed, 21 Mar 2012 17:44:31 +0100 Subject: [FFmpeg-user] FFmpeg's guidelines for cross-platform compilation? Message-ID: <20120321174431.27813dx6uxyo6yr3@webmail.tuwien.ac.at> I somewhat remember to have read some comment about that the FFmpeg-developers try to write the code in a way that it compiles on as many architectures/platforms as easily as possible. If I'm wrong, please correct me - but if there is such a text, could you please point me to it, as I would need it as a reference in an article. Thank you very much, Pb From cehoyos at ag.or.at Wed Mar 21 17:55:44 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 21 Mar 2012 16:55:44 +0000 (UTC) Subject: [FFmpeg-user] MPEG-TS trouble References: <1332307981.76400.YahooMailNeo@web162003.mail.bf1.yahoo.com> <1332339430.4061.14.camel@gw-laptop> Message-ID: gs_gail gmail.com> writes: > there is a bug in libavcodec/h264_mp4toannexb_bsf.c > when the ctx->length_size == 3 , the filter will report the error > "Invalid argument" Please post the command line together with the complete, uncut console output. Carl Eugen From Jeff_Hollar at prn.com Wed Mar 21 19:14:11 2012 From: Jeff_Hollar at prn.com (Hollar Jeff) Date: Wed, 21 Mar 2012 11:14:11 -0700 Subject: [FFmpeg-user] Need help understanding encoding error message Message-ID: I've been using FFMPEG [0.6-4:6-2] on ubuntu successfully encoding mpeg2 streams using the following command: %ffmpeg -qscale 5 -loop_input -I audio/demo.mp3 -I pngs/svg_tide.png -flags +ilme -f mpegts -b 18000000 -ab 192k -top 1 -aspect 16:9 -r 29.97 -t 10 -vcodec mpeg2video Preview155948.mpg This actually works but when I do this on windows XP using FFMPEG the -loop_input option is no longer working and other options have changed. The major issue is the error. Could someone explain what this error means and how to resolve it? ffmpeg -i audio/demo.mp3 -i pngs/svg_tide.png -q:v 5 -flags +ilme -b:v 18000000 -f mpegts -ab 192k -top 1 -aspect 16:9 -r 29.97/1001 -t 10 -vcodec mpeg2video Preview155948.mpg ffmpeg version N-38622-g1eabd71 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 7 2012 00:18:03 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 42.100 / 51. 42.100 libavcodec 54. 10.100 / 54. 10.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 63.100 / 2. 63.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 [mp3 @ 0216AE20] max_analyze_duration 5000000 reached at 5015510 [mp3 @ 0216AE20] Estimating duration from bitrate, this may be inaccurate Input #0, mp3, from 'audio/demo.mp3': Metadata: title : Track 2 genre : Unknown track : 02 TFLT : audio/mp3 artist : Unknown album : Untitled - 02-09-07 Duration: 00:03:37.45, start: 0.000000, bitrate: 191 kb/s Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16, 192 kb/s Input #1, image2, from 'pngs/svg_tide.png': Duration: 00:00:00.04, start: 0.000000, bitrate: N/A Stream #1:0: Video: png, rgba64be, 1920x1080, 25 tbr, 25 tbn, 25 tbc Incompatible pixel format 'rgba64be' for codec 'mpeg2video', auto-selecting format 'yuv420p' [buffer @ 02261260] w:1920 h:1080 pixfmt:rgba64be tb:1/1000000 sar:0/1 sws_param: [buffersink @ 02256800] auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'out' [scale @ 02256960] w:1920 h:1080 fmt:rgba64be -> w:1920 h:1080 fmt:yuv420p flags:0x4 [mpeg2video @ 02255B40] MPEG1/2 does not support 5/1 fps Output #0, mpegts, to 'Preview155948.mpg': Metadata: title : Track 2 genre : Unknown track : 02 TFLT : audio/mp3 artist : Unknown album : Untitled - 02-09-07 Stream #0:0: Video: mpeg2video, yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 18000 kb/s, 90k tbn, 5 tbc Stream #0:1: Audio: none, 44100 Hz, stereo, s16, 128 kb/s Stream mapping: Stream #1:0 -> #0:0 (png -> mpeg2video) Stream #0:0 -> #0:1 (mp3 -> mp2) Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height Thanks, Jeff From cehoyos at ag.or.at Wed Mar 21 20:53:20 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 21 Mar 2012 19:53:20 +0000 (UTC) Subject: [FFmpeg-user] Need help understanding encoding error message References: Message-ID: Hollar Jeff prn.com> writes: > I've been using FFMPEG [0.6-4:6-2] on ubuntu successfully encoding > mpeg2 streams using the following command: All releases of the 0.6 branch contain an extraordinary high number of regressions and bugs, please update to current git head. > %ffmpeg -qscale 5 -loop_input The name of the command is now "-loop 1" and contrary to before, it will now fail if it does not work (which I very much consider an improvement), so it's ffmpeg -loop 1 -i pngs/svg_tide.png. > -I audio/demo.mp3 "-I" is not supported by current versions, was it truly supported before? (It's "-i" now) [...] > -r 29.97/1001 > [mpeg2video @ 02255B40] MPEG1/2 does not support 5/1 fps Did you perhaps mean "-r 30000/1001"? Carl Eugen From roko98 at yahoo.com Wed Mar 21 15:12:24 2012 From: roko98 at yahoo.com (roko) Date: Wed, 21 Mar 2012 07:12:24 -0700 (PDT) Subject: [FFmpeg-user] MPEG-TS trouble In-Reply-To: References: <1332307981.76400.YahooMailNeo@web162003.mail.bf1.yahoo.com> Message-ID: <1332339144.32036.YahooMailNeo@web162001.mail.bf1.yahoo.com> Hi Andrey... thx for your response and your interest.? I'm using 0.10.? I' including the recorded files as attachments. Here is the full console output: ffmpeg -i rtsp://admin:admin at 192.168.99.146/11 -vcodec copy -y -r 25 algo.mp4 ffmpeg version 0.10 Copyright (c) 2000-2012 the FFmpeg developers ? built on Mar? 2 2012 14:36:33 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] ? configuration: --shlibdir=/usr/lib64 --prefix=/usr --mandir=/usr/share/man --libdir=/usr/lib64 --enable-shared --disable-static --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libspeex --enable-libxvid --enable-postproc --enable-gpl --enable-x11grab --extra-cflags='-fmessage-length=0 -O2 -Wall -D_FORTIFY_SOURCE=2 -fstack-protector -funwind-tables -fasynchronous-unwind-tables -g -fPIC -I/usr/include/gsm' --enable-debug --disable-stripping --enable-libgsm --enable-libschroedinger --enable-libdirac --enable-avfilter --enable-libvpx --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libdc1394 --enable-pthreads --enable-librtmp ? libavutil????? 51. 34.101 / 51. 34.101 ? libavcodec???? 53. 60.100 / 53. 60.100 ? libavformat??? 53. 31.100 / 53. 31.100 ? libavdevice??? 53.? 4.100 / 53.? 4.100 ? libavfilter???? 2. 60.100 /? 2. 60.100 ? libswscale????? 2.? 1.100 /? 2.? 1.100 ? libswresample?? 0.? 6.100 /? 0.? 6.100 ? libpostproc??? 52.? 0.100 / 52.? 0.100 [rtsp @ 0x638600] Estimating duration from bitrate, this may be inaccurate Input #0, rtsp, from 'rtsp://admin:admin at 192.168.99.146/11':??????????????????? ? Metadata: ??? title?????????? : \11 ? Duration: N/A, start: 0.039956, bitrate: N/A ??? Stream #0:0: Video: h264 (Constrained Baseline), yuv420p, 640x480, 25 tbr, 90k tbn, 180k tbc Output #0, mp4, to 'algo.mp4': ? Metadata: ??? title?????????? : \11 ??? encoder???????? : Lavf53.31.100 ??? Stream #0:0: Video: h264 (![0][0][0] / 0x0021), yuv420p, 640x480, q=2-31, 90k tbn, 90k tbc Stream mapping: ? Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help frame=?? 36 fps=? 0 q=-1.0 size=???? 123kB time=00:00:01.35 bitrate= 740.1kbits/frame=?? 49 fps= 47 q=-1.0 size=???? 156kB time=00:00:01.87 bitrate= 681.3kbits/frame=?? 62 fps= 40 q=-1.0 size=???? 217kB time=00:00:02.39 bitrate= 741.7kbits/frame=?? 75 fps= 36 q=-1.0 size=???? 252kB time=00:00:02.91 bitrate= 708.6kbits/frame=?? 88 fps= 34 q=-1.0 size=???? 286kB time=00:00:03.43 bitrate= 683.0kbits/frame=? 101 fps= 32 q=-1.0 size=???? 337kB time=00:00:03.95 bitrate= 697.9kbits/frame=? 114 fps= 31 q=-1.0 size=???? 382kB time=00:00:04.47 bitrate= 699.8kbits/frame=? 127 fps= 31 q=-1.0 size=???? 416kB time=00:00:04.99 bitrate= 682.6kbits/frame=? 140 fps= 30 q=-1.0 size=???? 450kB time=00:00:05.51 bitrate= 668.1kbits/frame=? 153 fps= 30 q=-1.0 size=???? 502kB time=00:00:06.03 bitrate= 681.1kbits/frame=? 166 fps= 29 q=-1.0 size=???? 545kB time=00:00:06.55 bitrate= 681.6kbits/frame=? 179 fps= 29 q=-1.0 size=???? 580kB time=00:00:07.07 bitrate= 671.3kbits/frame=? 192 fps= 28 q=-1.0 size=???? 613kB time=00:00:07.59 bitrate= 661.5kbits/frame=? 205 fps= 28 q=-1.0 size=???? 670kB time=00:00:08.11 bitrate= 676.5kbits/frame=? 218 fps= 28 q=-1.0 size=???? 711kB time=00:00:08.63 bitrate= 674.4kbits/frame=? 231 fps= 28 q=-1.0 size=???? 745kB time=00:00:09.15 bitrate= 667.0kbits/frame=? 244 fps= 28 q=-1.0 size=???? 779kB time=00:00:09.66 bitrate= 659.6kbits/frame=? 257 fps= 27 q=-1.0 size=???? 837kB time=00:00:10.18 bitrate= 673.2kbits/frame=? 266 fps= 27 q=-1.0 Lsize=???? 866kB time=00:00:10.54 bitrate= 672.2kbits/s??? video:863kB audio:0kB global headers:0kB muxing overhead 0.300009% I just save 10 seconds... The re-encoding of the file, as you suggest, works and the new file plays well with ffplay: ffmpeg -loglevel debug -i algo.mp4 -c copy -y -r 25 -vbsf h264_mp4toannexb algo.ts ffmpeg version 0.10 Copyright (c) 2000-2012 the FFmpeg developers ? built on Mar? 2 2012 14:36:33 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] ? configuration: --shlibdir=/usr/lib64 --prefix=/usr --mandir=/usr/share/man --libdir=/usr/lib64 --enable-shared --disable-static --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libspeex --enable-libxvid --enable-postproc --enable-gpl --enable-x11grab --extra-cflags='-fmessage-length=0 -O2 -Wall -D_FORTIFY_SOURCE=2 -fstack-protector -funwind-tables -fasynchronous-unwind-tables -g -fPIC -I/usr/include/gsm' --enable-debug --disable-stripping --enable-libgsm --enable-libschroedinger --enable-libdirac --enable-avfilter --enable-libvpx --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libdc1394 --enable-pthreads --enable-librtmp ? libavutil????? 51. 34.101 / 51. 34.101 ? libavcodec???? 53. 60.100 / 53. 60.100 ? libavformat??? 53. 31.100 / 53. 31.100 ? libavdevice??? 53.? 4.100 / 53.? 4.100 ? libavfilter???? 2. 60.100 /? 2. 60.100 ? libswscale????? 2.? 1.100 /? 2.? 1.100 ? libswresample?? 0.? 6.100 /? 0.? 6.100 ? libpostproc??? 52.? 0.100 / 52.? 0.100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x638600] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100???????????????????????????????????????????????????????? [mov,mp4,m4a,3gp,3g2,mj2 @ 0x638600] ISO: File Type Major Brand: isom?????????? [h264 @ 0x63ee00] err{or,}_recognition separate: 1; 1?????????????????????????? [h264 @ 0x63ee00] err{or,}_recognition combined: 1; 10001?????????????????????? [mov,mp4,m4a,3gp,3g2,mj2 @ 0x638600] All info found???????????????????????????? rfps: 24.416667 0.019750??????????????????????????????????????????????????????? ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 24.500000 0.014722 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 24.583333 0.010432 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 24.666667 0.006878 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 24.750000 0.004062 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 24.833333 0.001983 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 24.916667 0.000642 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 25.000000 0.000038 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 25.083333 0.000171 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 25.166667 0.001041 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 25.250000 0.002649 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 25.333333 0.004994 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 25.416667 0.008077 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 25.500000 0.011897 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 25.583333 0.016454 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 49.500000 0.016248 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 49.583333 0.011722 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 49.666667 0.007933 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 49.750000 0.004882 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 49.833333 0.002567 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 49.916667 0.000991 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 50.000000 0.000151 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 50.083333 0.000049 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 50.166667 0.000684 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 50.250000 0.002056 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 50.333333 0.004165 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 50.416667 0.007012 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 50.500000 0.010597 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 50.583333 0.014918 ??? Last message repeated 1 times?????????????????????????????????????????????? rfps: 50.666667 0.019977 ??? Last message repeated 1 times?????????????????????????????????????????????? Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'algo.mp4': ? Metadata: ??? major_brand???? : isom ??? minor_version?? : 512 ??? compatible_brands: isomiso2avc1mp41 ??? title?????????? : \11 ??? encoder???????? : Lavf53.31.100 ? Duration: 00:00:10.54, start: 0.000000, bitrate: 672 kb/s ??? Stream #0:0(und), 22, 1/90000: Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 640x480, 1/180000, 670 kb/s, 25.22 fps, 25 tbr, 90k tbn, 180k tbc ??? Metadata: ????? handler_name??? : [mpegts @ 0x63fe80] muxrate VBR, pcr every 9000 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to 'algo.ts': ? Metadata: ??? major_brand???? : isom ??? minor_version?? : 512 ??? compatible_brands: isomiso2avc1mp41 ??? title?????????? : \11 ??? encoder???????? : Lavf53.31.100 ??? Stream #0:0(und), 0, 1/90000: Video: h264 (avc1 / 0x31637661), yuv420p, 640x480, 1/90000, q=2-31, 670 kb/s, 25.22 fps, 90k tbn, 90k tbc ??? Metadata: ????? handler_name??? : Stream mapping: ? Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help frame=? 266 fps=? 0 q=-1.0 Lsize=???? 962kB time=00:00:10.54 bitrate= 747.4kbits/s??? video:863kB audio:0kB global headers:0kB muxing overhead 11.490276% Just for competition, I try to save the? RTSP stream using the same settings, it throws the same error: Input #0, rtsp, from 'rtsp://admin:admin at 192.168.99.146/11':??????????????????? ? Metadata: ??? title?????????? : \11 ? Duration: N/A, start: 0.039956, bitrate: N/A ??? Stream #0:0, 22, 1/90000: Video: h264 (Constrained Baseline), yuv420p, 640x480, 1/180000, 25 tbr, 90k tbn, 180k tbc [mpegts @ 0x778440] muxrate VBR, pcr every 9000 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to 'algo.ts': ? Metadata: ??? title?????????? : \11 ??? encoder???????? : Lavf53.31.100 ??? Stream #0:0, 0, 1/90000: Video: h264, yuv420p, 640x480, 1/90000, q=2-31, 90k tbn, 90k tbc Stream mapping: ? Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument???????????????????????????????????????????????????????????????? [mpegts @ 0x778440] H.264 bitstream malformed, no startcode found, use the h264_mp4toannexb bitstream filter??????????????????????????????????????????????????? av_interleaved_write_frame(): Invalid data found when processing input Any help will be greatly appreciated! ________________________________ 2012/3/21 Carl Eugen Hoyos : > Andrey Utkin gmail.com> writes: > >> 2012/3/21 Carl Eugen Hoyos ag.or.at>: >> > No transcoding takes place with above command line. >> >> Carl, i think you're wrong here, i use this bitstream filter with -c >> copy successfully. > > But a bitstream filter does no transcoding afaict. Ah, sorry, i misunderstood you. roko, First of all check that you use recent ffmpeg, at last 0.10 version. Then try doing ffmpeg -i rtsp://admin:admin at 192.168.99.146/11 -vcodec copy -y -r 25 algo.mp4 ffmpeg -loglevel debug -i algo.mp4 -c copy -y -r 25 -vbsf h264_mp4toannexb algo.ts And report here does error stays. If so, please share full log of second command execution, and algo.mp4 file. -- Andrey Utkin -------------- next part -------------- A non-text attachment was scrubbed... Name: algo.mp4 Type: application/octet-stream Size: 886291 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: algo.ts Type: application/octet-stream Size: 985496 bytes Desc: not available URL: From ryan.ismail2 at gmail.com Wed Mar 21 18:02:16 2012 From: ryan.ismail2 at gmail.com (ryan.ismail2 at gmail.com) Date: Wed, 21 Mar 2012 17:02:16 +0000 Subject: [FFmpeg-user] Ffplay feature request Message-ID: <1267970240-1332349330-cardhu_decombobulator_blackberry.rim.net-1772329587-@b4.c15.bise7.blackberry> Hi I would really appreciate it if ffplay could interact with lirc and have a -playlist switch similar to mplayer's. Thank you in advance From michaelni at gmx.at Wed Mar 21 21:38:42 2012 From: michaelni at gmx.at (Michael Niedermayer) Date: Wed, 21 Mar 2012 21:38:42 +0100 Subject: [FFmpeg-user] MPEG-TS trouble In-Reply-To: <1332339430.4061.14.camel@gw-laptop> References: <1332307981.76400.YahooMailNeo@web162003.mail.bf1.yahoo.com> <1332339430.4061.14.camel@gw-laptop> Message-ID: <20120321203842.GB13842@kiste2> On Wed, Mar 21, 2012 at 10:17:10PM +0800, gs_gail wrote: > there is a bug in libavcodec/h264_mp4toannexb_bsf.c > when the ctx->length_size == 3 , the filter will report the error > "Invalid argument" patch applied with some modifications Thanks [...] -- Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB Let us carefully observe those good qualities wherein our enemies excel us and endeavor to excel them, by avoiding what is faulty, and imitating what is excellent in them. -- Plutarch -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From langdon at gmail.com Thu Mar 22 13:02:59 2012 From: langdon at gmail.com (Langdon) Date: Thu, 22 Mar 2012 08:02:59 -0400 Subject: [FFmpeg-user] Trouble encoding h.264 Message-ID: I have some raw video (-f yuv4mpegpipe) and audio (-f u16le -acodec pcm_s16le) that I extracted from another file. When I try to re-encode the two parts as h.264, the sound plays OK, but the video is mostly gray in Media Player Classic Home Cinema, as well as when uploaded to YouTube. ffplay.exe seems to play it OK, but I'm creating the video for mass consumption. Am I doing something wrong? I had to tell ffmpeg about every attribute of the audio before it would work right, but I'm not sure what to tell it about the video. ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 48000 -i c:/alka-video/workspace/06A.cat.a -i c:/alka-video/workspace/06A.cat.v -vcodec libx264 -same_quant -y c:\alka-video/output/06A.avi > _h264.txt ffmpeg version N-38622-g1eabd71 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 7 2012 00:21:47 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 42.100 / 51. 42.100 libavcodec 54. 10.100 / 54. 10.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 63.100 / 2. 63.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 [u16le @ 000000000031F3C0] Estimating duration from bitrate, this may be inaccurate Input #0, u16le, from 'c:/alka-video/workspace/06A.cat.a': Duration: 00:00:03.79, start: 0.000000, bitrate: 1536 kb/s Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s [yuv4mpegpipe @ 0000000001E4F8C0] Estimating duration from bitrate, this may be inaccurate Input #1, yuv4mpegpipe, from 'c:/alka-video/workspace/06A.cat.v': Duration: N/A, bitrate: N/A Stream #1:0: Video: rawvideo (Y42B / 0x42323459), yuv422p, 1280x720, 23.98 fps, 23.98 tbr, 23.98 tbn, 23.98 tbc [buffer @ 0000000001E617D0] w:1280 h:720 pixfmt:yuv422p tb:1/1000000 sar:0/1 sws_param: [libx264 @ 0000000001E61350] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.1 Cache64 [libx264 @ 0000000001E61350] profile High 4:2:2, level 3.1, 4:2:2 8-bit Output #0, avi, to 'c:\alka-video/output/06A.avi': Metadata: ISFT : Lavf54.2.100 Stream #0:0: Video: h264 (H264 / 0x34363248), yuv422p, 1280x720, q=-1--1, 23.98 tbn, 23.98 tbc Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 48000 Hz, 2 channels, s16 Stream mapping: Stream #1:0 -> #0:0 (rawvideo -> libx264) Stream #0:0 -> #0:1 (pcm_s16le -> libmp3lame) Press [q] to stop, [?] for help Truncating packet of size 4096 to 3737 Truncating packet of size 4096 to 1 frame= 91 fps= 7 q=28.0 Lsize= 1187kB time=00:00:03.71 bitrate=2619.9kbits/s video:1099kB audio:72kB global headers:0kB muxing overhead 1.332733% [libx264 @ 0000000001E61350] frame I:2 Avg QP:22.39 size: 44095 [libx264 @ 0000000001E61350] frame P:51 Avg QP:24.42 size: 16121 [libx264 @ 0000000001E61350] frame B:38 Avg QP:27.17 size: 5667 [libx264 @ 0000000001E61350] consecutive B-frames: 22.0% 61.5% 16.5% 0.0% [libx264 @ 0000000001E61350] mb I I16..4: 11.8% 76.5% 11.7% [libx264 @ 0000000001E61350] mb P I16..4: 1.4% 7.2% 0.5% P16..4: 50.1% 16.1% 7.5% 0.0% 0.0% skip:17.1% [libx264 @ 0000000001E61350] mb B I16..4: 0.1% 1.1% 0.1% B16..8: 41.9% 4.9% 0.9% direct: 2.3% skip:48.8% L0:36.4% L1:56.4% BI: 7.2% [libx264 @ 0000000001E61350] 8x8 transform intra:78.7% inter:84.4% [libx264 @ 0000000001E61350] coded y,uvDC,uvAC intra: 64.4% 79.3% 10.1% inter: 18.2% 37.1% 0.1% [libx264 @ 0000000001E61350] i16 v,h,dc,p: 21% 28% 4% 47% [libx264 @ 0000000001E61350] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 19% 13% 10% 6% 11% 13% 10% 10% 8% [libx264 @ 0000000001E61350] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 24% 23% 10% 5% 11% 10% 8% 5% 4% [libx264 @ 0000000001E61350] i8c dc,h,v,p: 52% 17% 23% 9% [libx264 @ 0000000001E61350] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0000000001E61350] ref P L0: 59.5% 13.0% 17.9% 9.6% [libx264 @ 0000000001E61350] ref B L0: 87.1% 12.1% 0.7% [libx264 @ 0000000001E61350] ref B L1: 98.0% 2.0% [libx264 @ 0000000001E61350] kb/s:2372.67 Output is attached as well if that makes it easier to read. TIA! -------------- next part -------------- C:\alka-video>ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 48000 -i c:/alka-video/workspace/06A.cat.a -i c:/alka-video/workspace/06A.cat.v -vcodec libx264 -same_quant -y c:\alka-video/output/06A.avi > _h264.txt ffmpeg version N-38622-g1eabd71 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 7 2012 00:21:47 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 42.100 / 51. 42.100 libavcodec 54. 10.100 / 54. 10.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 63.100 / 2. 63.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 [u16le @ 000000000031F3C0] Estimating duration from bitrate, this may be inaccurate Input #0, u16le, from 'c:/alka-video/workspace/06A.cat.a': Duration: 00:00:03.79, start: 0.000000, bitrate: 1536 kb/s Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s [yuv4mpegpipe @ 0000000001E4F8C0] Estimating duration from bitrate, this may be inaccurate Input #1, yuv4mpegpipe, from 'c:/alka-video/workspace/06A.cat.v': Duration: N/A, bitrate: N/A Stream #1:0: Video: rawvideo (Y42B / 0x42323459), yuv422p, 1280x720, 23.98 fps, 23.98 tbr, 23.98 tbn, 23.98 tbc [buffer @ 0000000001E617D0] w:1280 h:720 pixfmt:yuv422p tb:1/1000000 sar:0/1 sws_param: [libx264 @ 0000000001E61350] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.1 Cache64 [libx264 @ 0000000001E61350] profile High 4:2:2, level 3.1, 4:2:2 8-bit Output #0, avi, to 'c:\alka-video/output/06A.avi': Metadata: ISFT : Lavf54.2.100 Stream #0:0: Video: h264 (H264 / 0x34363248), yuv422p, 1280x720, q=-1--1, 23.98 tbn, 23.98 tbc Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 48000 Hz, 2 channels, s16 Stream mapping: Stream #1:0 -> #0:0 (rawvideo -> libx264) Stream #0:0 -> #0:1 (pcm_s16le -> libmp3lame) Press [q] to stop, [?] for help Truncating packet of size 4096 to 3737 Truncating packet of size 4096 to 1 frame= 91 fps= 7 q=28.0 Lsize= 1187kB time=00:00:03.71 bitrate=2619.9kbits/s video:1099kB audio:72kB global headers:0kB muxing overhead 1.332733% [libx264 @ 0000000001E61350] frame I:2 Avg QP:22.39 size: 44095 [libx264 @ 0000000001E61350] frame P:51 Avg QP:24.42 size: 16121 [libx264 @ 0000000001E61350] frame B:38 Avg QP:27.17 size: 5667 [libx264 @ 0000000001E61350] consecutive B-frames: 22.0% 61.5% 16.5% 0.0% [libx264 @ 0000000001E61350] mb I I16..4: 11.8% 76.5% 11.7% [libx264 @ 0000000001E61350] mb P I16..4: 1.4% 7.2% 0.5% P16..4: 50.1% 16.1% 7.5% 0.0% 0.0% skip:17.1% [libx264 @ 0000000001E61350] mb B I16..4: 0.1% 1.1% 0.1% B16..8: 41.9% 4.9% 0.9% direct: 2.3% skip:48.8% L0:36.4% L1:56.4% BI: 7.2% [libx264 @ 0000000001E61350] 8x8 transform intra:78.7% inter:84.4% [libx264 @ 0000000001E61350] coded y,uvDC,uvAC intra: 64.4% 79.3% 10.1% inter: 18.2% 37.1% 0.1% [libx264 @ 0000000001E61350] i16 v,h,dc,p: 21% 28% 4% 47% [libx264 @ 0000000001E61350] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 19% 13% 10% 6% 11% 13% 10% 10% 8% [libx264 @ 0000000001E61350] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 24% 23% 10% 5% 11% 10% 8% 5% 4% [libx264 @ 0000000001E61350] i8c dc,h,v,p: 52% 17% 23% 9% [libx264 @ 0000000001E61350] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0000000001E61350] ref P L0: 59.5% 13.0% 17.9% 9.6% [libx264 @ 0000000001E61350] ref B L0: 87.1% 12.1% 0.7% [libx264 @ 0000000001E61350] ref B L1: 98.0% 2.0% [libx264 @ 0000000001E61350] kb/s:2372.67 From andrey.krieger.utkin at gmail.com Thu Mar 22 13:31:34 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Thu, 22 Mar 2012 14:31:34 +0200 Subject: [FFmpeg-user] Trouble encoding h.264 In-Reply-To: References: Message-ID: 2012/3/22 Langdon : > I have some raw video (-f yuv4mpegpipe) and audio (-f u16le -acodec > pcm_s16le) that I extracted from another file. > > When I try to re-encode the two parts as h.264, the sound plays OK, but the > video is mostly gray in Media Player Classic Home Cinema, as well as when > uploaded to YouTube. ?ffplay.exe seems to play it OK, but I'm creating the > video for mass consumption. > > Am I doing something wrong? ?I had to tell ffmpeg about every attribute of > the audio before it would work right, but I'm not sure what to tell it > about the video. > > ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 48000 -i > c:/alka-video/workspace/06A.cat.a -i c:/alka-video/workspace/06A.cat.v > -vcodec libx264 -same_quant -y c:\alka-video/output/06A.avi > _h264.txt Try dropping -same_quant option, add -b:v (video bitrate here), or other options for quality setting. -- Andrey Utkin From langdon at gmail.com Thu Mar 22 14:21:20 2012 From: langdon at gmail.com (Langdon) Date: Thu, 22 Mar 2012 09:21:20 -0400 Subject: [FFmpeg-user] Trouble encoding h.264 In-Reply-To: References: Message-ID: > 2012/3/22 Langdon : > > When I try to re-encode the two parts as h.264, the sound plays OK, but the > > video is mostly gray in Media Player Classic Home Cinema, as well as when > > uploaded to YouTube. ?ffplay.exe seems to play it OK, but I'm creating the > > video for mass consumption. > > > > ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 48000 -i > > c:/alka-video/workspace/06A.cat.a -i c:/alka-video/workspace/06A.cat.v > > -vcodec libx264 -same_quant -y c:\alka-video/output/06A.avi > _h264.txt > > Try dropping -same_quant option, add -b:v (video bitrate here), or > other options for quality setting. I tried various -b:v values (1000k, 10000k, and 81000k), and while all still mostly gray, some showed different variations of artifacts I then ran ffprobe on the source video to see what I could glean from it... Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'source/06A.mov': Metadata: creation_time : 2012-03-20 13:18:46 Duration: 00:00:02.25, start: 0.000000, bitrate: 84045 kb/s Stream #0:0(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, 2 channels, s16, 1536 kb/s Metadata: creation_time : 2012-03-20 13:18:47 handler_name : ?Apple Alias Data Handler Stream #0:1(eng): Video: prores (apch / 0x68637061), yuv422p10le, 1280x720, 81541 kb/s, 23.98 fps, 23.98 tbr, 23976 tbn, 23976 tbc Metadata: creation_time : 2012-03-20 13:18:47 handler_name : ?Apple Alias Data Handler Stream #0:2(eng): Data: none (tmcd / 0x64636D74) Metadata: creation_time : 2012-03-20 13:18:49 handler_name : ?Apple Alias Data Handler timecode : 01:00:32:21 Unsupported codec with id 0 for input stream 2 So I tried -b:v 81541k, to no avail. Interestingly enough, the source video converts to h.264 just fine (from ProRes), but I can't concatenate ProRes. =[ I don't see anything in -help to specify tbn or tbc (the only 2 values that different from input to output at this point). From cehoyos at ag.or.at Thu Mar 22 14:37:24 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 22 Mar 2012 13:37:24 +0000 (UTC) Subject: [FFmpeg-user] Trouble encoding h.264 References: Message-ID: Langdon gmail.com> writes: > Stream #1:0: Video: rawvideo (Y42B / 0x42323459), yuv422p Your input is yuv422p, you do not specify another colourspace, > Stream #0:0: Video: h264 (H264 / 0x34363248), yuv422p libx264 was compiled with yuv422p support, so ffmpeg encodes to h264 4:2:2 which is not supported by most playback applications. Add -pix_fmt yuv420p. Carl Eugen From miaou.pbl at gmail.com Thu Mar 22 15:04:28 2012 From: miaou.pbl at gmail.com (A BC) Date: Thu, 22 Mar 2012 15:04:28 +0100 Subject: [FFmpeg-user] png, pipe, image2pipe Message-ID: Hi, I searched on the list and found helpful information about what I want to do but I can't make it work. I try to pipe into ffmpeg a quantity of png files. I am working on making it work with a simple example. So, what I would do with a single file "Anim0001.png" is : ffmpeg -y -f image2 -i Anim0001.png -vcodec libx264 -filter pad=1080:814 Anim1.mkv wich works fine and make a single frame video. To do it with a pipe : cat Anim0001.png | ffmpeg -y -f image2pipe -i - -vcodec libx264 -filter pad=1080:814 Anim1.mkv wich does not work (classic "Could not find codec parameter (Video: none) So I put a "-vcodec png" before "-i -": cat Anim0001.png | ffmpeg -y -f image2pipe -vcodec png -i - -vcodec libx264 -filter pad=1080:814 Anim1.mkv wich fails (output below). So I thought, maybe "-vcodec png" does not work correctly, and I tried to do the first one again, with -vcodec png: ffmpeg -y -f image2 -vcodec png -i Anim0001.png -vcodec libx264 -filter pad=1080:814 Anim1.mkv which works fine too ! (I use the padding because my image is 1080x813 so I round for x264) Here's what does not work : cat Anim0001.png | ffmpeg -y -f image2pipe -vcodec png -i - -vcodec libx264 -filter pad="1080:814" Anim1.slow.mkv And the stderr : ffmpeg version git-2012-03-22-eb98412 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 22 2012 12:45:00 with gcc 4.4.3 configuration: --enable-gpl --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-version3 --enable-x11grab libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 65.102 / 2. 65.102 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 [png @ 0xa1792a0] chunk too big [png @ 0xa1792a0] Missing png signature Last message repeated 13 times [image2pipe @ 0xa16ab60] decoding for stream 0 failed [image2pipe @ 0xa16ab60] Could not find codec parameters (Video: png) [image2pipe @ 0xa16ab60] Estimating duration from bitrate, this may be inaccurate pipe:: could not find codec parameters For information, I used this page to compile ffmpeg (I work on Ubuntu 10.04) http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 Do you know how I can make it work ? Thanks for your help ! From langdon at gmail.com Thu Mar 22 15:41:50 2012 From: langdon at gmail.com (Langdon) Date: Thu, 22 Mar 2012 10:41:50 -0400 Subject: [FFmpeg-user] Trouble encoding h.264 In-Reply-To: References: Message-ID: > Your input is yuv422p, you do not specify another colourspace, > > libx264 was compiled with yuv422p support, > so ffmpeg encodes to h264 4:2:2 which is not supported > by most playback applications. > > Add -pix_fmt yuv420p. > > Carl Eugen Impressive, that did the trick! Thank you! It's not a huge deal, but I switched from MP3 to AAC and again, WMP choked on it (played just a portion of the audio, and sped up the video 2-3x), while ffplay and YouTube were able to handle it just fine. I switched the container from AVI to MKV, and that played fine in WMP-HC. It seemed like the only sample_fmt that AAC supports is flt... Do most playback options not support aac/flt in the avi container as well? Or is there something else I can tweak? I have latest CCCP installed, if that makes a difference... * Plays Well * > ffmpeg -strict experimental -f u16le -acodec pcm_s16le -ac 2 -ar 48000 -i c:/alka-video/workspace/06A.cat.a -i c:/alka-video/workspace/06A.cat.v -strict experimental -acodec aac -vcodec libx264 -same_quant -sample_fmt s32 -pix_fmt yuv420p -ac 2 -ar 48000 -y c:\alka-video/output/06A.mkv ffmpeg version N-38622-g1eabd71 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 7 2012 00:21:47 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libope ncore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger - -enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable- libxvid --enable-zlib libavutil 51. 42.100 / 51. 42.100 libavcodec 54. 10.100 / 54. 10.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 63.100 / 2. 63.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 [u16le @ 00000000002E0200] Estimating duration from bitrate, this may be inaccurate Input #0, u16le, from 'c:/alka-video/workspace/06A.cat.a': Duration: 00:00:03.79, start: 0.000000, bitrate: 1536 kb/s Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s [yuv4mpegpipe @ 00000000002F05E0] Estimating duration from bitrate, this may be inaccurate Input #1, yuv4mpegpipe, from 'c:/alka-video/workspace/06A.cat.v': Duration: N/A, bitrate: N/A Stream #1:0: Video: rawvideo (Y42B / 0x42323459), yuv422p, 1280x720, 23.98 fps, 23.98 tbr, 23.98 tbn, 23.98 tbc [buffer @ 0000000000302860] w:1280 h:720 pixfmt:yuv422p tb:1/1000000 sar:0/1 sws_param: [buffersink @ 000000000030D830] auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'out' [scale @ 000000000030DA00] w:1280 h:720 fmt:yuv422p -> w:1280 h:720 fmt:yuv420p flags:0x4 Incompatible sample format 's32' for codec 'aac', auto-selecting format 'flt' [libx264 @ 00000000003022F0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.1 Cache64 [libx264 @ 00000000003022F0] profile High, level 3.1 [libx264 @ 00000000003022F0] 264 - core 120 r2146 bcd41db - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 r ef=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pski p=1 chroma_qp_offset=-2 threads=3 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 dir ect=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=23 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 q pstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, matroska, to 'c:\alka-video/output/06A.mkv': Metadata: encoder : Lavf54.2.100 Stream #0:0: Video: h264, yuv420p, 1280x720, q=-1--1, 1k tbn, 23.98 tbc Stream #0:1: Audio: aac, 48000 Hz, 2 channels, flt, 128 kb/s Stream mapping: Stream #1:0 -> #0:0 (rawvideo -> libx264) Stream #0:0 -> #0:1 (pcm_s16le -> aac) Press [q] to stop, [?] for help Truncating packet of size 4096 to 373777kB time=00:00:01.70 bitrate=2767.2kbits/s Truncating packet of size 4096 to 1 frame= 91 fps= 10 q=28.0 Lsize= 1029kB time=00:00:03.71 bitrate=2271.5kbits/s video:961kB audio:65kB global headers:0kB muxing overhead 0.244409% [libx264 @ 00000000003022F0] frame I:2 Avg QP:22.31 size: 39945 [libx264 @ 00000000003022F0] frame P:51 Avg QP:24.38 size: 13822 [libx264 @ 00000000003022F0] frame B:38 Avg QP:27.25 size: 5233 [libx264 @ 00000000003022F0] consecutive B-frames: 22.0% 61.5% 16.5% 0.0% [libx264 @ 00000000003022F0] mb I I16..4: 11.0% 76.7% 12.3% [libx264 @ 00000000003022F0] mb P I16..4: 1.2% 7.0% 0.5% P16..4: 45.7% 15.0% 6.8% 0.0% 0.0% skip:23.8% [libx264 @ 00000000003022F0] mb B I16..4: 0.1% 1.0% 0.1% B16..8: 43.0% 5.0% 0.9% direct: 1.8% skip:48.1% L0:35.8% L1:57.4% BI: 6.8% [libx264 @ 00000000003022F0] 8x8 transform intra:79.3% inter:84.1% [libx264 @ 00000000003022F0] coded y,uvDC,uvAC intra: 65.5% 61.3% 10.5% inter: 17.8% 18.8% 0.1% [libx264 @ 00000000003022F0] i16 v,h,dc,p: 20% 31% 3% 47% [libx264 @ 00000000003022F0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 20% 12% 10% 7% 11% 13% 10% 9% 8% [libx264 @ 00000000003022F0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 23% 11% 5% 12% 10% 8% 6% 4% [libx264 @ 00000000003022F0] i8c dc,h,v,p: 55% 22% 18% 5% [libx264 @ 00000000003022F0] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 00000000003022F0] ref P L0: 61.5% 14.4% 15.6% 8.4% [libx264 @ 00000000003022F0] ref B L0: 87.5% 11.8% 0.8% [libx264 @ 00000000003022F0] ref B L1: 98.0% 2.0% [libx264 @ 00000000003022F0] kb/s:2073.35 * Only hear a portion of the audio, and video plays too fast in WMP-HC * >ffmpeg -strict experimental -f u16le -acodec pcm_s16le -ac 2 -ar 48000 -i c:/alka-video/workspace/06A.cat.a -i c:/alka-video/workspace/06A.cat.v -strict experimental -acodec aac -vcodec libx264 -same_quant -sample_fmt s32 -pix_fmt yuv420p -ac 2 -ar 48000 -y c:\alka-video/output/06A.avi ffmpeg version N-38622-g1eabd71 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 7 2012 00:21:47 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libope ncore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger - -enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable- libxvid --enable-zlib libavutil 51. 42.100 / 51. 42.100 libavcodec 54. 10.100 / 54. 10.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 63.100 / 2. 63.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 [u16le @ 00000000002E0200] Estimating duration from bitrate, this may be inaccurate Input #0, u16le, from 'c:/alka-video/workspace/06A.cat.a': Duration: 00:00:03.79, start: 0.000000, bitrate: 1536 kb/s Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s [yuv4mpegpipe @ 00000000002F05E0] Estimating duration from bitrate, this may be inaccurate Input #1, yuv4mpegpipe, from 'c:/alka-video/workspace/06A.cat.v': Duration: N/A, bitrate: N/A Stream #1:0: Video: rawvideo (Y42B / 0x42323459), yuv422p, 1280x720, 23.98 fps, 23.98 tbr, 23.98 tbn, 23.98 tbc [buffer @ 00000000003027B0] w:1280 h:720 pixfmt:yuv422p tb:1/1000000 sar:0/1 sws_param: [buffersink @ 00000000003029B0] auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'out' [scale @ 000000000030D960] w:1280 h:720 fmt:yuv422p -> w:1280 h:720 fmt:yuv420p flags:0x4 Incompatible sample format 's32' for codec 'aac', auto-selecting format 'flt' [libx264 @ 00000000003022F0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.1 Cache64 [libx264 @ 00000000003022F0] profile High, level 3.1 Output #0, avi, to 'c:\alka-video/output/06A.avi': Metadata: ISFT : Lavf54.2.100 Stream #0:0: Video: h264 (H264 / 0x34363248), yuv420p, 1280x720, q=-1--1, 23.98 tbn, 23.98 tbc Stream #0:1: Audio: aac ([255][0][0][0] / 0x00FF), 48000 Hz, 2 channels, flt, 128 kb/s Stream mapping: Stream #1:0 -> #0:0 (rawvideo -> libx264) Stream #0:0 -> #0:1 (pcm_s16le -> aac) Press [q] to stop, [?] for help Truncating packet of size 4096 to 373708kB time=00:00:01.70 bitrate=2918.1kbits/s Truncating packet of size 4096 to 1 frame= 91 fps= 9 q=28.0 Lsize= 1043kB time=00:00:03.71 bitrate=2301.0kbits/s video:961kB audio:65kB global headers:0kB muxing overhead 1.572291% [libx264 @ 00000000003022F0] frame I:2 Avg QP:22.31 size: 40316 [libx264 @ 00000000003022F0] frame P:51 Avg QP:24.38 size: 13822 [libx264 @ 00000000003022F0] frame B:38 Avg QP:27.25 size: 5233 [libx264 @ 00000000003022F0] consecutive B-frames: 22.0% 61.5% 16.5% 0.0% [libx264 @ 00000000003022F0] mb I I16..4: 11.0% 76.7% 12.3% [libx264 @ 00000000003022F0] mb P I16..4: 1.2% 7.0% 0.5% P16..4: 45.7% 15.0% 6.8% 0.0% 0.0% skip:23.8% [libx264 @ 00000000003022F0] mb B I16..4: 0.1% 1.0% 0.1% B16..8: 43.0% 5.0% 0.9% direct: 1.8% skip:48.1% L0:35.8% L1:57.4% BI: 6.8% [libx264 @ 00000000003022F0] 8x8 transform intra:79.3% inter:84.1% [libx264 @ 00000000003022F0] coded y,uvDC,uvAC intra: 65.5% 61.3% 10.5% inter: 17.8% 18.8% 0.1% [libx264 @ 00000000003022F0] i16 v,h,dc,p: 20% 31% 3% 47% [libx264 @ 00000000003022F0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 20% 12% 10% 7% 11% 13% 10% 9% 8% [libx264 @ 00000000003022F0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 23% 11% 5% 12% 10% 8% 6% 4% [libx264 @ 00000000003022F0] i8c dc,h,v,p: 55% 22% 18% 5% [libx264 @ 00000000003022F0] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 00000000003022F0] ref P L0: 61.5% 14.4% 15.6% 8.4% [libx264 @ 00000000003022F0] ref B L0: 87.5% 11.8% 0.8% [libx264 @ 00000000003022F0] ref B L1: 98.0% 2.0% [libx264 @ 00000000003022F0] kb/s:2074.91 From programmingkidx at gmail.com Thu Mar 22 15:52:28 2012 From: programmingkidx at gmail.com (Programmingkid) Date: Thu, 22 Mar 2012 10:52:28 -0400 Subject: [FFmpeg-user] How to make RealPlayer files Message-ID: <50955CC6-72B6-43AF-BAC7-ADB29EF4A8CC@gmail.com> Hi, I'm trying to make a RealPlayer video file that is compatible with RealPlayer 6. Here is the command I used: ffmpeg-0.5 -i file -s 176x128 -r 4 -ar 8000 -ac 1 -b 30k ~/desktop/test9.rm. The resulting file works only briefly until the player just stops only half a minute into the file. Does anyone know how to make a more compatible rm file? From rhodri at kynesim.co.uk Thu Mar 22 16:02:04 2012 From: rhodri at kynesim.co.uk (Rhodri James) Date: Thu, 22 Mar 2012 15:02:04 -0000 Subject: [FFmpeg-user] How to make RealPlayer files In-Reply-To: <50955CC6-72B6-43AF-BAC7-ADB29EF4A8CC@gmail.com> References: <50955CC6-72B6-43AF-BAC7-ADB29EF4A8CC@gmail.com> Message-ID: On Thu, 22 Mar 2012 14:52:28 -0000, Programmingkid wrote: > Hi, I'm trying to make a RealPlayer video file that is compatible with > RealPlayer 6. Here is the command I used: > ffmpeg-0.5 -i file -s 176x128 -r 4 -ar 8000 -ac 1 -b 30k > ~/desktop/test9.rm. > The resulting file works only briefly until the player just stops only > half a minute into the file. Does anyone know how to make a more > compatible rm file? It would help if you provided the complete, uncut output of the command, but I would hazard a guess from the "ffmpeg-0.5" that you are using an ancient version of ffmpeg. Have you tried using the current version off git head? -- Rhodri James Kynesim Ltd From cehoyos at ag.or.at Thu Mar 22 16:59:01 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 22 Mar 2012 15:59:01 +0000 (UTC) Subject: [FFmpeg-user] Trouble encoding h.264 References: Message-ID: Langdon gmail.com> writes: > ffmpeg -strict experimental -f u16le -acodec pcm_s16le -ac 2 > -ar 48000 -i c:/alka-video/workspace/06A.cat.a -i > c:/alka-video/workspace/06A.cat.v -strict experimental > -acodec aac -vcodec libx264 -same_quant -sample_fmt s32 > -pix_fmt yuv420p -ac 2 -ar 48000 -y c:\alka-video/output/06A.avi I find it surprising that you try to set sample_fmt (which should probably fail, but just has no effect) and channels and rate (which you do not change wrt input and input is the default for both), but you do not try to set the bitrate which I would try. Are you sure aac in avi is supported by WMP? I am not saying it isn't I am just surprised. otoh, if you don't use wmp, but another application: How do you know it works ok? Finally, please note that you had to use -strict experimental which basically tells you that you cannot expect the encoder to work perfectly... Carl Eugen From baptiste.coudurier at gmail.com Thu Mar 22 17:02:48 2012 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Thu, 22 Mar 2012 09:02:48 -0700 Subject: [FFmpeg-user] ProRes (LT) Quicktimes: Good for Windows. Not so good for Mac. In-Reply-To: References: <4F68F4F9.3040504@gmx.de> Message-ID: <4F6B4D28.8050409@gmail.com> Hi Thomas, On 03/20/2012 02:46 PM, Thomas Worth wrote: > On Tue, Mar 20, 2012 at 2:22 PM, Andreas Gumm wrote: >> Yes, this is a known issue! Your files has been correct encoded! FCP just >> complains sometimes about external encoded ProRes footage! >> It seem to be that some missing header informations or other stupid missing >> informations let FCP think the file looks not compatible! > > About this, I think the Final Cut media warning may be related to the > A/V interleaving, i.e. the way the chunk offset tables are written. I > don't think it has as much to do with information in the QuickTime > header. Files written without audio don't show the warning. Baptiste > Coudurier might comment on this. If anyone knows why this is > happening, is would be him. Yes, this is because FCP expects audio chunks to be approximately 48000 samples. -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From cehoyos at ag.or.at Thu Mar 22 17:01:50 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 22 Mar 2012 16:01:50 +0000 (UTC) Subject: [FFmpeg-user] png, pipe, image2pipe References: Message-ID: A BC gmail.com> writes: > I searched on the list and found helpful information about what > I want to do but I can't make it work. > I try to pipe into ffmpeg a quantity of png files. I am working > on making it work with a simple example. > Do you know how I can make it work ? Since it works for (very) small image dimensions, I suspect you have to increase (strongly) the input buffer used by FFmpeg, unfortunately, I don't know offhand where to change the size. Carl Eugen From osoleil at ubisoft.fr Thu Mar 22 16:51:09 2012 From: osoleil at ubisoft.fr (Olivier Soleil) Date: Thu, 22 Mar 2012 08:51:09 -0700 (PDT) Subject: [FFmpeg-user] How can I replace numbered sound tracks in my AVI ? In-Reply-To: <524EA12FEC145E4E86154A23DA429D2D5DE588AFD3@PDC-MAIL-CMS01.ubisoft.org> References: <524EA12FEC145E4E86154A23DA429D2D5DE588AFD3@PDC-MAIL-CMS01.ubisoft.org> Message-ID: <1332431469877-4495903.post@n4.nabble.com> Doesn't anyone have any idea about these issues ? I had no answer yet, but i'd like to remind you the main things i miss. *First question*, is there an easy way for the user to mux an audio stream to a desired stream number in the destination file. e.g. : I want to mux one soundtrack to the 100th stream in my video. Does it exist ? Is there any chance that this feature can be developed ? *Second question*, is there a way for FFMPEG to modify an already existing multimedia file ? In other words : can FFMPEG *not* create a file from scratch when adding/muxing/converting anything ? Thank you all for your attention. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-can-I-replace-numbered-sound-tracks-in-my-AVI-tp4469444p4495903.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From andreas.gumm at gmx.de Thu Mar 22 17:16:13 2012 From: andreas.gumm at gmx.de (Andreas Gumm) Date: Thu, 22 Mar 2012 17:16:13 +0100 Subject: [FFmpeg-user] ProRes (LT) Quicktimes: Good for Windows. Not so good for Mac. In-Reply-To: <4F6B5009.3090009@googlemail.com> References: <4F6B5009.3090009@googlemail.com> Message-ID: <4F6B504D.5010002@gmx.de> Can I define an argument in command line to get 48000 samples per audio chunk? Thank you for answering in advance! Andreas Am 22.03.2012 17:02, schrieb Baptiste Coudurier: > Hi Thomas, > > On 03/20/2012 02:46 PM, Thomas Worth wrote: >> On Tue, Mar 20, 2012 at 2:22 PM, Andreas Gumm wrote: >>> Yes, this is a known issue! Your files has been correct encoded! FCP just >>> complains sometimes about external encoded ProRes footage! >>> It seem to be that some missing header informations or other stupid missing >>> informations let FCP think the file looks not compatible! >> About this, I think the Final Cut media warning may be related to the >> A/V interleaving, i.e. the way the chunk offset tables are written. I >> don't think it has as much to do with information in the QuickTime >> header. Files written without audio don't show the warning. Baptiste >> Coudurier might comment on this. If anyone knows why this is >> happening, is would be him. > Yes, this is because FCP expects audio chunks to be approximately > 48000 samples. > From tom1.alter at gmail.com Thu Mar 22 18:04:45 2012 From: tom1.alter at gmail.com (Tom Alter) Date: Thu, 22 Mar 2012 22:34:45 +0530 Subject: [FFmpeg-user] Problem using H263 Encoder Message-ID: Hi All, I am using ffmpeg h263 encoder to encode the live stream and sending the encoded stream to peer over RTP. Solution works one start but slowly video quality gets degraded with time. After two to three hours video gets blurred. I have tweaked with various settings of H263 Encoder but did not able to fix the problem. Is anyone there who can suggest the reason of this behaviour? Any help would be greatly appreciated. Thanks From ban.an at list.ru Thu Mar 22 13:11:23 2012 From: ban.an at list.ru (iiser) Date: Thu, 22 Mar 2012 05:11:23 -0700 (PDT) Subject: [FFmpeg-user] Unable to demux 5.1 DVD-Audio rips Message-ID: <1332418283977-4495306.post@n4.nabble.com> Hello. I have a couple of DVD-Audio rips: one contains 44.1kHz/24-bit 5.1-channel MLP track and another one has 48kHz/24-bit 5.1-channel MLP track. For some reason ffmpeg is extracting only 10kB and 120kB files respectively from 1GB input. Am I doing something wrong? DVD-Audio Explorer and VLC do not have any problems with these files. Please see ffmpeg output below. ==================================== 44.1kHz/24-bit 5.1-channel: ffmpeg -i ATS_01_1.AOB -acodec copy -f mlp test.mlp ffmpeg version N-38938-ge01f478 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 19 2012 23:16:52 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 42.100 / 51. 42.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 65.101 / 2. 65.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, mpeg, from 'ATS_01_1.AOB': Duration: 00:29:45.39, start: 0.001167, bitrate: 4811 kb/s Stream #0:0[0xa1]: Audio: pcm_s16be, 96000 Hz, 8 channels, s16, 12288 kb/s Output #0, mlp, to 'test.mlp': Metadata: encoder : Lavf54.2.100 Stream #0:0: Audio: pcm_s16be, 96000 Hz, 8 channels, 12288 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help size= 10kB time=00:00:00.02 bitrate=3342.7kbits/s video:0kB audio:10kB global headers:0kB muxing overhead 0.000000% ==================================== 48kHz/24-bit 5.1-channel: ffmpeg -i ATS_01_1.AOB -acodec copy -f mlp test.mlp ffmpeg version N-38938-ge01f478 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 19 2012 23:16:52 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 42.100 / 51. 42.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 65.101 / 2. 65.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, mpeg, from 'ATS_01_1.AOB': Duration: 00:32:28.35, start: 0.007333, bitrate: 4408 kb/s Stream #0:0[0xa1]: Audio: pcm_s16be, 96000 Hz, 6 channels, s16, 9216 kb/s Output #0, mlp, to 'test.mlp': Metadata: encoder : Lavf54.2.100 Stream #0:0: Audio: pcm_s16be, 96000 Hz, 6 channels, 9216 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help size= 120kB time=00:00:02.62 bitrate= 373.9kbits/s video:0kB audio:120kB global headers:0kB muxing overhead 0.000000% ==================================== -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Unable-to-demux-5-1-DVD-Audio-rips-tp4495306p4495306.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From rhodri at kynesim.co.uk Thu Mar 22 18:34:42 2012 From: rhodri at kynesim.co.uk (Rhodri James) Date: Thu, 22 Mar 2012 17:34:42 -0000 Subject: [FFmpeg-user] Problem using H263 Encoder In-Reply-To: References: Message-ID: On Thu, 22 Mar 2012 17:04:45 -0000, Tom Alter wrote: > Hi All, > > I am using ffmpeg h263 encoder to encode the live stream and sending the > encoded stream to peer over RTP. Solution works one start but slowly > video > quality gets degraded with time. After two to three hours video gets > blurred. I have tweaked with various settings of H263 Encoder but did not > able to fix the problem. All together now: command line and complete, uncut output missing :-) > Is anyone there who can suggest the reason of this behaviour? Any help > would be greatly appreciated. It sounds like you are seeing small errors accumulate over a long time, which suggests that the encoder isn't producing I-frames often enough (or at all). It's impossible to tell without more information, but some combination of your target bit rate and encoder parameters may be, er, optimistic. -- Rhodri James Kynesim Ltd From joddo at jerfu.com Thu Mar 22 19:14:06 2012 From: joddo at jerfu.com (Jeremy Oddo) Date: Thu, 22 Mar 2012 11:14:06 -0700 Subject: [FFmpeg-user] ProRes (LT) Quicktimes: Good for Windows. Not so good for Mac. In-Reply-To: <4F6B504D.5010002@gmx.de> References: <4F6B5009.3090009@googlemail.com> <4F6B504D.5010002@gmx.de> Message-ID: >>> ?About this, I think the Final Cut media warning may be related to the >>> ?A/V interleaving, i.e. the way the chunk offset tables are written. I [...SNIP...] >> ?Yes, this is because FCP expects audio chunks to be approximately >> ?48000 samples. >> Could you get around this issue by supplying a null stream for the audio and setting -ar 48000? I can't test the /dev/zero because I'm on Windows and I don't know if there is a /dev/zero equivalent (I tried null w/o luck). From dev at rarevision.com Thu Mar 22 23:31:02 2012 From: dev at rarevision.com (Thomas Worth) Date: Thu, 22 Mar 2012 15:31:02 -0700 Subject: [FFmpeg-user] ProRes (LT) Quicktimes: Good for Windows. Not so good for Mac. In-Reply-To: References: <4F68F4F9.3040504@gmx.de> Message-ID: >> Also it should be noted that the media performance warning does not >> occur when QuickTime files are written with a co64 atom. So, Final Cut >> apparently likes FFmpeg MOV files written with 64 bit offsets. > > Could you explain the "co64 atom" thing a bit? I don't know what co64 atom > is. Is it possible for FFMPEG to write ProRes QTs with a co64 atom or > is that related to some other codec? The co64 atom is only used when the file is over 4GB. The reason is because an unsigned 32 bit value (e.g. uint32_t, which is 4 bytes) simply isn't big enough to hold a number over 4294967295. Sometimes the file is so big that the position of data relative to the beginning of the file goes past 4294967295, so without a 64 bit value (i.e. int64_t, which is 8 bytes), the file reader hits the 4GB limit and doesn't have any more numbers to seek over 4GB. This is a common limitation in computing in general and affects everything, most notably the FAT32 filesystem. >> Someone might take a gander at movenc.c and see if interleaving is >> done differently with 64 bit MOV offsets vs. 32 bit ones. > > Does this 32/64-bit offsets related to the OS or can a 64-bit MOV offset > be written from a 32-bit OS? Don't know if it matters but we're using the > 64-bit FFMPEG. The way the program is compiled doesn't have anything to do with it. A 32 bit ffmpeg binary can still read/write 64 bit QuickTime files. If it couldn't, then the file size would be limited to 4GB. >>> ?Yes, this is because FCP expects audio chunks to be approximately >>> ?48000 samples. >>> > > Could you get around this issue by supplying a null stream for the audio > and setting -ar 48000? I can't test the /dev/zero because I'm on Windows > and I don't know if there is a /dev/zero equivalent (I tried null w/o luck). This doesn't work. The problem is with movenc.c, and possibly other source files. It should also be pointed out that MOV/MP4 elst atoms aren't written properly in many cases, at least not in a way that Apple's software expects them. For example, sometimes the start value in elst will be wrong if performing stream copies with B-frame'd H.264. The fix is to go into a hex editor and change the start value to 1 or 1001, depending on the PTS duration. ffmbc isn't affected by this since Baptiste has taken the time to ensure the MOV/MP4 writer works in a way that Apple software expects. IMHO, someone should follow ffmbc's good example and fix the MOV muxer in ffmpeg, at least to the point where it addresses these issues. It'd be a pretty hard sell to convince me that writing MOVs/MP4s that are incompatible with software from the same company that invented the format in the first place is acceptable. Not to bash the effort that's been put into this, of course. avformat is a fabulous piece of software, it just could be a LITTLE better... :-) Baptiste, what would it take to fix these issues in ffmpeg? Would the changes be limited to movenc or would changes elsewhere need to be made as well? I would gladly fix this myself and submit patches, but unfortunately I don't know avformat well enough to not utterly break it to oblivion! From miaou.pbl at gmail.com Fri Mar 23 07:59:02 2012 From: miaou.pbl at gmail.com (A BC) Date: Fri, 23 Mar 2012 07:59:02 +0100 Subject: [FFmpeg-user] png, pipe, image2pipe In-Reply-To: References: Message-ID: On Thu, Mar 22, 2012 at 5:01 PM, Carl Eugen Hoyos wrote: > A BC gmail.com> writes: > > > I try to pipe into ffmpeg a quantity of png files. I am working > > on making it work with a simple example. > > > Do you know how I can make it work ? > > Since it works for (very) small image dimensions, I suspect you > have to increase (strongly) the input buffer used by FFmpeg, > unfortunately, I don't know offhand where to change the size. > > Carl Eugen > Thanks for your quick answer ! I will search in that direction. Do you think I may have to ask on the dev list and make a custom build ? Thanks again. From miaou.pbl at gmail.com Fri Mar 23 11:13:47 2012 From: miaou.pbl at gmail.com (A BC) Date: Fri, 23 Mar 2012 11:13:47 +0100 Subject: [FFmpeg-user] png, pipe, image2pipe In-Reply-To: References: Message-ID: Hi, Some people opened an issue (http://roundup.libav.org/issue1854) about the same problem, but links to the patch is dead. I have found an "old" conversation, and finally found a permalink to the most recent patch ( http://cache.gmane.org//gmane/comp/video/ffmpeg/devel/94083-001.bin found on page http://permalink.gmane.org/gmane.comp.video.ffmpeg.devel/94083). I tried to apply it roughly with "git apply png-patch.diff", (adding first a final '\n'), but the patch is not applicable anymore. Do you know what I can do from now on ? Thanks On Fri, Mar 23, 2012 at 7:59 AM, A BC wrote: > On Thu, Mar 22, 2012 at 5:01 PM, Carl Eugen Hoyos wrote: > >> A BC gmail.com> writes: >> >> > I try to pipe into ffmpeg a quantity of png files. I am working >> > on making it work with a simple example. >> >> > Do you know how I can make it work ? >> >> Since it works for (very) small image dimensions, I suspect you >> have to increase (strongly) the input buffer used by FFmpeg, >> unfortunately, I don't know offhand where to change the size. >> >> Carl Eugen >> > > > Thanks for your quick answer ! I will search in that direction. Do you > think I may have to ask on the dev list and make a custom build ? > > Thanks again. > From tom1.alter at gmail.com Fri Mar 23 13:28:58 2012 From: tom1.alter at gmail.com (Tom Alter) Date: Fri, 23 Mar 2012 17:58:58 +0530 Subject: [FFmpeg-user] Problem using H263 Encoder In-Reply-To: References: Message-ID: Thanks Rhodri for your response.. I am setting following properties of Encoder: bitrate: 150 kpbs gop-size: 100 I have also tried following options: mb-cmp: 2 ildct-cmp: 2 me-cmp: 2 me-sub-cmp: 2 One more thing, could you please share the difference between encoder parameter *bitrate* and *max-key-interval? *Thanks* * On Thu, Mar 22, 2012 at 11:04 PM, Rhodri James wrote: > On Thu, 22 Mar 2012 17:04:45 -0000, Tom Alter > wrote: > > Hi All, >> >> I am using ffmpeg h263 encoder to encode the live stream and sending the >> encoded stream to peer over RTP. Solution works one start but slowly video >> quality gets degraded with time. After two to three hours video gets >> blurred. I have tweaked with various settings of H263 Encoder but did not >> able to fix the problem. >> > > All together now: command line and complete, uncut output missing :-) > > > > Is anyone there who can suggest the reason of this behaviour? Any help >> would be greatly appreciated. >> > > It sounds like you are seeing small errors accumulate over a long time, > which suggests that the encoder isn't producing I-frames often enough > (or at all). It's impossible to tell without more information, but > some combination of your target bit rate and encoder parameters may > be, er, optimistic. > > -- > Rhodri James > Kynesim Ltd > ______________________________**_________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/**listinfo/ffmpeg-user > From rhodri at kynesim.co.uk Fri Mar 23 13:34:33 2012 From: rhodri at kynesim.co.uk (Rhodri James) Date: Fri, 23 Mar 2012 12:34:33 -0000 Subject: [FFmpeg-user] Problem using H263 Encoder In-Reply-To: References: Message-ID: On Fri, 23 Mar 2012 12:28:58 -0000, Tom Alter wrote: > Thanks Rhodri for your response.. > I am setting following properties of Encoder: > bitrate: 150 kpbs > gop-size: 100 > I have also tried following options: > mb-cmp: 2 > ildct-cmp: 2 > me-cmp: 2 > me-sub-cmp: 2 > One more thing, could you please share the difference between encoder > parameter *bitrate* and *max-key-interval? You still haven't shared your command line and complete uncut output with us, so I can't usefully answer your questions. Also, please don't top-post. It's considered rude on this list. -- Rhodri James Kynesim Ltd From jeffreylaut at gmail.com Fri Mar 23 13:06:25 2012 From: jeffreylaut at gmail.com (Jeff L) Date: Fri, 23 Mar 2012 05:06:25 -0700 (PDT) Subject: [FFmpeg-user] resolution problem with recording from webcam Message-ID: <1332504385473-4498565.post@n4.nabble.com> I am trying to record webcam (logitech pro 9000) video using ffmpeg... If I use cheese, the webcam will record at whatever resolution I tell it to. If I use the following command; ffmpeg -t 10 -f video4linux2 -s 1280x720 -r 9 -i /dev/video0 -an webcam.avi The webcam will record, and I'll have a video that is 1280x720 in size, but the quality is crap. It looks like it is just stretching a low resolution image to be 1280x720. Recording with cheese it looks great. I can't seem to figure out why. Any ideas? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/resolution-problem-with-recording-from-webcam-tp4498565p4498565.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From andrey.krieger.utkin at gmail.com Fri Mar 23 14:04:26 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Fri, 23 Mar 2012 15:04:26 +0200 Subject: [FFmpeg-user] resolution problem with recording from webcam In-Reply-To: <1332504385473-4498565.post@n4.nabble.com> References: <1332504385473-4498565.post@n4.nabble.com> Message-ID: 2012/3/23 Jeff L : > I am trying to record webcam (logitech pro 9000) video using ffmpeg... If I > use cheese, the webcam will record at whatever resolution I tell it to. What is cheese? > If I use the following command; > > ffmpeg -t 10 -f video4linux2 -s 1280x720 -r 9 -i /dev/video0 -an webcam.avi Why do you have _two_ inputs? Why do you set explicitly the resolution? > The webcam will record, and I'll have a video that is 1280x720 in size, but > the quality is crap. It looks like it is just stretching a low resolution > image to be 1280x720. Recording with cheese it looks great. I can't seem to > figure out why. > Any ideas? Stick with single video input, and use "-c copy". -- Andrey Utkin From jeffreylaut at gmail.com Fri Mar 23 14:25:37 2012 From: jeffreylaut at gmail.com (Jeff L) Date: Fri, 23 Mar 2012 06:25:37 -0700 (PDT) Subject: [FFmpeg-user] resolution problem with recording from webcam In-Reply-To: References: <1332504385473-4498565.post@n4.nabble.com> Message-ID: <1332509137268-4498790.post@n4.nabble.com> Andrey Utkin wrote > > 2012/3/23 Jeff L <jeffreylaut@>: >> I am trying to record webcam (logitech pro 9000) video using ffmpeg... If >> I >> use cheese, the webcam will record at whatever resolution I tell it to. > > What is cheese? > >> If I use the following command; >> >> ffmpeg -t 10 -f video4linux2 -s 1280x720 -r 9 -i /dev/video0 -an >> webcam.avi > > Why do you have _two_ inputs? > Why do you set explicitly the resolution? > >> The webcam will record, and I'll have a video that is 1280x720 in size, >> but >> the quality is crap. It looks like it is just stretching a low resolution >> image to be 1280x720. Recording with cheese it looks great. I can't seem >> to >> figure out why. >> Any ideas? > > Stick with single video input, and use "-c copy". > > -- > Andrey Utkin > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user@ > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > cheese is a webcam recording program. I have two inputs? I thought that /dev/video0 was my only input.... I explicitly set the resolution because it was using the default framerate (25 fps I think?). I thought that maybe it was using a lower image resolution to get that framerate, so I tried lowering it to something reasonable for that framesize. "-c" doesn't seem to be an option for me, but if I use "-vcodec copy", it seems to do the trick. Thanks! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/resolution-problem-with-recording-from-webcam-tp4498565p4498790.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From ibantxo28 at gmail.com Fri Mar 23 15:11:13 2012 From: ibantxo28 at gmail.com (Iban Garcia) Date: Fri, 23 Mar 2012 15:11:13 +0100 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO Message-ID: Hi All! I am trying to down convert a MXF: It is a Stream #0:5: Video: h264 (High 4:2:2 Intra), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc It has a MBAFF interlacing inside (Mediainfo tells it). With ffmpeg I can get a DV, but it is de-interlaced and I want to be interlaced as well as the original video is. How can I get it using ffmpeg? Is there any flag/option that allow it? Many Thanks Iban From de.techno at gmail.com Fri Mar 23 16:09:42 2012 From: de.techno at gmail.com (dE .) Date: Fri, 23 Mar 2012 20:39:42 +0530 Subject: [FFmpeg-user] Ffplay feature request In-Reply-To: <1267970240-1332349330-cardhu_decombobulator_blackberry.rim.net-1772329587-@b4.c15.bise7.blackberry> References: <1267970240-1332349330-cardhu_decombobulator_blackberry.rim.net-1772329587-@b4.c15.bise7.blackberry> Message-ID: <4F6C9236.5010704@gmail.com> On 03/21/12 22:32, ryan.ismail2 at gmail.com wrote: > Hi I would really appreciate it if ffplay could interact with lirc and have a -playlist switch similar to mplayer's. Thank you in advance > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user I don't think ffplay was actually designed to be a mainstream player. Correct me if I'm wrong. "It is mostly used as a testbed for the various FFmpeg APIs." From andrey.krieger.utkin at gmail.com Fri Mar 23 16:25:13 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Fri, 23 Mar 2012 17:25:13 +0200 Subject: [FFmpeg-user] resolution problem with recording from webcam In-Reply-To: <1332509137268-4498790.post@n4.nabble.com> References: <1332504385473-4498565.post@n4.nabble.com> <1332509137268-4498790.post@n4.nabble.com> Message-ID: 2012/3/23 Jeff L : > I have two inputs? ?I thought that /dev/video0 was my only input.... Terribly sorry for confusion, my wrong sight. > "-c" doesn't seem to be an option for me, but if I use "-vcodec copy", it > seems to do the trick. ?Thanks! This manifests you use old version of ffmpeg. -- Andrey Utkin From de.techno at gmail.com Fri Mar 23 16:28:41 2012 From: de.techno at gmail.com (dE .) Date: Fri, 23 Mar 2012 20:58:41 +0530 Subject: [FFmpeg-user] How can I replace numbered sound tracks in my AVI ? In-Reply-To: <1332431469877-4495903.post@n4.nabble.com> References: <524EA12FEC145E4E86154A23DA429D2D5DE588AFD3@PDC-MAIL-CMS01.ubisoft.org> <1332431469877-4495903.post@n4.nabble.com> Message-ID: <4F6C96A9.2030601@gmail.com> On 03/22/12 21:21, Olivier Soleil wrote: > Doesn't anyone have any idea about these issues ? > > I had no answer yet, but i'd like to remind you the main things i miss. > > *First question*, is there an easy way for the user to mux an audio stream > to a desired stream number in the destination file. > e.g. : I want to mux one soundtrack to the 100th stream in my video. > Does it exist ? Is there any chance that this feature can be developed ? Yes, it's possible, but I don't know how. It's there in the man page. From joddo at jerfu.com Fri Mar 23 17:02:34 2012 From: joddo at jerfu.com (Jeremy Oddo) Date: Fri, 23 Mar 2012 09:02:34 -0700 Subject: [FFmpeg-user] ProRes (LT) Quicktimes: Good for Windows. Not so good for Mac. In-Reply-To: References: <4F68F4F9.3040504@gmx.de> Message-ID: > The co64 atom is only used when the file is over 4GB. The reason is > because an unsigned 32 bit value (e.g. uint32_t, which is 4 bytes) > simply isn't big enough to hold a number over 4294967295. Sometimes > the file is so big that the position of data relative to the beginning > of the file goes past 4294967295, so without a 64 bit value (i.e. > int64_t, which is 8 bytes), the file reader hits the 4GB limit and > doesn't have any more numbers to seek over 4GB. Got it. Thanks for explaining. > It should also be pointed out that MOV/MP4 elst atoms aren't written > properly in many cases, at least not in a way that Apple's software > expects them. For example, sometimes the start value in elst will be > wrong if performing stream copies with B-frame'd H.264. The fix is to > go into a hex editor and change the start value to 1 or 1001, > depending on the PTS duration. ffmbc isn't affected by this since > Baptiste has taken the time to ensure the MOV/MP4 writer works in a > way that Apple software expects. IMHO, someone should follow ffmbc's > good example and fix the MOV muxer in ffmpeg, at least to the point > where it addresses these issues. For what it's worth, I encoded a ProRes QT using ffmbc and had the same issue (FCP not accepting the QT as ProRes (LT)). From your text above, I wasn't sure if you were pointing out that ffmbc could make "legal" ProRes (LT)'s or if you were merely saying that ffmbc deals with B-frame'd H.264's properly. (I should note that I didn't have an audio track if that is an issue.) From ianm at brick.net Fri Mar 23 17:33:18 2012 From: ianm at brick.net (ianm at brick.net) Date: Fri, 23 Mar 2012 11:33:18 -0500 Subject: [FFmpeg-user] -f vob does not appear to produce MPEG-2 Program Streams according to ffprobe Message-ID: I am seeking to concatenate video clips of what might well be random provenance, typically .MTS files (MEPG-2 TS container file, H.264 video, AC-3 audio) but not always. These are reasonably high resolution: 1920 x 1080. The FAQ lists the following as "privileged formats" capable of binary concatenation: "MPEG-1, MPEG-2 PS, DV" I wanted to give MPEG-2 PS a shot. I'm using the wonderful Zeranoe 64-bit static build on Windows 7. Here's my command line on a sample clip and the results: C:\data\pipeline\binaries\ffmpeg\bin>ffmpeg -i C:\data\pipeline\input\short-clip.mts -codec:v copy -codec:a copy -f vob short-clip.vob ffmpeg version N-38292-ga4c22e3 Copyright (c) 2000-2012 the FFmpeg developers built on Feb 27 2012 14:55:47 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 41.100 / 51. 41.100 libavcodec 54. 4.100 / 54. 4.100 libavformat 54. 1.100 / 54. 1.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 62.101 / 2. 62.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 [h264 @ 0000000001E7AAB0] Increasing reorder buffer to 1 Input #0, mpegts, from 'C:\data\pipeline\input\short-clip.mts': Duration: 00:02:00.03, start: 1.410000, bitrate: 24790 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 59.96 fps, 59.94 tbr, 90k tbn, 59.94 tbc Stream #0:1[0x101]: Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, stereo, s16, 256 kb/s [vob @ 0000000001E5ECC0] VBV buffer size not set, muxing may fail Output #0, vob, to 'short-clip.vob': Metadata: encoder : Lavf54.1.100 Stream #0:0: Video: h264 ([27][0][0][0] / 0x001B), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 59.96 fps, 90k tbn, 29.97 tbc Stream #0:1: Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, stereo, 256 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help frame= 7193 fps=692 q=-1.0 Lsize= 340006kB time=00:02:00.00 bitrate=23209.7kbits/s video:332130kB audio:3750kB global headers:0kB muxing overhead 1.228504% Yet when I ffprobe the file, I get C:\data\pipeline\binaries\ffmpeg\bin>ffprobe short-clip.vob ffprobe version N-38292-ga4c22e3 Copyright (c) 2007-2012 the FFmpeg developers built on Feb 27 2012 14:55:47 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 41.100 / 51. 41.100 libavcodec 54. 4.100 / 54. 4.100 libavformat 54. 1.100 / 54. 1.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 62.101 / 2. 62.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 [h264 @ 00000000003FF840] Increasing reorder buffer to 1 Input #0, mpeg, from 'short-clip.vob': Duration: 00:02:00.03, start: 0.999967, bitrate: 23204 kb/s Stream #0:0[0x1e0]: Video: h264 (High), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 59.96 fps, 59.94 tbr, 90k tbn, 59.94 tbc Stream #0:1[0x80]: Audio: ac3, 48000 Hz, stereo, s16, 256 kb/s Running a ffprobe -formats yields the relevant line "DE mpeg MPEG-1 System format," which suggests that I am not, in fact, getting an MPEG-2 Program Stream container format. How do I do that? Obviously, I would need to select the right codecs, but I am not that far yet, although suggestions would be appreciated. Alternatively, if I ought to be using MPEG-1 as the container format, what audio and video codecs ought I to use with it? I'm looking to retain the large resolution, bitrate, etc., until the final downmix. As I am trying to automate this process (clips get transcoded, concatenated, transcoded more, all without human eyes looking at them), I think I need be more aware of potential problems with timestamps but I do not know where to begin there and how my choice of concatenation format will or will not affect that. From nichot20 at yahoo.com Fri Mar 23 17:36:49 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Fri, 23 Mar 2012 16:36:49 +0000 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: References: Message-ID: <4F6CA6A1.1050607@yahoo.com> On 23/03/12 14:11, Iban Garcia wrote: > Hi All! > > I am trying to down convert a MXF: It is a Stream #0:5: Video: h264 (High > 4:2:2 Intra), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 > tbn, 50 tbc > > It has a MBAFF interlacing inside (Mediainfo tells it). > > With ffmpeg I can get a DV, but it is de-interlaced and I want to be > interlaced as well as the original video is. > > How can I get it using ffmpeg? Is there any flag/option that allow it? > You don't provide complete uncut console output. However from my experience MXF's always seem to be flagged as progressive even when not. In which case setfield=1 is your friend. If this does not suffice I refer you to my first comment.... -- Tim From tevans.uk at googlemail.com Fri Mar 23 17:43:41 2012 From: tevans.uk at googlemail.com (Tom Evans) Date: Fri, 23 Mar 2012 16:43:41 +0000 Subject: [FFmpeg-user] -f vob does not appear to produce MPEG-2 Program Streams according to ffprobe In-Reply-To: References: Message-ID: On Fri, Mar 23, 2012 at 4:33 PM, wrote: > > I am seeking to concatenate video clips of what might well be random > provenance, typically .MTS files (MEPG-2 TS container file, H.264 video, > AC-3 audio) but not always. These are reasonably high resolution: 1920 x > 1080. ?The FAQ lists the following as "privileged formats" capable of binary > concatenation: "MPEG-1, MPEG-2 PS, DV" ?I wanted to give MPEG-2 PS a shot. > ?I'm using the wonderful Zeranoe 64-bit static build on Windows 7. ?Here's > my command line on a sample clip and the results: > > C:\data\pipeline\binaries\ffmpeg\bin>ffmpeg -i > C:\data\pipeline\input\short-clip.mts -codec:v copy -codec:a copy -f vob > short-clip.vob Why "-f vob" and not "-f mpegts" ? You may also need to use a bitstream filter to ensure your H264 streams are annex b compliant, so they can be concatenated. IE, try something like this: ffmpeg -i src.mts -c:v copy -c:a copy -bsf:v h264_mp4toannexb -f mpegts out.ts Oh, I see, the FAQ says to use MPEG2 PS - hmm. I've used the above recipe to concatenate HD TV recordings in mpegts :/ Cheers Tom From tom1.alter at gmail.com Fri Mar 23 18:24:27 2012 From: tom1.alter at gmail.com (Tom Alter) Date: Fri, 23 Mar 2012 22:54:27 +0530 Subject: [FFmpeg-user] Problem using H263 Encoder In-Reply-To: References: Message-ID: Hi Rhodri, i am not using command line infact i am doing all the stuff in c code. Can u please specify what exactly you need from my side? Thanks On 3/23/12, Rhodri James wrote: > On Fri, 23 Mar 2012 12:28:58 -0000, Tom Alter wrote: > >> Thanks Rhodri for your response.. >> I am setting following properties of Encoder: >> bitrate: 150 kpbs >> gop-size: 100 >> I have also tried following options: >> mb-cmp: 2 >> ildct-cmp: 2 >> me-cmp: 2 >> me-sub-cmp: 2 >> One more thing, could you please share the difference between encoder >> parameter *bitrate* and *max-key-interval? > > You still haven't shared your command line and complete uncut output > with us, so I can't usefully answer your questions. > > Also, please don't top-post. It's considered rude on this list. > > -- > Rhodri James > Kynesim Ltd > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From rhodri at kynesim.co.uk Fri Mar 23 18:31:44 2012 From: rhodri at kynesim.co.uk (Rhodri James) Date: Fri, 23 Mar 2012 17:31:44 -0000 Subject: [FFmpeg-user] Problem using H263 Encoder In-Reply-To: References: Message-ID: On Fri, 23 Mar 2012 17:24:27 -0000, Tom Alter wrote: > Hi Rhodri, > i am not using command line infact i am doing all the stuff in c code. Then you are outside my experience. And still top-posting. > On 3/23/12, Rhodri James wrote: >> On Fri, 23 Mar 2012 12:28:58 -0000, Tom Alter >> wrote: >> >>> Thanks Rhodri for your response.. >>> I am setting following properties of Encoder: >>> bitrate: 150 kpbs >>> gop-size: 100 >>> I have also tried following options: >>> mb-cmp: 2 >>> ildct-cmp: 2 >>> me-cmp: 2 >>> me-sub-cmp: 2 >>> One more thing, could you please share the difference between encoder >>> parameter *bitrate* and *max-key-interval? >> >> You still haven't shared your command line and complete uncut output >> with us, so I can't usefully answer your questions. >> >> Also, please don't top-post. It's considered rude on this list. -- Rhodri James Kynesim Ltd From etienne.buira.lists at free.fr Fri Mar 23 18:57:18 2012 From: etienne.buira.lists at free.fr (Etienne Buira) Date: Fri, 23 Mar 2012 18:57:18 +0100 Subject: [FFmpeg-user] pcm_s16le (and flac) in TS fails Message-ID: <20120323175718.GU2315@epicure.lazyet.homelinux.net> Hi all. Tried on many kind of input files with HEAD (and earliers): 1. ffmpeg -i source.something -ss 120 -t 120 -c:a pcm_s16le -c:v libx264 -preset ultrafast -qp 0 pcm_s16le.ts 2. ffmpeg -i pcm_s16le.ts ffmpeg version N-39180-g4c38e8a Copyright (c) 2000-2012 the FFmpeg developers built on Mar 23 2012 18:16:41 with gcc 4.5.3 configuration: --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --disable-stripping --enable-nonfree --enable-libfaac --enable-libmp3lame --enable-libx264 libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 2.100 / 54. 2.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 65.102 / 2. 65.102 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 [mpegts @ 0x32f33a0] probed stream 1 failed [mpegts @ 0x32f33a0] Could not find codec parameters (Unknown: none ([6][0][0][0] / 0x0006)) Input #0, mpegts, from 'pcm_s16le.ts': Duration: 00:01:59.99, start: 1.400000, bitrate: 29007 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100]: Video: h264 (High 4:4:4 Predictive) ([27][0][0][0] / 0x001B), yuv420p, 704x528 [SAR 4:3 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0:1[0x101](fra): Unknown: none ([6][0][0][0] / 0x0006) At least one output file must be specified Did I do something wrong in my command line? I also tried flac, but similar issue. For the global picture, I want to do some processing on some streams, being as lossless as possible, and then concat them to encode in wished final quality. Somebody have a better idea for lossless step encodes? Regards From andrey.krieger.utkin at gmail.com Fri Mar 23 19:51:48 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Fri, 23 Mar 2012 20:51:48 +0200 Subject: [FFmpeg-user] pcm_s16le (and flac) in TS fails In-Reply-To: <20120323175718.GU2315@epicure.lazyet.homelinux.net> References: <20120323175718.GU2315@epicure.lazyet.homelinux.net> Message-ID: 2012/3/23 Etienne Buira : > Hi all. > > Tried on many kind of input files with HEAD (and earliers): > > 1. ffmpeg -i source.something -ss 120 -t 120 -c:a pcm_s16le -c:v libx264 -preset ultrafast -qp 0 pcm_s16le.ts > 2. ffmpeg -i pcm_s16le.ts > ffmpeg version N-39180-g4c38e8a Copyright (c) 2000-2012 the FFmpeg developers > ?built on Mar 23 2012 18:16:41 with gcc 4.5.3 > ?configuration: --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --disable-stripping --enable-nonfree --enable-libfaac --enable-libmp3lame --enable-libx264 > ?libavutil ? ? ?51. 44.100 / 51. 44.100 > ?libavcodec ? ? 54. 12.100 / 54. 12.100 > ?libavformat ? ?54. ?2.100 / 54. ?2.100 > ?libavdevice ? ?53. ?4.100 / 53. ?4.100 > ?libavfilter ? ? 2. 65.102 / ?2. 65.102 > ?libswscale ? ? ?2. ?1.100 / ?2. ?1.100 > ?libswresample ? 0. ?7.100 / ?0. ?7.100 > ?libpostproc ? ?52. ?0.100 / 52. ?0.100 > [mpegts @ 0x32f33a0] probed stream 1 failed > [mpegts @ 0x32f33a0] Could not find codec parameters (Unknown: none ([6][0][0][0] / 0x0006)) > Input #0, mpegts, from 'pcm_s16le.ts': > ?Duration: 00:01:59.99, start: 1.400000, bitrate: 29007 kb/s > ?Program 1 > ? ?Metadata: > ? ? ?service_name ? ?: Service01 > ? ? ?service_provider: FFmpeg > ? ?Stream #0:0[0x100]: Video: h264 (High 4:4:4 Predictive) ([27][0][0][0] / 0x001B), yuv420p, 704x528 [SAR 4:3 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc > ? ?Stream #0:1[0x101](fra): Unknown: none ([6][0][0][0] / 0x0006) > At least one output file must be specified > > Did I do something wrong in my command line? AFAIK pcm_s16le audio codec cannot be contained in mpegts. Check allowed codecs for your desired container, or choose container that supports lossless audio formats. -- Andrey Utkin From dev at rarevision.com Sat Mar 24 07:08:15 2012 From: dev at rarevision.com (Thomas Worth) Date: Fri, 23 Mar 2012 23:08:15 -0700 Subject: [FFmpeg-user] ProRes (LT) Quicktimes: Good for Windows. Not so good for Mac. In-Reply-To: References: <4F68F4F9.3040504@gmx.de> Message-ID: >> It should also be pointed out that MOV/MP4 elst atoms aren't written >> properly in many cases, at least not in a way that Apple's software >> expects them. For example, sometimes the start value in elst will be >> wrong if performing stream copies with B-frame'd H.264. The fix is to >> go into a hex editor and change the start value to 1 or 1001, >> depending on the PTS duration. ffmbc isn't affected by this since >> Baptiste has taken the time to ensure the MOV/MP4 writer works in a >> way that Apple software expects. IMHO, someone should follow ffmbc's >> good example and fix the MOV muxer in ffmpeg, at least to the point >> where it addresses these issues. > > For what it's worth, I encoded a ProRes QT using ffmbc and had the > same issue (FCP not accepting the QT as ProRes (LT)). From your > text above, I wasn't sure if you were pointing out that ffmbc could make > "legal" ProRes (LT)'s or if you were merely saying that ffmbc deals with > B-frame'd H.264's properly. (I should note that I didn't have an audio track > if that is an issue.) I don't know. I haven't used ffmbc to generate ProRes files. How did you determine that FCP wouldn't accept the file as ProRes LT? I'm not sure this even matters since ProRes should just be ProRes. As far as I know, the only difference between the different ProRes flavors is bitrate. Can someone confirm that? From cehoyos at ag.or.at Sat Mar 24 08:32:52 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 24 Mar 2012 07:32:52 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?pcm=5Fs16le_=28and_flac=29_in_TS_fails?= References: <20120323175718.GU2315@epicure.lazyet.homelinux.net> Message-ID: Etienne Buira free.fr> writes: > 1. ffmpeg -i source.something -ss 120 -t 120 -c:a pcm_s16le > -c:v libx264 -preset ultrafast -qp 0 pcm_s16le.ts Do you have any indication that pcm in mpeg-ts is supported by any application? If yes, the developers will seriously consider adding decoding (and possibly encoding) support to FFmpeg. If FFmpeg supports encoding arbitrary codecs to (quite) special-purpose containers (as opposed to for example avi) that can certainly not be decoded by any other application, this leads to guaranteed trouble. If you really know what you are doing, you will find that it is very simple to add support for arbitrary codecs to the mpeg-ts encoder and decoder. Carl Eugen From cehoyos at ag.or.at Sat Mar 24 08:38:59 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 24 Mar 2012 07:38:59 +0000 (UTC) Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO References: Message-ID: Iban Garcia gmail.com> writes: > It has a MBAFF interlacing inside (Mediainfo tells it). > > With ffmpeg I can get a DV, but it is de-interlaced and > I want to be interlaced as well as the original video is. Please provide the command line you used and the complete, uncut console output and please point to a very short sample. FFmpeg does no de-interlacing at all if not explicitly told to do so. (Otoh, progressive material can be encoded using MBAFF - and even PAFF - and since the H264 decoder can only output frames, you will get the - original - progressive material if you encode the frames coming from the decoder.) Carl Eugen From cehoyos at ag.or.at Sat Mar 24 08:35:16 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 24 Mar 2012 07:35:16 +0000 (UTC) Subject: [FFmpeg-user] -f vob does not appear to produce MPEG-2 Program Streams according to ffprobe References: Message-ID: brick.net> writes: > The FAQ lists the following as "privileged formats" capable of > binary concatenation: "MPEG-1, MPEG-2 PS, DV" I wanted to > give MPEG-2 PS a shot. I don't think there is any difference between a MPEG-1 and a MPEG-2 program stream (at least I believe there is no implementation difference in FFmpeg). Carl Eugen From cehoyos at ag.or.at Sat Mar 24 08:41:21 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 24 Mar 2012 07:41:21 +0000 (UTC) Subject: [FFmpeg-user] Unable to demux 5.1 DVD-Audio rips References: <1332418283977-4495306.post@n4.nabble.com> Message-ID: iiser list.ru> writes: > Hello. I have a couple of DVD-Audio rips: one contains 44.1kHz/24-bit > 5.1-channel MLP track and another one has 48kHz/24-bit 5.1-channel MLP > track. For some reason ffmpeg is extracting only 10kB and 120kB files > respectively from 1GB input. Please upload a sample, see http://ffmpeg.org/bugreports.html Thank you, Carl Eugen From cehoyos at ag.or.at Sat Mar 24 08:51:24 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 24 Mar 2012 07:51:24 +0000 (UTC) Subject: [FFmpeg-user] How can I replace numbered sound tracks in my AVI ? References: <524EA12FEC145E4E86154A23DA429D2D5DE588AFD3@PDC-MAIL-CMS01.ubisoft.org> Message-ID: Olivier Soleil ubisoft.fr> writes: > Let's say that I have to produce CGI movies for a game, and > voiced tracks (ENG, FRA, ...) are located above stream 100. > So, I have a video track on the 0 stream, surround ambient > sounds located between 1-5 streams, and voiced tracks above > 100th stream. Please be more specific in a different way. The important thing is not that you need movies for a game, but the technology you want to use (especially which container). MPEG-TS certainly supports tracks, but I am not sure that you can put something into stream 0. (Writing bink files is not supported by FFmpeg.) > Every time the soundmix for a language is updated, I would > like to replace the old one with the new one, > without changing the rest of the data in the AVI. Remux the AVI, this does not (necessarily) change the data. Carl Eugen From etienne.buira.lists at free.fr Sat Mar 24 09:52:20 2012 From: etienne.buira.lists at free.fr (Etienne Buira) Date: Sat, 24 Mar 2012 09:52:20 +0100 Subject: [FFmpeg-user] pcm_s16le (and flac) in TS fails In-Reply-To: References: <20120323175718.GU2315@epicure.lazyet.homelinux.net> Message-ID: <20120324085220.GV2315@epicure.lazyet.homelinux.net> On Fri, Mar 23, 2012 at 08:51:48PM +0200, Andrey Utkin wrote: > 2012/3/23 Etienne Buira : > > Hi all. > > > > Tried on many kind of input files with HEAD (and earliers): > > > > 1. ffmpeg -i source.something -ss 120 -t 120 -c:a pcm_s16le -c:v libx264 -preset ultrafast -qp 0 pcm_s16le.ts > > 2. ffmpeg -i pcm_s16le.ts > > ffmpeg version N-39180-g4c38e8a Copyright (c) 2000-2012 the FFmpeg developers > > ?built on Mar 23 2012 18:16:41 with gcc 4.5.3 > > ?configuration: --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --disable-stripping --enable-nonfree --enable-libfaac --enable-libmp3lame --enable-libx264 > > ?libavutil ? ? ?51. 44.100 / 51. 44.100 > > ?libavcodec ? ? 54. 12.100 / 54. 12.100 > > ?libavformat ? ?54. ?2.100 / 54. ?2.100 > > ?libavdevice ? ?53. ?4.100 / 53. ?4.100 > > ?libavfilter ? ? 2. 65.102 / ?2. 65.102 > > ?libswscale ? ? ?2. ?1.100 / ?2. ?1.100 > > ?libswresample ? 0. ?7.100 / ?0. ?7.100 > > ?libpostproc ? ?52. ?0.100 / 52. ?0.100 > > [mpegts @ 0x32f33a0] probed stream 1 failed > > [mpegts @ 0x32f33a0] Could not find codec parameters (Unknown: none ([6][0][0][0] / 0x0006)) > > Input #0, mpegts, from 'pcm_s16le.ts': > > ?Duration: 00:01:59.99, start: 1.400000, bitrate: 29007 kb/s > > ?Program 1 > > ? ?Metadata: > > ? ? ?service_name ? ?: Service01 > > ? ? ?service_provider: FFmpeg > > ? ?Stream #0:0[0x100]: Video: h264 (High 4:4:4 Predictive) ([27][0][0][0] / 0x001B), yuv420p, 704x528 [SAR 4:3 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc > > ? ?Stream #0:1[0x101](fra): Unknown: none ([6][0][0][0] / 0x0006) > > At least one output file must be specified > > > > Did I do something wrong in my command line? > > AFAIK pcm_s16le audio codec cannot be contained in mpegts. Check > allowed codecs for your desired container, or choose container that > supports lossless audio formats. OK, thanks, I'll look for another container/codecs tuple to do my encodes. From etienne.buira.lists at free.fr Sat Mar 24 10:16:47 2012 From: etienne.buira.lists at free.fr (Etienne Buira) Date: Sat, 24 Mar 2012 10:16:47 +0100 Subject: [FFmpeg-user] pcm_s16le (and flac) in TS fails In-Reply-To: References: <20120323175718.GU2315@epicure.lazyet.homelinux.net> Message-ID: <20120324091647.GW2315@epicure.lazyet.homelinux.net> On Sat, Mar 24, 2012 at 07:32:52AM +0000, Carl Eugen Hoyos wrote: > Etienne Buira free.fr> writes: > > > 1. ffmpeg -i source.something -ss 120 -t 120 -c:a pcm_s16le > > -c:v libx264 -preset ultrafast -qp 0 pcm_s16le.ts > > Do you have any indication that pcm in mpeg-ts is supported > by any application? > If yes, the developers will seriously consider adding decoding > (and possibly encoding) support to FFmpeg. > > If FFmpeg supports encoding arbitrary codecs to (quite) > special-purpose containers (as opposed to for example avi) that > can certainly not be decoded by any other application, this > leads to guaranteed trouble. > > If you really know what you are doing, you will find that it is > very simple to add support for arbitrary codecs to the mpeg-ts > encoder and decoder. > > Carl Eugen Hi Carl. As said, I do not intent to use the (pcm,h264 q=0) in mpegts for longer than temporary files. My final goal is just to extract parts of some streams (aka ffmpeg -i something_part1 -ss 10 t 120 tempout1 && ffmpeg -i something_part2 -ss 10 -t 120 tempout2) losslessly to concat them just after that and do a lossy encode (aka ffmpeg -i 'concat://tempout1|tempout2' -crf 15 final). I did this kind of thing with avi in the past, but it is not working anymore at the concat step, so I tried with a container designed to be concatenable (even if from what I understand, avi should be, as long as index is not wished). If you have a better idea of catable container/lossless video codec/lossless audio codec that is usable with ffmpeg (and that offers some compression, at least for video part), I'd buy it! Ideally, the container also haves timestamps, in order to keep AV sync even in case of slight corruption of the source. Thanks for your answer. From cehoyos at ag.or.at Sat Mar 24 10:31:18 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 24 Mar 2012 09:31:18 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?pcm=5Fs16le_=28and_flac=29_in_TS_fails?= References: <20120323175718.GU2315@epicure.lazyet.homelinux.net> <20120324091647.GW2315@epicure.lazyet.homelinux.net> Message-ID: Etienne Buira free.fr> writes: > I did this kind of thing with avi in the past, but it is not working > anymore at the concat step, so I tried with a container designed to be > concatenable (even if from what I understand, avi should be, as long as > index is not wished). Did you report this? (Although I don't understand how avi can be concatenated, given it has a header). > If you have a better idea of catable container/lossless video > codec/lossless audio codec that is usable with ffmpeg (and that offers > some compression, at least for video part), I'd buy it! Do you really hear any difference between lossless and -acodec (e)ac3 -ab 640k ? Carl Eugen From etienne.buira.lists at free.fr Sat Mar 24 10:44:41 2012 From: etienne.buira.lists at free.fr (Etienne Buira) Date: Sat, 24 Mar 2012 10:44:41 +0100 Subject: [FFmpeg-user] pcm_s16le (and flac) in TS fails In-Reply-To: References: <20120323175718.GU2315@epicure.lazyet.homelinux.net> <20120324091647.GW2315@epicure.lazyet.homelinux.net> Message-ID: <20120324094441.GX2315@epicure.lazyet.homelinux.net> On Sat, Mar 24, 2012 at 09:31:18AM +0000, Carl Eugen Hoyos wrote: > Etienne Buira free.fr> writes: > > > I did this kind of thing with avi in the past, but it is not working > > anymore at the concat step, so I tried with a container designed to be > > concatenable (even if from what I understand, avi should be, as long as > > index is not wished). > > Did you report this? > (Although I don't understand how avi can be concatenated, given it has > a header). I don't remember doing so, or maybe posting to this list. AVI can, to some extend be concatenated. I don't know the gory details of containers formats nor codecs, but for sure an mencoder FAQ entry stated that you could join two avi files sharing same codec, resolution, by cating them and then rebuilding index. > > If you have a better idea of catable container/lossless video > > codec/lossless audio codec that is usable with ffmpeg (and that offers > > some compression, at least for video part), I'd buy it! > > Do you really hear any difference between lossless and > -acodec (e)ac3 -ab 640k ? Didn't try actually, and I assume I don't, but streams might be processed many times before final lossy encode, so using lossless seams the most reasonable option to me. From Ajita-Pandey at hcl.com Sat Mar 24 13:30:40 2012 From: Ajita-Pandey at hcl.com (Ajita Pandey) Date: Sat, 24 Mar 2012 18:00:40 +0530 Subject: [FFmpeg-user] DTS Profile Issue In-Reply-To: References: <20120323175718.GU2315@epicure.lazyet.homelinux.net> <20120324091647.GW2315@epicure.lazyet.homelinux.net> Message-ID: <45F3FDEC759D21468F1F910D2B1CC43BC588003D39@NDA-HCLT-EVS04.HCLT.CORP.HCL.IN> Hi all, I am trying to decode some DTS profile which my hardware supports like DTS -low bit rate and DTS -boardcast But I don't any decoder in ffmpeg code . can any please tell where I get this . Also if this is supported by any existing profile. Regards, Ajita ::DISCLAIMER:: ----------------------------------------------------------------------------------------------------------------------- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. It shall not attach any liability on the originator or HCL or its affiliates. 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Before opening any mail and attachments please check them for viruses and defect. ----------------------------------------------------------------------------------------------------------------------- From stefasab at gmail.com Sat Mar 24 14:29:05 2012 From: stefasab at gmail.com (Stefano Sabatini) Date: Sat, 24 Mar 2012 14:29:05 +0100 Subject: [FFmpeg-user] Ffplay feature request In-Reply-To: <1267970240-1332349330-cardhu_decombobulator_blackberry.rim.net-1772329587-@b4.c15.bise7.blackberry> References: <1267970240-1332349330-cardhu_decombobulator_blackberry.rim.net-1772329587-@b4.c15.bise7.blackberry> Message-ID: <20120324132905.GE5283@arborea> On date Wednesday 2012-03-21 17:02:16 +0000, ryan.ismail2 at gmail.com encoded: > Hi I would really appreciate it if ffplay could interact with lirc > and have a -playlist switch similar to mplayer's. Thank you in > advance Please create an enhancement ticket on trac, or this request will be lost in the bits limbo: http://ffmpeg.org/trac/ffmpeg -- ffmpeg-user random tip #33 FFmpeg has several video sources which can be used for generating synthetic video. Example: ffplay -f lavfi testsrc=n=2 From smfabac at att.net Sat Mar 24 18:40:37 2012 From: smfabac at att.net (Steve M. Fabac, Jr.) Date: Sat, 24 Mar 2012 12:40:37 -0500 Subject: [FFmpeg-user] Note to document maintainer Message-ID: <4F6E0715.2010604@att.net> As a newbie I tried the following example: The following example shows how to use ffmpeg for creating a sequence of files ?img-001.jpeg?, ?img-002.jpeg?, ..., taking one image every second from the input video: ffmpeg -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg' So what's the change/addition to create an image every two seconds? Every three seconds? The needed switch may be buried in the overall documentation. Having it in the "section 9.3 image 2" would save newbie's a lot of searching. Also, the single quotes above fail under Windows XP and the correct command is: ffmpeg -i in.avi -vsync 1 -r 1 -f image2 img-%03d.jpeg -- Steve Fabac S.M. Fabac & Associates 816/765-1670 From eolinwen at gmail.com Sat Mar 24 20:42:59 2012 From: eolinwen at gmail.com (eolinwen at gmail.com) Date: Sat, 24 Mar 2012 20:42:59 +0100 Subject: [FFmpeg-user] Note to document maintainer In-Reply-To: <4F6E0715.2010604@att.net> References: <4F6E0715.2010604@att.net> Message-ID: Hi, I am just an new user and far away to be an expert but with news version of ffmpeg the option -f image2 is deprecated. you should replace this one by -f mjpeg. Just my 2 cents. 2012/3/24 Steve M. Fabac, Jr. > As a newbie I tried the following example: > > The following example shows how to use ffmpeg for creating > a sequence of files ?img-001.jpeg?, ?img-002.jpeg?, ..., > taking one image every second from the input video: > > ffmpeg -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg' > > So what's the change/addition to create an image every two seconds? > Every three seconds? The needed switch may be buried in > the overall documentation. Having it in the "section 9.3 image 2" > would save newbie's a lot of searching. > > Also, the single quotes above fail under Windows XP and the correct > command is: ffmpeg -i in.avi -vsync 1 -r 1 -f image2 img-%03d.jpeg > > > -- > Steve Fabac > S.M. Fabac & Associates > 816/765-1670 > ______________________________**_________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/**listinfo/ffmpeg-user > -- Olivier Cenwen un elfe sur la banquise/ an elve on the ice Mon blog perso sur le multim?dia, Ubuntu, Linux et OpenShot : http://linuxevolution.wordpress.com/ Le forum d'Openshot o? vous me trouverez : http://openshotusers.com/ http://openshotusers.com/forum/index.php Nothing is lost until the last second. The family motto : When we want, we can. Astuces, Actualit?s, Logiciels, bref sur tout ce que je ne fais d'articles dessus Google+ From cehoyos at ag.or.at Sat Mar 24 23:34:28 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 24 Mar 2012 22:34:28 +0000 (UTC) Subject: [FFmpeg-user] Note to document maintainer References: <4F6E0715.2010604@att.net> Message-ID: eolinwen gmail.com gmail.com> writes: > I am just an new user and far away to be an expert but with > news version of ffmpeg the option -f image2 is deprecated. > you should replace this one by -f mjpeg. image2 and mjpeg are two different formats, none of them is deprecated. Carl Eugen From roko98 at yahoo.com Sun Mar 25 03:21:26 2012 From: roko98 at yahoo.com (roko) Date: Sat, 24 Mar 2012 18:21:26 -0700 (PDT) Subject: [FFmpeg-user] MPEG-TS trouble In-Reply-To: <20120321203842.GB13842@kiste2> References: <1332307981.76400.YahooMailNeo@web162003.mail.bf1.yahoo.com> <1332339430.4061.14.camel@gw-laptop> <20120321203842.GB13842@kiste2> Message-ID: <1332638486.73254.YahooMailNeo@web162005.mail.bf1.yahoo.com> Thank you all for your answers.? A compiled version from Git save the file now: emelo at alfa:~/workspace/ffmpeg> ./ffmpeg -i rtsp://admin:admin at 192.168.99.146/11 -vcodec copy -y -r 25 -vbsf h264_mp4toannexb algo.ts ffmpeg version N-39223-gac6798d Copyright (c) 2000-2012 the FFmpeg developers ? built on Mar 24 2012 19:43:40 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] ? configuration: --shlibdir=/usr/lib64 --prefix=/usr --mandir=/usr/share/man --libdir=/usr/lib64 --enable-shared --disable-static --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libxvid --enable-postproc --enable-gpl --enable-x11grab --extra-cflags='-fmessage-length=0 -O2 -Wall -D_FORTIFY_SOURCE=2 -fstack-protector -funwind-tables -fasynchronous-unwind-tables -g -fPIC -I/usr/include/gsm' --enable-debug --disable-stripping --enable-libgsm --enable-libschroedinger --enable-libdirac --enable-avfilter --enable-libvpx --enable-version3 --enable-libx264 --enable-libdc1394 --enable-pthreads --enable-librtmp ? libavutil????? 51. 44.100 / 51. 44.100 ? libavcodec???? 54. 12.100 / 54. 12.100 ? libavformat??? 54.? 2.100 / 54.? 2.100 ? libavdevice??? 53.? 4.100 / 53.? 4.100 ? libavfilter???? 2. 65.102 /? 2. 65.102 ? libswscale????? 2.? 1.100 /? 2.? 1.100 ? libswresample?? 0. 10.100 /? 0. 10.100 ? libpostproc??? 52.? 0.100 / 52.? 0.100 [rtsp @ 0x6283a0] Estimating duration from bitrate, this may be inaccurate Input #0, rtsp, from 'rtsp://admin:admin at 192.168.99.146/11':??????????????????? ? Metadata: ??? title?????????? : \11 ? Duration: N/A, start: 0.039956, bitrate: N/A ??? Stream #0:0: Video: h264 (Constrained Baseline), yuv420p, 640x480, 25 tbr, 90k tbn, 180k tbc [mpegts @ 0x656820] muxrate VBR, pcr every 9000 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to 'algo.ts': ? Metadata: ??? title?????????? : \11 ??? encoder???????? : Lavf54.2.100 ??? Stream #0:0: Video: h264, yuv420p, 640x480, q=2-31, 90k tbn, 90k tbc Stream mapping: ? Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help And the file (algo.ts) plays well with ffplay, but a HUGE (realy... HUGE) amount of : Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument Is generated in the process.? Do you consider it "normal" or there is something more to be fixed ? ??? E. On Wed, Mar 21, 2012 at 10:17:10PM +0800, gs_gail wrote: > there is a bug in libavcodec/h264_mp4toannexb_bsf.c > when the ctx->length_size == 3 , the filter will report the error > "Invalid argument" patch applied with some modifications Thanks _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From funkyirish at gmail.com Sun Mar 25 03:25:02 2012 From: funkyirish at gmail.com (Josh long) Date: Sat, 24 Mar 2012 20:25:02 -0500 Subject: [FFmpeg-user] bgra to yuv In-Reply-To: References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: > > Alternatively, there should be a launchpad version: > https://launchpad.net/~jon-severinsson/+archive/ffmpeg > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Thanks I've now upgraded to ffmpeg 0.9.1. I tried the libschroedinger codec, but it's not to lossy for me (unless I'm doing it wrong). I ran this code (after compiling of course) #include #include #include main(){ char out[120]; int a; for(a=0;a<25;a++){ sprintf(out,"ffmpeg -i 0libschbgra%d.avi -vcodec libschroedinger -q 0 0libsch%d.avi\n",a,a); system(out); sprintf(out,"ffmpeg -i 0libsch%d.avi -vcodec rawvideo -pix_fmt bgra -q 0 0libschbgra%d.avi\n",a,a+1); system(out); } } When I was finished I could see clear distortion between the last and first videos. I know that ffmpeg has not had lossless enabled in the past, is that still the case? I could really use a tip for lossless compression; I tried converting to yuv444p and then using dirac encoder, but the results wouldn't play correctly. I'm going to run a script to convert back and forth that way and get back here. Thanks again for the help I'm getting. Joshua From guisheng315 at gmail.com Sun Mar 25 05:41:20 2012 From: guisheng315 at gmail.com (gs_gail) Date: Sun, 25 Mar 2012 11:41:20 +0800 Subject: [FFmpeg-user] MPEG-TS trouble In-Reply-To: <1332638486.73254.YahooMailNeo@web162005.mail.bf1.yahoo.com> References: <1332307981.76400.YahooMailNeo@web162003.mail.bf1.yahoo.com> <1332339430.4061.14.camel@gw-laptop> <20120321203842.GB13842@kiste2> <1332638486.73254.YahooMailNeo@web162005.mail.bf1.yahoo.com> Message-ID: <1332646880.2472.0.camel@gw-laptop> do you have a sample file? ? 2012-03-24?? 18:21 -0700?roko??? > Thank you all for your answers. A compiled version from Git save the file now: > > emelo at alfa:~/workspace/ffmpeg> ./ffmpeg -i rtsp://admin:admin at 192.168.99.146/11 -vcodec copy -y -r 25 -vbsf h264_mp4toannexb algo.ts > ffmpeg version N-39223-gac6798d Copyright (c) 2000-2012 the FFmpeg developers > built on Mar 24 2012 19:43:40 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] > configuration: --shlibdir=/usr/lib64 --prefix=/usr --mandir=/usr/share/man --libdir=/usr/lib64 --enable-shared --disable-static --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libxvid --enable-postproc --enable-gpl --enable-x11grab --extra-cflags='-fmessage-length=0 -O2 -Wall -D_FORTIFY_SOURCE=2 -fstack-protector -funwind-tables -fasynchronous-unwind-tables -g -fPIC -I/usr/include/gsm' --enable-debug --disable-stripping --enable-libgsm --enable-libschroedinger --enable-libdirac --enable-avfilter --enable-libvpx --enable-version3 --enable-libx264 --enable-libdc1394 --enable-pthreads --enable-librtmp > libavutil 51. 44.100 / 51. 44.100 > libavcodec 54. 12.100 / 54. 12.100 > libavformat 54. 2.100 / 54. 2.100 > libavdevice 53. 4.100 / 53. 4.100 > libavfilter 2. 65.102 / 2. 65.102 > libswscale 2. 1.100 / 2. 1.100 > libswresample 0. 10.100 / 0. 10.100 > libpostproc 52. 0.100 / 52. 0.100 > [rtsp @ 0x6283a0] Estimating duration from bitrate, this may be inaccurate > Input #0, rtsp, from 'rtsp://admin:admin at 192.168.99.146/11': > Metadata: > title : \11 > Duration: N/A, start: 0.039956, bitrate: N/A > Stream #0:0: Video: h264 (Constrained Baseline), yuv420p, 640x480, 25 tbr, 90k tbn, 180k tbc > [mpegts @ 0x656820] muxrate VBR, pcr every 9000 pkts, sdt every 200, pat/pmt every 40 pkts > Output #0, mpegts, to 'algo.ts': > Metadata: > title : \11 > encoder : Lavf54.2.100 > Stream #0:0: Video: h264, yuv420p, 640x480, q=2-31, 90k tbn, 90k tbc > Stream mapping: > Stream #0:0 -> #0:0 (copy) > Press [q] to stop, [?] for help > > > And the file (algo.ts) plays well with ffplay, but a HUGE (realy... HUGE) amount of : > > Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument > Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument > Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument > Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument > Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument > Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument > Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument > Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument > Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument > > Is generated in the process. Do you consider it "normal" or there is something more to be fixed ? > > E. > > > On Wed, Mar 21, 2012 at 10:17:10PM +0800, gs_gail wrote: > > there is a bug in libavcodec/h264_mp4toannexb_bsf.c > > when the ctx->length_size == 3 , the filter will report the error > > "Invalid argument" > > patch applied with some modifications > > Thanks > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From onemda at gmail.com Sun Mar 25 05:47:50 2012 From: onemda at gmail.com (Paul B Mahol) Date: Sun, 25 Mar 2012 03:47:50 +0000 Subject: [FFmpeg-user] bgra to yuv In-Reply-To: References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: On 3/25/12, Josh long wrote: >> >> Alternatively, there should be a launchpad version: >> https://launchpad.net/~jon-severinsson/+archive/ffmpeg >> >> Carl Eugen >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > Thanks I've now upgraded to ffmpeg 0.9.1. I tried the libschroedinger > codec, but it's not to lossy for me (unless I'm doing it wrong). I ran > this code (after compiling of course) > > #include > #include > #include > > main(){ > char out[120]; > int a; > for(a=0;a<25;a++){ > sprintf(out,"ffmpeg -i 0libschbgra%d.avi -vcodec libschroedinger -q 0 > 0libsch%d.avi\n",a,a); > system(out); > sprintf(out,"ffmpeg -i 0libsch%d.avi -vcodec rawvideo -pix_fmt bgra -q 0 > 0libschbgra%d.avi\n",a,a+1); > system(out); > } > } > When I was finished I could see clear distortion between the last and first > videos. I know that ffmpeg has not had lossless enabled in the past, is > that still the case? For lossless with that codec use "-global_quality 0". From renaux.jacky at orange.fr Sun Mar 25 08:32:09 2012 From: renaux.jacky at orange.fr (jacky) Date: Sun, 25 Mar 2012 08:32:09 +0200 Subject: [FFmpeg-user] sound balance on several files Message-ID: <4F6EBBE9.7020907@orange.fr> Hello I have a streaming server with contains about 100 x 90 minutes flv files, the sound is only talk not music the concern is on volume level which must be quite similar from files to files I can manage through batch files, ffmpeg to demux, modify the volume level and remux each files according to what I undestood, the volume level is next volume = (actual volume level) * k with k = (10 ^ (target_volume / 20)) questions to you are 1- is there a way to know the actual volume level on a given flv file ? could it be done with ffmpeg ? other programs ? 2- what is the average volume level standard (some said 20.1 other 20.3db) (it seems Toshiba PC need a larger volume level than other) many thanks to your answers jacky From eolinwen at gmail.com Sun Mar 25 10:40:51 2012 From: eolinwen at gmail.com (eolinwen at gmail.com) Date: Sun, 25 Mar 2012 10:40:51 +0200 Subject: [FFmpeg-user] Note to document maintainer In-Reply-To: References: <4F6E0715.2010604@att.net> Message-ID: Hi and sorry Carl. I was not aware about that like I have read recently that on Debian Facile ( Easy Debian in English) in their Wiki here : http://wiki.debian-facile.org/manuel:ffmpeg_codage. You could see this precision at the end of the third paragraph =>the remark with the light !!! What is the difference between image2 and mjpeg ? When must we use the both. Thanks and yet sorry. 2012/3/24 Carl Eugen Hoyos > eolinwen gmail.com gmail.com> writes: > > > I am just an new user and far away to be an expert but with > > news version of ffmpeg the option -f image2 is deprecated. > > you should replace this one by -f mjpeg. > > image2 and mjpeg are two different formats, none of them > is deprecated. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Olivier Cenwen un elfe sur la banquise/ an elve on the ice Mon blog perso sur le multim?dia, Ubuntu, Linux et OpenShot : http://linuxevolution.wordpress.com/ Le forum d'Openshot o? vous me trouverez : http://openshotusers.com/ http://openshotusers.com/forum/index.php Nothing is lost until the last second. The family motto : When we want, we can. Astuces, Actualit?s, Logiciels, bref sur tout ce que je ne fais d'articles dessus Google+ From ibantxo28 at gmail.com Sun Mar 25 11:51:43 2012 From: ibantxo28 at gmail.com (Iban Garcia) Date: Sun, 25 Mar 2012 11:51:43 +0200 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: <4F6CA6A1.1050607@yahoo.com> References: <4F6CA6A1.1050607@yahoo.com> Message-ID: First of all, many thanks for your help, Tim. My command is: ffmpeg -y -threads 8 -i input.mxf -b:v 50000000 -s 720x576 -pix_fmt yuv422p -aspect 16:9 -an -r 25 -vf setfield=1 output.dv OUTPUT: # ffmpeg -y -threads 8 -i input.mxf -b:v 50000000 -s 720x576 -pix_fmt yuv422p -aspect 16:9 -an -r 25 output.dv ffmpeg version 0.9.0.git-2169f97 Copyright (c) 2000-2012 the FFmpeg developers built on Jan 10 2012 12:08:32 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] configuration: --shlibdir=/usr/lib64 --prefix=/usr/local --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads --enable-shared --enable-libvorbis --enable-gpl --enable-x11grab --enable-libx264 --enable-libmp3lame --enable-nonfree --enable-postproc libavutil 51. 34.100 / 51. 34.100 libavcodec 53. 54.100 / 53. 54.100 libavformat 53. 29.100 / 53. 29.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 58.100 / 2. 58.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 51. 2.100 / 51. 2.100 [mxf @ 0x6248e0] could not resolve sub descriptor strong ref [mxf @ 0x6248e0] source track 6: stream 4, no descriptor found [mxf @ 0x6248e0] could not resolve sub descriptor strong ref Input #0, mxf, from 'input.mxf': Duration: 00:00:02.32, start: 0.000000, bitrate: 119235 kb/s Stream #0:0: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:1: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:2: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:3: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:4: Data: none Stream #0:5: Video: h264 (High 4:2:2 Intra), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc [buffer @ 0x628180] w:1920 h:1080 pixfmt:yuv422p10le tb:1/1000000 sar:1/1 sws_param: [scale @ 0x62c160] w:1920 h:1080 fmt:yuv422p10le -> w:720 h:576 fmt:yuv422p flags:0x4 Output #0, dv, to 'output.dv': Metadata: encoder : Lavf53.29.100 Stream #0:0: Video: dvvideo, yuv422p, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 50000 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0:5 -> #0:0 (h264 -> dvvideo) Press [q] to stop, [?] for help frame= 58 fps= 0 q=0.0 Lsize= 16312kB time=00:00:02.32 bitrate=57600.0kbits/s video:16312kB audio:0kB global headers:0kB muxing overhead 0.000000% I am testing the "setfield=1" video filter. But I think I have to patch my code first. 2012/3/23 Tim Nicholson > On 23/03/12 14:11, Iban Garcia wrote: > > Hi All! > > > > I am trying to down convert a MXF: It is a Stream #0:5: Video: h264 (High > > 4:2:2 Intra), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, > 25 > > tbn, 50 tbc > > > > It has a MBAFF interlacing inside (Mediainfo tells it). > > > > With ffmpeg I can get a DV, but it is de-interlaced and I want to be > > interlaced as well as the original video is. > > > > How can I get it using ffmpeg? Is there any flag/option that allow it? > > > > You don't provide complete uncut console output. > > However from my experience MXF's always seem to be flagged as progressive > even > when not. > > In which case setfield=1 is your friend. > > If this does not suffice I refer you to my first comment.... > > > -- > Tim > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ibantxo28 at gmail.com Sun Mar 25 11:53:16 2012 From: ibantxo28 at gmail.com (Iban Garcia) Date: Sun, 25 Mar 2012 11:53:16 +0200 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: References: Message-ID: First of all, many thanks for your help, Carl. My command is: ffmpeg -y -threads 8 -i input.mxf -b:v 50000000 -s 720x576 -pix_fmt yuv422p -aspect 16:9 -an -r 25 -vf setfield=1 output.dv OUTPUT: # ffmpeg -y -threads 8 -i input.mxf -b:v 50000000 -s 720x576 -pix_fmt yuv422p -aspect 16:9 -an -r 25 output.dv ffmpeg version 0.9.0.git-2169f97 Copyright (c) 2000-2012 the FFmpeg developers built on Jan 10 2012 12:08:32 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] configuration: --shlibdir=/usr/lib64 --prefix=/usr/local --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads --enable-shared --enable-libvorbis --enable-gpl --enable-x11grab --enable-libx264 --enable-libmp3lame --enable-nonfree --enable-postproc libavutil 51. 34.100 / 51. 34.100 libavcodec 53. 54.100 / 53. 54.100 libavformat 53. 29.100 / 53. 29.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 58.100 / 2. 58.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 51. 2.100 / 51. 2.100 [mxf @ 0x6248e0] could not resolve sub descriptor strong ref [mxf @ 0x6248e0] source track 6: stream 4, no descriptor found [mxf @ 0x6248e0] could not resolve sub descriptor strong ref Input #0, mxf, from 'input.mxf': Duration: 00:00:02.32, start: 0.000000, bitrate: 119235 kb/s Stream #0:0: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:1: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:2: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:3: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:4: Data: none Stream #0:5: Video: h264 (High 4:2:2 Intra), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc [buffer @ 0x628180] w:1920 h:1080 pixfmt:yuv422p10le tb:1/1000000 sar:1/1 sws_param: [scale @ 0x62c160] w:1920 h:1080 fmt:yuv422p10le -> w:720 h:576 fmt:yuv422p flags:0x4 Output #0, dv, to 'output.dv': Metadata: encoder : Lavf53.29.100 Stream #0:0: Video: dvvideo, yuv422p, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 50000 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0:5 -> #0:0 (h264 -> dvvideo) Press [q] to stop, [?] for help frame= 58 fps= 0 q=0.0 Lsize= 16312kB time=00:00:02.32 bitrate=57600.0kbits/s video:16312kB audio:0kB global headers:0kB muxing overhead 0.000000% Original MXF file is interlaced (MBAFF) The DV output has lost all interlacing information. 2012/3/24 Carl Eugen Hoyos > Iban Garcia gmail.com> writes: > > > It has a MBAFF interlacing inside (Mediainfo tells it). > > > > With ffmpeg I can get a DV, but it is de-interlaced and > > I want to be interlaced as well as the original video is. > > Please provide the command line you used and the complete, > uncut console output and please point to a very short sample. > > FFmpeg does no de-interlacing at all if not explicitly told > to do so. > (Otoh, progressive material can be encoded using MBAFF - and > even PAFF - and since the H264 decoder can only output frames, > you will get the - original - progressive material if you > encode the frames coming from the decoder.) > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ibantxo28 at gmail.com Sun Mar 25 16:42:07 2012 From: ibantxo28 at gmail.com (Iban Garcia) Date: Sun, 25 Mar 2012 16:42:07 +0200 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: References: <4F6CA6A1.1050607@yahoo.com> Message-ID: Hi, I have taken from git the last version of ffmpeg. And I have made next test. The output.dv fields are not interlaced. The interlaced information has been lost in the process. # ffmpeg -y -threads 8 -i input.mxf -s 720x576 -pix_fmt yuv411p -aspect 16:9 -an -r 25 -vf "setfield=1, fieldorder=bff" output.dv ffmpeg version N-39247-g6809818 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 25 2012 17:35:24 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] configuration: --shlibdir=/usr/lib64 --prefix=/usr/local --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads --enable-shared --enable-libvorbis --enable-gpl --enable-x11grab --enable-libx264 --enable-libmp3lame --enable-postproc libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 65.102 / 2. 65.102 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 [mxf @ 0x6284e0] could not resolve sub descriptor strong ref [mxf @ 0x6284e0] source track 6: stream 4, no descriptor found [mxf @ 0x6284e0] could not resolve sub descriptor strong ref [mxf @ 0x6284e0] Unknown frame layout type: 1 Input #0, mxf, from 'input.mxf': Metadata: timecode : 11:31:46:14 Duration: 00:00:33.48, start: 0.000000, bitrate: 118911 kb/s Stream #0:0: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:1: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:2: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:3: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:4: Data: none Stream #0:5: Video: h264 (High 4:2:2 Intra), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc [buffer @ 0x719f80] w:1920 h:1080 pixfmt:yuv422p10le tb:1/1000000 sar:1/1 sws_param: [fieldorder @ 0x646e80] output field order: bff [scale @ 0x6283a0] w:1920 h:1080 fmt:yuv422p10le sar:1/1 -> w:720 h:576 fmt:yuv411p sar:64/45 flags:0x4 Output #0, dv, to 'output.dv': Metadata: timecode : 11:31:46:14 encoder : Lavf54.3.100 Stream #0:0: Video: dvvideo, yuv411p, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0:5 -> #0:0 (h264 -> dvvideo) Press [q] to stop, [?] for help frame= 837 fps= 89 q=0.0 Lsize= 117703kB time=00:00:33.48 bitrate=28800.0kbits/s video:117703kB audio:0kB global headers:0kB muxing overhead 0.000000% 2012/3/25 Iban Garcia > First of all, many thanks for your help, Tim. > > My command is: ffmpeg -y -threads 8 -i input.mxf -b:v 50000000 -s 720x576 > -pix_fmt yuv422p -aspect 16:9 -an -r 25 -vf setfield=1 output.dv > > OUTPUT: > # ffmpeg -y -threads 8 -i input.mxf -b:v 50000000 -s 720x576 -pix_fmt > yuv422p -aspect 16:9 -an -r 25 output.dv > ffmpeg version 0.9.0.git-2169f97 Copyright (c) 2000-2012 the FFmpeg > developers > built on Jan 10 2012 12:08:32 with gcc 4.5.1 20101208 [gcc-4_5-branch > revision 167585] > configuration: --shlibdir=/usr/lib64 --prefix=/usr/local > --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads > --enable-shared --enable-libvorbis --enable-gpl --enable-x11grab > --enable-libx264 --enable-libmp3lame --enable-nonfree --enable-postproc > libavutil 51. 34.100 / 51. 34.100 > libavcodec 53. 54.100 / 53. 54.100 > libavformat 53. 29.100 / 53. 29.100 > libavdevice 53. 4.100 / 53. 4.100 > libavfilter 2. 58.100 / 2. 58.100 > libswscale 2. 1.100 / 2. 1.100 > libswresample 0. 6.100 / 0. 6.100 > libpostproc 51. 2.100 / 51. 2.100 > [mxf @ 0x6248e0] could not resolve sub descriptor strong ref > [mxf @ 0x6248e0] source track 6: stream 4, no descriptor found > [mxf @ 0x6248e0] could not resolve sub descriptor strong ref > Input #0, mxf, from 'input.mxf': > Duration: 00:00:02.32, start: 0.000000, bitrate: 119235 kb/s > Stream #0:0: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s > Stream #0:1: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s > Stream #0:2: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s > Stream #0:3: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s > Stream #0:4: Data: none > Stream #0:5: Video: h264 (High 4:2:2 Intra), yuv422p10le, 1920x1080 > [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc > [buffer @ 0x628180] w:1920 h:1080 pixfmt:yuv422p10le tb:1/1000000 sar:1/1 > sws_param: > [scale @ 0x62c160] w:1920 h:1080 fmt:yuv422p10le -> w:720 h:576 > fmt:yuv422p flags:0x4 > Output #0, dv, to 'output.dv': > Metadata: > encoder : Lavf53.29.100 > Stream #0:0: Video: dvvideo, yuv422p, 720x576 [SAR 64:45 DAR 16:9], > q=2-31, 50000 kb/s, 90k tbn, 25 tbc > Stream mapping: > Stream #0:5 -> #0:0 (h264 -> dvvideo) > Press [q] to stop, [?] for help > frame= 58 fps= 0 q=0.0 Lsize= 16312kB time=00:00:02.32 > bitrate=57600.0kbits/s > video:16312kB audio:0kB global headers:0kB muxing overhead 0.000000% > > I am testing the "setfield=1" video filter. But I think I have to patch my > code first. > > > > > 2012/3/23 Tim Nicholson > >> On 23/03/12 14:11, Iban Garcia wrote: >> > Hi All! >> > >> > I am trying to down convert a MXF: It is a Stream #0:5: Video: h264 >> (High >> > 4:2:2 Intra), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 >> tbr, 25 >> > tbn, 50 tbc >> > >> > It has a MBAFF interlacing inside (Mediainfo tells it). >> > >> > With ffmpeg I can get a DV, but it is de-interlaced and I want to be >> > interlaced as well as the original video is. >> > >> > How can I get it using ffmpeg? Is there any flag/option that allow it? >> > >> >> You don't provide complete uncut console output. >> >> However from my experience MXF's always seem to be flagged as progressive >> even >> when not. >> >> In which case setfield=1 is your friend. >> >> If this does not suffice I refer you to my first comment.... >> >> >> -- >> Tim >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > From ibantxo28 at gmail.com Sun Mar 25 16:43:12 2012 From: ibantxo28 at gmail.com (Iban Garcia) Date: Sun, 25 Mar 2012 16:43:12 +0200 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: References: Message-ID: Hi, I have taken from git the last version of ffmpeg. And I have made next test. The output.dv fields are not interlaced. The interlaced information has been lost in the process. # ffmpeg -y -threads 8 -i input.mxf -s 720x576 -pix_fmt yuv411p -aspect 16:9 -an -r 25 -vf "setfield=1, fieldorder=bff" output.dv ffmpeg version N-39247-g6809818 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 25 2012 17:35:24 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] configuration: --shlibdir=/usr/lib64 --prefix=/usr/local --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads --enable-shared --enable-libvorbis --enable-gpl --enable-x11grab --enable-libx264 --enable-libmp3lame --enable-postproc libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 65.102 / 2. 65.102 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 [mxf @ 0x6284e0] could not resolve sub descriptor strong ref [mxf @ 0x6284e0] source track 6: stream 4, no descriptor found [mxf @ 0x6284e0] could not resolve sub descriptor strong ref [mxf @ 0x6284e0] Unknown frame layout type: 1 Input #0, mxf, from 'input.mxf': Metadata: timecode : 11:31:46:14 Duration: 00:00:33.48, start: 0.000000, bitrate: 118911 kb/s Stream #0:0: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:1: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:2: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:3: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:4: Data: none Stream #0:5: Video: h264 (High 4:2:2 Intra), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc [buffer @ 0x719f80] w:1920 h:1080 pixfmt:yuv422p10le tb:1/1000000 sar:1/1 sws_param: [fieldorder @ 0x646e80] output field order: bff [scale @ 0x6283a0] w:1920 h:1080 fmt:yuv422p10le sar:1/1 -> w:720 h:576 fmt:yuv411p sar:64/45 flags:0x4 Output #0, dv, to 'output.dv': Metadata: timecode : 11:31:46:14 encoder : Lavf54.3.100 Stream #0:0: Video: dvvideo, yuv411p, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0:5 -> #0:0 (h264 -> dvvideo) Press [q] to stop, [?] for help frame= 837 fps= 89 q=0.0 Lsize= 117703kB time=00:00:33.48 bitrate=28800.0kbits/s video:117703kB audio:0kB global headers:0kB muxing overhead 0.000000% 2012/3/25 Iban Garcia > First of all, many thanks for your help, Carl. > > My command is: ffmpeg -y -threads 8 -i input.mxf -b:v 50000000 -s 720x576 > -pix_fmt yuv422p -aspect 16:9 -an -r 25 -vf setfield=1 output.dv > > OUTPUT: > # ffmpeg -y -threads 8 -i input.mxf -b:v 50000000 -s 720x576 -pix_fmt > yuv422p -aspect 16:9 -an -r 25 output.dv > ffmpeg version 0.9.0.git-2169f97 Copyright (c) 2000-2012 the FFmpeg > developers > built on Jan 10 2012 12:08:32 with gcc 4.5.1 20101208 [gcc-4_5-branch > revision 167585] > configuration: --shlibdir=/usr/lib64 --prefix=/usr/local > --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads > --enable-shared --enable-libvorbis --enable-gpl --enable-x11grab > --enable-libx264 --enable-libmp3lame --enable-nonfree --enable-postproc > libavutil 51. 34.100 / 51. 34.100 > libavcodec 53. 54.100 / 53. 54.100 > libavformat 53. 29.100 / 53. 29.100 > libavdevice 53. 4.100 / 53. 4.100 > libavfilter 2. 58.100 / 2. 58.100 > libswscale 2. 1.100 / 2. 1.100 > libswresample 0. 6.100 / 0. 6.100 > libpostproc 51. 2.100 / 51. 2.100 > [mxf @ 0x6248e0] could not resolve sub descriptor strong ref > [mxf @ 0x6248e0] source track 6: stream 4, no descriptor found > [mxf @ 0x6248e0] could not resolve sub descriptor strong ref > Input #0, mxf, from 'input.mxf': > Duration: 00:00:02.32, start: 0.000000, bitrate: 119235 kb/s > Stream #0:0: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s > Stream #0:1: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s > Stream #0:2: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s > Stream #0:3: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s > Stream #0:4: Data: none > Stream #0:5: Video: h264 (High 4:2:2 Intra), yuv422p10le, 1920x1080 > [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc > [buffer @ 0x628180] w:1920 h:1080 pixfmt:yuv422p10le tb:1/1000000 sar:1/1 > sws_param: > [scale @ 0x62c160] w:1920 h:1080 fmt:yuv422p10le -> w:720 h:576 > fmt:yuv422p flags:0x4 > Output #0, dv, to 'output.dv': > Metadata: > encoder : Lavf53.29.100 > Stream #0:0: Video: dvvideo, yuv422p, 720x576 [SAR 64:45 DAR 16:9], > q=2-31, 50000 kb/s, 90k tbn, 25 tbc > Stream mapping: > Stream #0:5 -> #0:0 (h264 -> dvvideo) > Press [q] to stop, [?] for help > frame= 58 fps= 0 q=0.0 Lsize= 16312kB time=00:00:02.32 > bitrate=57600.0kbits/s > video:16312kB audio:0kB global headers:0kB muxing overhead 0.000000% > > Original MXF file is interlaced (MBAFF) > The DV output has lost all interlacing information. > > > > > 2012/3/24 Carl Eugen Hoyos > >> Iban Garcia gmail.com> writes: >> >> > It has a MBAFF interlacing inside (Mediainfo tells it). >> > >> > With ffmpeg I can get a DV, but it is de-interlaced and >> > I want to be interlaced as well as the original video is. >> >> Please provide the command line you used and the complete, >> uncut console output and please point to a very short sample. >> >> FFmpeg does no de-interlacing at all if not explicitly told >> to do so. >> (Otoh, progressive material can be encoded using MBAFF - and >> even PAFF - and since the H264 decoder can only output frames, >> you will get the - original - progressive material if you >> encode the frames coming from the decoder.) >> >> Carl Eugen >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > From funkyirish at gmail.com Sun Mar 25 19:38:43 2012 From: funkyirish at gmail.com (Josh long) Date: Sun, 25 Mar 2012 12:38:43 -0500 Subject: [FFmpeg-user] bgra to yuv In-Reply-To: References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: > > > For lossless with that codec use "-global_quality 0". > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Thanks, but it seems that I'm not doing this correctly as there is a lot of distortion again. Could you tell me where I'm going wrong here? I also tried it (first) just using -global_quality 0, and it didn't work well so I added the -q 0 before it. #include #include #include main(){ char out[120]; int a; for(a=0;a<25;a++){ sprintf(out,"ffmpeg -i 0libschbgra%d.avi -vcodec libschroedinger -q 0 -global_quality 0 0libsch%d.avi\n",a,a); system(out); sprintf(out,"ffmpeg -i 0libsch%d.avi -vcodec rawvideo -pix_fmt bgra -q 0 0libschbgra%d.avi\n",a,a+1); system(out); } } thanks again, Joshua From onemda at gmail.com Sun Mar 25 20:32:38 2012 From: onemda at gmail.com (Paul B Mahol) Date: Sun, 25 Mar 2012 18:32:38 +0000 Subject: [FFmpeg-user] bgra to yuv In-Reply-To: References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: On 3/25/12, Josh long wrote: >> >> >> For lossless with that codec use "-global_quality 0". >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > Thanks, but it seems that I'm not doing this correctly as there is a lot of > distortion again. Could you tell me where I'm going wrong here? I also > tried it (first) just using -global_quality 0, and it didn't work well so I > added the -q 0 before it. Converting from one colorspace to another one may not be lossless. So if you do not want distortions at all only colorspace conversions which are actually lossless should be used. > > #include > #include > #include > > main(){ > char out[120]; > int a; > for(a=0;a<25;a++){ > sprintf(out,"ffmpeg -i 0libschbgra%d.avi -vcodec libschroedinger -q 0 > -global_quality 0 0libsch%d.avi\n",a,a); > system(out); > sprintf(out,"ffmpeg -i 0libsch%d.avi -vcodec rawvideo -pix_fmt bgra -q 0 > 0libschbgra%d.avi\n",a,a+1); > system(out); > } > } > > thanks again, > > Joshua > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ffmpeg-user at herveybayaustralia.com.au Mon Mar 26 07:16:32 2012 From: ffmpeg-user at herveybayaustralia.com.au (Da Rock) Date: Mon, 26 Mar 2012 15:16:32 +1000 Subject: [FFmpeg-user] ffmpeg build - how build finds indevs Message-ID: <4F6FFBB0.7000708@herveybayaustralia.com.au> I'm trying to get video4linux working in ffmpeg port on FreeBSD. I'm not getting many answers there, so I thought I'd enquire here. When I run ffmpeg with -f video4linux I get unknown input format: video4linux. Same for v4l2. In the build it only shows up 4 indevs: OSS, BKTR, and 2 others (sorry, lost the screen with it on) - but no video4linux(2). In the configure script there is a disable indevs, but no enable. What ritual do I have to perform to get this to work? Cut off a toe or something? Can anyone point me in the right direction here? Cheers From vrparekh at gmail.com Mon Mar 26 08:07:20 2012 From: vrparekh at gmail.com (vrparekh at gmail.com) Date: Sun, 25 Mar 2012 23:07:20 -0700 (PDT) Subject: [FFmpeg-user] aspect ratio is changed with codec copy and sameq In-Reply-To: References: <1332254972184-4489075.post@n4.nabble.com> <1332262763259-4489550.post@n4.nabble.com> <1332307414992-4491324.post@n4.nabble.com> <201203211931.52053.rodney.baker@iinet.net.au> Message-ID: <1332742040114-4504864.post@n4.nabble.com> i have done like find actual aspect ratio using mediainfo, and pass -aspect x:y as ffmpeg param. Thanks, Vishal Parekh -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/aspect-ratio-is-changed-with-codec-copy-and-sameq-tp4489075p4504864.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From nichot20 at yahoo.com Mon Mar 26 09:29:57 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Mon, 26 Mar 2012 08:29:57 +0100 Subject: [FFmpeg-user] Note to document maintainer In-Reply-To: <4F6E0715.2010604@att.net> References: <4F6E0715.2010604@att.net> Message-ID: <4F701AF5.6030200@yahoo.com> On 24/03/12 17:40, Steve M. Fabac, Jr. wrote: > As a newbie I tried the following example: > > The following example shows how to use ffmpeg for creating > a sequence of files ?img-001.jpeg?, ?img-002.jpeg?, ..., > taking one image every second from the input video: > > ffmpeg -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg' > > So what's the change/addition to create an image every two seconds? the frame rate is specified by the -r, so -r 1 => 1fps. Not sure how well this handles fractional framerates but give it a try. > Every three seconds? The needed switch may be buried in > the overall documentation. Having it in the "section 9.3 image 2" > would save newbie's a lot of searching. > > Also, the single quotes above fail under Windows XP and the correct > command is: ffmpeg -i in.avi -vsync 1 -r 1 -f image2 img-%03d.jpeg > > It is almost impossible to cover every nuance of syntax difference between different OS's and different shells. In particular quoting, and escaping can be particularly thorny. I believe the documentation is, by and large, consistent with bash, and reference is made to the above issues. So for those using other shells you just need to learn the differences. -- Tim From nichot20 at yahoo.com Mon Mar 26 09:43:44 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Mon, 26 Mar 2012 08:43:44 +0100 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: References: Message-ID: <4F701E30.6000500@yahoo.com> On 25/03/12 15:43, Iban Garcia wrote: > Hi, > > I have taken from git the last version of ffmpeg. And I have made next > test. The output.dv fields are not interlaced. The interlaced information > has been lost in the process. > > # ffmpeg -y -threads 8 -i input.mxf -s 720x576 -pix_fmt yuv411p -aspect > 16:9 -an -r 25 -vf "setfield=1, fieldorder=bff" output.dv > [..] > sws_param: > [fieldorder @ 0x646e80] output field order: bff > [scale @ 0x6283a0] w:1920 h:1080 fmt:yuv422p10le sar:1/1 -> w:720 h:576 > fmt:yuv411p sar:64/45 flags:0x4 > As your source is HD and you wish to output SD, you should really scale the material properly using the "scale" filter with interlacing turned on. Something like:- ffmpeg -y -threads 8 -i input.mxf \ -vf "setfield=1, scale=720:576:1, fieldorder=bff" \ -pix_fmt yuv411p -aspect 16:9 -an -r 25 output.dv -- Tim From cehoyos at ag.or.at Mon Mar 26 09:48:01 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 26 Mar 2012 07:48:01 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg build - how build finds indevs References: <4F6FFBB0.7000708@herveybayaustralia.com.au> Message-ID: Da Rock herveybayaustralia.com.au> writes: > I'm trying to get video4linux working in ffmpeg port on FreeBSD. I'm not > getting many answers there, so I thought I'd enquire here. > In the configure script there is a disable indevs, but no enable. They are auto-detected if they are available. > What ritual do I have to perform to get this to work? Search for "check_header linux/videodev2.h" in config.log, the following lines should explain why the device is not auto-detected. Carl Eugen From cehoyos at ag.or.at Mon Mar 26 09:55:38 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 26 Mar 2012 07:55:38 +0000 (UTC) Subject: [FFmpeg-user] Note to document maintainer References: <4F6E0715.2010604@att.net> Message-ID: eolinwen gmail.com gmail.com> writes: > I was not aware about that like I have read recently that > on Debian Facile ( Easy Debian in English) in their Wiki here : > http://wiki.debian-facile.org/manuel:ffmpeg_codage. You > could see this precision at the end of the third paragraph > =>the remark with the light !!! For the command line given there, "-f ..." is not necessary, "fichier_vignette.jpg" signals a jpeg. > What is the difference between image2 and mjpeg ? The "image2" format is for images (!), mjpeg for videos (that consist of jpeg images). Note that you generally don't need to set the output format, it is sufficient to use file suffixes: out.jpg vs out.mjpg Carl Eugen From scottf at tvw.org Mon Mar 26 10:29:59 2012 From: scottf at tvw.org (Scott Freeman) Date: Mon, 26 Mar 2012 01:29:59 -0700 Subject: [FFmpeg-user] overlay w/librtmp Message-ID: Has anyone gotten an overlay image to work when the output is to an rtmp stream? I get no errors on my attempts but the logo simply does not appear over the live video. Thanks. From dave.bevan at bbc.co.uk Mon Mar 26 10:57:44 2012 From: dave.bevan at bbc.co.uk (Dave Bevan) Date: Mon, 26 Mar 2012 09:57:44 +0100 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: <4F701E30.6000500@yahoo.com> References: <4F701E30.6000500@yahoo.com> Message-ID: -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Tim Nicholson Sent: 26 March 2012 08:44 To: FFmpeg user questions and RTFMs Subject: Re: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO On 25/03/12 15:43, Iban Garcia wrote: > Hi, > > I have taken from git the last version of ffmpeg. And I have made next > test. The output.dv fields are not interlaced. The interlaced > information has been lost in the process. > > # ffmpeg -y -threads 8 -i input.mxf -s 720x576 -pix_fmt yuv411p > -aspect > 16:9 -an -r 25 -vf "setfield=1, fieldorder=bff" output.dv > [..] > sws_param: > [fieldorder @ 0x646e80] output field order: bff [scale @ 0x6283a0] > w:1920 h:1080 fmt:yuv422p10le sar:1/1 -> w:720 h:576 fmt:yuv411p > sar:64/45 flags:0x4 > As your source is HD and you wish to output SD, you should really scale the material properly using the "scale" filter with interlacing turned on. Something like:- ffmpeg -y -threads 8 -i input.mxf \ -vf "setfield=1, scale=720:576:1, fieldorder=bff" \ -pix_fmt yuv411p -aspect 16:9 -an -r 25 output.dv ------------------------------------------------------------------------ ------------------------- And you should also ensure you change colourspace too - HD is bt709 whereas SD is bt601. Add this to Tim's filter chain: ,colormatrix=bt709:bt601 --Dave. ------------------------------------------------------------------------ ------------------------- -- Tim _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From nichot20 at yahoo.com Mon Mar 26 11:18:07 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Mon, 26 Mar 2012 10:18:07 +0100 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: References: <4F701E30.6000500@yahoo.com> Message-ID: <4F70344F.1000502@yahoo.com> On 26/03/12 09:57, Dave Bevan wrote: > > [...] > ------------------------------------------------------------------------ > ------------------------- > And you should also ensure you change colourspace too - HD is bt709 > whereas SD is bt601. Add this to Tim's filter chain: > > ,colormatrix=bt709:bt601 > I didn't think colormatrix was in git head yet..... > --Dave. -- Tim From dave.bevan at bbc.co.uk Mon Mar 26 11:22:54 2012 From: dave.bevan at bbc.co.uk (Dave Bevan) Date: Mon, 26 Mar 2012 10:22:54 +0100 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: <4F70344F.1000502@yahoo.com> References: <4F701E30.6000500@yahoo.com> <4F70344F.1000502@yahoo.com> Message-ID: -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Tim Nicholson Sent: 26 March 2012 10:18 To: FFmpeg user questions and RTFMs Subject: Re: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO On 26/03/12 09:57, Dave Bevan wrote: > > [...] > ---------------------------------------------------------------------- > -- > ------------------------- > And you should also ensure you change colourspace too - HD is bt709 > whereas SD is bt601. Add this to Tim's filter chain: > > ,colormatrix=bt709:bt601 > I didn't think colormatrix was in git head yet..... ---------------------------------------------------------------------- Ah... That'll be ffmbc vs ffmpeg then ;-) Personally, for broadcast grade work (such as AVCi to DVCPRO), I would switch from using ffmpeg to ffmbc - http://code.google.com/p/ffmbc/ - it has much broader support for professional formats and for us in our small part of the BBC, meets most of our professional requirements - such as having colorspace conversion support etc, etc. --Dave. ---------------------------------------------------------------------- > --Dave. -- Tim _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From mlefe74 at gmail.com Mon Mar 26 11:10:42 2012 From: mlefe74 at gmail.com (LsBender) Date: Mon, 26 Mar 2012 02:10:42 -0700 (PDT) Subject: [FFmpeg-user] blackframe and blackdetect In-Reply-To: References: <1331653320238-4469396.post@n4.nabble.com> <1331716310370-4471324.post@n4.nabble.com> <20120314093803.GB26897@leki> <1331719651679-4471452.post@n4.nabble.com> <20120314095835.5eb6f895@lrcd.com> <1331826664285-4475432.post@n4.nabble.com> Message-ID: <1332753042896-4505179.post@n4.nabble.com> I can find blackframes on my videos by using this command: ffmpeg -i test.avi -vf blackdetect=d=1:pic_th=0.70:pix_th=0.10 -an -f null - Thanks for your help, Marc -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/blackframe-and-blackdetect-tp4469396p4505179.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From nichot20 at yahoo.com Mon Mar 26 11:33:07 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Mon, 26 Mar 2012 10:33:07 +0100 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: References: <4F701E30.6000500@yahoo.com> <4F70344F.1000502@yahoo.com> Message-ID: <4F7037D3.4010100@yahoo.com> On 26/03/12 10:22, Dave Bevan wrote: > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org > [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Tim Nicholson > Sent: 26 March 2012 10:18 > To: FFmpeg user questions and RTFMs > Subject: Re: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC > Intra) - MBAFF "interlacing" to DVCPRO > > On 26/03/12 09:57, Dave Bevan wrote: >> >> [...] >> ---------------------------------------------------------------------- >> -- >> ------------------------- >> And you should also ensure you change colourspace too - HD is bt709 >> whereas SD is bt601. Add this to Tim's filter chain: >> >> ,colormatrix=bt709:bt601 >> > > I didn't think colormatrix was in git head yet..... > > > ---------------------------------------------------------------------- > Ah... That'll be ffmbc vs ffmpeg then ;-) > > Personally, for broadcast grade work (such as AVCi to DVCPRO), I would > switch from using ffmpeg to ffmbc - http://code.google.com/p/ffmbc/ - it > has much broader support for professional formats and for us in our > small part of the BBC, meets most of our professional requirements - > such as having colorspace conversion support etc, etc. > > --Dave. Although curiously at the moment ffmpeg has a couple of filters that I cannot live without that are not (yet?) in ffmbc...... Btw Cl?ment has submitted a patch to include colorspace, and another one to include mov timecode so the difference is shrinking. -- Tim From scottf at tvw.org Mon Mar 26 11:34:07 2012 From: scottf at tvw.org (Scott Freeman) Date: Mon, 26 Mar 2012 02:34:07 -0700 Subject: [FFmpeg-user] passing commands to filters at runtime to change logo Message-ID: I am interested in changing a logo while encoding. I tried to use the "c" key at runtime and pasted in the string I used for starting the encode "movie=overlays/MyLogo.png [logo]; [in][logo] overlay=10:10 [out]" but I got this error: Parse error, at least 3 arguments were expected, only 1 given in string '-vf Parse error, at least 3 arguments were expected, only 1 given in string Does anyone know the proper way to send a logo change through the filter at runtime? thanks in advance From tevans.uk at googlemail.com Mon Mar 26 12:07:24 2012 From: tevans.uk at googlemail.com (Tom Evans) Date: Mon, 26 Mar 2012 11:07:24 +0100 Subject: [FFmpeg-user] ffmpeg build - how build finds indevs In-Reply-To: <4F6FFBB0.7000708@herveybayaustralia.com.au> References: <4F6FFBB0.7000708@herveybayaustralia.com.au> Message-ID: On Mon, Mar 26, 2012 at 6:16 AM, Da Rock wrote: > I'm trying to get video4linux working in ffmpeg port on FreeBSD. I'm not > getting many answers there, so I thought I'd enquire here. Subscribed to multimedia@, stable@ and current@ - not getting too many questions there.. > > When I run ffmpeg with -f video4linux I get unknown input format: > video4linux. Same for v4l2. > > In the build it only shows up 4 indevs: OSS, BKTR, and 2 others (sorry, lost > the screen with it on) - but no video4linux(2). > > In the configure script there is a disable indevs, but no enable. > > What ritual do I have to perform to get this to work? Cut off a toe or > something? Can anyone point me in the right direction here? > > Cheers What outcome are you trying to get at, what device are you using and what driver are you using with that device? If it is a USB V4L1 device, and you are using cuse4bsd/webcamd, you will need a very recent version of cuse4bsd to translate v4l2 ioctls into v4l1 ioctls that your device will understand. Cheers Tom From andrey.krieger.utkin at gmail.com Mon Mar 26 12:14:34 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Mon, 26 Mar 2012 13:14:34 +0300 Subject: [FFmpeg-user] overlay w/librtmp In-Reply-To: References: Message-ID: 2012/3/26 Scott Freeman : > Has anyone gotten an overlay image to work when the output is to an rtmp > stream? I get no errors on my attempts but the logo simply does not appear > over the live video. In this scenario, rtmp should be irrelevant. 1. Check for newest ffmpeg version. 2. Please try with other overlay input. 3. If it reproduces only with your rtmp input, save it to file and make overlay from that file, and look what happens. -- Andrey Utkin From ibantxo28 at gmail.com Mon Mar 26 12:34:27 2012 From: ibantxo28 at gmail.com (Iban Garcia) Date: Mon, 26 Mar 2012 12:34:27 +0200 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: <4F701E30.6000500@yahoo.com> References: <4F701E30.6000500@yahoo.com> Message-ID: Umm... it does not work for me. The output.dv is still "progressive" (de-interlaced): both fields of output.dv file are equal. The interlacing has been lost... I have download and installed "ffmbc". I am doing some test now with it. But, by the moment, all of the tests has the same "de-interlaced" output... :-((( 2012/3/26 Tim Nicholson > On 25/03/12 15:43, Iban Garcia wrote: > > Hi, > > > > I have taken from git the last version of ffmpeg. And I have made next > > test. The output.dv fields are not interlaced. The interlaced information > > has been lost in the process. > > > > # ffmpeg -y -threads 8 -i input.mxf -s 720x576 -pix_fmt yuv411p -aspect > > 16:9 -an -r 25 -vf "setfield=1, fieldorder=bff" output.dv > > > [..] > > > sws_param: > > [fieldorder @ 0x646e80] output field order: bff > > [scale @ 0x6283a0] w:1920 h:1080 fmt:yuv422p10le sar:1/1 -> w:720 h:576 > > fmt:yuv411p sar:64/45 flags:0x4 > > > > As your source is HD and you wish to output SD, you should really scale the > material properly using the "scale" filter with interlacing turned on. > > Something like:- > > ffmpeg -y -threads 8 -i input.mxf \ > -vf "setfield=1, scale=720:576:1, fieldorder=bff" \ > -pix_fmt yuv411p -aspect 16:9 -an -r 25 output.dv > > -- > Tim > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ibantxo28 at gmail.com Mon Mar 26 12:41:34 2012 From: ibantxo28 at gmail.com (Iban Garcia) Date: Mon, 26 Mar 2012 12:41:34 +0200 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: References: <4F701E30.6000500@yahoo.com> Message-ID: Hi Dave, Umm... it does not work for me. The output.dv is still "progressive" (de-interlaced): both fields of output.dv file are equal. The interlacing has been lost... I have download and installed "ffmbc". I am doing some test now with it. But, by the moment, all of the tests has the same "de-interlaced" output... :-((( # ffmpeg -y -threads 8 -i input.mxf -vf "setfield=1, scale=720:576:1, fieldorder=bff" -pix_fmt yuv411p -aspect 16:9 -an -r 25 output.dv ffmpeg version N-39247-g6809818 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 25 2012 17:35:24 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] configuration: --shlibdir=/usr/lib64 --prefix=/usr/local --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads --enable-shared --enable-libvorbis --enable-gpl --enable-x11grab --enable-libx264 --enable-libmp3lame --enable-postproc libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 65.102 / 2. 65.102 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 [mxf @ 0x6284e0] could not resolve sub descriptor strong ref [mxf @ 0x6284e0] source track 6: stream 4, no descriptor found [mxf @ 0x6284e0] could not resolve sub descriptor strong ref [mxf @ 0x6284e0] Unknown frame layout type: 1 Input #0, mxf, from 'input.mxf': Metadata: timecode : 11:31:46:14 Duration: 00:00:33.48, start: 0.000000, bitrate: 118911 kb/s Stream #0:0: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:1: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:2: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:3: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:4: Data: none Stream #0:5: Video: h264 (High 4:2:2 Intra), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc [buffer @ 0x719f80] w:1920 h:1080 pixfmt:yuv422p10le tb:1/1000000 sar:1/1 sws_param: [fieldorder @ 0x646f80] output field order: bff [scale @ 0x62fbc0] w:1920 h:1080 fmt:yuv422p10le sar:1/1 -> w:720 h:576 fmt:yuv411p sar:64/45 flags:0x4 Output #0, dv, to 'output.dv': Metadata: timecode : 11:31:46:14 encoder : Lavf54.3.100 Stream #0:0: Video: dvvideo, yuv411p, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0:5 -> #0:0 (h264 -> dvvideo) Press [q] to stop, [?] for help frame= 837 fps= 88 q=0.0 Lsize= 117703kB time=00:00:33.48 bitrate=28800.0kbits/s video:117703kB audio:0kB global headers:0kB muxing overhead 0.000000% 2012/3/26 Dave Bevan > > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org > [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Tim Nicholson > Sent: 26 March 2012 08:44 > To: FFmpeg user questions and RTFMs > Subject: Re: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC > Intra) - MBAFF "interlacing" to DVCPRO > > On 25/03/12 15:43, Iban Garcia wrote: > > Hi, > > > > I have taken from git the last version of ffmpeg. And I have made next > > > test. The output.dv fields are not interlaced. The interlaced > > information has been lost in the process. > > > > # ffmpeg -y -threads 8 -i input.mxf -s 720x576 -pix_fmt yuv411p > > -aspect > > 16:9 -an -r 25 -vf "setfield=1, fieldorder=bff" output.dv > > > [..] > > > sws_param: > > [fieldorder @ 0x646e80] output field order: bff [scale @ 0x6283a0] > > w:1920 h:1080 fmt:yuv422p10le sar:1/1 -> w:720 h:576 fmt:yuv411p > > sar:64/45 flags:0x4 > > > > As your source is HD and you wish to output SD, you should really scale > the material properly using the "scale" filter with interlacing turned > on. > > Something like:- > > ffmpeg -y -threads 8 -i input.mxf \ > -vf "setfield=1, scale=720:576:1, fieldorder=bff" \ -pix_fmt yuv411p > -aspect 16:9 -an -r 25 output.dv > > > ------------------------------------------------------------------------ > ------------------------- > And you should also ensure you change colourspace too - HD is bt709 > whereas SD is bt601. Add this to Tim's filter chain: > > ,colormatrix=bt709:bt601 > > --Dave. > ------------------------------------------------------------------------ > ------------------------- > > > > -- > Tim > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > http://www.bbc.co.uk/ > This e-mail (and any attachments) is confidential and may contain personal > views which are not the views of the BBC unless specifically stated. > If you have received it in error, please delete it from your system. > Do not use, copy or disclose the information in any way nor act in > reliance on it and notify the sender immediately. > Please note that the BBC monitors e-mails sent or received. > Further communication will signify your consent to this. > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ebisumartin at gmail.com Mon Mar 26 13:01:44 2012 From: ebisumartin at gmail.com (Martin G) Date: Mon, 26 Mar 2012 20:01:44 +0900 Subject: [FFmpeg-user] -t option is ignored Message-ID: FFMPEG users, I am trying to edit a subclip from a larger movie. I just want a two minute section. This is the command I have been working with: ffmpeg -i input.avi -ss 01:05:16 -t 120 -vcodec copy -acodec copy output.avi However, instead of getting a two minute (120 second) clip, I get a clip that starts at the selected time, and runs all the way to the end of the film (about 45 minutes long). What is wrong with this command? How can I get ffmpeg to only output a specific time length of video? Thank you. -- Martin From cehoyos at ag.or.at Mon Mar 26 13:46:52 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 26 Mar 2012 11:46:52 +0000 (UTC) Subject: [FFmpeg-user] -t option is ignored References: Message-ID: Martin G gmail.com> writes: > ffmpeg -i input.avi -ss 01:05:16 -t 120 -vcodec copy -acodec copy output.avi (Complete, uncut console output missing.) -t does not work well with -vcodec copy -acodec copy. Carl Eugen From Donald.McLachlan at crc.ca Mon Mar 26 15:13:22 2012 From: Donald.McLachlan at crc.ca (Donald McLachlan) Date: Mon, 26 Mar 2012 09:13:22 -0400 Subject: [FFmpeg-user] Note to document maintainer In-Reply-To: <4F6E0715.2010604@att.net> References: <4F6E0715.2010604@att.net> Message-ID: <4F706B72.2000804@crc.ca> On 24/03/2012 1:40 PM, Steve M. Fabac, Jr. wrote: > As a newbie I tried the following example: > > The following example shows how to use ffmpeg for creating > a sequence of files ?img-001.jpeg?, ?img-002.jpeg?, ..., > taking one image every second from the input video: > > ffmpeg -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg' > > So what's the change/addition to create an image every two seconds? > Every three seconds? The needed switch may be buried in > the overall documentation. Having it in the "section 9.3 image 2" > would save newbie's a lot of searching. > > Also, the single quotes above fail under Windows XP and the correct > command is: ffmpeg -i in.avi -vsync 1 -r 1 -f image2 img-%03d.jpeg > > I have recently used -f image2 to extract a video to a series of pegs and it works just fine. In the example you provided, "-r 1" sets the output frame rate to be 1 frame per second. Did you try "-r 0.5" for one frame every 2 seconds? If your video is 30 frames/sec I guess you could extract all the frames and then hand pick the ones you want from the lot of them. From osoleil at ubisoft.fr Mon Mar 26 15:00:55 2012 From: osoleil at ubisoft.fr (Olivier Soleil) Date: Mon, 26 Mar 2012 06:00:55 -0700 (PDT) Subject: [FFmpeg-user] How can I replace numbered sound tracks in my AVI ? In-Reply-To: References: <524EA12FEC145E4E86154A23DA429D2D5DE588AFD3@PDC-MAIL-CMS01.ubisoft.org> Message-ID: <1332766855393-4505694.post@n4.nabble.com> Hello and thank you. /"The important thing is not that you need movies for a game, but the technology you want to use"/ You're right, i forgot to mention my containers/codecs. So far, i'd like to produce AVI / H264 and probably MOV / H264. And I take for granted that these containers/formats can use multiple audio tracks. The problem is i cannot associate an audio stream to any track number I want to, considering FFMPEG's doc. Would this feature be feasible, I'd like to see that developed within FFMPEG. I add that track 0 will always be a video stream. /"Remux the AVI, this does not (necessarily) change the data."/ On the other hand, do you really imply that one can replace the audio tracks and/or the video track of an AVI/H264 (or a MOV/H264 ?) via FFMPEG without creating a new file every time ? That would prove helpful to us. Thanks again -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-can-I-replace-numbered-sound-tracks-in-my-AVI-tp4469444p4505694.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From nichot20 at yahoo.com Mon Mar 26 15:58:13 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Mon, 26 Mar 2012 14:58:13 +0100 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: References: <4F701E30.6000500@yahoo.com> Message-ID: <4F7075F5.5090403@yahoo.com> On 26/03/12 11:34, Iban Garcia wrote: > Umm... it does not work for me. The output.dv is still "progressive" > (de-interlaced): both fields of output.dv file are equal. The interlacing > has been lost... > OK so something is not working as it should. I have had a look at some of my H264 10bit interlaced material and that is also behaving as you describe. The source is definitely interlaced, as can be seen from the interlace artifacts in a resized ffplay window. But the output is a blend of the two fields. I will try and get some more details and then raise a bug report. > I have download and installed "ffmbc". I am doing some test now with it. > But, by the moment, all of the tests has the same "de-interlaced" output... > :-((( > > > 2012/3/26 Tim Nicholson > >> On 25/03/12 15:43, Iban Garcia wrote: >>> Hi, >>> >>> I have taken from git the last version of ffmpeg. And I have made next >>> test. The output.dv fields are not interlaced. The interlaced information >>> has been lost in the process. >>> >>> # ffmpeg -y -threads 8 -i input.mxf -s 720x576 -pix_fmt yuv411p -aspect >>> 16:9 -an -r 25 -vf "setfield=1, fieldorder=bff" output.dv >> >>> [..] >> >>> sws_param: >>> [fieldorder @ 0x646e80] output field order: bff >>> [scale @ 0x6283a0] w:1920 h:1080 fmt:yuv422p10le sar:1/1 -> w:720 h:576 >>> fmt:yuv411p sar:64/45 flags:0x4 >>> >> >> As your source is HD and you wish to output SD, you should really scale the >> material properly using the "scale" filter with interlacing turned on. >> >> Something like:- >> >> ffmpeg -y -threads 8 -i input.mxf \ >> -vf "setfield=1, scale=720:576:1, fieldorder=bff" \ >> -pix_fmt yuv411p -aspect 16:9 -an -r 25 output.dv >> >> -- >> Tim >> _______________________________________________ -- Tim From rodney.baker at iinet.net.au Mon Mar 26 15:58:38 2012 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Tue, 27 Mar 2012 00:28:38 +1030 Subject: [FFmpeg-user] -t option is ignored In-Reply-To: References: Message-ID: <201203270028.38628.rodney.baker@iinet.net.au> On Mon, 26 Mar 2012 21:31:44 Martin G wrote: > FFMPEG users, > > I am trying to edit a subclip from a larger movie. I just want a two > minute section. > > This is the command I have been working with: > > ffmpeg -i input.avi -ss 01:05:16 -t 120 -vcodec copy -acodec copy > output.avi > In my experience it is faster to put -ss and -t options before -i, otherwise ffmpeg parses the whole file up to -ss before starting conversion. With -ss first, it begins reading at that point (or the closest i-frame or reference frame). Unless, of course, the behaviour has changed in recent versions but I have developed the habit of always putting the seek and duration options first. (YMMV). > However, instead of getting a two minute (120 second) clip, I get a > clip that starts at the selected time, and runs all the way to the end > of the film (about 45 minutes long). > > What is wrong with this command? > > How can I get ffmpeg to only output a specific time length of video? > > Thank you. -- ========================================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au ========================================================================== From ibantxo28 at gmail.com Mon Mar 26 16:29:20 2012 From: ibantxo28 at gmail.com (Iban Garcia) Date: Mon, 26 Mar 2012 16:29:20 +0200 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: <4F7075F5.5090403@yahoo.com> References: <4F701E30.6000500@yahoo.com> <4F7075F5.5090403@yahoo.com> Message-ID: Exact!! But the issue *has been solved* using next: -vf "scale=720:576:interl=1,fieldorder=tff" -vcodec dvvideo I think ffmpeg should "detect" the interlaced without "interl=1" option. Many thanks, Tim!!! 2012/3/26 Tim Nicholson > On 26/03/12 11:34, Iban Garcia wrote: > > Umm... it does not work for me. The output.dv is still "progressive" > > (de-interlaced): both fields of output.dv file are equal. The interlacing > > has been lost... > > > > OK so something is not working as it should. I have had a look at some of > my > H264 10bit interlaced material and that is also behaving as you describe. > The > source is definitely interlaced, as can be seen from the interlace > artifacts in > a resized ffplay window. But the output is a blend of the two fields. > > I will try and get some more details and then raise a bug report. > > > I have download and installed "ffmbc". I am doing some test now with it. > > But, by the moment, all of the tests has the same "de-interlaced" > output... > > :-((( > > > > > > 2012/3/26 Tim Nicholson > > > >> On 25/03/12 15:43, Iban Garcia wrote: > >>> Hi, > >>> > >>> I have taken from git the last version of ffmpeg. And I have made next > >>> test. The output.dv fields are not interlaced. The interlaced > information > >>> has been lost in the process. > >>> > >>> # ffmpeg -y -threads 8 -i input.mxf -s 720x576 -pix_fmt yuv411p -aspect > >>> 16:9 -an -r 25 -vf "setfield=1, fieldorder=bff" output.dv > >> > >>> [..] > >> > >>> sws_param: > >>> [fieldorder @ 0x646e80] output field order: bff > >>> [scale @ 0x6283a0] w:1920 h:1080 fmt:yuv422p10le sar:1/1 -> w:720 h:576 > >>> fmt:yuv411p sar:64/45 flags:0x4 > >>> > >> > >> As your source is HD and you wish to output SD, you should really scale > the > >> material properly using the "scale" filter with interlacing turned on. > >> > >> Something like:- > >> > >> ffmpeg -y -threads 8 -i input.mxf \ > >> -vf "setfield=1, scale=720:576:1, fieldorder=bff" \ > >> -pix_fmt yuv411p -aspect 16:9 -an -r 25 output.dv > >> > >> -- > >> Tim > >> _______________________________________________ > > -- > Tim > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From nichot20 at yahoo.com Mon Mar 26 16:30:55 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Mon, 26 Mar 2012 15:30:55 +0100 Subject: [FFmpeg-user] -t option is ignored In-Reply-To: <201203270028.38628.rodney.baker@iinet.net.au> References: <201203270028.38628.rodney.baker@iinet.net.au> Message-ID: <4F707D9F.6000905@yahoo.com> On 26/03/12 14:58, Rodney Baker wrote: > On Mon, 26 Mar 2012 21:31:44 Martin G wrote: >> FFMPEG users, >> >> I am trying to edit a subclip from a larger movie. I just want a two >> minute section. >> >> This is the command I have been working with: >> >> ffmpeg -i input.avi -ss 01:05:16 -t 120 -vcodec copy -acodec copy >> output.avi >> > > In my experience it is faster to put -ss and -t options before -i, otherwise > ffmpeg parses the whole file up to -ss before starting conversion. With -ss > first, it begins reading at that point (or the closest i-frame or reference > frame). > faster, yes, but sometimes less accurate..... > Unless, of course, the behaviour has changed in recent versions but I have > developed the habit of always putting the seek and duration options first. > > [...] > -- Tim From scottf at tvw.org Mon Mar 26 16:34:33 2012 From: scottf at tvw.org (Scott Freeman) Date: Mon, 26 Mar 2012 07:34:33 -0700 Subject: [FFmpeg-user] overlay w/librtmp In-Reply-To: References: Message-ID: Thanks for the reply, I figured out what was going on (running 0.10). Evidently you cannot use the "-acodec copy -vcodec copy" for the codec setting from the source media for your output. The overlay filter will throw no errors, but will not work. You have to specify the output codec parameters (even if they are the same as the inputs) to get the overlay to work. I would stretch to call this a bug or an oversight but I think I understand the workflow and the logic there being a programmer. It does work now. I hope that this could though be at least documented or updated in the next revision. Thanks On Mon, Mar 26, 2012 at 3:14 AM, Andrey Utkin < andrey.krieger.utkin at gmail.com> wrote: > 2012/3/26 Scott Freeman : > > Has anyone gotten an overlay image to work when the output is to an rtmp > > stream? I get no errors on my attempts but the logo simply does not > appear > > over the live video. > > In this scenario, rtmp should be irrelevant. > 1. Check for newest ffmpeg version. > 2. Please try with other overlay input. > 3. If it reproduces only with your rtmp input, save it to file and > make overlay from that file, and look what happens. > > -- > Andrey Utkin > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From tevans.uk at googlemail.com Mon Mar 26 16:54:55 2012 From: tevans.uk at googlemail.com (Tom Evans) Date: Mon, 26 Mar 2012 15:54:55 +0100 Subject: [FFmpeg-user] overlay w/librtmp In-Reply-To: References: Message-ID: On Mon, Mar 26, 2012 at 3:34 PM, Scott Freeman wrote: > Thanks for the reply, I figured out what was going on (running 0.10). > Evidently you cannot use the "-acodec copy -vcodec copy" for the codec > setting from the source media for your output. The overlay filter will > throw no errors, but will not work. You have to specify the output codec > parameters (even if they are the same as the inputs) to get the overlay to > work. > I would stretch to call this a bug or an oversight but I think I understand > the workflow and the logic there being a programmer. > It does work now. I hope that this could though be at least documented or > updated in the next revision. > > It's definitely not a bug - how can ffmpeg render a logo on the output video when you tell it to copy the video data? "-vcodec copy" literally means "copy, byte for byte, the data in the video stream", it definitely does not mean "re-encode the video using the same parameters it was originally encoded with" (for a start, most of the parameters that it was encoded with cannot even be known at that point!) Cheers Tom From nichot20 at yahoo.com Mon Mar 26 17:06:12 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Mon, 26 Mar 2012 16:06:12 +0100 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: References: <4F701E30.6000500@yahoo.com> <4F7075F5.5090403@yahoo.com> Message-ID: <4F7085E4.8080609@yahoo.com> On 26/03/12 15:29, Iban Garcia wrote: > Exact!! > > But the issue *has been solved* using next: > -vf "scale=720:576:interl=1,fieldorder=tff" -vcodec dvvideo > Ahh, the difference between scale=720:576:1 and scale=720:576:interl=1 !! I think the documentation is unclear on the fact that for width and height you just specify an expression, but for interlace you need to specify the parameter as well. *But* your command, as above is setting the output field order to tff when DV is bff so you will get motion judder. I really think you should use:- ...vf "setfield=1, scale=720:576:interl=1, fieldorder=bff"... or:- -vf "setfield=0, scale=720:576:interl=1"... if your source is already bottom field first (unlikely). > I think ffmpeg should "detect" the interlaced without "interl=1" option. > That option has nothing to do with detection, its how the scaling filter should work. ffmpeg *can* detect interlace for which you set interl=-1 to let the filter decide automatically how to behave, however by default I believe it is set to 0, i.e off. *However* as I mentioned previously. MXF material seems to be always detected as progressive even when it is not. It is not just ffmpeg that makes this mistake. The way you detect interlace in MXF seems to be different to most other formats. Hence the forcing of the interlace flag by setfield, followed by telling the scaler to use interlaced scaling. > Many thanks, Tim!!! > > > 2012/3/26 Tim Nicholson > >> On 26/03/12 11:34, Iban Garcia wrote: >>> Umm... it does not work for me. The output.dv is still "progressive" >>> (de-interlaced): both fields of output.dv file are equal. The interlacing >>> has been lost... >>> >> >> OK so something is not working as it should. I have had a look at some of >> my >> H264 10bit interlaced material and that is also behaving as you describe. >> The >> source is definitely interlaced, as can be seen from the interlace >> artifacts in >> a resized ffplay window. But the output is a blend of the two fields. >> >> I will try and get some more details and then raise a bug report. >> >>> I have download and installed "ffmbc". I am doing some test now with it. >>> But, by the moment, all of the tests has the same "de-interlaced" >> output... >>> :-((( >>> >>> >>> 2012/3/26 Tim Nicholson >>> >>>> On 25/03/12 15:43, Iban Garcia wrote: >>>>> Hi, >>>>> >>>>> I have taken from git the last version of ffmpeg. And I have made next >>>>> test. The output.dv fields are not interlaced. The interlaced >> information >>>>> has been lost in the process. >>>>> >>>>> # ffmpeg -y -threads 8 -i input.mxf -s 720x576 -pix_fmt yuv411p -aspect >>>>> 16:9 -an -r 25 -vf "setfield=1, fieldorder=bff" output.dv >>>> >>>>> [..] >>>> >>>>> sws_param: >>>>> [fieldorder @ 0x646e80] output field order: bff >>>>> [scale @ 0x6283a0] w:1920 h:1080 fmt:yuv422p10le sar:1/1 -> w:720 h:576 >>>>> fmt:yuv411p sar:64/45 flags:0x4 >>>>> >>>> >>>> As your source is HD and you wish to output SD, you should really scale >> the >>>> material properly using the "scale" filter with interlacing turned on. >>>> >>>> Something like:- >>>> >>>> ffmpeg -y -threads 8 -i input.mxf \ >>>> -vf "setfield=1, scale=720:576:1, fieldorder=bff" \ >>>> -pix_fmt yuv411p -aspect 16:9 -an -r 25 output.dv >>>> >>>> -- >>>> Tim >>>> _______________________________________________ >> >> -- >> Tim >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Tim From scottf at tvw.org Mon Mar 26 19:19:06 2012 From: scottf at tvw.org (Scott Freeman) Date: Mon, 26 Mar 2012 10:19:06 -0700 Subject: [FFmpeg-user] overlay w/librtmp In-Reply-To: References: Message-ID: Tom, thanks for the clarification. Like I mentioned, I thought it would be a stretch to call it a bug. And you confirmed my guess on how "copy" worked, thank you. I am curious though about what you said and this may be a topic for another thread. Why couldn't ffmpeg be able to detect the input video information and create a "similar" output string to essentially copy the inputs settings? I could surely do that or something similar with a script and some program logic to generate a output string before starting the encode. I think it would make for a good feature built-in, especially when users sometimes are just re-encoding a video to do effects and overlays and such. Am I way off here? Thanks again Tom, for taking the time to reply to my post. Cheers, Scott On Mon, Mar 26, 2012 at 7:54 AM, Tom Evans wrote: > On Mon, Mar 26, 2012 at 3:34 PM, Scott Freeman wrote: > > Thanks for the reply, I figured out what was going on (running 0.10). > > Evidently you cannot use the "-acodec copy -vcodec copy" for the codec > > setting from the source media for your output. The overlay filter will > > throw no errors, but will not work. You have to specify the output codec > > parameters (even if they are the same as the inputs) to get the overlay > to > > work. > > I would stretch to call this a bug or an oversight but I think I > understand > > the workflow and the logic there being a programmer. > > It does work now. I hope that this could though be at least documented or > > updated in the next revision. > > > > > > It's definitely not a bug - how can ffmpeg render a logo on the output > video when you tell it to copy the video data? > > "-vcodec copy" literally means "copy, byte for byte, the data in the > video stream", it definitely does not mean "re-encode the video using > the same parameters it was originally encoded with" (for a start, most > of the parameters that it was encoded with cannot even be known at > that point!) > > Cheers > > Tom > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ubitux at gmail.com Mon Mar 26 23:38:14 2012 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Mon, 26 Mar 2012 23:38:14 +0200 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: <4F7037D3.4010100@yahoo.com> References: <4F701E30.6000500@yahoo.com> <4F70344F.1000502@yahoo.com> <4F7037D3.4010100@yahoo.com> Message-ID: <20120326213814.GG5169@leki> On Mon, Mar 26, 2012 at 10:33:07AM +0100, Tim Nicholson wrote: > On 26/03/12 10:22, Dave Bevan wrote: > > -----Original Message----- > > From: ffmpeg-user-bounces at ffmpeg.org > > [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Tim Nicholson > > Sent: 26 March 2012 10:18 > > To: FFmpeg user questions and RTFMs > > Subject: Re: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC > > Intra) - MBAFF "interlacing" to DVCPRO > > > > On 26/03/12 09:57, Dave Bevan wrote: > >> > >> [...] > >> ---------------------------------------------------------------------- > >> -- > >> ------------------------- > >> And you should also ensure you change colourspace too - HD is bt709 > >> whereas SD is bt601. Add this to Tim's filter chain: > >> > >> ,colormatrix=bt709:bt601 > >> > > > > I didn't think colormatrix was in git head yet..... > > > > > > ---------------------------------------------------------------------- > > Ah... That'll be ffmbc vs ffmpeg then ;-) > > > > Personally, for broadcast grade work (such as AVCi to DVCPRO), I would > > switch from using ffmpeg to ffmbc - http://code.google.com/p/ffmbc/ - it > > has much broader support for professional formats and for us in our > > small part of the BBC, meets most of our professional requirements - > > such as having colorspace conversion support etc, etc. > > > > --Dave. > > Although curiously at the moment ffmpeg has a couple of filters that I cannot > live without that are not (yet?) in ffmbc...... > > Btw Cl?ment has submitted a patch to include colorspace, and another one to Nope I just did a review of the recently submitted colormatrix filter :) > include mov timecode so the difference is shrinking. > Yep, but need to be fixed as you pointed it out, but as usual time is the biggest issue. -- Cl?ment B. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 490 bytes Desc: not available URL: From brendan.brewster at gmail.com Tue Mar 27 01:22:39 2012 From: brendan.brewster at gmail.com (Brendan Brewster) Date: Mon, 26 Mar 2012 19:22:39 -0400 Subject: [FFmpeg-user] Compressibility Check Message-ID: Hi, I am trying to implement a compressibility check via ffmpeg to determine an appropriate resolution for the full encode. My script is currently determining the interval and length of snippets to test with and then running a sequence of calls to ffmpeg to gather a total size for the samples as follows: $ ffmpeg -i VTS_01_PGC_01_1.VOB -ss 0 -t 9 -filter:v yadif -target ntsc-dvd -f mpeg2video -mbd rd -trellis 1 -cmp 2 -subcmp 2 -g 15 - -an 2>/dev/null | wc -c $ ffmpeg -i VTS_01_PGC_01_1.VOB -ss 179 -t 9 -filter:v yadif -target ntsc-dvd -f mpeg2video -mbd rd -trellis 1 -cmp 2 -subcmp 2 -g 15 - -an 2>/dev/null | wc -c ... (and so on) This works but the problem with this really stems from how long it takes to seek and process subsequent segments of the input especially as we approach the latter portion. Is there a better way encode multiple segments like this, say, in one pass? Any other ideas or feedback? Thanks, Brendan From funkyirish at gmail.com Tue Mar 27 04:12:52 2012 From: funkyirish at gmail.com (Josh long) Date: Mon, 26 Mar 2012 21:12:52 -0500 Subject: [FFmpeg-user] bgra to yuv In-Reply-To: References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: > > > > Converting from one colorspace to another one may not be lossless. > > So if you do not want distortions at all only colorspace conversions which > are > actually lossless should be used. > > Aah, I've realized that ffmpeg first converts to yuv420 and then employs the libschroedinger codec. For some reason I've been under the impression that yuv444p and bgra are lossless equivalents, so now I use the following code: #include #include #include main(){ char out[120]; int a; for(a=0;a<25;a++){ sprintf(out,"ffmpeg -i 0libschbgra%d.avi -vcodec rawvideo -pix_fmt yuv444p -q 0 -global_quality 0 0libsch444%d.avi\n",a,a); system(out); sprintf(out,"ffmpeg -i 0libsch444%d.avi -vcodec libschroedinger -q 0 -global_quality 0 0libsch%d.avi\n",a,a); system(out); sprintf(out,"ffmpeg -i 0libsch%d.avi -vcodec rawvideo -pix_fmt bgra -q 0 0libschbgra%d.avi\n",a,a+1); system(out); } } This ends up in garbage as well so I'm going to try to convert to yuv444p once and then stay there; I'll post that when it finishes. However I've noticed that after converting from bgra to yuv444p my video is all grayscale and split into 9 sections which consist of 3 seemingly repeated columns of three different rows. Is this a bug in vlc, or am I just doing this wrong? Thanks for the help again, I think this is getting closer. I'm also going to try converting to yuva444p, and then yuv444p in case that works better. Joshua From onemda at gmail.com Tue Mar 27 04:29:13 2012 From: onemda at gmail.com (Paul B Mahol) Date: Tue, 27 Mar 2012 02:29:13 +0000 Subject: [FFmpeg-user] bgra to yuv In-Reply-To: References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: On 3/27/12, Josh long wrote: >> >> >> >> Converting from one colorspace to another one may not be lossless. >> >> So if you do not want distortions at all only colorspace conversions which >> are >> actually lossless should be used. >> >> Aah, I've realized that ffmpeg first converts to yuv420 and then employs > the libschroedinger codec. For some reason I've been under the impression > that yuv444p and bgra are lossless equivalents, so now I use the following > code: libschroedinger encoder in FFmpeg supports only yuv420p, yuv422p and yuv444p. If your input is not one of this above you can not expect encoding to be lossless. > > #include > #include > #include > > main(){ > char out[120]; > int a; > for(a=0;a<25;a++){ > sprintf(out,"ffmpeg -i 0libschbgra%d.avi -vcodec rawvideo -pix_fmt > yuv444p -q 0 -global_quality 0 0libsch444%d.avi\n",a,a); > system(out); > sprintf(out,"ffmpeg -i 0libsch444%d.avi -vcodec libschroedinger -q 0 > -global_quality 0 0libsch%d.avi\n",a,a); > system(out); > sprintf(out,"ffmpeg -i 0libsch%d.avi -vcodec rawvideo -pix_fmt bgra -q 0 > 0libschbgra%d.avi\n",a,a+1); > system(out); > } > } > > This ends up in garbage as well so I'm going to try to convert to yuv444p > once and then stay there; I'll post that when it finishes. However I've > noticed that after converting from bgra to yuv444p my video is all > grayscale and split into 9 sections which consist of 3 seemingly repeated > columns of three different rows. Is this a bug in vlc, or am I just doing > this wrong? > Thanks for the help again, I think this is getting closer. > > I'm also going to try converting to yuva444p, and then yuv444p in case that > works better. > > Joshua > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From funkyirish at gmail.com Tue Mar 27 04:35:34 2012 From: funkyirish at gmail.com (Josh long) Date: Mon, 26 Mar 2012 21:35:34 -0500 Subject: [FFmpeg-user] bgra to yuv In-Reply-To: References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: > > libschroedinger encoder in FFmpeg supports only yuv420p, yuv422p and > yuv444p. > > If your input is not one of this above you can not expect encoding to be > lossless. > > Thanks. I realized that I made a typo above; I'm now converting to yuva420p, and then to yuv444p. Any ideas on why the image splits? From onemda at gmail.com Tue Mar 27 04:38:31 2012 From: onemda at gmail.com (Paul B Mahol) Date: Tue, 27 Mar 2012 02:38:31 +0000 Subject: [FFmpeg-user] bgra to yuv In-Reply-To: References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: On 3/27/12, Josh long wrote: >> >> libschroedinger encoder in FFmpeg supports only yuv420p, yuv422p and >> yuv444p. >> >> If your input is not one of this above you can not expect encoding to be >> lossless. >> >> > Thanks. I realized that I made a typo above; I'm now converting to > yuva420p, and then to yuv444p. Any ideas on why the image splits? yuva420->yuv444 is not lossless either. From funkyirish at gmail.com Tue Mar 27 04:43:58 2012 From: funkyirish at gmail.com (Josh long) Date: Mon, 26 Mar 2012 21:43:58 -0500 Subject: [FFmpeg-user] bgra to yuv In-Reply-To: References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: On Mon, Mar 26, 2012 at 9:38 PM, Paul B Mahol wrote: > On 3/27/12, Josh long wrote: > >> > >> libschroedinger encoder in FFmpeg supports only yuv420p, yuv422p and > >> yuv444p. > >> > >> If your input is not one of this above you can not expect encoding to be > >> lossless. > >> > >> > > Thanks. I realized that I made a typo above; I'm now converting to > > yuva420p, and then to yuv444p. Any ideas on why the image splits? > > yuva420->yuv444 is not lossless either. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > aha, thanks I suppose I've not done my homework. Do you know of any routes that would be lossless? I have strong suspicions that the alpha channel in my bgra files is not significant (meaning that it is a constant ff when looking at it in a hex editor). From onemda at gmail.com Tue Mar 27 04:49:16 2012 From: onemda at gmail.com (Paul B Mahol) Date: Tue, 27 Mar 2012 02:49:16 +0000 Subject: [FFmpeg-user] bgra to yuv In-Reply-To: References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: On 3/27/12, Josh long wrote: > On Mon, Mar 26, 2012 at 9:38 PM, Paul B Mahol wrote: > >> On 3/27/12, Josh long wrote: >> >> >> >> libschroedinger encoder in FFmpeg supports only yuv420p, yuv422p and >> >> yuv444p. >> >> >> >> If your input is not one of this above you can not expect encoding to >> >> be >> >> lossless. >> >> >> >> >> > Thanks. I realized that I made a typo above; I'm now converting to >> > yuva420p, and then to yuv444p. Any ideas on why the image splits? >> >> yuva420->yuv444 is not lossless either. >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > aha, thanks I suppose I've not done my homework. Do you know of any routes > that would be lossless? I have strong suspicions that the alpha channel in > my bgra files is not significant (meaning that it is a constant ff when > looking at it in a hex editor). Note that it may be vlc bug if it fails to properly decode yuv444. From ban.an at list.ru Sun Mar 25 20:33:03 2012 From: ban.an at list.ru (iiser) Date: Sun, 25 Mar 2012 11:33:03 -0700 (PDT) Subject: [FFmpeg-user] Unable to demux 5.1 DVD-Audio rips In-Reply-To: References: <1332418283977-4495306.post@n4.nabble.com> Message-ID: <1332700383267-4503722.post@n4.nabble.com> I've uploaded 10MB samples from both DVD-A rips: DVDA-44100-24-51--MLP_ATS_01_1.AOB DVDA-44100-24-51--MLP_ATS_01_1.txt DVDA-48000-24-51--MLP_ATS_01_1.AOB DVDA-48000-24-51--MLP_ATS_01_1.txt -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Unable-to-demux-5-1-DVD-Audio-rips-tp4495306p4503722.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From ndummy001 at yahoo.co.jp Mon Mar 26 13:38:56 2012 From: ndummy001 at yahoo.co.jp (bluestring) Date: Mon, 26 Mar 2012 04:38:56 -0700 (PDT) Subject: [FFmpeg-user] How ot avoid "invalid droping" assertion message ? Message-ID: <1332761936574-4505484.post@n4.nabble.com> I've implementing own DirectShow source filter. This filter correctly push samples to FFmpeg and FFmpeg renders images. But FFmpeg outputs "ts:0 invalid droping" assertion messages. How to avoid this assertion ? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-ot-avoid-invalid-droping-assertion-message-tp4505484p4505484.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From esq.mail at gmail.com Mon Mar 26 04:18:02 2012 From: esq.mail at gmail.com (Dan Esquibel) Date: Sun, 25 Mar 2012 20:18:02 -0600 Subject: [FFmpeg-user] Help Message-ID: Hello I hope someone can help me. I get the following comment in the terminal:*"ERROR: libmp3lame >= 3.98.3 not found"* Then:* "If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solving the problem."* I have no idea how to get the *"config.log"* and it doesn't look like any one goes to the #ffmpeg freenode IRC chat. Can anyone help? I'm copying and pasting from *" ubuntuforums.org/showpost.php?p=11483157&postcount=1945" * * "HOW TO: Install and use the latest FFmpeg and x264 "* I'm on no. 7: *cd git clone --depth 1 git://source.ffmpeg.org/ffmpeg cd ffmpeg ./configure --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb \ --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx \ --enable-libx264 --enable-nonfree --enable-version3 --enable-x11grab make * *sudo checkinstall --pkgname=ffmpeg --pkgversion="5:$(./version.sh)" --backup=no \ --deldoc=yes --default hash x264 ffmpeg ffplay ffprobe* The error comes when I copy and past enter: *"./configure --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb \ --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx \ --enable-libx264 --enable-nonfree --enable-version3 ?enable-x11grab"* This is my third reinstall of Ubuntu because I thought I messed it up. From ibantxo28 at gmail.com Tue Mar 27 08:05:15 2012 From: ibantxo28 at gmail.com (Iban Garcia) Date: Tue, 27 Mar 2012 08:05:15 +0200 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: <4F7085E4.8080609@yahoo.com> References: <4F701E30.6000500@yahoo.com> <4F7075F5.5090403@yahoo.com> <4F7085E4.8080609@yahoo.com> Message-ID: 2012/3/26 Tim Nicholson > On 26/03/12 15:29, Iban Garcia wrote: > > Exact!! > > > > But the issue *has been solved* using next: > > -vf "scale=720:576:interl=1,fieldorder=tff" -vcodec dvvideo > > > > Ahh, the difference between scale=720:576:1 and scale=720:576:interl=1 !! > > I think the documentation is unclear on the fact that for width and height > you > just specify an expression, but for interlace you need to specify the > parameter > as well. > > *But* your command, as above is setting the output field order to tff when > DV is > bff so you will get motion judder. I really think you should use:- > > ...vf "setfield=1, scale=720:576:interl=1, fieldorder=bff"... > > or:- > > -vf "setfield=0, scale=720:576:interl=1"... > if your source is already bottom field first (unlikely). > > Umm... it is strange, because if I use -vf "setfield=1, scale=720:576:interl=1, fieldorder=bff" I get incorrect field order when playing (motion judder) The only way I have got the video correctly played is using "tff" instead of "bff". > > > I think ffmpeg should "detect" the interlaced without "interl=1" option. > > > > That option has nothing to do with detection, its how the scaling filter > should > work. > > ffmpeg *can* detect interlace for which you set interl=-1 to let the filter > decide automatically how to behave, however by default I believe it is set > to 0, > i.e off. > > *However* as I mentioned previously. MXF material seems to be always > detected as > progressive even when it is not. It is not just ffmpeg that makes this > mistake. > The way you detect interlace in MXF seems to be different to most other > formats. > Hence the forcing of the interlace flag by setfield, followed by telling > the > scaler to use interlaced scaling. > > Understood. Thanks Tim. > > Many thanks, Tim!!! > > > > > > 2012/3/26 Tim Nicholson > > > >> On 26/03/12 11:34, Iban Garcia wrote: > >>> Umm... it does not work for me. The output.dv is still "progressive" > >>> (de-interlaced): both fields of output.dv file are equal. The > interlacing > >>> has been lost... > >>> > >> > >> OK so something is not working as it should. I have had a look at some > of > >> my > >> H264 10bit interlaced material and that is also behaving as you > describe. > >> The > >> source is definitely interlaced, as can be seen from the interlace > >> artifacts in > >> a resized ffplay window. But the output is a blend of the two fields. > >> > >> I will try and get some more details and then raise a bug report. > >> > >>> I have download and installed "ffmbc". I am doing some test now with > it. > >>> But, by the moment, all of the tests has the same "de-interlaced" > >> output... > >>> :-((( > >>> > >>> > >>> 2012/3/26 Tim Nicholson > >>> > >>>> On 25/03/12 15:43, Iban Garcia wrote: > >>>>> Hi, > >>>>> > >>>>> I have taken from git the last version of ffmpeg. And I have made > next > >>>>> test. The output.dv fields are not interlaced. The interlaced > >> information > >>>>> has been lost in the process. > >>>>> > >>>>> # ffmpeg -y -threads 8 -i input.mxf -s 720x576 -pix_fmt yuv411p > -aspect > >>>>> 16:9 -an -r 25 -vf "setfield=1, fieldorder=bff" output.dv > >>>> > >>>>> [..] > >>>> > >>>>> sws_param: > >>>>> [fieldorder @ 0x646e80] output field order: bff > >>>>> [scale @ 0x6283a0] w:1920 h:1080 fmt:yuv422p10le sar:1/1 -> w:720 > h:576 > >>>>> fmt:yuv411p sar:64/45 flags:0x4 > >>>>> > >>>> > >>>> As your source is HD and you wish to output SD, you should really > scale > >> the > >>>> material properly using the "scale" filter with interlacing turned on. > >>>> > >>>> Something like:- > >>>> > >>>> ffmpeg -y -threads 8 -i input.mxf \ > >>>> -vf "setfield=1, scale=720:576:1, fieldorder=bff" \ > >>>> -pix_fmt yuv411p -aspect 16:9 -an -r 25 output.dv > >>>> > >>>> -- > >>>> Tim > >>>> _______________________________________________ > >> > >> -- > >> Tim > >> _______________________________________________ > >> ffmpeg-user mailing list > >> ffmpeg-user at ffmpeg.org > >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > >> > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > -- > Tim > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From softteam at hotmail.com Tue Mar 27 09:49:01 2012 From: softteam at hotmail.com (Peter Dickten) Date: Tue, 27 Mar 2012 07:49:01 +0000 Subject: [FFmpeg-user] set total number of output frames when extracting pictures from movie Message-ID: Hi, I'm extracting pictures of movies using ffmpeg. In the past I was abled to specify the number of frames to export (total number, not fps!)This was very handy, because I sometimes needed more pictures than the movie contained (because of gaps in the recording)and ffmpeg filled the gaps with duplicate pictures. Now I can't find my old shell command and I'm either too stupid or too blind [or both] to find the command line parameter :-(Can you give me hint! Thanks a lot!J.F.Sebastian From nichot20 at yahoo.com Tue Mar 27 10:31:22 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Tue, 27 Mar 2012 09:31:22 +0100 Subject: [FFmpeg-user] Help In-Reply-To: References: Message-ID: <4F717ADA.9010106@yahoo.com> On 26/03/12 03:18, Dan Esquibel wrote: > Hello I hope someone can help me. I get the following comment in the > terminal:*"ERROR: libmp3lame >= 3.98.3 not found"* > Then:* "If you think configure made a mistake, make sure you are using the > latest version from Git. If the latest version fails, report the problem to > the ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. > Include the log file "config.log" produced by configure as this will help > solving the problem."* > > I have no idea how to get the *"config.log"* and it doesn't look like any > one goes to the #ffmpeg freenode IRC chat. #ffmpeg is usually pretty active, but maybe not in your timezone.. The file in question is in the root of your ffmpeg source tree. > > Can anyone help? > > I'm copying and pasting from *" > ubuntuforums.org/showpost.php?p=11483157&postcount=1945" > * > * "HOW TO: Install and use the latest FFmpeg and x264 "* > > I'm on no. 7: > > *cd > > git clone --depth 1 git://source.ffmpeg.org/ffmpeg > > cd ffmpeg > > ./configure --enable-gpl --enable-libfaac --enable-libmp3lame > --enable-libopencore-amrnb \ So you are asking it to include the external lame library and the error you list above is telling you that it cannot find a version of that library no earlier than version 3.98.3 You need to install a version of lame that matches this criteria, making sure you include the "-devel" packages for the compiler to link against. > > --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis > --enable-libvpx \ > > --enable-libx264 --enable-nonfree --enable-version3 --enable-x11grab > -- Tim From tevans.uk at googlemail.com Tue Mar 27 10:50:33 2012 From: tevans.uk at googlemail.com (Tom Evans) Date: Tue, 27 Mar 2012 09:50:33 +0100 Subject: [FFmpeg-user] overlay w/librtmp In-Reply-To: References: Message-ID: On Mon, Mar 26, 2012 at 6:19 PM, Scott Freeman wrote: > Tom, thanks for the clarification. Like I mentioned, I thought it would be > a stretch to call it a bug. And you confirmed my guess on how > "copy" worked, thank you. > > I am curious though about what you said and this may be a topic for another > thread. Why couldn't ffmpeg be able to detect the input video information > and create a "similar" output string to essentially copy the inputs > settings? I could surely do that or something similar with a script and > some program logic to generate a output string before starting the encode. > I think it would make for a good feature built-in, especially when users > sometimes are just re-encoding a video to do effects and overlays and such. > Am I way off here? > > Thanks again Tom, for taking the time to reply to my post. > > Cheers, > Scott Please don't top post, people will get angry. Certain parameters - the dimensions of the video, the colourspace, the codec used, the average bitrate - can be deduced, and ffmpeg can detect those. What about things like whether the video was 2 or 3 pass encoded, how will it deduce that? How about codec/encoder specific things - if x264, what preset to use, ultrafast or ultraslow? Did the original have an x264 tuning (film, cartoon, etc). If you are planning to overlay a logo on videos, it would be better to do so before you do the lossy encoding. I don't know if that is appropriate for you, maybe you start with quite lossy videos already? Cheers Tom From nichot20 at yahoo.com Tue Mar 27 10:56:22 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Tue, 27 Mar 2012 09:56:22 +0100 Subject: [FFmpeg-user] DOWN CONVERSION ISSUE: From MXF (1080i AVC Intra) - MBAFF "interlacing" to DVCPRO In-Reply-To: References: <4F701E30.6000500@yahoo.com> <4F7075F5.5090403@yahoo.com> <4F7085E4.8080609@yahoo.com> Message-ID: <4F7180B6.2030205@yahoo.com> On 27/03/12 07:05, Iban Garcia wrote: > 2012/3/26 Tim Nicholson > >> On 26/03/12 15:29, Iban Garcia wrote: >>> Exact!! >>> >>> But the issue *has been solved* using next: >>> -vf "scale=720:576:interl=1,fieldorder=tff" -vcodec dvvideo >>> >> >> Ahh, the difference between scale=720:576:1 and scale=720:576:interl=1 !! >> >> I think the documentation is unclear on the fact that for width and height >> you >> just specify an expression, but for interlace you need to specify the >> parameter >> as well. >> >> *But* your command, as above is setting the output field order to tff when >> DV is >> bff so you will get motion judder. I really think you should use:- >> >> ...vf "setfield=1, scale=720:576:interl=1, fieldorder=bff"... >> >> or:- >> >> -vf "setfield=0, scale=720:576:interl=1"... >> if your source is already bottom field first (unlikely). >> >> Umm... it is strange, because if I use > -vf "setfield=1, scale=720:576:interl=1, fieldorder=bff" > I get incorrect field order when playing (motion judder) > The only way I have got the video correctly played is using "tff" instead > of "bff". This would suggest your original material is already bff but flagged as tff, that way ffmpeg thinks nothing needs to be done when you specify tff as it is that already, wheras my command forces a flip. It shows how important it is to test the parameters thoroughly before use...:) -- Tim From de.techno at gmail.com Tue Mar 27 05:38:25 2012 From: de.techno at gmail.com (dE .) Date: Tue, 27 Mar 2012 09:08:25 +0530 Subject: [FFmpeg-user] -t option is ignored In-Reply-To: <4F707D9F.6000905@yahoo.com> References: <201203270028.38628.rodney.baker@iinet.net.au> <4F707D9F.6000905@yahoo.com> Message-ID: <4F713631.5050502@gmail.com> On 03/26/12 20:00, Tim Nicholson wrote: > On 26/03/12 14:58, Rodney Baker wrote: >> On Mon, 26 Mar 2012 21:31:44 Martin G wrote: >>> FFMPEG users, >>> >>> I am trying to edit a subclip from a larger movie. I just want a two >>> minute section. >>> >>> This is the command I have been working with: >>> >>> ffmpeg -i input.avi -ss 01:05:16 -t 120 -vcodec copy -acodec copy >>> output.avi >>> >> In my experience it is faster to put -ss and -t options before -i, otherwise >> ffmpeg parses the whole file up to -ss before starting conversion. With -ss >> first, it begins reading at that point (or the closest i-frame or reference >> frame). >> > faster, yes, but sometimes less accurate..... > >> Unless, of course, the behaviour has changed in recent versions but I have >> developed the habit of always putting the seek and duration options first. >> >> [...] >> > Ok, now I see why did that happen. Thanks for the tip. From de.techno at gmail.com Tue Mar 27 05:53:51 2012 From: de.techno at gmail.com (dE .) Date: Tue, 27 Mar 2012 09:23:51 +0530 Subject: [FFmpeg-user] Compressibility Check In-Reply-To: References: Message-ID: <4F7139CF.1080209@gmail.com> On 03/27/12 04:52, Brendan Brewster wrote: > Hi, > > I am trying to implement a compressibility check via ffmpeg to determine an > appropriate resolution for the full encode. Can you please be more clear about what you're trying to do? What do you mean by 'compressibility'. Changing resolution is just one of changing compression level. From mike.scheutzow at alcatel-lucent.com Tue Mar 27 14:42:45 2012 From: mike.scheutzow at alcatel-lucent.com (Mike Scheutzow) Date: Tue, 27 Mar 2012 08:42:45 -0400 Subject: [FFmpeg-user] set total number of output frames when extracting pictures from movie In-Reply-To: References: Message-ID: <4F71B5C5.4040602@alcatel-lucent.com> Peter Dickten wrote: > I'm extracting pictures of movies using ffmpeg. > In the past I was abled to specify the number of frames to export (total number, not fps!)This was very handy, because I sometimes needed more pictures than the movie contained (because of gaps in the recording)and ffmpeg filled the gaps with duplicate pictures. > Now I can't find my old shell command and I'm either too stupid or too blind [or both] to find the command line parameter :-(Can you give me hint! > Thanks a lot!J.F.Sebastian -vframes N In recent versions of ffmpeg, I had to move it after -vcodec to make it to work. Mike Scheutzow From brendan.brewster at gmail.com Tue Mar 27 18:04:44 2012 From: brendan.brewster at gmail.com (Brendan Brewster) Date: Tue, 27 Mar 2012 12:04:44 -0400 Subject: [FFmpeg-user] Compressibility Check In-Reply-To: <4F7139CF.1080209@gmail.com> References: <4F7139CF.1080209@gmail.com> Message-ID: On Mon, Mar 26, 2012 at 11:53 PM, dE . wrote: > On 03/27/12 04:52, Brendan Brewster wrote: > >> Hi, >> >> I am trying to implement a compressibility check via ffmpeg to determine >> an >> appropriate resolution for the full encode. >> > > Can you please be more clear about what you're trying to do? What do you > mean by 'compressibility'. Changing resolution is just one of changing > compression level. > The idea is to take some percentage of the input, say 5%, at various points throughout and encode this with the default (input) resolution and bitrate. I have a specific file size to meet, such as DVD5, and therefore know the bitrate cap in order to fit my output onto the destination media (given a certain audio bitrate). I can then determine the number of bits per pixel for the default encode and prepare ratios with regard to lesser resolutions with the enforced bitrate. We choose a lower resolution if the ratio is poor. This is what I mean by a compressibility check. Similar functionality was implemented by the legacy Gordian Knot as well as within Tuxrip. I would just like to speed up my analysis by avoiding the majority of the seek time incurred especially in passes that pinpoint points later in the input video stream. I was really hoping that there might be a way to achieve the same end via one pass. Thanks, Brendan From anders.branderud at gmail.com Tue Mar 27 21:41:22 2012 From: anders.branderud at gmail.com (Anders Branderud) Date: Tue, 27 Mar 2012 21:41:22 +0200 Subject: [FFmpeg-user] Code which encodes file using MPEG4-codec, decodes the encoded data and then plays the file? In-Reply-To: References: Message-ID: Hello! I wondered if anyone has created a code which does the following: Encodes file using MPEG4-codec, decodes the encoded data; and play each 25 frames continiously as 25 new frames are decoded (If so, I would be very thankful to get it mailed to me!)? I am aware of the decoding_encoding.c-example (in /doc/examples of the FFMPEG-library) and wonder which changes that are needed to be made in order to accomplish this? Thanks in advance! -- *Kind regards, Anders * [Personal blog] Will of the Creator : Logical reasons - based on scientific premises - for the existence of a Super intelligent and Orderly Creator and that He hasn't left His sapient creatures without an Instruction Manual - Torah ['books of Moses'] - to ascertain, and aspire to, His purpose. [Company] Anders Branderud IT Solutions - www.abitsolutions.org -- *Kind regards,* [Personal blog] Will of the Creator : Logical reasons - based on scientific premises - for the existence of a Super intelligent and Orderly Creator and that He hasn't left His sapient creatures without an Instruction Manual - Torah ['books of Moses'] - to ascertain, and aspire to, His purpose. [Company] Anders Branderud IT Solutions - www.abitsolutions.org From de.techno at gmail.com Wed Mar 28 01:28:09 2012 From: de.techno at gmail.com (dE .) Date: Wed, 28 Mar 2012 04:58:09 +0530 Subject: [FFmpeg-user] libavcodec/dv_tablegen.h:47:34: fatal error: libavcodec/dv_tables.h: No such file or directory Message-ID: <4F724D09.1080907@gmail.com> Compiling from GIT fails with this. https://409945.bugs.gentoo.org/attachment.cgi?id=306925 From cehoyos at ag.or.at Wed Mar 28 08:10:10 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 28 Mar 2012 06:10:10 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?libavcodec/dv=5Ftablegen=2Eh=3A47=3A34=3A?= =?utf-8?q?_fatal_error=3A_libavcodec/dv=5Ftables=2Eh=3A_No_such_fi?= =?utf-8?q?le_or_directory?= References: <4F724D09.1080907@gmail.com> Message-ID: dE . gmail.com> writes: > Compiling from GIT fails with this. Please understand that this report is nearly useless: A build problem report should always contain the used configure line (especially if the problem is not reproducible with ./configure && make) and all information should be posted here on the list, external resources may disappear (especially if only two lines are needed to describe the problem). Should be fixed, Carl Eugen From salmjuh at hotmail.com Wed Mar 28 09:38:05 2012 From: salmjuh at hotmail.com (juha s.) Date: Wed, 28 Mar 2012 07:38:05 +0000 Subject: [FFmpeg-user] =?windows-1256?q?moov_atom_not_found=FE?= Message-ID: Hi I'm capturing live h264 stream over RTSP protocol and time to time when I stop recordind I get corrupted file. Here is ffmpeg line what I use for recording: ffmpeg -i rtsp://192.169.23.28:8555 -aspect 16:9 -copyts -vcodec libx264 -preset fast -b:v 900k -crf 22 -y /videos/video1.mp4 . I must stop this record using "pkill 2" (=ctrl+c) command, because I like to split this program for peaces. Moust of files are OK and I can play those without any problem. Here is one of those working files output: root at CVod1:/DVB# ffmpeg -i /videos/Video_2.mp4 ffmpeg version N-34781-gab31db0 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 27 2012 08:48:53 with gcc 4.6.1 configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab --enable-libvpx libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 66.100 / 2. 66.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/videos/Video_2.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf54.3.100 Duration: 00:00:30.26, start: 0.000000, bitrate: 642 kb/s Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 508 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc Metadata: handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 127 kb/s Metadata: handler_name : At least one output file must be specified root at CVod1:/DVB# And sometimes from same source and same stream, the recording fails for " moov atom not found": root at Vod1:/DVB# ffmpeg -i /videos/Video_53.mp4 ffmpeg version N-34781-gab31db0 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 27 2012 08:48:53 with gcc 4.6.1 configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab --enable-libvpx libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 66.100 / 2. 66.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x16d43e0] moov atom not found /videos/Video_53.mp4: Invalid data found when processing input root at Vod1:/DVB# So what's the trik here or how I should do this to get all files to work without any problem ? Cheers, Jii From de.techno at gmail.com Wed Mar 28 04:14:07 2012 From: de.techno at gmail.com (dE .) Date: Wed, 28 Mar 2012 07:44:07 +0530 Subject: [FFmpeg-user] Compressibility Check In-Reply-To: References: <4F7139CF.1080209@gmail.com> Message-ID: <4F7273EF.5020503@gmail.com> On 03/27/12 21:34, Brendan Brewster wrote: > On Mon, Mar 26, 2012 at 11:53 PM, dE . wrote: > >> On 03/27/12 04:52, Brendan Brewster wrote: >> >>> Hi, >>> >>> I am trying to implement a compressibility check via ffmpeg to determine >>> an >>> appropriate resolution for the full encode. >>> >> Can you please be more clear about what you're trying to do? What do you >> mean by 'compressibility'. Changing resolution is just one of changing >> compression level. >> > The idea is to take some percentage of the input, say 5%, at various points > throughout and encode this with the default (input) resolution and bitrate. > I have a specific file size to meet, such as DVD5, and therefore know the > bitrate cap in order to fit my output onto the destination media (given a > certain audio bitrate). I can then determine the number of bits per pixel > for the default encode and prepare ratios with regard to lesser resolutions > with the enforced bitrate. We choose a lower resolution if the ratio is > poor. This is what I mean by a compressibility check. Similar functionality > was implemented by the legacy Gordian Knot as well as within Tuxrip. > > I would just like to speed up my analysis by avoiding the majority of the > seek time incurred especially in passes that pinpoint points later in the > input video stream. I was really hoping that there might be a way to > achieve the same end via one pass. > > Thanks, > Brendan > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From what I understand, you're asking for CBR, but that can be done using a combination of -maxrate and -minrate. That will reduce on the quality not resolution. This way you can keep the resolution constant. From de.techno at gmail.com Wed Mar 28 04:16:41 2012 From: de.techno at gmail.com (dE .) Date: Wed, 28 Mar 2012 07:46:41 +0530 Subject: [FFmpeg-user] Code which encodes file using MPEG4-codec, decodes the encoded data and then plays the file? In-Reply-To: References: Message-ID: <4F727489.7010704@gmail.com> On 03/28/12 01:11, Anders Branderud wrote: > Hello! > I wondered if anyone has created a code which does the following: > Encodes file using MPEG4-codec, decodes the encoded data; and play each 25 > frames continiously as 25 new frames are decoded (If so, I would be very > thankful to get it mailed to me!)? > > I am aware of the decoding_encoding.c-example (in /doc/examples of the > FFMPEG-library) and wonder which changes that are needed to be made in > order to accomplish this? > > Thanks in advance! > Are you going to incorporate this in a code? If not, you may use pipe, but if you want blocks of 25 frames, I'm not sure... From cehoyos at ag.or.at Wed Mar 28 09:47:03 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 28 Mar 2012 07:47:03 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?moov_atom_not_found=E2=80=8F?= References: Message-ID: juha s. hotmail.com> writes: > Here is ffmpeg line what I use for recording: ffmpeg -i > rtsp://192.169.23.28:8555 -aspect 16:9 -copyts > -vcodec libx264 -preset fast -b:v 900k -crf 22 -y > /videos/video1.mp4 . I must stop this record using > "pkill 2" (=ctrl+c) command, because I like to > split this program for peaces. Use "q" to stop recording, this ensures that the moov atom is written (without it, the file can not be played). Carl Eugen From de.techno at gmail.com Wed Mar 28 04:24:56 2012 From: de.techno at gmail.com (dE .) Date: Wed, 28 Mar 2012 07:54:56 +0530 Subject: [FFmpeg-user] libavcodec/dv_tablegen.h:47:34: fatal error: libavcodec/dv_tables.h: No such file or directory In-Reply-To: References: <4F724D09.1080907@gmail.com> Message-ID: <4F727678.7090207@gmail.com> On 03/28/12 11:40, Carl Eugen Hoyos wrote: > dE . gmail.com> writes: > >> Compiling from GIT fails with this. > Please understand that this report is nearly useless: > A build problem report should always contain the used configure > line (especially if the problem is not reproducible > with ./configure&& make) and all information should be > posted here on the list, external resources may disappear > (especially if only two lines are needed to describe the problem). > > Should be fixed, Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user ./configure --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-ar --optflags='-march=native -O2 -fomit-frame-pointer -floop-interchange -floop-strip-mine -floop-block -fgraphite-identity' --extra-cflags='-march=native -O2 -fomit-frame-pointer -floop-interchange -floop-strip-mine -floop-block -fgraphite-identity' --extra-cxxflags='-march=native -O2 -fomit-frame-pointer -floop-interchange -floop-strip-mine -floop-block -fgraphite-identity' --disable-static --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --disable-stripping --disable-debug --disable-doc --disable-network --disable-vaapi --disable-vdpau --enable-libmp3lame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libaacplus --enable-nonfree --enable-openal --disable-indev=v4l --disable-indev=v4l2 --disable-indev=oss --disable-indev=jack --enable-x11grab --disable-outdev=oss --enable-libfreetype --enable-pthreads --enable-libopencore-amrwb --enable-libopencore-amrnb --enable-libgsm --enable-libdirac --enable-libschroedinger --enable-libspeex --enable-libvpx --enable-libopenjpeg --disable-altivec --disable-avx --disable-ssse3 --disable-vis --disable-neon --cpu=host --enable-hardcoded-tables From ebisumartin at gmail.com Wed Mar 28 10:31:16 2012 From: ebisumartin at gmail.com (Martin G) Date: Wed, 28 Mar 2012 17:31:16 +0900 Subject: [FFmpeg-user] -t option is ignored In-Reply-To: <4F713631.5050502@gmail.com> References: <201203270028.38628.rodney.baker@iinet.net.au> <4F707D9F.6000905@yahoo.com> <4F713631.5050502@gmail.com> Message-ID: I tried moving the time specs to the beginning of the command like so: ffmpeg -ss 01:05:16 -t 120 -i input.avi -vcodec copy -acodec copy output.avi ... however this hasn't changed anything. Instead of 120 seconds worth, I get over 45 minutes. If -vcodec copy and -acodec copy "don't play well" with -t, then what should I use instead to preserve the same quality of video in the sub clip? Here is the output: f$ ffmpeg -ss 01:05:16 -t 120 -i input.avi -vcodec copy -acodec copy output.avi ffmpeg version 0.7.3-4:0.7.3-0ubuntu0.11.10.1, Copyright (c) 2000-2011 the Libav developers built on Jan 4 2012 16:08:51 with gcc 4.6.1 configuration: --extra-version='4:0.7.3-0ubuntu0.11.10.1' --arch=amd64 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static WARNING: library configuration mismatch avutil configuration: --extra-version='4:0.7.3ubuntu0.11.10.1' --arch=amd64 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-libopenjpeg --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdirac --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid --enable-libvo-aacenc --enable-version3 --enable-libvo-amrwbenc --enable-version3 --enable-libdc1394 --enable-shared --disable-static avcodec configuration: --extra-version='4:0.7.3ubuntu0.11.10.1' --arch=amd64 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-libopenjpeg --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdirac --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid --enable-libvo-aacenc --enable-version3 --enable-libvo-amrwbenc --enable-version3 --enable-libdc1394 --enable-shared --disable-static libavutil 51. 7. 0 / 51. 7. 0 libavcodec 53. 6. 0 / 53. 6. 0 libavformat 53. 3. 0 / 53. 3. 0 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 4. 0 / 2. 4. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 52. 0. 0 / 52. 0. 0 [mpeg4 @ 0x1f67540] Invalid and inefficient vfw-avi packed B frames detected Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 23.98 (24000/1001) Input #0, avi, from 'input.avi': Metadata: title : Triumph Des Willens comment : Ripped by Reevel encoder : VirtualDubMod 1.4.13 Duration: 01:50:30.50, start: 0.000000, bitrate: 886 kb/s Stream #0.0: Video: mpeg4, yuv420p, 512x400 [PAR 1:1 DAR 32:25], 23.98 fps, 23.98 tbr, 23.98 tbn, 30k tbc Stream #0.1: Audio: mp3, 44100 Hz, mono, s16, 93 kb/s Output #0, avi, to 'output.avi': Metadata: INAM : Triumph Des Willens ILNG : Undefined ICMT : Ripped by Reevel ISFT : Lavf53.3.0 Stream #0.0: Video: mpeg4, yuv420p, 512x400 [PAR 1:1 DAR 32:25], q=2-31, 23.98 tbn, 23.98 tbc Stream #0.1: Audio: libmp3lame, 44100 Hz, mono, 93 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press ctrl-c to stop encoding [NULL @ 0x1f67540] Invalid and inefficient vfw-avi packed B frames detected frame= 5057 fps=4915 q=-1.0 size= 13866kB time=120.01 bitrate= 946.5kbits/s frame= 8317 fps=5456 q=-1.0 size= 13866kB time=120.01 bitrate= 946.5kbits/s frame=15897 fps=7796 q=-1.0 size= 13866kB time=120.01 bitrate= 946.5kbits/s frame=19543 fps=7651 q=-1.0 size= 13866kB time=120.01 bitrate= 946.5kbits/s frame=22683 fps=7426 q=-1.0 size= 13866kB time=120.01 bitrate= 946.5kbits/s frame=29647 fps=8338 q=-1.0 size= 13866kB time=120.01 bitrate= 946.5kbits/s frame=33261 fps=8199 q=-1.0 size= 13866kB time=120.01 bitrate= 946.5kbits/s frame=37141 fps=8150 q=-1.0 size= 13866kB time=120.01 bitrate= 946.5kbits/s frame=40029 fps=7915 q=-1.0 size= 13866kB time=120.01 bitrate= 946.5kbits/s frame=44628 fps=8016 q=-1.0 size= 13866kB time=120.01 bitrate= 946.5kbits/s frame=47354 fps=7766 q=-1.0 size= 13866kB time=120.01 bitrate= 946.5kbits/s frame=51237 fps=7765 q=-1.0 size= 13866kB time=120.01 bitrate= 946.5kbits/s frame=56325 fps=7934 q=-1.0 size= 13866kB time=120.01 bitrate= 946.5kbits/s frame=61463 fps=8087 q=-1.0 size= 13866kB time=120.01 bitrate= 946.5kbits/s frame=65294 fps=4227 q=-1.0 Lsize= 301685kB time=120.01 bitrate=20593.9kbits/s video:298534kB audio:1474kB global headers:0kB muxing overhead 0.558787% From stozher at gmail.com Wed Mar 28 10:31:29 2012 From: stozher at gmail.com (John Saturday) Date: Wed, 28 Mar 2012 11:31:29 +0300 Subject: [FFmpeg-user] sound balance on several files In-Reply-To: <4F6EBBE9.7020907@orange.fr> References: <4F6EBBE9.7020907@orange.fr> Message-ID: Read chapter 2 and end table of document: http://www.dolby.com/uploadedFiles/Assets/US/Doc/Professional/18_Metadata.Guide.pdf Normalize to multiply of 3 dB level (-15, -18, -21 dB ...) without clips. I use Audacity ( http://audacity.sourceforge.net/download/ ): Analyze / Contrast... to see clips at waveform check View / Show Clippings. This is an actual level of right channel... to balance channels Split Stereo Track (dropdown menu left at waveform), remove one channel and measure contrast (dialogue level); Undo, remove other channel and measure... Undo and amplify each channel separately and Make Stereo Track (dropdown menu left at waveform). DVD producers usually normalized at -27 dB but this is not a fixed standard. First you need to measure and choice dialogue level for your files: load files one by one, amplify to not clipping level and round down measured contrast to multiply of 3 dB... After this choice your dialogue level and calculate volume parameter for FFmpeg. From cehoyos at ag.or.at Wed Mar 28 10:39:09 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 28 Mar 2012 08:39:09 +0000 (UTC) Subject: [FFmpeg-user] -t option is ignored References: <201203270028.38628.rodney.baker@iinet.net.au> <4F707D9F.6000905@yahoo.com> <4F713631.5050502@gmail.com> Message-ID: Martin G gmail.com> writes: > ffmpeg version 0.7.3-4:0.7.3-0ubuntu0.11.10.1 This is an intentionally broken version of FFmpeg that contains several hundred regressions, some of them security relevant. We therefore cannot support this version. Please test current git head, see http://ffmpeg.org/download.html Carl Eugen From ebisumartin at gmail.com Wed Mar 28 11:00:52 2012 From: ebisumartin at gmail.com (Martin G) Date: Wed, 28 Mar 2012 18:00:52 +0900 Subject: [FFmpeg-user] -t option is ignored In-Reply-To: References: <201203270028.38628.rodney.baker@iinet.net.au> <4F707D9F.6000905@yahoo.com> <4F713631.5050502@gmail.com> Message-ID: > This is an intentionally broken version of FFmpeg What? The ffmpeg in the repositories is an intentionally broken version?? Is it just me, or is that insane? Why would that be put out to where regular Ubuntu users would download it? -- Martin From ubitux at gmail.com Wed Mar 28 11:23:25 2012 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Wed, 28 Mar 2012 11:23:25 +0200 Subject: [FFmpeg-user] -t option is ignored In-Reply-To: References: <201203270028.38628.rodney.baker@iinet.net.au> <4F707D9F.6000905@yahoo.com> <4F713631.5050502@gmail.com> Message-ID: <20120328092325.GI5169@leki> On Wed, Mar 28, 2012 at 06:00:52PM +0900, Martin G wrote: > > This is an intentionally broken version of FFmpeg > > What? The ffmpeg in the repositories is an intentionally broken version?? > Ubuntu isn't packaging FFmpeg, despite the name. It's a fork called Libav; You may check https://launchpad.net/~jon-severinsson/+archive/ffmpeg for FFmpeg. [...] -- Cl?ment B. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 490 bytes Desc: not available URL: From salmjuh at hotmail.com Wed Mar 28 11:33:41 2012 From: salmjuh at hotmail.com (juha s.) Date: Wed, 28 Mar 2012 09:33:41 +0000 Subject: [FFmpeg-user] =?windows-1256?q?FW=3A_moov_atom_not_found=FE?= In-Reply-To: References: Message-ID: Hi I changed that so that PKILL is now using signal -3 (3 = QUIT Quit 19 CONT Continued) But no help: root at CVod1:/DVB# ffmpeg -i /videos/19010521.mp4 ffmpeg version N-34813-g96d0494 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 28 2012 08:37:13 with gcc 4.6.1 configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab --enable-libvpx libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 66.100 / 2. 66.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x2f5f3e0] moov atom not found /videos/19010521.mp4: Invalid data found when processing input root at CVod1:/DVB# Any suggestions would be highly appreciated !! br, jiiii From: salmjuh at hotmail.com To: ffmpeg-user at ffmpeg.org Subject: moov atom not found? Date: Wed, 28 Mar 2012 07:38:05 +0000 Hi I'm capturing live h264 stream over RTSP protocol and time to time when I stop recordind I get corrupted file. Here is ffmpeg line what I use for recording: ffmpeg -i rtsp://192.169.23.28:8555 -aspect 16:9 -copyts -vcodec libx264 -preset fast -b:v 900k -crf 22 -y /videos/video1.mp4 . I must stop this record using "pkill 2" (=ctrl+c) command, because I like to split this program for peaces. Moust of files are OK and I can play those without any problem. Here is one of those working files output: root at CVod1:/DVB# ffmpeg -i /videos/Video_2.mp4 ffmpeg version N-34781-gab31db0 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 27 2012 08:48:53 with gcc 4.6.1 configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab --enable-libvpx libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 66.100 / 2. 66.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/videos/Video_2.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf54.3.100 Duration: 00:00:30.26, start: 0.000000, bitrate: 642 kb/s Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 508 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc Metadata: handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 127 kb/s Metadata: handler_name : At least one output file must be specified root at CVod1:/DVB# And sometimes from same source and same stream, the recording fails for " moov atom not found": root at Vod1:/DVB# ffmpeg -i /videos/Video_53.mp4 ffmpeg version N-34781-gab31db0 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 27 2012 08:48:53 with gcc 4.6.1 configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab --enable-libvpx libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 66.100 / 2. 66.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x16d43e0] moov atom not found /videos/Video_53.mp4: Invalid data found when processing input root at Vod1:/DVB# So what's the trik here or how I should do this to get all files to work without any problem ? Cheers, Jii From ibantxo28 at gmail.com Wed Mar 28 12:10:50 2012 From: ibantxo28 at gmail.com (Iban Garcia) Date: Wed, 28 Mar 2012 12:10:50 +0200 Subject: [FFmpeg-user] Segmentation fault when opening source file (MXF). Debugging? Message-ID: Hi all! I have launched this command and I am getting "Segmentation fault" message. I have tried to "debug" and no information appears. "Segmentation fault" appears inmediatly, but no more information about the problem. The input.mxf file is correct (played with other video-player and mediainfo is telling it is OK) Iban # ffmpeg -loglevel debug -i input.mxf -threads 8 -pix_fmt yuv422p -aspect 16:9 -vcodec dvvideo -an -y output.dv ffmpeg version N-39247-g6809818 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 28 2012 11:41:42 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] configuration: --shlibdir=/usr/lib64 --prefix=/usr/local --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads --enable-shared --enable-libvorbis --enable-gpl --enable-x11grab --enable-libx264 --enable-libmp3lame --enable-nonfree --enable-postproc --enable-debug=3 --disable-optimizations --disable-mmx libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 65.102 / 2. 65.102 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 [mxf @ 0x62d320] Format mxf probed with size=2048 and score=100 Segmentation fault Same for: # ffmpeg -loglevel debug -i input.mxf ffmpeg version N-39247-g6809818 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 28 2012 11:41:42 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] configuration: --shlibdir=/usr/lib64 --prefix=/usr/local --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads --enable-shared --enable-libvorbis --enable-gpl --enable-x11grab --enable-libx264 --enable-libmp3lame --enable-nonfree --enable-postproc --enable-debug=3 --disable-optimizations --disable-mmx libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 65.102 / 2. 65.102 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 [mxf @ 0x62d320] Format mxf probed with size=2048 and score=100 Segmentation fault From cehoyos at ag.or.at Wed Mar 28 12:34:21 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 28 Mar 2012 10:34:21 +0000 (UTC) Subject: [FFmpeg-user] Segmentation fault when opening source file (MXF). Debugging? References: Message-ID: Iban Garcia gmail.com> writes: > I have launched this command and I am getting > "Segmentation fault" message. Please provide the sample, segmentation faults are always important bugs. Carl Eugen From renaux.jacky at orange.fr Wed Mar 28 12:37:23 2012 From: renaux.jacky at orange.fr (jacky) Date: Wed, 28 Mar 2012 12:37:23 +0200 Subject: [FFmpeg-user] sound balance on several files In-Reply-To: References: <4F6EBBE9.7020907@orange.fr> Message-ID: <4F72E9E3.4000803@orange.fr> Le 28/03/2012 10:31, John Saturday a ?crit : > Read chapter 2 and end table of document: > http://www.dolby.com/uploadedFiles/Assets/US/Doc/Professional/18_Metadata.Guide.pdf > > Normalize to multiply of 3 dB level (-15, -18, -21 dB ...) without > clips. I use Audacity ( http://audacity.sourceforge.net/download/ ): > Analyze / Contrast... to see clips at waveform check View / Show > Clippings. > > This is an actual level of right channel... to balance channels Split > Stereo Track (dropdown menu left at waveform), remove one channel and > measure contrast (dialogue level); Undo, remove other channel and > measure... Undo and amplify each channel separately and Make Stereo > Track (dropdown menu left at waveform). > > DVD producers usually normalized at -27 dB but this is not a fixed > standard. First you need to measure and choice dialogue level for your > files: load files one by one, amplify to not clipping level and round > down measured contrast to multiply of 3 dB... After this choice your > dialogue level and calculate volume parameter for FFmpeg. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user many thanks John I start to better undestand, I did read the paper and gave me a direction to look at I found an interresting one http://www.iis.fraunhofer.de/bf/amm/download/whitepapers/Metadata-whitepaper-092011.pdf then is it obvious it depend on sound coder setting , il will go more into details it is quite difficult to figure out what level is dbFS for mp3, aac and ac3, I will avoid ac3 at it is not compatible with flv container I do only streams speachs then either mp3 or aac are still eligible to be selected I cannot use audacity as it does not read flv container (I did not succed) there are still confusion (for me on) depending on sound format used which metadata name I should specify would this metadata be forwarded by the red5 streaming server (rtmp mode) does customer located player will handle such then I will explore few tracks Now with you answer, I know what are the points to look at thanks you to gave me a great help jacky From ibantxo28 at gmail.com Wed Mar 28 12:41:35 2012 From: ibantxo28 at gmail.com (Iban Garcia) Date: Wed, 28 Mar 2012 12:41:35 +0200 Subject: [FFmpeg-user] Segmentation fault when opening source file (MXF). Debugging? In-Reply-To: References: Message-ID: Well... it is a bit strange... In a machine I get "segmentation fault" and in another different machine I don't get it. # ffmpeg -i input.MXF ffmpeg version N-39247-g6809818 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 28 2012 11:41:42 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] configuration: --shlibdir=/usr/lib64 --prefix=/usr/local --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads --enable-shared --enable-libvorbis --enable-gpl --enable-x11grab --enable-libx264 --enable-libmp3lame --enable-nonfree --enable-postproc --enable-debug=3 --disable-optimizations --disable-mmx libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 65.102 / 2. 65.102 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 Segmentation fault # ffmpeg -i imput.MXF ffmpeg version 0.9.0.git, Copyright (c) 2000-2012 the FFmpeg developers built on Jan 5 2012 12:22:34 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] configuration: --shlibdir=/usr/lib64 --prefix=/usr/local --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads --enable-shared --enable-libvorbis --enable-libfaac --enable-gpl --enable-x11grab --enable-libx264 --enable-libmp3lame --enable-libtheora --enable-libxvid --enable-nonfree --enable-postproc --enable-pthreads libavutil 51. 33.100 / 51. 33.100 libavcodec 53. 50.100 / 53. 50.100 libavformat 53. 29.100 / 53. 29.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 57.101 / 2. 57.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 5.100 / 0. 5.100 libpostproc 51. 2.100 / 51. 2.100 Input #0, mxf, from '/net/online/satabio/HD_FROGAK/AC100/remen2.MXF': Duration: 00:00:35.00, start: 0.000000, bitrate: 117557 kb/s Stream #0:0: Video: h264 (High 4:2:2 Intra), yuv422p10le, 1280x720 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 50 tbn, 100 tbc Stream #0:1: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:2: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:3: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:4: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s 2012/3/28 Carl Eugen Hoyos > Iban Garcia gmail.com> writes: > > > I have launched this command and I am getting > > "Segmentation fault" message. > > Please provide the sample, segmentation faults > are always important bugs. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ibantxo28 at gmail.com Wed Mar 28 12:43:27 2012 From: ibantxo28 at gmail.com (Iban Garcia) Date: Wed, 28 Mar 2012 12:43:27 +0200 Subject: [FFmpeg-user] Segmentation fault when opening source file (MXF). Debugging? In-Reply-To: References: Message-ID: Well... it is a bit strange... In a machine I get "segmentation fault" and in another different machine I don't get it. # ffmpeg -i input.MXF ffmpeg version N-39247-g6809818 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 28 2012 11:41:42 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] configuration: --shlibdir=/usr/lib64 --prefix=/usr/local --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads --enable-shared --enable-libvorbis --enable-gpl --enable-x11grab --enable-libx264 --enable-libmp3lame --enable-nonfree --enable-postproc --enable-debug=3 --disable-optimizations --disable-mmx libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 65.102 / 2. 65.102 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 Segmentation fault # ffmpeg -i imput.MXF ffmpeg version 0.9.0.git, Copyright (c) 2000-2012 the FFmpeg developers built on Jan 5 2012 12:22:34 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] configuration: --shlibdir=/usr/lib64 --prefix=/usr/local --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads --enable-shared --enable-libvorbis --enable-libfaac --enable-gpl --enable-x11grab --enable-libx264 --enable-libmp3lame --enable-libtheora --enable-libxvid --enable-nonfree --enable-postproc --enable-pthreads libavutil 51. 33.100 / 51. 33.100 libavcodec 53. 50.100 / 53. 50.100 libavformat 53. 29.100 / 53. 29.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 57.101 / 2. 57.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 5.100 / 0. 5.100 libpostproc 51. 2.100 / 51. 2.100 Input #0, mxf, from 'input.MXF': Duration: 00:00:35.00, start: 0.000000, bitrate: 117557 kb/s Stream #0:0: Video: h264 (High 4:2:2 Intra), yuv422p10le, 1280x720 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 50 tbn, 100 tbc Stream #0:1: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:2: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:3: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:4: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s 2012/3/28 Carl Eugen Hoyos > Iban Garcia gmail.com> writes: > > > I have launched this command and I am getting > > "Segmentation fault" message. > > Please provide the sample, segmentation faults > are always important bugs. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From deepesh.basu at gmail.com Wed Mar 28 12:44:40 2012 From: deepesh.basu at gmail.com (Deepesh Basu) Date: Wed, 28 Mar 2012 16:14:40 +0530 Subject: [FFmpeg-user] Segmentation fault when opening source file (MXF). Debugging? In-Reply-To: References: Message-ID: Could you please provide us with the Core-Dumps and DEBUG logs, from both the systems, as well as a Minimal H/W configuration (binutils, glibc wud suffice) differences, please? Thnx/BR, Deepesh +49-89-2351-4997 On Wed, Mar 28, 2012 at 4:11 PM, Iban Garcia wrote: > Well... it is a bit strange... In a machine I get "segmentation fault" and > in another different machine I don't get it. > > # ffmpeg -i input.MXF > ffmpeg version N-39247-g6809818 Copyright (c) 2000-2012 the FFmpeg > developers > built on Mar 28 2012 11:41:42 with gcc 4.5.1 20101208 [gcc-4_5-branch > revision 167585] > configuration: --shlibdir=/usr/lib64 --prefix=/usr/local > --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads > --enable-shared --enable-libvorbis --enable-gpl --enable-x11grab > --enable-libx264 --enable-libmp3lame --enable-nonfree --enable-postproc > --enable-debug=3 --disable-optimizations --disable-mmx > libavutil 51. 44.100 / 51. 44.100 > libavcodec 54. 12.100 / 54. 12.100 > libavformat 54. 3.100 / 54. 3.100 > libavdevice 53. 4.100 / 53. 4.100 > libavfilter 2. 65.102 / 2. 65.102 > libswscale 2. 1.100 / 2. 1.100 > libswresample 0. 10.100 / 0. 10.100 > libpostproc 52. 0.100 / 52. 0.100 > Segmentation fault > > > # ffmpeg -i imput.MXF > ffmpeg version 0.9.0.git, Copyright (c) 2000-2012 the FFmpeg developers > built on Jan 5 2012 12:22:34 with gcc 4.5.1 20101208 [gcc-4_5-branch > revision 167585] > configuration: --shlibdir=/usr/lib64 --prefix=/usr/local > --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads > --enable-shared --enable-libvorbis --enable-libfaac --enable-gpl > --enable-x11grab --enable-libx264 --enable-libmp3lame --enable-libtheora > --enable-libxvid --enable-nonfree --enable-postproc --enable-pthreads > libavutil 51. 33.100 / 51. 33.100 > libavcodec 53. 50.100 / 53. 50.100 > libavformat 53. 29.100 / 53. 29.100 > libavdevice 53. 4.100 / 53. 4.100 > libavfilter 2. 57.101 / 2. 57.101 > libswscale 2. 1.100 / 2. 1.100 > libswresample 0. 5.100 / 0. 5.100 > libpostproc 51. 2.100 / 51. 2.100 > Input #0, mxf, from '/net/online/satabio/HD_FROGAK/AC100/remen2.MXF': > Duration: 00:00:35.00, start: 0.000000, bitrate: 117557 kb/s > Stream #0:0: Video: h264 (High 4:2:2 Intra), yuv422p10le, 1280x720 [SAR > 1:1 DAR 16:9], 50 fps, 50 tbr, 50 tbn, 100 tbc > Stream #0:1: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s > Stream #0:2: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s > Stream #0:3: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s > Stream #0:4: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s > > > > 2012/3/28 Carl Eugen Hoyos > > > Iban Garcia gmail.com> writes: > > > > > I have launched this command and I am getting > > > "Segmentation fault" message. > > > > Please provide the sample, segmentation faults > > are always important bugs. > > > > Carl Eugen > > > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From deepesh.basu at gmail.com Wed Mar 28 12:47:17 2012 From: deepesh.basu at gmail.com (Deepesh Basu) Date: Wed, 28 Mar 2012 16:17:17 +0530 Subject: [FFmpeg-user] Segmentation fault when opening source file (MXF). Debugging? In-Reply-To: References: Message-ID: Well, without running gdb, its very difficult to shed any light or to gauge and say, what might be going wrong! The CORE files would help! Thnx/BR, Deepesh +49-89-2351-4997 On Wed, Mar 28, 2012 at 4:14 PM, Deepesh Basu wrote: > Could you please provide us with the Core-Dumps and DEBUG logs, from both > the systems, as well as a Minimal H/W configuration (binutils, glibc wud > suffice) differences, please? > > Thnx/BR, > Deepesh > +49-89-2351-4997 > > > On Wed, Mar 28, 2012 at 4:11 PM, Iban Garcia wrote: > >> Well... it is a bit strange... In a machine I get "segmentation fault" and >> in another different machine I don't get it. >> >> # ffmpeg -i input.MXF >> ffmpeg version N-39247-g6809818 Copyright (c) 2000-2012 the FFmpeg >> developers >> built on Mar 28 2012 11:41:42 with gcc 4.5.1 20101208 [gcc-4_5-branch >> revision 167585] >> configuration: --shlibdir=/usr/lib64 --prefix=/usr/local >> --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads >> --enable-shared --enable-libvorbis --enable-gpl --enable-x11grab >> --enable-libx264 --enable-libmp3lame --enable-nonfree --enable-postproc >> --enable-debug=3 --disable-optimizations --disable-mmx >> libavutil 51. 44.100 / 51. 44.100 >> libavcodec 54. 12.100 / 54. 12.100 >> libavformat 54. 3.100 / 54. 3.100 >> libavdevice 53. 4.100 / 53. 4.100 >> libavfilter 2. 65.102 / 2. 65.102 >> libswscale 2. 1.100 / 2. 1.100 >> libswresample 0. 10.100 / 0. 10.100 >> libpostproc 52. 0.100 / 52. 0.100 >> Segmentation fault >> >> >> # ffmpeg -i imput.MXF >> ffmpeg version 0.9.0.git, Copyright (c) 2000-2012 the FFmpeg developers >> built on Jan 5 2012 12:22:34 with gcc 4.5.1 20101208 [gcc-4_5-branch >> revision 167585] >> configuration: --shlibdir=/usr/lib64 --prefix=/usr/local >> --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads >> --enable-shared --enable-libvorbis --enable-libfaac --enable-gpl >> --enable-x11grab --enable-libx264 --enable-libmp3lame --enable-libtheora >> --enable-libxvid --enable-nonfree --enable-postproc --enable-pthreads >> libavutil 51. 33.100 / 51. 33.100 >> libavcodec 53. 50.100 / 53. 50.100 >> libavformat 53. 29.100 / 53. 29.100 >> libavdevice 53. 4.100 / 53. 4.100 >> libavfilter 2. 57.101 / 2. 57.101 >> libswscale 2. 1.100 / 2. 1.100 >> libswresample 0. 5.100 / 0. 5.100 >> libpostproc 51. 2.100 / 51. 2.100 >> Input #0, mxf, from '/net/online/satabio/HD_FROGAK/AC100/remen2.MXF': >> Duration: 00:00:35.00, start: 0.000000, bitrate: 117557 kb/s >> Stream #0:0: Video: h264 (High 4:2:2 Intra), yuv422p10le, 1280x720 [SAR >> 1:1 DAR 16:9], 50 fps, 50 tbr, 50 tbn, 100 tbc >> Stream #0:1: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s >> Stream #0:2: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s >> Stream #0:3: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s >> Stream #0:4: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s >> >> >> >> 2012/3/28 Carl Eugen Hoyos >> >> > Iban Garcia gmail.com> writes: >> > >> > > I have launched this command and I am getting >> > > "Segmentation fault" message. >> > >> > Please provide the sample, segmentation faults >> > are always important bugs. >> > >> > Carl Eugen >> > >> > _______________________________________________ >> > ffmpeg-user mailing list >> > ffmpeg-user at ffmpeg.org >> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > From softteam at hotmail.com Wed Mar 28 13:08:18 2012 From: softteam at hotmail.com (J.F. Sebastian) Date: Wed, 28 Mar 2012 11:08:18 +0000 Subject: [FFmpeg-user] set total number of output frames when extracting pictures from movie In-Reply-To: <4F71B5C5.4040602@alcatel-lucent.com> References: , <4F71B5C5.4040602@alcatel-lucent.com> Message-ID: > > I'm extracting pictures of movies using ffmpeg. > > In the past I was abled to specify the number of frames to export (total number, not fps!)This was very handy, because I sometimes needed more pictures than the movie contained (because of gaps in the recording)and ffmpeg filled the gaps with duplicate pictures. > > Now I can't find my old shell command and I'm either too stupid or too blind [or both] to find the command line parameter :-(Can you give me hint! > > Thanks a lot!J.F.Sebastian > > > -vframes N > > In recent versions of ffmpeg, I had to move it after -vcodec to make it > to work. Thanks for helping me!I already tried -vframes but my fault was that I also had the -r parameter (fps).When using -r the parameter -vframes is ignored. J.F. From cehoyos at ag.or.at Wed Mar 28 13:17:46 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 28 Mar 2012 11:17:46 +0000 (UTC) Subject: [FFmpeg-user] set total number of output frames when extracting pictures from movie References: , <4F71B5C5.4040602@alcatel-lucent.com> Message-ID: J.F. Sebastian hotmail.com> writes: > When using -r the parameter -vframes is ignored. Command line and complete, uncut console output missing. (Works fine here.) Carl Eugen From cehoyos at ag.or.at Wed Mar 28 13:20:18 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 28 Mar 2012 11:20:18 +0000 (UTC) Subject: [FFmpeg-user] Segmentation fault when opening source file (MXF). Debugging? References: Message-ID: Iban Garcia gmail.com> writes: > Segmentation fault Please provide the sample - or at least backtrace etc. as explained on http://ffmpeg.org/bugreports.html - segmentation faults are always important bugs. Please do not top-post here, it is considered rude. Carl Eugen From anders.branderud at gmail.com Wed Mar 28 13:29:15 2012 From: anders.branderud at gmail.com (Anders Branderud) Date: Wed, 28 Mar 2012 13:29:15 +0200 Subject: [FFmpeg-user] Fwd: Code which encodes file using MPEG4-codec, decodes the encoded data and then plays the file? In-Reply-To: <4F727489.7010704@gmail.com> References: <4F727489.7010704@gmail.com> Message-ID: Thanks for the reply! I want to incorportate this in a code. When I have found a way to code the below, I will also add Application-FEC (Forward Erasure Correction) from openfec.org .. It takes chunks of a certain size - e.g. 1024 bytes per each; and then for all the units of e.g. 1024 bytes it is given, it create FEC-encoded units of data. Some FEC-encoded units will contain only parts of a frame; and some FEC-encoded units will contain several frames. After FEC-decoding the FEC-encoded units, how will I be able to distinguish the different frames in the FEC-decoded data? Thinking of it now when I am writing, I realize that I can add this information in some bytes of each packet that I send over the network. Say that I have 25 MPEG4-encoded frames per FEC-encoded unit. E.g. if I have five frames in one packet, I could add the following data in the beginning of the packet sent over UDP: Nr of the packet sent, number of bytes containing meta data of the MPEG4-encoded data, byte containing info. whether the first frame of the packet is the continuation of a frame sent in the previous packet and info whether the last frame also is contained in the next packet; nr of bytes of first frame, nr of bytes of second frame, nr of bytes of third frame, nr of bytes of fourth frame, nr of bytes of fifth frame. All bytes, except the first byte are encoded with FEC. Receiver-side: A fix number of packets are included in each FEC-session - and which FEC-session a packet belongs to is determined based on the package-number in the first byte of the received package. The FEC-decoder examines for each received package of a FEC-session whether the data is decoded yet or not. When the data is decoded, the FEC-decoder decodes the data. Now, using the meta data specified above, the program starts doing the following: Check the second byte, and go through the number of bytes specified by it. Based on the data in these bytes, reassemble the frames and put them in the kind of queue/similar that FFMPEG is using for playback (I haven't examined this yet). -Do playback continiously of the frames that are next in order to be played (in a separate thread), at the same time as the decoding takes place. What do you think of this proposed solution? Would it work? Any more efficient/better solution? Any example code available for parts of what I am desiring to implement? Thanks! Kind regards, Anders [Personal blog] Will of the Creator : Logical reasons - based on scientific premises - for the existence of a Super intelligent and Orderly Creator and that He hasn't left His sapient creatures without an Instruction Manual - Torah ['books of Moses'] - to ascertain, and aspire to, His purpose. [Company] Anders Branderud IT Solutions - www.abitsolutions.org ---------- Forwarded message ---------- From: dE . Date: 2012/3/28 Subject: Re: [FFmpeg-user] Code which encodes file using MPEG4-codec, decodes the encoded data and then plays the file? To: FFmpeg user questions and RTFMs On 03/28/12 01:11, Anders Branderud wrote: > Hello! > I wondered if anyone has created a code which does the following: > Encodes file using MPEG4-codec, decodes the encoded data; and play each 25 > frames continiously as 25 new frames are decoded (If so, I would be very > thankful to get it mailed to me!)? > > I am aware of the decoding_encoding.c-example (in /doc/examples of the > FFMPEG-library) and wonder which changes that are needed to be made in > order to accomplish this? > > Thanks in advance! > > Are you going to incorporate this in a code? If not, you may use pipe, but if you want blocks of 25 frames, I'm not sure... ______________________________**_________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/**listinfo/ffmpeg-user From salmjuh at hotmail.com Wed Mar 28 13:55:26 2012 From: salmjuh at hotmail.com (juha s.) Date: Wed, 28 Mar 2012 11:55:26 +0000 Subject: [FFmpeg-user] =?windows-1256?q?moov_atom_not_found=FE___/_av=5Fin?= =?windows-1256?q?terleaved=5Fwrite=5Fframe=28=29=3A_Invalid_argument?= Message-ID: Hi Here is debug information about one file that is broken. Please, can somebody help me with this one ? root at CVod1:/DVB# ffmpeg -i /videos/Lives/18928653.mp4 ffmpeg version N-34813-g96d0494 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 28 2012 08:37:13 with gcc 4.6.1 configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab --enable-libvpx libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 66.100 / 2. 66.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x30aa3e0] moov atom not found /videos/Lives/18928653.mp4: Invalid data found when processing input root at CVod1:/DVB# root at CVod1:/DVB# cat /DVB/Log/18928653.Lives ffmpeg version N-34813-g96d0494 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 28 2012 08:37:13 with gcc 4.6.1 configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab --enable-libvpx libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 66.100 / 2. 66.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 [rtsp @ 0x27b13e0] SDP: v=0 o=1332932278761000 1 IN IP4 10.71.31.232 s=Lives i=--- t=0 0 a=tool:--- Streaming a=type:broadcast a=control:* a=source-filter: incl IN IP4 * 10.71.31.232 a=rtcp:unicast reflection a=range:npt=0- a=x-qt-text-nam:Lives a=x-qt-text-inf:--- m=audio 0 RTP/AVP 96 c=IN IP4 0.0.0.0 a=rtpmap:96 MPEG4-GENERIC/90000 a=fmtp:96 streamtype=5;profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;bitrate=2;config=1390 a=control:trackID=1 m=video 0 RTP/AVP 97 c=IN IP4 0.0.0.0 a=rtpmap:97 H264/90000 a=fmtp:97 packetization-mode=1;profile-level-id=4D4020;sprop-parameter-sets=Z01AIJZWBQF/y/+AAYACCAAAAwAIAAADAZQg,aO8GDMg= a=control:trackID=2 [rtsp @ 0x27b13e0] audio codec set to: aac [rtsp @ 0x27b13e0] audio samplerate set to: 90000 [rtsp @ 0x27b13e0] audio channels set to: 1 [rtsp @ 0x27b13e0] video codec set to: h264 [NULL @ 0x27bffe0] RTP Packetization Mode: 1 [NULL @ 0x27bffe0] RTP Profile IDC: 4d Profile IOP: 40 Level: 20 [NULL @ 0x27bffe0] Extradata set to 0x27c05a0 (size: 40)!hello state=0 [h264 @ 0x27bffe0] Missing reference picture [h264 @ 0x27bffe0] decode_slice_header error [h264 @ 0x27bffe0] concealing 920 DC, 920 AC, 920 MV errors [h264 @ 0x27bffe0] Current profile doesn't provide more RBSP data in PPS, skipping Last message repeated 1 times [rtsp @ 0x27b13e0] All info found [rtsp @ 0x27b13e0] Estimating duration from bitrate, this may be inaccurate Input #0, rtsp, from 'rtsp://10.71.31.232:8554': Metadata: title : Lives comment : --- Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0, 19, 1/90000: Audio: aac, 44100 Hz, stereo, s16 Stream #0:1, 22, 1/90000: Video: h264 (Main), yuv420p, 640x360 [SAR 3:4 DAR 4:3], 1/50, 25 fps, 25 tbr, 90k tbn, 50 tbc [buffer @ 0x280ad60] w:640 h:360 pixfmt:yuv420p tb:1/1000000 sar:3/4 sws_param: [libx264 @ 0x2809780] using mv_range_thread = 24 [libx264 @ 0x2809780] using SAR=1/1 [libx264 @ 0x2809780] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x2809780] profile High, level 3.0 [libx264 @ 0x2809780] 264 - core 122 r2184 5c85e0a - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=2 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=6 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=24 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=1 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=30 rc=abr mbtree=1 bitrate=900 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 [h264 @ 0x27bffe0] detected 16 logical cores Output #0, mp4, to '/videos/Lives/18928653.mp4': Metadata: title : Lives comment : --- encoder : Lavf54.3.100 Stream #0:0, 0, 1/25: Video: h264 (![0][0][0] / 0x0021), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 1/25, q=-1--1, 900 kb/s, 25 tbn, 25 tbc Stream #0:1, 0, 1/44100: Audio: aac (@[0][0][0] / 0x0040), 44100 Hz, stereo, s16, 128 kb/s Stream mapping: Stream #0:1 -> #0:0 (h264 -> libx264) Stream #0:0 -> #0:1 (aac -> libfaac) Press [q] to stop, [?] for help [h264 @ 0x401af80] Missing reference picture [h264 @ 0x401af80] decode_slice_header error [h264 @ 0x401af80] concealing 920 DC, 920 AC, 920 MV errors [h264 @ 0x441c640] Increasing reorder buffer to 1 [h264 @ 0x441c640] no picture *** 1 dup! [libx264 @ 0x2809780] using mv_range_thread = 24 [h264 @ 0x27bffe0] Current profile doesn't provide more RBSP data in PPS, skipping Current profile doesn't provide more RBSP data in PPS, skippingate= 0.0kbits/s dup=1 drop=0 Current profile doesn't provide more RBSP data in PPS, skippingate= 0.0kbits/s dup=1 drop=0 [libx264 @ 0x2809780] scene cut at 10 Icost:192175 Pcost:192175 ratio:0.0000 bias:0.0400 gop:10 (imb:798 pmb:0) [h264 @ 0x27bffe0] Current profile doesn't provide more RBSP data in PPS, skipping Current profile doesn't provide more RBSP data in PPS, skippingate= 0.0kbits/s dup=1 drop=0 [libx264 @ 0x2809780] frame= 0 QP=10.00 NAL=3 Slice:I Poc:0 I:920 P:0 SKIP:0 size=87 bytes [libx264 @ 0x2809780] frame= 1 QP=29.00 NAL=2 Slice:P Poc:8 I:0 P:0 SKIP:920 size=19 bytes [libx264 @ 0x2809780] frame= 2 QP=29.00 NAL=2 Slice:B Poc:4 I:0 P:0 SKIP:920 size=17 bytes [libx264 @ 0x2809780] frame= 3 QP=29.00 NAL=0 Slice:B Poc:2 I:0 P:0 SKIP:920 size=17 bytes frame= 4 QP=29.00 NAL=0 Slice:B Poc:6 I:0 P:0 SKIP:920 size=16 bytes/s dup=1 drop=0 [libx264 @ 0x2809780] frame= 5 QP=29.00 NAL=2 Slice:P Poc:16 I:0 P:0 SKIP:920 size=20 bytes [libx264 @ 0x2809780] frame= 6 QP=29.00 NAL=2 Slice:B Poc:12 I:0 P:0 SKIP:920 size=18 bytes [libx264 @ 0x2809780] frame= 7 QP=29.00 NAL=0 Slice:B Poc:10 I:0 P:0 SKIP:920 size=17 bytes [libx264 @ 0x2809780] frame= 8 QP=29.00 NAL=0 Slice:B Poc:14 I:0 P:0 SKIP:920 size=16 bytes [libx264 @ 0x2809780] frame= 9 QP=29.00 NAL=2 Slice:P Poc:18 I:0 P:0 SKIP:920 size=20 bytes [libx264 @ 0x2809780] frame= 10 QP=19.54 NAL=2 Slice:I Poc:20 I:920 P:0 SKIP:0 size=11958 bytes [libx264 @ 0x2809780] frame= 11 QP=22.51 NAL=2 Slice:P Poc:28 I:17 P:651 SKIP:252 size=2099 bytes [libx264 @ 0x2809780] frame= 12 QP=24.66 NAL=2 Slice:B Poc:24 I:3 P:414 SKIP:503 size=690 bytes [h264 @ 0x27bffe0] Current profile doesn't provide more RBSP data in PPS, skipping [libx264 @ 0x2809780] frame= 13 QP=26.27 NAL=0 Slice:B Poc:22 I:1 P:280 SKIP:639 size=397 bytes [libx264 @ 0x2809780] frame= 14 QP=25.29 NAL=0 Slice:B Poc:26 I:2 P:359 SKIP:559 size=499 bytes [libx264 @ 0x2809780] frame= 15 QP=22.92 NAL=2 Slice:P Poc:36 I:16 P:622 SKIP:282 size=2078 bytes frame= 16 QP=24.69 NAL=2 Slice:B Poc:32 I:14 P:452 SKIP:454 size=863 bytess dup=1 drop=0 [libx264 @ 0x2809780] frame= 17 QP=25.94 NAL=0 Slice:B Poc:30 I:1 P:232 SKIP:687 size=393 bytes [libx264 @ 0x2809780] frame= 18 QP=25.30 NAL=0 Slice:B Poc:34 I:1 P:218 SKIP:701 size=324 bytes [libx264 @ 0x2809780] frame= 19 QP=23.44 NAL=2 Slice:P Poc:44 I:34 P:683 SKIP:203 size=2630 bytes [libx264 @ 0x2809780] frame= 20 QP=24.34 NAL=2 Slice:B Poc:40 I:10 P:379 SKIP:531 size=753 bytes [libx264 @ 0x2809780] frame= 21 QP=26.11 NAL=0 Slice:B Poc:38 I:15 P:356 SKIP:549 size=584 bytes [libx264 @ 0x2809780] frame= 22 QP=25.70 NAL=0 Slice:B Poc:42 I:2 P:192 SKIP:726 size=304 bytes [libx264 @ 0x2809780] frame= 23 QP=22.63 NAL=2 Slice:P Poc:52 I:52 P:721 SKIP:147 size=3692 bytes [libx264 @ 0x2809780] frame= 24 QP=24.73 NAL=2 Slice:B Poc:48 I:10 P:472 SKIP:435 size=1066 bytes [libx264 @ 0x2809780] frame= 25 QP=26.20 NAL=0 Slice:B Poc:46 I:14 P:391 SKIP:515 size=704 bytes [h264 @ 0x27bffe0] Current profile doesn't provide more RBSP data in PPS, skipping [libx264 @ 0x2809780] frame= 26 QP=26.35 NAL=0 Slice:B Poc:50 I:6 P:200 SKIP:713 size=389 bytes [libx264 @ 0x2809780] frame= 27 QP=18.93 NAL=2 Slice:P Poc:60 I:87 P:785 SKIP:48 size=7833 bytes [libx264 @ 0x2809780] frame= 28 QP=22.92 NAL=2 Slice:B Poc:56 I:13 P:541 SKIP:362 size=1405 bytes [libx264 @ 0x2809780] frame= 29 QP=24.92 NAL=0 Slice:B Poc:54 I:2 P:321 SKIP:597 size=627 bytes [libx264 @ 0x2809780] frame= 30 QP=23.25 NAL=0 Slice:B Poc:58 I:9 P:500 SKIP:409 size=1057 bytes frame= 31 QP=11.36 NAL=2 Slice:P Poc:68 I:223 P:697 SKIP:0 size=25242 bytesdup=1 drop=0 [libx264 @ 0x2809780] frame= 32 QP=17.26 NAL=2 Slice:B Poc:64 I:23 P:778 SKIP:93 size=4188 bytes [libx264 @ 0x2809780] frame= 33 QP=20.60 NAL=0 Slice:B Poc:62 I:6 P:591 SKIP:320 size=1399 bytes [libx264 @ 0x2809780] frame= 34 QP=16.77 NAL=0 Slice:B Poc:66 I:5 P:635 SKIP:262 size=2147 bytes [libx264 @ 0x2809780] scene cut at 61 Icost:143962 Pcost:141511 ratio:0.0170 bias:0.1480 gop:61 (imb:769 pmb:29) [libx264 @ 0x2809780] frame= 35 QP=9.76 NAL=2 Slice:P Poc:76 I:651 P:269 SKIP:0 size=39634 bytes [libx264 @ 0x2809780] frame= 36 QP=13.43 NAL=2 Slice:B Poc:72 I:77 P:739 SKIP:40 size=12781 bytes [libx264 @ 0x2809780] frame= 37 QP=14.42 NAL=0 Slice:B Poc:70 I:50 P:771 SKIP:63 size=6841 bytes [libx264 @ 0x2809780] frame= 38 QP=13.95 NAL=0 Slice:B Poc:74 I:6 P:698 SKIP:189 size=3218 bytes [h264 @ 0x27bffe0] Current profile doesn't provide more RBSP data in PPS, skipping [libx264 @ 0x2809780] frame= 39 QP=8.53 NAL=2 Slice:P Poc:84 I:197 P:723 SKIP:0 size=29804 bytes [mp4 @ 0x2809240] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 199680 >= 199212 av_interleaved_write_frame(): Invalid argument root at CVod1:/DVB# From ebisumartin at gmail.com Wed Mar 28 14:50:17 2012 From: ebisumartin at gmail.com (Martin G) Date: Wed, 28 Mar 2012 21:50:17 +0900 Subject: [FFmpeg-user] -t option is ignored In-Reply-To: <20120328092325.GI5169@leki> References: <201203270028.38628.rodney.baker@iinet.net.au> <4F707D9F.6000905@yahoo.com> <4F713631.5050502@gmail.com> <20120328092325.GI5169@leki> Message-ID: > You may check https://launchpad.net/~jon-severinsson/+archive/ffmpeg for > FFmpeg Okay, I installed that repository, and it worked much better. Not perfectly, though. The subclip being created is about 6 minutes long, but after two minutes it freezes, and the remaining 4 minutes there is no motion or sound. Is there a way I can make two minutes be actually two minutes without this extraneous time tacked onto the end? This is the output: $ ffmpeg -ss 01:05:16 -t 120 -i input.avi -vcodec copy -acodec copy output.avi ffmpeg version 0.10.2-4:0.10.2-0ubuntu0jon1~oneiric1 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 18 2012 11:07:55 with gcc 4.6.1 configuration: --extra-version='4:0.10.2-0ubuntu0jon1~oneiric1' --arch=amd64 --prefix=/usr --libdir=/usr/lib/x86_64-linux-gnu --disable-stripping --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-libfreetype --enable-vaapi --enable-frei0r --enable-gpl --enable-postproc --enable-x11grab --enable-librtmp --enable-libvo-aacenc --enable-version3 --enable-libvo-amrwbenc --enable-version3 --enable-libdc1394 --shlibdir=/usr/lib/x86_64-linux-gnu --enable-shared --disable-static avutil configuration: --extra-version='4:0.10.2.0jon1~oneiric1' --arch=amd64 --prefix=/usr --libdir=/usr/lib/x86_64-linux-gnu --disable-stripping --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-libfreetype --enable-vaapi --enable-frei0r --enable-libopenjpeg --enable-gpl --enable-postproc --enable-x11grab --enable-libdirac --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libopencore-amrnb --enable-version3 --enable-libopencore-amrwb --enable-version3 --enable-libvo-aacenc --enable-version3 --enable-libvo-amrwbenc --enable-version3 --enable-libdc1394 --shlibdir=/usr/lib/x86_64-linux-gnu --enable-shared --disable-static avcodec configuration: --extra-version='4:0.10.2.0jon1~oneiric1' --arch=amd64 --prefix=/usr --libdir=/usr/lib/x86_64-linux-gnu --disable-stripping --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-libfreetype --enable-vaapi --enable-frei0r --enable-libopenjpeg --enable-gpl --enable-postproc --enable-x11grab --enable-libdirac --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libopencore-amrnb --enable-version3 --enable-libopencore-amrwb --enable-version3 --enable-libvo-aacenc --enable-version3 --enable-libvo-amrwbenc --enable-version3 --enable-libdc1394 --shlibdir=/usr/lib/x86_64-linux-gnu --enable-shared --disable-static libavutil 51. 35.100 / 51. 35.100 libavcodec 53. 61.100 / 53. 61.100 libavformat 53. 32.100 / 53. 32.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 61.100 / 2. 61.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 52. 0.100 / 52. 0.100 [mpeg4 @ 0x1defa40] Invalid and inefficient vfw-avi packed B frames detected Input #0, avi, from 'input.avi': Metadata: title : Triumph Des Willens comment : Ripped by Reevel encoder : VirtualDubMod 1.4.13 Duration: 01:50:30.50, start: 0.000000, bitrate: 886 kb/s Stream #0:0: Video: mpeg4 (DX50 / 0x30355844), yuv420p, 512x400 [SAR 1:1 DAR 32:25], 23.98 fps, 23.98 tbr, 23.98 tbn, 30k tbc Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 44100 Hz, mono, s16, 32 kb/s File 'output.avi' already exists. Overwrite ? [y/N] y Output #0, avi, to 'output.avi': Metadata: INAM : Triumph Des Willens ILNG : Undefined ICMT : Ripped by Reevel ISFT : Lavf53.32.100 Stream #0:0: Video: mpeg4 (DX50 / 0x30355844), yuv420p, 512x400 [SAR 1:1 DAR 32:25], q=2-31, 23.98 fps, 23.98 tbn, 23.98 tbc Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 44100 Hz, mono, 32 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help [NULL @ 0x1defa40] Invalid and inefficient vfw-avi packed B frames detected frame= 2520 fps= 0 q=-1.0 size= 10466kB time=00:01:36.30 bitrate= 890.3kbits/frame= 3089 fps= 0 q=-1.0 Lsize= 14005kB time=00:02:00.00 bitrate= 956.0kbits/s video:12322kB audio:1480kB global headers:0kB muxing overhead 1.473870% From matthew.salisbury at autonomy.com Wed Mar 28 15:00:47 2012 From: matthew.salisbury at autonomy.com (matthew.salisbury) Date: Wed, 28 Mar 2012 15:00:47 +0200 Subject: [FFmpeg-user] FFmpeg & referenced .mov files Message-ID: <019a01cd0ce2$cb6098a0$6221c9e0$@salisbury@autonomy.com> Hello, Is it possible generate referenced .mov files with FFmpeg? If so could you please provide a hint as to where I would find more information on this topic? I have been searching Google without success. Regards, Matthew Salisbury From ubitux at gmail.com Wed Mar 28 15:07:29 2012 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Wed, 28 Mar 2012 15:07:29 +0200 Subject: [FFmpeg-user] -t option is ignored In-Reply-To: References: <201203270028.38628.rodney.baker@iinet.net.au> <4F707D9F.6000905@yahoo.com> <4F713631.5050502@gmail.com> <20120328092325.GI5169@leki> Message-ID: <20120328130729.GJ5169@leki> On Wed, Mar 28, 2012 at 09:50:17PM +0900, Martin G wrote: > > You may check https://launchpad.net/~jon-severinsson/+archive/ffmpeg for > > FFmpeg > > Okay, I installed that repository, and it worked much better. > > Not perfectly, though. The subclip being created is about 6 minutes > long, but after two minutes it freezes, and the remaining 4 minutes > there is no motion or sound. > > Is there a way I can make two minutes be actually two minutes without > this extraneous time tacked onto the end? > > This is the output: > > $ ffmpeg -ss 01:05:16 -t 120 -i input.avi -vcodec copy -acodec copy output.avi Could you try moving the -ss and -t after the -i now? :) before -i: input option, faster, less accurate after -i: output option, slower, more accurate Note: you can now use "-c copy" (or "-c:v copy -c:a copy") instead of the long -Xcodec version. [...] -- Cl?ment B. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 490 bytes Desc: not available URL: From licanjah at gmail.com Wed Mar 28 09:30:53 2012 From: licanjah at gmail.com (=?Big5?B?rHi4UrX6?=) Date: Wed, 28 Mar 2012 15:30:53 +0800 Subject: [FFmpeg-user] report a configure bug Message-ID: Hi, I'm configuring ffmpeg under linux os, and it turn out like below: ERROR: libcelt not found my computer is lenovo t60p and with linux fedora 12-i686 i'have install celt, celt-devel with yum, and version is 0.7.0.1.fc12.i686 thanks for reading. Best Regards, Destin Hung -------------- next part -------------- A non-text attachment was scrubbed... Name: config.log Type: text/x-log Size: 155905 bytes Desc: not available URL: From xjf_hw at sina.com Tue Mar 27 15:00:30 2012 From: xjf_hw at sina.com (xjf_hw at sina.com) Date: Tue, 27 Mar 2012 21:00:30 +0800 Subject: [FFmpeg-user] for help in ffmpeg-0.10.2 Message-ID: <20120327130030.A8243DD8001@webmail.sinamail.sina.com.cn> Dear,Sir or Madam: For help , In the ffmpeg-0.10.2 , # ./configure --prefix=/usr/local/linphone/linphone_x86/ --enable-gpl --enable-shared --enable-swscale --enable-pthreads yasm not found, use --disable-yasm for a crippled build config.log in the enclosure. Please check. Thank you! Yours truly. -------------- next part -------------- A non-text attachment was scrubbed... Name: config.log Type: application/octet-stream Size: 110163 bytes Desc: not available URL: From brendan.brewster at gmail.com Wed Mar 28 17:42:32 2012 From: brendan.brewster at gmail.com (Brendan Brewster) Date: Wed, 28 Mar 2012 11:42:32 -0400 Subject: [FFmpeg-user] Compressibility Check In-Reply-To: <4F7273EF.5020503@gmail.com> References: <4F7139CF.1080209@gmail.com> <4F7273EF.5020503@gmail.com> Message-ID: On Tue, Mar 27, 2012 at 10:14 PM, dE . wrote: > On 03/27/12 21:34, Brendan Brewster wrote: > >> On Mon, Mar 26, 2012 at 11:53 PM, dE . wrote: >> >> On 03/27/12 04:52, Brendan Brewster wrote: >>> >>> Hi, >>>> >>>> I am trying to implement a compressibility check via ffmpeg to determine >>>> an >>>> appropriate resolution for the full encode. >>>> >>>> Can you please be more clear about what you're trying to do? What do >>> you >>> mean by 'compressibility'. Changing resolution is just one of changing >>> compression level. >>> >>> The idea is to take some percentage of the input, say 5%, at various >> points >> throughout and encode this with the default (input) resolution and >> bitrate. >> I have a specific file size to meet, such as DVD5, and therefore know the >> bitrate cap in order to fit my output onto the destination media (given a >> certain audio bitrate). I can then determine the number of bits per pixel >> for the default encode and prepare ratios with regard to lesser >> resolutions >> with the enforced bitrate. We choose a lower resolution if the ratio is >> poor. This is what I mean by a compressibility check. Similar >> functionality >> was implemented by the legacy Gordian Knot as well as within Tuxrip. >> >> I would just like to speed up my analysis by avoiding the majority of the >> seek time incurred especially in passes that pinpoint points later in the >> input video stream. I was really hoping that there might be a way to >> achieve the same end via one pass. >> >> Thanks, >> Brendan >> ______________________________**_________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/**listinfo/ffmpeg-user >> > > From what I understand, you're asking for CBR, but that can be done using > a combination of -maxrate and -minrate. That will reduce on the quality not > resolution. This way you can keep the resolution constant. > > ______________________________**_________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/**listinfo/ffmpeg-user > I appreciate your response but no, I'm not looking for guidance on achieving CBR. My intent is to pick an appropriate resolution for the given bitrate. I have no issues staying within my target output size during the full encode no matter the resolution. All I'm trying to do is optimize my current solution for the initial analysis prior to the actual full encode. Maybe someone else has some thoughts for me :) Thanks, Brendan From koxaniy at mail.ru Wed Mar 28 17:26:32 2012 From: koxaniy at mail.ru (Tuuls) Date: Wed, 28 Mar 2012 08:26:32 -0700 (PDT) Subject: [FFmpeg-user] FFmpeg & referenced .mov files In-Reply-To: <019a01cd0ce2$cb6098a0$6221c9e0$@salisbury@autonomy.com> References: <019a01cd0ce2$cb6098a0$6221c9e0$@salisbury@autonomy.com> Message-ID: <1332948392193-4512597.post@n4.nabble.com> Unfortunately, ffmpeg does not know how to read them :( -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/FFmpeg-referenced-mov-files-tp4512216p4512597.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From oussama.stiti at gmail.com Wed Mar 28 18:37:12 2012 From: oussama.stiti at gmail.com (Oussama Stiti) Date: Thu, 29 Mar 2012 01:37:12 +0900 Subject: [FFmpeg-user] input rtp stream Message-ID: Hello, I'm broadcasting a video with vlc on rtp://127.0.0.1:5004. When i try the command : ffmpeg -report -loglevel warning -i rtp:// 127.0.0.1:5004 -stats -f null - It gives me the following result: [h264 @ 0x9cbe0a0] non-existing PPS referenced [h264 @ 0x9cbe0a0] non-existing PPS 0 referenced [h264 @ 0x9cbe0a0] decode_slice_header error [h264 @ 0x9cbe0a0] no frame! [h264 @ 0x9cbe0a0] non-existing PPS referenced [h264 @ 0x9cbe0a0] non-existing PPS 0 referenced [h264 @ 0x9cbe0a0] decode_slice_header error [h264 @ 0x9cbe0a0] no frame! [h264 @ 0x9cbe0a0] non-existing PPS referenced [h264 @ 0x9cbe0a0] non-existing PPS 0 referenced [h264 @ 0x9cbe0a0] decode_slice_header error [h264 @ 0x9cbe0a0] no frame! [...] DTS 4519, next:92863 st:0 invalid droping DTS 6609, next:116082 st:0 invalid droping DTS 8698, next:139301 st:0 invalid droping DTS 10788, next:162520 st:0 invalid droping DTS 9039, next:185739 st:0 invalid droping DTS 11128, next:208958 st:0 invalid droping DTS 13558, next:232177 st:0 invalid droping DTS 15648, next:255396 st:0 invalid droping DTS 17737, next:278615 st:0 invalid droping DTS 15988, next:301834 st:0 invalid droping DTS 18078, next:325053 st:0 invalid droping DTS 20167, next:348272 st:0 invalid droping What's wrong ? Regards -- *Oussama Stiti* From cehoyos at ag.or.at Wed Mar 28 18:38:09 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 28 Mar 2012 16:38:09 +0000 (UTC) Subject: [FFmpeg-user] FFmpeg & referenced .mov files References: <019a01cd0ce2$cb6098a0$6221c9e0$@salisbury@autonomy.com> <1332948392193-4512597.post@n4.nabble.com> Message-ID: Tuuls mail.ru> writes: > Unfortunately, ffmpeg does not know how to read them :( Could you provide the files of a non-working sample? There is no such open ticket and I thought this works fine. Carl Eugen From oussama.stiti at gmail.com Wed Mar 28 18:41:47 2012 From: oussama.stiti at gmail.com (Oussama Stiti) Date: Thu, 29 Mar 2012 01:41:47 +0900 Subject: [FFmpeg-user] Referenced frames Message-ID: Hello, I want to detect in a h264 video, which frames are related ( referenced) to which frames. For example : Frame 62 : I Frame. Frame 63 : P Frame [referenced to frame 62] Frame 64 : B Frame [referenced to frame 63 & 65] Is it possible to display such result ? What kind of command line, should i run ? Thank you Regards -- *Oussama Stiti* ?l?ve ing?nieur en t?l?communications ? Sup'com (?cole sup?rieure des communications de Tunis) T?l: +21652363164 E-mail: oussama.stiti at gmail.com From ibantxo28 at gmail.com Wed Mar 28 19:48:09 2012 From: ibantxo28 at gmail.com (Iban Garcia) Date: Wed, 28 Mar 2012 19:48:09 +0200 Subject: [FFmpeg-user] Segmentation fault when opening source file (MXF). Debugging? In-Reply-To: References: Message-ID: Well... very good news... I have downloaded the snapshot and re-installed again. "Segmentation Fault" message dissapeared!!! Many Thanks cehoyos ffmpeg version 0.10.2.git Copyright (c) 2000-2012 the FFmpeg developers built on Mar 28 2012 18:51:09 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] configuration: --shlibdir=/usr/lib64 --prefix=/usr/local --mandir=/usr/share/man --libdir=/usr/lib64 --enable-pthreads --enable-shared --enable-libvorbis --enable-gpl --enable-x11grab --enable-libx264 --enable-libmp3lame --enable-postproc libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 66.100 / 2. 66.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 2012/3/28 Carl Eugen Hoyos > Iban Garcia gmail.com> writes: > > > I have launched this command and I am getting > > "Segmentation fault" message. > > Please provide the sample, segmentation faults > are always important bugs. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From anders.branderud at gmail.com Wed Mar 28 22:39:06 2012 From: anders.branderud at gmail.com (Anders Branderud) Date: Wed, 28 Mar 2012 22:39:06 +0200 Subject: [FFmpeg-user] How to play two mp4-files after each other without noticable lag? Message-ID: How does one program (in C++/C-code) a program to play two short mp4-files subsequently after each other meeting the below constraint?: I want them to be played in such a way that it won't be noticed that there are two separate files. I will have hundreds of short mp4-files and want to play all of these subsequentially without that the viewer should notice it. One way would be to separate the mp4-files into their frames -- if FFMPEG provide some kind of player which can be fed with frames. I am looking for code of how to do this. Thanks in advance! -- *Kind regards Anders* [Personal blog] Will of the Creator : Logical reasons - based on scientific premises - for the existence of a Super intelligent and Orderly Creator and that He hasn't left His sapient creatures without an Instruction Manual - Torah ['books of Moses'] - to ascertain, and aspire to, His purpose. [Company] Anders Branderud IT Solutions - www.abitsolutions.org From koxaniy at mail.ru Wed Mar 28 23:48:08 2012 From: koxaniy at mail.ru (Tuuls) Date: Wed, 28 Mar 2012 14:48:08 -0700 (PDT) Subject: [FFmpeg-user] FFmpeg & referenced .mov files In-Reply-To: References: <019a01cd0ce2$cb6098a0$6221c9e0$@salisbury@autonomy.com> <1332948392193-4512597.post@n4.nabble.com> Message-ID: <1332971288900-4513843.post@n4.nabble.com> You can create this file yourself. This can be done about QuickTime Pro player. Open the file in it and "save as" a link to the movie. You will receive a reference file, which contains only a reference to the media, and if you delete a file to which it refers, it will not play. Still, you can create a reference file using Final Cut. There's more complicated. Created with the help of the reference file will contain all of the audio from the project Final Cut , and be sure to segments requiring rendering. Video does not require render to be due to references to the source video. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/FFmpeg-referenced-mov-files-tp4512216p4513843.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From funkyirish at gmail.com Thu Mar 29 05:46:16 2012 From: funkyirish at gmail.com (Josh long) Date: Wed, 28 Mar 2012 22:46:16 -0500 Subject: [FFmpeg-user] bgra to yuv In-Reply-To: References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: On Mon, Mar 26, 2012 at 9:49 PM, Paul B Mahol wrote: > On 3/27/12, Josh long wrote: > > On Mon, Mar 26, 2012 at 9:38 PM, Paul B Mahol wrote: > > > >> On 3/27/12, Josh long wrote: > >> >> > >> >> libschroedinger encoder in FFmpeg supports only yuv420p, yuv422p and > >> >> yuv444p. > >> >> > >> >> If your input is not one of this above you can not expect encoding to > >> >> be > >> >> lossless. > >> >> > >> >> > >> > Thanks. I realized that I made a typo above; I'm now converting to > >> > yuva420p, and then to yuv444p. Any ideas on why the image splits? > >> > >> yuva420->yuv444 is not lossless either. > >> _______________________________________________ > >> ffmpeg-user mailing list > >> ffmpeg-user at ffmpeg.org > >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > >> > > > > aha, thanks I suppose I've not done my homework. Do you know of any > routes > > that would be lossless? I have strong suspicions that the alpha channel > in > > my bgra files is not significant (meaning that it is a constant ff when > > looking at it in a hex editor). > > Note that it may be vlc bug if it fails to properly decode yuv444. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > This may be of interest to someone. When converting bgra to rgb24 ffmpeg messes up the colors. However if one first converts to rgba and then rgb24 this undesired effect is avoided. Also it doesn't fix everything, but mplayer and Gnome Mplayer play certain files accurately where VLC will not. Thanks again for all the information and help, Joshua From cehoyos at ag.or.at Thu Mar 29 08:21:03 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 29 Mar 2012 06:21:03 +0000 (UTC) Subject: [FFmpeg-user] bgra to yuv References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: Josh long gmail.com> writes: > When converting bgra to rgb24 ffmpeg messes up the colors. This sounds like a very important bug, please provide a command line and complete, uncut console output so I can reproduce the issue. Thank you, Carl Eugen From nicolas.george at normalesup.org Thu Mar 29 11:49:37 2012 From: nicolas.george at normalesup.org (Nicolas George) Date: Thu, 29 Mar 2012 11:49:37 +0200 Subject: [FFmpeg-user] report a configure bug In-Reply-To: References: Message-ID: <20120329094937.GB23293@phare.normalesup.org> Le nonidi 9 germinal, an CCXX, ??? a ?crit?: > I'm configuring ffmpeg under linux os, > and it turn out like below: > > ERROR: libcelt not found > > my computer is lenovo t60p > and with linux fedora 12-i686 > > i'have install celt, celt-devel with yum, > and version is 0.7.0.1.fc12.i686 > BEGIN /tmp/ffconf.tKzcQxfU.c > 1 extern int celt_decode(); > 2 int main(void){ celt_decode(); } > END /tmp/ffconf.tKzcQxfU.c > gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -std=c99 -fomit-frame-pointer -pthread -I/usr/include/freetype2 -c -o /tmp/ffconf.0OqoKRBI.o /tmp/ffconf.tKzcQxfU.c > gcc -Wl,--as-needed -o /tmp/ffconf.IXxn8Obe /tmp/ffconf.0OqoKRBI.o -lcelt0 -lbluray -lass -laacplus -lm -pthread -lbz2 -lz > /usr/bin/ld: cannot find -lcelt0 The excerpt of configure you quoted indicate your compiler is unable to find the necessary file to link with -lcelt0: it does look like a build environment problem, not a problem on ffmpeg's side. I tried to query the Fedora packages on the web but did not manage to find how to do it. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From tevans.uk at googlemail.com Thu Mar 29 12:41:33 2012 From: tevans.uk at googlemail.com (Tom Evans) Date: Thu, 29 Mar 2012 11:41:33 +0100 Subject: [FFmpeg-user] for help in ffmpeg-0.10.2 In-Reply-To: <20120327130030.A8243DD8001@webmail.sinamail.sina.com.cn> References: <20120327130030.A8243DD8001@webmail.sinamail.sina.com.cn> Message-ID: 2012/3/27 : > Dear,Sir or Madam: ? ? ? For help , > ? In the ffmpeg-0.10.2 , ? ?# ./configure --prefix=/usr/local/linphone/linphone_x86/ --enable-gpl --enable-shared --enable-swscale --enable-pthreads ? ? ? ? yasm not found, use --disable-yasm for a crippled build ? ? ?config.log ?in the enclosure. ? ? ? ? Please check. > ? ? Thank you! > ? ? Yours truly. Install yasm, or configure with --disable-yasm if you don't mind ffmpeg running ridiculously slowly. Cheers Tom From mobilityrulezy3h at gmail.com Thu Mar 29 17:46:32 2012 From: mobilityrulezy3h at gmail.com (Mobility Lab) Date: Thu, 29 Mar 2012 11:46:32 -0400 Subject: [FFmpeg-user] Obtaining SDP information from ffmpeg command line Message-ID: Hello, I am working on a project where one instance of ffmpeg will transmit video data to another via RTP across the internet to another ffmpeg instance. Under this scheme, I require the SDP information to give to the recieving end. >From looking on various posts on StackOverFlow and other's personal blogs, it seemsthat the SDP information should be printed to the console when ffmpeg is invoked with RTP as an output format. However, when I call ffmpeg, I see no SDP related information. Is there some special flag I need to define in order to display the desired information? As I understand, it should be explicitly stated in the information printed to the console, not hidden. I am making the following call in terminal: ffmpeg -fflags +genpts -i files\sample.mpg -an -vcodec libx264 -threads 0 -r 10 -g 45 -s 352x240 -deinterlace -flags +global_header -f rtp rtp://192.168.200.198:9008 I am using ffmpeg-0.8-win64-shared from 23-Jun-2011, on a Windows 7 machine. The project will eventually be ported to linux, but for now I need to develop it on Windows. However, I do get the same result on my Ubuntu VM. Does anybody know why I am not getting the SDP information, or how to get ffmpeg to output the information? Thank you, Andrew P.S. The attached .txt file contains the output from the console . -------------- next part -------------- ffmpeg version 0.8, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 23 2011 14:22:23 with gcc 4.5.3 configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --enable-memalign-hack --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib --disable-outdev=sdl libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 7. 0 / 53. 7. 0 libavformat 53. 4. 0 / 53. 4. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpeg @ 0000000000635A00] max_analyze_duration 5000000 reached at 5005000 Input #0, mpeg, from 'files\sample.mpg': Duration: 00:12:13.41, start: 0.213367, bitrate: 1388 kb/s Stream #0.0[0x1e0]: Video: mpeg1video, yuv420p, 352x240 [PAR 200:219 DAR 880:657], 1150 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 29.97 tbc Stream #0.1[0x1c0]: Audio: mp2, 44100 Hz, stereo, s16, 224 kb/s [buffer @ 0000000002331F60] w:352 h:240 pixfmt:yuv420p tb:1/1000000 sar:200/219 sws_param: [libx264 @ 000000000079ADA0] Default settings detected, using medium profile [libx264 @ 000000000079ADA0] using SAR=200/219 [libx264 @ 000000000079ADA0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.1 Cache64 [libx264 @ 000000000079ADA0] profile High, level 1.2 [libx264 @ 000000000079ADA0] 264 - core 115 r2008 4c552d8 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=3 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=10 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, rtp, to 'rtp://192.168.200.198:9008': Metadata: encoder : Lavf53.4.0 Stream #0.0: Video: libx264, yuv420p, 352x240 [PAR 200:219 DAR 880:657], q=2-31, 200 kb/s, 90k tbn, 10 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop, [?] for help frame= 55 fps= 9 q=25.0 size= 2kB time=00:00:00.60 bitrate= 28.4kbits/s dup=0 drop=105 frame= 142 fps= 21 q=25.0 size= 127kB time=00:00:09.30 bitrate= 112.2kbits/s dup=0 drop=279 frame= 230 fps= 32 q=25.0 size= 248kB time=00:00:18.10 bitrate= 112.4kbits/s dup=0 drop=455 frame= 263 fps= 22 q=25.0 size= 292kB time=00:00:21.40 bitrate= 111.9kbits/s dup=0 drop=520 frame= 323 fps= 18 q=25.0 size= 351kB time=00:00:27.40 bitrate= 104.8kbits/s dup=0 drop=640 frame= 354 fps= 15 q=25.0 size= 369kB time=00:00:30.50 bitrate= 99.0kbits/s dup=0 drop=702 frame= 450 fps= 18 q=25.0 size= 416kB time=00:00:40.10 bitrate= 85.0kbits/s dup=0 drop=894 frame= 533 fps= 21 q=25.0 size= 517kB time=00:00:48.40 bitrate= 87.5kbits/s dup=0 drop=1060 frame= 585 fps= 18 q=25.0 size= 654kB time=00:00:53.60 bitrate= 100.0kbits/s dup=0 drop=1163 frame= 616 fps= 16 q=25.0 size= 721kB time=00:00:56.70 bitrate= 104.1kbits/s dup=0 drop=1225 frame= 672 fps= 17 q=25.0 size= 834kB time=00:01:02.30 bitrate= 109.7kbits/s dup=0 drop=1337 frame= 726 fps= 19 q=25.0 size= 970kB time=00:01:07.70 bitrate= 117.3kbits/s dup=0 drop=1445 frame= 740 fps= 17 q=25.0 size= 1008kB time=00:01:09.10 bitrate= 119.5kbits/s dup=0 drop=1473 frame= 795 fps= 18 q=25.0 size= 1163kB time=00:01:14.60 bitrate= 127.7kbits/s dup=0 drop=1583 frame= 833 fps= 17 q=25.0 size= 1274kB time=00:01:18.40 bitrate= 133.1kbits/s dup=0 drop=1659 frame= 885 fps= 17 q=25.0 size= 1414kB time=00:01:23.60 bitrate= 138.5kbits/s dup=0 drop=1763 frame= 933 fps= 17 q=25.0 size= 1537kB time=00:01:28.40 bitrate= 142.5kbits/s dup=0 drop=1858 frame= 968 fps= 16 q=25.0 size= 1653kB time=00:01:31.90 bitrate= 147.3kbits/s dup=0 drop=1928 frame= 998 fps= 16 q=25.0 size= 1756kB time=00:01:34.90 bitrate= 151.6kbits/s dup=0 drop=1988 frame= 1050 fps= 17 q=25.0 size= 1914kB time=00:01:40.10 bitrate= 156.6kbits/s dup=0 drop=2092 frame= 1101 fps= 17 q=25.0 size= 2035kB time=00:01:45.20 bitrate= 158.5kbits/s dup=0 drop=2194 frame= 1141 fps= 18 q=25.0 size= 2150kB time=00:01:49.20 bitrate= 161.3kbits/s dup=0 drop=2274 frame= 1193 fps= 18 q=25.0 size= 2281kB time=00:01:54.40 bitrate= 163.3kbits/s dup=0 drop=2378 frame= 1243 fps= 19 q=25.0 size= 2420kB time=00:01:59.40 bitrate= 166.0kbits/s dup=0 drop=2477 frame= 1295 fps= 20 q=25.0 size= 2562kB time=00:02:04.60 bitrate= 168.4kbits/s dup=0 drop=2581 frame= 1318 fps= 18 q=25.0 size= 2617kB time=00:02:06.90 bitrate= 168.9kbits/s dup=0 drop=2627 frame= 1374 fps= 19 q=25.0 size= 2738kB time=00:02:12.50 bitrate= 169.3kbits/s dup=0 drop=2739 frame= 1434 fps= 20 q=25.0 size= 2871kB time=00:02:18.50 bitrate= 169.8kbits/s dup=0 drop=2859 frame= 1549 fps= 21 q=25.0 size= 2953kB time=00:02:30.00 bitrate= 161.3kbits/s dup=0 drop=3089 frame= 1581 fps= 20 q=25.0 size= 2988kB time=00:02:33.20 bitrate= 159.8kbits/s dup=0 drop=3152 frame= 1675 fps= 21 q=25.0 size= 3078kB time=00:02:42.60 bitrate= 155.1kbits/s dup=0 drop=3340 frame= 1726 fps= 21 q=25.0 size= 3212kB time=00:02:47.70 bitrate= 156.9kbits/s dup=0 drop=3442 frame= 1775 fps= 22 q=22.0 size= 3338kB time=00:02:52.60 bitrate= 158.4kbits/s dup=0 drop=3542 frame= 1781 fps= 20 q=25.0 size= 3359kB time=00:02:53.20 bitrate= 158.9kbits/s dup=0 drop=3552 frame= 1831 fps= 21 q=25.0 size= 3495kB time=00:02:58.20 bitrate= 160.7kbits/s dup=0 drop=3652 frame= 1880 fps= 21 q=25.0 size= 3616kB time=00:03:03.10 bitrate= 161.8kbits/s dup=0 drop=3750 frame= 1930 fps= 22 q=25.0 size= 3734kB time=00:03:08.10 bitrate= 162.6kbits/s dup=0 drop=3849 frame= 1978 fps= 22 q=25.0 size= 3872kB time=00:03:12.90 bitrate= 164.4kbits/s dup=0 drop=3945 frame= 2028 fps= 22 q=25.0 size= 4006kB time=00:03:17.90 bitrate= 165.8kbits/s dup=0 drop=4046 frame= 2053 fps= 21 q=25.0 size= 4064kB time=00:03:20.40 bitrate= 166.1kbits/s dup=0 drop=4095 frame= 2105 fps= 22 q=25.0 size= 4180kB time=00:03:25.60 bitrate= 166.5kbits/s dup=0 drop=4199 frame= 2152 fps= 22 q=25.0 size= 4291kB time=00:03:30.30 bitrate= 167.2kbits/s dup=0 drop=4293 frame= 2199 fps= 23 q=25.0 size= 4404kB time=00:03:35.00 bitrate= 167.8kbits/s dup=0 drop=4387 frame= 2221 fps= 21 q=25.0 size= 4458kB time=00:03:37.20 bitrate= 168.1kbits/s dup=0 drop=4431 frame= 2265 fps= 22 q=25.0 size= 4600kB time=00:03:41.60 bitrate= 170.1kbits/s dup=0 drop=4518 frame= 2314 fps= 22 q=25.0 size= 4744kB time=00:03:46.50 bitrate= 171.6kbits/s dup=0 drop=4616 frame= 2363 fps= 22 q=25.0 size= 4878kB time=00:03:51.40 bitrate= 172.7kbits/s dup=0 drop=4714 frame= 2406 fps= 21 q=25.0 size= 4948kB time=00:03:55.70 bitrate= 172.0kbits/s dup=0 drop=4800 frame= 2444 fps= 22 q=25.0 size= 4993kB time=00:03:59.50 bitrate= 170.8kbits/s dup=0 drop=4876 frame= 2498 fps= 22 q=25.0 size= 5069kB time=00:04:04.90 bitrate= 169.6kbits/s dup=0 drop=4984 frame= 2567 fps= 23 q=25.0 size= 5156kB time=00:04:11.80 bitrate= 167.7kbits/s dup=0 drop=5122 frame= 2621 fps= 23 q=25.0 size= 5266kB time=00:04:17.20 bitrate= 167.7kbits/s dup=0 drop=5229 frame= 2679 fps= 23 q=25.0 size= 5380kB time=00:04:23.00 bitrate= 167.6kbits/s dup=0 drop=5347 frame= 2736 fps= 24 q=25.0 size= 5498kB time=00:04:28.70 bitrate= 167.6kbits/s dup=0 drop=5459 frame= 2812 fps= 24 q=25.0 size= 5600kB time=00:04:36.30 bitrate= 166.0kbits/s dup=0 drop=5611 frame= 2859 fps= 25 q=25.0 size= 5755kB time=00:04:41.00 bitrate= 167.8kbits/s dup=0 drop=5705 frame= 2907 fps= 25 q=25.0 size= 5890kB time=00:04:45.80 bitrate= 168.8kbits/s dup=0 drop=5800 frame= 2910 fps= 24 q=25.0 size= 5898kB time=00:04:46.10 bitrate= 168.9kbits/s dup=0 drop=5806 frame= 2963 fps= 24 q=25.0 size= 6033kB time=00:04:51.40 bitrate= 169.6kbits/s dup=0 drop=5912 frame= 3014 fps= 24 q=25.0 size= 6158kB time=00:04:56.50 bitrate= 170.2kbits/s dup=0 drop=6014 frame= 3066 fps= 25 q=25.0 size= 6291kB time=00:05:01.70 bitrate= 170.8kbits/s dup=0 drop=6118 frame= 3115 fps= 25 q=25.0 size= 6380kB time=00:05:06.60 bitrate= 170.5kbits/s dup=0 drop=6218 frame= 3170 fps= 25 q=25.0 size= 6515kB time=00:05:12.10 bitrate= 171.0kbits/s dup=0 drop=6326 frame= 3210 fps= 25 q=25.0 size= 6624kB time=00:05:16.10 bitrate= 171.7kbits/s dup=0 drop=6406 frame= 3254 fps= 25 q=25.0 size= 6768kB time=00:05:20.50 bitrate= 173.0kbits/s dup=0 drop=6493 frame= 3303 fps= 25 q=25.0 size= 6921kB time=00:05:25.40 bitrate= 174.2kbits/s dup=0 drop=6591 frame= 3353 fps= 25 q=25.0 size= 7069kB time=00:05:30.40 bitrate= 175.3kbits/s dup=0 drop=6691 frame= 3403 fps= 26 q=25.0 size= 7195kB time=00:05:35.40 bitrate= 175.7kbits/s dup=0 drop=6791 frame= 3454 fps= 26 q=25.0 size= 7348kB time=00:05:40.50 bitrate= 176.8kbits/s dup=0 drop=6893 frame= 3499 fps= 26 q=25.0 size= 7520kB time=00:05:45.00 bitrate= 178.6kbits/s dup=0 drop=6983 frame= 3512 fps= 25 q=25.0 size= 7561kB time=00:05:46.30 bitrate= 178.9kbits/s dup=0 drop=7009 frame= 3570 fps= 26 q=25.0 size= 7700kB time=00:05:52.10 bitrate= 179.1kbits/s dup=0 drop=7125 frame= 3626 fps= 26 q=25.0 size= 7834kB time=00:05:57.70 bitrate= 179.4kbits/s dup=0 drop=7236 frame= 3683 fps= 26 q=25.0 size= 7965kB time=00:06:03.40 bitrate= 179.5kbits/s dup=0 drop=7350 frame= 3742 fps= 27 q=25.0 size= 8114kB time=00:06:09.30 bitrate= 180.0kbits/s dup=0 drop=7468 frame= 3805 fps= 27 q=25.0 size= 8246kB time=00:06:15.60 bitrate= 179.9kbits/s dup=0 drop=7594 frame= 3862 fps= 27 q=25.0 size= 8400kB time=00:06:21.30 bitrate= 180.5kbits/s dup=0 drop=7708 frame= 3917 fps= 28 q=25.0 size= 8548kB time=00:06:26.80 bitrate= 181.0kbits/s dup=0 drop=7819 frame= 3974 fps= 27 q=25.0 size= 8690kB time=00:06:32.50 bitrate= 181.4kbits/s dup=0 drop=7931 frame= 4053 fps= 27 q=25.0 size= 8783kB time=00:06:40.40 bitrate= 179.7kbits/s dup=0 drop=8089 frame= 4098 fps= 27 q=25.0 size= 8883kB time=00:06:44.90 bitrate= 179.7kbits/s dup=0 drop=8179 frame= 4101 fps= 27 q=25.0 size= 8891kB time=00:06:45.20 bitrate= 179.7kbits/s dup=0 drop=8185 frame= 4157 fps= 27 q=25.0 size= 9018kB time=00:06:50.80 bitrate= 179.8kbits/s dup=0 drop=8297 frame= 4217 fps= 27 q=25.0 size= 9162kB time=00:06:56.80 bitrate= 180.1kbits/s dup=0 drop=8417 frame= 4279 fps= 27 q=25.0 size= 9295kB time=00:07:03.00 bitrate= 180.0kbits/s dup=0 drop=8540 frame= 4332 fps= 28 q=25.0 size= 9414kB time=00:07:08.30 bitrate= 180.1kbits/s dup=0 drop=8646 frame= 4354 fps= 27 q=25.0 size= 9468kB time=00:07:10.50 bitrate= 180.2kbits/s dup=0 drop=8690 frame= 4357 fps= 26 q=25.0 size= 9475kB time=00:07:10.80 bitrate= 180.2kbits/s dup=0 drop=8696 frame= 4397 fps= 26 q=25.0 size= 9583kB time=00:07:14.80 bitrate= 180.5kbits/s dup=0 drop=8776 frame= 4448 fps= 26 q=25.0 size= 9727kB time=00:07:19.90 bitrate= 181.1kbits/s dup=0 drop=8878 frame= 4498 fps= 27 q=25.0 size= 9870kB time=00:07:24.90 bitrate= 181.7kbits/s dup=0 drop=8978 frame= 4547 fps= 26 q=25.0 size= 10001kB time=00:07:29.80 bitrate= 182.1kbits/s dup=0 drop=9076 frame= 4568 fps= 26 q=25.0 size= 10061kB time=00:07:31.90 bitrate= 182.4kbits/s dup=0 drop=9118 frame= 4591 fps= 26 q=25.0 size= 10125kB time=00:07:34.20 bitrate= 182.6kbits/s dup=0 drop=9163 frame= 4618 fps= 25 q=25.0 size= 10191kB time=00:07:36.90 bitrate= 182.7kbits/s dup=0 drop=9217 frame= 4652 fps= 24 q=25.0 size= 10270kB time=00:07:40.30 bitrate= 182.8kbits/s dup=0 drop=9285 frame= 4705 fps= 24 q=25.0 size= 10448kB time=00:07:45.60 bitrate= 183.8kbits/s dup=0 drop=9391 frame= 4756 fps= 24 q=25.0 size= 10556kB time=00:07:50.70 bitrate= 183.7kbits/s dup=0 drop=9493 frame= 4789 fps= 24 q=25.0 size= 10641kB time=00:07:54.00 bitrate= 183.9kbits/s dup=0 drop=9559 frame= 4813 fps= 24 q=25.0 size= 10699kB time=00:07:56.40 bitrate= 184.0kbits/s dup=0 drop=9607 frame= 4862 fps= 25 q=25.0 size= 10851kB time=00:08:01.30 bitrate= 184.7kbits/s dup=0 drop=9705 frame= 4908 fps= 25 q=25.0 size= 10961kB time=00:08:05.90 bitrate= 184.8kbits/s dup=0 drop=9796 frame= 4937 fps= 25 q=25.0 size= 11042kB time=00:08:08.80 bitrate= 185.1kbits/s dup=0 drop=9854 frame= 4964 fps= 24 q=25.0 size= 11111kB time=00:08:11.50 bitrate= 185.2kbits/s dup=0 drop=9908 frame= 5021 fps= 24 q=25.0 size= 11247kB time=00:08:17.20 bitrate= 185.3kbits/s dup=0 drop=10022 frame= 5083 fps= 25 q=25.0 size= 11375kB time=00:08:23.40 bitrate= 185.1kbits/s dup=0 drop=10146 frame= 5141 fps= 25 q=25.0 size= 11481kB time=00:08:29.20 bitrate= 184.7kbits/s dup=0 drop=10262 frame= 5198 fps= 25 q=25.0 size= 11611kB time=00:08:34.90 bitrate= 184.7kbits/s dup=0 drop=10376 frame= 5231 fps= 24 q=25.0 size= 11688kB time=00:08:38.20 bitrate= 184.8kbits/s dup=0 drop=10442 frame= 5287 fps= 25 q=25.0 size= 11768kB time=00:08:43.80 bitrate= 184.1kbits/s dup=0 drop=10553 frame= 5296 fps= 24 q=25.0 size= 11779kB time=00:08:44.70 bitrate= 183.9kbits/s dup=0 drop=10571 frame= 5372 fps= 24 q=25.0 size= 11866kB time=00:08:52.30 bitrate= 182.6kbits/s dup=0 drop=10723 frame= 5428 fps= 25 q=25.0 size= 12008kB time=00:08:57.90 bitrate= 182.9kbits/s dup=0 drop=10835 frame= 5471 fps= 24 q=25.0 size= 12112kB time=00:09:02.20 bitrate= 183.0kbits/s dup=0 drop=10921 frame= 5520 fps= 24 q=25.0 size= 12240kB time=00:09:07.10 bitrate= 183.3kbits/s dup=0 drop=11020 frame= 5577 fps= 24 q=22.0 size= 12371kB time=00:09:12.80 bitrate= 183.3kbits/s dup=0 drop=11132 frame= 5633 fps= 25 q=25.0 size= 12512kB time=00:09:18.40 bitrate= 183.6kbits/s dup=0 drop=11244 frame= 5693 fps= 25 q=25.0 size= 12637kB time=00:09:24.40 bitrate= 183.4kbits/s dup=0 drop=11364 frame= 5750 fps= 25 q=25.0 size= 12765kB time=00:09:30.10 bitrate= 183.4kbits/s dup=0 drop=11478 frame= 5805 fps= 25 q=25.0 size= 12831kB time=00:09:35.60 bitrate= 182.6kbits/s dup=0 drop=11588 frame= 5863 fps= 25 q=25.0 size= 12895kB time=00:09:41.40 bitrate= 181.7kbits/s dup=0 drop=11704 frame= 5896 fps= 25 q=25.0 size= 12984kB time=00:09:44.70 bitrate= 181.9kbits/s dup=0 drop=11770 frame= 5906 fps= 24 q=25.0 size= 13003kB time=00:09:45.70 bitrate= 181.9kbits/s dup=0 drop=11790 frame= 5957 fps= 24 q=25.0 size= 13110kB time=00:09:50.80 bitrate= 181.8kbits/s dup=0 drop=11891 frame= 6012 fps= 25 q=25.0 size= 13271kB time=00:09:56.30 bitrate= 182.3kbits/s dup=0 drop=12001 frame= 6058 fps= 24 q=25.0 size= 13387kB time=00:10:00.90 bitrate= 182.5kbits/s dup=0 drop=12093 frame= 6068 fps= 24 q=25.0 size= 13415kB time=00:10:01.90 bitrate= 182.6kbits/s dup=0 drop=12113 frame= 6127 fps= 24 q=25.0 size= 13571kB time=00:10:07.80 bitrate= 182.9kbits/s dup=0 drop=12231 frame= 6175 fps= 24 q=25.0 size= 13748kB time=00:10:12.60 bitrate= 183.8kbits/s dup=0 drop=12329 frame= 6209 fps= 23 q=25.0 size= 13874kB time=00:10:16.00 bitrate= 184.5kbits/s dup=0 drop=12395 frame= 6215 fps= 23 q=25.0 size= 13890kB time=00:10:16.60 bitrate= 184.5kbits/s dup=0 drop=12407 frame= 6233 fps= 22 q=25.0 size= 13947kB time=00:10:18.40 bitrate= 184.8kbits/s dup=0 drop=12443 frame= 6282 fps= 23 q=25.0 size= 14117kB time=00:10:23.30 bitrate= 185.5kbits/s dup=0 drop=12540 frame= 6290 fps= 22 q=25.0 size= 14138kB time=00:10:24.10 bitrate= 185.6kbits/s dup=0 drop=12556 frame= 6332 fps= 22 q=25.0 size= 14234kB time=00:10:28.30 bitrate= 185.6kbits/s dup=0 drop=12641 frame= 6340 fps= 22 q=25.0 size= 14246kB time=00:10:29.10 bitrate= 185.5kbits/s dup=0 drop=12656 frame= 6371 fps= 22 q=25.0 size= 14307kB time=00:10:32.20 bitrate= 185.4kbits/s dup=0 drop=12718 frame= 6389 fps= 22 q=25.0 size= 14351kB time=00:10:34.00 bitrate= 185.4kbits/s dup=0 drop=12754 frame= 6418 fps= 22 q=25.0 size= 14424kB time=00:10:36.90 bitrate= 185.5kbits/s dup=0 drop=12812 frame= 6452 fps= 22 q=25.0 size= 14476kB time=00:10:40.30 bitrate= 185.2kbits/s dup=0 drop=12880 frame= 6475 fps= 22 q=25.0 size= 14502kB time=00:10:42.60 bitrate= 184.9kbits/s dup=0 drop=12926 frame= 6492 fps= 21 q=25.0 size= 14534kB time=00:10:44.30 bitrate= 184.8kbits/s dup=0 drop=12960 frame= 6515 fps= 21 q=25.0 size= 14579kB time=00:10:46.60 bitrate= 184.7kbits/s dup=0 drop=13006 frame= 6562 fps= 21 q=25.0 size= 14613kB time=00:10:51.30 bitrate= 183.8kbits/s dup=0 drop=13100 frame= 6587 fps= 21 q=25.0 size= 14682kB time=00:10:53.80 bitrate= 184.0kbits/s dup=0 drop=13149 frame= 6608 fps= 21 q=25.0 size= 14734kB time=00:10:55.90 bitrate= 184.0kbits/s dup=0 drop=13191 frame= 6631 fps= 21 q=25.0 size= 14788kB time=00:10:58.20 bitrate= 184.0kbits/s dup=0 drop=13237 frame= 6658 fps= 21 q=25.0 size= 14842kB time=00:11:00.90 bitrate= 184.0kbits/s dup=0 drop=13291 frame= 6670 fps= 21 q=25.0 size= 14874kB time=00:11:02.10 bitrate= 184.0kbits/s dup=0 drop=13315 frame= 6675 fps= 20 q=25.0 size= 14887kB time=00:11:02.60 bitrate= 184.1kbits/s dup=0 drop=13325 frame= 6696 fps= 20 q=25.0 size= 14930kB time=00:11:04.70 bitrate= 184.0kbits/s dup=0 drop=13367 frame= 6702 fps= 20 q=25.0 size= 14947kB time=00:11:05.30 bitrate= 184.0kbits/s dup=0 drop=13379 frame= 6754 fps= 20 q=25.0 size= 15074kB time=00:11:10.50 bitrate= 184.2kbits/s dup=0 drop=13483 frame= 6797 fps= 20 q=25.0 size= 15184kB time=00:11:14.80 bitrate= 184.3kbits/s dup=0 drop=13569 frame= 6845 fps= 20 q=25.0 size= 15290kB time=00:11:19.60 bitrate= 184.3kbits/s dup=0 drop=13665 frame= 6877 fps= 20 q=25.0 size= 15379kB time=00:11:22.80 bitrate= 184.5kbits/s dup=0 drop=13729 frame= 6905 fps= 20 q=25.0 size= 15462kB time=00:11:25.60 bitrate= 184.7kbits/s dup=0 drop=13785 frame= 6930 fps= 20 q=25.0 size= 15528kB time=00:11:28.10 bitrate= 184.9kbits/s dup=0 drop=13834 frame= 6936 fps= 20 q=25.0 size= 15544kB time=00:11:28.70 bitrate= 184.9kbits/s dup=0 drop=13846 frame= 6975 fps= 20 q=25.0 size= 15638kB time=00:11:32.60 bitrate= 185.0kbits/s dup=0 drop=13924 frame= 7027 fps= 20 q=25.0 size= 15771kB time=00:11:37.80 bitrate= 185.1kbits/s dup=0 drop=14030 frame= 7097 fps= 20 q=25.0 size= 15926kB time=00:11:44.80 bitrate= 185.1kbits/s dup=0 drop=14168 frame= 7125 fps= 20 q=25.0 size= 15947kB time=00:11:47.60 bitrate= 184.6kbits/s dup=0 drop=14224 frame= 7143 fps= 19 q=25.0 size= 15962kB time=00:11:49.40 bitrate= 184.3kbits/s dup=0 drop=14260 frame= 7200 fps= 19 q=25.0 size= 16002kB time=00:11:55.10 bitrate= 183.3kbits/s dup=0 drop=14374 frame= 7243 fps= 19 q=25.0 size= 16034kB time=00:11:59.40 bitrate= 182.6kbits/s dup=0 drop=14459 frame= 7324 fps= 19 q=25.0 size= 16116kB time=00:12:07.50 bitrate= 181.5kbits/s dup=0 drop=14621 frame= 7336 fps= 19 q=-1.0 Lsize= 16161kB time=00:12:13.40 bitrate= 180.5kbits/s dup=0 drop=14645 video:15992kB audio:0kB global headers:0kB muxing overhead 1.053730% frame I:39 Avg QP:16.75 size: 7893 [libx264 @ 000000000079ADA0] frame P:4389 Avg QP:21.24 size: 3127 [libx264 @ 000000000079ADA0] frame B:2908 Avg QP:22.97 size: 806 [libx264 @ 000000000079ADA0] consecutive B-frames: 30.8% 45.7% 10.1% 13.4% [libx264 @ 000000000079ADA0] mb I I16..4: 27.8% 53.6% 18.5% [libx264 @ 000000000079ADA0] mb P I16..4: 7.2% 8.8% 1.8% P16..4: 36.6% 21.2% 11.7% 0.0% 0.0% skip:12.7% [libx264 @ 000000000079ADA0] mb B I16..4: 0.7% 0.7% 0.1% B16..8: 32.8% 8.7% 2.0% direct: 5.2% skip:49.8% L0:37.9% L1:41.9% BI:20.3% [libx264 @ 000000000079ADA0] 8x8 transform intra:49.5% inter:69.2% [libx264 @ 000000000079ADA0] coded y,uvDC,uvAC intra: 42.7% 61.9% 21.5% inter: 28.0% 31.2% 2.0% [libx264 @ 000000000079ADA0] i16 v,h,dc,p: 38% 28% 8% 27% [libx264 @ 000000000079ADA0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 18% 33% 3% 4% 6% 4% 5% 5% [libx264 @ 000000000079ADA0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 24% 25% 14% 4% 7% 8% 6% 6% 5% [libx264 @ 000000000079ADA0] i8c dc,h,v,p: 46% 26% 23% 5% [libx264 @ 000000000079ADA0] Weighted P-Frames: Y:4.7% UV:2.8% [libx264 @ 000000000079ADA0] ref P L0: 62.5% 16.4% 14.0% 7.0% 0.1% [libx264 @ 000000000079ADA0] ref B L0: 83.7% 14.6% 1.7% [libx264 @ 000000000079ADA0] ref B L1: 97.3% 2.7% [libx264 @ 000000000079ADA0] kb/s:178.58 From andrey.krieger.utkin at gmail.com Thu Mar 29 18:57:05 2012 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Thu, 29 Mar 2012 19:57:05 +0300 Subject: [FFmpeg-user] Obtaining SDP information from ffmpeg command line In-Reply-To: References: Message-ID: 2012/3/29 Mobility Lab : > Does anybody know why I am not getting the SDP information, or how to > get ffmpeg to output the information? Add option -loglevel verbose (According to rtspdec.c:136 in git master HEAD revision) -- Andrey Utkin From funkyirish at gmail.com Thu Mar 29 23:55:07 2012 From: funkyirish at gmail.com (Josh long) Date: Thu, 29 Mar 2012 16:55:07 -0500 Subject: [FFmpeg-user] bgra to yuv In-Reply-To: References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: On Thu, Mar 29, 2012 at 1:21 AM, Carl Eugen Hoyos wrote: > Josh long gmail.com> writes: > > > When converting bgra to rgb24 ffmpeg messes up the colors. > > This sounds like a very important bug, please provide a > command line and complete, uncut console output so I can > reproduce the issue. > > Thank you, Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > For my 1920x1080 30fps file the following produces an rgb24 file with messed up colors: ffmpeg -i 0_10_sec.avi -vcodec rawvideo -pix_fmt rgb24 0_10_secrgb..avi ffmpeg version 0.9.1, Copyright (c) 2000-2012 the FFmpeg developers built on Mar 16 2012 02:44:10 with gcc 4.6.1 configuration: --enable-libschroedinger libavutil 51. 32. 0 / 51. 32. 0 libavcodec 53. 42. 4 / 53. 42. 4 libavformat 53. 24. 2 / 53. 24. 2 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 53. 0 / 2. 53. 0 libswscale 2. 1. 0 / 2. 1. 0 Input #0, avi, from '0_10_sec.avi': Duration: 00:00:10.00, start: 0.000000, bitrate: 1990665 kb/s Stream #0:0: Video: rawvideo, bgra, 1920x1080, 30 tbr, 30 tbn, 30 tbc [buffer @ 0x1e08c80] w:1920 h:1080 pixfmt:bgra tb:1/1000000 sar:0/1 sws_param: [buffersink @ 0x1e01420] auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'out' [scale @ 0x1e01c20] w:1920 h:1080 fmt:bgra -> w:1920 h:1080 fmt:rgb24 flags:0x4 Output #0, avi, to '0_10_secrgb..avi': Metadata: ISFT : Lavf53.24.2 Stream #0:0: Video: rawvideo, rgb24, 1920x1080, q=2-31, 200 kb/s, 30 tbn, 30 tbc Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> rawvideo) Press [q] to stop, [?] for help frame= 9 fps= 0 q=0.0 size= 54681kB time=00:00:00.30 bitrate=1493145.2kbitframe= 17 fps= 16 q=0.0 size= 103281kB time=00:00:00.56 bitrate=1493073.1kbitframe= 22 fps= 14 q=0.0 size= 133656kB time=00:00:00.73 bitrate=1493056.5kbitframe= 27 fps= 13 q=0.0 size= 164031kB time=00:00:00.90 bitrate=1493044.3kbitframe= 31 fps= 12 q=0.0 size= 188331kB time=00:00:01.03 bitrate=1493038.3kbitframe= 34 fps= 10 q=0.0 size= 206556kB time=00:00:01.13 bitrate=1493034.4kbitframe= 38 fps= 10 q=0.0 size= 230856kB time=00:00:01.26 bitrate=1493029.4kbitframe= 40 fps= 9 q=0.0 size= 243006kB time=00:00:01.33 bitrate=1493028.3kbitframe= 43 fps= 8 q=0.0 size= 261231kB time=00:00:01.43 bitrate=1493025.9kbitframe= 46 fps= 8 q=0.0 size= 279456kB time=00:00:01.53 bitrate=1493023.8kbitframe= 49 fps= 8 q=0.0 size= 297681kB time=00:00:01.63 bitrate=1493022.0kbitframe= 53 fps= 8 q=0.0 size= 321981kB time=00:00:01.76 bitrate=1493019.3kbitframe= 56 fps= 8 q=0.0 size= 340206kB time=00:00:01.86 bitrate=1493018.0kbitframe= 59 fps= 7 q=0.0 size= 358431kB time=00:00:01.96 bitrate=1493016.7kbitframe= 63 fps= 7 q=0.0 size= 382731kB time=00:00:02.10 bitrate=1493015.5kbitframe= 66 fps= 7 q=0.0 size= 400956kB time=00:00:02.20 bitrate=1493014.5kbitframe= 69 fps= 7 q=0.0 size= 419181kB time=00:00:02.30 bitrate=1493013.6kbitframe= 72 fps= 7 q=0.0 size= 437406kB time=00:00:02.40 bitrate=1493012.8kbitframe= 75 fps= 7 q=0.0 size= 455631kB time=00:00:02.50 bitrate=1493012.1kbitframe= 78 fps= 7 q=0.0 size= 473856kB time=00:00:02.60 bitrate=1493011.4kbitframe= 81 fps= 7 q=0.0 size= 492081kB time=00:00:02.70 bitrate=1493010.7kbitframe= 85 fps= 7 q=0.0 size= 516381kB time=00:00:02.83 bitrate=1493010.1kbitframe= 88 fps= 7 q=0.0 size= 534606kB time=00:00:02.93 bitrate=1493009.6kbitframe= 91 fps= 6 q=0.0 size= 552831kB time=00:00:03.03 bitrate=1493009.0kbitframe= 94 fps= 6 q=0.0 size= 571056kB time=00:00:03.13 bitrate=1493008.6kbitframe= 96 fps= 6 q=0.0 size= 583206kB time=00:00:03.20 bitrate=1493008.1kbitframe= 100 fps= 6 q=0.0 size= 607506kB time=00:00:03.33 bitrate=1493007.7kbitframe= 103 fps= 6 q=0.0 size= 625731kB time=00:00:03.43 bitrate=1493007.3kbitframe= 106 fps= 6 q=0.0 size= 643956kB time=00:00:03.53 bitrate=1493006.9kbitframe= 108 fps= 6 q=0.0 size= 656106kB time=00:00:03.60 bitrate=1493006.5kbitframe= 112 fps= 6 q=0.0 size= 680406kB time=00:00:03.73 bitrate=1493006.2kbitframe= 115 fps= 6 q=0.0 size= 698631kB time=00:00:03.83 bitrate=1493005.9kbitframe= 118 fps= 6 q=0.0 size= 716856kB time=00:00:03.93 bitrate=1493005.6kbitframe= 121 fps= 6 q=0.0 size= 735081kB time=00:00:04.03 bitrate=1493005.3kbitframe= 124 fps= 6 q=0.0 size= 753307kB time=00:00:04.13 bitrate=1493005.0kbitframe= 127 fps= 6 q=0.0 size= 771532kB time=00:00:04.23 bitrate=1493004.8kbitframe= 130 fps= 6 q=0.0 size= 789757kB time=00:00:04.33 bitrate=1493004.5kbitframe= 133 fps= 6 q=0.0 size= 807982kB time=00:00:04.43 bitrate=1493004.3kbitframe= 136 fps= 6 q=0.0 size= 826207kB time=00:00:04.53 bitrate=1493004.0kbitframe= 139 fps= 6 q=0.0 size= 844432kB time=00:00:04.63 bitrate=1493003.8kbitframe= 142 fps= 6 q=0.0 size= 862657kB time=00:00:04.73 bitrate=1493003.6kbitframe= 144 fps= 6 q=0.0 size= 874807kB time=00:00:04.80 bitrate=1493003.4kbitframe= 147 fps= 6 q=0.0 size= 893032kB time=00:00:04.90 bitrate=1493003.2kbitframe= 149 fps= 6 q=0.0 size= 905182kB time=00:00:04.96 bitrate=1493003.0kbitframe= 151 fps= 6 q=0.0 size= 917332kB time=00:00:05.03 bitrate=1493003.0kbitframe= 155 fps= 6 q=0.0 size= 941632kB time=00:00:05.16 bitrate=1493002.6kbitframe= 158 fps= 6 q=0.0 size= 959857kB time=00:00:05.26 bitrate=1493002.4kbitframe= 160 fps= 6 q=0.0 size= 972007kB time=00:00:05.33 bitrate=1493002.5kbitframe= 164 fps= 6 q=0.0 size= 996307kB time=00:00:05.46 bitrate=1493002.1kbitframe= 168 fps= 6 q=0.0 size= 1020607kB time=00:00:05.60 bitrate=1493002.0kbitframe= 171 fps= 6 q=0.0 size= 1038832kB time=00:00:05.70 bitrate=1493001.9kbitframe= 174 fps= 6 q=0.0 size= 1057061kB time=00:00:05.80 bitrate=1493007.6kbitframe= 177 fps= 6 q=0.0 size= 1075286kB time=00:00:05.90 bitrate=1493007.3kbitframe= 180 fps= 6 q=0.0 size= 1093511kB time=00:00:06.00 bitrate=1493007.1kbitframe= 184 fps= 6 q=0.0 size= 1117811kB time=00:00:06.13 bitrate=1493006.9kbitframe= 188 fps= 6 q=0.0 size= 1142111kB time=00:00:06.26 bitrate=1493006.5kbitframe= 191 fps= 6 q=0.0 size= 1160336kB time=00:00:06.36 bitrate=1493006.3kbitframe= 193 fps= 6 q=0.0 size= 1172486kB time=00:00:06.43 bitrate=1493006.3kbitframe= 196 fps= 6 q=0.0 size= 1190711kB time=00:00:06.53 bitrate=1493006.1kbitframe= 199 fps= 6 q=0.0 size= 1208936kB time=00:00:06.63 bitrate=1493005.9kbitframe= 202 fps= 6 q=0.0 size= 1227161kB time=00:00:06.73 bitrate=1493005.7kbitframe= 205 fps= 6 q=0.0 size= 1245386kB time=00:00:06.83 bitrate=1493005.6kbitframe= 208 fps= 5 q=0.0 size= 1263611kB time=00:00:06.93 bitrate=1493005.4kbitframe= 211 fps= 5 q=0.0 size= 1281836kB time=00:00:07.03 bitrate=1493005.2kbitframe= 214 fps= 5 q=0.0 size= 1300061kB time=00:00:07.13 bitrate=1493005.1kbitframe= 217 fps= 5 q=0.0 size= 1318286kB time=00:00:07.23 bitrate=1493004.9kbitframe= 220 fps= 5 q=0.0 size= 1336511kB time=00:00:07.33 bitrate=1493004.8kbitframe= 223 fps= 5 q=0.0 size= 1354736kB time=00:00:07.43 bitrate=1493004.6kbitframe= 226 fps= 5 q=0.0 size= 1372961kB time=00:00:07.53 bitrate=1493004.5kbitframe= 228 fps= 5 q=0.0 size= 1385111kB time=00:00:07.60 bitrate=1493004.3kbitframe= 232 fps= 5 q=0.0 size= 1409411kB time=00:00:07.73 bitrate=1493004.2kbitframe= 235 fps= 5 q=0.0 size= 1427636kB time=00:00:07.83 bitrate=1493004.1kbitframe= 238 fps= 5 q=0.0 size= 1445862kB time=00:00:07.93 bitrate=1493004.0kbitframe= 241 fps= 5 q=0.0 size= 1464087kB time=00:00:08.03 bitrate=1493003.8kbitframe= 244 fps= 5 q=0.0 size= 1482312kB time=00:00:08.13 bitrate=1493003.7kbitframe= 246 fps= 5 q=0.0 size= 1494462kB time=00:00:08.20 bitrate=1493003.6kbitframe= 249 fps= 5 q=0.0 size= 1512687kB time=00:00:08.30 bitrate=1493003.5kbitframe= 252 fps= 5 q=0.0 size= 1530912kB time=00:00:08.40 bitrate=1493003.3kbitframe= 255 fps= 5 q=0.0 size= 1549137kB time=00:00:08.50 bitrate=1493003.2kbitframe= 258 fps= 5 q=0.0 size= 1567362kB time=00:00:08.60 bitrate=1493003.1kbitframe= 261 fps= 5 q=0.0 size= 1585587kB time=00:00:08.70 bitrate=1493003.0kbitframe= 264 fps= 5 q=0.0 size= 1603812kB time=00:00:08.80 bitrate=1493002.9kbitframe= 267 fps= 5 q=0.0 size= 1622037kB time=00:00:08.90 bitrate=1493002.8kbitframe= 270 fps= 5 q=0.0 size= 1640262kB time=00:00:09.00 bitrate=1493002.7kbitframe= 273 fps= 5 q=0.0 size= 1658487kB time=00:00:09.10 bitrate=1493002.6kbitframe= 275 fps= 5 q=0.0 size= 1670637kB time=00:00:09.16 bitrate=1493002.5kbitframe= 278 fps= 5 q=0.0 size= 1688862kB time=00:00:09.26 bitrate=1493002.4kbitframe= 281 fps= 5 q=0.0 size= 1707087kB time=00:00:09.36 bitrate=1493002.3kbitframe= 284 fps= 5 q=0.0 size= 1725312kB time=00:00:09.46 bitrate=1493002.2kbitframe= 287 fps= 5 q=0.0 size= 1743537kB time=00:00:09.56 bitrate=1493002.1kbitframe= 289 fps= 5 q=0.0 size= 1755687kB time=00:00:09.63 bitrate=1493002.2kbitframe= 293 fps= 5 q=0.0 size= 1779987kB time=00:00:09.76 bitrate=1493002.0kbitframe= 296 fps= 5 q=0.0 size= 1798212kB time=00:00:09.86 bitrate=1493001.9kbitframe= 299 fps= 5 q=0.0 size= 1816437kB time=00:00:09.96 bitrate=1493001.8kbitframe= 300 fps= 5 q=0.0 Lsize= 1822513kB time=00:00:10.00 bitrate=1493002.7kbits/s video:1822500kB audio:0kB global headers:0kB muxing overhead 0.000715% these commands in order produce an rgb24 file with the correct colors: ffmpeg -i 0bgra0.avi -vcodec rawvideo -pix_fmt rgba -q 0 0rgba0.avi ffmpeg -i 0rgba0.avi -vcodec rawvideo -pix_fmt rgb24 -q 0 0240.avi also this command morphs the video into a nine segment copy of the original, with some wierd color changes in each one. ffmpeg -i 0240.avi -vcodec rawvideo -pix_fmt yuv444p -q 0 04440.avi From wtfux.dev at googlemail.com Fri Mar 30 05:40:44 2012 From: wtfux.dev at googlemail.com (wtfux) Date: Fri, 30 Mar 2012 05:40:44 +0200 Subject: [FFmpeg-user] Need help converting videos for on-demand-streaming In-Reply-To: References: Message-ID: Hello. I'm looking for some help to properly convert videos for on-demand-streaming. The input files are in all kind of formats (mainly h264, aac but also divx, mp3 or flac). Some videos contain .ass subtitles that need to be burned into the video. I need an automated way to convert those videos and after searching for a long time I couldn't find anything so I decided to write my own little script using ffmpeg. However I'm a newbie at this video stuff. The output should be 720p and 480p for >=720p video, 480p for >=480p and the original size for everything else. I think I'm going to use mediainfo to get that info before conversion. The videos are computer-animated and contain many steady scenes. The output file should have an acceptable file size / bit rate for streaming but should not lose (much) in quality. This is what I got so far: A video in 720p x264/aac without subtitles: ffmpeg -i input.mp4 -codec:a copy -codec:v libx264 -preset slow -tune animation -level 3.1 -crf 21 -threads 8 -f mp4 720p.mp4 => encoding with 15 fps ffmpeg -i input.mp4 -codec:a libfaac -b:a 128k -codec:v libx264 -preset slow -tune animation -level 3.1 -crf 21 -vf "scale=854:480" -threads 8 -f mp4 480p.mp4 => encoding with 35 fps Input file: 299MB (mediainfo: http://pastebin.com/WdD2v7Qs ) Output file 720p: 251MB (mediainfo: http://pastebin.com/AJnJQaU1 ) Output file 480p: 147MB (mediainfo: http://pastebin.com/hsuV55aN ) Video in 720p x264/aac with .ass subtitles: ffmpeg -i input.mkv -codec:a copy -codec:v libx264 -preset slow -tune animation -level 3.1 -crf 21 -vf "ass=a.ass" -threads 8 -f mp4 720p.mp4 => forgot to notice fps but slower than the video itself ffmpeg -i input.mkv -codec:a libfaac -b:a 128k -codec:v libx264 -preset slow -tune animation -level 3.1 -crf 21 -vf "scale=854:480,ass=a.ass" -threads 8 -f mp4 480p.mp4 => encoding with 43 fps Input file: 422MB (mediainfo: http://pastebin.com/HCHLdzNS ) Output file 720p: 144 MB (mediainfo: http://pastebin.com/4pyiE1n8 ) Output file 480p: 77.3 MB (mediainfo: http://pastebin.com/vRQicQBS ) Huge difference in file size but I can't see any video quality difference. I have several questions I couldn't find answers for. I hope someone on this mailing list could help out: 1. How does variable bitrate in aac work? Instead of copying the stream and using a fixed bitrate for the 480p videos or videos with other codecs I think it'd be better to use a variable bitrate depending on the source material. How do I do this? 2. Subtitle: Read subtitle and fonts from stream? In order to burn the subtitle I need to extract the subtitle file and fonts from the file (and place the fonts in ~/.fonts). Is it possible to tell ffmpeg to read them from the input file? 3. -tune animation What exactly is this for? Since the videos are animated I think this is a good option? 4. What is a good value for -crf? I tried 20, 21 and 22 and couldn't notice a difference. I want small files so it's possible to watch the videos without buffering but you shouldn't notice (much) difference in video quality. 5. Faster conversion? If I need the conversion faster, should I change -preset slow to -preset medium or something else? Will the file get larger or the quality worse? I noticed that the conversion isn't slower when I run multiple at the same time so I guess ffmpeg isn't using the full CPU. I set "-threads 8" because i have 4 cores (8 in HT) but no difference. 6. Other params for improvement? While googling for ffmpeg examples I found a lot of other parameters. I omitted them since I don't know what I'm doing. I see "-partitions +parti4x4+parti8x8+partp4x4+partp8x8+partb8x8" often. Are there any important parameters I should use to make the video/sound better? 7. I'm not sure if this is related to ffmpeg but I can't seek when streamed over HTTP (I do use qt-faststart after ffmpeg!). Other vids I converted some months ago however work. The only difference I know of is that I used that -partitions parameter from the question above. I know these are a lot of questions but I spend so much time reading lots of blogs and forum topics and couldn't find the right answers. Thank you! From batguano999 at hotmail.com Fri Mar 30 06:41:32 2012 From: batguano999 at hotmail.com (bat guano) Date: Fri, 30 Mar 2012 04:41:32 +0000 Subject: [FFmpeg-user] Need help converting videos for on-demand-streaming In-Reply-To: References: , Message-ID: > I have several questions I couldn't find answers for. I hope someone > on this mailing list could help out: > > 1. How does variable bitrate in aac work? > Instead of copying the stream and using a fixed bitrate for the 480p > videos or videos with other codecs I think it'd be better to use a > variable bitrate depending on the source material. How do I do this? > ******************************************************************************* Quality-related options: ? -q ??? Set default variable bitrate (VBR) quantizer quality in percent. ??? ??? (default: 100, averages at approx. 120 kbps VBR for a normal ??? ??? stereo input file with 16 bit and 44.1 kHz sample rate; max. ??? ??? value 500, min. 10). ******************************************************************************* Information above is excerpt from "faac --long-help". So when using libfaac with FFmpeg.... You would use it like this..... -codec:a libfaac -q:a 100 Choose a quality value from 10 to 500. faac --long-help is here ---> http://pastebin.com/Xgfdc3CS From salmjuh at hotmail.com Fri Mar 30 08:15:02 2012 From: salmjuh at hotmail.com (juha s.) Date: Fri, 30 Mar 2012 06:15:02 +0000 Subject: [FFmpeg-user] YouTube Settings Message-ID: Hi What settings I need to use if I like to produce same quolity videos than youtube are using ? Br, jiii From oussama.stiti at gmail.com Fri Mar 30 08:35:14 2012 From: oussama.stiti at gmail.com (Oussama Stiti) Date: Fri, 30 Mar 2012 15:35:14 +0900 Subject: [FFmpeg-user] Fwd: input rtp stream In-Reply-To: References: Message-ID: ---------- Forwarded message ---------- From: Oussama Stiti Date: 2012/3/29 Subject: input rtp stream To: FFmpeg user questions and RTFMs Hello, I'm broadcasting a video with vlc on rtp://127.0.0.1:5004. When i try the command : ffmpeg -report -loglevel warning -i rtp:// 127.0.0.1:5004 -stats -f null - It gives me the following result: [h264 @ 0x9cbe0a0] non-existing PPS referenced [h264 @ 0x9cbe0a0] non-existing PPS 0 referenced [h264 @ 0x9cbe0a0] decode_slice_header error [h264 @ 0x9cbe0a0] no frame! [h264 @ 0x9cbe0a0] non-existing PPS referenced [h264 @ 0x9cbe0a0] non-existing PPS 0 referenced [h264 @ 0x9cbe0a0] decode_slice_header error [h264 @ 0x9cbe0a0] no frame! [h264 @ 0x9cbe0a0] non-existing PPS referenced [h264 @ 0x9cbe0a0] non-existing PPS 0 referenced [h264 @ 0x9cbe0a0] decode_slice_header error [h264 @ 0x9cbe0a0] no frame! [...] DTS 4519, next:92863 st:0 invalid droping DTS 6609, next:116082 st:0 invalid droping DTS 8698, next:139301 st:0 invalid droping DTS 10788, next:162520 st:0 invalid droping DTS 9039, next:185739 st:0 invalid droping DTS 11128, next:208958 st:0 invalid droping DTS 13558, next:232177 st:0 invalid droping DTS 15648, next:255396 st:0 invalid droping DTS 17737, next:278615 st:0 invalid droping DTS 15988, next:301834 st:0 invalid droping DTS 18078, next:325053 st:0 invalid droping DTS 20167, next:348272 st:0 invalid droping What's wrong ? Regards -- *Oussama Stiti* -- *Oussama Stiti* ?l?ve ing?nieur en t?l?communications ? Sup'com (?cole sup?rieure des communications de Tunis) T?l: +21652363164 E-mail: oussama.stiti at gmail.com From oussama.stiti at gmail.com Fri Mar 30 08:35:45 2012 From: oussama.stiti at gmail.com (Oussama Stiti) Date: Fri, 30 Mar 2012 15:35:45 +0900 Subject: [FFmpeg-user] Fwd: Referenced frames In-Reply-To: References: Message-ID: ---------- Forwarded message ---------- From: Oussama Stiti Date: 2012/3/29 Subject: Referenced frames To: FFmpeg user questions and RTFMs Hello, I want to detect in a h264 video, which frames are related ( referenced) to which frames. For example : Frame 62 : I Frame. Frame 63 : P Frame [referenced to frame 62] Frame 64 : B Frame [referenced to frame 63 & 65] Is it possible to display such result ? What kind of command line, should i run ? Thank you Regards -- *Oussama Stiti* ?l?ve ing?nieur en t?l?communications ? Sup'com (?cole sup?rieure des communications de Tunis) T?l: +21652363164 E-mail: oussama.stiti at gmail.com -- *Oussama Stiti* ?l?ve ing?nieur en t?l?communications ? Sup'com (?cole sup?rieure des communications de Tunis) T?l: +21652363164 E-mail: oussama.stiti at gmail.com From bostjan.strojan at gmail.com Fri Mar 30 08:48:52 2012 From: bostjan.strojan at gmail.com (=?UTF-8?Q?Bo=C5=A1tjan_Strojan?=) Date: Fri, 30 Mar 2012 08:48:52 +0200 Subject: [FFmpeg-user] Need help converting videos for on-demand-streaming In-Reply-To: References: Message-ID: > 3. -tune animation > What exactly is this for? Since the videos are animated I think this > is a good option? Depends, it is not meant for 3d animation, this is more for old fashioned 2d anim/cartoonish stuff. maybe check http://doom10.org/index.php?topic=253.0 > 4. What is a good value for -crf? > I tried 20, 21 and 22 and couldn't notice a difference. I want small > files so it's possible to watch the videos without buffering but you > shouldn't notice (much) difference in video quality. My usual default is 21, bigger number = worse quality. Also people seems to use higher values (worse quality) for higher resolutions. > 5. Faster conversion? > If I need the conversion faster, should I change -preset slow to > -preset medium or something else? Will the file get larger or the > quality worse? afaik they are not directly comparable, but i could be wrong, my default is -preset medium. > 7. I'm not sure if this is related to ffmpeg but I can't seek when > streamed over HTTP (I do use qt-faststart after ffmpeg!). Other vids I > converted some months ago however work. The only difference I know of > is that I used that -partitions parameter from the question above. how are you streaming? progressive download or real streaming? For progressive download qt-faststart is working for me. b. From wtfux.dev at googlemail.com Fri Mar 30 12:52:16 2012 From: wtfux.dev at googlemail.com (wtfux) Date: Fri, 30 Mar 2012 12:52:16 +0200 Subject: [FFmpeg-user] Need help converting videos for on-demand-streaming In-Reply-To: References: Message-ID: >> 3. -tune animation >> What exactly is this for? Since the videos are animated I think this >> is a good option? > > Depends, it is not meant for 3d animation, this is more for old > fashioned 2d anim/cartoonish stuff. > maybe check http://doom10.org/index.php?topic=253.0 It's actually this kind of video, I just used the wrong term in my post. So it should fit. >> 5. Faster conversion? >> If I need the conversion faster, should I change -preset slow to >> -preset medium or something else? Will the file get larger or the >> quality worse? > > afaik they are not directly comparable, but i could be wrong, my > default is -preset medium. > It looks like they can be used together. With `-preset superfast` it'll encode with ~80fps and output a huge file. I'll try medium and check the difference to slow. >> 7. I'm not sure if this is related to ffmpeg but I can't seek when >> streamed over HTTP (I do use qt-faststart after ffmpeg!). Other vids I >> converted some months ago however work. The only difference I know of >> is that I used that -partitions parameter from the question above. > > how are you streaming? progressive download or real streaming? For > progressive download ?qt-faststart ?is working for me. progressive download. Without qt-faststart the whole file needs to be downloaded before flash player can play it. However seeking / jumping to a part that has not yet been loaded won't work (with videoJS or flowplayer; in vlc it works (from http)). From bostjan.strojan at gmail.com Fri Mar 30 17:32:15 2012 From: bostjan.strojan at gmail.com (=?UTF-8?Q?Bo=C5=A1tjan_Strojan?=) Date: Fri, 30 Mar 2012 17:32:15 +0200 Subject: [FFmpeg-user] Need help converting videos for on-demand-streaming In-Reply-To: References: Message-ID: > However seeking / jumping to a part that has not yet been loaded won't > work (with videoJS or flowplayer; in vlc it works (from http)). What's funny is that it appears that html5 mode playback (in chrome) is also able to jump to arbitrary position. (Just noticed it, but i did upgrade my old karmic box to lucid yesterday (finally), so i wonder if it has something to do with new apache or just chrome got smarter?) Just an observation i can't explain..., b. From de.techno at gmail.com Fri Mar 30 12:18:08 2012 From: de.techno at gmail.com (dE .) Date: Fri, 30 Mar 2012 15:48:08 +0530 Subject: [FFmpeg-user] YouTube Settings In-Reply-To: References: Message-ID: <4F758860.3010403@gmail.com> On 03/30/12 11:45, juha s. wrote: > Hi > > What settings I need to use if I like to produce same quolity videos than youtube are using ? > > > Br, > jiii They use h264. From nichot20 at yahoo.com Fri Mar 30 18:06:28 2012 From: nichot20 at yahoo.com (Tim Nicholson) Date: Fri, 30 Mar 2012 17:06:28 +0100 Subject: [FFmpeg-user] -flags2 +ivlc+non_linear_q issues Message-ID: <4F75DA04.3040200@yahoo.com> Trying to generate some imx50 material using suggested parameters from the web but when I add in the -flags2 options I get an odd error:- ffmpeg -i BRD17568301-16x9-ffv1.mkv -map 0:v -map 0:a -an -vf pad=720:608:0:32:black -tag:v mx5p -c:v mpeg2video -r 25 -pix_fmt yuv422p -minrate 50000k -maxrate 50000k -b:v 50000k -intra -flags +ildct+ilme+low_delay -flags2 +ivlc+non_linear_q -ps 1 -qmin 1 -qmax 3 -top 1 -dc 10 -bufsize 2000000 -rc_init_occupancy 2000000 -rc_buf_aggressivity 0.25 -y -f mxf_d10 ./BRD17568301-16x9-ffv1-d10.mxf ffmpeg version N-39411-g2b7c0c9-by_Tim Copyright (c) 2000-2012 the FFmpeg developers built on Mar 30 2012 09:42:53 with gcc 4.6.2 configuration: --extra-version=by_Tim --enable-static --disable-shared --enable-gpl --enable-nonfree --enable-version3 --prefix=/mnt/msds-store-0/tim/ffmpeg-tux/usr/local --libdir=/mnt/msds-store-0/tim/ffmpeg-tux/usr/local/lib64 --enable-runtime-cpudetect --extra-cflags='-static -I/mnt/msds-store-0/tim/ffmpeg-tux/usr/local/include' --extra-ldflags='-static -L/mnt/msds-store-0/tim/ffmpeg-tux/usr/local/lib64' --progs-suffix=_Mar-30 --enable-libfaac --enable-libx264 --enable-libfreetype --disable-ffplay libavutil 51. 44.100 / 51. 44.100 libavcodec 54. 12.100 / 54. 12.100 libavformat 54. 3.100 / 54. 3.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 66.101 / 2. 66.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 10.100 / 0. 10.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, matroska,webm, from 'BRD17568301-16x9-ffv1.mkv': Metadata: TIMECODE : 00:00:00:00 ENCODER : Lavf54.3.100 Duration: 00:01:15.72, start: 0.000000, bitrate: 106171 kb/s Stream #0:0: Video: ffv1 (FFV1 / 0x31564646), yuv422p10le, 720x576, SAR 1:1 DAR 5:4, 25 fps, 25 tbr, 1k tbn, 1k tbc (default) Stream #0:1: Audio: pcm_s24le, 48000 Hz, 1 channels, s32, 1152 kb/s (default) Stream #0:2: Audio: pcm_s24le, 48000 Hz, 1 channels, s32, 1152 kb/s (default) Stream #0:3: Audio: pcm_s24le, 48000 Hz, 1 channels, s32, 1152 kb/s (default) Stream #0:4: Audio: pcm_s24le, 48000 Hz, 1 channels, s32, 1152 kb/s (default) [buffer @ 0x18424a0] w:720 h:576 pixfmt:yuv422p10le tb:1/1000000 sar:1/1 sws_param:flags=2 [pad @ 0x17c5dc0] auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'Parsed_pad_0' [scale @ 0x1841480] w:720 h:576 fmt:yuv422p10le sar:1/1 -> w:720 h:576 fmt:yuv422p sar:1/1 flags:0x4 [pad @ 0x17c5dc0] w:720 h:576 -> w:720 h:608 x:0 y:32 color:0x000000FF [NULL @ 0x1823f20] [Eval @ 0x7fffd3cdd210] Undefined constant or missing '(' in 'ivlc' [NULL @ 0x1823f20] Unable to parse option value "ivlc+non_linear_q" [NULL @ 0x1823f20] Error setting option flags2 to value +ivlc+non_linear_q. Output #0, mxf_d10, to './BRD17568301-16x9-ffv1-d10.mxf': I have tried using just '-flags2 +non_linear_q issues' and get a very similar error:- Undefined constant or missing '(' in 'non_linear_q' Can anyone see something I have missed, or is this a regression? -- Tim From dashing.meng at gmail.com Fri Mar 30 18:22:23 2012 From: dashing.meng at gmail.com (littlebat) Date: Sat, 31 Mar 2012 00:22:23 +0800 Subject: [FFmpeg-user] YouTube Settings In-Reply-To: <4F758860.3010403@gmail.com> References: <4F758860.3010403@gmail.com> Message-ID: <20120331002223.1f648ed8.dashing.meng@gmail.com> On Fri, 30 Mar 2012 15:48:08 +0530 "dE ." wrote: > On 03/30/12 11:45, juha s. wrote: > > Hi > > > > What settings I need to use if I like to produce same quolity > > videos than youtube are using ? > > > > > > Br, > > jiii > > They use h264. > _______________________________________________ They have 3 formats(webm, flv, mp4) and several resolutions(1028, 720, 480, 360, 240?). You can use "ffmpeg -i filename" or "mediainfo filename" to analyze it's acodec, vcodec, container, fps, bit rate, etc.. Then use the same setting to convert your videos. littlebat From florabbi at gmail.com Fri Mar 30 09:45:59 2012 From: florabbi at gmail.com (flora) Date: Fri, 30 Mar 2012 00:45:59 -0700 (PDT) Subject: [FFmpeg-user] Need help on error concealment Message-ID: <1333093559101-4518323.post@n4.nabble.com> Hi guys, I have a question about the settings of error resilience, which seems should be decided by the parameters :error resilience and error concealment Now the function ff_er_add_slice() always return without doing EC, and I found the statement in the code, but where should i set them? * Error resilience; higher values will detect more errors but may * misdetect some more or less valid parts as errors. * - encoding: unused * - decoding: Set by user. #define FF_ER_CAREFUL 1 #define FF_ER_COMPLIANT 2 #define FF_ER_AGGRESSIVE 3 #define FF_ER_VERY_AGGRESSIVE 4 /** * error concealment flags * - encoding: unused * - decoding: Set by user. */ int error_concealment; #define FF_EC_GUESS_MVS 1 #define FF_EC_DEBLOCK 2 Could anyone please give me some suggestions? Thanks so much! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Need-help-on-error-concealment-tp4518323p4518323.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From wtfux.dev at googlemail.com Fri Mar 30 20:13:20 2012 From: wtfux.dev at googlemail.com (wtfux) Date: Fri, 30 Mar 2012 20:13:20 +0200 Subject: [FFmpeg-user] Need help converting videos for on-demand-streaming In-Reply-To: References: Message-ID: > ******************************************************************************* > Quality-related options: > ? -q ??? Set default variable bitrate (VBR) quantizer quality in percent. > ??? ??? (default: 100, averages at approx. 120 kbps VBR for a normal > ??? ??? stereo input file with 16 bit and 44.1 kHz sample rate; max. > ??? ??? value 500, min. 10). > ******************************************************************************* I tried that but the output is constant bitrate. -codec:a libfaac -q:a 120 Audio ID : 2 Format : AAC Format/Info : Advanced Audio Codec Format profile : LC Codec ID : 40 Duration : 2mn 0s Bit rate mode : Constant Bit rate : 144 Kbps Channel(s) : 2 channels Channel positions : Front: L R Sampling rate : 48.0 KHz Compression mode : Lossy Delay relative to video : -2s 2ms Stream size : 2.05 MiB (8%) -codec:a libfaac -q:a 200 Audio ID : 2 Format : AAC Format/Info : Advanced Audio Codec Format profile : LC Codec ID : 40 Duration : 2mn 0s Bit rate mode : Constant Bit rate : 187 Kbps Channel(s) : 2 channels Channel positions : Front: L R Sampling rate : 48.0 KHz Compression mode : Lossy Delay relative to video : -2s 2ms Stream size : 2.68 MiB (11%) From batguano999 at hotmail.com Fri Mar 30 21:22:12 2012 From: batguano999 at hotmail.com (bat guano) Date: Fri, 30 Mar 2012 19:22:12 +0000 Subject: [FFmpeg-user] Need help converting videos for on-demand-streaming In-Reply-To: References: , , , Message-ID: > > I tried that but the output is constant bitrate. > Hi faac is a variable bit rate encoder. I think that MediaInfo is mistaken - but I can't explain it. >From "faac --long-help" -q ??? Set default variable bitrate (VBR) quantizer quality in percent. ??? ??? (default: 100, averages at approx. 120 kbps VBR for a normal ??? ??? stereo input file with 16 bit and 44.1 kHz sample rate; max. ??? ??? value 500, min. 10). -b ??? Set average bitrate (ABR) to approximately kbps. ??? ??? (max. value 152 kbps/stereo with a 16 kHz cutoff, can be raised ??? ??? with a higher -c setting). From mobilityrulezy3h at gmail.com Fri Mar 30 22:04:05 2012 From: mobilityrulezy3h at gmail.com (Mobility Lab) Date: Fri, 30 Mar 2012 16:04:05 -0400 Subject: [FFmpeg-user] Obtaining SDP information from ffmpeg command line In-Reply-To: References: Message-ID: On Thu, Mar 29, 2012 at 12:57 PM, Andrey Utkin wrote: > 2012/3/29 Mobility Lab : >> Does anybody know why I am not getting the SDP information, or how to >> get ffmpeg to output the information? > > Add option -loglevel verbose > (According to rtspdec.c:136 in git master HEAD revision) > > -- > Andrey Utkin Thank you, that work perfectly. - Andrew From joshdotnet at gmail.com Fri Mar 30 16:24:32 2012 From: joshdotnet at gmail.com (Josh) Date: Fri, 30 Mar 2012 14:24:32 +0000 (UTC) Subject: [FFmpeg-user] .m4a to .flv - image/video fails References: Message-ID: Nancy, did you ever find a workaround for this problem? I am trying to convert .m4a podcast files with embedded images and chapters to .flv so I can more easily display and play them on my website. Any advice would be much appreciated. From mobilityrulezy3h at gmail.com Fri Mar 30 22:38:56 2012 From: mobilityrulezy3h at gmail.com (Mobility Lab) Date: Fri, 30 Mar 2012 16:38:56 -0400 Subject: [FFmpeg-user] ffserver Error when streaming video Message-ID: Hello, So, I am trying to use ffserver to stream video retrieved from RTP via ffmpeg. However, when I test the video with VLC, I never see any video, and the terminal for ffserver shows a large number of lines like below. Error writing frame to output Too large number of skiped frames 13331376327 I don't know if it is related, but I also get a good number of the following errors in the ffmpeg instance that is feeding the ffserver. [h264 @ 0x21e51a0] reference picture missing during reorder [h264 @ 0x21e51a0] Missing reference picture Can anyone explain why I am getting either of these errors? I am testing on a local wireless network, and the VLC and ffserver instances are on the same linux machine. Thanks, Andrew P.S. Attached is my ffserver.conf -------------- next part -------------- A non-text attachment was scrubbed... Name: ffserver.conf Type: application/octet-stream Size: 575 bytes Desc: not available URL: From cehoyos at ag.or.at Sat Mar 31 01:42:26 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 30 Mar 2012 23:42:26 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?-flags2_+ivlc+non=5Flinear=5Fq_issues?= References: <4F75DA04.3040200@yahoo.com> Message-ID: Tim Nicholson yahoo.com> writes: > ffmpeg ... -flags2 +ivlc+non_linear_q Please try "-intra_vlc -non_linear_quant" instead. (And please consider searching ffmpeg -h) Carl Eugen From cehoyos at ag.or.at Sat Mar 31 01:44:35 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 30 Mar 2012 23:44:35 +0000 (UTC) Subject: [FFmpeg-user] bgra to yuv References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: Josh long gmail.com> writes: > ffmpeg -i 0_10_sec.avi -vcodec rawvideo -pix_fmt rgb24 0_10_secrgb..avi rawvideo in avi does not allow rgb24, only bgr24 is supported for that combination. Carl Eugen From brendan.brewster at gmail.com Sat Mar 31 02:12:36 2012 From: brendan.brewster at gmail.com (Brendan Brewster) Date: Fri, 30 Mar 2012 20:12:36 -0400 Subject: [FFmpeg-user] Referenced frames In-Reply-To: References: Message-ID: On Wed, Mar 28, 2012 at 12:41 PM, Oussama Stiti wrote: > Hello, > > I want to detect in a h264 video, which frames are related ( referenced) to > which frames. > For example : Frame 62 : I Frame. > Frame 63 : P Frame [referenced to frame 62] > Frame 64 : B Frame [referenced to frame 63 & 65] > Is it possible to display such result ? > What kind of command line, should i run ? > Thank you > > Regards > > -- > *Oussama Stiti* > > ?l?ve ing?nieur en t?l?communications ? Sup'com (?cole sup?rieure des > communications de Tunis) > > T?l: +21652363164 > E-mail: oussama.stiti at gmail.com > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Hi there, I think your only option is take a look at the select filter. http://ffmpeg.org/libavfilter.html#showinfo If you just want the showinfo output, an example invocation could be: ffmpeg -i *infile* -filter:v showinfo -f h264 - -an > /dev/null Hope this helps even a little. -Brendan From rickcorteza at gmail.com Sat Mar 31 03:25:36 2012 From: rickcorteza at gmail.com (Rick C.) Date: Sat, 31 Mar 2012 09:25:36 +0800 Subject: [FFmpeg-user] converting from drm protected videos Message-ID: Hi, If I'm not allowed to discuss this my apologies but how to convert a video that has drm protection? I know some apps make use of certain installed components to do this but I haven't figured it out yet. This can be very useful for let's say taking my own dvd and putting it on my iPod. thanks, rc From funkyirish at gmail.com Sat Mar 31 03:49:32 2012 From: funkyirish at gmail.com (Josh long) Date: Fri, 30 Mar 2012 20:49:32 -0500 Subject: [FFmpeg-user] bgra to yuv In-Reply-To: References: <1330634734197-4436338.post@n4.nabble.com> Message-ID: On Fri, Mar 30, 2012 at 6:44 PM, Carl Eugen Hoyos wrote: > Josh long gmail.com> writes: > > > ffmpeg -i 0_10_sec.avi -vcodec rawvideo -pix_fmt rgb24 0_10_secrgb..avi > > rawvideo in avi does not allow rgb24, only bgr24 is > supported for that combination. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > ah I see. Again I suppose I need to do more homework on this. So I suppose it's just a fluke that it plays the correct colors when I first go to rgba and then rgb24? Also why is it that yuv444p ends up so distorted, is it because I used rawvideo? Thanks again for all of your help. Joshua From robert at theMakers.com Sat Mar 31 06:14:25 2012 From: robert at theMakers.com (Robert Reinhardt) Date: Sat, 31 Mar 2012 04:14:25 +0000 Subject: [FFmpeg-user] .m4a to .flv - image/video fails In-Reply-To: References: , Message-ID: <2D405CD275952E49B92B7F48B3A0308A2B8CFC62@nakedex.flaction.com> Why are you going to FLV? Flash Player supports H.264/MP4, and most HTML5 browsers support H.264 MP4 as well. You'll get better quality for equivalent nitrates than the H.263 FLV compression ffmpeg offers. -Robert Robert Reinhardt Author, original Flash Bible series and ActionScript Bible series The difference knowledge + experience makes | Consultant @ [theMAKERS] { work: http://www.theMakers.com } { video: http://videoRx.com } { blog: http://probablyjustme.com } ________________________________________ From: ffmpeg-user-bounces at ffmpeg.org [ffmpeg-user-bounces at ffmpeg.org] on behalf of Josh [joshdotnet at gmail.com] Sent: Friday, March 30, 2012 7:24 AM To: ffmpeg-user at ffmpeg.org Subject: Re: [FFmpeg-user] .m4a to .flv - image/video fails Nancy, did you ever find a workaround for this problem? I am trying to convert .m4a podcast files with embedded images and chapters to .flv so I can more easily display and play them on my website. Any advice would be much appreciated. _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From phil_rhodes at rocketmail.com Sat Mar 31 13:57:03 2012 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Sat, 31 Mar 2012 12:57:03 +0100 Subject: [FFmpeg-user] converting from drm protected videos In-Reply-To: References: Message-ID: If there's a pushbutton approach to this, I'm all ears. Historically it has been quite tricky because there's no obvious way to discover which of the several chunks of video on a DVD is the main feature, and there are often problems with audio sync. P On Sat, 31 Mar 2012 02:25:36 +0100, Rick C. wrote: > Hi, > > If I'm not allowed to discuss this my apologies but how to convert a > video that has drm protection? I know some apps make use of certain > installed components to do this but I haven't figured it out yet. This > can be very useful for let's say taking my own dvd and putting it on my > iPod. thanks, > > rc > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From zammargu at marvell.com Sat Mar 31 17:28:38 2012 From: zammargu at marvell.com (Zahira Ammarguellat) Date: Sat, 31 Mar 2012 08:28:38 -0700 Subject: [FFmpeg-user] Buildin ffmpeg on QNX/ARM Message-ID: Hello, I have been trying to build the ffmeg library with ffplay enabled (I have built it without ffplay) but I don't seem to find the right configuration (ffplay is always disabled) Has someone even found the right configuration for QNX/ARM. Attached is the config file I am using. Thanks. Any help is appreciated. -Zahira -------------- next part -------------- A non-text attachment was scrubbed... Name: config.qnx Type: application/octet-stream Size: 494 bytes Desc: config.qnx URL: From daverice at mac.com Sat Mar 31 17:43:34 2012 From: daverice at mac.com (David Rice) Date: Sat, 31 Mar 2012 11:43:34 -0400 Subject: [FFmpeg-user] using flac parameters in ffmpeg Message-ID: Is it possible to use flac's --keep-foreign-metadata feature in ffmpeg when decoding or encoding flac files? Dave Rice From szumlins at gmail.com Sat Mar 31 18:11:39 2012 From: szumlins at gmail.com (Mike Szumlinski) Date: Sat, 31 Mar 2012 12:11:39 -0400 Subject: [FFmpeg-user] Oddity decoding DVCProHD and XDCAM Message-ID: <879C5526-21D2-4F68-9658-471BF8ED8829@gmail.com> I did some searching on the list and did not see anything relevant, so I hope this isn't a repost. I have compiled and installed the latest git version as of today (N-39412-gec0965b) and have tried throwing a few broadcast formats at ffmpeg for encode to H264. Doing DNxHD and AppleProRes 422 both worked and the result was a viewable H264 mp4 file. However, when my input file is a DVCProHD quicktime movie or an XDCAM MXF file, ffmpeg seems to start encoding without issue or error, but upon opening the output file all I see is gibberish and digital distortion. I'm new to ffmpeg, so I'm not sure if there is something I should be doing to properly feed these files (or certain types of files versus other input files) to make sure the output is proper. Any help would be appreciated. From cehoyos at ag.or.at Sat Mar 31 18:50:23 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 31 Mar 2012 16:50:23 +0000 (UTC) Subject: [FFmpeg-user] Oddity decoding DVCProHD and XDCAM References: <879C5526-21D2-4F68-9658-471BF8ED8829@gmail.com> Message-ID: Mike Szumlinski gmail.com> writes: > However, when my input file is a DVCProHD quicktime movie > or an XDCAM MXF file, ffmpeg seems to start encoding > without issue or error, but upon opening the output file > all I see is gibberish and digital distortion. Command line together with complete, uncut console output (and samples) missing. Carl Eugen From cehoyos at ag.or.at Sat Mar 31 18:51:34 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 31 Mar 2012 16:51:34 +0000 (UTC) Subject: [FFmpeg-user] using flac parameters in ffmpeg References: Message-ID: David Rice mac.com> writes: > Is it possible to use flac's --keep-foreign-metadata > feature in ffmpeg when decoding or encoding flac files? (Command line and complete, uncut console output missing.) Isn't this the default? Or do I misunderstand? Carl Eugen From cehoyos at ag.or.at Sat Mar 31 18:53:33 2012 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 31 Mar 2012 16:53:33 +0000 (UTC) Subject: [FFmpeg-user] Buildin ffmpeg on QNX/ARM References: Message-ID: Zahira Ammarguellat marvell.com> writes: > I have been trying to build the ffmeg library with ffplay enabled > (I have built it without ffplay) but I don't seem to find the > right configuration (ffplay is always disabled) Without looking at the attachment (shouldn't it be plain/text?), I suspect you have to provide SDL library, this is the only requirement for ffplay that I remember atm. Carl Eugen From szumlins at gmail.com Sat Mar 31 19:07:08 2012 From: szumlins at gmail.com (Mike Szumlinski) Date: Sat, 31 Mar 2012 13:07:08 -0400 Subject: [FFmpeg-user] Oddity decoding DVCProHD and XDCAM In-Reply-To: References: <879C5526-21D2-4F68-9658-471BF8ED8829@gmail.com> Message-ID: <178B20FC-6CCA-41EE-A6DF-C46F9E6B4A95@gmail.com> Carl, I've put the output into pastebin as to not clog up everyone's email with excessive output DVCProHD Output http://pastebin.com/NaiUVciH XDCam Output http://pastebin.com/f1QE0rny Upon digging deeper, the output actually does seem valid, but only in certain players (mplayer, vlc) and not widely available players (WMPlayer, Quicktime X, embedded HTML5