From doadam at gmail.com Tue Oct 1 11:32:12 2013 From: doadam at gmail.com (Adam) Date: Tue, 1 Oct 2013 11:32:12 +0200 Subject: [FFmpeg-user] Turning an image set into an FLV with transparent background Message-ID: I have an image set (over 900 PNGs) that I want to merge into FLV. I've been searching on Google and other places and the best result I got is a working movie, with a black background. The command I've been trying: avconv -r 24 -i HippoBingo_B_Render_%04d.png -i music.wav -r 24 -ac 1 -vcodec flv -strict experimental -s 256x384 -ab 128k z.flv console output: adam at ubuntu:~/render$ avconv -r 24 -i HippoBingo_B_Render_%04d.png -i music.wav -r 24 -ac 1 -vcodec flv -strict experimental -s 256x384 -ab 128k z.flv avconv version 0.8.6-6:0.8.6-1ubuntu2, Copyright (c) 2000-2013 the Libav developers built on Mar 30 2013 22:20:06 with gcc 4.7.2 Input #0, image2, from 'HippoBingo_B_Render_%04d.png': Duration: 00:00:34.16, start: 0.000000, bitrate: N/A Stream #0.0: Video: png, bgra, 400x600, 24 fps, 24 tbr, 24 tbn, 24 tbc [wav @ 0x1ba8840] max_analyze_duration reached Input #1, wav, from 'music.wav': Duration: 00:00:32.66, bitrate: 1412 kb/s Stream #1.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s File 'z.flv' already exists. Overwrite ? [y/N] y Incompatible pixel format 'bgra' for codec 'flv', auto-selecting format 'yuv420p' [buffer @ 0x21091a0] w:400 h:600 pixfmt:bgra [scale @ 0x211fda0] w:400 h:600 fmt:bgra -> w:256 h:384 fmt:yuv420p flags:0x4 Output #0, flv, to 'z.flv': Metadata: encoder : Lavf53.21.1 Stream #0.0: Video: flv, yuv420p, 256x384, q=2-31, 200 kb/s, 1k tbn, 24 tbc Stream #0.1: Audio: adpcm_swf, 44100 Hz, 1 channels, s16, 176 kb/s Stream mapping: Stream #0:0 -> #0:0 (png -> flv) Stream #1:0 -> #0:1 (pcm_s16le -> adpcm_swf) Press ctrl-c to stop encoding frame= 820 fps=198 q=2.0 Lsize= 1776kB time=32.69 bitrate= 445.1kbits/s video:1046kB audio:706kB global headers:0kB muxing overhead 1.376779% adam at ubuntu:~/render$ avconv -r 24 -i HippoBingo_B_Render_%04d.png -i music.wav -r 24 -ac 1 -vcodec flv -strict experimental -s 256x384 -ab 128k z.flv avconv version 0.8.6-6:0.8.6-1ubuntu2, Copyright (c) 2000-2013 the Libav developers built on Mar 30 2013 22:20:06 with gcc 4.7.2 Input #0, image2, from 'HippoBingo_B_Render_%04d.png': Duration: 00:00:34.16, start: 0.000000, bitrate: N/A Stream #0.0: Video: png, bgra, 400x600, 24 fps, 24 tbr, 24 tbn, 24 tbc [wav @ 0x1ba8840] max_analyze_duration reached Input #1, wav, from 'music.wav': Duration: 00:00:32.66, bitrate: 1412 kb/s Stream #1.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s File 'z.flv' already exists. Overwrite ? [y/N] y Incompatible pixel format 'bgra' for codec 'flv', auto-selecting format 'yuv420p' [buffer @ 0x21091a0] w:400 h:600 pixfmt:bgra [scale @ 0x211fda0] w:400 h:600 fmt:bgra -> w:256 h:384 fmt:yuv420p flags:0x4 Output #0, flv, to 'z.flv': Metadata: encoder : Lavf53.21.1 Stream #0.0: Video: flv, yuv420p, 256x384, q=2-31, 200 kb/s, 1k tbn, 24 tbc Stream #0.1: Audio: adpcm_swf, 44100 Hz, 1 channels, s16, 176 kb/s Stream mapping: Stream #0:0 -> #0:0 (png -> flv) Stream #1:0 -> #0:1 (pcm_s16le -> adpcm_swf) Press ctrl-c to stop encoding frame= 820 fps=198 q=2.0 Lsize= 1776kB time=32.69 bitrate= 445.1kbits/s video:1046kB audio:706kB global headers:0kB muxing overhead 1.376779% Why do I get a black background? Sincerely, Adam. From soho123.2012 at gmail.com Tue Oct 1 14:38:54 2013 From: soho123.2012 at gmail.com (Huang Soho) Date: Tue, 1 Oct 2013 20:38:54 +0800 Subject: [FFmpeg-user] ffserver+ffmpeg how to inform VLC player the frame rate is decrease? Message-ID: Hi All, I got a big problem when I use VLC 2.1.0-rc2 to play live H.264 stream that is output by ffserver. the stream format is rtp. it includes Audio+ video. One rtp stream is for Audio and another is for Video. The payload type is 96 for Video, and 97 for Audio. Video format: H.264, 1280x720, 30fps. Audio format: PCM_S16_LE, 2 channel, 48000 sample rate. I see the error when VLC debug message is enabled. "avcodec error: more than 5 seconds of late video -> dropping frame (computer too slow ?)" then the video stream is freeze. Because the frame rate is decrease if the light is dark when ffmpeg capture video data . Then the frame gap is larger than the first 10 minutes. VLC does not know frame rate is decrease. VLC just know rtp timestamp has larger gap in the afterward packets. Is there any method to inform VLC the timestamp will get more larger gap? From andrey.krieger.utkin at gmail.com Tue Oct 1 15:40:09 2013 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Tue, 1 Oct 2013 16:40:09 +0300 Subject: [FFmpeg-user] activating rtmp_* options In-Reply-To: References: Message-ID: 2013/9/30 Mustafa Yavuz <89.yavuz at gmail.com>: > Hi, > I am enabling rtmp while configuration with --enable-librtmp of ffmpeg, > however, most of rtmp options were not installed (only -rtmp_app and > -rtmp_playpath), how will I enable the others like rtmp_swfurl, > rtmp_pageurl? There are two RTMP implementations usable with ffmpeg: 1. using librtmp, 2. ffmpeg internal protocol driver. Options you need (rtmp_swfurl etc.) belong to the second one. So drop --enable-librtmp flag, and add --enable-rtmp. -- Andrey Utkin From blacktrash at gmx.net Tue Oct 1 17:22:21 2013 From: blacktrash at gmx.net (Christian Ebert) Date: Tue, 1 Oct 2013 16:22:21 +0100 Subject: [FFmpeg-user] aresample out_channel_layout values Message-ID: <20131001152221.GA24454@krille.blacktrash.org> Hi, What are the correct values to pass to in/out_channel_layout as aresample option? The aformat filter accepts aformat=channel_layouts=mono, but aresample=22050:out_channel_layout=mono gives the following error: [SWR @ 0x7fe9bb000600] [Eval @ 0x7fff570741d0] Undefined constant or missing '(' in 'mono' [SWR @ 0x7fe9bb000600] Unable to parse option value "mono" [AVFilterGraph @ 0x7fe9b9c16140] Error initializing filter 'aresample' with args '22050:ocl=mono' Error configuring filters. I also tried out_channel_layout=FC (taken from ffmpeg -layouts) with the same error. Finally: What would be the difference between: aformat=channel_layouts=stereo and aresample=out_channel_layout= -- \black\trash movie _COWBOY CANOE COMA_ Ein deutscher Western/A German Western --->> http://www.blacktrash.org/underdogma/ccc.php From s0527705277 at gmail.com Tue Oct 1 21:00:21 2013 From: s0527705277 at gmail.com (Stephan Asovski) Date: Tue, 1 Oct 2013 22:00:21 +0300 Subject: [FFmpeg-user] Can Intel Xeon Phi improve ffmpeg performance? Message-ID: I'm building a server for video encoding and I'm planning to use the latest Xenon e5-2697 v2. Can Xeon Phi help me improve ffmpeg encoding performance? a bit confused by this new "processor" http://ark.intel.com/products/family/71840/Intel-Xeon-Phi-Coprocessors/server From lou at lrcd.com Tue Oct 1 21:09:26 2013 From: lou at lrcd.com (Lou) Date: Tue, 1 Oct 2013 11:09:26 -0800 Subject: [FFmpeg-user] Turning an image set into an FLV with transparent background In-Reply-To: References: Message-ID: <20131001110926.3167db9e@lrcd.com> On Tue, 1 Oct 2013 11:32:12 +0200 Adam wrote: > avconv version 0.8.6-6:0.8.6-1ubuntu2, Copyright (c) 2000-2013 the Libav > developers This is not from the FFmpeg project and is therefore not supported here. If you would like help, then please use ffmpeg from FFmpeg; otherwise you will need to get help from Libav. For most Ubuntu users the easiest way to use ffmpeg is by downloading a Linux build: http://ffmpeg.org/download.html#LinuxBuilds Or you could always compile ffmpeg: http://trac.ffmpeg.org/wiki/UbuntuCompilationGuide From buddhabauch at gmx.at Tue Oct 1 12:41:53 2013 From: buddhabauch at gmx.at (buddhabauch) Date: Tue, 01 Oct 2013 12:41:53 +0200 Subject: [FFmpeg-user] ffvhuff codec improvement Message-ID: <524AA6F1.7040600@gmx.at> Hello, Huffyuv is know as fast codec which can handle the pixel formats yuv422p and rgb24 ffvhuff is a improved huffyuv version of ffmpeg which can handle yuv422p yuv420p and rgb24 and ffvhuff is faster than huffyuv. Is it possible to improve this codec that it can handle more pixel formats and multithreading encoding even like FFv1.3? e.g. yuv411p yuv422p10le yuv420p10le .... If yes, ffvhuff would be a very good lossless mezzanine codec for SD and HD. Best Regards Christoph Gerstbauer From andrey.krieger.utkin at gmail.com Tue Oct 1 22:54:07 2013 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Tue, 1 Oct 2013 23:54:07 +0300 Subject: [FFmpeg-user] Can Intel Xeon Phi improve ffmpeg performance? In-Reply-To: References: Message-ID: 2013/10/1 Stephan Asovski : > I'm building a server for video encoding and I'm planning to use the latest > Xenon e5-2697 v2. > Can Xeon Phi help me improve ffmpeg encoding performance? As far as i googled out, Phi coprocessor occurs logically as separate network host. So it can be leveraged easily in two ways: 1) Running generic application straightly on it. You can ssh to the coprocessor and do whatever you need in its internal Linux. 2) Using computation distribution technologies like MPI, OpenMP or NUMA. But as far as i know, ffmpeg does not leverage any of MPI, OpenMP and NUMA. -- Andrey Utkin From cehoyos at ag.or.at Tue Oct 1 23:13:08 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Oct 2013 21:13:08 +0000 (UTC) Subject: [FFmpeg-user] ERROR: libass not found although I install it References: <1380261096301-4661468.post@n4.nabble.com> Message-ID: SrApy msn.com> writes: > ERROR: libass not found Please provide the last lines of config.log Carl Eugen From cehoyos at ag.or.at Tue Oct 1 23:14:54 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Oct 2013 21:14:54 +0000 (UTC) Subject: [FFmpeg-user] Stripping OGG with the same command References: Message-ID: Shay Yaish gmail.com> writes: > I won't explain much as all of the information for this > issue is presented under this page: If you are not searching for -flags bitexact, please provide all necessary information to understand your problem here on the mailing list, please do not use external resources. Carl Eugen From chronek at interia.eu Tue Oct 1 23:24:32 2013 From: chronek at interia.eu (mike) Date: Tue, 01 Oct 2013 23:24:32 +0200 Subject: [FFmpeg-user] sei type truncated Message-ID: <524B3D90.4040609@interia.eu> Hello, I do not know if it is a bug or i am doing something wrong. I recorded with Open Broadcaster Software with enabled quick sync some game in 1920x1080 resolution , video is playable in vlc , then was trying convert with ffmpeg to huffyuv , video is playable , but image is broken , there are only multicolor lines on whole image. In log i have that error : SEI type truncated , need advice. Log: C:\inne\ffmpeg.exe -loglevel verbose -i obs-out.mp4 -an -vcodec huffyuv out.avi ffmpeg version N-56793-g01e3340 Copyright (c) 2000-2013 the FFmpeg developers built on Sep 30 2013 18:07:49 with gcc 4.8.1 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --ena le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --en ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libs eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwav ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 46.100 / 52. 46.100 libavcodec 55. 33.101 / 55. 33.101 libavformat 55. 18.104 / 55. 18.104 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 88.100 / 3. 88.100 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 [h264 @ 0000000002908d40] SEI type 5 truncated at 1126 [h264 @ 0000000002908d40] SEI type 0 truncated at 49 [h264 @ 0000000002908d40] SEI type 1 truncated at 53 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'obs-out.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 2013-10-01 21:02:31 encoder : Open Broadcaster Software v0.571b Duration: 00:00:02.80, start: 0.000000, bitrate: 61217 kb/s Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 3 kb/s (default) Metadata: creation_time : 2013-10-01 21:02:31 handler_name : Sound Media Handler Stream #0:1(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 61931 kb/s, 30.01 fps, 30 tbr, 1k tbn, 60 tbc ( efault) Metadata: creation_time : 2013-10-01 21:02:31 handler_name : Video Media Handler File 'out.avi' already exists. Overwrite ? [y/N] y [graph 0 input from stream 0:1 @ 00000000048b3c80] w:1920 h:1080 pixfmt:yuv420p tb:1/1000 fr:30/1 sar:1/1 sws_param:flags=2 [auto-inserted scaler 0 @ 0000000002909ea0] w:iw h:ih flags:'0x4' interl:0 [format @ 0000000002909cc0] auto-inserting filter 'auto-inserted scaler 0' between the filter 'Parsed_null_0' and the filter 'format' [auto-inserted scaler 0 @ 0000000002909ea0] w:1920 h:1080 fmt:yuv420p sar:1/1 -> w:1920 h:1080 fmt:yuv422p sar:1/1 flags:0x4 [huffyuv @ 000000000497eae0] using huffyuv 2.2.0 or newer interlacing flag Output #0, avi, to 'out.avi': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 ISFT : Lavf55.18.104 Stream #0:0(und): Video: huffyuv (HFYU / 0x55594648), yuv422p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 30 tbn, 30 tbc (default) Metadata: creation_time : 2013-10-01 21:02:31 handler_name : Video Media Handler Stream mapping: Stream #0:1 -> #0:0 (h264 -> huffyuv) Press [q] to stop, [?] for help [h264 @ 00000000048b1b20] SEI type 5 truncated at 1126 [h264 @ 00000000048b1b20] SEI type 0 truncated at 49 [h264 @ 00000000048b1b20] SEI type 1 truncated at 53 [h264 @ 0000000005839460] SEI type 1 truncated at 53 [h264 @ 00000000058398c0] SEI type 1 truncated at 53 [h264 @ 00000000048a3b20] SEI type 1 truncated at 53 [h264 @ 00000000048a4160] SEI type 1 truncated at 53 [h264 @ 0000000004f3ddc0] SEI type 1 truncated at 53 [h264 @ 0000000004d33d60] SEI type 1 truncated at 53 [h264 @ 0000000004d361e0] SEI type 1 truncated at 53 [h264 @ 0000000004d37420] SEI type 1 truncated at 53 [h264 @ 00000000048b1b20] SEI type 1 truncated at 53 [h264 @ 0000000005839460] SEI type 1 truncated at 53 [h264 @ 00000000058398c0] SEI type 1 truncated at 53 [h264 @ 00000000048a3b20] SEI type 1 truncated at 53 [h264 @ 00000000048a4160] SEI type 1 truncated at 53 [h264 @ 0000000004f3ddc0] SEI type 1 truncated at 53 [h264 @ 0000000004d33d60] SEI type 1 truncated at 53 [h264 @ 0000000004d361e0] SEI type 1 truncated at 53 [h264 @ 0000000004d37420] SEI type 1 truncated at 53 [h264 @ 00000000048b1b20] SEI type 1 truncated at 53 [h264 @ 0000000005839460] SEI type 1 truncated at 53 [h264 @ 00000000058398c0] SEI type 1 truncated at 53 [h264 @ 00000000048a3b20] SEI type 1 truncated at 53 [h264 @ 00000000048a4160] SEI type 1 truncated at 53 [h264 @ 0000000004f3ddc0] SEI type 1 truncated at 53 [h264 @ 0000000004d33d60] SEI type 1 truncated at 53 [h264 @ 0000000004d361e0] SEI type 1 truncated at 53 [h264 @ 0000000004d37420] SEI type 1 truncated at 53 [h264 @ 00000000048b1b20] SEI type 1 truncated at 53 [h264 @ 0000000005839460] SEI type 1 truncated at 53 ......... From cehoyos at ag.or.at Tue Oct 1 23:22:19 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Oct 2013 21:22:19 +0000 (UTC) Subject: [FFmpeg-user] any way to force mov stsc atom to <1MB References: <85CB68C6-0356-42BC-9C57-2EEDE7D2FEE7@mac.com> Message-ID: Dave Rice mac.com> writes: > I use ffmpeg to create files to be used in a broadcast > system. After much digging, testing, and working with > the vendor I've found that the quicktime demuxer of > the broadcast server only reads the first megabyte of > the stsc atom and if the stsc atom is larger than one > megabyte then the audio fails to play. > Looking at QuickTime files from other tools I see that > stsc is typically much smaller than 1MB. Are there any > commands I could use to reduce the size of the stsc atom? Please provide the command line that produces the failing output file together with the complete, uncut console output. Carl Eugen From cehoyos at ag.or.at Tue Oct 1 23:25:46 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Oct 2013 21:25:46 +0000 (UTC) Subject: [FFmpeg-user] ffvhuff codec improvement References: <524AA6F1.7040600@gmx.at> Message-ID: buddhabauch gmx.at> writes: > If yes, ffvhuff would be a very good lossless > mezzanine codec for SD and HD. What's wrong with ffv1? (Which produces significantly smaller files iirc.) Multi-threading should be possible, patch certainly welcome. Carl Eugen From cehoyos at ag.or.at Tue Oct 1 23:27:00 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Oct 2013 21:27:00 +0000 (UTC) Subject: [FFmpeg-user] Error to compile app with Xcode 5 References: Message-ID: John Bishop gmail.com> writes: > I try to compile one app (linphone) on Mac Lion 10.8.5 > with new Xcode5 (before have XCode 4.6.3 and can copile > without any problem), i tried to download by git last > version of ffmpeg but i got same error, follow the > config.log file with the error What was your configure line? Carl Eugen From chronek at interia.eu Tue Oct 1 23:39:39 2013 From: chronek at interia.eu (mike) Date: Tue, 01 Oct 2013 23:39:39 +0200 Subject: [FFmpeg-user] ffvhuff codec improvement In-Reply-To: References: <524AA6F1.7040600@gmx.at> Message-ID: <524B411B.502@interia.eu> > What's wrong with ffv1? > (Which produces significantly smaller files iirc.) For me ffv1 is usable only if i need record in 444p, but it is not usable for any editing purpose, cause it not operate on single frame (seeking taking too long), From cehoyos at ag.or.at Tue Oct 1 23:37:07 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Oct 2013 21:37:07 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?Setting_option_video=5Fsize_/_progressive?= =?utf-8?q?_/_interlaced?= References: <1380553147996-4661522.post@n4.nabble.com> Message-ID: billyx gmx.at> writes: > The command line I think should be correct is: > ffmpeg.exe -f h264 -r 12 -s 704x576 -i > ch00000000000001-130930-095853-105853-01p101000000.264 > -vcodec copy -acodec copy test.mp4 Difficult to answer... You hopefully should not have to specify "-f h264", FFmpeg should auto-detect it, if it fails there may be a bug (sample welcome). You cannot specify a size for h264 streams, so please remove -s 704x576. You actually can specify an input frame rate, but this is only a (particularly ugly) hack around a bug in FFmpeg, I suggest you remove "-r 12" until the speed is the only thing wrong about the output file. -acodec copy makes no sense if your input is raw h264. I wonder why you are using -vcodec copy, I suspect what you actually want is to reencode the input file... But the real question is: Why do you get so many errors while decoding? Is the file broken (because it comes from an transmission that is known not to be error- free)? Or is this an intentionally broken file from a camera (cctv or similar)? In that case please provide a sample and please tell us about the camera (type). Re-reading your command line, I realize that this might be the stream of a mxf file. Is FFmpeg unable to read the mxf file? If yes, a sample would be very welcome! (mxf as input should fix the problem you are trying to solve with -r 12.) If all you need is to scale, please use the scale filter, if you want merge two fields that are encoded after each other, please see the filter documentation, one of the interlace filter should support it. Carl Eugen From cehoyos at ag.or.at Tue Oct 1 23:17:09 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Oct 2013 21:17:09 +0000 (UTC) Subject: [FFmpeg-user] configure failed for ffmpeg in debian system References: Message-ID: kostas kapetanakis gmail.com> writes: > root mclab1:/home/kapekost/ffmpeg# ./configure > --enable-shared --disable-debug --enable-libx264 > libx264 is gpl and --enable-gpl is not specified. FFmpeg by default is licensed under the LGPL, x264 is more restrictive and cannot be distributed under the rules of the LGPL. If you want to link FFmpeg against x264 (this is what --enable-libx264 does), you have to enable the GPL for all FFmpeg components (to make it clear that it is not LGPL anymore). Hope that helps, Carl Eugen From cehoyos at ag.or.at Tue Oct 1 23:20:45 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Oct 2013 21:20:45 +0000 (UTC) Subject: [FFmpeg-user] Demultiplexing multichannel audio References: Message-ID: Hecken Penner googlemail.com> writes: > "ffmpeg -i "$movie" -filter:v 'channelsplit I did not test but I assume that you cannot use an audio filter in a video filter chain. Please understand that since avconv contains several hundred known bugs not reproducible with FFmpeg, some of them security relevant, we cannot support it here. Carl Eugen From cehoyos at ag.or.at Tue Oct 1 23:53:28 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Oct 2013 21:53:28 +0000 (UTC) Subject: [FFmpeg-user] Turning an image set into an FLV with transparent background References: <20131001110926.3167db9e@lrcd.com> Message-ID: Lou lrcd.com> writes: > > avconv version 0.8.6-6:0.8.6-1ubuntu2, Copyright > > (c) 2000-2013 the Libav developers > > This is not from the FFmpeg project and is therefore > not supported here. To elaborate: avconv contains several hundred known bugs that are not reproducible with FFmpeg (among them both security relevant and transparency-related issues), so please test current FFmpeg to rule out a known issue with avconv (that we cannot fix for you). Carl Eugen From cehoyos at ag.or.at Tue Oct 1 23:50:30 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Oct 2013 21:50:30 +0000 (UTC) Subject: [FFmpeg-user] ffvhuff codec improvement References: <524AA6F1.7040600@gmx.at> <524B411B.502@interia.eu> Message-ID: mike interia.eu> writes: > > What's wrong with ffv1? > > (Which produces significantly smaller files iirc.) > > For me ffv1 is usable only if i need record in 444p, > but it is not usable for any editing purpose, cause > it not operate on single frame (seeking taking too > long), I may misunderstand but you can use ffv1 intra-only (and for some input files, its output will still be several orders of magnitude smaller than huffyuv). In any case: Please report your problems with ffv1 in a way that allows developers to reproduce them, fixes for ffv1 are significantly more likely than new features for huffyuv (except maybe multi- threading). Carl Eugen From cehoyos at ag.or.at Tue Oct 1 23:38:45 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Oct 2013 21:38:45 +0000 (UTC) Subject: [FFmpeg-user] sei type truncated References: <524B3D90.4040609@interia.eu> Message-ID: mike interia.eu> writes: > I recorded with Open Broadcaster Software with enabled > quick sync some game in 1920x1080 resolution , video is > playable in vlc , then was trying convert with ffmpeg to > huffyuv , video is playable , but image is broken Please provide the input sample. Carl Eugen From leonard at kcfchurch.org Wed Oct 2 03:32:06 2013 From: leonard at kcfchurch.org (Leonard Bogard) Date: Tue, 1 Oct 2013 18:32:06 -0700 Subject: [FFmpeg-user] Anybody got experience with ffmpeg+YouTube live event? Message-ID: I looked at their setting recommendations and followed them best I could. YouTube reported that it was receiving the RTMP stream but it showed nothing but a black screen. If nobody has any experience with it before that's okay. It's really new and I don't expect anybody to have messed with it (yet.) tia From chronek at interia.eu Wed Oct 2 03:36:16 2013 From: chronek at interia.eu (mike) Date: Wed, 02 Oct 2013 03:36:16 +0200 Subject: [FFmpeg-user] sei type truncated In-Reply-To: References: <524B3D90.4040609@interia.eu> Message-ID: <524B7890.4080609@interia.eu> > Please provide the input sample. > > Carl Eugen I do not know if it send corect , so trying again - www.sendspace.com/file/3t3nzx Mike From billyx at gmx.at Wed Oct 2 02:21:15 2013 From: billyx at gmx.at (billyx) Date: Tue, 1 Oct 2013 17:21:15 -0700 (PDT) Subject: [FFmpeg-user] Setting option video_size / progressive / interlaced In-Reply-To: References: <1380553147996-4661522.post@n4.nabble.com> Message-ID: <1380673275621-4661553.post@n4.nabble.com> Hello Carl, Thanks for your try to support. I have some kind of video stream encoded by a hardware video encoder. I assume there are some kind of errors in the video stream (e.g. wrong size information). The transmission should be without errors. What I would like to get is corrected video stream with a correct container and, when possible, without reencoding, because it is not necessary. And for this goal I try to choose the correct parameters: just copy the video stream, but set correct framerate and resolution. If I remove all parameters I get the error: Invalid data found when processing input Adding -f h264 I get a video running twice as fast (25 frames/s) as it should be (12 frames/s) with the half length and with the half height (704x288 instead of 704x576) Adding -r 12 I get a video running with the correct speed and the correct length but with the half height (704x288 instead of 704x576) I could perhaps make an example if it helps. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Setting-option-video-size-progressive-interlaced-tp4661522p4661553.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From srapy at msn.com Tue Oct 1 23:52:13 2013 From: srapy at msn.com (SrApy) Date: Tue, 1 Oct 2013 14:52:13 -0700 (PDT) Subject: [FFmpeg-user] ERROR: libass not found although I install it In-Reply-To: References: <1380261096301-4661468.post@n4.nabble.com> Message-ID: <1380664333994-4661549.post@n4.nabble.com> Thank you for replying, this is the last few lines in config.log ======== 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 -c -o /usr/local/src/tmp/ffconf.Kgz18516.o /usr/local/src/tmp/ffconf.Yto18508.c gcc -Wl,--as-needed -o /usr/local/src/tmp/ffconf.fOq18512 /usr/local/src/tmp/ffconf.Kgz18516.o -lm -pthread -lbz2 -lz -lrt check_pkg_config libass ass/ass.h ass_library_init pkg-config --exists --print-errors libass Package libass was not found in the pkg-config search path. Perhaps you should add the directory containing `libass.pc' to the PKG_CONFIG_PATH environment variable No package 'libass' found ERROR: libass not found ======== -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ERROR-libass-not-found-although-I-install-it-tp4661468p4661549.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From chronek at interia.eu Wed Oct 2 01:07:37 2013 From: chronek at interia.eu (mike) Date: Wed, 02 Oct 2013 01:07:37 +0200 Subject: [FFmpeg-user] sei type truncated In-Reply-To: References: <524B3D90.4040609@interia.eu> Message-ID: <524B55B9.4000903@interia.eu> > Please provide the input sample. > > Carl Eugen I do not know how to send , so distribute file by sendspace: http://www.sendspace.com/file/3t3nzx Mike From cehoyos at ag.or.at Wed Oct 2 10:57:11 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 2 Oct 2013 08:57:11 +0000 (UTC) Subject: [FFmpeg-user] ERROR: libass not found although I install it References: <1380261096301-4661468.post@n4.nabble.com> <1380664333994-4661549.post@n4.nabble.com> Message-ID: SrApy msn.com> writes: > this is the last few lines in config.log > pkg-config --exists --print-errors libass > Package libass was not found in the pkg-config search path. > Perhaps you should add the directory containing `libass.pc' > to the PKG_CONFIG_PATH environment variable > No package 'libass' found I assume that you get the exact same error message if you type: $ pkg-config --exists --print-errors libass You will have to fix that if you want to use libass from FFmpeg. I suspect it is possible to change FFmpeg so that pkg-config will be unneeded but this will need an explanation why it doesn't work for you. Carl Eugen From cehoyos at ag.or.at Wed Oct 2 10:59:04 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 2 Oct 2013 08:59:04 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?Setting_option_video=5Fsize_/_progressive?= =?utf-8?q?_/=09interlaced?= References: <1380553147996-4661522.post@n4.nabble.com> <1380673275621-4661553.post@n4.nabble.com> Message-ID: billyx gmx.at> writes: > If I remove all parameters I get the error: Invalid data > found when processing input Please provide a sample (and please tell us the name of the hardware encoder). Carl Eugen From doadam at gmail.com Wed Oct 2 11:41:16 2013 From: doadam at gmail.com (Adam) Date: Wed, 2 Oct 2013 11:41:16 +0200 Subject: [FFmpeg-user] Turning an image set into an FLV with transparent background In-Reply-To: References: <20131001110926.3167db9e@lrcd.com> Message-ID: Hey, I tested with this command and received the same results: ffmpeg -r 24 -i HippoBingo_B_Render_%04d.png -i music.wav -r 24 -ac 1 -vcodec flv -strict experimental -s 256x384 -ab 128k z.flv On Wed, Oct 2, 2013 at 12:53 AM, Carl Eugen Hoyos wrote: > Lou lrcd.com> writes: > > > > avconv version 0.8.6-6:0.8.6-1ubuntu2, Copyright > > > (c) 2000-2013 the Libav developers > > > > This is not from the FFmpeg project and is therefore > > not supported here. > > To elaborate: > avconv contains several hundred known bugs that are > not reproducible with FFmpeg (among them both security > relevant and transparency-related issues), so please > test current FFmpeg to rule out a known issue with > avconv (that we cannot fix for you). > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From andrey.krieger.utkin at gmail.com Wed Oct 2 11:44:58 2013 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Wed, 2 Oct 2013 12:44:58 +0300 Subject: [FFmpeg-user] Multicast UDP Port Issues In-Reply-To: <1380040634012-4661430.post@n4.nabble.com> References: <1379612334315-4661352.post@n4.nabble.com> <1380040634012-4661430.post@n4.nabble.com> Message-ID: 2013/9/24 Goldsmith81 : > I have narrowed the issue down, but I am still not sure why it is happening. > > When I use two instances of ffmpeg, since I am pulling the streams from the > same mpts UDP port, whenever I start the second instance of ffmpeg, the > first one just stops. > > Which ever one I start last gets control and the other stream pauses... > > Can anyone shed some light on this for me, please? I'm really stuck > here.... Please tell us your platform (OS etc.), ffmpeg version, full ffmpeg output, maybe with -loglevel debug. -- Andrey Utkin From cehoyos at ag.or.at Wed Oct 2 13:44:09 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 2 Oct 2013 11:44:09 +0000 (UTC) Subject: [FFmpeg-user] Turning an image set into an FLV with transparent background References: <20131001110926.3167db9e@lrcd.com> Message-ID: Adam gmail.com> writes: > I tested with this command and received the same results: > ffmpeg -r 24 -i HippoBingo_B_Render_%04d.png -i music.wav > -r 24 -ac 1 -vcodec flv -strict experimental -s 256x384 > -ab 128k z.flv (Complete, uncut console output missing.) I am not sure I understand your original report but if your question is why the output file does not contain transparency information then the answer is that FFmpeg's flv encoder does not support transparency (or in FFmpeg speech: Does not support yuva420p colour space). I am not sure if this is likely to get implemented but I created ticket #3022. Carl Eugen From cehoyos at ag.or.at Wed Oct 2 14:13:41 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 2 Oct 2013 12:13:41 +0000 (UTC) Subject: [FFmpeg-user] Turning an image set into an FLV with transparent background References: <20131001110926.3167db9e@lrcd.com> Message-ID: Carl Eugen Hoyos ag.or.at> writes: > I am not sure I understand your original report but if > your question is why the output file does not contain > transparency information then the answer is that FFmpeg's > flv encoder does not support transparency The reason is of course not that FFmpeg's encoder does not support transparency but that Sorenson H263 ("flv1") does not support transparency. Sorry, Carl Eugen From dashing.meng at gmail.com Wed Oct 2 14:52:51 2013 From: dashing.meng at gmail.com (littlebat) Date: Wed, 2 Oct 2013 20:52:51 +0800 Subject: [FFmpeg-user] ERROR: libass not found although I install it In-Reply-To: <1380664333994-4661549.post@n4.nabble.com> References: <1380261096301-4661468.post@n4.nabble.com> <1380664333994-4661549.post@n4.nabble.com> Message-ID: <20131002205251.5a38abb5.dashing.meng@gmail.com> On Tue, 1 Oct 2013 14:52:13 -0700 (PDT) SrApy wrote: > Thank you for replying, > this is the last few lines in config.log > ======== > > 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 -c > -o /usr/local/src/tmp/ffconf.Kgz18516.o /usr/local/src/tmp/ffconf.Yto18508.c > gcc -Wl,--as-needed -o /usr/local/src/tmp/ffconf.fOq18512 > /usr/local/src/tmp/ffconf.Kgz18516.o -lm -pthread -lbz2 -lz -lrt > check_pkg_config libass ass/ass.h ass_library_init > pkg-config --exists --print-errors libass > Package libass was not found in the pkg-config search path. > Perhaps you should add the directory containing `libass.pc' > to the PKG_CONFIG_PATH environment variable > No package 'libass' found > ERROR: libass not found > How about compile and install the latest libass from source? http://code.google.com/p/libass/ From shapableline at gmail.com Wed Oct 2 14:54:18 2013 From: shapableline at gmail.com (Sam Logan) Date: Wed, 2 Oct 2013 08:54:18 -0400 Subject: [FFmpeg-user] Combining 24fps Video with 23.976fps Audio In-Reply-To: References: Message-ID: On 9/24/13, Sam Logan wrote: > On 9/20/13, Carl Eugen Hoyos wrote: >>> The atempo and the setpts filter should allow you to change >>> the speed of the files: >>> http://ffmpeg.org/ffmpeg-filters.html#setpts_002c-asetpts >>> http://ffmpeg.org/ffmpeg-filters.html#atempo > > One more question on this topic: would it also be possible to also > change audio speed with > > -filter:a aresample > > I ask because the documentation at > http://ffmpeg.org/ffmpeg-filters.html#aresample-1 says "This filter is > also able to stretch/squeeze the audio data", which seems like the > same thing. But there are no examples either there or at > http://ffmpeg.org/ffmpeg-resampler.html on what exact arguments must > be used to do this. Basically, I'm asking, is it possible to duplicate > the effect of... > > asetpts=1000/1001*PTS > > ...or... > > atempo=1001/1000 > > ...using aresample, and if so, what exactly are the arguments to be used? No one responded to this from a week ago... does anyone know? Thanks. From pb at das-werkstatt.com Wed Oct 2 15:47:51 2013 From: pb at das-werkstatt.com (Peter B.) Date: Wed, 02 Oct 2013 15:47:51 +0200 Subject: [FFmpeg-user] ffvhuff codec improvement In-Reply-To: References: <524AA6F1.7040600@gmx.at> Message-ID: <524C2407.20406@das-werkstatt.com> On 10/01/2013 11:25 PM, Carl Eugen Hoyos wrote: > buddhabauch gmx.at> writes: > >> If yes, ffvhuff would be a very good lossless >> mezzanine codec for SD and HD. > What's wrong with ffv1? > (Which produces significantly smaller files iirc.) Of course, we all know that FFV1 rocks for lossless use! ;) I think what buddhabauch is looking for is just a very fast lossless codec for mezzanine usage. If used as a fast mezzanine format (e.g. during realtime capture), I think he's interested in being able to store the same pix_fmts with ffvhuff that ffv1 already supports. Regards, Pb From pb at das-werkstatt.com Wed Oct 2 15:51:40 2013 From: pb at das-werkstatt.com (Peter B.) Date: Wed, 02 Oct 2013 15:51:40 +0200 Subject: [FFmpeg-user] ffvhuff codec improvement In-Reply-To: <524B411B.502@interia.eu> References: <524AA6F1.7040600@gmx.at> <524B411B.502@interia.eu> Message-ID: <524C24EC.8010308@das-werkstatt.com> On 10/01/2013 11:39 PM, mike wrote: > >> What's wrong with ffv1? >> (Which produces significantly smaller files iirc.) > > For me ffv1 is usable only if i need record in 444p, but it is not > usable for any editing purpose, cause it not operate on single frame > (seeking taking too long), Interesting. May I ask which application, which container (avi, mov, mkv, ...) and which version of ffv1 (1 or 3) you are using? I'm asking, because we're using FFV1 for editing (over the network even) on a daily basis and it works super smooth. I do have noticed however, that the speed of seeking varies greatly from application to application. Regards, Pb From michaeljthorpe at yahoo.com Wed Oct 2 16:15:56 2013 From: michaeljthorpe at yahoo.com (Mike Thorpe) Date: Wed, 2 Oct 2013 07:15:56 -0700 (PDT) Subject: [FFmpeg-user] Anybody got experience with ffmpeg+YouTube live event? In-Reply-To: References: Message-ID: <1380723356.37362.YahooMailNeo@web140903.mail.bf1.yahoo.com> ________________________________ From: Leonard B? I looked at their setting recommendations and followed them best I could. YouTube reported that it was receiving the RTMP stream but it showed nothing but a black screen. If nobody has any experience with it before that's okay.? It's really new and I don't expect anybody to have messed with it (yet.) tia Leonard, I am interested in FFmpeg to you tube live and had posted a question here? about it on 8/22. ?In my case FFmpeg seemed to be running normally however? you tube said it was not receiving a stream. ?My settings were also following? you tube recommendations. ?Any advice or troubleshooting suggestions from the group would be appreciated. Mike From chronek at interia.eu Wed Oct 2 16:54:21 2013 From: chronek at interia.eu (mike) Date: Wed, 02 Oct 2013 16:54:21 +0200 Subject: [FFmpeg-user] ffvhuff codec improvement In-Reply-To: <524C24EC.8010308@das-werkstatt.com> References: <524AA6F1.7040600@gmx.at> <524B411B.502@interia.eu> <524C24EC.8010308@das-werkstatt.com> Message-ID: <524C339D.9090204@interia.eu> > Interesting. > May I ask which application, which container (avi, mov, mkv, ...) and > which version of ffv1 (1 or 3) you are using? I usually using Zeranoe FFmpeg builds, encoding to avi container, not know what version ffv1, just filled in vcodec , opening by avisynth (2.5.8) script with FFVideoSource (google code ffms2-2.18-rc1.7) From cehoyos at ag.or.at Wed Oct 2 17:11:25 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 2 Oct 2013 15:11:25 +0000 (UTC) Subject: [FFmpeg-user] ffvhuff codec improvement References: <524AA6F1.7040600@gmx.at> <524C2407.20406@das-werkstatt.com> Message-ID: Peter B. das-werkstatt.com> writes: > >> If yes, ffvhuff would be a very good lossless > >> mezzanine codec for SD and HD. > > > > What's wrong with ffv1? > > (Which produces significantly smaller files iirc.) > > Of course, we all know that FFV1 rocks for lossless use! ;) > I think what buddhabauch is looking for is just a very > fast lossless codec for mezzanine usage. I simply didn't understand "mezzanine". Is ffvhuff always faster than ffv1? Even with heavy multi-threading? Carl Eugen From pb at das-werkstatt.com Wed Oct 2 17:25:26 2013 From: pb at das-werkstatt.com (Peter B.) Date: Wed, 02 Oct 2013 17:25:26 +0200 Subject: [FFmpeg-user] ffvhuff codec improvement In-Reply-To: <524C339D.9090204@interia.eu> References: <524AA6F1.7040600@gmx.at> <524B411B.502@interia.eu> <524C24EC.8010308@das-werkstatt.com> <524C339D.9090204@interia.eu> Message-ID: <524C3AE6.5040300@das-werkstatt.com> On 10/02/2013 04:54 PM, mike wrote: > >> Interesting. >> May I ask which application, which container (avi, mov, mkv, ...) and >> which version of ffv1 (1 or 3) you are using? > > I usually using Zeranoe FFmpeg builds, encoding to avi container, not > know what version ffv1, just filled in vcodec , > opening by avisynth (2.5.8) script with FFVideoSource (google code > ffms2-2.18-rc1.7) Hm... >From that I'd assume you're using FFV1.1. If the resolution is beyond SD, then I'd rather assume performance problems with playback of yuv444p rather than seeking being the issue. Do the files playback in realtime without problems, once you've seeked to the desired position using avisynth/ffvideosource? Pb From chronek at interia.eu Wed Oct 2 17:52:16 2013 From: chronek at interia.eu (mike) Date: Wed, 02 Oct 2013 17:52:16 +0200 Subject: [FFmpeg-user] ffvhuff codec improvement In-Reply-To: <524C3AE6.5040300@das-werkstatt.com> References: <524AA6F1.7040600@gmx.at> <524B411B.502@interia.eu> <524C24EC.8010308@das-werkstatt.com> <524C339D.9090204@interia.eu> <524C3AE6.5040300@das-werkstatt.com> Message-ID: <524C4130.60705@interia.eu> >>From that I'd assume you're using FFV1.1. > If the resolution is beyond SD, then I'd rather assume performance > problems with playback of yuv444p rather than seeking being the issue. it could be performance problems cause video resolution was always around 3 mega pixels From belcampo at zonnet.nl Wed Oct 2 18:10:04 2013 From: belcampo at zonnet.nl (Henk D. Schoneveld) Date: Wed, 2 Oct 2013 18:10:04 +0200 Subject: [FFmpeg-user] I am sure something is wrong with configure In-Reply-To: References: Message-ID: <0DF48890-265D-4C04-9BD6-B89A85F927F6@zonnet.nl> On Sep 27, 2013, at 2:59 PM, SrAp - wrote: > Hello, I am trying to compile latest ffmpeg and configure it like shown ./configure --enable-version3 --enable-libopencore-amrnb \ --enable-libopencore-amrwb --enable-libvpx --enable-libfaac \ --enable-libmp3lame --enable-libtheora --enable-libvorbis \ --enable-libx264 --enable-libxvid --enable-gpl --enable-postproc \ --enable-nonfree --enable-fontconfig --enable-libass > I got this error root at server1 [~/ffmpeg/ffmpeg]# ./configure --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvpx --enable-libfaac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-gpl --enable-postproc --enable-nonfree --enable-fontconfig --enable-libass ERROR: libass not found You need the development package of libass something like libass-devel > =============xxxx============ > I install libass and when I run whereis libass ----- root at server1 [~/ffmpeg/ffmpeg]# whereis libass libass: /usr/lib/libass.so /usr/local/lib/libass.so /usr/local/lib/libass.a /usr/local/lib/libass.la ----- > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From elliottbalsley at gmail.com Wed Oct 2 18:28:03 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Wed, 2 Oct 2013 09:28:03 -0700 Subject: [FFmpeg-user] Hardware Acceleration on OS X Message-ID: I've read bits and pieces about ffmpeg's hardware acceleration, but can't find a full answer. On OS X, is VDA the only supported hwaccel? In what ways can it actually help? I primarily use ffmpeg for x264 encodes, and libx264 does all the heavy lifting. I have an octo-core Mac Pro with GTX570 (2.5GB), so it would be great to utilize that powerful GPU if possible. From tevans.uk at googlemail.com Wed Oct 2 18:35:42 2013 From: tevans.uk at googlemail.com (Tom Evans) Date: Wed, 2 Oct 2013 17:35:42 +0100 Subject: [FFmpeg-user] ffvhuff codec improvement In-Reply-To: References: <524AA6F1.7040600@gmx.at> <524C2407.20406@das-werkstatt.com> Message-ID: On Wed, Oct 2, 2013 at 4:11 PM, Carl Eugen Hoyos wrote: > I simply didn't understand "mezzanine". A mezzanine floor is a floor between two levels, a mezzanine codec is a codec that is between uncompressed recordings and heavily compressed recordings, like Dirac. Cheers Tom From tevans.uk at googlemail.com Wed Oct 2 18:39:31 2013 From: tevans.uk at googlemail.com (Tom Evans) Date: Wed, 2 Oct 2013 17:39:31 +0100 Subject: [FFmpeg-user] Hardware Acceleration on OS X In-Reply-To: References: Message-ID: On Wed, Oct 2, 2013 at 5:28 PM, Elliott Balsley wrote: > I've read bits and pieces about ffmpeg's hardware acceleration, but can't find a full answer. > On OS X, is VDA the only supported hwaccel? In what ways can it actually help? I primarily use ffmpeg for x264 encodes, and libx264 does all the heavy lifting. > I have an octo-core Mac Pro with GTX570 (2.5GB), so it would be great to utilize that powerful GPU if possible. libx264 is not hardware accelerated, and the video decode acceleration (VDA or VDPAU) on your graphics card is limited to presentation. VD in both acronyms stands for 'Video Decode'. Cheers Tom From elliottbalsley at gmail.com Wed Oct 2 18:43:11 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Wed, 2 Oct 2013 09:43:11 -0700 Subject: [FFmpeg-user] Hardware Acceleration on OS X In-Reply-To: References: Message-ID: OK that makes sense. Too bad. And am I correct that OpenCL only helps with a couple of filters (unsharp and deshake)? On Oct 2, 2013, at 9:39 AM, Tom Evans wrote: > On Wed, Oct 2, 2013 at 5:28 PM, Elliott Balsley > wrote: >> I've read bits and pieces about ffmpeg's hardware acceleration, but can't find a full answer. >> On OS X, is VDA the only supported hwaccel? In what ways can it actually help? I primarily use ffmpeg for x264 encodes, and libx264 does all the heavy lifting. >> I have an octo-core Mac Pro with GTX570 (2.5GB), so it would be great to utilize that powerful GPU if possible. > > libx264 is not hardware accelerated, and the video decode acceleration > (VDA or VDPAU) on your graphics card is limited to presentation. VD in > both acronyms stands for 'Video Decode'. > > Cheers > > Tom > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From alina at vicon.co.il Wed Oct 2 14:16:19 2013 From: alina at vicon.co.il (Alina lifshits) Date: Wed, 2 Oct 2013 15:16:19 +0300 Subject: [FFmpeg-user] H.265 codec Message-ID: <51813287-2833-4885-aa94-885fc26aa021@vicon.co.il> Hi, When H.265 codec will be supported in ffmpeg? Thanks, Alina From sgoldsmith at vibble.tv Wed Oct 2 15:19:53 2013 From: sgoldsmith at vibble.tv (Goldsmith81) Date: Wed, 2 Oct 2013 06:19:53 -0700 (PDT) Subject: [FFmpeg-user] Multicast UDP Port Issues In-Reply-To: References: <1379612334315-4661352.post@n4.nabble.com> <1380040634012-4661430.post@n4.nabble.com> Message-ID: Centos 6.3 FFmpeg version 0.6.5 The output I feel is irrelevant because all of the output just stops. Only one ffmpeg instance will continue running. The others just freeze. There has to be a way to "share" the udp port in ffmpeg. Thanks for the reply! -Stephen From: Andrey Utkin [via FFmpeg-users] [mailto:ml-node+s933282n4661560h16 at n4.nabble.com] Sent: Wednesday, October 02, 2013 5:46 AM To: Stephen Goldsmith Subject: Re: Multicast UDP Port Issues 2013/9/24 Goldsmith81 <[hidden email]>: > I have narrowed the issue down, but I am still not sure why it is happening. > > When I use two instances of ffmpeg, since I am pulling the streams from the > same mpts UDP port, whenever I start the second instance of ffmpeg, the > first one just stops. > > Which ever one I start last gets control and the other stream pauses... > > Can anyone shed some light on this for me, please? I'm really stuck > here.... Please tell us your platform (OS etc.), ffmpeg version, full ffmpeg output, maybe with -loglevel debug. -- Andrey Utkin _______________________________________________ ffmpeg-user mailing list [hidden email] http://ffmpeg.org/mailman/listinfo/ffmpeg-user ________________________________ If you reply to this email, your message will be added to the discussion below: http://ffmpeg-users.933282.n4.nabble.com/Multicast-UDP-Port-Issues-tp4661352p4661560.html To unsubscribe from Multicast UDP Port Issues, click here. NAML -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Multicast-UDP-Port-Issues-tp4661352p4661566.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From srapy at msn.com Wed Oct 2 12:16:52 2013 From: srapy at msn.com (SrApy) Date: Wed, 2 Oct 2013 03:16:52 -0700 (PDT) Subject: [FFmpeg-user] ERROR: libass not found although I install it In-Reply-To: References: <1380261096301-4661468.post@n4.nabble.com> <1380664333994-4661549.post@n4.nabble.com> Message-ID: Yes I got the exact same error when I run$ pkg-config --exists --print-errors libass do you know how to fix it? Date: Wed, 2 Oct 2013 01:58:11 -0700 From: ml-node+s933282n4661557h77 at n4.nabble.com To: srapy at msn.com Subject: Re: ERROR: libass not found although I install it SrApy msn.com> writes: > this is the last few lines in config.log > pkg-config --exists --print-errors libass > Package libass was not found in the pkg-config search path. > Perhaps you should add the directory containing `libass.pc' > to the PKG_CONFIG_PATH environment variable > No package 'libass' found I assume that you get the exact same error message if you type: $ pkg-config --exists --print-errors libass You will have to fix that if you want to use libass from FFmpeg. I suspect it is possible to change FFmpeg so that pkg-config will be unneeded but this will need an explanation why it doesn't work for you. Carl Eugen _______________________________________________ ffmpeg-user mailing list [hidden email] http://ffmpeg.org/mailman/listinfo/ffmpeg-user If you reply to this email, your message will be added to the discussion below: http://ffmpeg-users.933282.n4.nabble.com/ERROR-libass-not-found-although-I-install-it-tp4661468p4661557.html To unsubscribe from ERROR: libass not found although I install it, click here. NAML -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ERROR-libass-not-found-although-I-install-it-tp4661468p4661561.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From s0527705277 at gmail.com Wed Oct 2 19:53:45 2013 From: s0527705277 at gmail.com (Stephan Asovski) Date: Wed, 2 Oct 2013 20:53:45 +0300 Subject: [FFmpeg-user] H.265 codec In-Reply-To: <51813287-2833-4885-aa94-885fc26aa021@vicon.co.il> References: <51813287-2833-4885-aa94-885fc26aa021@vicon.co.il> Message-ID: You can download a patched version here https://code.google.com/p/x265/downloads/list 2013/10/2 Alina lifshits > Hi, > > > When H.265 codec will be supported in ffmpeg? > > > > Thanks, > Alina > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From elliottbalsley at gmail.com Wed Oct 2 20:28:56 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Wed, 2 Oct 2013 11:28:56 -0700 Subject: [FFmpeg-user] IVTC with pullup filter Message-ID: I have a source video that's 29.97 with 3:2 pulldown, and I want to encode it at 23.98p. This ffmpeg command produces video at 23.98, but with a pattern of 2 progressive frames followed by 2 interlaced frames. ffmpeg -ss 208 -i NCIS.ts -t 30 -vf "pullup,fps=24000/1001" -acodec copy -vcodec libx264 pullup.mkv On the other hand, I can achieve perfect results with this simple AviSynth script: MPEG2Source("NCIS.d2v", cpu=0) TFM().TDecimate() Trim(5000,5300) Since AviSynth isn't doing any deinterlacing, I don't think I should need to add yadif in ffmpeg. How can I make the pullup filter work correctly? From s0527705277 at gmail.com Wed Oct 2 20:35:52 2013 From: s0527705277 at gmail.com (Stephan Asovski) Date: Wed, 2 Oct 2013 21:35:52 +0300 Subject: [FFmpeg-user] Can Intel Xeon Phi improve ffmpeg performance? In-Reply-To: References: Message-ID: hmm sad I can't use the second option. What else can I try to speed encoding time? 2013/10/1 Andrey Utkin > 2013/10/1 Stephan Asovski : > > I'm building a server for video encoding and I'm planning to use the > latest > > Xenon e5-2697 v2. > > Can Xeon Phi help me improve ffmpeg encoding performance? > > As far as i googled out, Phi coprocessor occurs logically as separate > network host. So it can be leveraged easily in two ways: > 1) Running generic application straightly on it. You can ssh to the > coprocessor and do whatever you need in its internal Linux. > 2) Using computation distribution technologies like MPI, OpenMP or NUMA. > > But as far as i know, ffmpeg does not leverage any of MPI, OpenMP and NUMA. > > -- > Andrey Utkin > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Wed Oct 2 20:39:36 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 2 Oct 2013 18:39:36 +0000 (UTC) Subject: [FFmpeg-user] IVTC with pullup filter References: Message-ID: Elliott Balsley gmail.com> writes: > This ffmpeg command produces video at 23.98, but with > a pattern of 2 progressive frames followed by 2 > interlaced frames. > > ffmpeg -ss 208 -i NCIS.ts -t 30 -vf "pullup,fps=24000/1001" > -acodec copy -vcodec libx264 pullup.mkv (Complete, uncut console output missing.) Please provide the input sample. Note that I don't think you can combine pullup and a deinterlacer, this only works with fieldmatch iiuc. Carl Eugen From u at pkh.me Wed Oct 2 20:46:40 2013 From: u at pkh.me (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Wed, 2 Oct 2013 20:46:40 +0200 Subject: [FFmpeg-user] IVTC with pullup filter In-Reply-To: References: Message-ID: <20131002184640.GB3676@leki.pkh.me> On Wed, Oct 02, 2013 at 11:28:56AM -0700, Elliott Balsley wrote: [...] > On the other hand, I can achieve perfect results with this simple AviSynth script: > > MPEG2Source("NCIS.d2v", cpu=0) > TFM().TDecimate() > Trim(5000,5300) > TFM/TDecimate are ported to FFmpeg as fieldmatch/decimate; you may want to try. [...] -- 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 cehoyos at ag.or.at Wed Oct 2 20:52:56 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 2 Oct 2013 18:52:56 +0000 (UTC) Subject: [FFmpeg-user] sei type truncated References: <524B3D90.4040609@interia.eu> Message-ID: mike interia.eu> writes: > I do not know if it is a bug or i am doing something wrong. > I recorded with Open Broadcaster Software with enabled > quick sync some game in 1920x1080 resolution , video is > playable in vlc , then was trying convert with ffmpeg to > huffyuv , video is playable , but image is > broken , there are only multicolor lines on whole image. I cannot reproduce the multicolor lines, which application did you use to see them? I also see the warnings, I don't know if showing them is a bug. Carl Eugen $ md5sum obs-out.mp4 0558c5d6927abe421719d00a5d0bdffd obs-out.mp4 $ ffmpeg -i obs-out.mp4 -vcodec huffyuv out.avi ffmpeg version N-56801-g286beeb Copyright (c) 2000-2013 the FFmpeg developers built on Oct 2 2013 11:26:12 with gcc 4.3 (SUSE Linux) configuration: --enable-gpl libavutil 52. 46.100 / 52. 46.100 libavcodec 55. 33.101 / 55. 33.101 libavformat 55. 18.104 / 55. 18.104 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 88.100 / 3. 88.100 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 [h264 @ 0x91058e0] SEI type 5 truncated at 1126 [h264 @ 0x91058e0] SEI type 0 truncated at 49 [h264 @ 0x91058e0] SEI type 1 truncated at 53 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'obs-out.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 2013-10-01 21:09:44 encoder : Open Broadcaster Software v0.571b Duration: 00:00:04.10, start: 0.000000, bitrate: 65973 kb/s Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 3 kb/s (default) Metadata: creation_time : 2013-10-01 21:09:44 handler_name : Sound Media Handler Stream #0:1(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 66493 kb/s, 30 fps, 30 tbr, 1k tbn, 60 tbc (default) Metadata: creation_time : 2013-10-01 21:09:44 handler_name : Video Media Handler [huffyuv @ 0x9108d00] using huffyuv 2.2.0 or newer interlacing flag Output #0, avi, to 'out.avi': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 ISFT : Lavf55.18.104 Stream #0:0(und): Video: huffyuv (HFYU / 0x55594648), yuv422p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 30 tbn, 30 tbc (default) Metadata: creation_time : 2013-10-01 21:09:44 handler_name : Video Media Handler Stream #0:1(eng): Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, stereo, fltp, 192 kb/s (default) Metadata: creation_time : 2013-10-01 21:09:44 handler_name : Sound Media Handler Stream mapping: Stream #0:1 -> #0:0 (h264 -> huffyuv) Stream #0:0 -> #0:1 (aac -> ac3) Press [q] to stop, [?] for help [h264 @ 0x91058e0] SEI type 5 truncated at 1126 [h264 @ 0x91058e0] SEI type 0 truncated at 49 [h264 @ 0x91058e0] SEI type 1 truncated at 53 Last message repeated 2 times frame= 2 fps=0.0 q=0.0 size= 2098kB time=00:00:00.10 bitrate=171833.6kbits/s ... [h264 @ 0x91058e0] SEI type 1 truncated at 53 Last message repeated 1 times frame= 123 fps=3.2 q=0.0 size= 258943kB time=00:00:04.13 bitrate=513207.8kbits/s frame= 123 fps=3.2 q=0.0 Lsize= 265441kB time=00:00:04.13 bitrate=526087.7kbits/s video:265330kB audio:95kB subtitle:0 global headers:0kB muxing overhead 0.005890% out.avi plays fine here. From elliottbalsley at gmail.com Wed Oct 2 21:07:55 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Wed, 2 Oct 2013 12:07:55 -0700 Subject: [FFmpeg-user] IVTC with pullup filter In-Reply-To: <20131002184640.GB3676@leki.pkh.me> References: <20131002184640.GB3676@leki.pkh.me> Message-ID: <2C73C31B-C467-4961-9025-1B7C35F4D623@gmail.com> Here's the full output: http://pastebin.com/kWmqgKRA I tried fieldmatch/decimate, but it made the audio out of sync. I reported it as a bug here: https://trac.ffmpeg.org/ticket/3019 I thought pullup was supposed to be the modern, better way of doing reverse pulldown. Here's an input sample. MPEG2 video with 5.1 AC3 audio. The header might be broken because I trimmed it using dd, but it still works. https://docs.google.com/file/d/0B52QuT8oHvtZcDNFNjdTaXpNdms/ Thanks for your help. On Oct 2, 2013, at 11:46 AM, Cl?ment B?sch wrote: > On Wed, Oct 02, 2013 at 11:28:56AM -0700, Elliott Balsley wrote: > [...] >> On the other hand, I can achieve perfect results with this simple AviSynth script: >> > >> MPEG2Source("NCIS.d2v", cpu=0) >> TFM().TDecimate() >> Trim(5000,5300) >> > > TFM/TDecimate are ported to FFmpeg as fieldmatch/decimate; you may want to > try. > > [...] > > -- > Cl?ment B. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From chronek at interia.eu Wed Oct 2 21:50:30 2013 From: chronek at interia.eu (mike) Date: Wed, 02 Oct 2013 21:50:30 +0200 Subject: [FFmpeg-user] sei type truncated In-Reply-To: References: <524B3D90.4040609@interia.eu> Message-ID: <524C7906.3000309@interia.eu> > I cannot reproduce the multicolor lines, which application did > you use to see them? After you wrote that I tryed open it in virtualdub and file is ok, looks like it is not a ffmpeg bug, but maybe vlc 2.1.0 bug (attached thumbnail) but other video files playing good, and other huffyuv files too, so it is a strange bug Mike -------------- next part -------------- A non-text attachment was scrubbed... Name: screen.jpg Type: image/jpeg Size: 63429 bytes Desc: not available URL: From s0527705277 at gmail.com Wed Oct 2 23:01:52 2013 From: s0527705277 at gmail.com (Stephan Asovski) Date: Thu, 3 Oct 2013 00:01:52 +0300 Subject: [FFmpeg-user] How to remove atoms from MP4? Message-ID: I'm merging two mp4 files for video streaming. the problem is, the merged file contains two atoms, 1 atom from each file and that is a problem when it comes to streaming. the client needs to wait until the first atom loads from file1, than when it's time to play second part from file2 client will wait again for the atom to be loaded. when your atom is small you may not notice this, but when you stream a large video, atom could weight 7-10mb. How can I remove those two atoms from merged file and create a new one? what i've tried: ffmpeg -i file1.mp4 -c copy -bsf:v h264_mp4toannexb -f mpegts intermediate1.ts ffmpeg -i file2.mp4 -c copy -bsf:v h264_mp4toannexb -f mpegts intermediate2.ts ffmpeg -i "concat:intermediate1.ts|intermediate2.ts" -c copy -bsf:a aac_adtstoasc m.file.mp4 MP4Box -add m.file.mp4 -isma mf.file.mp4 mv mf.file.mp4 m.file.mp4 From blacktrash at gmx.net Wed Oct 2 23:32:58 2013 From: blacktrash at gmx.net (Christian Ebert) Date: Wed, 2 Oct 2013 22:32:58 +0100 Subject: [FFmpeg-user] aresample out_channel_layout values In-Reply-To: <20131001152221.GA24454@krille.blacktrash.org> References: <20131001152221.GA24454@krille.blacktrash.org> Message-ID: <20131002213258.GE24454@krille.blacktrash.org> * Christian Ebert on Tuesday, October 01, 2013 at 16:22:21 +0100 > What are the correct values to pass to in/out_channel_layout as > aresample option? > > The aformat filter accepts aformat=channel_layouts=mono, but > aresample=22050:out_channel_layout=mono gives the following error: > > [SWR @ 0x7fe9bb000600] [Eval @ 0x7fff570741d0] Undefined constant or missing '(' in 'mono' > [SWR @ 0x7fe9bb000600] Unable to parse option value "mono" > [AVFilterGraph @ 0x7fe9b9c16140] Error initializing filter 'aresample' with args '22050:ocl=mono' > Error configuring filters. > > I also tried out_channel_layout=FC (taken from ffmpeg -layouts) > with the same error. Attempt to answer my own question: 1c or 2c (from libavutil/channel_layout.h) do not throw an error but do not seem to have any effect. Plain integers 1 or 2 seem to work, but what's the difference then to specifying out_channel_count=1 ? > Finally: What would be the difference between: > > aformat=channel_layouts=stereo > and > aresample=out_channel_layout= Still curious about this one. -- \black\trash movie _COWBOY CANOE COMA_ Ein deutscher Western/A German Western --->> http://www.blacktrash.org/underdogma/ccc.php From elliottbalsley at gmail.com Thu Oct 3 00:21:28 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Wed, 2 Oct 2013 15:21:28 -0700 Subject: [FFmpeg-user] mailing list filter Message-ID: <915C5CC7-ACB0-445C-BA61-A856C8BC29D3@gmail.com> Is there some way to receive only emails in response to my own questions, so my inbox is not bombarded with all messages on the ffmpeg-user list? From lou at lrcd.com Thu Oct 3 00:37:38 2013 From: lou at lrcd.com (Lou) Date: Wed, 2 Oct 2013 14:37:38 -0800 Subject: [FFmpeg-user] mailing list filter In-Reply-To: <915C5CC7-ACB0-445C-BA61-A856C8BC29D3@gmail.com> References: <915C5CC7-ACB0-445C-BA61-A856C8BC29D3@gmail.com> Message-ID: <20131002143738.63e9508a@lrcd.com> On Wed, 2 Oct 2013 15:21:28 -0700 Elliott Balsley wrote: > Is there some way to receive only emails in response to my own > questions, so my inbox is not bombarded with all messages on the > ffmpeg-user list? Not via the Mailman interface, but you can possibly setup some sort of filtering in your mail client, or perhaps Nabble or Gmane may offer an interface or feature more to your liking (I'm not sure; I've never used them). http://ffmpeg-users.933282.n4.nabble.com/ http://dir.gmane.org/gmane.comp.video.ffmpeg.user From jshupert at pps-inc.com Thu Oct 3 00:41:46 2013 From: jshupert at pps-inc.com (Jim Shupert) Date: Wed, 02 Oct 2013 18:41:46 -0400 Subject: [FFmpeg-user] buffer problems with dvd encoding Message-ID: <524CA12A.7010900@pps-inc.com> friends I am having a difficulty with a dvd encode I am coming from a 10bit uncompressed avi I have tried setting -bufsize 4096k which I know is high. I think -bufsize 1835k is more standard it will start ok ffmpeg -y -i /media/data/cap/soc_1.avi -vf "yadif" -vcodec mpeg2video -pix_fmt yuv420p -acodec mp2 -target ntsc-dvd -sample_fmt flt -minrate 1000k -maxrate 6000k -bufsize 4096k -aspect 4:3 /media/SSD-015/SofC/SofC2.mpg ffmpeg version git-2012-09-24-fd63c2f Copyright (c) 2000-2012 the FFmpeg developers built on Sep 24 2012 14:09:13 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1) configuration: --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 libavutil 51. 73.101 / 51. 73.101 libavcodec 54. 58.100 / 54. 58.100 libavformat 54. 28.101 / 54. 28.101 libavdevice 54. 2.101 / 54. 2.101 libavfilter 3. 17.100 / 3. 17.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 [avi @ 0x24d7240] non-interleaved AVI Guessed Channel Layout for Input Stream #0.1 : stereo Input #0, avi, from '/media/data/cap/soc_1.avi': Duration: 00:01:40.26, start: 0.000000, bitrate: 226520 kb/s Stream #0:0: Video: v210 (v210 / 0x30313276), yuv422p10le, 720x486, 29.97 tbr, 29.97 tbn, 29.97 tbc Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s [mpeg2video @ 0x24ff600] Warning min_rate > 0 but min_rate != max_rate isn't recommended! [mpeg2video @ 0x24ff600] impossible bitrate constraints, this will fail Output #0, dvd, to '/media/SSD-015/SofC/SofC2.mpg': Metadata: encoder : Lavf54.28.101 Stream #0:0: Video: mpeg2video, yuv420p, 720x480 [SAR 8:9 DAR 4:3], q=2-31, 6000 kb/s, 90k tbn, 29.97 tbc Stream #0:1: Audio: ac3, 48000 Hz, stereo, flt, 448 kb/s Stream mapping: Stream #0:0 -> #0:0 (v210 -> mpeg2video) Stream #0:1 -> #0:1 (pcm_s16le -> ac3) Press [q] to stop, [?] for help frame= 25 fps=0.0 q=2.0 size= 188kB time=00:00:00.85 bitrate=1793.6kbits/sframe= 51 fps= 50 q=2.0 size= 382kB time=00:00:01.72 bitrate=1816.6kbits/sframe= 76 fps= 49 q=2.0 size= 814kB time=00:00:02.52 bitrate=2643.3kbits/sframe= 98 fps= 48 q=2.0 size= 1538kB time=00:00:03.25 bitrate=3866.4kbits/sframe= 122 fps= 47 q=2.0 size= 2358kB time=00:00:04.09 bitrate=4722.1kbits/sframe= 146 fps= 47 q=2.2 size= 3060kB time=00:00:04.89 bitrate=5125.6kbits/sframe= 167 fps= 47 q=2.5 Lsize= 3680kB time=00:00:05.56 bitrate=5419.4kbits/s but "hit a bad spot" [dvd @ 0x31febc0] buffer underflow i=1 bufi=328 size=1792 [dvd @ 0x31febc0] buffer underflow i=1 bufi=551 size=1792 [dvd @ 0x31febc0] buffer underflow i=1 bufi=774 size=1792 [dvd @ 0x31febc0] buffer underflow i=1 bufi=997 size=1792 [dvd @ 0x31febc0] buffer underflow i=1 bufi=1220 size=1792 [dvd @ 0x31febc0] buffer underflow i=1 bufi=1443 size=1792 lots & lots of that - and then the audio is out of sync it also alwyas happens at the same spot in the file ( ~ 1:26 deep in a 1:40 file ) also a transcode into a mp4 is OK here are some things I have tried ffmpeg -y -i /media/data/cap/soc_1.avi -vf "yadif" -vcodec mpeg2video -pix_fmt yuv420p -acodec mp2 -target ntsc-dvd -sample_fmt flt -minrate 1000k -bufsize 4096k -aspect 4:3 /media/SSD-015/SofC/SofC.mpg ffmpeg -y -i /media/data/cap/soc_1.avi -vf "yadif" -vcodec mpeg2video -pix_fmt yuv420p -acodec mp2 -target ntsc-dvd -sample_fmt flt -minrate 1000k -maxrate 6000k -bufsize 4096k -aspect 4:3 /media/SSD-015/SofC/SofC2.mpg ffmpeg -y -i /media/data/cap/soc_1.avi -vf "yadif" -f dvd -vcodec mpeg2video -pix_fmt yuv420p -acodec mp2 -target ntsc-dvd -sample_fmt flt -b:v 6110k -minrate 2000k -maxrate 8000k -aspect 4:3 /media/SSD-015/SofC/SofC4.mpg ffmpeg -y -i /media/data/cap/soc_1.avi -f dvd -vcodec mpeg2video -pix_fmt yuv420p -b:v 7000k -minrate 1000k -maxrate 8000k -bufsize 1835k -acodec ac3 -b:a 192K -aspect 4:3 /media/SSD-015/SofC/SofC9.mpg ================== also the below did NOT err with a buffer underun ffmpeg -y -i /media/data/cap/soc_1.avi -vcodec mpeg2video -pix_fmt yuv420p -b:v 7000k -acodec ac3 -b:a 192K -aspect 4:3 /media/SSD-015/SofC/SofC10.mpg nor does ffmpeg -y -i /media/data/cap/soc_1.avi /media/SSD-015/SofC/S1.mpg a simple mpg1 =================== so the difference is no -target ntsc-dvd MY Q is what are the "details" of -target ntsc-dvd and or what combo of minrate maxrate datarate bufsize vbv - might resolve this problem again it seems to be -target ntsc-dvd and mpg2video Thanks jS From cehoyos at ag.or.at Thu Oct 3 00:59:25 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 2 Oct 2013 22:59:25 +0000 (UTC) Subject: [FFmpeg-user] buffer problems with dvd encoding References: <524CA12A.7010900@pps-inc.com> Message-ID: Jim Shupert pps-inc.com> writes: > ffmpeg -y -i /media/data/cap/soc_1.avi -vf "yadif" > -vcodec mpeg2video -pix_fmt yuv420p -acodec mp2 > -target ntsc-dvd -sample_fmt flt -minrate > 1000k -maxrate 6000k -bufsize 4096k -aspect 4:3 > /media/SSD-015/SofC/SofC2.mpg Does the following work? ffmpeg -i soc_1.avi -vf yadif -target ntsc-dvd SofC2.mpg Carl Eugen From cehoyos at ag.or.at Thu Oct 3 01:01:04 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 2 Oct 2013 23:01:04 +0000 (UTC) Subject: [FFmpeg-user] How to remove atoms from MP4? References: Message-ID: Stephan Asovski gmail.com> writes: > I'm merging two mp4 files for video streaming. the > problem is, the merged file contains two atoms, 1 atom > from each file and that is a problem when it comes to > streaming. Please provide the failing ffmpeg command line together with the complete, uncut console output. Carl Eugen From s0527705277 at gmail.com Thu Oct 3 01:22:54 2013 From: s0527705277 at gmail.com (Stephan Asovski) Date: Thu, 3 Oct 2013 02:22:54 +0300 Subject: [FFmpeg-user] How to remove atoms from MP4? In-Reply-To: References: Message-ID: it's not failing, it's creating a file with two MOOV atoms. 2013/10/3 Carl Eugen Hoyos > Stephan Asovski gmail.com> writes: > > > I'm merging two mp4 files for video streaming. the > > problem is, the merged file contains two atoms, 1 atom > > from each file and that is a problem when it comes to > > streaming. > > Please provide the failing ffmpeg command line together > with the complete, uncut console output. > > 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 Oct 3 02:03:30 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 3 Oct 2013 00:03:30 +0000 (UTC) Subject: [FFmpeg-user] How to remove atoms from MP4? References: Message-ID: Stephan Asovski gmail.com> writes: > it's not failing, it's creating a file with two MOOV atoms. Please provide the ffmpeg command line creating a file with two moov atoms together with the complete, uncut console output. Please do not top-post here, Carl Eugen From s0527705277 at gmail.com Thu Oct 3 02:10:29 2013 From: s0527705277 at gmail.com (Stephan Asovski) Date: Thu, 3 Oct 2013 03:10:29 +0300 Subject: [FFmpeg-user] How to remove atoms from MP4? In-Reply-To: References: Message-ID: it's in my first post ffmpeg -i file1.mp4 -c copy -bsf:v h264_mp4toannexb -f mpegts intermediate1.ts ffmpeg -i file2.mp4 -c copy -bsf:v h264_mp4toannexb -f mpegts intermediate2.ts ffmpeg -i "concat:intermediate1.ts|intermediate2.ts" -c copy -bsf:a aac_adtstoasc m.file.mp4 MP4Box -add m.file.mp4 -isma mf.file.mp4 mv mf.file.mp4 m.file.mp4 2013/10/3 Carl Eugen Hoyos > Stephan Asovski gmail.com> writes: > > > it's not failing, it's creating a file with two MOOV atoms. > > Please provide the ffmpeg command line creating a file with > two moov atoms together with the complete, uncut console > output. > > Please do not top-post here, Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From stefasab at gmail.com Thu Oct 3 09:14:33 2013 From: stefasab at gmail.com (Stefano Sabatini) Date: Thu, 3 Oct 2013 09:14:33 +0200 Subject: [FFmpeg-user] aresample out_channel_layout values In-Reply-To: <20131002213258.GE24454@krille.blacktrash.org> References: <20131001152221.GA24454@krille.blacktrash.org> <20131002213258.GE24454@krille.blacktrash.org> Message-ID: <20131003071433.GG21336@barisone> On date Wednesday 2013-10-02 22:32:58 +0100, Christian Ebert wrote: > * Christian Ebert on Tuesday, October 01, 2013 at 16:22:21 +0100 > > What are the correct values to pass to in/out_channel_layout as > > aresample option? > > > > The aformat filter accepts aformat=channel_layouts=mono, but > > aresample=22050:out_channel_layout=mono gives the following error: > > > > [SWR @ 0x7fe9bb000600] [Eval @ 0x7fff570741d0] Undefined constant or missing '(' in 'mono' > > [SWR @ 0x7fe9bb000600] Unable to parse option value "mono" > > [AVFilterGraph @ 0x7fe9b9c16140] Error initializing filter 'aresample' with args '22050:ocl=mono' > > Error configuring filters. > > > > I also tried out_channel_layout=FC (taken from ffmpeg -layouts) > > with the same error. > > Attempt to answer my own question: > > 1c or 2c (from libavutil/channel_layout.h) do not throw an error > but do not seem to have any effect. > > Plain integers 1 or 2 seem to work, but what's the difference > then to specifying out_channel_count=1 ? > > > Finally: What would be the difference between: > > > > aformat=channel_layouts=stereo > > and > > aresample=out_channel_layout= > > Still curious about this one. They use different parsers. aresample accepts an integer, while aformat understand the more human-friendly symbolic notation. We should probably extend the option system to accept channel layout string specifications. From senthil at real-image.com Thu Oct 3 14:19:37 2013 From: senthil at real-image.com (SK Cinema) Date: Thu, 3 Oct 2013 05:19:37 -0700 (PDT) Subject: [FFmpeg-user] xyz to rgb conversion In-Reply-To: References: <522B9A71.5090209@googlemail.com> <522E5387.4000503@googlemail.com> <522F3F4F.5040307@googlemail.com> Message-ID: <1380802777422-4661599.post@n4.nabble.com> >The FFmpeg prores encoder does not care about >colourinformation, RGB colourspace or bits. >It needs input data in YUVx4xxP10 (pix_fmts defined >within FFmpeg), nothing else can be read. If your >input data is in another format, ffmpeg (the >application) will automatically insert a conversion >filter (the scale filter). I too am in need of raw XYZ output from a DCI MXF and I tested output to a TIFF file since that would allow for XYZ colour space without the need to convert to Rec.709 RGB. However, the output TIF seems to always be RGB Rec.709. Is there any way to disable the automatic conversion from XYZ to RGB when decoding JPEG2000? The command line I used was: ffmpeg -i videofile.MXF -pix_fmt xyz12le -f image2 -vframes 1 output.tiff The output.tiff is in RGB Rec.709. I used the Sep 18 ffmpeg build. Thanks anyone that can help! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/xyz-to-rgb-conversion-tp4661196p4661599.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From joolzg at btinternet.com Thu Oct 3 15:16:35 2013 From: joolzg at btinternet.com (JULIAN GARDNER) Date: Thu, 3 Oct 2013 14:16:35 +0100 (BST) Subject: [FFmpeg-user] Help with a damaged file Message-ID: <1380806195.10320.YahooMailNeo@web87801.mail.ir2.yahoo.com> I am trying to convert a series of .TS files to .MP4 and 99% are working but a few have PES errors. Now when playing using ffplay i get output like this [mpegts @ 0xb1701980] PES packet size mismatch15KB sq=??? 0B f=0/0?? [aac @ 0xb1706320] channel element 1.8 is not allocated?? 0B f=0/0?? [aac @ 0xb1706320] Prediction is not allowed in AAC-LC. [aac @ 0xb1706320] More than one AAC RDB per ADTS frame is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented. [aac @ 0xb1706320] Reserved bit set. [aac @ 0xb1706320] channel element 2.10 is not allocated But it plays with a little glitch in the sound Bu the problem is when i transcode, all works upto the glitch but then i get Input #0, mpegts, from 'cu-111981-207.ts': ? Duration: 00:31:00.37, start: 68474.392000, bitrate: 778 kb/s ? Program 1 ??? Metadata: ????? service_name??? : Service01 ????? service_provider: FFmpeg ??? Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 528x288 [SAR 32:33 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc ??? Stream #0:1[0x101](dan): Audio: aac ([15][0][0][0] / 0x000F), 32000 Hz, stereo, fltp, 104 kb/s Output #0, mpegts, to 'a.ts': ? Metadata: ??? encoder???????? : Lavf55.18.104 ??? Stream #0:0: Video: h264 ([27][0][0][0] / 0x001B), yuv420p, 528x288 [SAR 32:33 DAR 16:9], q=2-31, 25 fps, 90k tbn, 25 tbc ??? Stream #0:1(dan): Audio: aac ([15][0][0][0] / 0x000F), 32000 Hz, stereo, 104 kb/s Stream mapping: ? Stream #0:0 -> #0:0 (copy) ? Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help [mpegts @ 0xaddae00] PES packet size mismatch ??? Last message repeated 1 times [mpegts @ 0xaddae00] PES packet size mismatchtime=00:07:10.00 bitrate= 766.8kbits/s??? [mpegts @ 0xaddff40] AAC bitstream not in ADTS format and extradata missing av_interleaved_write_frame(): Invalid data found when processing input and thats it, is there a commandline that can make the transcode ignore this and carry on! joolz From renaux.jacky at orange.fr Thu Oct 3 15:20:29 2013 From: renaux.jacky at orange.fr (jacky) Date: Thu, 03 Oct 2013 15:20:29 +0200 Subject: [FFmpeg-user] speeding transfert Message-ID: <524D6F1D.3070008@orange.fr> Hello I am a volunter in a senior culture association, responsable for streaming server application and have a question to vid?o gourous We have a large DV file (about 4Gb) from each conference (once a week) this dv file is encoded using ffmpeg supplying 2 differents flv files (one 400kb/s second 600kb/s) We all are senior this means which we are working from home and we must transfert from the team to the admin then from the admin to the server this process is requiring about 7h total download and would like to avoid or to keep smaller (the admin and the video team are separated and can do both tasks) One supposed solution might be to slice a file in several parts et send these to Dropbox (as an exemple) and let the serveur fetching, concatenate and running ffmpeg (a very large file is keeping vpn active and in case it stops we have to restart from start) Do you thing it is possible to slice DV file and concatenate without too much trouble ? it seems it cannot be on flv easily does somebody knows a better solution ? even complex requiring programming thanks to all of you , ffmpeg is really a very powerfull tool regards jacky From jshupert at pps-inc.com Thu Oct 3 18:08:36 2013 From: jshupert at pps-inc.com (Jim Shupert) Date: Thu, 03 Oct 2013 12:08:36 -0400 Subject: [FFmpeg-user] buffer problems with dvd encoding In-Reply-To: References: <524CA12A.7010900@pps-inc.com> Message-ID: <524D9684.4050004@pps-inc.com> On 10/2/2013 6:59 PM, Carl Eugen Hoyos wrote: > Jim Shupert pps-inc.com> writes: > >> ffmpeg -y -i /media/data/cap/soc_1.avi -vf "yadif" >> -vcodec mpeg2video -pix_fmt yuv420p -acodec mp2 >> -target ntsc-dvd -sample_fmt flt -minrate >> 1000k -maxrate 6000k -bufsize 4096k -aspect 4:3 >> /media/SSD-015/SofC/SofC2.mpg > Does the following work? > ffmpeg -i soc_1.avi -vf yadif -target ntsc-dvd SofC2.mpg > > Carl Eugen :: starts OK but hits a 'bad spot at 1:26 of a 1:40 file ffmpeg -y -i /media/data/cap/soc_1.avi -vf yadif -target ntsc-dvd /media/SSD-015/SofC/SofC10e.mpg ffmpeg -y -i /media/data/cap/soc_1.avi -vf yadif -target ntsc-dvd /media/SSD-015/SofC/SofC10e.mpg ffmpeg version git-2012-09-24-fd63c2f Copyright (c) 2000-2012 the FFmpeg developers built on Sep 24 2012 14:09:13 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1) configuration: --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 libavutil 51. 73.101 / 51. 73.101 libavcodec 54. 58.100 / 54. 58.100 libavformat 54. 28.101 / 54. 28.101 libavdevice 54. 2.101 / 54. 2.101 libavfilter 3. 17.100 / 3. 17.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 [avi @ 0x2f2b240] non-interleaved AVI Guessed Channel Layout for Input Stream #0.1 : stereo Input #0, avi, from '/media/data/cap/soc_1.avi': Duration: 00:01:40.26, start: 0.000000, bitrate: 226520 kb/s Stream #0:0: Video: v210 (v210 / 0x30313276), yuv422p10le, 720x486, 29.97 tbr, 29.97 tbn, 29.97 tbc Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s Output #0, dvd, to '/media/SSD-015/SofC/SofC10e.mpg': Metadata: encoder : Lavf54.28.101 Stream #0:0: Video: mpeg2video, yuv420p, 720x480, q=2-31, 6000 kb/s, 90k tbn, 29.97 tbc Stream #0:1: Audio: ac3, 48000 Hz, stereo, flt, 448 kb/s Stream mapping: Stream #0:0 -> #0:0 (v210 -> mpeg2video) Stream #0:1 -> #0:1 (pcm_s16le -> ac3) Press [q] to stop, [?] for help frame= 23 fps=0.0 q=2.0 size= 162kB time=00:00:00.76 bitrate=1740.1kbits/sframe= 50 fps= 49 q=2.0 size= 358kB time=00:00:01.65 bitrate=1768.1kbits/sframe= 75 fps= 49 q=2.0 size= 774kB time=00:00:02.52 bitrate=2513.5kbits/sframe= 99 fps= 48 q=2.0 size= 1562kB time=00:00:03.32 bitrate=3851.1kbits/sframe= 122 fps= 48 q=2.0 size= 2346kB time=00:00:04.09 bitrate=4698.1kbits/sframe= 145 fps= 47 q=2.0 size= 3160kB time=00:00:04.82 bitrate=5363.3kbits/sframe= 165 fps= 46 q=2.0 size= 3808kB time=00:00:05.49 bitrate=5673.2kbits/sframe= 186 fps= 46 q=2.0 size= 4476kB time=00:00:06.20 bitrate=5911.6kbits/sframe= 206 fps= 45 q=2.0 size= 5096kB time=00:00:06.87 bitrate=6072.5kbits/sframe= 220 fps= 43 q=2.0 size= 5524kB time=00:00:07.35 bitrate=6152.9kbits/sframe= 241 fps= 43 q=2.0 size= 6180kB time=00:00:08.05 bitrate=6282.2kbits/sframe= 261 fps= 42 q=2.0 size= 6818kB time=00:00:08.73 bitrate=6397.3kbits/sframe= 282 fps= 42 q=2.0 size= 7494kB time=00:00:09.40 bitrate=6529.1kbits/sframe= 303 fps= 42 q=1.6 size= 8154kB time=00:00:10.10 bitrate=6609.3kbits/sframe= 324 fps= 42 q=2.6 size= 8716kB time=00:00:10.81 bitrate=6604.7kbits/sframe= 345 fps= 42 q=2.4 size= 9282kB time=00:00:11.51 bitrate=6603.6kbits/sframe= 367 fps= 42 q=2.2 size= 9852kB frame= 2512 fps= 44 q=2.0 size= 37922kB time=00:01:23.83 bitrate=3705.6kbits/sframe= 2534 fps= 44 q=2.0 size= 38596kB time=00:01:24.53 bitrate=3740.0kbits/sframe= 2556 fps= 44 q=2.0 size= 39556kB time=00:01:25.27 bitrate=3800.0kbits/sframe= 2577 fps= 44 q=2.0 size= 40400kB time=00:01:26.01 bitrate=3847.9kbits/src buffer underflow [mpeg2video @ 0x2f53600] rc buffer underflow Last message repeated 3 times frame= 2599 fps= 44 q=2.0 size= 41304kB time=00:01:26.71 bitrate=3902.0kbits/src buffer underflow [mpeg2video @ 0x2f53600] rc buffer underflow Last message repeated 6 times frame= 2619 fps= 44 q=2.0 size= 42126kB time=00:01:27.38 bitrate=3949.1kbits/src buffer underflow so not a solution __________________________ but, This seems to have worked ------------note: minRate == maxRate == bRate ffmpeg -y -i /media/data/cap/soc_1.avi -vf yadif -f dvd -vcodec mpeg2video -pix_fmt yuv420p -target ntsc-dvd -b:v 7000k -minrate 7000k -maxrate 7000k -bufsize 4096k -aspect 4:3 /media/SSD-015/SofC/SofC11.mpg - and dvdauthor seems to have not puked on it - the disc is burning as i write this... dvdauthor -x /media/SSD-015/SofC/SofC11.xml DVDAuthor::dvdauthor, version 0.7.0. INFO: default video format is NTSC INFO: dvdauthor creating VTS STAT: Picking VTS 01 STAT: Processing /media/SSD-015/SofC/SofC11.mpg... STAT: VOBU 160 at 80MB, 1 PGCs INFO: Video pts = 0.533 .. 100.800 STAT: VOBU 170 at 85MB, 1 PGCs CHAPTERS: VTS[1/1] 0.000 INFO: Generating VTS with the following video attributes: INFO: MPEG version: mpeg2 INFO: TV standard: ntsc INFO: Aspect ratio: 4:3 INFO: Resolution: 720x480 STAT: fixed 170 VOBUs INFO: dvdauthor creating table of contents INFO: Scanning /media/SSD-015/dvd_product/soc//VIDEO_TS/VTS_01_0.IFO so having minrate == maxrate is needed , maybe, what do you think? I may need to do more testing to see if bufsize is needed. I do bufsize of `high' -bufsize 4096k not the standard -bufsize 1835k ___________ I would love to see where -target ntsc-dvd is fully defined. meaning what are all the details of the various params that are encapsulated in that target. And can I declare them all out in `long form` , meaning declare each - and be able to tweek each. where in the c code can i find that? would love to know more..... It is not in ffmpeg man ( which does have lots of good stuff ) Thanks Much! js From blacktrash at gmx.net Thu Oct 3 18:31:50 2013 From: blacktrash at gmx.net (Christian Ebert) Date: Thu, 3 Oct 2013 17:31:50 +0100 Subject: [FFmpeg-user] aresample out_channel_layout values In-Reply-To: <20131003071433.GG21336@barisone> References: <20131001152221.GA24454@krille.blacktrash.org> <20131002213258.GE24454@krille.blacktrash.org> <20131003071433.GG21336@barisone> Message-ID: <20131003163150.GF24454@krille.blacktrash.org> * Stefano Sabatini on Thursday, October 03, 2013 at 09:14:33 +0200 > On date Wednesday 2013-10-02 22:32:58 +0100, Christian Ebert wrote: >> * Christian Ebert on Tuesday, October 01, 2013 at 16:22:21 +0100 >>> What are the correct values to pass to in/out_channel_layout as >>> aresample option? >> >> Attempt to answer my own question: >> >> 1c or 2c (from libavutil/channel_layout.h) do not throw an error >> but do not seem to have any effect. >> >> Plain integers 1 or 2 seem to work, but what's the difference >> then to specifying out_channel_count=1 ? >> >>> Finally: What would be the difference between: >>> >>> aformat=channel_layouts=stereo >>> and >>> aresample=out_channel_layout= >> >> Still curious about this one. > > They use different parsers. aresample accepts an integer, while > aformat understand the more human-friendly symbolic notation. ok, but how would you denote e.g. aformat=channel_layouts=downmix in an integer? 2? But that's already stereo. In what way does this integer differ from channel count? > We should probably extend the option system to accept channel > layout string specifications. Or document a mapping of those integers to their layout meanings, yes, that would be helpful. At the moment I'm not 100% sure whether out_channel_layout=2 has the same effect as out_channel_count=2 by coincidence. -- \black\trash movie _MORALISK ANSTALT_ "Nix verstanden." --->> http://www.blacktrash.org/underdogma/moraliskanstalt.php From onemda at gmail.com Thu Oct 3 19:11:16 2013 From: onemda at gmail.com (Paul B Mahol) Date: Thu, 3 Oct 2013 17:11:16 +0000 Subject: [FFmpeg-user] aresample out_channel_layout values In-Reply-To: <20131003163150.GF24454@krille.blacktrash.org> References: <20131001152221.GA24454@krille.blacktrash.org> <20131002213258.GE24454@krille.blacktrash.org> <20131003071433.GG21336@barisone> <20131003163150.GF24454@krille.blacktrash.org> Message-ID: On 10/3/13, Christian Ebert wrote: > * Stefano Sabatini on Thursday, October 03, 2013 at 09:14:33 +0200 >> On date Wednesday 2013-10-02 22:32:58 +0100, Christian Ebert wrote: >>> * Christian Ebert on Tuesday, October 01, 2013 at 16:22:21 +0100 >>>> What are the correct values to pass to in/out_channel_layout as >>>> aresample option? >>> >>> Attempt to answer my own question: >>> >>> 1c or 2c (from libavutil/channel_layout.h) do not throw an error >>> but do not seem to have any effect. >>> >>> Plain integers 1 or 2 seem to work, but what's the difference >>> then to specifying out_channel_count=1 ? >>> >>>> Finally: What would be the difference between: >>>> >>>> aformat=channel_layouts=stereo >>>> and >>>> aresample=out_channel_layout= >>> >>> Still curious about this one. >> >> They use different parsers. aresample accepts an integer, while >> aformat understand the more human-friendly symbolic notation. > > ok, but how would you denote e.g. aformat=channel_layouts=downmix > in an integer? 2? But that's already stereo. > > In what way does this integer differ from channel count? > >> We should probably extend the option system to accept channel >> layout string specifications. > > Or document a mapping of those integers to their layout meanings, > yes, that would be helpful. > > At the moment I'm not 100% sure whether out_channel_layout=2 has > the same effect as out_channel_count=2 by coincidence. count is for number of channels, layout is for actual layout, in your case 2 maps to stereo, for actual mapping see: channel_layout_map in libavutil/channel_layout.c So no, you can not use integer to denote 'downmix'. > > -- > \black\trash movie _MORALISK ANSTALT_ > "Nix verstanden." > > --->> http://www.blacktrash.org/underdogma/moraliskanstalt.php > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From jwpassmore at gmail.com Thu Oct 3 19:55:38 2013 From: jwpassmore at gmail.com (john passmore) Date: Thu, 3 Oct 2013 13:55:38 -0400 Subject: [FFmpeg-user] PTS/DTS st:0 invailid dropping when concatenating two WAV files into one mp3. Message-ID: I'm attempting to concatenate two 44.1/16 wav files into a single 128k mp3. The concatenation seems to work, but I'm getting this crazy out put that runs for a few minutes before the process ends. I'm not sure what it means or if I should just ignore it, but I couldn't find any documentation. Any ideas would be appreciated! I'm using the Zeranoe install on a Windows XP machine. ffmpeg -f concat -i temp.txt -ab 128k -metadata date="1993" -metadata COMM="NYPR Archive ID 40996" -metadata URL="http://www.wnyc.org/archives" -metadata title="New York and Company" -metadata copyright="NYPR" -metadata publisher="NYPR" archive_import40996.mp3 ffmpeg version N-56297-g7ac6c63 Copyright (c) 2000-2013 the FFmpeg developers built on Sep 15 2013 18:02:28 with gcc 4.7.3 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --ena ble-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-li bvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 43.100 / 52. 43.100 libavcodec 55. 31.101 / 55. 31.101 libavformat 55. 16.102 / 55. 16.102 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 84.100 / 3. 84.100 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 [concat @ 029b7d60] Estimating duration from bitrate, this may be inaccurate Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, concat, from 'temp.txt': Duration: 00:00:00.00, start: 0.000000, bitrate: 1411 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s *PTS -406750697871134784, next:8258932804 invalid dropping st:0* *DTS -406750697871133760, next:8258956023 st:0 invalid dropping* *PTS -406750697871133760, next:8258956023 invalid dropping st:0* *DTS -406750697871132736, next:8258979242 st:0 invalid dropping* *PTS -406750697871132736, next:8258979242 invalid dropping st:0* *DTS -406750697871131712, next:8259002461 st:0 invalid dropping* *PTS -406750697871131712, next:8259002461 invalid dropping st:0* *DTS -406750697871130688, next:8259025680 st:0 invalid dropping* *PTS -406750697871130688, next:8259025680 invalid dropping st:0* *DTS -406750697871129664, next:8259048899 st:0 invalid dropping* *PTS -406750697871129664, next:8259048899 invalid dropping st:0* *DTS -406750697871128640, next:8259072118 st:0 invalid dropping* *PTS -406750697871128640, next:8259072118 invalid dropping st:0* *DTS -406750697871127616, next:8259095337 st:0 invalid dropping* *PTS -406750697871127616, next:8259095337 invalid dropping st:0* *DTS -406750697871126592, next:8259118556 st:0 invalid dropping* *PTS -406750697871126592, next:8259118556 invalid dropping st:0* *DTS -406750697871125568, next:8259141775 st:0 invalid dropping* *PTS -406750697871125568, next:8259141775 invalid dropping st:0* *DTS -406750697871124544, next:8259164994 st:0 invalid dropping* *PTS -406750697871124544, next:8259164994 invalid dropping st:0* *DTS -406750697871123520, next:8259188213 st:0 invalid dropping* *PTS -406750697871123520, next:8259188213 invalid dropping st:0* *DTS -406750697871122496, next:8259211432 st:0 invalid dropping* *PTS -406750697871122496, next:8259211432 invalid dropping st:0* *DTS -406750697871121472, next:8259234651 st:0 invalid dropping* *PTS -406750697871121472, next:8259234651 invalid dropping st:0* *DTS -406750697871120448, next:8259257870 st:0 invalid dropping* *PTS -406750697871120448, next:8259257870 invalid dropping st:0* *DTS -406750697871119424, next:8259281089 st:0 invalid dropping* *PTS -406750697871119424, next:8259281089 invalid dropping st:0* *DTS -406750697871118400, next:8259304308 st:0 invalid dropping* *PTS -406750697871118400, next:8259304308 invalid dropping st:0* *DTS -406750697871117376, next:8259327527 st:0 invalid dropping* *PTS -406750697871117376, next:8259327527 invalid dropping st:0* *DTS -406750697871116352, next:8259350746 st:0 invalid dropping* *PTS -406750697871116352, next:8259350746 invalid dropping st:0* *DTS -406750697871115328, next:8259373965 st:0 invalid dropping* *PTS -406750697871115328, next:8259373965 invalid dropping st:0* *DTS -406750697871114304, next:8259397184 st:0 invalid dropping* *PTS -406750697871114304, next:8259397184 invalid dropping st:0* *DTS -406750697871113280, next:8259420403 st:0 invalid dropping* *PTS -406750697871113280, next:8259420403 invalid dropping st:0* *DTS -406750697871112256, next:8259443622 st:0 invalid dropping* *PTS -406750697871112256, next:8259443622 invalid dropping st:0* *DTS -406750697871111232, next:8259466841 st:0 invalid dropping* *PTS -406750697871111232, next:8259466841 invalid dropping st:0* *DTS -406750697871110208, next:8259490060 st:0 invalid dropping* *PTS -406750697871110208, next:8259490060 invalid dropping st:0* *DTS -406750697871109184, next:8259513279 st:0 invalid dropping* *PTS -406750697871109184, next:8259513279 invalid dropping st:0* *DTS -406750697871108160, next:8259536498 st:0 invalid dropping* *PTS -406750697871108160, next:8259536498 invalid dropping st:0* *DTS -406750697871107136, next:8259559717 st:0 invalid dropping* *PTS -406750697871107136, next:8259559717 invalid dropping st:0* *DTS -406750697871106112, next:8259582936 st:0 invalid dropping* *PTS -406750697871106112, next:8259582936 invalid dropping st:0* *DTS -406750697871105088, next:8259606155 st:0 invalid dropping* *PTS -406750697871105088, next:8259606155 invalid dropping st:0* *DTS -406750697871104064, next:8259629374 st:0 invalid dropping* *PTS -406750697871104064, next:8259629374 invalid dropping st:0* *DTS -406750697871103040, next:8259652593 st:0 invalid dropping* *PTS -406750697871103040, next:8259652593 invalid dropping st:0* *DTS -406750697871102016, next:8259675812 st:0 invalid dropping* *PTS -406750697871102016, next:8259675812 invalid dropping st:0* *DTS -406750697871100992, next:8259699031 st:0 invalid dropping* *PTS -406750697871100992, next:8259699031 invalid dropping st:0* *DTS -406750697871099968, next:8259722250 st:0 invalid dropping* *PTS -406750697871099968, next:8259722250 invalid dropping st:0* *DTS -406750697871098944, next:8259745469 st:0 invalid dropping* *PTS -406750697871098944, next:8259745469 invalid dropping st:0* *DTS -406750697871097920, next:8259768688 st:0 invalid dropping* *PTS -406750697871097920, next:8259768688 invalid dropping st:0* *DTS -406750697871096896, next:8259791907 st:0 invalid dropping* *PTS -406750697871096896, next:8259791907 invalid dropping st:0* *DTS -406750697871095872, next:8259815126 st:0 invalid dropping* *PTS -406750697871095872, next:8259815126 invalid dropping st:0* *DTS -406750697871094848, next:8259838345 st:0 invalid dropping* *PTS -406750697871094848, next:8259838345 invalid dropping st:0* *DTS -406750697871093824, next:8259861564 st:0 invalid dropping* *PTS -406750697871093824, next:8259861564 invalid dropping st:0* *DTS -406750697871092800, next:8259884783 st:0 invalid dropping* *PTS -406750697871092800, next:8259884783 invalid dropping st:0* *DTS -406750697871091776, next:8259908002 st:0 invalid dropping* *PTS -406750697871091776, next:8259908002 invalid dropping st:0* *DTS -406750697871090752, next:8259931221 st:0 invalid dropping* *PTS -406750697871090752, next:8259931221 invalid dropping st:0* *DTS -406750697871089728, next:8259954440 st:0 invalid dropping* *PTS -406750697871089728, next:8259954440 invalid dropping st:0* *DTS -406750697871088704, next:8259977659 st:0 invalid dropping* *PTS -406750697871088704, next:8259977659 invalid dropping st:0* *DTS -406750697871087680, next:8260000878 st:0 invalid dropping* *PTS -406750697871087680, next:8260000878 invalid dropping st:0* *DTS -406750697871086656, next:8260024097 st:0 invalid dropping* *PTS -406750697871086656, next:8260024097 invalid dropping st:0* *DTS -406750697871085632, next:8260047316 st:0 invalid dropping* *PTS -406750697871085632, next:8260047316 invalid dropping st:0* *DTS -406750697871084608, next:8260070535 st:0 invalid dropping* *PTS -406750697871084608, next:8260070535 invalid dropping st:0* *DTS -406750697871083584, next:8260093754 st:0 invalid dropping* *PTS -406750697871083584, next:8260093754 invalid dropping st:0* *DTS -406750697871082560, next:8260116973 st:0 invalid dropping* *PTS -406750697871082560, next:8260116973 invalid dropping st:0* *DTS -406750697871081536, next:8260140192 st:0 invalid dropping* *PTS -406750697871081536, next:8260140192 invalid dropping st:0* *DTS -406750697871080512, next:8260163411 st:0 invalid dropping* *PTS -406750697871080512, next:8260163411 invalid dropping st:0* *DTS -406750697871079488, next:8260186630 st:0 invalid dropping* *PTS -406750697871079488, next:8260186630 invalid dropping st:0* *DTS -406750697871078464, next:8260209849 st:0 invalid dropping* *PTS -406750697871078464, next:8260209849 invalid dropping st:0* *DTS -406750697871077440, next:8260233068 st:0 invalid dropping* *PTS -406750697871077440, next:8260233068 invalid dropping st:0* *DTS -406750697871076416, next:8260256287 st:0 invalid dropping* *PTS -406750697871076416, next:8260256287 invalid dropping st:0* *DTS -406750697871075392, next:8260279506 st:0 invalid dropping* *PTS -406750697871075392, next:8260279506 invalid dropping st:0* *DTS -406750697871074368, next:8260302725 st:0 invalid dropping* *PTS -406750697871074368, next:8260302725 invalid dropping st:0* *DTS -406750697871073344, next:8260325944 st:0 invalid dropping* *PTS -406750697871073344, next:8260325944 invalid dropping st:0* *DTS -406750697871072320, next:8260349163 st:0 invalid dropping* *PTS -406750697871072320, next:8260349163 invalid dropping st:0* *DTS -406750697871071296, next:8260372382 st:0 invalid dropping* *PTS -406750697871071296, next:8260372382 invalid dropping st:0* *DTS -406750697871070272, next:8260395601 st:0 invalid dropping* *PTS -406750697871070272, next:8260395601 invalid dropping st:0* *DTS -406750697871069248, next:8260418820 st:0 invalid dropping* *PTS -406750697871069248, next:8260418820 invalid dropping st:0* *DTS -406750697871068224, next:8260442039 st:0 invalid dropping* *PTS -406750697871068224, next:8260442039 invalid dropping st:0* *DTS -406750697871067200, next:8260465258 st:0 invalid dropping* *PTS -406750697871067200, next:8260465258 invalid dropping st:0* *DTS -406750697871066176, next:8260488477 st:0 invalid dropping* *PTS -406750697871066176, next:8260488477 invalid dropping st:0* *DTS -406750697871065152, next:8260511696 st:0 invalid dropping* *PTS -406750697871065152, next:8260511696 invalid dropping st:0* *DTS -406750697871064128, next:8260534915 st:0 invalid dropping* *PTS -406750697871064128, next:8260534915 invalid dropping st:0* *DTS -406750697871063104, next:8260558134 st:0 invalid dropping* *PTS -406750697871063104, next:8260558134 invalid dropping st:0* *DTS -406750697871062080, next:8260581353 st:0 invalid dropping* *PTS -406750697871062080, next:8260581353 invalid dropping st:0* *DTS -406750697871061056, next:8260604572 st:0 invalid dropping* *PTS -406750697871061056, next:8260604572 invalid dropping st:0* *DTS -406750697871060032, next:8260627791 st:0 invalid dropping* *PTS -406750697871060032, next:8260627791 invalid dropping st:0* *DTS -406750697871059008, next:8260651010 st:0 invalid dropping* *PTS -406750697871059008, next:8260651010 invalid dropping st:0* *DTS -406750697871057984, next:8260674229 st:0 invalid dropping* *PTS -406750697871057984, next:8260674229 invalid dropping st:0* *DTS -406750697871056960, next:8260697448 st:0 invalid dropping* *PTS -406750697871056960, next:8260697448 invalid dropping st:0* *DTS -406750697871055936, next:8260720667 st:0 invalid dropping* *PTS -406750697871055936, next:8260720667 invalid dropping st:0* *DTS -406750697871054912, next:8260743886 st:0 invalid dropping* *PTS -406750697871054912, next:8260743886 invalid dropping st:0* *DTS -406750697871053888, next:8260767105 st:0 invalid dropping* *PTS -406750697871053888, next:8260767105 invalid dropping st:0* *DTS -406750697871052864, next:8260790324 st:0 invalid dropping* *PTS -406750697871052864, next:8260790324 invalid dropping st:0* *DTS -406750697871051840, next:8260813543 st:0 invalid dropping* *PTS -406750697871051840, next:8260813543 invalid dropping st:0* *DTS -406750697871050816, next:8260836762 st:0 invalid dropping* *PTS -406750697871050816, next:8260836762 invalid dropping st:0* *DTS -406750697871049792, next:8260859981 st:0 invalid dropping* *PTS -406750697871049792, next:8260859981 invalid dropping st:0* *DTS -406750697871048768, next:8260883200 st:0 invalid dropping* *PTS -406750697871048768, next:8260883200 invalid dropping st:0* *DTS -406750697871047744, next:8260906419 st:0 invalid dropping* *PTS -406750697871047744, next:8260906419 invalid dropping st:0* *DTS -406750697871046720, next:8260929638 st:0 invalid dropping* *PTS -406750697871046720, next:8260929638 invalid dropping st:0* *DTS -406750697871045696, next:8260952857 st:0 invalid dropping* *PTS -406750697871045696, next:8260952857 invalid dropping st:0* *DTS -406750697871044672, next:8260976076 st:0 invalid dropping* *PTS -406750697871044672, next:8260976076 invalid dropping st:0* *DTS -406750697871043648, next:8260999295 st:0 invalid dropping* *PTS -406750697871043648, next:8260999295 invalid dropping st:0* *DTS -406750697871042624, next:8261022514 st:0 invalid dropping* *PTS -406750697871042624, next:8261022514 invalid dropping st:0* *DTS -406750697871041600, next:8261045733 st:0 invalid dropping* *PTS -406750697871041600, next:8261045733 invalid dropping st:0* *DTS -406750697871040576, next:8261068952 st:0 invalid dropping* *PTS -406750697871040576, next:8261068952 invalid dropping st:0* *DTS -406750697871039552, next:8261092171 st:0 invalid dropping* *PTS -406750697871039552, next:8261092171 invalid dropping st:0* *DTS -406750697871038528, next:8261115390 st:0 invalid dropping* *PTS -406750697871038528, next:8261115390 invalid dropping st:0* *DTS -406750697871037504, next:8261138609 st:0 invalid dropping* *PTS -406750697871037504, next:8261138609 invalid dropping st:0* *DTS -406750697871036480, next:8261161828 st:0 invalid dropping* *PTS -406750697871036480, next:8261161828 invalid dropping st:0* *DTS -406750697871035456, next:8261185047 st:0 invalid dropping* *PTS -406750697871035456, next:8261185047 invalid dropping st:0* *DTS -406750697871034432, next:8261208266 st:0 invalid dropping* *PTS -406750697871034432, next:8261208266 invalid dropping st:0* *DTS -406750697871033408, next:8261231485 st:0 invalid dropping* *PTS -406750697871033408, next:8261231485 invalid dropping st:0* *DTS -406750697871032384, next:8261254704 st:0 invalid dropping* *PTS -406750697871032384, next:8261254704 invalid dropping st:0* *DTS -406750697871031360, next:8261277923 st:0 invalid dropping* *PTS -406750697871031360, next:8261277923 invalid dropping st:0* *DTS -406750697871030336, next:8261301142 st:0 invalid dropping* *PTS -406750697871030336, next:8261301142 invalid dropping st:0* *DTS -406750697871029312, next:8261324361 st:0 invalid dropping* *PTS -406750697871029312, next:8261324361 invalid dropping st:0* *DTS -406750697871028288, next:8261347580 st:0 invalid dropping* *PTS -406750697871028288, next:8261347580 invalid dropping st:0* *DTS -406750697871027264, next:8261370799 st:0 invalid dropping* *PTS -406750697871027264, next:8261370799 invalid dropping st:0* *DTS -406750697871026240, next:8261394018 st:0 invalid dropping* *PTS -406750697871026240, next:8261394018 invalid dropping st:0* *DTS -406750697871025216, next:8261417237 st:0 invalid dropping* *PTS -406750697871025216, next:8261417237 invalid dropping st:0* *DTS -406750697871024192, next:8261440456 st:0 invalid dropping* *PTS -406750697871024192, next:8261440456 invalid dropping st:0* *DTS -406750697871023168, next:8261463675 st:0 invalid dropping* *PTS -406750697871023168, next:8261463675 invalid dropping st:0* *DTS -406750697871022144, next:8261486894 st:0 invalid dropping* *PTS -406750697871022144, next:8261486894 invalid dropping st:0* *DTS -406750697871021120, next:8261510113 st:0 invalid dropping* *PTS -406750697871021120, next:8261510113 invalid dropping st:0* *DTS -406750697871020096, next:8261533332 st:0 invalid dropping* *PTS -406750697871020096, next:8261533332 invalid dropping st:0* *DTS -406750697871019072, next:8261556551 st:0 invalid dropping* *PTS -406750697871019072, next:8261556551 invalid dropping st:0* *DTS -406750697871018048, next:8261579770 st:0 invalid dropping* *PTS -406750697871018048, next:8261579770 invalid dropping st:0* *DTS -406750697871017024, next:8261602989 st:0 invalid dropping* *PTS -406750697871017024, next:8261602989 invalid dropping st:0* *DTS -406750697871016000, next:8261626208 st:0 invalid dropping* *PTS -406750697871016000, next:8261626208 invalid dropping st:0* *DTS -406750697871014976, next:8261649427 st:0 invalid dropping* *PTS -406750697871014976, next:8261649427 invalid dropping st:0* *DTS -406750697871013952, next:8261672646 st:0 invalid dropping* *PTS -406750697871013952, next:8261672646 invalid dropping st:0* *DTS -406750697871012928, next:8261695865 st:0 invalid dropping* *PTS -406750697871012928, next:8261695865 invalid dropping st:0* *DTS -406750697871011904, next:8261719084 st:0 invalid dropping* *PTS -406750697871011904, next:8261719084 invalid dropping st:0* *DTS -406750697871010880, next:8261742303 st:0 invalid dropping* *PTS -406750697871010880, next:8261742303 invalid dropping st:0* *DTS -406750697871009856, next:8261765522 st:0 invalid dropping* *PTS -406750697871009856, next:8261765522 invalid dropping st:0* *DTS -406750697871008832, next:8261788741 st:0 invalid dropping* *PTS -406750697871008832, next:8261788741 invalid dropping st:0* *DTS -406750697871007808, next:8261811960 st:0 invalid dropping* *PTS -406750697871007808, next:8261811960 invalid dropping st:0* *DTS -406750697871006784, next:8261835179 st:0 invalid dropping* *PTS -406750697871006784, next:8261835179 invalid dropping st:0* *DTS -406750697871005760, next:8261858398 st:0 invalid dropping* *PTS -406750697871005760, next:8261858398 invalid dropping st:0* *DTS -406750697871004736, next:8261881617 st:0 invalid dropping* *PTS -406750697871004736, next:8261881617 invalid dropping st:0* *DTS -406750697871003712, next:8261904836 st:0 invalid dropping* *PTS -406750697871003712, next:8261904836 invalid dropping st:0* *DTS -406750697871002688, next:8261928055 st:0 invalid dropping* *PTS -406750697871002688, next:8261928055 invalid dropping st:0* *DTS -406750697871001664, next:8261951274 st:0 invalid dropping* *PTS -406750697871001664, next:8261951274 invalid dropping st:0* *DTS -406750697871000640, next:8261974493 st:0 invalid dropping* *PTS -406750697871000640, next:8261974493 invalid dropping st:0* *DTS -406750697870999616, next:8261997712 st:0 invalid dropping* *PTS -406750697870999616, next:8261997712 invalid dropping st:0* *DTS -406750697870998592, next:8262020931 st:0 invalid dropping* *PTS -406750697870998592, next:8262020931 invalid dropping st:0* *DTS -406750697870997568, next:8262044150 st:0 invalid dropping* *PTS -406750697870997568, next:8262044150 invalid dropping st:0* *DTS -406750697870996544, next:8262067369 st:0 invalid dropping* *PTS -406750697870996544, next:8262067369 invalid dropping st:0* *DTS -406750697870995520, next:8262090588 st:0 invalid dropping* *PTS -406750697870995520, next:8262090588 invalid dropping st:0* *DTS -406750697870994496, next:8262113807 st:0 invalid dropping* *PTS -406750697870994496, next:8262113807 invalid dropping st:0* *DTS -406750697870993472, next:8262137026 st:0 invalid dropping* *PTS -406750697870993472, next:8262137026 invalid dropping st:0* *DTS -406750697870992448, next:8262160245 st:0 invalid dropping* *PTS -406750697870992448, next:8262160245 invalid dropping st:0* *DTS -406750697870991424, next:8262183464 st:0 invalid dropping* *PTS -406750697870991424, next:8262183464 invalid dropping st:0* *DTS -406750697870990400, next:8262206683 st:0 invalid dropping* *PTS -406750697870990400, next:8262206683 invalid dropping st:0* *DTS -406750697870989376, next:8262229902 st:0 invalid dropping* *PTS -406750697870989376, next:8262229902 invalid dropping st:0* *DTS -406750697870988352, next:8262253121 st:0 invalid dropping* *PTS -406750697870988352, next:8262253121 invalid dropping st:0* *DTS -406750697870987328, next:8262276340 st:0 invalid dropping* *PTS -406750697870987328, next:8262276340 invalid dropping st:0* *DTS -406750697870986304, next:8262299559 st:0 invalid dropping* *PTS -406750697870986304, next:8262299559 invalid dropping st:0* *size= 129102kB time=02:17:42.31 bitrate= 128.0kbits/s* *video:0kB audio:129102kB subtitle:0 global headers:0kB muxing overhead 0.000498%* * * From sheng.peisi.luo at gmail.com Thu Oct 3 02:43:07 2013 From: sheng.peisi.luo at gmail.com (Sheng LUO) Date: Thu, 3 Oct 2013 08:43:07 +0800 Subject: [FFmpeg-user] How can I get code at r20544? Message-ID: Hi all, I am new to ffmpeg, a project of mine is required to use r20544, and I tried to get it. How can I get code at 20544? Best, Sheng From jshupert at pps-inc.com Thu Oct 3 21:07:42 2013 From: jshupert at pps-inc.com (Jim Shupert) Date: Thu, 03 Oct 2013 15:07:42 -0400 Subject: [FFmpeg-user] speeding transfert In-Reply-To: <524D6F1D.3070008@orange.fr> References: <524D6F1D.3070008@orange.fr> Message-ID: <524DC07E.1060201@pps-inc.com> well - this isn't a ffmpeg solution because it sounds like your difficulty is data move/store/access. How about you have your own ftp server This would allow for not chopping up the video into dropBox valid size then just use an ftp client, filezilla etc. so you have dv25 --~ 4 GB .... so that is ~ 20 min? do you really need 25Mbps of d1 411 & pcm audio?? could you deal w a really good mp4 , mov or wmv meaning can your edit/view needs be satisfied with a smaller file? So you might acquire as DV due to real world working means [ cam etc ] but transcode that into something smaller - then send it .then That is > flv & stream. do not send the DV .... no need to chop -send -concat back together - transcode to flv a mp4 @ 5Mbps would be alot smaller and it ends up as a flv @ ~ 600kbps anyhoo On 10/3/2013 9:20 AM, jacky wrote: > Hello > I am a volunter in a senior culture association, responsable for > streaming server application and have a question to vid?o gourous > > We have a large DV file (about 4Gb) from each conference (once a week) > this dv file is encoded using ffmpeg supplying 2 differents flv files > (one 400kb/s > second 600kb/s) > > We all are senior this means which we are working from home and we must > transfert from the team to the admin then from the admin to the server > this process is requiring about 7h total download and would like to > avoid or to > keep smaller (the admin and the video team are separated and can do > both tasks) > > One supposed solution might be to slice a file in several parts et > send these > to Dropbox (as an exemple) and let the serveur fetching, concatenate and > running ffmpeg (a very large file is keeping vpn active and in case it > stops we have to > restart from start) > Do you thing it is possible to slice DV file and concatenate without > too much trouble ? > it seems it cannot be on flv easily > does somebody knows a better solution ? even complex requiring > programming > thanks to all of you , ffmpeg is really a very powerfull tool > > regards > jacky > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From lou at lrcd.com Thu Oct 3 21:11:58 2013 From: lou at lrcd.com (Lou) Date: Thu, 3 Oct 2013 11:11:58 -0800 Subject: [FFmpeg-user] How can I get code at r20544? In-Reply-To: References: Message-ID: <20131003111158.476e6c2e@lrcd.com> On Thu, 3 Oct 2013 08:43:07 +0800 Sheng LUO wrote: > Hi all, > > I am new to ffmpeg, a project of mine is required to use r20544, and I > tried to get it. The obligatory question is: Why exactly do you need 20544? This is old, outdated, and unsupported. From s0527705277 at gmail.com Thu Oct 3 21:21:26 2013 From: s0527705277 at gmail.com (Stephan Asovski) Date: Thu, 3 Oct 2013 22:21:26 +0300 Subject: [FFmpeg-user] How can I get code at r20544? In-Reply-To: <20131003111158.476e6c2e@lrcd.com> References: <20131003111158.476e6c2e@lrcd.com> Message-ID: here's your answer http://stackoverflow.com/questions/12882859/get-ffmpeg-source-code-at-a-specific-revision 2013/10/3 Lou > On Thu, 3 Oct 2013 08:43:07 +0800 > Sheng LUO wrote: > > > Hi all, > > > > I am new to ffmpeg, a project of mine is required to use r20544, and I > > tried to get it. > > The obligatory question is: Why exactly do you need 20544? This is old, > outdated, and unsupported. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From aku.radiaku at gmail.com Thu Oct 3 23:48:12 2013 From: aku.radiaku at gmail.com (radiaku) Date: Fri, 4 Oct 2013 04:48:12 +0700 Subject: [FFmpeg-user] Fwd: ffmpeg dropping framerate when joining image with video? In-Reply-To: References: Message-ID: Hey everyone, when I joining video with image using itsoffset, why ffmpeg is dropping frame? ( my video is 25fps ) and its take forever... here my command: ffmpeg.exe -r 25 -itsoffset 5 -i "C:\\after-earth\After_A.mp4" -r 25 -loop > 1 -i "C:\\after-earth\Art.png" -filter_complex "[1:v] > fade=out:125:25:alpha=1 [intro]; [0:v][intro] overlay [v]" -map "[v]" -map > 0:a -acodec copy -strict experimental -y -threads 1 > "C:\\after-earth\After_A1.mp4" > I using zeranoe build for windows ffmpeg version N-56800-gad8fbdd > built on Oct 1 2013 18:01:59 with gcc 4.8.1 (GCC) > configuration: --enable-gpl --enable-version3 --disable-w32threads > --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r > --enable-gnutls --enable-iconv --enable-libass --enable-libbluray > --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc > --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb > --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus > --enable-librtmp --enable-libschroedinger --enable-libsoxr > --enable-libspeex --enable-libtheora --enable-libtwolame > --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc > --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264 > --enable-libxavs --enable-libxvid --enable-zlib > libavutil 52. 46.100 / 52. 46.100 > libavcodec 55. 33.101 / 55. 33.101 > libavformat 55. 18.104 / 55. 18.104 > libavdevice 55. 3.100 / 55. 3.100 > libavfilter 3. 88.100 / 3. 88.100 > libswscale 2. 5.100 / 2. 5.100 > libswresample 0. 17.103 / 0. 17.103 > libpostproc 52. 3.100 / 52. 3.100 > My logs: ffmpeg version N-56800-gad8fbdd Copyright (c) 2000-2013 the FFmpeg > developers > built on Oct 1 2013 18:01:59 with gcc 4.8.1 (GCC) > configuration: --enable-gpl --enable-version3 --disable-w32threads > --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r > --enable-gnutls --enable-iconv --enable-libass --enable-libbluray > --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc > --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb > --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus > --enable-librtmp --enable-libschroedinger --enable-libsoxr > --enable-libspeex --enable-libtheora --enable-libtwolame > --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc > --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264 > --enable-libxavs --enable-libxvid --enable-zlib > libavutil 52. 46.100 / 52. 46.100 > libavcodec 55. 33.101 / 55. 33.101 > libavformat 55. 18.104 / 55. 18.104 > libavdevice 55. 3.100 / 55. 3.100 > libavfilter 3. 88.100 / 3. 88.100 > libswscale 2. 5.100 / 2. 5.100 > libswresample 0. 17.103 / 0. 17.103 > libpostproc 52. 3.100 / 52. 3.100 > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\\after-earth\After_A.mp4': > Metadata: > major_brand : isom > minor_version : 512 > compatible_brands: isomiso2avc1mp41 > encoder : Lavf55.18.104 > Duration: 00:02:31.15, start: 0.036281, bitrate: 521 kb/s > Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, > 640x360, 404 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default) > Metadata: > handler_name : VideoHandler > Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, > fltp, 128 kb/s (default) > Metadata: > handler_name : SoundHandler > Input #1, image2, from 'C:\\after-earth\Art.png': > Duration: 00:00:00.04, start: 0.000000, bitrate: N/A > Stream #1:0: Video: png, rgba, 640x360, 25 fps, 25 tbr, 25 tbn, 25 tbc > [libx264 @ 0027a000] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 > [libx264 @ 0027a000] profile High, level 3.0 > [libx264 @ 0027a000] 264 - core 138 r2358 9e941d1 - H.264/MPEG-4 AVC codec > - Copyleft 2003-2013 - 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=1 > lookahead_threads=1 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=25 > 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, mp4, to 'C:\\after-earth\After_A1.mp4': > Metadata: > major_brand : isom > minor_version : 512 > compatible_brands: isomiso2avc1mp41 > encoder : Lavf55.18.104 > Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, > 640x360, q=-1--1, 12800 tbn, 25 tbc > Stream #0:1(und): Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, > stereo, 128 kb/s (default) > Metadata: > handler_name : SoundHandler > Stream mapping: > Stream #0:0 (h264) -> overlay:main > Stream #1:0 (png) -> fade > overlay -> Stream #0:0 (libx264) > Stream #0:1 -> #0:1 (copy) > Thank you :) From cehoyos at ag.or.at Fri Oct 4 00:02:21 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 3 Oct 2013 22:02:21 +0000 (UTC) Subject: [FFmpeg-user] Help with a damaged file References: <1380806195.10320.YahooMailNeo@web87801.mail.ir2.yahoo.com> Message-ID: JULIAN GARDNER btinternet.com> writes: > Now when playing using ffplay i get output like this > > [mpegts 0xb1701980] PES packet size mismatch Command line and complete, uncut console output missing. I suspect you can use dd to cut a small part of the original file that allows to reproduce the problem. Carl Eugen From cehoyos at ag.or.at Fri Oct 4 00:03:36 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 3 Oct 2013 22:03:36 +0000 (UTC) Subject: [FFmpeg-user] buffer problems with dvd encoding References: <524CA12A.7010900@pps-inc.com> <524D9684.4050004@pps-inc.com> Message-ID: Jim Shupert pps-inc.com> writes: > :: starts OK but hits a 'bad spot at 1:26 of a 1:40 file Please try if you can extract the problematic part: Perhaps ffmpeg -i input -ss 1:10 -t 20 -codec copy out.avi Carl Eugen From cehoyos at ag.or.at Fri Oct 4 00:14:48 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 3 Oct 2013 22:14:48 +0000 (UTC) Subject: [FFmpeg-user] xyz to rgb conversion References: <522B9A71.5090209@googlemail.com> <522E5387.4000503@googlemail.com> <522F3F4F.5040307@googlemail.com> <1380802777422-4661599.post@n4.nabble.com> Message-ID: SK Cinema real-image.com> writes: > Is there any way to disable the automatic conversion from > XYZ to RGB when decoding JPEG2000? Please allow me to repeat that there is NO automatic conversion (every time) when you decode xyz jpeg2000. Or to say it differently: The automatic conversion that happens has nothing to do with jpeg2000 (but only with the used encoder). > The command line I used was: > ffmpeg -i videofile.MXF -pix_fmt xyz12le -f image2 > -vframes 1 output.tiff (The tiff encoder currently does not accept xyz.) Please provide a tiff sample containing xyz. Carl Eugen From s0527705277 at gmail.com Fri Oct 4 00:35:08 2013 From: s0527705277 at gmail.com (Stephan Asovski) Date: Fri, 4 Oct 2013 01:35:08 +0300 Subject: [FFmpeg-user] How to multiplex two videos with ffmpeg Message-ID: I'm converting videos using the -s option for multiple resulutions and I want to include a short intro at the begining of the video. but for some reason it skips the second `-i` so i'm only getting the intro `Opener_4.mp4` witout the video `hdvd.mkv`. ------------------------------------------------------------------------------------------------------------------- What I've tried: (* i tried with and witout the map option) ------------------------------------------------------------------------------------------------------------------- ffmpeg -i Opener_4.mp4 -i hdvd.mkv -map 0:0 -map 0:1 -s 1836x1080 -map 0:0 -map 0:1 -vf -vcodec libx264 -acodec aac -strict experimental -movflags faststart hdvd_1080p.mp4 -s 1224x720 -map 0:0 -map 0:1 -vf -vcodec libx264 -acodec aac -strict experimental -movflags faststart hdvd_720p.mp4 -s 816x480 -map 0:0 -map 0:1 -vf -vcodec libx264 -acodec aac -strict experimental -movflags faststart hdvd_480p.mp4 -s 612x360 -map 0:0 -map 0:1 -vf -vcodec libx264 -acodec aac -strict experimental -movflags faststart hdvd_360p.mp4 -s 408x240 -map 0:0 -map 0:1 -vf -vcodec libx264 -acodec aac -strict experimental -movflags faststart hdvd_240p.mp4 -s 244x144 -map 0:0 -map 0:1 -vcodec libx264 -acodec aac -strict experimental -movflags faststart hdvd_144p.mp4 ------------------------------------------------------------------------------------------------------------------- the log ------------------------------------------------------------------------------------------------------------------- ffmpeg version 1.2.3 Copyright (c) 2000-2013 the FFmpeg developers built on Sep 5 2013 03:04:34 with gcc 4.7 (Debian 4.7.1-2) configuration: --enable-gpl --enable-libass --enable-libfaac --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libspeex --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-x11grab --enable-libx264 --enable-nonfree --enable-version3 libavutil 52. 18.100 / 52. 18.100 libavcodec 54. 92.100 / 54. 92.100 libavformat 54. 63.104 / 54. 63.104 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 42.103 / 3. 42.103 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Opener_4.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42mp41 creation_time : 2013-03-03 06:57:29 Duration: 00:00:13.35, start: 0.000000, bitrate: 6744 kb/s Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 6611 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc Metadata: creation_time : 2013-03-03 06:57:29 handler_name : ?Mainconcept Video Media Handler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 125 kb/s Metadata: creation_time : 2013-03-03 06:57:29 handler_name : #Mainconcept MP4 Sound Media Handler Input #1, matroska,webm, from 'hdvd2.mkv': Metadata: creation_time : 2007-04-07 03:28:47 Duration: 00:02:01.90, start: 0.000000, bitrate: 19153 kb/s Stream #1:0(eng): Video: vc1 (Advanced) (WVC1 / 0x31435657), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 29.97 tbr, 1k tbn, 59.94 tbc (default) Metadata: title : 1080p VC-1 Stream #1:1(eng): Audio: ac3, 48000 Hz, stereo, fltp, 640 kb/s (default) Metadata: title : Dolby Digital 2.0 640kbps Stream #1:2(eng): Audio: eac3, 48000 Hz, 5.1(side), fltp, 640 kb/s Metadata: title : Dolby Digital Plus 5.1 640kbps [libx264 @ 0x2df64c0] using SAR=160/153 [libx264 @ 0x2df64c0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX [libx264 @ 0x2df64c0] profile High, level 4.0 [libx264 @ 0x2df64c0] 264 - core 125 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - 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=48 lookahead_threads=8 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=25 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 [libx264 @ 0x2d8dd80] using SAR=160/153 [libx264 @ 0x2d8dd80] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX [libx264 @ 0x2d8dd80] profile High, level 3.1 [libx264 @ 0x2d8dd80] 264 - core 125 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - 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=48 lookahead_threads=8 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=25 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 [libx264 @ 0x2de9c00] using SAR=160/153 [libx264 @ 0x2de9c00] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX [libx264 @ 0x2de9c00] profile High, level 3.1 [libx264 @ 0x2de9c00] 264 - core 125 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - 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=48 lookahead_threads=7 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=25 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 [libx264 @ 0x2dee860] using SAR=160/153 [libx264 @ 0x2dee860] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX [libx264 @ 0x2dee860] profile High, level 3.0 [libx264 @ 0x2dee860] 264 - core 125 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - 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=48 lookahead_threads=5 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=25 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 [libx264 @ 0x2df1400] using SAR=160/153 [libx264 @ 0x2df1400] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX [libx264 @ 0x2df1400] profile High, level 1.3 [libx264 @ 0x2df1400] 264 - core 125 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - 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=48 lookahead_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=25 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 [libx264 @ 0x2dcf1e0] using SAR=64/61 [libx264 @ 0x2dcf1e0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX [libx264 @ 0x2dcf1e0] profile High, level 1.2 [libx264 @ 0x2dcf1e0] 264 - core 125 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - 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=48 lookahead_threads=2 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=25 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, mp4, to 'hdvd2.mkv_1080p.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42mp41 encoder : Lavf54.63.104 Stream #0:0(eng): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 1836x1080 [SAR 160:153 DAR 16:9], q=-1--1, 30k tbn, 29.97 tbc Metadata: creation_time : 2013-03-03 06:57:29 handler_name : ?Mainconcept Video Media Handler Stream #0:1(eng): Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, fltp, 128 kb/s Metadata: creation_time : 2013-03-03 06:57:29 handler_name : #Mainconcept MP4 Sound Media Handler Output #1, mp4, to 'hdvd2.mkv_720p.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42mp41 encoder : Lavf54.63.104 Stream #1:0(eng): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 1224x720 [SAR 160:153 DAR 16:9], q=-1--1, 30k tbn, 29.97 tbc Metadata: creation_time : 2013-03-03 06:57:29 handler_name : ?Mainconcept Video Media Handler Stream #1:1(eng): Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, fltp, 128 kb/s Metadata: creation_time : 2013-03-03 06:57:29 handler_name : #Mainconcept MP4 Sound Media Handler Output #2, mp4, to 'hdvd2.mkv_480p.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42mp41 encoder : Lavf54.63.104 Stream #2:0(eng): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 816x480 [SAR 160:153 DAR 16:9], q=-1--1, 30k tbn, 29.97 tbc Metadata: creation_time : 2013-03-03 06:57:29 handler_name : ?Mainconcept Video Media Handler Stream #2:1(eng): Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, fltp, 128 kb/s Metadata: creation_time : 2013-03-03 06:57:29 handler_name : #Mainconcept MP4 Sound Media Handler Output #3, mp4, to 'hdvd2.mkv_360p.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42mp41 encoder : Lavf54.63.104 Stream #3:0(eng): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 612x360 [SAR 160:153 DAR 16:9], q=-1--1, 30k tbn, 29.97 tbc Metadata: creation_time : 2013-03-03 06:57:29 handler_name : ?Mainconcept Video Media Handler Stream #3:1(eng): Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, fltp, 128 kb/s Metadata: creation_time : 2013-03-03 06:57:29 handler_name : #Mainconcept MP4 Sound Media Handler Output #4, mp4, to 'hdvd2.mkv_240p.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42mp41 encoder : Lavf54.63.104 Stream #4:0(eng): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 408x240 [SAR 160:153 DAR 16:9], q=-1--1, 30k tbn, 29.97 tbc Metadata: creation_time : 2013-03-03 06:57:29 handler_name : ?Mainconcept Video Media Handler Stream #4:1(eng): Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, fltp, 128 kb/s Metadata: creation_time : 2013-03-03 06:57:29 handler_name : #Mainconcept MP4 Sound Media Handler Output #5, mp4, to 'hdvd2.mkv_144p.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42mp41 encoder : Lavf54.63.104 Stream #5:0(eng): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 244x144 [SAR 64:61 DAR 16:9], q=-1--1, 30k tbn, 29.97 tbc Metadata: creation_time : 2013-03-03 06:57:29 handler_name : ?Mainconcept Video Media Handler Stream #5:1(eng): Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, fltp, 128 kb/s Metadata: creation_time : 2013-03-03 06:57:29 handler_name : #Mainconcept MP4 Sound Media Handler Stream mapping: Stream #0:0 -> #0:0 (h264 -> libx264) Stream #0:1 -> #0:1 (aac -> aac) Stream #0:0 -> #1:0 (h264 -> libx264) Stream #0:1 -> #1:1 (aac -> aac) Stream #0:0 -> #2:0 (h264 -> libx264) Stream #0:1 -> #2:1 (aac -> aac) Stream #0:0 -> #3:0 (h264 -> libx264) Stream #0:1 -> #3:1 (aac -> aac) Stream #0:0 -> #4:0 (h264 -> libx264) Stream #0:1 -> #4:1 (aac -> aac) Stream #0:0 -> #5:0 (h264 -> libx264) Stream #0:1 -> #5:1 (aac -> aac) Press [q] to stop, [?] for help Starting second pass: moving header on top of the file q=29.0 size= 3630kB time=00:00:13.31 bitrate=2233.6kbits/s dup=6 drop=0 [mp4 @ 0x2d8d4e0] Starting second pass: moving header on top of the file [mp4 @ 0x2de9360] Starting second pass: moving header on top of the file [mp4 @ 0x2dedec0] Starting second pass: moving header on top of the file [mp4 @ 0x2df0ba0] Starting second pass: moving header on top of the file [mp4 @ 0x2dcea60] Starting second pass: moving header on top of the file frame= 401 fps= 12 q=-1.0 Lq=-1.0 q=-1.0 q=-1.0 q=-1.0 q=-1.0 size= 4548kB time=00:00:13.35 bitrate=2789.7kbits/s dup=6 drop=0 video:8523kB audio:1165kB subtitle:0 global headers:0kB muxing overhead -53.057493% [libx264 @ 0x2df64c0] frame I:3 Avg QP:17.44 size: 42261 [libx264 @ 0x2df64c0] frame P:166 Avg QP:22.23 size: 18663 [libx264 @ 0x2df64c0] frame B:232 Avg QP:25.06 size: 5246 [libx264 @ 0x2df64c0] consecutive B-frames: 16.0% 17.5% 9.7% 56.9% [libx264 @ 0x2df64c0] mb I I16..4: 55.8% 38.3% 6.0% [libx264 @ 0x2df64c0] mb P I16..4: 13.8% 10.0% 0.5% P16..4: 40.1% 5.7% 2.9% 0.0% 0.0% skip:26.9% [libx264 @ 0x2df64c0] mb B I16..4: 0.4% 0.2% 0.0% B16..8: 31.4% 1.7% 0.2% direct: 1.4% skip:64.7% L0:39.9% L1:56.8% BI: 3.3% [libx264 @ 0x2df64c0] 8x8 transform intra:40.7% inter:82.6% [libx264 @ 0x2df64c0] coded y,uvDC,uvAC intra: 16.4% 37.0% 3.3% inter: 6.3% 16.0% 0.1% [libx264 @ 0x2df64c0] i16 v,h,dc,p: 37% 26% 7% 30% [libx264 @ 0x2df64c0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 26% 21% 34% 3% 4% 4% 3% 3% 3% [libx264 @ 0x2df64c0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 17% 41% 4% 5% 4% 3% 3% 2% [libx264 @ 0x2df64c0] i8c dc,h,v,p: 61% 21% 14% 3% [libx264 @ 0x2df64c0] Weighted P-Frames: Y:31.3% UV:29.5% [libx264 @ 0x2df64c0] ref P L0: 71.5% 8.9% 15.0% 4.5% 0.0% [libx264 @ 0x2df64c0] ref B L0: 90.9% 7.7% 1.5% [libx264 @ 0x2df64c0] ref B L1: 95.8% 4.2% [libx264 @ 0x2df64c0] kb/s:2655.87 [libx264 @ 0x2d8dd80] frame I:3 Avg QP:17.88 size: 25469 [libx264 @ 0x2d8dd80] frame P:141 Avg QP:22.69 size: 9944 [libx264 @ 0x2d8dd80] frame B:257 Avg QP:25.83 size: 2403 [libx264 @ 0x2d8dd80] consecutive B-frames: 11.5% 8.0% 3.7% 76.8% [libx264 @ 0x2d8dd80] mb I I16..4: 48.3% 42.9% 8.8% [libx264 @ 0x2d8dd80] mb P I16..4: 8.8% 9.5% 0.6% P16..4: 42.2% 8.5% 4.2% 0.0% 0.0% skip:26.1% [libx264 @ 0x2d8dd80] mb B I16..4: 0.2% 0.1% 0.0% B16..8: 30.2% 2.4% 0.4% direct: 1.8% skip:64.8% L0:38.7% L1:55.8% BI: 5.5% [libx264 @ 0x2d8dd80] 8x8 transform intra:49.2% inter:74.2% [libx264 @ 0x2d8dd80] coded y,uvDC,uvAC intra: 23.3% 39.3% 4.8% inter: 6.7% 12.8% 0.2% [libx264 @ 0x2d8dd80] i16 v,h,dc,p: 38% 22% 5% 35% [libx264 @ 0x2d8dd80] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 25% 23% 25% 4% 6% 5% 6% 4% 4% [libx264 @ 0x2d8dd80] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 20% 20% 33% 5% 6% 5% 5% 4% 3% [libx264 @ 0x2d8dd80] i8c dc,h,v,p: 63% 20% 13% 4% [libx264 @ 0x2d8dd80] Weighted P-Frames: Y:28.4% UV:27.0% [libx264 @ 0x2d8dd80] ref P L0: 67.9% 12.2% 14.6% 4.9% 0.4% [libx264 @ 0x2d8dd80] ref B L0: 94.3% 4.8% 0.9% [libx264 @ 0x2d8dd80] ref B L1: 98.2% 1.8% [libx264 @ 0x2d8dd80] kb/s:1253.22 [libx264 @ 0x2de9c00] frame I:3 Avg QP:18.17 size: 15799 [libx264 @ 0x2de9c00] frame P:132 Avg QP:22.99 size: 5465 [libx264 @ 0x2de9c00] frame B:266 Avg QP:26.61 size: 1050 [libx264 @ 0x2de9c00] consecutive B-frames: 10.0% 3.0% 5.2% 81.8% [libx264 @ 0x2de9c00] mb I I16..4: 45.1% 43.3% 11.7% [libx264 @ 0x2de9c00] mb P I16..4: 6.1% 8.5% 0.5% P16..4: 42.7% 12.0% 6.5% 0.0% 0.0% skip:23.8% [libx264 @ 0x2de9c00] mb B I16..4: 0.1% 0.1% 0.0% B16..8: 30.1% 3.0% 0.6% direct: 2.1% skip:64.0% L0:38.0% L1:55.4% BI: 6.6% [libx264 @ 0x2de9c00] 8x8 transform intra:54.3% inter:64.3% [libx264 @ 0x2de9c00] coded y,uvDC,uvAC intra: 30.7% 42.3% 6.9% inter: 7.3% 10.8% 0.5% [libx264 @ 0x2de9c00] i16 v,h,dc,p: 36% 22% 6% 36% [libx264 @ 0x2de9c00] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 21% 24% 4% 6% 6% 7% 4% 5% [libx264 @ 0x2de9c00] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 21% 23% 6% 8% 6% 6% 5% 4% [libx264 @ 0x2de9c00] i8c dc,h,v,p: 63% 20% 13% 5% [libx264 @ 0x2de9c00] Weighted P-Frames: Y:25.0% UV:22.7% [libx264 @ 0x2de9c00] ref P L0: 65.8% 15.4% 13.4% 5.0% 0.4% [libx264 @ 0x2de9c00] ref B L0: 95.2% 4.1% 0.7% [libx264 @ 0x2de9c00] ref B L1: 98.3% 1.7% [libx264 @ 0x2de9c00] kb/s:626.67 [libx264 @ 0x2dee860] frame I:3 Avg QP:18.31 size: 11356 [libx264 @ 0x2dee860] frame P:130 Avg QP:23.11 size: 3716 [libx264 @ 0x2dee860] frame B:268 Avg QP:26.88 size: 592 [libx264 @ 0x2dee860] consecutive B-frames: 9.2% 3.5% 4.5% 82.8% [libx264 @ 0x2dee860] mb I I16..4: 42.9% 43.6% 13.5% [libx264 @ 0x2dee860] mb P I16..4: 4.5% 8.6% 0.7% P16..4: 42.2% 15.0% 7.7% 0.0% 0.0% skip:21.3% [libx264 @ 0x2dee860] mb B I16..4: 0.2% 0.2% 0.0% B16..8: 30.3% 2.7% 0.6% direct: 2.0% skip:64.0% L0:36.6% L1:57.5% BI: 5.9% [libx264 @ 0x2dee860] 8x8 transform intra:58.8% inter:59.5% [libx264 @ 0x2dee860] coded y,uvDC,uvAC intra: 35.8% 47.0% 8.6% inter: 7.8% 10.4% 0.8% [libx264 @ 0x2dee860] i16 v,h,dc,p: 35% 25% 7% 33% [libx264 @ 0x2dee860] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 23% 24% 21% 4% 6% 5% 6% 4% 6% [libx264 @ 0x2dee860] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 20% 30% 18% 5% 7% 6% 6% 4% 4% [libx264 @ 0x2dee860] i8c dc,h,v,p: 59% 22% 13% 5% [libx264 @ 0x2dee860] Weighted P-Frames: Y:21.5% UV:18.5% [libx264 @ 0x2dee860] ref P L0: 64.5% 17.5% 12.8% 5.0% 0.2% [libx264 @ 0x2dee860] ref B L0: 94.1% 4.9% 1.0% [libx264 @ 0x2dee860] ref B L1: 98.0% 2.0% [libx264 @ 0x2dee860] kb/s:404.07 [libx264 @ 0x2df1400] frame I:2 Avg QP:18.75 size: 7873 [libx264 @ 0x2df1400] frame P:143 Avg QP:23.38 size: 1808 [libx264 @ 0x2df1400] frame B:256 Avg QP:27.86 size: 178 [libx264 @ 0x2df1400] consecutive B-frames: 13.5% 3.0% 3.7% 79.8% [libx264 @ 0x2df1400] mb I I16..4: 56.9% 23.1% 20.0% [libx264 @ 0x2df1400] mb P I16..4: 3.2% 6.0% 0.4% P16..4: 41.8% 18.2% 9.7% 0.0% 0.0% skip:20.8% [libx264 @ 0x2df1400] mb B I16..4: 0.1% 0.0% 0.0% B16..8: 30.1% 1.3% 0.3% direct: 1.0% skip:67.1% L0:33.7% L1:63.0% BI: 3.4% [libx264 @ 0x2df1400] 8x8 transform intra:57.2% inter:54.7% [libx264 @ 0x2df1400] coded y,uvDC,uvAC intra: 40.7% 48.7% 11.3% inter: 8.7% 10.8% 1.4% [libx264 @ 0x2df1400] i16 v,h,dc,p: 44% 20% 8% 28% [libx264 @ 0x2df1400] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 15% 26% 20% 5% 7% 6% 8% 5% 7% [libx264 @ 0x2df1400] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 17% 25% 14% 6% 9% 7% 8% 7% 6% [libx264 @ 0x2df1400] i8c dc,h,v,p: 61% 22% 12% 5% [libx264 @ 0x2df1400] Weighted P-Frames: Y:19.6% UV:17.5% [libx264 @ 0x2df1400] ref P L0: 63.5% 21.5% 10.5% 4.4% 0.0% [libx264 @ 0x2df1400] ref B L0: 95.3% 3.8% 0.8% [libx264 @ 0x2df1400] ref B L1: 98.0% 2.0% [libx264 @ 0x2df1400] kb/s:191.33 [libx264 @ 0x2dcf1e0] frame I:3 Avg QP:20.44 size: 3117 [libx264 @ 0x2dcf1e0] frame P:258 Avg QP:23.63 size: 482 [libx264 @ 0x2dcf1e0] frame B:140 Avg QP:28.53 size: 58 [libx264 @ 0x2dcf1e0] consecutive B-frames: 52.1% 3.0% 3.0% 41.9% [libx264 @ 0x2dcf1e0] mb I I16..4: 37.5% 47.2% 15.3% [libx264 @ 0x2dcf1e0] mb P I16..4: 1.3% 2.8% 0.5% P16..4: 34.8% 13.6% 7.5% 0.0% 0.0% skip:39.4% [libx264 @ 0x2dcf1e0] mb B I16..4: 0.1% 0.1% 0.0% B16..8: 23.6% 0.5% 0.1% direct: 0.3% skip:75.4% L0:24.2% L1:73.3% BI: 2.5% [libx264 @ 0x2dcf1e0] 8x8 transform intra:57.7% inter:58.0% [libx264 @ 0x2dcf1e0] coded y,uvDC,uvAC intra: 46.2% 56.9% 18.9% inter: 11.2% 13.8% 2.6% [libx264 @ 0x2dcf1e0] i16 v,h,dc,p: 51% 18% 11% 21% [libx264 @ 0x2dcf1e0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 12% 28% 21% 5% 7% 5% 9% 5% 9% [libx264 @ 0x2dcf1e0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 8% 53% 9% 4% 6% 5% 6% 3% 7% [libx264 @ 0x2dcf1e0] i8c dc,h,v,p: 57% 27% 10% 5% [libx264 @ 0x2dcf1e0] Weighted P-Frames: Y:10.5% UV:8.9% [libx264 @ 0x2dcf1e0] ref P L0: 68.0% 21.4% 7.2% 3.4% 0.0% [libx264 @ 0x2dcf1e0] ref B L0: 90.9% 6.8% 2.3% [libx264 @ 0x2dcf1e0] ref B L1: 97.6% 2.4% [libx264 @ 0x2dcf1e0] kb/s:84.71 real 0m35.185s user 4m51.602s sys 0m30.390s From blacktrash at gmx.net Fri Oct 4 10:29:35 2013 From: blacktrash at gmx.net (Christian Ebert) Date: Fri, 4 Oct 2013 09:29:35 +0100 Subject: [FFmpeg-user] aresample out_channel_layout values In-Reply-To: References: <20131001152221.GA24454@krille.blacktrash.org> <20131002213258.GE24454@krille.blacktrash.org> <20131003071433.GG21336@barisone> <20131003163150.GF24454@krille.blacktrash.org> Message-ID: <20131004082935.GG24454@krille.blacktrash.org> * Paul B Mahol on Thursday, October 03, 2013 at 17:11:16 +0000 > On 10/3/13, Christian Ebert wrote: >> * Stefano Sabatini on Thursday, October 03, 2013 at 09:14:33 +0200 >>> On date Wednesday 2013-10-02 22:32:58 +0100, Christian Ebert wrote: >> ok, but how would you denote e.g. aformat=channel_layouts=downmix >> in an integer? 2? But that's already stereo. >> >> In what way does this integer differ from channel count? >> >>> We should probably extend the option system to accept channel >>> layout string specifications. >> >> Or document a mapping of those integers to their layout meanings, >> yes, that would be helpful. >> >> At the moment I'm not 100% sure whether out_channel_layout=2 has >> the same effect as out_channel_count=2 by coincidence. > > count is for number of channels, > > layout is for actual layout, in your case 2 maps to stereo, for actual > mapping see: > > channel_layout_map in libavutil/channel_layout.c Ah, should've looke there instead of channel_layout.h, duh. > So no, you can not use integer to denote 'downmix'. ok - so, one last question: The value for channels returned by ffprobe is the channel count or the channel layout? -- \black\trash movie _SAME TIME SAME PLACE_ New York, in the summer of 2001 --->> http://www.blacktrash.org/underdogma/stsp.php From stefasab at gmail.com Fri Oct 4 10:35:29 2013 From: stefasab at gmail.com (Stefano Sabatini) Date: Fri, 4 Oct 2013 10:35:29 +0200 Subject: [FFmpeg-user] aresample out_channel_layout values In-Reply-To: <20131004082935.GG24454@krille.blacktrash.org> References: <20131001152221.GA24454@krille.blacktrash.org> <20131002213258.GE24454@krille.blacktrash.org> <20131003071433.GG21336@barisone> <20131003163150.GF24454@krille.blacktrash.org> <20131004082935.GG24454@krille.blacktrash.org> Message-ID: <20131004083529.GA14539@barisone> On date Friday 2013-10-04 09:29:35 +0100, Christian Ebert wrote: > * Paul B Mahol on Thursday, October 03, 2013 at 17:11:16 +0000 [...] > The value for channels returned by ffprobe is the channel count > or the channel layout? It's the number of channels. From joolzg at btinternet.com Fri Oct 4 11:14:26 2013 From: joolzg at btinternet.com (JULIAN GARDNER) Date: Fri, 4 Oct 2013 10:14:26 +0100 (BST) Subject: [FFmpeg-user] Help with a damaged file Message-ID: <1380878066.61437.YahooMailNeo@web87803.mail.ir2.yahoo.com> file uploaded here http://joolzg.myetrayz.net:8080/rapidbox/rapidFileDownload.php?r_id=d645920e395fedad7bbbed0eca3fe2e0 Playpack ffmpeg problem.ts ffplay version N-56801-g286beeb Copyright (c) 2003-2013 the FFmpeg developers ? built on Oct? 2 2013 09:25:25 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) ? configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libx264 --enable-libzvbi --cc='ccache cc' ? libavutil????? 52. 46.100 / 52. 46.100 ? libavcodec???? 55. 33.101 / 55. 33.101 ? libavformat??? 55. 18.104 / 55. 18.104 ? libavdevice??? 55.? 3.100 / 55.? 3.100 ? libavfilter???? 3. 88.100 /? 3. 88.100 ? libswscale????? 2.? 5.100 /? 2.? 5.100 ? libswresample?? 0. 17.103 /? 0. 17.103 ? libpostproc??? 52.? 3.100 / 52.? 3.100 [h264 @ 0xb1705c80] non-existing PPS referencedKB sq=??? 0B f=0/0?? [h264 @ 0xb1705c80] non-existing PPS 0 referenced [h264 @ 0xb1705c80] decode_slice_header error [h264 @ 0xb1705c80] no frame! [h264 @ 0xb1705c80] non-existing PPS referenced [h264 @ 0xb1705c80] non-existing PPS 0 referenced [h264 @ 0xb1705c80] decode_slice_header error [h264 @ 0xb1705c80] no frame! [mpegts @ 0xb1701980] PES packet size mismatch0KB sq=??? 0B f=0/0?? ??? Last message repeated 3 times Input #0, mpegts, from 'problem.ts': ? Duration: 00:00:03.52, start: 69067.800000, bitrate: 854 kb/s ? Program 1 ??? Metadata: ????? service_name??? : Service01 ????? service_provider: FFmpeg ??? Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 528x288 [SAR 32:33 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc ??? Stream #0:1[0x101](dan): Audio: aac ([15][0][0][0] / 0x000F), 32000 Hz, stereo, fltp, 97 kb/s [h264 @ 0xb5e7c620] Missing reference picture, default is 0 [h264 @ 0xb5e7c620] decode_slice_header error [h264 @ 0xb5e7d1c0] Missing reference picture, default is 65554/0?? ??? Last message repeated 1 times [mpegts @ 0xb1701980] PES packet size mismatch16KB sq=??? 0B f=0/0?? [aac @ 0xb1706240] channel element 1.8 is not allocated?? 0B f=0/0?? [aac @ 0xb1706240] Prediction is not allowed in AAC-LC. [aac @ 0xb1706240] More than one AAC RDB per ADTS frame is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented. [aac @ 0xb1706240] Reserved bit set. [aac @ 0xb1706240] channel element 2.10 is not allocated [h264 @ 0xb5e7d1c0] reference picture missing during reorder [h264 @ 0xb5e7d1c0] Missing reference picture, default is 65626 [h264 @ 0xb5e7d620] reference picture missing during reorder [h264 @ 0xb5e7d620] Missing reference picture, default is 65632 [h264 @ 0xb5e7d1c0] mmco: unref short failure [mpegts @ 0xb1701980] PES packet size mismatch46KB sq=??? 0B f=0/0?? [h264 @ 0xb5e7c620] error while decoding MB 9 12, bytestream (-8)0?? [h264 @ 0xb5e7c620] concealing 238 DC, 238 AC, 238 MV errors in P frame [h264 @ 0xb5e7c620] Missing reference picture, default is 0B f=0/0?? [h264 @ 0xb5e7c620] decode_slice_header error [h264 @ 0xb5e7d1c0] Missing reference picture, default is 65532 [h264 @ 0xb5e7d620] Missing reference picture, default is 65532 [h264 @ 0xb5e7c620] reference picture missing during reorder [h264 @ 0xb5e7c620] Missing reference picture, default is 65532 [h264 @ 0xb5e7d620] mmco: unref short failure [mpegts @ 0xb1701980] PES packet size mismatch43KB sq=??? 0B f=0/0?? [h264 @ 0xb5e7c620] error while decoding MB 9 12, bytestream (-8)0?? [h264 @ 0xb5e7c620] concealing 238 DC, 238 AC, 238 MV errors in P frame 69070.48 A-V: -0.727 fd=? 12 aq=??? 7KB vq=??? 0KB sq=??? 0B f=0/0?? Transcode ffmpeg -i problem.ts -vcodec copy -acodec copy -bsf:a aac_adtstoasc -y problem.mp4 ffmpeg version N-56801-g286beeb Copyright (c) 2000-2013 the FFmpeg developers ? built on Oct? 2 2013 09:25:25 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) ? configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libx264 --enable-libzvbi --cc='ccache cc' ? libavutil????? 52. 46.100 / 52. 46.100 ? libavcodec???? 55. 33.101 / 55. 33.101 ? libavformat??? 55. 18.104 / 55. 18.104 ? libavdevice??? 55.? 3.100 / 55.? 3.100 ? libavfilter???? 3. 88.100 /? 3. 88.100 ? libswscale????? 2.? 5.100 /? 2.? 5.100 ? libswresample?? 0. 17.103 /? 0. 17.103 ? libpostproc??? 52.? 3.100 / 52.? 3.100 [h264 @ 0xaeb3b00] non-existing PPS referenced [h264 @ 0xaeb3b00] non-existing PPS 0 referenced [h264 @ 0xaeb3b00] decode_slice_header error [h264 @ 0xaeb3b00] no frame! [h264 @ 0xaeb3b00] non-existing PPS referenced [h264 @ 0xaeb3b00] non-existing PPS 0 referenced [h264 @ 0xaeb3b00] decode_slice_header error [h264 @ 0xaeb3b00] no frame! [mpegts @ 0xaeafd60] PES packet size mismatch ??? Last message repeated 3 times Input #0, mpegts, from 'problem.ts': ? Duration: 00:00:03.52, start: 69067.800000, bitrate: 854 kb/s ? Program 1 ??? Metadata: ????? service_name??? : Service01 ????? service_provider: FFmpeg ??? Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 528x288 [SAR 32:33 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc ??? Stream #0:1[0x101](dan): Audio: aac ([15][0][0][0] / 0x000F), 32000 Hz, stereo, fltp, 97 kb/s Output #0, mp4, to 'problem.mp4': ? Metadata: ??? encoder???????? : Lavf55.18.104 ??? Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 528x288 [SAR 32:33 DAR 16:9], q=2-31, 25 fps, 90k tbn, 90k tbc ??? Stream #0:1(dan): Audio: aac ([64][0][0][0] / 0x0040), 32000 Hz, stereo, 97 kb/s Stream mapping: ? Stream #0:0 -> #0:0 (copy) ? Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help [mpegts @ 0xaeafd60] PES packet size mismatch [NULL @ 0xaeb5ce0] Multiple RDBs per frame with CRC is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented. Failed to open bitstream filter aac_adtstoasc for stream 0 with codec copy: Not yet implemented in FFmpeg, patches welcome [mp4 @ 0xaeb4d60] aac bitstream error [mpegts @ 0xaeafd60] PES packet size mismatch frame=?? 80 fps=0.0 q=-1.0 Lsize=???? 325kB time=00:00:03.44 bitrate= 773.1kbits/s??? video:285kB audio:37kB subtitle:0 global headers:0kB muxing overhead 0.926851% Failure is a 69069 (btw multiply value for a 32 bit random number) Now i think the contact ffmpeg is bogus as the 1st part of the real file convert fine, its when we have this little glitch it messes up joolz From shadowing71 at gmail.com Fri Oct 4 11:19:14 2013 From: shadowing71 at gmail.com (Young Kim) Date: Fri, 4 Oct 2013 02:19:14 -0700 Subject: [FFmpeg-user] flv error Message-ID: Hello, I'm currently attempt to stream to a rtmp server using bmdcapture as the input. However, after a few minutes (or hours), the ffmpeg command I'm using eventually spits out this error: [flv @ 0x2279ba0] Failed to update header with correct duration [flv @ 0x2279ba0] Failed to update header with correct filesize. Here's the command I'm using: bmdcapture -C 1 -A 2 -p 8 -c 2 -V 4 -m 1 -s 16 -F nut -f pipe:1 | ffmpeg -rtbufsize 2147483647 -threads 8 -re -copyts -i - -pix_fmt yuv420p -profile high -ab 256k -vf yadif -vcodec libx264 -maxrate:v 1024k -minrate:v 1024k -bufsize:v 1024k -tune zerolatency -ar 44100 -preset veryfast -acodec libfaac -f flv "rtmp://localhost:1935/live/test live=1" Does anyone have suggestions on where I can start to fix this? Thanks, Young Kim From senthil at real-image.com Fri Oct 4 11:32:59 2013 From: senthil at real-image.com (SK Cinema) Date: Fri, 4 Oct 2013 02:32:59 -0700 (PDT) Subject: [FFmpeg-user] xyz to rgb conversion In-Reply-To: References: <522B9A71.5090209@googlemail.com> <522E5387.4000503@googlemail.com> <522F3F4F.5040307@googlemail.com> <1380802777422-4661599.post@n4.nabble.com> Message-ID: <1380879179953-4661619.post@n4.nabble.com> (The tiff encoder currently does not accept xyz.) Please provide a tiff sample containing xyz. Here's a sample of a tiff containing X'Y'Z'. Google Viewer seems unable to display this tiff correctly so please download and view locally. https://docs.google.com/file/d/0B1U1rZVj1f7pbXdlZkFucjhTQk0/edit?usp=sharing Ironically, I made this tiff with an older ffmpeg build (2013-04-18) which did not yet have the XYZ-RGB conversion! In my understanding, any output that supports RGB 4:4:4 will be perfectly capable of supporting XYZ because XYZ is really just a much wider gamut version of RGB where the color primaries are the absolute values based on which RGB primaries for color spaces such as Rec.709 are defined. I also tried to use a custom 3D LUT using the following option: -vf lut3d=XYZ-sRGB.3dl But again, this is defeated by the XYZ-RGB conversion happening prior to the 3D LUT being applied, which is not the desired result. Thanks for looking into this! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/xyz-to-rgb-conversion-tp4661196p4661619.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From andreas.gumm at googlemail.com Fri Oct 4 12:18:40 2013 From: andreas.gumm at googlemail.com (Andreas Gumm) Date: Fri, 4 Oct 2013 12:18:40 +0200 Subject: [FFmpeg-user] xyz to rgb conversion In-Reply-To: <1380879179953-4661619.post@n4.nabble.com> References: <522B9A71.5090209@googlemail.com> <522E5387.4000503@googlemail.com> <522F3F4F.5040307@googlemail.com> <1380802777422-4661599.post@n4.nabble.com> <1380879179953-4661619.post@n4.nabble.com> Message-ID: May I'm wrong with the idea, but is it not good enough to simply unwrap the JPEG2000 stream to image sequence? Andreas Andreas Gumm ______________________________________ metakraft ? - DVD CD und Web-Produktion Berlin Kiefholzstr. 19 2HH 4OG 12435 Berlin www.metakraft.de andreas.gumm at metakraft.de Mobil: +49 177 / 6 444 754 Tel.: +49 30 / 48 49 29 -39 Fax : +49 30 / 48 49 29 -40 Gesch?ftsinhaber: Leander v. Kraft Diese Nachricht ist ausschlie?lich f?r den im Adressfeld ausgewiesenen Empf?nger bestimmt. Sollten Sie nicht der vorgesehene Adressat sein, so bitten wir um eine kurze Nachricht. Jede unbefugte Weiterleitung oder Fertigung einer Kopie dieser Mitteilung ist unzul?ssig. 2013/10/4 SK Cinema > > (The tiff encoder currently does not accept xyz.) > Please provide a tiff sample containing xyz. > > Here's a sample of a tiff containing X'Y'Z'. Google Viewer seems unable to > display this tiff correctly so please download and view locally. > > https://docs.google.com/file/d/0B1U1rZVj1f7pbXdlZkFucjhTQk0/edit?usp=sharing > > Ironically, I made this tiff with an older ffmpeg build (2013-04-18) which > did not yet have the XYZ-RGB conversion! In my understanding, any output > that supports RGB 4:4:4 will be perfectly capable of supporting XYZ because > XYZ is really just a much wider gamut version of RGB where the color > primaries are the absolute values based on which RGB primaries for color > spaces such as Rec.709 are defined. > > I also tried to use a custom 3D LUT using the following option: > -vf lut3d=XYZ-sRGB.3dl > > But again, this is defeated by the XYZ-RGB conversion happening prior to > the > 3D LUT being applied, which is not the desired result. > > Thanks for looking into this! > > > > > -- > View this message in context: > http://ffmpeg-users.933282.n4.nabble.com/xyz-to-rgb-conversion-tp4661196p4661619.html > Sent from the FFmpeg-users mailing list archive at Nabble.com. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From blacktrash at gmx.net Fri Oct 4 13:24:30 2013 From: blacktrash at gmx.net (Christian Ebert) Date: Fri, 4 Oct 2013 12:24:30 +0100 Subject: [FFmpeg-user] aresample out_channel_layout values In-Reply-To: <20131004083529.GA14539@barisone> References: <20131001152221.GA24454@krille.blacktrash.org> <20131002213258.GE24454@krille.blacktrash.org> <20131003071433.GG21336@barisone> <20131003163150.GF24454@krille.blacktrash.org> <20131004082935.GG24454@krille.blacktrash.org> <20131004083529.GA14539@barisone> Message-ID: <20131004112430.GA982@krille.blacktrash.org> * Stefano Sabatini on Friday, October 04, 2013 at 10:35:29 +0200 > On date Friday 2013-10-04 09:29:35 +0100, Christian Ebert wrote: >> The value for channels returned by ffprobe is the channel count >> or the channel layout? > > It's the number of channels. Thanks. @Carl Eugen - for once mediainfo seems to be more informative than ffprobe: mediainfo --Inform='Audio;%ChannelLayout%' test.mov L R (assuming the info is reliable) ;-) -- Was hei?t hier Dogma, ich bin Underdogma! [ What the hell do you mean dogma, I am underdogma. ] free movies --->>> http://www.blacktrash.org/underdogma http://itunes.apple.com/podcast/underdogma-movies/id363423596 From sjtech9 at gmail.com Fri Oct 4 14:15:26 2013 From: sjtech9 at gmail.com (Sudarshan) Date: Fri, 4 Oct 2013 17:45:26 +0530 Subject: [FFmpeg-user] not able to build libmp3lame.so using command Message-ID: Hi All, I am compiling the libmp3lame, and wanted to creat libmp3lame.so but failed to do it I used following steps from the Centos OS compilation guild by the ffmpeg team. cd ~/ffmpeg_sources curl -L -O http://downloads.sourceforge.net/project/lame/lame/3.99/lame-3.99.5.tar.gz tar xzvf lame-3.99.5.tar.gz cd lame-3.99.5 ./configure --prefix="$HOME/ffmpeg_build" --bindir="$HOME/bin" --enable-shared --enable-static make also I have tried to remove static flag from configuration and compilated but it don't generate libmp3lame.so if you need my log, let me know I will provide it. thanks, Sudarshan From h.reindl at thelounge.net Fri Oct 4 14:31:43 2013 From: h.reindl at thelounge.net (Reindl Harald) Date: Fri, 04 Oct 2013 14:31:43 +0200 Subject: [FFmpeg-user] not able to build libmp3lame.so using command In-Reply-To: References: Message-ID: <524EB52F.90209@thelounge.net> Am 04.10.2013 14:15, schrieb Sudarshan: > Hi All, > I am compiling the libmp3lame, and wanted to creat libmp3lame.so but failed > to do it > I used following steps from the Centos OS compilation guild by the ffmpeg > team. > cd ~/ffmpeg_sources > curl -L -O > http://downloads.sourceforge.net/project/lame/lame/3.99/lame-3.99.5.tar.gz > tar xzvf lame-3.99.5.tar.gz > cd lame-3.99.5 > ./configure --prefix="$HOME/ffmpeg_build" --bindir="$HOME/bin" > --enable-shared --enable-static > make > > also I have tried to remove static flag from configuration and compilated > but it don't generate libmp3lame.so > > if you need my log, let me know I will provide it %build sed -i -e 's/^\(\s*hardcode_libdir_flag_spec\s*=\).*/\1/' configure export CFLAGS="%{optflags} -fPIC -fPIE -ffast-math" export LDFLAGS="-Wl,-z,now -Wl,-z,relro,-z,noexecstack -pie" export SH_LDFLAGS="-Wl,-z,now -Wl,-z,relro,-z,noexecstack -pie" %configure --disable-static --enable-mp3x --enable-mp3rtp --enable-decode-layer1 --enable-expopt=norm %{__make} %{?_smp_mflags} ____________________________________ [builduser at testserver:~]$ rpm -q --filesbypkg lame-devel lame-devel /usr/include/lame lame-devel /usr/include/lame.h lame-devel /usr/include/lame/lame.h lame-devel /usr/lib64/libmp3lame.so -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature URL: From stefasab at gmail.com Fri Oct 4 15:36:23 2013 From: stefasab at gmail.com (Stefano Sabatini) Date: Fri, 4 Oct 2013 15:36:23 +0200 Subject: [FFmpeg-user] aresample out_channel_layout values In-Reply-To: <20131004112430.GA982@krille.blacktrash.org> References: <20131001152221.GA24454@krille.blacktrash.org> <20131002213258.GE24454@krille.blacktrash.org> <20131003071433.GG21336@barisone> <20131003163150.GF24454@krille.blacktrash.org> <20131004082935.GG24454@krille.blacktrash.org> <20131004083529.GA14539@barisone> <20131004112430.GA982@krille.blacktrash.org> Message-ID: <20131004133623.GI14539@barisone> On date Friday 2013-10-04 12:24:30 +0100, Christian Ebert wrote: > * Stefano Sabatini on Friday, October 04, 2013 at 10:35:29 +0200 > > On date Friday 2013-10-04 09:29:35 +0100, Christian Ebert wrote: > >> The value for channels returned by ffprobe is the channel count > >> or the channel layout? > > > > It's the number of channels. > > Thanks. > > @Carl Eugen - for once mediainfo seems to be more informative > than ffprobe: > > mediainfo --Inform='Audio;%ChannelLayout%' test.mov > L R > > (assuming the info is reliable) ffprobe -show_entries stream=channel_layout,channels -select_streams a -of compact slow.flv [...] stream|channels=1|channel_layout=mono You need a recent version. -- FFmpeg = Faithless and Foolish Multipurpose Portable Everlasting God From onemda at gmail.com Fri Oct 4 16:00:30 2013 From: onemda at gmail.com (Paul B Mahol) Date: Fri, 4 Oct 2013 14:00:30 +0000 Subject: [FFmpeg-user] IVTC with pullup filter In-Reply-To: <2C73C31B-C467-4961-9025-1B7C35F4D623@gmail.com> References: <20131002184640.GB3676@leki.pkh.me> <2C73C31B-C467-4961-9025-1B7C35F4D623@gmail.com> Message-ID: On 10/2/13, Elliott Balsley wrote: > Here's the full output: > http://pastebin.com/kWmqgKRA > > I tried fieldmatch/decimate, but it made the audio out of sync. I reported > it as a bug here: > https://trac.ffmpeg.org/ticket/3019 > I thought pullup was supposed to be the modern, better way of doing reverse > pulldown. It was silly bug in pullup filter. The latest master from git should not have this bug. From cehoyos at ag.or.at Fri Oct 4 20:25:18 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 4 Oct 2013 18:25:18 +0000 (UTC) Subject: [FFmpeg-user] Hi, I want ask a quesiton about -vcodec copy, mine doesn't work correct, thanks. References: Message-ID: Wei Gao gmail.com> writes: > ffmpeg -i ./testfile/wwwq.mp4 -vcodec copy > ./testfile/out.h264, ffmpeg > print that > > Output file is empty, nothing was encoded > (check -ss / -t / -frames parameters if used) This should be fixed (by Michael), thank you for the report! Carl Eugen From cehoyos at ag.or.at Fri Oct 4 20:28:15 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 4 Oct 2013 18:28:15 +0000 (UTC) Subject: [FFmpeg-user] How to multiplex two videos with ffmpeg References: Message-ID: Stephan Asovski gmail.com> writes: > I'm converting videos using the -s option for multiple > resulutions and I want to include a short intro at the > begining of the video. but for some reason it skips the > second `-i` (I don't think it gets skipped, but -i file1 -i file2 never meant "please concatenate file1 and file2.) FFmpeg offers three possibilities to concatenate files: http://ffmpeg.org/ffmpeg-all.html#concat-2 http://ffmpeg.org/ffmpeg-all.html#concat-1 http://ffmpeg.org/ffmpeg-all.html#concat-3 I don't think the concat protocol can work for mkv, please test demuxer or filter. Carl Eugen From s0527705277 at gmail.com Fri Oct 4 20:52:34 2013 From: s0527705277 at gmail.com (Stephan Asovski) Date: Fri, 4 Oct 2013 21:52:34 +0300 Subject: [FFmpeg-user] How to multiplex two videos with ffmpeg In-Reply-To: References: Message-ID: joining won't help me here because the second video is uploaded by the client and can be in many different formats (avi, mp4, mpeg, mpg, ts, mp4...) my original thought was to convert both of the videos to mp4 and join them, but than I have problem with two MOOV atoms and it's not suitable for streaming http://superuser.com/questions/653513/how-to-merge-2-mp4-files-with-1-atom 2013/10/4 Carl Eugen Hoyos > Stephan Asovski gmail.com> writes: > > > I'm converting videos using the -s option for multiple > > resulutions and I want to include a short intro at the > > begining of the video. but for some reason it skips the > > second `-i` > > (I don't think it gets skipped, but -i file1 -i file2 > never meant "please concatenate file1 and file2.) > > FFmpeg offers three possibilities to concatenate files: > http://ffmpeg.org/ffmpeg-all.html#concat-2 > http://ffmpeg.org/ffmpeg-all.html#concat-1 > http://ffmpeg.org/ffmpeg-all.html#concat-3 > > I don't think the concat protocol can work > for mkv, please test demuxer or filter. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Fri Oct 4 21:24:03 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 4 Oct 2013 19:24:03 +0000 (UTC) Subject: [FFmpeg-user] Help with a damaged file References: <1380878066.61437.YahooMailNeo@web87803.mail.ir2.yahoo.com> Message-ID: JULIAN GARDNER btinternet.com> writes: > file uploaded here Thank you for the sample! I apparently don't understand your problem. On playback, both ffplay and MPlayer show several warnings after "PES packet size mismatch" (indicating that the following warnings are probably invalid), you maybe here an unintended sound (I don't know if that should be avoided but that is not what you report iiuc) but playback continues. On remuxing, the same warnings are shown and remuxing continues after that (afaict). The warnings are shown after approximately 1.9 seconds, approximately three seconds are remuxed. What do I miss? Probably unrelated: The file you uploaded was created with FFmpeg: How is that possible? I mean isn't this a recording (dvb or similar)? Carl Eugen From joolzg at btinternet.com Fri Oct 4 23:14:49 2013 From: joolzg at btinternet.com (JULIAN GARDNER) Date: Fri, 4 Oct 2013 22:14:49 +0100 (BST) Subject: [FFmpeg-user] Help with a damaged file In-Reply-To: Message-ID: <1380921289.29164.YahooMailAndroidMobile@web87806.mail.ir2.yahoo.com> The 1st pes errors are due to cutting the file to get a nice short sample.? I'll put a longer one on the ftp so you don't get the extra errors. And as for transcoding my git pulled ffmpeg stop with an error at 69069 secs. Yes the file is a recording from the output of a transcoded steam. But the source had little glitches hence me asking of a way of getting past these. I will put the file on the server to a point 10 seconds after the failure,? but 1st I will pull a fresh version from git and test that. Joolz Sent from Yahoo! Mail on Android From cehoyos at ag.or.at Fri Oct 4 23:25:18 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 4 Oct 2013 21:25:18 +0000 (UTC) Subject: [FFmpeg-user] Help with a damaged file References: <1380921289.29164.YahooMailAndroidMobile@web87806.mail.ir2.yahoo.com> Message-ID: JULIAN GARDNER btinternet.com> writes: > The 1st pes errors are due to cutting the file to > get a nice short sample. I meant the second pes error, sorry for being unclear. > I'll put a longer one on the ftp so > you don't get the extra errors. > And as for transcoding my git pulled ffmpeg stop with > an error at 69069 secs. Is this reproducible for you with the file you uploaded or not? > Yes the file is a recording from the output of a > transcoded steam. (I probably misunderstand above sentence...) Why does the uploaded file contains metadata "service_provider: FFmpeg"? This usually means that it is not a recording. Carl Eugen From jay.c.kemper at gmail.com Sat Oct 5 00:20:25 2013 From: jay.c.kemper at gmail.com (jaykemper) Date: Fri, 4 Oct 2013 15:20:25 -0700 (PDT) Subject: [FFmpeg-user] UDP Multicast streaming In-Reply-To: References: Message-ID: <1380925225632-4661632.post@n4.nabble.com> I'm working on something similar. I can get unicast to work like this: On 192.168.5.16: ffmpeg -f x11grab -s 2560x1600 -r 20 -i :0.0+nomouse -s 1280x800 -f mpegts udp://192.168.4.111:12345 In VLC on 192.168.4.111: Open Network Stream->udp://@:12345 What needs to change to multicast over the network? Netmask is 255.255.248.0. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/UDP-Multicast-streaming-tp2339775p4661632.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From renaux.jacky at orange.fr Sat Oct 5 10:05:52 2013 From: renaux.jacky at orange.fr (jacky) Date: Sat, 05 Oct 2013 10:05:52 +0200 Subject: [FFmpeg-user] speeding transfert In-Reply-To: <524DC07E.1060201@pps-inc.com> References: <524D6F1D.3070008@orange.fr> <524DC07E.1060201@pps-inc.com> Message-ID: <524FC860.5010406@orange.fr> Le 03/10/2013 21:07, Jim Shupert a ?crit : > well - this isn't a ffmpeg solution because it sounds like your > difficulty is data move/store/access. > > How about you have your own ftp server > This would allow for not chopping up the video into dropBox valid size > then just use an ftp client, filezilla etc. > > so you have dv25 --~ 4 GB .... so that is ~ 20 min? > > do you really need 25Mbps of d1 411 & pcm audio?? > > could you deal w a really good mp4 , mov or wmv > meaning can your edit/view needs be satisfied with a smaller file? > So you might acquire as DV due to real world working means [ cam etc ] > but transcode that into something smaller - then send it .then That is > > flv & stream. > do not send the DV .... no need to chop -send -concat back together - > transcode to flv > a mp4 @ 5Mbps would be alot smaller and it ends up as a flv @ ~ > 600kbps anyhoo > > On 10/3/2013 9:20 AM, jacky wrote: >> Hello >> I am a volunter in a senior culture association, responsable for >> streaming server application and have a question to vid?o gourous >> >> We have a large DV file (about 4Gb) from each conference (once a week) >> this dv file is encoded using ffmpeg supplying 2 differents flv files >> (one 400kb/s >> second 600kb/s) >> >> We all are senior this means which we are working from home and we must >> transfert from the team to the admin then from the admin to the server >> this process is requiring about 7h total download and would like to >> avoid or to >> keep smaller (the admin and the video team are separated and can do >> both tasks) >> >> One supposed solution might be to slice a file in several parts et >> send these >> to Dropbox (as an exemple) and let the serveur fetching, concatenate and >> running ffmpeg (a very large file is keeping vpn active and in case >> it stops we have to >> restart from start) >> Do you thing it is possible to slice DV file and concatenate without >> too much trouble ? >> it seems it cannot be on flv easily >> does somebody knows a better solution ? even complex requiring >> programming >> thanks to all of you , ffmpeg is really a very powerfull tool >> >> regards >> jacky >> >> _______________________________________________ >> 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 > Many thanks Jim i was quite affraid on making a ftp server (I have a lot of security constraints) but I think it is the best solution and I will move on by the way is there a way to exhange programming tricks related to ffmpeg ? we are several using batch files it will be great to share solved points exemple : In case you run ffmpeg -list_devices true -f dshow -i dummy you'll get webcam model name which might include acents and/or special caracters (mine is video="P?riph?rique vid?o USB") accents cannot be handled very well with dos batch I found a trick which may help the community and ready to share regards jacky From codingpotatolinda at gmail.com Sat Oct 5 19:51:01 2013 From: codingpotatolinda at gmail.com (Li) Date: Sat, 5 Oct 2013 12:51:01 -0500 Subject: [FFmpeg-user] What is a good way to calculate the frame rate of a video file using ffmpeg? In-Reply-To: References: Message-ID: Oh, it does not work for one video file .mp4. It should be 30 fps, but it returns 50 fmps. But all Android Video players can play it correctly, which means the correct information is in the file. On Thu, Sep 26, 2013 at 4:55 PM, Li wrote: > It works! Thanks. > I missed this method. > > > On Thu, Sep 26, 2013 at 8:24 AM, Carl Eugen Hoyos wrote: > >> Li gmail.com> writes: >> >> [...] >> >> Any reason you don't use av_guess_frame_rate() ? >> >> Carl Eugen >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > From harneets at unisysinfo.in Sat Oct 5 13:09:20 2013 From: harneets at unisysinfo.in (harneets at unisysinfo.in) Date: Sat, 05 Oct 2013 16:39:20 +0530 Subject: [FFmpeg-user] Required script for 3GP (h.263) Message-ID: <20458242f61541d885255beef56bd92e@unisysinfo.in> Dear All Team please suggest me for transcode mp4 to 3gp in ffmpeg with h.263 codec requirements are : 1. video codec 4.263 2. video frame rate 25 3. video bitrate - 150 4. resolution - 320x240 5. audio bitrate -64k 6. sample rate 44100 Thanks & Regards Harneet Virk. From dashing.meng at gmail.com Sun Oct 6 03:41:05 2013 From: dashing.meng at gmail.com (littlebat) Date: Sun, 6 Oct 2013 09:41:05 +0800 Subject: [FFmpeg-user] Required script for 3GP (h.263) In-Reply-To: <20458242f61541d885255beef56bd92e@unisysinfo.in> References: <20458242f61541d885255beef56bd92e@unisysinfo.in> Message-ID: <20131006094105.a699978e51d3a24c60115513@gmail.com> On Sat, 05 Oct 2013 16:39:20 +0530 harneets at unisysinfo.in wrote: > > > Dear All Team > > please suggest me for transcode mp4 to 3gp in > ffmpeg with h.263 codec > > requirements are : > > 1. video codec 4.263 > > 2. > video frame rate 25 > > 3. video bitrate - 150 > > 4. resolution - 320x240 h.263 can't support 320x240 > > > 5. audio bitrate -64k > > 6. sample rate 44100 > /opt/ffmpeg/bin/ffmpeg -i INPUT -c:v h263 -r 25 -vb 150k -s 352x288 -c:a libfaac -ar 44100 -ab 64k -f 3gp out.3gp From bjorn.ramakers at gmail.com Sun Oct 6 14:56:11 2013 From: bjorn.ramakers at gmail.com (Bjorn Ramakers) Date: Sun, 6 Oct 2013 14:56:11 +0200 Subject: [FFmpeg-user] Capture from Decklink card on Windows 7 and display it (preview) and on demand record it (while still displaying/previewing). Message-ID: Hey, Cant seem to wrap my head around this, seemingly easy, but cant put it all together. I am using Windows 7 to accomplish this. My requirement is to capture video+audio from a Blackmagic Decklink card which delivers 720p from a video source. This capture should be displayed (preview) and on demand I should be able to record (and stop recording) the same stream to a file. My initial though on how to solve this was having ffmpeg capture the video/audio from the Decklink card and create a network stream. I would have vlc player pick up the network stream and display that. Now I would have a second ffmpeg process started and record the same network stream to a file whenever it's needed. This is the command I used to setup the network stream : ffmpeg -f dshow -i video="Decklink Video Capture" -f dshow -i audio="Decklink Audio Capture" -r 60 -vcodec mpeg4 -q 20 -acodec libmp3lame -ab 128k -f mpegts udp://127.0.0.1:6666?pkt_size=188?buffer_size=65535 Using VLC I tried to open this network stream via the address : rtp:// 127.0.0.1:6666 But immediately I get the following error : SDP required: A description in SDP format is required to receive the RTP stream. Note that rtp:// URIs cannot work with dynamic RTP payload format (65). Following some posts I found on forums I tried to add > config.sdp after the command I create the network stream with, but the file never gets any input. My question, would this be a good way to handle the requirements I have? And if so, what should I do to create the correct SDP for the stream? Many thanks in advance. Best Regards, Bjorn. From pb at das-werkstatt.com Sun Oct 6 21:47:29 2013 From: pb at das-werkstatt.com (Peter B.) Date: Sun, 06 Oct 2013 21:47:29 +0200 Subject: [FFmpeg-user] FFV1 and ffprobe Message-ID: <5251BE51.6050909@das-werkstatt.com> Hello. I wanted to ask if there's a way to read the parameters used when encoding FFV1 (version 1 and 3), such as coder/context/slices/slicecrc/etc? Thank you very much, Pb From dave at dericed.com Sun Oct 6 21:51:29 2013 From: dave at dericed.com (Dave Rice) Date: Sun, 6 Oct 2013 15:51:29 -0400 Subject: [FFmpeg-user] FFV1 and ffprobe In-Reply-To: <5251BE51.6050909@das-werkstatt.com> References: <5251BE51.6050909@das-werkstatt.com> Message-ID: On Oct 6, 2013, at 3:47 PM, Peter B. wrote: > I wanted to ask if there's a way to read the parameters used when > encoding FFV1 (version 1 and 3), such as coder/context/slices/slicecrc/etc? As noted here http://trac.ffmpeg.org/ticket/1534 this data is accessible with ffplay -debug 1 . Dave Rice From onemda at gmail.com Sun Oct 6 21:53:00 2013 From: onemda at gmail.com (Paul B Mahol) Date: Sun, 6 Oct 2013 19:53:00 +0000 Subject: [FFmpeg-user] FFV1 and ffprobe In-Reply-To: <5251BE51.6050909@das-werkstatt.com> References: <5251BE51.6050909@das-werkstatt.com> Message-ID: On 10/6/13, Peter B. wrote: > Hello. > > I wanted to ask if there's a way to read the parameters used when > encoding FFV1 (version 1 and 3), such as coder/context/slices/slicecrc/etc? Nope, instead it could be displayed via verbose log messages in encoder/decoder. And/Or flag when enabled would instruct encoder to write those as frame metadata and that frame metadata is stored in container (nut?). > > Thank you very much, > Pb > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Mon Oct 7 00:21:43 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 6 Oct 2013 22:21:43 +0000 (UTC) Subject: [FFmpeg-user] xyz to rgb conversion References: <522B9A71.5090209@googlemail.com> <522E5387.4000503@googlemail.com> <522F3F4F.5040307@googlemail.com> <1380802777422-4661599.post@n4.nabble.com> <1380879179953-4661619.post@n4.nabble.com> Message-ID: SK Cinema real-image.com> writes: > > Please provide a tiff sample containing xyz. > > Here's a sample of a tiff containing X'Y'Z'. Google > Viewer seems unable to display this tiff correctly so > please download and view locally. > https://docs.google.com/file/d/0B1U1rZVj1f7pbXdlZkFucjhTQk0/edit?usp=sharing > > Ironically, I made this tiff with an older ffmpeg build > (2013-04-18) which did not yet have the XYZ-RGB conversion! I may misunderstand but this sounds to me as if no software would be able to recognize this file as XYZ or do I miss something? [...] As said elsewhere, please try to explain what is wrong with the conversion by FFmpeg? I suspect (maybe I am wrong) that no developer know why other coefficients may be needed. (In the sense of: Which usecases need other coefficients and which ones.) > But again, this is defeated by the XYZ-RGB conversion > happening prior to the 3D LUT being applied, which is > not the desired result. This sounds as if it could be fixed but it would need (for me, probably not for other developers) a proper report. Please fix your quoting, Carl Eugen From endofsummer1981 at gmail.com Sun Oct 6 16:36:32 2013 From: endofsummer1981 at gmail.com (lulalala) Date: Sun, 6 Oct 2013 07:36:32 -0700 (PDT) Subject: [FFmpeg-user] Encoding x264 hangs in a high turn over pipeline In-Reply-To: References: <500FD481.8050507@audiomotion.com> <20120725114318.GA25200@phare.normalesup.org> <5010640A.6060502@audiomotion.com> Message-ID: <1381070192796-4661641.post@n4.nabble.com> I am currently using x264.exe directly, and when encoding ac AviSynth script, I encountered the same issue as well. The encoding sometimes works, sometimes hangs. No error message. After the hang the memory usage stops at *300mb* too, and CPU usage falls back to 0. x264 --crf 16 --bframes 0 --colormatrix GBR --qp 0 --preset placebo --range pc --input-range pc --input-csp rgb --output-csp rgb -o a.avi ss.avs I am using the lastest x264 (core:138 r2358 9e941d1). Not sure if this will help ffmpeg -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Encoding-x264-hangs-in-a-high-turn-over-pipeline-tp4652044p4661641.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From timothyhiles at gmail.com Sun Oct 6 05:36:26 2013 From: timothyhiles at gmail.com (Tim Hiles) Date: Sat, 5 Oct 2013 20:36:26 -0700 Subject: [FFmpeg-user] Stereo WMA to right / left mono WMA without reencoding Message-ID: So.. been trying to figure out if this is possible but can't seem to find this through google searches or in past ffmpeg list threads. All i want to do is split a stereo wma file into left and right mono files without reencoding... i.e. no conversion to wav. the reason is, I have all of these wma recordings that are recorded on one side of a stereo file. I'd like to save some space by getting rid of the blank side. converting to wav only creates a huge file, same goes when you convert to flac.. defeating the purpose of all of this in the first place. Here's what I have so far: WS400104.WMA file that is 204,040 KB command: c:\ffmpeg\ffmpeg\bin\ffmpeg.exe -i WS400104.WMA -map_channel 0.0.1:0.1 -acodec copy out.wma output: ffmpeg version N-51683-g9dc88ac Copyright (c) 2000-2013 the FFmpeg developers built on Apr 8 2013 21:19:21 with gcc 4.8.0 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfi g --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-li bopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheo ra --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 25.100 / 52. 25.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 1.100 / 55. 1.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 49.101 / 3. 49.101 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, asf, from 'WS400104.WMA': Duration: 03:36:32.54, start: 0.000000, bitrate: 128 kb/s Stream #0:0: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, stereo, fltp, 128 kb/s File 'out.wma' already exists. Overwrite ? [y/N] y Output #0, asf, to 'out.wma': Metadata: WM/EncodingSettings: Lavf55.1.100 Stream #0:0: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, stereo, 128 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help size= 218588kB time=03:36:32.51 bitrate= 137.8kbits/s video:0kB audio:203047kB subtitle:0 global headers:0kB muxing overhead 7.653751% but as you see.. my output file is a stereo file. So, I tried this: command: c:\ffmpeg\ffmpeg\bin\ffmpeg.exe -i WS400104.WMA -map_channel 0.0.1:0.1 -ac 1 out.wma but got this: output: H:\asis\ansem2013>c:\ffmpeg\ffmpeg\bin\ffmpeg.exe -i WS400104.WMA -map_channel 0.0.1:0.1 -ac 1 out.wma ffmpeg version N-51683-g9dc88ac Copyright (c) 2000-2013 the FFmpeg developers built on Apr 8 2013 21:19:21 with gcc 4.8.0 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfi g --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-li bopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheo ra --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 25.100 / 52. 25.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 1.100 / 55. 1.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 49.101 / 3. 49.101 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, asf, from 'WS400104.WMA': Duration: 03:36:32.54, start: 0.000000, bitrate: 128 kb/s Stream #0:0: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, stereo, fltp, 128 kb/s File 'out.wma' already exists. Overwrite ? [y/N] y Output #0, asf, to 'out.wma': Metadata: WM/EncodingSettings: Lavf55.1.100 Stream #0:0: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, mono, fltp, 128 kb/s Stream mapping: Stream #0:0 -> #0:0 (wmav2 -> wmav2) Press [q] to stop, [?] for help size= 218572kB time=03:36:32.51 bitrate= 137.8kbits/s video:0kB audio:202999kB subtitle:0 global headers:0kB muxing overhead 7.671857% which gives me a mono wma that is now bigger in size than the stereo wma I had to start out with. and also.. my bitrate is different (ffmpeg default?) Would love and appreciate your help. Tim From dashing.meng at gmail.com Mon Oct 7 07:23:39 2013 From: dashing.meng at gmail.com (littlebat) Date: Mon, 7 Oct 2013 13:23:39 +0800 Subject: [FFmpeg-user] Stereo WMA to right / left mono WMA without reencoding In-Reply-To: References: Message-ID: <20131007132339.a8b132a102d19e95f112edd0@gmail.com> On Sat, 5 Oct 2013 20:36:26 -0700 Tim Hiles wrote: > So.. been trying to figure out if this is possible but can't seem to > find this through google searches or in past ffmpeg list threads. > All i want to do is split a stereo wma file into left and right mono > files without reencoding... i.e. no conversion to wav. the reason > is, I have all of these wma recordings that are recorded on one side > of a stereo file. I'd like to save some space by getting rid of the > blank side. converting to wav only creates a huge file, same goes > when you convert to flac.. defeating the purpose of all of this in > the first place. > > > > Here's what I have so far: > > WS400104.WMA file that is 204,040 KB > > command: > > c:\ffmpeg\ffmpeg\bin\ffmpeg.exe -i WS400104.WMA -map_channel 0.0.1:0.1 > -acodec copy out.wma > > output: > > ffmpeg version N-51683-g9dc88ac Copyright (c) 2000-2013 the FFmpeg > developers > built on Apr 8 2013 21:19:21 with gcc 4.8.0 (GCC) > configuration: --enable-gpl --enable-version3 --disable-w32threads > --enable-avisynth --enable-bzlib --enable-fontconfi > g --enable-frei0r --enable-gnutls --enable-iconv --enable-libass > --enable-libbluray --enable-libcaca --enable-libfreetyp > e --enable-libgsm --enable-libilbc --enable-libmp3lame > --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-li > bopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger > --enable-libsoxr --enable-libspeex --enable-libtheo > ra --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc > --enable-libvorbis --enable-libvpx --enable-libx264 > --enable-libxavs --enable-libxvid --enable-zlib > libavutil 52. 25.100 / 52. 25.100 > libavcodec 55. 2.100 / 55. 2.100 > libavformat 55. 1.100 / 55. 1.100 > libavdevice 55. 0.100 / 55. 0.100 > libavfilter 3. 49.101 / 3. 49.101 > libswscale 2. 2.100 / 2. 2.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 2.100 / 52. 2.100 > Guessed Channel Layout for Input Stream #0.0 : stereo > Input #0, asf, from 'WS400104.WMA': > Duration: 03:36:32.54, start: 0.000000, bitrate: 128 kb/s > Stream #0:0: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, stereo, > fltp, 128 kb/s > File 'out.wma' already exists. Overwrite ? [y/N] y > Output #0, asf, to 'out.wma': > Metadata: > WM/EncodingSettings: Lavf55.1.100 > Stream #0:0: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, > stereo, 128 kb/s > Stream mapping: > Stream #0:0 -> #0:0 (copy) > Press [q] to stop, [?] for help > size= 218588kB time=03:36:32.51 bitrate= 137.8kbits/s > video:0kB audio:203047kB subtitle:0 global headers:0kB muxing overhead > 7.653751% > > but as you see.. my output file is a stereo file. So, I tried this: > > command: > > c:\ffmpeg\ffmpeg\bin\ffmpeg.exe -i WS400104.WMA -map_channel > 0.0.1:0.1 -ac 1 out.wma > > but got this: > > output: > > H:\asis\ansem2013>c:\ffmpeg\ffmpeg\bin\ffmpeg.exe -i WS400104.WMA > -map_channel 0.0.1:0.1 -ac 1 out.wma > ffmpeg version N-51683-g9dc88ac Copyright (c) 2000-2013 the FFmpeg > developers > built on Apr 8 2013 21:19:21 with gcc 4.8.0 (GCC) > configuration: --enable-gpl --enable-version3 --disable-w32threads > --enable-avisynth --enable-bzlib --enable-fontconfi > g --enable-frei0r --enable-gnutls --enable-iconv --enable-libass > --enable-libbluray --enable-libcaca --enable-libfreetyp > e --enable-libgsm --enable-libilbc --enable-libmp3lame > --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-li > bopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger > --enable-libsoxr --enable-libspeex --enable-libtheo > ra --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc > --enable-libvorbis --enable-libvpx --enable-libx264 > --enable-libxavs --enable-libxvid --enable-zlib > libavutil 52. 25.100 / 52. 25.100 > libavcodec 55. 2.100 / 55. 2.100 > libavformat 55. 1.100 / 55. 1.100 > libavdevice 55. 0.100 / 55. 0.100 > libavfilter 3. 49.101 / 3. 49.101 > libswscale 2. 2.100 / 2. 2.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 2.100 / 52. 2.100 > Guessed Channel Layout for Input Stream #0.0 : stereo > Input #0, asf, from 'WS400104.WMA': > Duration: 03:36:32.54, start: 0.000000, bitrate: 128 kb/s > Stream #0:0: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, stereo, > fltp, 128 kb/s > File 'out.wma' already exists. Overwrite ? [y/N] y > Output #0, asf, to 'out.wma': > Metadata: > WM/EncodingSettings: Lavf55.1.100 > Stream #0:0: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, mono, > fltp, 128 kb/s > Stream mapping: > Stream #0:0 -> #0:0 (wmav2 -> wmav2) > Press [q] to stop, [?] for help > size= 218572kB time=03:36:32.51 bitrate= 137.8kbits/s > video:0kB audio:202999kB subtitle:0 global headers:0kB muxing overhead > 7.671857% > > which gives me a mono wma that is now bigger in size than the stereo > wma I had to start out with. and also.. my bitrate is different > (ffmpeg default?) > > Would love and appreciate your help. > Can't find a way without re-encoding. Try filter_complex: channelsplit, like this: ffmpeg -i test.wma \ -filter_complex 'channelsplit=channel_layout=stereo[FL][FR]' \ -map '[FL]' -b:a 64k left.wma -map '[FR]' -b:a 64k right.wma See doc: http://ffmpeg.org/ffmpeg-filters.html#channelsplit From news2013 at thegoosefamily.plus.com Mon Oct 7 12:15:25 2013 From: news2013 at thegoosefamily.plus.com (Martin Goose) Date: Mon, 7 Oct 2013 10:15:25 +0000 (UTC) Subject: [FFmpeg-user] Using ffmpeg with HD video broadcasts Message-ID: I use ffmpeg to tidy up UK digital satellite TV recordings using a command such as: ffmpeg -i mm.ts -vcodec copy -acodec copy -ss 00:07:01 -t 00:51:28 -y mm.mp4 This works very well with standard definition recordings but fails with high definition recordings. Example uncut output is below. I can transcode these videos using Handbrake which I think uses ffmpeg so I guess the error is mine and not something that ffmpeg cannot do. Can someone please advise me where I am going wrong? --------- start output -------- ffmpeg version 0.7.15, Copyright (c) 2000-2013 the FFmpeg developers built on Jul 25 2013 06:16:13 with gcc 4.7.2 configuration: --prefix=/usr --enable-shared --libdir=/usr/lib64 -- shlibdir=/usr/lib64 --incdir=/usr/include --disable-stripping --enable- postproc --enable-gpl --enable-pthreads --enable-libtheora --enable- libvorbis --disable-encoder=vorbis --enable-libvpx --enable-x11grab -- enable-runtime-cpudetect --enable-libdc1394 --enable-libschroedinger -- enable-librtmp --enable-libmp3lame --enable-libopencore-amrnb --enable- libopencore-amrwb --enable-version3 --enable-libx264 --enable-nonfree -- enable-libfaac --enable-libxvid 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 [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS referenced [h264 @ 0x1505300] non-existing SPS 0 referenced in buffering period [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] non-existing PPS 0 referenced [h264 @ 0x1505300] decode_slice_header error [h264 @ 0x1505300] no frame! [h264 @ 0x1505300] mmco: unref short failure Last message repeated 1 times [mpegts @ 0x14fd0e0] max_analyze_duration 5000000 reached at 5024000 [NULL @ 0x1522c80] start time is not set in av_estimate_timings_from_pts [NULL @ 0x1524d80] start time is not set in av_estimate_timings_from_pts [NULL @ 0x1526e80] start time is not set in av_estimate_timings_from_pts [NULL @ 0x1528f80] start time is not set in av_estimate_timings_from_pts [NULL @ 0x152d180] start time is not set in av_estimate_timings_from_pts [NULL @ 0x152f280] start time is not set in av_estimate_timings_from_pts [NULL @ 0x1531380] start time is not set in av_estimate_timings_from_pts [NULL @ 0x1533480] start time is not set in av_estimate_timings_from_pts [NULL @ 0x1535580] start time is not set in av_estimate_timings_from_pts [NULL @ 0x1537680] start time is not set in av_estimate_timings_from_pts [NULL @ 0x1539780] start time is not set in av_estimate_timings_from_pts [NULL @ 0x153b880] start time is not set in av_estimate_timings_from_pts [NULL @ 0x153fa80] start time is not set in av_estimate_timings_from_pts [NULL @ 0x1541b80] start time is not set in av_estimate_timings_from_pts Input #0, mpegts, from 'bam.ts': Duration: 01:09:52.33, start: 92096.160644, bitrate: 6758 kb/s Program 6940 Program 6941 Stream #0.0[0x1518]: Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 30.05 fps, 25 tbr, 90k tbn, 50 tbc Stream #0.1[0x151a](NAR): Audio: mp2, 48000 Hz, stereo, s16, 256 kb/s Stream #0.2[0x151b](eng): Subtitle: [6][0][0][0] / 0x0006 Stream #0.3[0xf06]: Data: [11][0][0][0] / 0x000B Stream #0.4[0xf07]: Data: [11][0][0][0] / 0x000B Stream #0.5[0xf09]: Data: [11][0][0][0] / 0x000B Stream #0.6[0x151c](eng): Subtitle: dvbsub Stream #0.7[0xf00]: Data: [5][0][0][0] / 0x0005 Stream #0.8[0xf01]: Data: [5][0][0][0] / 0x0005 Stream #0.9[0xf02]: Data: [5][0][0][0] / 0x0005 Stream #0.10[0xf03]: Data: [5][0][0][0] / 0x0005 Stream #0.11[0xf04]: Data: [5][0][0][0] / 0x0005 Stream #0.12[0x911]: Data: [5][0][0][0] / 0x0005 Stream #0.13[0x912]: Data: [5][0][0][0] / 0x0005 Stream #0.14[0x913]: Data: [5][0][0][0] / 0x0005 Stream #0.15[0x1519](eng): Audio: ac3, 48000 Hz, stereo, s16, 192 kb/s Stream #0.16[0x919]: Data: [5][0][0][0] / 0x0005 Stream #0.17[0x91a]: Data: [5][0][0][0] / 0x0005 Program 6943 Program 6945 Output #0, mp4, to 'bam.mp4': Metadata: encoder : Lavf52.111.0 Stream #0.0: Video: libx264, yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], q=2-31, 25 tbn, 25 tbc Stream #0.1(NAR): Audio: mp2, 48000 Hz, stereo, 256 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop, [?] for help [mp4 @ 0x14fbb80] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 1475 >= 1475 av_interleaved_write_frame(): Invalid data found when processing input --------- end output -------- From dashing.meng at gmail.com Mon Oct 7 12:54:50 2013 From: dashing.meng at gmail.com (littlebat) Date: Mon, 7 Oct 2013 18:54:50 +0800 Subject: [FFmpeg-user] Using ffmpeg with HD video broadcasts In-Reply-To: References: Message-ID: <20131007185450.098af87f31ec6ed777d2bf0d@gmail.com> On Mon, 7 Oct 2013 10:15:25 +0000 (UTC) Martin Goose wrote: > I use ffmpeg to tidy up UK digital satellite TV recordings using a > command such as: > ffmpeg -i mm.ts -vcodec copy -acodec copy -ss 00:07:01 -t 00:51:28 -y > mm.mp4 > > This works very well with standard definition recordings but fails > with high definition recordings. Example uncut output is below. I can > transcode these videos using Handbrake which I think uses ffmpeg so I > guess the error is mine and not something that ffmpeg cannot do. > > Can someone please advise me where I am going wrong? > > --------- start output -------- > ffmpeg version 0.7.15, Copyright (c) 2000-2013 the FFmpeg developers > built on Jul 25 2013 06:16:13 with gcc 4.7.2 > Too old version, try the lastest: https://ffmpeg.org/trac/ffmpeg/wiki/CompilationGuide From dashing.meng at gmail.com Mon Oct 7 13:04:36 2013 From: dashing.meng at gmail.com (littlebat) Date: Mon, 7 Oct 2013 19:04:36 +0800 Subject: [FFmpeg-user] How to select x264 preset when do ffmpeg 2-pass encoding is helpful for quality ? Message-ID: <20131007190436.82c1c4827334cef5a465023f@gmail.com> Hi, I read the 2-pass encoding guide at: https://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide#twopass I found the guide use commands below as a two-pass example: ffmpeg -y -i input -c:v libx264 -preset medium -b:v 555k \ -pass 1 -an -f mp4 /dev/null && \ ffmpeg -i input -c:v libx264 -preset medium -b:v 555k \ -pass 2 -c:a libfdk_aac -b:a 128k output.mp4 A question: If a slower preset is helpful for the quality at a specified bitrate when do a 2-pass encoding with libx264 encoder? For example: select "ultrafast" in pass 1 and "veryslow" in pass 2, or, select "ultrafast" for both 2 passes? From news2013 at thegoosefamily.plus.com Mon Oct 7 13:59:21 2013 From: news2013 at thegoosefamily.plus.com (Martin Goose) Date: Mon, 7 Oct 2013 11:59:21 +0000 (UTC) Subject: [FFmpeg-user] Using ffmpeg with HD video broadcasts References: <20131007185450.098af87f31ec6ed777d2bf0d@gmail.com> Message-ID: On Mon, 07 Oct 2013 18:54:50 +0800, littlebat wrote: > Too old version, try the lastest: > https://ffmpeg.org/trac/ffmpeg/wiki/CompilationGuide Thanks. That is the version that comes with my Linux distro PCLinuxOS. I will ask for the latest version of ffmpeg and/or try another distro. From dashing.meng at gmail.com Mon Oct 7 15:23:50 2013 From: dashing.meng at gmail.com (littlebat) Date: Mon, 7 Oct 2013 21:23:50 +0800 Subject: [FFmpeg-user] Using ffmpeg with HD video broadcasts In-Reply-To: References: <20131007185450.098af87f31ec6ed777d2bf0d@gmail.com> Message-ID: <20131007212350.e248c3fddcf4ef707d9bca4a@gmail.com> On Mon, 7 Oct 2013 11:59:21 +0000 (UTC) Martin Goose wrote: > On Mon, 07 Oct 2013 18:54:50 +0800, littlebat wrote: > > > Too old version, try the lastest: > > https://ffmpeg.org/trac/ffmpeg/wiki/CompilationGuide > > Thanks. That is the version that comes with my Linux distro > PCLinuxOS. I will ask for the latest version of ffmpeg and/or try > another distro. > Try static binary for test: http://ffmpeg.gusari.org/static/ From lou at lrcd.com Mon Oct 7 19:43:27 2013 From: lou at lrcd.com (Lou) Date: Mon, 7 Oct 2013 09:43:27 -0800 Subject: [FFmpeg-user] How to select x264 preset when do ffmpeg 2-pass encoding is helpful for quality ? In-Reply-To: <20131007190436.82c1c4827334cef5a465023f@gmail.com> References: <20131007190436.82c1c4827334cef5a465023f@gmail.com> Message-ID: <20131007094327.2f39c82f@lrcd.com> On Mon, 7 Oct 2013 19:04:36 +0800 littlebat wrote: > A question: If a slower preset is helpful for the quality at a > specified bitrate when do a 2-pass encoding with libx264 encoder? Sorry, but I don't quite understand the question. > For example: select "ultrafast" in pass 1 and "veryslow" in pass 2, or, > select "ultrafast" for both 2 passes? You should use the same preset for both passes to avoid errors such as: [libx264 @ 0x20ac580] different weightp setting than first pass (1 vs 2) Note that the first pass should have "turbo" settings applied (disable slower options generally not useful for first pass) unless x264 option --slow-firstpass is used (maybe there are some exceptions). From highgod0401 at gmail.com Tue Oct 8 03:09:32 2013 From: highgod0401 at gmail.com (Wei Gao) Date: Tue, 8 Oct 2013 09:09:32 +0800 Subject: [FFmpeg-user] Hi, I want ask a quesiton about -vcodec copy, mine doesn't work correct, thanks. In-Reply-To: References: Message-ID: 2013/10/5 Carl Eugen Hoyos > Wei Gao gmail.com> writes: > > > ffmpeg -i ./testfile/wwwq.mp4 -vcodec copy > > ./testfile/out.h264, ffmpeg > > print that > > > > Output file is empty, nothing was encoded > > (check -ss / -t / -frames parameters if used) > > This should be fixed (by Michael), thank you for the > report! > > Hi, Can you tell me what the bug of the video? thanks. Best regards. > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From rfg at tristatelogic.com Tue Oct 8 03:30:21 2013 From: rfg at tristatelogic.com (Ronald F. Guilmette) Date: Mon, 07 Oct 2013 18:30:21 -0700 Subject: [FFmpeg-user] Changing DAR ? Message-ID: <54126.1381195821@server1.tristatelogic.com> I am trying to use ffmpeg (2.0.1) to make what I believe should be a simple change in a video file I have, and it's just not happening, so I need to ask what I am doing wrong. (Please note that I am laboring under the disadvantage of substantial ignorance about a lot of this stuff, so please bear with me and be kind.) Basically, the file I have, when played on various players, just looks wrong, i.e. a big stretched horizontally. I've run mediainfo on the file and the output of that is attached below. I cannot help but notice that in this output, the display aspect ratio is set to a rather strange value, i.e. 2.011. I do suspect that the proper value should be 16:9 (1.7777). Anyway, I would like to try setting it to 16:9 and then see if the video looks reasonably normal after that. (I confess that I don't even have any idea as to whether that 2.011 value that mediainfo is reporting is relevant to the container or to the video stream, so any enlightenment on that point would be appreciated too.) Of course, if I can avoid wholesale re-encoding while ``fixing'' the DAR, then I'd like to do that. So anyway, I've tried ffmpeg (2.0.1) with the "-acodec copy" and "-vcodec copy" options and using those, I have run two different experiments. In the first, I added "-aspect 16:9" and that had no apparent effect at all. In the second I tried instead adding -vf setdar=16:9 (although this appears to be a magic undocumented thing that, although mentioned in various online posts is most definitely _not_ documented in the ffmpeg online documentation at all). Anyway, that also had no apparent effect. In both cases, running mediainfo on the resulting output files shows the DAR _still_ set to the value 2.011, which is not at all what I want. And help or advice would be appreciated. Regards, rfg P.S. I am beginning to think that what the world really needs, in addition to a good 5 cent cigar, is a nice tutotial page someplace that would go into some depth explaining all of the ins and outs of DAR, PAR, and SAR, and which would also clarify not only how to simply _find_ those values for any given video (in any given popular format), _both_ as they may be encoded in the video stream _and_ also in the container(s), for various popular container formats, but also and further, how to _modify_ each of those independent aspect ratios, using ffmpeg and other tools, e.g WINARChanger, Yamb, etc., again both on the container level and on the stream level. After some searching, I have found that this information is hard to come by on the web, and when one does find some info on this topic, it is fragmentary and almost always unenligntening. (For example, I've seen where several people suggest using YAMB to change DAR. OK, so I downloaded the thing and installed on omy Win7 system, and managed to get it to open the file of interest. Swell. NOW WHAT? If there is a way to get this thing to change the DAR I sure as hell don't see it. Whatever it is, if it is there, it is non-intutive and non-obvious. Sigh. That's OK, I guess. I would prefer to be using ffmpeg from the command line over on my FreeBSD system anyway.) Anyway, maybe I myself might take a stab at writing up a web page like what I have suggested just above... the "Video Aspect Ratios for Dummies". I would probably be the perfect guy to write this, since I *am* a dummy, and so I'd be starting almost from scratch, and (thus) I'd have to go 'round and badger all sorts of experts to gather all the info for the page, and I would not be polluted by any pre-established under- standings of this stuff, since I have (essentially) none. mediainfo output: ============================================================================ General Complete name : Kr.avi Format : AVI Format/Info : Audio Video Interleave File size : 105 MiB Duration : 5mn 14s Overall bit rate : 2 797 Kbps Writing library : VirtualDub build 32817/release Video ID : 0 Format : MPEG-4 Visual Format profile : Advanced Simple at L5 Format settings, BVOP : 1 Format settings, QPel : No Format settings, GMC : No warppoints Format settings, Matrix : Default (MPEG) Codec ID : XVID Codec ID/Hint : XviD Duration : 5mn 14s Bit rate : 2 593 Kbps Width : 720 pixels Height : 358 pixels Display aspect ratio : 2.011 Frame rate : 29.970 fps Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Progressive Compression mode : Lossy Bits/(Pixel*Frame) : 0.336 Stream size : 97.2 MiB (93%) Writing library : XviD 64 Audio ID : 1 Format : AC-3 Format/Info : Audio Coding 3 Mode extension : CM (complete main) Codec ID : 2000 Duration : 5mn 14s Bit rate mode : Constant Bit rate : 192 Kbps Channel(s) : 2 channels Channel positions : Front: L R Sampling rate : 48.0 KHz Bit depth : 16 bits Compression mode : Lossy Stream size : 7.20 MiB (7%) Alignment : Aligned on interleaves Interleave, duration : 33 ms (1.00 video frame) Interleave, preload duration : 512 ms From dashing.meng at gmail.com Tue Oct 8 05:01:54 2013 From: dashing.meng at gmail.com (littlebat) Date: Tue, 8 Oct 2013 11:01:54 +0800 Subject: [FFmpeg-user] How to select x264 preset when do ffmpeg 2-pass encoding is helpful for quality ? In-Reply-To: <20131007094327.2f39c82f@lrcd.com> References: <20131007190436.82c1c4827334cef5a465023f@gmail.com> <20131007094327.2f39c82f@lrcd.com> Message-ID: <20131008110154.068c31dd88a962b427502abd@gmail.com> On Mon, 7 Oct 2013 09:43:27 -0800 Lou wrote: > On Mon, 7 Oct 2013 19:04:36 +0800 > littlebat wrote: > > > A question: If a slower preset is helpful for the quality at a > > specified bitrate when do a 2-pass encoding with libx264 encoder? > > Sorry, but I don't quite understand the question. For example, a given video bitrate is 1024kbps, when do only one-pass encoding, we select a slower preset will gain a better quality. But, when I do the 2-pass encoding, if I select a slower preset in both the first and second pass, if will I can still gain a better quality than we use a faster preset in both the first and second pass? Or, the preset is no influence on the quality in a 2-pass encoding? I read the FFMpeg wiki page: https://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide#twopass It seems don't clarify this question. > > For example: select "ultrafast" in pass 1 and "veryslow" in pass 2, > > or, select "ultrafast" for both 2 passes? > > You should use the same preset for both passes to avoid errors such > as: [libx264 @ 0x20ac580] different weightp setting than first pass > (1 vs 2) > > Note that the first pass should have "turbo" settings applied (disable > slower options generally not useful for first pass) unless x264 option > --slow-firstpass is used (maybe there are some exceptions). From adishavit at gmail.com Tue Oct 8 08:22:32 2013 From: adishavit at gmail.com (Adi Shavit) Date: Tue, 8 Oct 2013 08:22:32 +0200 Subject: [FFmpeg-user] How to dumping a full muxed multiprogram stream. Message-ID: Hi! I have a muxed UDP MPEG-2 multi-program transport stream. How can I use ffmpeg to dump this stream, with all its muxed elementary streams, to a .ts file without any transcoding? Thanks, Adi From news2013 at thegoosefamily.plus.com Tue Oct 8 08:46:03 2013 From: news2013 at thegoosefamily.plus.com (Martin Goose) Date: Tue, 8 Oct 2013 06:46:03 +0000 (UTC) Subject: [FFmpeg-user] Using ffmpeg with HD video broadcasts References: <20131007185450.098af87f31ec6ed777d2bf0d@gmail.com> <20131007212350.e248c3fddcf4ef707d9bca4a@gmail.com> Message-ID: On Mon, 07 Oct 2013 21:23:50 +0800, littlebat wrote: > On Mon, 7 Oct 2013 11:59:21 +0000 (UTC) > Martin Goose wrote: > >> On Mon, 07 Oct 2013 18:54:50 +0800, littlebat wrote: >> >> > Too old version, try the lastest: >> > https://ffmpeg.org/trac/ffmpeg/wiki/CompilationGuide >> >> Thanks. That is the version that comes with my Linux distro PCLinuxOS. >> I will ask for the latest version of ffmpeg and/or try another distro. >> >> > Try static binary for test: http://ffmpeg.gusari.org/static/ I had a quick look at Ubuntu and ran the ffmpeg version there. It gave a warning that ffmpeg is deprecated and 'avconv' should be used instead. The version of 'libav' that contains 'avconv' in the PCLinuxOS repos does the job! From anatol2002 at gmail.com Tue Oct 8 11:54:27 2013 From: anatol2002 at gmail.com (Anatol) Date: Tue, 8 Oct 2013 12:54:27 +0300 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: <54126.1381195821@server1.tristatelogic.com> References: <54126.1381195821@server1.tristatelogic.com> Message-ID: Check: http://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar On Tue, Oct 8, 2013 at 4:30 AM, Ronald F. Guilmette wrote: > > I am trying to use ffmpeg (2.0.1) to make what I believe should be > a simple change in a video file I have, and it's just not happening, > so I need to ask what I am doing wrong. (Please note that I am laboring > under the disadvantage of substantial ignorance about a lot of this stuff, > so please bear with me and be kind.) > > Basically, the file I have, when played on various players, just looks > wrong, i.e. a big stretched horizontally. I've run mediainfo on the > file and the output of that is attached below. > > I cannot help but notice that in this output, the display aspect ratio > is set to a rather strange value, i.e. 2.011. I do suspect that the > proper value should be 16:9 (1.7777). Anyway, I would like to try > setting it to 16:9 and then see if the video looks reasonably normal > after that. (I confess that I don't even have any idea as to whether > that 2.011 value that mediainfo is reporting is relevant to the container > or to the video stream, so any enlightenment on that point would be > appreciated too.) > > Of course, if I can avoid wholesale re-encoding while ``fixing'' the > DAR, then I'd like to do that. > > So anyway, I've tried ffmpeg (2.0.1) with the "-acodec copy" and > "-vcodec copy" options and using those, I have run two different > experiments. In the first, I added "-aspect 16:9" and that had no > apparent effect at all. In the second I tried instead adding > -vf setdar=16:9 (although this appears to be a magic undocumented > thing that, although mentioned in various online posts is most > definitely _not_ documented in the ffmpeg online documentation at > all). Anyway, that also had no apparent effect. In both cases, > running mediainfo on the resulting output files shows the DAR > _still_ set to the value 2.011, which is not at all what I want. > > And help or advice would be appreciated. > > > Regards, > rfg > > > P.S. I am beginning to think that what the world really needs, in addition > to a good 5 cent cigar, is a nice tutotial page someplace that would go > into some depth explaining all of the ins and outs of DAR, PAR, and SAR, > and which would also clarify not only how to simply _find_ those values > for any given video (in any given popular format), _both_ as they may be > encoded in the video stream _and_ also in the container(s), for various > popular container formats, but also and further, how to _modify_ each of > those independent aspect ratios, using ffmpeg and other tools, e.g > WINARChanger, Yamb, etc., again both on the container level and on the > stream level. After some searching, I have found that this information > is hard to come by on the web, and when one does find some info on this > topic, it is fragmentary and almost always unenligntening. (For example, > I've seen where several people suggest using YAMB to change DAR. OK, so > I downloaded the thing and installed on omy Win7 system, and managed to > get it to open the file of interest. Swell. NOW WHAT? If there is a > way to get this thing to change the DAR I sure as hell don't see it. > Whatever it is, if it is there, it is non-intutive and non-obvious. > Sigh. That's OK, I guess. I would prefer to be using ffmpeg from the > command line over on my FreeBSD system anyway.) > > Anyway, maybe I myself might take a stab at writing up a web page like > what I have suggested just above... the "Video Aspect Ratios for Dummies". > I would probably be the perfect guy to write this, since I *am* a dummy, > and so I'd be starting almost from scratch, and (thus) I'd have to > go 'round and badger all sorts of experts to gather all the info for > the page, and I would not be polluted by any pre-established under- > standings of this stuff, since I have (essentially) none. > > mediainfo output: > > ============================================================================ > General > Complete name : Kr.avi > Format : AVI > Format/Info : Audio Video Interleave > File size : 105 MiB > Duration : 5mn 14s > Overall bit rate : 2 797 Kbps > Writing library : VirtualDub build 32817/release > > Video > ID : 0 > Format : MPEG-4 Visual > Format profile : Advanced Simple at L5 > Format settings, BVOP : 1 > Format settings, QPel : No > Format settings, GMC : No warppoints > Format settings, Matrix : Default (MPEG) > Codec ID : XVID > Codec ID/Hint : XviD > Duration : 5mn 14s > Bit rate : 2 593 Kbps > Width : 720 pixels > Height : 358 pixels > Display aspect ratio : 2.011 > Frame rate : 29.970 fps > Color space : YUV > Chroma subsampling : 4:2:0 > Bit depth : 8 bits > Scan type : Progressive > Compression mode : Lossy > Bits/(Pixel*Frame) : 0.336 > Stream size : 97.2 MiB (93%) > Writing library : XviD 64 > > Audio > ID : 1 > Format : AC-3 > Format/Info : Audio Coding 3 > Mode extension : CM (complete main) > Codec ID : 2000 > Duration : 5mn 14s > Bit rate mode : Constant > Bit rate : 192 Kbps > Channel(s) : 2 channels > Channel positions : Front: L R > Sampling rate : 48.0 KHz > Bit depth : 16 bits > Compression mode : Lossy > Stream size : 7.20 MiB (7%) > Alignment : Aligned on interleaves > Interleave, duration : 33 ms (1.00 video frame) > Interleave, preload duration : 512 ms > > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Tue Oct 8 12:20:01 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Oct 2013 10:20:01 +0000 (UTC) Subject: [FFmpeg-user] Changing DAR ? References: <54126.1381195821@server1.tristatelogic.com> Message-ID: Ronald F. Guilmette tristatelogic.com> writes: > I am trying to use ffmpeg (2.0.1) to make what I believe > should be a simple change in a video file I have, and > it's just not happening, so I need to ask what I am > doing wrong. Please post your command line together with the complete, uncut console output. Carl Eugen From cehoyos at ag.or.at Tue Oct 8 12:24:01 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Oct 2013 10:24:01 +0000 (UTC) Subject: [FFmpeg-user] Using ffmpeg with HD video broadcasts References: <20131007185450.098af87f31ec6ed777d2bf0d@gmail.com> <20131007212350.e248c3fddcf4ef707d9bca4a@gmail.com> Message-ID: Martin Goose thegoosefamily.plus.com> writes: > > Try static binary for test: http://ffmpeg.gusari.org/static/ > > I had a quick look at Ubuntu and ran the ffmpeg version there. > It gave a warning that ffmpeg is deprecated This is an intentional lie, see http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html for more information. > and 'avconv' should be used instead. avconv contains several hundred bugs that are not reproducible with FFmpeg, some of them are possibly security-relevant, please understand that we therefore cannot support avconv here. Carl Eugen From cehoyos at ag.or.at Tue Oct 8 12:26:43 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Oct 2013 10:26:43 +0000 (UTC) Subject: [FFmpeg-user] Stereo WMA to right / left mono WMA without reencoding References: Message-ID: Tim Hiles gmail.com> writes: > All i want to do is split a stereo wma file into left > and right mono files without reencoding This is simply impossible, please try to read something about audio encoding. (I cannot really explain.) > ... i.e. no conversion to wav. Depending on how you mean it, this is of course possible. (But since wav usually contains lossless codecs, it makes no difference if you convert to wav and then to another file format or to any other file format directly.) Carl Eugen From cehoyos at ag.or.at Tue Oct 8 12:21:17 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Oct 2013 10:21:17 +0000 (UTC) Subject: [FFmpeg-user] How to dumping a full muxed multiprogram stream. References: Message-ID: Adi Shavit gmail.com> writes: > I have a muxed UDP MPEG-2 multi-program transport stream. > How can I use ffmpeg to dump this stream I always suggest (and use) mplayer -dumpstream It depends on your use-case if you should use FFmpeg, for a simple 1:1 dump of the incoming stream, I don't think it is the right tool. Carl Eugen From adishavit at gmail.com Tue Oct 8 14:45:03 2013 From: adishavit at gmail.com (Adi Shavit) Date: Tue, 8 Oct 2013 14:45:03 +0200 Subject: [FFmpeg-user] How to dumping a full muxed multiprogram stream. In-Reply-To: References: Message-ID: Hi Carl, MPlayer doesn't seem to support udp streams... Adi On Tue, Oct 8, 2013 at 1:21 PM, Carl Eugen Hoyos wrote: > Adi Shavit gmail.com> writes: > > > I have a muxed UDP MPEG-2 multi-program transport stream. > > How can I use ffmpeg to dump this stream > > I always suggest (and use) mplayer -dumpstream > > It depends on your use-case if you should use FFmpeg, > for a simple 1:1 dump of the incoming stream, I don't > think it is the right tool. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From news2013 at thegoosefamily.plus.com Tue Oct 8 14:55:59 2013 From: news2013 at thegoosefamily.plus.com (Martin Goose) Date: Tue, 8 Oct 2013 12:55:59 +0000 (UTC) Subject: [FFmpeg-user] Using ffmpeg with HD video broadcasts References: <20131007185450.098af87f31ec6ed777d2bf0d@gmail.com> <20131007212350.e248c3fddcf4ef707d9bca4a@gmail.com> Message-ID: On Tue, 08 Oct 2013 10:24:01 +0000, Carl Eugen Hoyos wrote: > This is an intentional lie, see > http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html for more > information. Thank you for the link to this information. I was unaware of this. PCLinuxOS currently has ffmpeg 2.0.1 in the testing repo. The current position is "This package is in testing since there are many apps that need patching since they do not build with this version." I will await its release from testing and resume using it then. From phil_rhodes at rocketmail.com Tue Oct 8 17:10:58 2013 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Tue, 8 Oct 2013 08:10:58 -0700 (PDT) Subject: [FFmpeg-user] Using ffmpeg with HD video broadcasts In-Reply-To: References: <20131007185450.098af87f31ec6ed777d2bf0d@gmail.com> <20131007212350.e248c3fddcf4ef707d9bca4a@gmail.com> Message-ID: <1381245058.36233.YahooMailNeo@web121102.mail.ne1.yahoo.com> > > It gave a warning that ffmpeg is deprecated > This is an intentional lie No, it isn't. The people who distribute ffmpeg as part of a linux distribution are free to deprecate it. Please do not accuse people of lying on this list. It is considered very rude. P From onemda at gmail.com Tue Oct 8 18:21:03 2013 From: onemda at gmail.com (Paul B Mahol) Date: Tue, 8 Oct 2013 16:21:03 +0000 Subject: [FFmpeg-user] Using ffmpeg with HD video broadcasts In-Reply-To: <1381245058.36233.YahooMailNeo@web121102.mail.ne1.yahoo.com> References: <20131007185450.098af87f31ec6ed777d2bf0d@gmail.com> <20131007212350.e248c3fddcf4ef707d9bca4a@gmail.com> <1381245058.36233.YahooMailNeo@web121102.mail.ne1.yahoo.com> Message-ID: On 10/8/13, Phil Rhodes wrote: > > >> > It gave a warning that ffmpeg is deprecated > >> This is an intentional lie > > > No, it isn't. The people who distribute ffmpeg as part of a linux > distribution are free to deprecate it. But not in the way how it was done. > > Please do not accuse people of lying on this list. It is considered very > rude. You are troll and nobody takes you seriously. From jshupert at pps-inc.com Tue Oct 8 19:08:46 2013 From: jshupert at pps-inc.com (Jim Shupert) Date: Tue, 08 Oct 2013 13:08:46 -0400 Subject: [FFmpeg-user] speeding transfert In-Reply-To: <524FC860.5010406@orange.fr> References: <524D6F1D.3070008@orange.fr> <524DC07E.1060201@pps-inc.com> <524FC860.5010406@orange.fr> Message-ID: <52543C1E.2000800@pps-inc.com> Many thanks Jim > i was quite affraid on making a ftp server (I have a lot of security > constraints) > but I think it is the best solution and I will move on you may wish to look at vsftp { very secure ftp } on a linux machine in fact - run ffmpeg on the same machine > > by the way is there a way to exhange programming tricks related to > ffmpeg ? > we are several using batch files it will be great to share solved points well. You could always post here I would love to know best of luck From leccine+szopjalki at gmail.com Tue Oct 8 19:44:49 2013 From: leccine+szopjalki at gmail.com (=?UTF-8?Q?Istv=C3=A1n?=) Date: Tue, 8 Oct 2013 10:44:49 -0700 Subject: [FFmpeg-user] Unable to parse option value "-1" as pixel format Message-ID: Hey, I am trying to convert a video from wmv/g2m to mp4/x264 but no success. The debug log is below. http://pastebin.com/raw.php?i=3itcBs4m Any suggestion how to get this working? Thanks in advance, Istvan From elliottbalsley at gmail.com Tue Oct 8 19:48:37 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Tue, 8 Oct 2013 10:48:37 -0700 Subject: [FFmpeg-user] Install updates through git Message-ID: I'm running a fairly recent build of ffmpeg through git, and I want to update to the latest version. Is there some way to configure the new installation using the exact same parameters as my current installation? I used a lot of flags during the configure last time, and it takes some time to research and remember them all. Is there a simple "update" feature? From andrey.aleksandrovich at googlemail.com Tue Oct 8 20:02:12 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Tue, 8 Oct 2013 21:02:12 +0300 Subject: [FFmpeg-user] How to dumping a full muxed multiprogram stream. In-Reply-To: References: Message-ID: >>> MPlayer doesn't seem to support udp streams... No, you are wrong, it does. However sometimes MPlayer creates files with transmission errors. I think the ffmpeg command must be like that: ffmpeg -fflags genpts -i -c:v copy -c:a copy -c:s copy "-map 0:0 -map 0:1 -map 0:2 ... -map 0:N" -f mpegts dump.ts From lou at lrcd.com Tue Oct 8 20:11:52 2013 From: lou at lrcd.com (Lou) Date: Tue, 8 Oct 2013 10:11:52 -0800 Subject: [FFmpeg-user] Install updates through git In-Reply-To: References: Message-ID: <20131008101152.107a0f6a@lrcd.com> On Tue, 8 Oct 2013 10:48:37 -0700 Elliott Balsley wrote: > Is there a simple "update" feature? Not in a way that I think you are looking for, but you can do this: $ cd ffmpeg $ make distclean $ git pull $ ./configure #with all of your options $ make You could make a script. From elliottbalsley at gmail.com Tue Oct 8 20:26:27 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Tue, 8 Oct 2013 11:26:27 -0700 Subject: [FFmpeg-user] Install updates through git In-Reply-To: <20131008101152.107a0f6a@lrcd.com> References: <20131008101152.107a0f6a@lrcd.com> Message-ID: I want to make sure I'm understanding you here. So this assumes I keep the ffmpeg source download each time? What's the benefit of "git pull" instead of downloading with "git clone" ? Sounds like a shell script is definitely the way to go to save the options. On Oct 8, 2013, at 11:11 AM, Lou wrote: > On Tue, 8 Oct 2013 10:48:37 -0700 > Elliott Balsley wrote: > >> Is there a simple "update" feature? > > Not in a way that I think you are looking for, but you can do this: > > $ cd ffmpeg > $ make distclean > $ git pull > $ ./configure #with all of your options > $ make > > You could make a script. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From bjorn.ramakers at gmail.com Tue Oct 8 20:30:11 2013 From: bjorn.ramakers at gmail.com (Bjorn Ramakers) Date: Tue, 8 Oct 2013 20:30:11 +0200 Subject: [FFmpeg-user] ffmpeg on windows with dshow, support for blackmagic intensity pro with HDYC uyvy422 support. Message-ID: Hey, Please bear with me, I have very little experience with ffmpeg but I dont mind listening (reading) and am usually a quick study. I am trying to use ffmpeg (http://ffmpeg.zeranoe.com windows build) with my blackmagic intensity pro, but cant get it to work. Input should be HDYC uyvy422. Whatever I do I keep getting no video (neither on ffmpeg to file nor with ffplay). If I try : ffplay -f dshow -i video="Decklink Video Capture" this is the output : ffplay version N-56892-ge1f8184 Copyright (c) 2003-2013 the FFmpeg developers built on Oct 4 2013 18:01:41 with gcc 4.8.1 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 46.100 / 52. 46.100 libavcodec 55. 34.100 / 55. 34.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 88.101 / 3. 88.101 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, dshow, from 'video=Decklink Video Capture': 0B f=0/0 Duration: N/A, start: 0.030097, bitrate: N/A Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422(tv), 1280x720, 59. 94 tbr, 10000k tbn, 59.94 tbc [dshow @ 042dd000] real-time buffer 121% full! frame dropped! Last message repeated 1 times [dshow @ 042dd000] real-time buffer 121% full! frame dropped!=0/0 Last message repeated 6 times Last message repeated 6 times Even specifying the formats gives the same buffer full error : c:\ffmpeg>ffplay -video_size 1280x720 -pixel_format uyvy422 -framerate 59.94 -fdshow -i video="Decklink Video Capture" -format x264 Gives : ffplay version N-56892-ge1f8184 Copyright (c) 2003-2013 the FFmpeg developers built on Oct 4 2013 18:01:41 with gcc 4.8.1 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 46.100 / 52. 46.100 libavcodec 55. 34.100 / 55. 34.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 88.101 / 3. 88.101 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, dshow, from 'video=Decklink Video Capture': 0B f=0/0 Duration: N/A, start: 0.030254, bitrate: N/A Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422(tv), 1280x720, 59. 94 tbr, 10000k tbn, 59.94 tbc [dshow @ 0437d220] real-time buffer 121% full! frame dropped! Last message repeated 1 times [dshow @ 0437d220] real-time buffer 121% full! frame dropped!=0/0 Last message repeated 4 times Last message repeated 4 times Anyone have an idea what I am doing wrong? Or is this setup just not supported? Bjorn. From lou at lrcd.com Tue Oct 8 20:33:40 2013 From: lou at lrcd.com (Lou) Date: Tue, 8 Oct 2013 10:33:40 -0800 Subject: [FFmpeg-user] Unable to parse option value "-1" as pixel format In-Reply-To: References: Message-ID: <20131008103340.171cfd85@lrcd.com> On Tue, 8 Oct 2013 10:44:49 -0700 Istv?n wrote: > Hey, > > I am trying to convert a video from wmv/g2m to mp4/x264 but no success. The > debug log is below. For future reference please include the console output in your message. > ffmpeg -loglevel debug -i test.wmv output.mp4 > ffmpeg version 1.2.4 Copyright (c) 2000-2013 the FFmpeg developers > built on Oct 8 2013 10:13:30 with Apple LLVM version 4.2 (clang-425.0.28) (based on LLVM 3.2svn) > configuration: --prefix=/usr/local/Cellar/ffmpeg/1.2.4 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --e > nable-avresample --enable-vda --cc=cc --host-cflags= --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libxvid > libavutil 52. 18.100 / 52. 18.100 > libavcodec 54. 92.100 / 54. 92.100 > libavformat 54. 63.104 / 54. 63.104 > libavdevice 54. 3.103 / 54. 3.103 > libavfilter 3. 42.103 / 3. 42.103 > libswscale 2. 2.100 / 2. 2.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 2.100 / 52. 2.100 > Splitting the commandline. > Reading option '-loglevel' ... matched as option 'loglevel' (set libav* logging level) with argument 'debug'. > Reading option '-i' ... matched as input file with argument 'test.wmv'. > Reading option 'output.mp4' ... matched as output file. > Finished splitting the commandline. > Parsing a group of options: global . > Applying option loglevel (set libav* logging level) with argument debug. > Successfully parsed a group of options. > Parsing a group of options: input file test.wmv. > Successfully parsed a group of options. > Opening an input file: test.wmv. > [asf @ 0x7f820c006600] Format asf probed with size=2048 and score=100 > [asf @ 0x7f820c006600] gpos mismatch our pos=24, end=26 > [asf @ 0x7f820c006600] gpos mismatch our pos=24, end=3421 > [asf @ 0x7f820c006600] Payload extension 50 2 > [asf @ 0x7f820c006600] gpos mismatch our pos=24, end=42 > [asf @ 0x7f820c006600] Unsupported byte array in tag ASFLeakyBucketPairs. > [asf @ 0x7f820c006600] gpos mismatch our pos=24, end=170 > [asf @ 0x7f820c006600] gpos mismatch our pos=24, end=44 > [asf @ 0x7f820c006600] File position before avformat_find_stream_info() is 5400 > [asf @ 0x7f820c006600] parser not found for codec wmav2, packets or times may be invalid. > Last message repeated 1 times > [asf @ 0x7f820c006600] All info found > [asf @ 0x7f820c006600] File position after avformat_find_stream_info() is 301284 > Guessed Channel Layout for Input Stream #0.0 : mono > Input #0, asf, from 'test.wmv': > Metadata: > DeviceConformanceTemplate: L2 > WMFSDKNeeded : 0.0.0.0000 > WMFSDKVersion : 12.0.7601.17514 > IsVBR : 1 > WM/ToolVersion : 5.8 Build 1189 > WM/ToolName : GoToMeeting > BitRateFrom the writer: 1180997 > Audio samples : 228 > Video samples : 216 > recording time : Tue, 08 Oct 2013 09:31:19 Pacific Daylight Time > Duration: 00:00:23.60, start: 0.000000, bitrate: 1159 kb/s > Stream #0:0, 11, 1/1000: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, mono, fltp, 48 kb/s > Stream #0:1, 1, 1/1000: Data: none, 0/1, 2 kb/s > Stream #0:2, 41, 1/1000: Video: g2m (G2M4 / 0x344D3247), 1024x750, 1/1000, 1132 kb/s, 100 tbr, 1k tbn, 1k tbc > Successfully opened the file. > Parsing a group of options: output file output.mp4. > Successfully parsed a group of options. > Opening an output file: output.mp4. > File 'output.mp4' already exists. Overwrite ? [y/N] > Successfully opened the file. > [buffer @ 0x7f820bc107c0] Setting entry with key 'video_size' to value '1024x750' > [buffer @ 0x7f820bc107c0] Setting entry with key 'pix_fmt' to value '-1' > [buffer @ 0x7f820bc107c0] Unable to parse option value "-1" as pixel format > Error opening filters! > Statistics: 327680 bytes read, 0 seeks > > Any suggestion how to get this working? Your build is too old. Use ffmpeg from git head, or if you must use a release for some reason use a newer release branch. From lou at lrcd.com Tue Oct 8 20:37:06 2013 From: lou at lrcd.com (Lou) Date: Tue, 8 Oct 2013 10:37:06 -0800 Subject: [FFmpeg-user] Install updates through git In-Reply-To: References: <20131008101152.107a0f6a@lrcd.com> Message-ID: <20131008103706.007e36a6@lrcd.com> On Tue, 8 Oct 2013 11:26:27 -0700 Elliott Balsley wrote: > So this assumes I keep the ffmpeg source download each time? Yes. > What's the benefit of "git pull" instead of downloading with "git clone" ? You don't have to download everything again but only the new stuff. For more info see "git help pull". > Sounds like a shell script is definitely the way to go to save the options. If your distro supports a builds-like system then that could be another option. Please do not top-post on this mailing-list. From rfg at tristatelogic.com Tue Oct 8 21:11:41 2013 From: rfg at tristatelogic.com (Ronald F. Guilmette) Date: Tue, 08 Oct 2013 12:11:41 -0700 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: Message-ID: <59815.1381259501@server1.tristatelogic.com> In message Anatol wrote: >Check: >http://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar Thank you. The information I found there has proven to be somewhat helpful, however... 1) It's relly too bad that a link to the page you just pointed me to isn't found under the description of the -vf option on this page: http://www.ffmpeg.org/ffmpeg.html#Video-Options or even here: http://www.ffmpeg.org/ffmpeg.html#filter_005foption 2) The descriptions of both setdar= and setsar= sub-options at the location you just pointed me to are extraordinarily unclear. More specifically, what, exactly is that equation near the beginning supposed to represent? The verbage at the top of this section seems to be saying that setdar= will cause the *SAR* to be changed! "The setdar filter sets the Display Aspect Ratio ... This is done by changing the specified Sample (aka Pixel) Aspect Ratio..." Huh? Obviously, what is said here is not what was actually meant. And also, of course, the true meaning and significance of the equation shown is nowhere explained. Then there is this: The filters accept the following options: 'r, ratio, dar (setdar only), sar (setsar only)' Huh?? Is that a literal 'r'? Should that be written with or without the surrounding single quotes? My impression is that 'r' in this context is supposed to represent an actual literal (floating point?) number, such as "1.7777", but that sure is not clear. Likewise, I guess that "ratio" in this context really means something like 16/9, however even that is quite confusing. Earlier, I had tried using setdar=16:9 because that (colon) notation seems to be supported by other parts of ffmpeg. However I got neither any error nor even any warning, as far as I could see, and yet there was no effect produced when I tried using -vf setdar=16:9". So I guess that ffmpeg demands different notations for the same ratio in different context, yes? Nothing like a bit of inconsistancy to confuse the users! I also have no idea what context, if any, the 'max' sub-parameter might be useful in. Do people ever really try something foolish like this? -vf setdar=10000/1 And if they do, and things go haywire, don't they get what they deserve? Last but not least, we have these two examples in the following section: setdar=dar=16/9 setdar=ratio=16/9:max=1000 Now I am totally flumoxed! What would be the difference between these two video filters? setdar=dar=16/9 setdar=ratio=16/9 and also, what would happen if I just said: -vf setdar=16/9 ? (I believe this last thing above is actually what I tried to use in my earlier experiments. I don't remember seeing any errors or warning when trying to use that. Is it in fact erroneous?) Obviously, "-vf" is an option for ffmpeg. To my way of thinking, that makes "setdar=" what I would call a "sub-option". But now it appears that that sub-option has its own specific, special, and peculiar set of sub-sub-options (e.g. "dar", "ratio", "max"). To say that this is all made somewhat less than clear by the documentation would be an understatement, I think. From nlewis at crawford.com Tue Oct 8 21:17:44 2013 From: nlewis at crawford.com (Nathan Lewis) Date: Tue, 8 Oct 2013 15:17:44 -0400 Subject: [FFmpeg-user] Install updates through git In-Reply-To: References: Message-ID: On Tue, Oct 8, 2013 at 1:48 PM, Elliott Balsley wrote: > Is there some way to configure the new installation using the exact same > parameters as my current installation? I used a lot of flags during the > configure last time, and it takes some time to research and remember them > all. Does calling ffmpeg -i not display all of the parameters passed to configure for the current installation? For example: $ ffmpeg -i ffmpeg version 1.0.3 Copyright (c) 2000-2012 the FFmpeg developers built on Jan 27 2013 20:29:40 with llvm-gcc 4.2.1 (LLVM build 2335.15.00) configuration: --enable-nonfree --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libvpx --enable-pthreads --enable-libx264 --enable-version3 libavutil 51. 73.101 / 51. 73.101 libavcodec 54. 59.100 / 54. 59.100 libavformat 54. 29.104 / 54. 29.104 libavdevice 54. 2.101 / 54. 2.101 libavfilter 3. 17.100 / 3. 17.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 From elliottbalsley at gmail.com Tue Oct 8 21:20:24 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Tue, 8 Oct 2013 12:20:24 -0700 Subject: [FFmpeg-user] Install updates through git In-Reply-To: References: Message-ID: <53BABC68-7C2F-42A0-9AE5-AD46C75B3F83@gmail.com> > Does calling ffmpeg -i not display all of the parameters passed to > configure for the current installation? For example: I think the modern option for that is -version. Yes, that does work, I was just wondering about a more streamlined way. But this is fine. From elliottbalsley at gmail.com Tue Oct 8 21:23:17 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Tue, 8 Oct 2013 12:23:17 -0700 Subject: [FFmpeg-user] libavfilter install errors Message-ID: <19820CD7-261F-4661-881F-A6D7C35F0676@gmail.com> When I installed ffmpeg from git today, I got lots of libavfilter errors during the make. Is this something I should worry about? Full output is here: configure: http://pastebin.com/TUzLaH1s make: http://pastebin.com/0HEcJPRz From onemda at gmail.com Tue Oct 8 21:24:26 2013 From: onemda at gmail.com (Paul B Mahol) Date: Tue, 8 Oct 2013 19:24:26 +0000 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: <59815.1381259501@server1.tristatelogic.com> References: <59815.1381259501@server1.tristatelogic.com> Message-ID: On 10/8/13, Ronald F. Guilmette wrote: > > In message > > Anatol wrote: > >>Check: >>http://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar > > Thank you. The information I found there has proven to be somewhat > helpful, however... > > 1) It's relly too bad that a link to the page you just pointed me to > isn't found under the description of the -vf option on this page: -vf sets list of video filters in filtergraph. Each filter may have own options. see: ffmpeg -h filter=setdar > > http://www.ffmpeg.org/ffmpeg.html#Video-Options > > or even here: > > http://www.ffmpeg.org/ffmpeg.html#filter_005foption > > 2) The descriptions of both setdar= and setsar= sub-options at the > location > you just pointed me to are extraordinarily unclear. More specifically, > what, exactly is that equation near the beginning supposed to represent? > > The verbage at the top of this section seems to be saying that setdar= > will cause the *SAR* to be changed! > > "The setdar filter sets the Display Aspect Ratio ... > > This is done by changing the specified Sample (aka Pixel) Aspect > Ratio..." > > Huh? Obviously, what is said here is not what was actually meant. And > also, of course, the true meaning and significance of the equation shown > is nowhere explained. > > Then there is this: > > The filters accept the following options: > > 'r, ratio, dar (setdar only), sar (setsar only)' > > Huh?? Is that a literal 'r'? Should that be written with or without the > surrounding single quotes? > > My impression is that 'r' in this context is supposed to represent an > actual literal (floating point?) number, such as "1.7777", but that sure > is not clear. Likewise, I guess that "ratio" in this context really means > something like 16/9, however even that is quite confusing. Earlier, I had > tried using setdar=16:9 because that (colon) notation seems to be supported > by other parts of ffmpeg. However I got neither any error nor even any > warning, as far as I could see, and yet there was no effect produced when > I tried using -vf setdar=16:9". So I guess that ffmpeg demands different > notations for the same ratio in different context, yes? > > Nothing like a bit of inconsistancy to confuse the users! Please elaborate what is wrong with documentation so it can be improved. Or even provide fixed text. > > I also have no idea what context, if any, the 'max' sub-parameter might be > useful in. Do people ever really try something foolish like this? > > -vf setdar=10000/1 > > And if they do, and things go haywire, don't they get what they deserve? > > Last but not least, we have these two examples in the following section: > > setdar=dar=16/9 > setdar=ratio=16/9:max=1000 > > Now I am totally flumoxed! What would be the difference between these > two video filters? > > setdar=dar=16/9 > setdar=ratio=16/9 > > and also, what would happen if I just said: > > -vf setdar=16/9 > ? > > (I believe this last thing above is actually what I tried to use in my > earlier experiments. I don't remember seeing any errors or warning when > trying to use that. Is it in fact erroneous?) > > Obviously, "-vf" is an option for ffmpeg. To my way of thinking, that > makes "setdar=" what I would call a "sub-option". But now it appears > that that sub-option has its own specific, special, and peculiar set of > sub-sub-options (e.g. "dar", "ratio", "max"). Perculiar? What you propose instead? > > To say that this is all made somewhat less than clear by the documentation > would be an understatement, I think. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From rfg at tristatelogic.com Tue Oct 8 21:25:52 2013 From: rfg at tristatelogic.com (Ronald F. Guilmette) Date: Tue, 08 Oct 2013 12:25:52 -0700 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: Message-ID: <59881.1381260352@server1.tristatelogic.com> In message , Carl Eugen Hoyos wrote: >Ronald F. Guilmette tristatelogic.com> writes: > >> I am trying to use ffmpeg (2.0.1) to make what I believe >> should be a simple change in a video file I have, and >> it's just not happening, so I need to ask what I am >> doing wrong. > >Please post your command line together with the complete, >uncut console output. Thank you. I would, but it might be embarassing. (1/2 :-) But seriously, I've (now, finally) managed to get done what I wanted to get done, as far as manipulating the aspect ratio(s) in the actual video file, but as I described at some length in my immediately prior posting, the documentation of this stuff could use some improvement. As I mentioned, earlier I had tried this, which did not seem to have any effect at all: ffmpeg -i foo.avi -acodec copy -vcodec copy -vf setdar=16:9 bar.avi Now I have tried instead: ffmpeg -i foo.avi -acodec copy -vcodec copy -vf setdar=16/9 bar.avi the subtle difference being the use of a '/' instead of a colon when specifying the desired DAR. Anyway, that seemed to work, although going by the examples here: http://ffmpeg.org/ffmpeg-filters.html#Examples-19 it should not have, because I neglected to insert also a "dar=" or "ratio=" sub-sub-clause. I understand now that within the arguments given in/for the setdar and setsar sub-options, when it is desired to express a ratio, that ratio must be written in the form: numerator/denominator and now that I've looked at the examples located at the URL just above I can even understand why this is necessary (i.e. because colons are used as sub-sub-option delimiters/separators, so they can't be used within the actual arguments given for the sub-sub-options) but still, I could have sworn that I had seen some examples somewhere which showed ratios of the form: numerator:denominator being used or specified within some different ffmpeg options (or sub-options). If so, then I'd like to just re-assert that it is quite confusing to the end-luser (like me) to have to write ratios in different formats, depending in the specific context within the ffmpeg command line one is dealing with. Regards, rfg From onemda at gmail.com Tue Oct 8 21:28:51 2013 From: onemda at gmail.com (Paul B Mahol) Date: Tue, 8 Oct 2013 19:28:51 +0000 Subject: [FFmpeg-user] libavfilter install errors In-Reply-To: <19820CD7-261F-4661-881F-A6D7C35F0676@gmail.com> References: <19820CD7-261F-4661-881F-A6D7C35F0676@gmail.com> Message-ID: On 10/8/13, Elliott Balsley wrote: > When I installed ffmpeg from git today, I got lots of libavfilter errors > during the make. Is this something I should worry about? Full output is > here: Those are not errors but warnings caused by useless renames and are introduced by Libav fork. You should not worry about them. > > configure: http://pastebin.com/TUzLaH1s > make: http://pastebin.com/0HEcJPRz > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From rene.herman at gmail.com Tue Oct 8 21:29:43 2013 From: rene.herman at gmail.com (Rene Herman) Date: Tue, 08 Oct 2013 21:29:43 +0200 Subject: [FFmpeg-user] How to convert 48000/2/FLOATLE AC3 to 48000/2/S16LE MP3 Message-ID: <52545D27.1090707@gmail.com> Good day. I have a (M2TS) input stream with floatle AC3 audio: === (mplayer) Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders AUDIO: 48000 Hz, 2 ch, floatle, 640.0 kbit/20.83% (ratio: 80000->384000) Selected audio codec: [ffac3] afm: ffmpeg (FFmpeg AC-3) === which I wish to transcode to s16le MP3 audio. I have by now tried probably a few hundred ways of writing: $ ffmpeg -i "$SRC" -c:v mpeg4 -vtag xvid -vf crop=1280:550:0:$(((720-550)/2+2)),scale=640:-1 -b:v 900k -c:a libmp3lame -af aformat=s16 -b:a 128k -map_metadata -1 "$DST" ... but every such attempt (which didn't bomb out due to bad parameter syntax to start with) has still resulted in floatle MP3: === (mplayer) Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders AUDIO: 48000 Hz, 2 ch, floatle, 128.0 kbit/4.17% (ratio: 16000->384000) Selected audio codec: [ffmp3float] afm: ffmpeg (FFmpeg MPEG layer-3 audio) === I'm using ffmpeg 2.0.1 on arch linux. Would anyone know how this can be achieved? Regards, Rene. From elliottbalsley at gmail.com Tue Oct 8 21:31:44 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Tue, 8 Oct 2013 12:31:44 -0700 Subject: [FFmpeg-user] libavfilter install errors In-Reply-To: References: <19820CD7-261F-4661-881F-A6D7C35F0676@gmail.com> Message-ID: > Those are not errors but warnings caused by useless renames and are introduced > by Libav fork. > > You should not worry about them. Just to be clear, I'm not talking about all the deprecated remarks, but the errors such as this: make: [libavcodec/x86/h264_intrapred.o] Error 1 (ignored) YASM libavcodec/x86/h264_intrapred_10bit.o STRIP libavcodec/x86/h264_intrapred_10bit.o strip: unrecognized option: -wN Usage: strip [-AnuSXx] [-] [-d filename] [-s filename] [-R filename] [-o output] file [...] From onemda at gmail.com Tue Oct 8 21:33:19 2013 From: onemda at gmail.com (Paul B Mahol) Date: Tue, 8 Oct 2013 19:33:19 +0000 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: <59881.1381260352@server1.tristatelogic.com> References: <59881.1381260352@server1.tristatelogic.com> Message-ID: On 10/8/13, Ronald F. Guilmette wrote: > > In message , > Carl Eugen Hoyos wrote: > >>Ronald F. Guilmette tristatelogic.com> writes: >> >>> I am trying to use ffmpeg (2.0.1) to make what I believe >>> should be a simple change in a video file I have, and >>> it's just not happening, so I need to ask what I am >>> doing wrong. >> >>Please post your command line together with the complete, >>uncut console output. > > > Thank you. I would, but it might be embarassing. (1/2 :-) > > But seriously, I've (now, finally) managed to get done what I wanted to > get done, as far as manipulating the aspect ratio(s) in the actual video > file, but as I described at some length in my immediately prior posting, > the documentation of this stuff could use some improvement. > > As I mentioned, earlier I had tried this, which did not seem to have > any effect at all: > > ffmpeg -i foo.avi -acodec copy -vcodec copy -vf setdar=16:9 bar.avi > > Now I have tried instead: > > ffmpeg -i foo.avi -acodec copy -vcodec copy -vf setdar=16/9 bar.avi > > the subtle difference being the use of a '/' instead of a colon when > specifying the desired DAR. Anyway, that seemed to work, although > going by the examples here: > > http://ffmpeg.org/ffmpeg-filters.html#Examples-19 > > it should not have, because I neglected to insert also a "dar=" or > "ratio=" sub-sub-clause. > > I understand now that within the arguments given in/for the setdar and > setsar sub-options, when it is desired to express a ratio, that ratio > must be written in the form: > > numerator/denominator > > and now that I've looked at the examples located at the URL just above > I can even understand why this is necessary (i.e. because colons are > used as sub-sub-option delimiters/separators, so they can't be used within > the actual arguments given for the sub-sub-options) but still, I could > have sworn that I had seen some examples somewhere which showed ratios > of the form: > > numerator:denominator > > being used or specified within some different ffmpeg options (or > sub-options). > If so, then I'd like to just re-assert that it is quite confusing to the > end-luser (like me) to have to write ratios in different formats, depending > in the specific context within the ffmpeg command line one is dealing with. The documentation clearly states that in that case ':' should be escaped. > > > Regards, > rfg > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From onemda at gmail.com Tue Oct 8 21:34:22 2013 From: onemda at gmail.com (Paul B Mahol) Date: Tue, 8 Oct 2013 19:34:22 +0000 Subject: [FFmpeg-user] libavfilter install errors In-Reply-To: References: <19820CD7-261F-4661-881F-A6D7C35F0676@gmail.com> Message-ID: On 10/8/13, Elliott Balsley wrote: >> Those are not errors but warnings caused by useless renames and are >> introduced >> by Libav fork. >> >> You should not worry about them. > > Just to be clear, I'm not talking about all the deprecated remarks, but the > errors such as this: > > make: [libavcodec/x86/h264_intrapred.o] Error 1 (ignored) > YASM libavcodec/x86/h264_intrapred_10bit.o > STRIP libavcodec/x86/h264_intrapred_10bit.o > strip: unrecognized option: -wN > Usage: strip [-AnuSXx] [-] [-d filename] [-s filename] [-R filename] [-o > output] file [...] Ah, that is bug, and should be reported. From jrunta at gmail.com Tue Oct 8 23:02:06 2013 From: jrunta at gmail.com (Jason Runta) Date: Tue, 8 Oct 2013 14:02:06 -0700 Subject: [FFmpeg-user] ffplay pipe output to ffmpeg (Windows) Message-ID: I'm trying to get a preview of my webcam BEFORE doing an encode pass on it. So I thought, 'why can't I just pipe the output from ffplay to ffmpeg'? ffplay -f dshow -i video="Microsoft LifeCam Cinema":audio="Desktop Microphone (Cinema - Mi" >(ffmpeg -f dshow -s 640x480 -i - -c:v libx264 -tune zerolatency -preset ultrafast -b:v 400k -pix_fmt yuv420p -c:a:0 libmp3lame -q:a 2 -b:a:0 56k -maxrate 750k -c:v:1 copy -f flv mycamera.flv) I'm on Windows, and unfortunately get a: "pipe:: Not enough space" error and then the ffmpeg process terminates. Is there a proper way to do this? From cehoyos at ag.or.at Wed Oct 9 00:05:14 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Oct 2013 22:05:14 +0000 (UTC) Subject: [FFmpeg-user] How to convert 48000/2/FLOATLE AC3 to 48000/2/S16LE MP3 References: <52545D27.1090707@gmail.com> Message-ID: Rene Herman gmail.com> writes: > $ ffmpeg -i "$SRC" -c:v mpeg4 -vtag xvid -vf > crop=1280:550:0:$(((720-550)/2+2)),scale=640:-1 > -b:v 900k -c:a libmp3lame -af aformat=s16 -b:a 128k > -map_metadata -1 "$DST" lame accepts s32, s16 and float as input. I suspect that for speed and quality reasons, it is best not to force a conversion. (This may be wrong, it assumes that lame internally does no slow conversion.) Concerning the decoding: There are two mp3 decoders in FFmpeg (and both can be used from MPlayer), one outputs floats and is faster on typical hardware (modern x86 and ppc), one is faster on some embedded hardware. Except for the speed, there should be no measurable difference between the two, if there is, it may be a bug. Generally, if you want help on this mailing list, please post your command line together with the complete, uncut console output. Carl Eugen From cehoyos at ag.or.at Wed Oct 9 00:09:14 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Oct 2013 22:09:14 +0000 (UTC) Subject: [FFmpeg-user] Install updates through git References: Message-ID: Elliott Balsley gmail.com> writes: > Is there some way to configure the new installation > using the exact same parameters as my current > installation? Assuming "installation" means an existing git checkout: $ git pull && make config If you meant that you want your custom FFmpeg build to have the same configuration as the installed version (that you did not build yourself) just run "ffmpeg", all sane versions show the configuration, no option needed. In case you don't know and if you are worried about your existing system configuration: You don't have to install FFmpeg, the default build works fine from its build directory. Carl Eugen From cehoyos at ag.or.at Wed Oct 9 00:14:14 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Oct 2013 22:14:14 +0000 (UTC) Subject: [FFmpeg-user] Changing DAR ? References: <59881.1381260352@server1.tristatelogic.com> Message-ID: Ronald F. Guilmette tristatelogic.com> writes: > ffmpeg -i foo.avi -acodec copy -vcodec copy -vf setdar=16/9 bar.avi This cannot work: Video filters ("vf") don't work with -vcodec copy. Sorry if I was unclear before: I suspect that what you want is possible with FFmpeg but if you cannot provide both the command line you tried and the console output, at least I am unable to help you. Carl Eugen From cehoyos at ag.or.at Wed Oct 9 00:10:36 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Oct 2013 22:10:36 +0000 (UTC) Subject: [FFmpeg-user] How to dumping a full muxed multiprogram stream. References: Message-ID: Adi Shavit gmail.com> writes: > MPlayer doesn't seem to support udp streams... You can try: $ mplayer ffmpeg://udp://1.2.3.4 -dumpstream (untested) Please do not top-post here, Carl Eugen From rene.herman at gmail.com Wed Oct 9 00:42:29 2013 From: rene.herman at gmail.com (Rene Herman) Date: Wed, 09 Oct 2013 00:42:29 +0200 Subject: [FFmpeg-user] How to convert 48000/2/FLOATLE AC3 to 48000/2/S16LE MP3 In-Reply-To: References: <52545D27.1090707@gmail.com> Message-ID: <52548A55.5050509@gmail.com> On 09-10-13 00:05, Carl Eugen Hoyos wrote: > Rene Herman gmail.com> writes: > >> $ ffmpeg -i "$SRC" -c:v mpeg4 -vtag xvid -vf >> crop=1280:550:0:$(((720-550)/2+2)),scale=640:-1 >> -b:v 900k -c:a libmp3lame -af aformat=s16 -b:a 128k >> -map_metadata -1 "$DST" > > lame accepts s32, s16 and float as input. > I suspect that for speed and quality reasons, it > is best not to force a conversion. > (This may be wrong, it assumes that lame internally > does no slow conversion.) Thanks for the answer -- but speed and quality are not an issue. I'm transcoding "nice BD source" to "pathetic sub-dvd destination" anyway. It's just that the target does not support mp3float: I really want it to be just bog-standard s16le. > Concerning the decoding: > There are two mp3 decoders in FFmpeg (and both can > be used from MPlayer), one outputs floats and is > faster on typical hardware (modern x86 and ppc), > one is faster on some embedded hardware. > Except for the speed, there should be no measurable > difference between the two, if there is, it may be > a bug. Sorry, but if you in the above provided an answer it seems I'm missing it. The mplayer output that I showed was just for showing how the input and output are recognized by mplayer (which is not the target: I'm transcoding for someone else). Really all that I want to know is how to get ffmpeg to transcode 48k fltp AC3 to s16le MP3. By now I've also tried "-af aresample=48000:osf=s16" but the result is still the same: floatle, and not s16. Is it just a bug that the s16 conversion specifier seems to be ignored, whatever I try? > Generally, if you want help on this mailing list, > please post your command line together with the > complete, uncut console output. Note how it says "s16p" in its output. I'm not actually sure what the "p" signifies, but it is in fact still producing floatle MP3 audio, and not s16: [rene at e600 ~/Videos]$ ffmpeg -i 00000.m2ts -c:v mpeg4 -vtag xvid -vf crop=1280:550:0:$(((720-550)/2+2)),scale=640:-1 -b:v 900k -pass 2 -c:a libmp3lame -af aformat=s16 -b:a 128k -map_metadata -1 00000.avi ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers built on Aug 11 2013 14:58:31 with gcc 4.8.1 (GCC) 20130725 (prerelease) configuration: --prefix=/usr --disable-debug --disable-static --enable-avresample --enable-dxva2 --enable-fontconfig --enable-gpl --enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libv4l2 --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-pic --enable-postproc --enable-runtime-cpudetect --enable-shared --enable-swresample --enable-vdpau --enable-version3 --enable-x11grab libavutil 52. 38.100 / 52. 38.100 libavcodec 55. 18.102 / 55. 18.102 libavformat 55. 12.100 / 55. 12.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 79.101 / 3. 79.101 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 Input #0, mpegts, from '00000.m2ts': Duration: 01:30:14.57, start: 599.958311, bitrate: 6838 kb/s Program 1 Stream #0:0[0x1011](und): Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 1280x720, 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc Stream #0:1[0x1100](bul): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 640 kb/s Output #0, avi, to '00000.avi': Metadata: ISFT : Lavf55.12.100 Stream #0:0: Video: mpeg4 (xvid / 0x64697678), yuv420p, 640x275, q=2-31, pass 2, 900 kb/s, 23.98 tbn, 23.98 tbc Stream #0:1: Audio: mp3 (libmp3lame) (U[0][0][0] / 0x0055), 48000 Hz, stereo, s16p, 128 kb/s Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg4) Stream #0:1 -> #0:1 (ac3 -> libmp3lame) Press [q] to stop, [?] for help frame= 755 fps= 37 q=4.1 Lsize= 3239kB time=00:00:31.65 bitrate= 838.2kbits/s video:2686kB audio:494kB subtitle:0 global headers:0kB muxing overhead 1.846697% And only to show the result: [rene at e600 ~/Videos]$ mplayer 00000.avi MPlayer SVN-r36285-4.8.1 (C) 2000-2013 MPlayer Team Cannot test OS support for SSE, disabling to be safe. 205 audio & 424 video codecs Playing 00000.avi. libavformat version 55.7.100 (internal) AVI file format detected. [aviheader] Video stream found, -vid 0 [aviheader] Audio stream found, -aid 1 VIDEO: [xvid] 640x275 24bpp 23.976 fps 696.9 kbps (85.1 kbyte/s) Clip info: Software: Lavf55.12.100 Load subtitles in ./ ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family libavcodec version 55.12.100 (internal) Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) ========================================================================== ========================================================================== Opening audio decoder: [mpg123] MPEG 1.0/2.0/2.5 layers I, II, III mpg123 init error: Error reading the stream. (code 18) ADecoder init failed :( ADecoder init failed :( Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders AUDIO: 48000 Hz, 2 ch, floatle, 128.0 kbit/4.17% (ratio: 16000->384000) Selected audio codec: [ffmp3float] afm: ffmpeg (FFmpeg MPEG layer-3 audio) ========================================================================== AO: [alsa] 48000Hz 2ch floatle (4 bytes per sample) Starting playback... From leccine+szopjalki at gmail.com Wed Oct 9 01:21:48 2013 From: leccine+szopjalki at gmail.com (=?UTF-8?Q?Istv=C3=A1n?=) Date: Tue, 8 Oct 2013 16:21:48 -0700 Subject: [FFmpeg-user] Unable to parse option value "-1" as pixel format In-Reply-To: <20131008103340.171cfd85@lrcd.com> References: <20131008103340.171cfd85@lrcd.com> Message-ID: You did not answer my question though... Anyways, the solution is for future reference: ffmpeg's error messages are totally useless because those have nothing to do with the actual error, in this case the input file was broken at some extent and it triggered this meaningless error message. On Tue, Oct 8, 2013 at 11:33 AM, Lou wrote: > On Tue, 8 Oct 2013 10:44:49 -0700 > Istv?n wrote: > > > Hey, > > > > I am trying to convert a video from wmv/g2m to mp4/x264 but no success. > The > > debug log is below. > > For future reference please include the console output in your message. > > > ffmpeg -loglevel debug -i test.wmv output.mp4 > > ffmpeg version 1.2.4 Copyright (c) 2000-2013 the FFmpeg developers > > built on Oct 8 2013 10:13:30 with Apple LLVM version 4.2 > (clang-425.0.28) (based on LLVM 3.2svn) > > configuration: --prefix=/usr/local/Cellar/ffmpeg/1.2.4 --enable-shared > --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree > --enable-hardcoded-tables --e > > nable-avresample --enable-vda --cc=cc --host-cflags= --host-ldflags= > --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libxvid > > libavutil 52. 18.100 / 52. 18.100 > > libavcodec 54. 92.100 / 54. 92.100 > > libavformat 54. 63.104 / 54. 63.104 > > libavdevice 54. 3.103 / 54. 3.103 > > libavfilter 3. 42.103 / 3. 42.103 > > libswscale 2. 2.100 / 2. 2.100 > > libswresample 0. 17.102 / 0. 17.102 > > libpostproc 52. 2.100 / 52. 2.100 > > Splitting the commandline. > > Reading option '-loglevel' ... matched as option 'loglevel' (set libav* > logging level) with argument 'debug'. > > Reading option '-i' ... matched as input file with argument 'test.wmv'. > > Reading option 'output.mp4' ... matched as output file. > > Finished splitting the commandline. > > Parsing a group of options: global . > > Applying option loglevel (set libav* logging level) with argument debug. > > Successfully parsed a group of options. > > Parsing a group of options: input file test.wmv. > > Successfully parsed a group of options. > > Opening an input file: test.wmv. > > [asf @ 0x7f820c006600] Format asf probed with size=2048 and score=100 > > [asf @ 0x7f820c006600] gpos mismatch our pos=24, end=26 > > [asf @ 0x7f820c006600] gpos mismatch our pos=24, end=3421 > > [asf @ 0x7f820c006600] Payload extension 50 2 > > [asf @ 0x7f820c006600] gpos mismatch our pos=24, end=42 > > [asf @ 0x7f820c006600] Unsupported byte array in tag ASFLeakyBucketPairs. > > [asf @ 0x7f820c006600] gpos mismatch our pos=24, end=170 > > [asf @ 0x7f820c006600] gpos mismatch our pos=24, end=44 > > [asf @ 0x7f820c006600] File position before avformat_find_stream_info() > is 5400 > > [asf @ 0x7f820c006600] parser not found for codec wmav2, packets or > times may be invalid. > > Last message repeated 1 times > > [asf @ 0x7f820c006600] All info found > > [asf @ 0x7f820c006600] File position after avformat_find_stream_info() > is 301284 > > Guessed Channel Layout for Input Stream #0.0 : mono > > Input #0, asf, from 'test.wmv': > > Metadata: > > DeviceConformanceTemplate: L2 > > WMFSDKNeeded : 0.0.0.0000 > > WMFSDKVersion : 12.0.7601.17514 > > IsVBR : 1 > > WM/ToolVersion : 5.8 Build 1189 > > WM/ToolName : GoToMeeting > > BitRateFrom the writer: 1180997 > > Audio samples : 228 > > Video samples : 216 > > recording time : Tue, 08 Oct 2013 09:31:19 Pacific Daylight Time > > Duration: 00:00:23.60, start: 0.000000, bitrate: 1159 kb/s > > Stream #0:0, 11, 1/1000: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 > Hz, mono, fltp, 48 kb/s > > Stream #0:1, 1, 1/1000: Data: none, 0/1, 2 kb/s > > Stream #0:2, 41, 1/1000: Video: g2m (G2M4 / 0x344D3247), 1024x750, > 1/1000, 1132 kb/s, 100 tbr, 1k tbn, 1k tbc > > Successfully opened the file. > > Parsing a group of options: output file output.mp4. > > Successfully parsed a group of options. > > Opening an output file: output.mp4. > > File 'output.mp4' already exists. Overwrite ? [y/N] > > Successfully opened the file. > > [buffer @ 0x7f820bc107c0] Setting entry with key 'video_size' to value > '1024x750' > > [buffer @ 0x7f820bc107c0] Setting entry with key 'pix_fmt' to value '-1' > > [buffer @ 0x7f820bc107c0] Unable to parse option value "-1" as pixel > format > > Error opening filters! > > Statistics: 327680 bytes read, 0 seeks > > > > Any suggestion how to get this working? > > Your build is too old. Use ffmpeg from git head, or if you must use a > release for some reason use a newer release branch. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Wed Oct 9 01:33:54 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Oct 2013 23:33:54 +0000 (UTC) Subject: [FFmpeg-user] Unable to parse option value "-1" as pixel format References: <20131008103340.171cfd85@lrcd.com> Message-ID: Istv?n gmail.com> writes: > in this case the input file was broken at some > extent Judging from your console output, this is very unlikely. Carl Eugen From cehoyos at ag.or.at Wed Oct 9 01:38:48 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Oct 2013 23:38:48 +0000 (UTC) Subject: [FFmpeg-user] How to convert 48000/2/FLOATLE AC3 to 48000/2/S16LE MP3 References: <52545D27.1090707@gmail.com> <52548A55.5050509@gmail.com> Message-ID: Rene Herman gmail.com> writes: > It's just that the target does not support mp3float Sorry for being unclear before: There is no mp3float FORMAT. There is just mp3, this is what lame outputs, and it outputs mp3 no matter if the input was s16, s32 or float. There is a mp3float DECODER in FFmpeg which outputs float (that is what you see with MPlayer by default because it is fastest), there is also a mp3 decoder which outputs s16, you can use it with: mplayer -ac ffmp3 So your target simply can neither support nor not support mp3float, it either supports mp3 or does not support it (but it doesn't matter what you feed the encoder with, it can only output mp3). Carl Eugen From rene.herman at gmail.com Wed Oct 9 01:58:04 2013 From: rene.herman at gmail.com (Rene Herman) Date: Wed, 09 Oct 2013 01:58:04 +0200 Subject: [FFmpeg-user] How to convert 48000/2/FLOATLE AC3 to 48000/2/S16LE MP3 In-Reply-To: References: <52545D27.1090707@gmail.com> <52548A55.5050509@gmail.com> Message-ID: <52549C0C.4090603@gmail.com> On 09-10-13 01:38, Carl Eugen Hoyos wrote: > Rene Herman gmail.com> writes: > >> It's just that the target does not support mp3float > > Sorry for being unclear before: > There is no mp3float FORMAT. > There is just mp3, this is what lame outputs, and > it outputs mp3 no matter if the input was s16, s32 > or float. > > There is a mp3float DECODER in FFmpeg which outputs > float (that is what you see with MPlayer by default > because it is fastest), there is also a mp3 decoder > which outputs s16, you can use it with: > mplayer -ac ffmp3 Ah. Thanks much for clearing that up.... It seems, then, that the answer to my problem could in fact be that I don't actually have a problem. I'll not complain about that. (the target-hardware bombed out on, an earlier attempt at, this same transcode and once I noticed this supposedly new-fangled "floating point mp3" format I assumed that must have been it, so I wanted to get it to standard mp3 before I reshared it with target-owner; but it seems that if it in fact still has a problem, I'll have to go look elsewhere) > So your target simply can neither support nor not > support mp3float, it either supports mp3 or does > not support it (but it doesn't matter what you > feed the encoder with, it can only output mp3). Regards, Rene From gerion.entrup at t-online.de Wed Oct 9 01:59:15 2013 From: gerion.entrup at t-online.de (Gerion Entrup) Date: Wed, 09 Oct 2013 01:59:15 +0200 Subject: [FFmpeg-user] Use cropdetect only every xth frame Message-ID: <31395406.M0gE8IemH8@gentoo> Hello, is it possible to say the cropdetect filter, that it should analyse the picture only every 10th frame (or something similar)? The background is: I want to find the correct values for the crop and volume filters, the simplest way, I know, is: ffmpeg -i inputfile -filter:v cropdetect -filter:a volumedetect -map 0 -f null /dev/null This command takes a long time. Another way would be: ffmpeg -i inputfile -filter:a volumedetect -map 0:a -f null /dev/null ffmpeg -ss 20 -i inputfile -filter:v cropdetect -t 2 -map 0:v -f null /dev/null or something like this. The two commands need much less time than the first. My Question is: Is there one command that do the things of the two commands in one step, e.g. to decode the videotrack only every x seconds for one second and analyse with cropdetect or decode only every xth I-Frame and analyse this but decode in the same time every audio frame? Regards, Gerion From cehoyos at ag.or.at Wed Oct 9 02:02:52 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 9 Oct 2013 00:02:52 +0000 (UTC) Subject: [FFmpeg-user] Use cropdetect only every xth frame References: <31395406.M0gE8IemH8@gentoo> Message-ID: Gerion Entrup t-online.de> writes: > is it possible to say the cropdetect filter, that it > should analyse the picture only every 10th frame (or > something similar)? The select filter should allow you to do this. Carl Eugen From lou at lrcd.com Wed Oct 9 02:07:56 2013 From: lou at lrcd.com (Lou) Date: Tue, 8 Oct 2013 16:07:56 -0800 Subject: [FFmpeg-user] Unable to parse option value "-1" as pixel format In-Reply-To: References: <20131008103340.171cfd85@lrcd.com> Message-ID: <20131008160756.189aeecc@lrcd.com> On Tue, 8 Oct 2013 16:21:48 -0700 Istv?n wrote: > You did not answer my question though... Anyways, the solution is for > future reference: I answered your question. Did you miss it? Scroll to the bottom of my previous message. 1.2.4 does not contain libavcodec/g2meet.c, so I assumed it is too old to decode your input. However, I may be incorrect and I did not test this time. It would have been most useful if you would have supplied the input if this was possible for you to do. > ffmpeg's error messages are totally useless because those have nothing to > do with the actual error In your case the message could be better (if I am correct in assuming that g2m decoding is not possible with 1.2.4), but your point may be somewhat moot if recent ffmpeg works as expected. Please do not top-post on this mailing list. From gerion.entrup at t-online.de Wed Oct 9 02:56:30 2013 From: gerion.entrup at t-online.de (Gerion Entrup) Date: Wed, 09 Oct 2013 02:56:30 +0200 Subject: [FFmpeg-user] Use cropdetect only every xth frame In-Reply-To: References: <31395406.M0gE8IemH8@gentoo> Message-ID: <5377384.STYxkmvPO0@gentoo> Am Mittwoch, 9. Oktober 2013, 00:02:52 schrieb Carl Eugen Hoyos: > Gerion Entrup t-online.de> writes: > > is it possible to say the cropdetect filter, that it > > should analyse the picture only every 10th frame (or > > something similar)? > > The select filter should allow you to do this. I think, this is not in the way I want it, because ffmpeg decodes every frame and then drop it (or is this wrong?). Here a few test results: File is in RAM: > $ pv test.mp4 > /dev/null > 104MiB 0:00:00 [2,77GiB/s] [=================================================================================>] 100% Every 10th I-Frame, if I get the concept: > $ time ffmpeg -i test.mp4 -filter:v select='if(eq(pict_type\,I)\,not(mod(st(0\,ld(0)+1)\,10)))',cropdetect - filter:a volumedetect -map 0 -f null /dev/null >/dev/null 2>&1 > > real 0m20.554s > user 1m8.585s > sys 0m0.509s Every I-Frame: > $ time ffmpeg -i test.mp4 -filter:v select='eq(pict_type\,I)',cropdetect - filter:a volumedetect -map 0 -f null /dev/null >/dev/null 2>&1 > > real 0m20.593s > user 1m8.687s > sys 0m0.487s Every Frame: > $ time ffmpeg -i test.mp4 -filter:v cropdetect -filter:a volumedetect -map 0 -f null /dev/null >/dev/null 2>&1 > > real 0m20.810s > user 1m9.323s > sys 0m0.438s Only Audioframes: > $ time ffmpeg -i test.mp4 -filter:v cropdetect -filter:a volumedetect -map 0:a -f null /dev/null >/dev/null 2>&1 > > real 0m0.388s > user 0m0.370s > sys 0m0.013s Only Video and I-Frames > $ time ffmpeg -i test.mp4 -filter:v select='eq(pict_type\,I)',cropdetect -map 0:v -f null /dev/null >/dev/null 2>&1 > > real 0m19.962s > user 1m7.830s > sys 0m0.440s Every 5 seconds 1 second via Bash: > $ time for i in `seq 0 5 $(ffprobe -show_format test.mp4 2>/dev/null | grep duration | sed 's,duration=\(.*\)\..*,\1,g')`; do ffmpeg -ss "$i" -i test.mp4 - t 1 -filter:v cropdetect -map 0:v -f null /dev/null >/dev/null 2>&1; done > > real 0m10.269s > user 0m26.083s > sys 0m1.192s Every 10 seconds 1 second: > $ time for i in `seq 0 10 $(ffprobe -show_format test.mp4 2>/dev/null | grep duration | sed 's,duration=\(.*\)\..*,\1,g')`; do ffmpeg -ss "$i" -i test.mp4 - t 1 -filter:v cropdetect -map 0:v -f null /dev/null >/dev/null 2>&1; done > > real 0m5.745s > user 0m13.658s > sys 0m0.600s I hope, you get, what I mean. Gerion From nickrobbins at yahoo.com Wed Oct 9 03:32:02 2013 From: nickrobbins at yahoo.com (Nicholas Robbins) Date: Tue, 8 Oct 2013 18:32:02 -0700 (PDT) Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: References: <59881.1381260352@server1.tristatelogic.com> Message-ID: <1381282322.15551.YahooMailNeo@web160806.mail.bf1.yahoo.com> > ffmpeg -i foo.avi -acodec copy -vcodec copy -vf setdar=16/9 bar.avi You might want to try mkvmerge. $ mkvmerge -o foo.mkv --aspect-ratio 16/9? foo.avi It will repackage the file as an mkv, don't know if that is or isn't a problem. From shadowing71 at gmail.com Wed Oct 9 03:41:48 2013 From: shadowing71 at gmail.com (Young Kim) Date: Tue, 8 Oct 2013 18:41:48 -0700 Subject: [FFmpeg-user] flv error In-Reply-To: References: Message-ID: <5A72E60305EC45D6BC8505D84749F54F@gmail.com> Anyone have any suggestions? I've tried futzing around with the encode commands, and I still get this error. On Friday, October 4, 2013 at 2:19 AM, Young Kim wrote: > Hello, > > I'm currently attempt to stream to a rtmp server using bmdcapture as the input. However, after a few minutes (or hours), the ffmpeg command I'm using eventually spits out this error: > > [flv @ 0x2279ba0] Failed to update header with correct duration > [flv @ 0x2279ba0] Failed to update header with correct filesize. > > > Here's the command I'm using: > > bmdcapture -C 1 -A 2 -p 8 -c 2 -V 4 -m 1 -s 16 -F nut -f pipe:1 | ffmpeg -rtbufsize 2147483647 -threads 8 -re -copyts -i - -pix_fmt yuv420p -profile high -ab 256k -vf yadif -vcodec libx264 -maxrate:v 1024k -minrate:v 1024k -bufsize:v 1024k -tune zerolatency -ar 44100 -preset veryfast -acodec libfaac -f flv "rtmp://localhost:1935/live/test live=1" > > Does anyone have suggestions on where I can start to fix this? > > Thanks, > Young Kim > > > From senthil at real-image.com Tue Oct 8 14:22:24 2013 From: senthil at real-image.com (SK Cinema) Date: Tue, 8 Oct 2013 05:22:24 -0700 (PDT) Subject: [FFmpeg-user] xyz to rgb conversion In-Reply-To: References: <522B9A71.5090209@googlemail.com> <522E5387.4000503@googlemail.com> <522F3F4F.5040307@googlemail.com> <1380802777422-4661599.post@n4.nabble.com> <1380879179953-4661619.post@n4.nabble.com> Message-ID: <1381234944129-4661664.post@n4.nabble.com> Carl Eugen Hoyos said: >I may misunderstand but this sounds to me as if no software >would be able to recognize this file as XYZ or do I miss >something? Yes, that's correct. >As said elsewhere, please try to explain what is wrong with >the conversion by FFmpeg? I suspect (maybe I am wrong) that >no developer know why other coefficients may be needed. >(In the sense of: Which usecases need other coefficients >and which ones.) The current ffmpeg conversion would use matrix coefficients for XYZ to Rec.709 conversion and assume specific gamma values for the source X'Y'Z' image and the target Rec.709 RGB image. A possible alternative requirement would be for converting to the DCI P3 colour space rather than Rec.709. Also, one might wish to use different gamma values. Also, XYZ is the color space with the widest gamut one would encounter. There would likely be material that is out of gamut in Rec.709 and this can produce nasty artifacts. Using a 3D LUT that is built to gently handle such out of gamut issues would often be preferable to a matrix color space conversion. For reference, here's the command line I'd like to use: ffmpeg -i video-source.mxf -vf scale=1920:808,lut3d=XYZ-sRGB.3dl,pad=width=1920:height=1080:x=0:y=136 -codec prores output.mov -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/xyz-to-rgb-conversion-tp4661196p4661664.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From adishavit at gmail.com Wed Oct 9 08:37:46 2013 From: adishavit at gmail.com (Adi Shavit) Date: Wed, 9 Oct 2013 08:37:46 +0200 Subject: [FFmpeg-user] How to dumping a full muxed multiprogram stream. In-Reply-To: References: Message-ID: Thanks. I managed to do it using https://github.com/gfto/**tsdumper2 . On Wed, Oct 9, 2013 at 1:10 AM, Carl Eugen Hoyos wrote: > Adi Shavit gmail.com> writes: > > > MPlayer doesn't seem to support udp streams... > > You can try: > $ mplayer ffmpeg://udp://1.2.3.4 -dumpstream > (untested) > > Please do not top-post here, 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 Oct 9 09:32:45 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 9 Oct 2013 07:32:45 +0000 (UTC) Subject: [FFmpeg-user] Use cropdetect only every xth frame References: <31395406.M0gE8IemH8@gentoo> <5377384.STYxkmvPO0@gentoo> Message-ID: Gerion Entrup t-online.de> writes: > > The select filter should allow you to do this. > I think, this is not in the way I want it, because > ffmpeg decodes every frame and then drop it (or is > this wrong?). You can use -skip_frame (see the documentation). Carl Eugen From cehoyos at ag.or.at Wed Oct 9 10:05:24 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 9 Oct 2013 08:05:24 +0000 (UTC) Subject: [FFmpeg-user] IVTC with pullup filter References: Message-ID: Elliott Balsley gmail.com> writes: > I have a source video that's 29.97 with 3:2 pulldown, > and I want to encode it at 23.98p. This ffmpeg command > produces video at 23.98, but with a pattern of 2 > progressive frames followed by 2 interlaced frames. This bug was fixed by Paul. > ffmpeg -ss 208 -i NCIS.ts -t 30 -vf "pullup,fps=24000/1001" > -acodec copy -vcodec libx264 pullup.mkv Unfortunately, this command line still does not work: Instead of producing different frames, for every five (progressive and interlaced) input frames, four progressive output frames are produced, but only three different ones: The wrong frame is dropped;-( The following works fine here for your sample: ffmpeg -i input -vf pullup -r 24000/1001 out.mkv Carl Eugen From onemda at gmail.com Wed Oct 9 11:08:02 2013 From: onemda at gmail.com (Paul B Mahol) Date: Wed, 9 Oct 2013 09:08:02 +0000 Subject: [FFmpeg-user] IVTC with pullup filter In-Reply-To: References: Message-ID: On 10/9/13, Carl Eugen Hoyos wrote: > Elliott Balsley gmail.com> writes: > >> I have a source video that's 29.97 with 3:2 pulldown, >> and I want to encode it at 23.98p. This ffmpeg command >> produces video at 23.98, but with a pattern of 2 >> progressive frames followed by 2 interlaced frames. > > This bug was fixed by Paul. > >> ffmpeg -ss 208 -i NCIS.ts -t 30 -vf "pullup,fps=24000/1001" >> -acodec copy -vcodec libx264 pullup.mkv > > Unfortunately, this command line still does not work: > Instead of producing different frames, for every five > (progressive and interlaced) input frames, four > progressive output frames are produced, but only three > different ones: The wrong frame is dropped;-( > > The following works fine here for your sample: > ffmpeg -i input -vf pullup -r 24000/1001 out.mkv Wouldn't using decimate after pullup be better approach? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From onemda at gmail.com Wed Oct 9 11:15:48 2013 From: onemda at gmail.com (Paul B Mahol) Date: Wed, 9 Oct 2013 09:15:48 +0000 Subject: [FFmpeg-user] Use cropdetect only every xth frame In-Reply-To: References: <31395406.M0gE8IemH8@gentoo> <5377384.STYxkmvPO0@gentoo> Message-ID: On 10/9/13, Carl Eugen Hoyos wrote: > Gerion Entrup t-online.de> writes: > >> > The select filter should allow you to do this. >> I think, this is not in the way I want it, because >> ffmpeg decodes every frame and then drop it (or is >> this wrong?). > > You can use -skip_frame (see the documentation). I think user wants to use 'enable' option within cropdetect. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From onemda at gmail.com Wed Oct 9 11:30:43 2013 From: onemda at gmail.com (Paul B Mahol) Date: Wed, 9 Oct 2013 09:30:43 +0000 Subject: [FFmpeg-user] How to convert 48000/2/FLOATLE AC3 to 48000/2/S16LE MP3 In-Reply-To: <52549C0C.4090603@gmail.com> References: <52545D27.1090707@gmail.com> <52548A55.5050509@gmail.com> <52549C0C.4090603@gmail.com> Message-ID: On 10/8/13, Rene Herman wrote: > On 09-10-13 01:38, Carl Eugen Hoyos wrote: > >> Rene Herman gmail.com> writes: >> >>> It's just that the target does not support mp3float >> >> Sorry for being unclear before: >> There is no mp3float FORMAT. >> There is just mp3, this is what lame outputs, and >> it outputs mp3 no matter if the input was s16, s32 >> or float. >> >> There is a mp3float DECODER in FFmpeg which outputs >> float (that is what you see with MPlayer by default >> because it is fastest), there is also a mp3 decoder >> which outputs s16, you can use it with: >> mplayer -ac ffmp3 > > Ah. Thanks much for clearing that up.... It seems, then, that the answer > to my problem could in fact be that I don't actually have a problem. > I'll not complain about that. > > (the target-hardware bombed out on, an earlier attempt at, this same > transcode and once I noticed this supposedly new-fangled "floating point > mp3" format I assumed that must have been it, so I wanted to get it to > standard mp3 before I reshared it with target-owner; but it seems that > if it in fact still has a problem, I'll have to go look elsewhere) mp3float decoder outputs float sample format. mp3 decoder outputs s16/s16p sample format. > >> So your target simply can neither support nor not >> support mp3float, it either supports mp3 or does >> not support it (but it doesn't matter what you >> feed the encoder with, it can only output mp3). > > Regards, > Rene > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Wed Oct 9 12:08:37 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 9 Oct 2013 10:08:37 +0000 (UTC) Subject: [FFmpeg-user] IVTC with pullup filter References: Message-ID: Paul B Mahol gmail.com> writes: > > The following works fine here for your sample: > > ffmpeg -i input -vf pullup -r 24000/1001 out.mkv > > Wouldn't using decimate after pullup be better approach? This does not really help with the original issue, don't you agree? ;-) Carl Eugen From onemda at gmail.com Wed Oct 9 12:10:58 2013 From: onemda at gmail.com (Paul B Mahol) Date: Wed, 9 Oct 2013 10:10:58 +0000 Subject: [FFmpeg-user] IVTC with pullup filter In-Reply-To: References: Message-ID: On 10/9/13, Carl Eugen Hoyos wrote: > Paul B Mahol gmail.com> writes: > >> > The following works fine here for your sample: >> > ffmpeg -i input -vf pullup -r 24000/1001 out.mkv >> >> Wouldn't using decimate after pullup be better approach? > > This does not really help with the original issue, don't > you agree? > ;-) Indeed, thus pts issue(s) is decimate filter should be resolved. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From onemda at gmail.com Wed Oct 9 12:16:58 2013 From: onemda at gmail.com (Paul B Mahol) Date: Wed, 9 Oct 2013 10:16:58 +0000 Subject: [FFmpeg-user] libavfilter install errors In-Reply-To: References: <19820CD7-261F-4661-881F-A6D7C35F0676@gmail.com> Message-ID: On 10/8/13, Paul B Mahol wrote: > On 10/8/13, Elliott Balsley wrote: >>> Those are not errors but warnings caused by useless renames and are >>> introduced >>> by Libav fork. >>> >>> You should not worry about them. >> >> Just to be clear, I'm not talking about all the deprecated remarks, but >> the >> errors such as this: >> >> make: [libavcodec/x86/h264_intrapred.o] Error 1 (ignored) >> YASM libavcodec/x86/h264_intrapred_10bit.o >> STRIP libavcodec/x86/h264_intrapred_10bit.o >> strip: unrecognized option: -wN >> Usage: strip [-AnuSXx] [-] [-d filename] [-s filename] [-R filename] [-o >> output] file [...] > > Ah, that is bug, and should be reported. > What OS is this? From tevans.uk at googlemail.com Wed Oct 9 13:54:34 2013 From: tevans.uk at googlemail.com (Tom Evans) Date: Wed, 9 Oct 2013 12:54:34 +0100 Subject: [FFmpeg-user] ffplay pipe output to ffmpeg (Windows) In-Reply-To: References: Message-ID: On Tue, Oct 8, 2013 at 10:02 PM, Jason Runta wrote: > I'm trying to get a preview of my webcam BEFORE doing an encode pass on it. > So I thought, 'why can't I just pipe the output from ffplay to ffmpeg'? > Because ffplay is a program that only displays media, and so it does not produce a stream of that media that could be passed via a pipe to ffmpeg to encode. In fact, ffplay produces no output on stdout at all, and only information on stderr. Cheers Tom From soho123.2012 at gmail.com Wed Oct 9 15:41:56 2013 From: soho123.2012 at gmail.com (Huang Soho) Date: Wed, 9 Oct 2013 21:41:56 +0800 Subject: [FFmpeg-user] ffmpeg how to output multiple file Message-ID: Hi All, Can ffmpeg output to multiple file? for example: if the command line is : ffmpeg -sn -f video4linux2 -r 30 -s 1280x720 -input_format h264 -i /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy -map 0:0 -map 1:0 http://localhost:8090/feed2.ffm I can capture video and audio from device, then output to feed2.ffm, Can ffmpeg output the video and audio data to local file? for example .avi or others? From elliottbalsley at gmail.com Wed Oct 9 16:57:05 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Wed, 9 Oct 2013 07:57:05 -0700 Subject: [FFmpeg-user] libavfilter install errors In-Reply-To: References: <19820CD7-261F-4661-881F-A6D7C35F0676@gmail.com> Message-ID: <5506BC5C-7C1B-4100-83BC-951E1E912706@gmail.com> > > What OS is this? OS X 10.8.5 From jrunta at gmail.com Wed Oct 9 20:04:44 2013 From: jrunta at gmail.com (Jason Runta) Date: Wed, 9 Oct 2013 11:04:44 -0700 Subject: [FFmpeg-user] ffmpeg how to output multiple file In-Reply-To: References: Message-ID: It can. See this article for more information: http://ffmpeg.org/trac/ffmpeg/wiki/Creating%20multiple%20outputs The important part is the tee command and getting the mappings correct. Here's an example I was using to test: ffmpeg -y -f dshow -s 640x480 -r 29.97 -i video="Microsoft LifeCam Cinema":audio="Desktop Microphone (Cinema - Mi" -c:v libx264 -preset ultrafast -b:v 800k -c:a aac -strict experimental -ar 44100 -b:a 56k -f tee -map 0:0 -map 0:1 "[f=rtsp] -rtscp_transport tcp rtsp://localhost:1234/live.sdp|C:\\users\\jasonr\\mymovie.mkv" and then using ffplay I could watch my stream using: ffplay -rtsp_flags listen rtsp://localhost:1234/live.sdp On Wed, Oct 9, 2013 at 6:41 AM, Huang Soho wrote: > Hi All, > > Can ffmpeg output to multiple file? > for example: > if the command line is : > ffmpeg -sn -f video4linux2 -r 30 -s 1280x720 -input_format h264 -i > /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy -map > 0:0 -map 1:0 http://localhost:8090/feed2.ffm > > I can capture video and audio from device, then output to feed2.ffm, > Can ffmpeg output the video and audio data to local file? for example .avi > or others? > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- *-_-=Jason Runta=-_-* From leonard at kcfchurch.org Wed Oct 9 21:14:17 2013 From: leonard at kcfchurch.org (Leonard Bogard) Date: Wed, 9 Oct 2013 12:14:17 -0700 Subject: [FFmpeg-user] Two-pass and RTMP. Message-ID: I search google for some example commands to use FFmpeg for h.264 2-pass encoding to an RTMP server but I couldn't find any. All I could find were examples to 2-pass encode to a file using multiple lines of execution. Is it even possible to do LIVE 2-pass encoding to an RTMP server with FFmpeg? tia. From jyothish.bg at srishtis.com Wed Oct 9 06:19:18 2013 From: jyothish.bg at srishtis.com (jyothish.bg at srishtis.com) Date: Tue, 08 Oct 2013 21:19:18 -0700 Subject: [FFmpeg-user] ffmpeg not working fine Message-ID: <20131008211918.9d03c5fe7a4a417631eb0f03f5ce441b.9455c8c1e6.wbe@email01.secureserver.net> A non-text attachment was scrubbed... Name: config.log Type: text/x-c Size: 124820 bytes Desc: not available URL: From lou at lrcd.com Wed Oct 9 21:58:43 2013 From: lou at lrcd.com (Lou) Date: Wed, 9 Oct 2013 11:58:43 -0800 Subject: [FFmpeg-user] ffmpeg not working fine In-Reply-To: <20131008211918.9d03c5fe7a4a417631eb0f03f5ce441b.9455c8c1e6.wbe@email01.secureserver.net> References: <20131008211918.9d03c5fe7a4a417631eb0f03f5ce441b.9455c8c1e6.wbe@email01.secureserver.net> Message-ID: <20131009115843.646fdf63@lrcd.com> On Tue, 08 Oct 2013 21:19:18 -0700 wrote: > ./configure: line 754: yasm: command not found Install yasm. Unfortunately you have not provided any additional information so I can not give you any specific instructions. From h.reindl at thelounge.net Wed Oct 9 22:01:25 2013 From: h.reindl at thelounge.net (Reindl Harald) Date: Wed, 09 Oct 2013 22:01:25 +0200 Subject: [FFmpeg-user] ffmpeg not working fine In-Reply-To: <20131008211918.9d03c5fe7a4a417631eb0f03f5ce441b.9455c8c1e6.wbe@email01.secureserver.net> References: <20131008211918.9d03c5fe7a4a417631eb0f03f5ce441b.9455c8c1e6.wbe@email01.secureserver.net> Message-ID: <5255B615.7010003@thelounge.net> Am 09.10.2013 06:19, schrieb jyothish.bg at srishtis.com: > yasm not found, use --disable-yasm for a crippled build what do you not understand in the fact that you need to install "yasm-devel" or whatever your distribution calls devel-packsges containing headers? i would understand if this is spitted in the middle of the output but in that case have it at the end your post is ueseless without mention what distribution and other environment you are using -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature URL: From jrunta at gmail.com Wed Oct 9 23:35:28 2013 From: jrunta at gmail.com (Jason Runta) Date: Wed, 9 Oct 2013 14:35:28 -0700 Subject: [FFmpeg-user] Correct way to get preview stream with no encoding? Message-ID: I'm trying to make a program that behaves similar to FMLE where I need to display a preview of the video from my webcam as well as a preview of the encoded output. I was wondering if anyone could suggest the proper way of getting a preview stream without any encoding being done to it. Do I need to make one ffmpeg call that uses -vcodec copy -acodec copy and then redirect the output from that into subsequent ffmpeg calls? If anyone has a good idea of how to get the preview stream into a C# app I'm all ears as well =) I'm pretty sure FMLE basically runs on top of ffmpeg so if they're doing it, I should be able to do it... -- *-_-=Jason Runta=-_-* From thiles at confex.com Thu Oct 10 02:01:42 2013 From: thiles at confex.com (Tim Hiles) Date: Wed, 9 Oct 2013 17:01:42 -0700 Subject: [FFmpeg-user] Stereo WMA to right / left mono WMA without reencoding In-Reply-To: References: Message-ID: I first sent this message as a non-subscriber and didn't receive word back but the more I thought about it, I came to the decision that being on this list I'm sure will definitely be useful as time progresses so I'm now an official subscriber.. and on to my question... On Sat, Oct 5, 2013 at 8:36 PM, wrote: > So.. been trying to figure out if this is possible but can't seem to find > this through google searches or in past ffmpeg list threads. All i want to > do is split a stereo wma file into left and right mono files without > reencoding... i.e. no conversion to wav. the reason is, I have all of > these wma recordings that are recorded on one side of a stereo file. I'd > like to save some space by getting rid of the blank side. converting to wav > only creates a huge file, same goes when you convert to flac.. defeating > the purpose of all of this in the first place. > > > > Here's what I have so far: > > WS400104.WMA file that is 204,040 KB > > command: > > c:\ffmpeg\ffmpeg\bin\ffmpeg.exe -i WS400104.WMA -map_channel 0.0.1:0.1 > -acodec copy out.wma > > output: > > ffmpeg version N-51683-g9dc88ac Copyright (c) 2000-2013 the FFmpeg > developers > built on Apr 8 2013 21:19:21 with gcc 4.8.0 (GCC) > configuration: --enable-gpl --enable-version3 --disable-w32threads > --enable-avisynth --enable-bzlib --enable-fontconfi > g --enable-frei0r --enable-gnutls --enable-iconv --enable-libass > --enable-libbluray --enable-libcaca --enable-libfreetyp > e --enable-libgsm --enable-libilbc --enable-libmp3lame > --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-li > bopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger > --enable-libsoxr --enable-libspeex --enable-libtheo > ra --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc > --enable-libvorbis --enable-libvpx --enable-libx264 > --enable-libxavs --enable-libxvid --enable-zlib > libavutil 52. 25.100 / 52. 25.100 > libavcodec 55. 2.100 / 55. 2.100 > libavformat 55. 1.100 / 55. 1.100 > libavdevice 55. 0.100 / 55. 0.100 > libavfilter 3. 49.101 / 3. 49.101 > libswscale 2. 2.100 / 2. 2.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 2.100 / 52. 2.100 > Guessed Channel Layout for Input Stream #0.0 : stereo > Input #0, asf, from 'WS400104.WMA': > Duration: 03:36:32.54, start: 0.000000, bitrate: 128 kb/s > Stream #0:0: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, stereo, > fltp, 128 kb/s > File 'out.wma' already exists. Overwrite ? [y/N] y > Output #0, asf, to 'out.wma': > Metadata: > WM/EncodingSettings: Lavf55.1.100 > Stream #0:0: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, stereo, 128 > kb/s > Stream mapping: > Stream #0:0 -> #0:0 (copy) > Press [q] to stop, [?] for help > size= 218588kB time=03:36:32.51 bitrate= 137.8kbits/s > video:0kB audio:203047kB subtitle:0 global headers:0kB muxing overhead > 7.653751% > > but as you see.. my output file is a stereo file. So, I tried this: > > command: > > c:\ffmpeg\ffmpeg\bin\ffmpeg.exe -i WS400104.WMA -map_channel 0.0.1:0.1 -ac > 1 out.wma > > but got this: > > output: > > H:\asis\ansem2013>c:\ffmpeg\ffmpeg\bin\ffmpeg.exe -i WS400104.WMA > -map_channel 0.0.1:0.1 -ac 1 out.wma > ffmpeg version N-51683-g9dc88ac Copyright (c) 2000-2013 the FFmpeg > developers > built on Apr 8 2013 21:19:21 with gcc 4.8.0 (GCC) > configuration: --enable-gpl --enable-version3 --disable-w32threads > --enable-avisynth --enable-bzlib --enable-fontconfi > g --enable-frei0r --enable-gnutls --enable-iconv --enable-libass > --enable-libbluray --enable-libcaca --enable-libfreetyp > e --enable-libgsm --enable-libilbc --enable-libmp3lame > --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-li > bopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger > --enable-libsoxr --enable-libspeex --enable-libtheo > ra --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc > --enable-libvorbis --enable-libvpx --enable-libx264 > --enable-libxavs --enable-libxvid --enable-zlib > libavutil 52. 25.100 / 52. 25.100 > libavcodec 55. 2.100 / 55. 2.100 > libavformat 55. 1.100 / 55. 1.100 > libavdevice 55. 0.100 / 55. 0.100 > libavfilter 3. 49.101 / 3. 49.101 > libswscale 2. 2.100 / 2. 2.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 2.100 / 52. 2.100 > Guessed Channel Layout for Input Stream #0.0 : stereo > Input #0, asf, from 'WS400104.WMA': > Duration: 03:36:32.54, start: 0.000000, bitrate: 128 kb/s > Stream #0:0: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, stereo, > fltp, 128 kb/s > File 'out.wma' already exists. Overwrite ? [y/N] y > Output #0, asf, to 'out.wma': > Metadata: > WM/EncodingSettings: Lavf55.1.100 > Stream #0:0: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, mono, fltp, > 128 kb/s > Stream mapping: > Stream #0:0 -> #0:0 (wmav2 -> wmav2) > Press [q] to stop, [?] for help > size= 218572kB time=03:36:32.51 bitrate= 137.8kbits/s > video:0kB audio:202999kB subtitle:0 global headers:0kB muxing overhead > 7.671857% > > which gives me a mono wma that is now bigger in size than the stereo wma I > had to start out with. and also.. my bitrate is different (ffmpeg default?) > > Would love and appreciate your help. > > Tim > From rfg at tristatelogic.com Thu Oct 10 02:06:09 2013 From: rfg at tristatelogic.com (Ronald F. Guilmette) Date: Wed, 09 Oct 2013 17:06:09 -0700 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: Message-ID: <81960.1381363569@server1.tristatelogic.com> In message Paul B Mahol wrote: >> Now I am totally flumoxed! What would be the difference between these >> two video filters? >> >> setdar=dar=16/9 >> setdar=ratio=16/9 I don't think that I saw any answer to this yet, so I'm still wondering about it. >> and also, what would happen if I just said: >> >> -vf setdar=16/9 >> ? >> >> (I believe this last thing above is actually what I tried to use in my >> earlier experiments. I don't remember seeing any errors or warning when >> trying to use that. Is it in fact erroneous?) >> >> Obviously, "-vf" is an option for ffmpeg. To my way of thinking, that >> makes "setdar=" what I would call a "sub-option". But now it appears >> that that sub-option has its own specific, special, and peculiar set of >> sub-sub-options (e.g. "dar", "ratio", "max"). > >Perculiar? Well, since you asked, yes, "peculiar" is the right word. I find it peculiar that a sub-option, suich as "setdar" has its own sub-sub-option which allows the user to specifically request setting of the DAR, i.e.: setdar=dar= ^^^ I mean seriously... WTF? To anybody who finds this notation NON-peculiar... please raise your hand. >What you propose instead? At the moment, I have no specific change proposal because I still am very much less than clear about _either_ what the various sub-sub-options of setdar and/or setsar do actually do, _or_ what they are intended to do (which may perhaps not in fact be what they are actually doing at present). And to be clear, when I say that I don't know... and that online docu- mentation currently fails to adequately explain... what any of these things actually do, or what they are intended to do, I am saying that with respect to *numerous* different input and/or output video formats, each of which is likely to have different inherent capabilities and thus, each of which may be affecetd by these sub-options and sub-sub- options in different ways... perhaps radically different. For example, even though it is an entirely different option that the ones that have been discussed in this exchange, I've been playing around also with the -aspect option and so far I have found that it works great when the video stream is MPEG-4/10, but otherwise, may not do anything at all, e.g. when used on a file which is a WVM container containing a WMV9 video stream. I think that it is safe to say that all this aspect ratio stuff is a big muddle, and at present, neither the online ffmpeg documentation nor ffmpeg itself is making any of it any less muddled. Of course, it is not ffmpeg's fault in any sense that we humans have repeatedly elected to erect Towers of Bable in the form of numerous incompatible languages and dialects, even when it comes to video stream formats (and also container formats for same) but as I say, the lack of clear documentation about the relevant ffmpeg options, sub-options, and sub-sub-options, when combined with ffmpegs clearly inconsistant actual behavior (e.g. when the -aspect option is used with various file formats) isn't making the Tower of Babel any more cognizable by ordinary humans. I hate to have to do this, but it appears that if I personally want to actually understand either ffmpeg as a tool and/or the set of things I can and can't do to manipulate various file formats I'm probably going to have to reach down to first principals, and even though I'm not familiar with any of this stuff I'm gonna have to grub around down in the code to find the various relevant header formats and fields, e.g. for MPEG-4/10, MPEG-4/2, WMV9, WMV8, WMV7, MPEG-2, etc. Maybe once I gather all this info and begin to unsderstand it I may then be able to understand why sometimes (e.g. when working with WVM material) I can't seem to get ffmpeg to diddle the DAR without also diddling the PAR and vise versa, and maybe I'll then be able to understand other ffmpeg behaviorial quirks I've encountered also. Sigh. Oh well. I didn't actually have that much else to do this decade. ;-) From rfg at tristatelogic.com Thu Oct 10 02:10:24 2013 From: rfg at tristatelogic.com (Ronald F. Guilmette) Date: Wed, 09 Oct 2013 17:10:24 -0700 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: Message-ID: <81991.1381363824@server1.tristatelogic.com> In message Paul B Mahol wrote: >On 10/8/13, Ronald F. Guilmette wrote: >> I understand now that within the arguments given in/for the setdar and >> setsar sub-options, when it is desired to express a ratio, that ratio >> must be written in the form: >> >> numerator/denominator >> >> and now that I've looked at the examples located at the URL just above >> I can even understand why this is necessary (i.e. because colons are >> used as sub-sub-option delimiters/separators, so they can't be used within >> the actual arguments given for the sub-sub-options) but still, I could >> have sworn that I had seen some examples somewhere which showed ratios >> of the form: >> >> numerator:denominator >> >> being used or specified within some different ffmpeg options (or >> sub-options). >> If so, then I'd like to just re-assert that it is quite confusing to the >> end-luser (like me) to have to write ratios in different formats, depending >> in the specific context within the ffmpeg command line one is dealing with. > >The documentation clearly states that in that case ':' should be escaped. Where? I sure didn't see it. But in any case that is almost besides the point, since what I was bemoaning is just the fact that in different contexts within the ffmpeg command line, the usual and _normative_ notation one uses to express a ratio is different... probably unnecessarily so. (And that somewhat bewildering inconsistancy is reflected very clearly in the online ffmpeg documentation itself.) From cehoyos at ag.or.at Thu Oct 10 02:22:59 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 10 Oct 2013 00:22:59 +0000 (UTC) Subject: [FFmpeg-user] Changing DAR ? References: <81960.1381363569@server1.tristatelogic.com> Message-ID: Ronald F. Guilmette tristatelogic.com> writes: > I'm gonna have to grub around down in the code to find > the various relevant header formats and fields This is possible and certainly not unwanted! (On the contrary.) If you just to want to solve your problem, the alternative is of course still to post the command line you tried together with the complete, uncut console output. Carl Eugen From rfg at tristatelogic.com Thu Oct 10 02:30:00 2013 From: rfg at tristatelogic.com (Ronald F. Guilmette) Date: Wed, 09 Oct 2013 17:30:00 -0700 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: Message-ID: <82111.1381365000@server1.tristatelogic.com> In message , Carl Eugen Hoyos wrote: >Ronald F. Guilmette tristatelogic.com> writes: > >> ffmpeg -i foo.avi -acodec copy -vcodec copy -vf setdar=16/9 bar.avi > >This cannot work: Video filters ("vf") don't work with >-vcodec copy. Ahhhhhhh!! That is quite obviously the epiphany I was looking for, even without knowing it. Thank you! That would explain some things. >Sorry if I was unclear before: You are e gentleman, and greatly respect and appreciate that. However _you_ certainly have nothing to apologize for. I'm not sure that I can say the same for ffmpeg itself however. If what you say is true... and I have no reason whatever to disbelieve your assertion... then ffmpeg, being a reasonable and otherwise well-behaved program, conforming to common, pre-existing and long-standing general rules of polite program behaviorial etiquette, should produce a fatal error and refuse to do anything at all when it is invoked with a clearly erroneous command line like the following, wouldn't you agree? ffmpeg -i input.mp4 -acodec copy -vcodec copy -vf setsar=sar=16/9 output.mp4 Strangely however, it does not do so. Instead, it proceeds and generates an output file, even while giving -no- helpful hint whatsoever to the user, either in the form of an error or even a warning message, that he's done anything wrong at all. So, you know, I am ernestly puzzled. How can this be, if what you said about is actually true (and as I've already said, I _do_ believe you, in no small part because what you said makes perfect sense). >I suspect that what you >want is possible with FFmpeg but if you cannot provide >both the command line you tried and the console output, >at least I am unable to help you. Actually, you already _have_ helped me a great deal, just by informing me that the "-acodec copy" option is fundamentally incompatible with the -vf option... a rather important fact which I ernestly did not know _and_ which, more importantly, ffmpeg did not have the simple courtesy to inform me about... you know... before you did. I'm going to go back now and try playing with some of my video files to see what other useful bits of enlightenment I can come up with on my own, but I am likely to have more questions later on. For now, if you have the time, I'd just like to know what ffmpeg can and cannot do with respect to modifications to aspect ratio (either DAR or PAR or SAR) *without* full-scale re-encoding and for each of the following: MPEG-4/10 MPEG-4/2 WMV9 WMV8 WMV7 MPEG-2 ... and anything else you happen to know about... Thanks! From cehoyos at ag.or.at Thu Oct 10 02:36:33 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 10 Oct 2013 00:36:33 +0000 (UTC) Subject: [FFmpeg-user] Changing DAR ? References: <82111.1381365000@server1.tristatelogic.com> Message-ID: Ronald F. Guilmette tristatelogic.com> writes: > Actually, you already _have_ helped me a great deal, > just by informing me that the "-acodec copy" option > is fundamentally incompatible with the -vf option... This is not correct. As said, what you want is possible afaict, if you want help, please avoid writing enormous emails, instead just post ... (well, you know already). Carl Eugen From rfg at tristatelogic.com Thu Oct 10 02:48:38 2013 From: rfg at tristatelogic.com (Ronald F. Guilmette) Date: Wed, 09 Oct 2013 17:48:38 -0700 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: <1381282322.15551.YahooMailNeo@web160806.mail.bf1.yahoo.com> Message-ID: <82291.1381366118@server1.tristatelogic.com> In message <1381282322.15551.YahooMailNeo at web160806.mail.bf1.yahoo.com>, Nicholas Robbins wrote: >> ffmpeg -i foo.avi -acodec copy -vcodec copy -vf setdar=3D16/9 bar.avi > >You might want to try mkvmerge. > >$ mkvmerge -o foo.mkv --aspect-ratio 16/9=A0 foo.avi > >It will repackage the file as an mkv, don't know if that is or isn't a prob= >lem. Thank you for the suggestion. I've noticed that even ffmpeg alone can fix-up certain aspect ratio problems I've encountered if I'm willing to have the output go into a .mkv container. From cehoyos at ag.or.at Thu Oct 10 02:52:29 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 10 Oct 2013 00:52:29 +0000 (UTC) Subject: [FFmpeg-user] h264 screenshot from multicast References: <20130701100604.GA20160@putsch.kolbu.ws> Message-ID: St?le Kristoffersen ifi.uio.no> writes: > Here is an example file that does not work: > http://kolbu.ws/~chiller/h264fail.ts > > When I run it like this: > # ffmpeg -i h264fail.ts -vframes 1 badframe.jpg This was fixed by Michael, thank you for the report! Carl Eugen From rfg at tristatelogic.com Thu Oct 10 03:23:02 2013 From: rfg at tristatelogic.com (Ronald F. Guilmette) Date: Wed, 09 Oct 2013 18:23:02 -0700 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: Message-ID: <86382.1381368182@server1.tristatelogic.com> In message , Carl Eugen Hoyos wrote: >Ronald F. Guilmette tristatelogic.com> writes: > >> I'm gonna have to grub around down in the code to find >> the various relevant header formats and fields > >This is possible and certainly not unwanted! > >(On the contrary.) Yes. The only problem is finding the time. (It appears, just glancing at the initial output of "make" that most or all of this stuff is in C, in which I have excellent fluency, so like I say, it is only a question of finding the time.) >If you just to want to solve your problem, the alternative >is of course still to post the command line you tried >together with the complete, uncut console output. Thank you, ernestly. So far I have found workarounds for eveything I've wanted or needed to do, with the help of some small Windoze utilities (specifically WMVARChanger for modern WVM9 stuff, MPEG4Modifier for MPEG-4/2 stuff, and DVDPatcher for MPEG2 stuff), and I seem to be able to use ffmpeg to diddle aspect ratios to my liking when I use it on MPEG-4/10 stuff... as long as I'm willing to repackage it into output .MKV files. So I think I'm good for now, but aas a general principal I _do_ like to understand anything and everything that I ever mess with, so eventually I'll come back to all this and try to get a better understanding of it all. In an ideal world, there would be one siingle tool (to rule them all? :-) which would be able to do everything that can be done, even theoretically, as regards to changing aspect ratios on all these different formats, and more, *without* full re-encoding, and perhaps someday I might even create such a thing. But alas, today is not that day. (I gather that in most or all of the video stream formats there are one or more fields in the headers that describe aspect ratio(s)... either "storage" or "pixel" or "display" or all of the above, and that separately, in certain formats at least, there may be separate fields which also try to specify aspect ratios, but that these are embedded at various points with the actual video stream itself. It seems clear that those could be diddled (programatically) too, without full re-encoding, but with the exception of DVDPatcher, which is limited to only MPEG-2, I've yet to find tools that do this for other formats, let alone generally, for all formats, which would be most helpful. Oh well. Someday.) From rfg at tristatelogic.com Thu Oct 10 03:31:34 2013 From: rfg at tristatelogic.com (Ronald F. Guilmette) Date: Wed, 09 Oct 2013 18:31:34 -0700 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: Message-ID: <86429.1381368694@server1.tristatelogic.com> In message , Carl Eugen Hoyos wrote: >Ronald F. Guilmette tristatelogic.com> writes: > >> Actually, you already _have_ helped me a great deal, >> just by informing me that the "-acodec copy" option >> is fundamentally incompatible with the -vf option... > >This is not correct. Uhhhhhhh... excuse me??? Come again? There was some guy using your e-mail address who just recently posted the following to this very mailing list: >This cannot work: Video filters ("vf") don't work with >-vcodec copy. Was that not you who posted that? Did someone hijack your e-mail account? Sorry, I don't mean to be flip, but you really have succeeded in confusing me. First you say that "-acodec copy" and -vf don't work together, and then, when I repeat what seems like that exact same thing you said, you tell me I'm wrong. I do hope you will excuse me if I say that this leaves me more than a bit confused. Please clarify whatever it was that you had actually intended to say. From dashing.meng at gmail.com Thu Oct 10 03:49:58 2013 From: dashing.meng at gmail.com (littlebat) Date: Thu, 10 Oct 2013 09:49:58 +0800 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: <82111.1381365000@server1.tristatelogic.com> References: <82111.1381365000@server1.tristatelogic.com> Message-ID: <20131010094958.1ae3cbb930bb3fb428021f92@gmail.com> On Wed, 09 Oct 2013 17:30:00 -0700 "Ronald F. Guilmette" wrote: > > In message , > Carl Eugen Hoyos wrote: > > >Ronald F. Guilmette tristatelogic.com> writes: > > > >> ffmpeg -i foo.avi -acodec copy -vcodec copy -vf setdar=16/9 bar.avi > > > >This cannot work: Video filters ("vf") don't work with > >-vcodec copy. > > Ahhhhhhh!! That is quite obviously the epiphany I was looking for, > even without knowing it. Thank you! That would explain some things. > Change DAR without re-encoding the video seems hasn't implemented in FFMpeg yet, I read some from list archive, maybe a clue: [FFmpeg-user] Change aspect ratio of MPEG-PS without re-encoding? http://ffmpeg.org/pipermail/ffmpeg-user/2012-August/008742.html But, maybe you can try other tool, e.g., mplayer mentioned here: http://ffmpeg.org/pipermail/ffmpeg-user/2012-August/008749.html Or, mkv tools mentioned in this thread. From elliottbalsley at gmail.com Thu Oct 10 08:31:36 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Wed, 9 Oct 2013 23:31:36 -0700 Subject: [FFmpeg-user] IVTC with pullup filter In-Reply-To: References: Message-ID: >> Paul B Mahol gmail.com> writes: >> >>>> The following works fine here for your sample: >>>> ffmpeg -i input -vf pullup -r 24000/1001 out.mkv >>> >>> Wouldn't using decimate after pullup be better approach? >> >> This does not really help with the original issue, don't >> you agree? >> ;-) > > Indeed, thus pts issue(s) is decimate filter should be resolved. > >> >> Carl Eugen I'm now getting proper results with "-vf pullup -r 24000/1001". Are you saying decimate after pullup would be better? Why? From cehoyos at ag.or.at Thu Oct 10 08:43:47 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 10 Oct 2013 06:43:47 +0000 (UTC) Subject: [FFmpeg-user] Changing DAR ? References: <82111.1381365000@server1.tristatelogic.com> <20131010094958.1ae3cbb930bb3fb428021f92@gmail.com> Message-ID: littlebat gmail.com> writes: > Change DAR without re-encoding the video seems hasn't > implemented in FFMpeg yet It works fine here. Carl Eugen From cehoyos at ag.or.at Thu Oct 10 08:45:39 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 10 Oct 2013 06:45:39 +0000 (UTC) Subject: [FFmpeg-user] IVTC with pullup filter References: Message-ID: Elliott Balsley gmail.com> writes: > I'm now getting proper results with > "-vf pullup -r 24000/1001". This is - afaict - the only variant that works atm. > Are you saying decimate after pullup would > be better? "-r" is generally deprecated (vf fps should be used), decimate is the filter to be used after the inverse telecine filters (but it does not work correctly atm). Carl Eugen From onemda at gmail.com Thu Oct 10 10:23:00 2013 From: onemda at gmail.com (Paul B Mahol) Date: Thu, 10 Oct 2013 08:23:00 +0000 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: References: <82111.1381365000@server1.tristatelogic.com> <20131010094958.1ae3cbb930bb3fb428021f92@gmail.com> Message-ID: On 10/10/13, Carl Eugen Hoyos wrote: > littlebat gmail.com> writes: > >> Change DAR without re-encoding the video seems hasn't >> implemented in FFMpeg yet > > It works fine here. And what command line you use? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From mapandrei at gmail.com Thu Oct 10 11:09:58 2013 From: mapandrei at gmail.com (Andrei Petru Mura) Date: Thu, 10 Oct 2013 12:09:58 +0300 Subject: [FFmpeg-user] Issues with avi file Message-ID: Hello, I'm trying to convert an AVI file from a format that I don't have codecs for to a usable one using ffmpeg. This is a resulted file from a GeoVision camera recording. On their site you will find codecs for Windows only, not for Linux. (I'm running a Linux OS). When I'm trying to watch this video with VLC, I get this error: "No suitable decoder module: VLC does not support the audio or video format "GAVC". Unfortunately there is no way for you to fix this." I tried to convert this file using ffmpeg, but without any results. Bellow is what I tried and the output. (I'm a very new one to codecs, video formats and convertion, and especially to ffmpeg. So please excuse me if I'm not clear in some of my explanations. Thanks) So, the conversion trial: ./bin/ffmpeg -i /path/to/my/file.avi -c:v libxvid output.avi The output is: ffmpeg version git-2013-10-10-65c2fe7 Copyright (c) 2000-2013 the FFmpeg developers built on Oct 10 2013 09:14:31 with gcc 4.7 (Ubuntu/Linaro 4.7.3-1ubuntu1) configuration: --prefix=/home/artaxerxe/ffmpeg_build --extra-cflags=-I/home/artaxerxe/ffmpeg_build/include --extra-ldflags=-L/home/artaxerxe/ffmpeg_build/lib --bindir=/home/artaxerxe/bin --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-x11grab libavutil 52. 46.101 / 52. 46.101 libavcodec 55. 35.100 / 55. 35.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.101 / 3. 88.101 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 [avi @ 0xa670d40] probed stream 1 failed [avi @ 0xa670d40] Could not find codec parameters for stream 0 (Video: none (GAVC / 0x43564147), 1920x1080): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options [avi @ 0xa670d40] Could not find codec parameters for stream 1 (Subtitle: none): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options [avi @ 0xa670d40] Could not find codec parameters for stream 2 (Subtitle: none): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options /media/artaxerxe/AC1B-69A3/avi from server/Event20131003095553005.avi: could not find codec parameters Can someone have some suggestions for me how to deal with this kind of video? Thank you in advance! From dashing.meng at gmail.com Thu Oct 10 11:22:03 2013 From: dashing.meng at gmail.com (littlebat) Date: Thu, 10 Oct 2013 17:22:03 +0800 Subject: [FFmpeg-user] Issues with avi file In-Reply-To: References: Message-ID: <20131010172203.a14fc6b696b1406f083c7500@gmail.com> On Thu, 10 Oct 2013 12:09:58 +0300 Andrei Petru Mura wrote: > Hello, > > I'm trying to convert an AVI file from a format that I don't have > codecs for to a usable one using ffmpeg. This is a resulted file from > a GeoVision camera recording. On their site you will find codecs for > Windows only, not for Linux. (I'm running a Linux OS). > When I'm trying to watch this video with VLC, I get this error: > > "No suitable decoder module: > > VLC does not support the audio or video format "GAVC". Unfortunately > there is no way for you to fix this." > > > I tried to convert this file using ffmpeg, but without any results. > Bellow is what I tried and the output. (I'm a very new one to codecs, > video formats and convertion, and especially to ffmpeg. So please > excuse me if I'm not clear in some of my explanations. Thanks) > > So, the conversion trial: > > > ./bin/ffmpeg -i /path/to/my/file.avi -c:v libxvid output.avi > > > The output is: > > > ffmpeg version git-2013-10-10-65c2fe7 Copyright (c) 2000-2013 the > FFmpeg developers > ... > > [avi @ 0xa670d40] Could not find codec parameters for stream 0 > (Video: none (GAVC / 0x43564147), 1920x1080): unknown codec No suitable decoder in FFMpeg yet. Try play it with mplayer, if it can't work, try install mplayer's binary codecs: http://www.mplayerhq.hu/MPlayer/releases/codecs/all-20110131.tar.bz2 and play it again. If it works, maybe you can convert your file using mencoder currently. From cehoyos at ag.or.at Thu Oct 10 11:22:08 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 10 Oct 2013 09:22:08 +0000 (UTC) Subject: [FFmpeg-user] Issues with avi file References: Message-ID: Andrei Petru Mura gmail.com> writes: > I'm trying to convert an AVI file from a format that > I don't have codecs for to a usable one using ffmpeg. Please provide a sample. Carl Eugen From tevans.uk at googlemail.com Thu Oct 10 12:34:33 2013 From: tevans.uk at googlemail.com (Tom Evans) Date: Thu, 10 Oct 2013 11:34:33 +0100 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: <86429.1381368694@server1.tristatelogic.com> References: <86429.1381368694@server1.tristatelogic.com> Message-ID: On Thu, Oct 10, 2013 at 2:31 AM, Ronald F. Guilmette wrote: > > In message , > Carl Eugen Hoyos wrote: > >>Ronald F. Guilmette tristatelogic.com> writes: >> >>> Actually, you already _have_ helped me a great deal, >>> just by informing me that the "-acodec copy" option >>> is fundamentally incompatible with the -vf option... >> >>This is not correct. > > Uhhhhhhh... excuse me??? Come again? > > There was some guy using your e-mail address who just recently posted > the following to this very mailing list: > >>This cannot work: Video filters ("vf") don't work with >>-vcodec copy. > > Was that not you who posted that? Did someone hijack your e-mail account? > > Sorry, I don't mean to be flip, but you really have succeeded in > confusing me. First you say that "-acodec copy" and -vf don't > work together, and then, when I repeat what seems like that exact > same thing you said, you tell me I'm wrong. Different letters have different meanings. Carl said "VIDEO filters (-vf) dont work with -VCODEC copy". You said "-ACODEC copy is fundamentally incompatible with -vf" acodec is not the same as vcodec. Carl did not say "-acodec copy" and -vf do not work together, you did. I assume you meant to say "vcodec", but it is not what you wrote. Cheers Tom From tevans.uk at googlemail.com Thu Oct 10 12:37:40 2013 From: tevans.uk at googlemail.com (Tom Evans) Date: Thu, 10 Oct 2013 11:37:40 +0100 Subject: [FFmpeg-user] Two-pass and RTMP. In-Reply-To: References: Message-ID: On Wed, Oct 9, 2013 at 8:14 PM, Leonard Bogard wrote: > I search google for some example commands to use FFmpeg for h.264 2-pass > encoding to an RTMP server but I couldn't find any. All I could find were > examples to 2-pass encode to a file using multiple lines of execution. Is > it even possible to do LIVE 2-pass encoding to an RTMP server with FFmpeg? > RTMP is live streaming. 2 pass encoding is about taking information from the first pass to better target codec bandwidth in the 2nd pass. How do you propose this could possibly work in a live streaming context? When would your 2nd pass run? Cheers Tom From gerion.entrup at t-online.de Thu Oct 10 16:30:43 2013 From: gerion.entrup at t-online.de (Gerion Entrup) Date: Thu, 10 Oct 2013 16:30:43 +0200 Subject: [FFmpeg-user] Use cropdetect only every xth frame In-Reply-To: References: <31395406.M0gE8IemH8@gentoo> <5377384.STYxkmvPO0@gentoo> Message-ID: <1635729.v0BkWSdRbW@gentoo> Am Mittwoch, 9. Oktober 2013, 07:32:45 schrieb Carl Eugen Hoyos: > Gerion Entrup t-online.de> writes: > > > The select filter should allow you to do this. > > > > I think, this is not in the way I want it, because > > ffmpeg decodes every frame and then drop it (or is > > this wrong?). > > You can use -skip_frame (see the documentation). Does not change that much: $ time ffmpeg -skip_frame nokey -i test.mp4 -filter:v select='not(mod(n\,10))',cropdetect -filter:a volumedetect -map 0 -f null /dev/null 2>/dev/null real 0m21.118s user 0m58.030s sys 0m1.508s $ time ffmpeg -skip_frame nokey -i test.mp4 -filter:v cropdetect -filter:a volumedetect -map 0 -f null /dev/null 2>/dev/null real 0m19.979s user 0m59.553s sys 0m1.434s Many thanks for your help, by the way. Gerion From leonard at kcfchurch.org Thu Oct 10 18:07:18 2013 From: leonard at kcfchurch.org (Leonard Bogard) Date: Thu, 10 Oct 2013 09:07:18 -0700 Subject: [FFmpeg-user] Two-pass and RTMP. In-Reply-To: References: Message-ID: On Thu, Oct 10, 2013 at 3:37 AM, Tom Evans wrote: > How do you propose this could possibly work in a live streaming > context? When would your 2nd pass run? > I was thinking the same thing but I figured I would ask just in case someone had come up with a solution. Thanks anyways. From sayaprashantha at gmail.com Thu Oct 10 07:38:35 2013 From: sayaprashantha at gmail.com (prashantha S) Date: Thu, 10 Oct 2013 11:08:35 +0530 Subject: [FFmpeg-user] stream RTP over UDP from specific source to destination (negotiated using SIP SDP) using ffmpeg Message-ID: Hi, I am trying to stream RTP over UDP from specific source to destination (negotiated using SIP SDP) using ffmpeg I would like to open the source port and send the packets to destination port from this port This is because 1. my destination will always check from where the RTP is coming from (source verification) 2. Another usage by opening the source port is - receive packets sent from destination, otherwise destination will complain "ICMP unreachable" SIP session INVITE source SDP c=IN IP4 172.27.6.45 t=0 0 m=video 20002 RTP/AVP 120 b=AS:768 a=rtpmap:120 VP8/90000 a=sendrecv SIP session INVITE destination SDP m=video 40154 RTP/AVP 120 c=IN IP4 10.211.5.13 b=AS:768 a=rtpmap:120 VP8/90000 a=sendrecv So i have tried the below in source ffmpeg -re -i HD.webm -t 00:01:00 -an -vcodec libvpx -s 352x288 -b:v 768k -r 30 -f rtp rtp://10.211.5.13:40154?localaddr=172.27.6.45&localport=20002 But using ffmpeg i am not able to achieve this 1. FFMPEG does not stream from source IP&port i.e. localaddr=172.27.6.45&localport=20002. It uses random ports Does the usage of "?localaddr=172.27.6.45&localport=20002" is correct? 2. Destination complain that "ICMP unreachable" for 172.27.6.45:20002 Is there any solution for these 2 issues? Thank you Psaya From rfg at tristatelogic.com Fri Oct 11 01:14:37 2013 From: rfg at tristatelogic.com (Ronald F. Guilmette) Date: Thu, 10 Oct 2013 16:14:37 -0700 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: Message-ID: <99969.1381446877@server1.tristatelogic.com> In message , you wrote: >On Thu, Oct 10, 2013 at 2:31 AM, Ronald F. Guilmette > wrote: >> Sorry, I don't mean to be flip, but you really have succeeded in >> confusing me. First you say that "-acodec copy" and -vf don't >> work together, and then, when I repeat what seems like that exact >> same thing you said, you tell me I'm wrong. > >Different letters have different meanings. > >Carl said "VIDEO filters (-vf) dont work with -VCODEC copy". > >You said "-ACODEC copy is fundamentally incompatible with -vf" Arrrrrrgggggg! Yes. May bad. I'm sorry. I never have been a good typist. >acodec is not the same as vcodec. This, at least, I *am* aware of. >I assume you meant to say "vcodec", but it is not what you wrote. Correct. I misspoke. sorry. So, just to be sure now, please confirm that it _is_ indeed the case that "-vcodec copy" is incompatible with the "-vf " option, yes? Assuming so, I go back to what I said yesterday... It would have been minimally polite and decent if ffmpeg had told me that those options are in fact incompatible, you know, like fer instance by doing what 99.99% of all other well written programs do when given incompatible command line options, i.e. issue an error and quit, _without_ producing any output file. At least that way I would know that I needed to delve more deeply into the documentation, and find out what the issue is, rather than believing that command line options I had given were being obeyed, when in fact they were just being silently ignored. It may seem like a small point, but it's an important one, I think. Regards, rfg From mapandrei at gmail.com Fri Oct 11 07:19:42 2013 From: mapandrei at gmail.com (Andrei Petru Mura) Date: Fri, 11 Oct 2013 08:19:42 +0300 Subject: [FFmpeg-user] Issues with avi file In-Reply-To: References: Message-ID: On Thu, Oct 10, 2013 at 12:22 PM, Carl Eugen Hoyos wrote: > Andrei Petru Mura gmail.com> writes: > > > I'm trying to convert an AVI file from a format that > > I don't have codecs for to a usable one using ffmpeg. > > Please provide a sample. > > Carl Eugen > > I put at this link a sample video. Sorry for being so late. I tried to put it on attachment, but it was rejected due to attachment size limit. From cehoyos at ag.or.at Fri Oct 11 10:05:51 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 11 Oct 2013 08:05:51 +0000 (UTC) Subject: [FFmpeg-user] Issues with avi file References: Message-ID: Andrei Petru Mura gmail.com> writes: > I'm trying to convert an AVI file from a format that > I don't have codecs for to a usable one using ffmpeg. Will be fixed in a few hours, thank you for the sample! Carl Eugen From mapandrei at gmail.com Fri Oct 11 10:10:04 2013 From: mapandrei at gmail.com (Andrei Petru Mura) Date: Fri, 11 Oct 2013 11:10:04 +0300 Subject: [FFmpeg-user] Issues with avi file In-Reply-To: References: Message-ID: Do you mean it will be possible to convert this kind of files with ffmpeg? If yes, I'm very glad to hear this. On Fri, Oct 11, 2013 at 11:05 AM, Carl Eugen Hoyos wrote: > Andrei Petru Mura gmail.com> writes: > > > I'm trying to convert an AVI file from a format that > > I don't have codecs for to a usable one using ffmpeg. > > Will be fixed in a few hours, thank you for the sample! > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From bouke at videotoolshed.com Fri Oct 11 12:31:21 2013 From: bouke at videotoolshed.com (Bouke (VideoToolShed)) Date: Fri, 11 Oct 2013 12:31:21 +0200 Subject: [FFmpeg-user] Changing DAR ? References: <99969.1381446877@server1.tristatelogic.com> Message-ID: <068B1B8693FF41FC93C7864BCE24927B@HPKANTOOR> ----- Original Message ----- From: "Ronald F. Guilmette" > > So, just to be sure now, please confirm that it _is_ indeed the case that > "-vcodec copy" is incompatible with the "-vf " option, yes? Perhaps i'm missing the point, but -vcodec copy -aspect 16:9 does work fine. No need for -vf if you just want to change DAR. Bouke From onemda at gmail.com Fri Oct 11 12:52:20 2013 From: onemda at gmail.com (Paul B Mahol) Date: Fri, 11 Oct 2013 10:52:20 +0000 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: <068B1B8693FF41FC93C7864BCE24927B@HPKANTOOR> References: <99969.1381446877@server1.tristatelogic.com> <068B1B8693FF41FC93C7864BCE24927B@HPKANTOOR> Message-ID: On 10/11/13, Bouke (VideoToolShed) wrote: > > ----- Original Message ----- > From: "Ronald F. Guilmette" >> >> So, just to be sure now, please confirm that it _is_ indeed the case that >> "-vcodec copy" is incompatible with the "-vf " option, yes? > > Perhaps i'm missing the point, but -vcodec copy -aspect 16:9 does work > fine. > No need for -vf if you just want to change DAR. Depends on scenario. DAR/SAR can be stored in container and/or bitstream. > > Bouke > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From praveen.vatt at gmail.com Fri Oct 11 13:04:40 2013 From: praveen.vatt at gmail.com (praveen vattipalli) Date: Fri, 11 Oct 2013 16:34:40 +0530 Subject: [FFmpeg-user] how to use ffserver+ ffmpeg Message-ID: hi i connected a webcam to my arm cortexa8 board now i want to stream video in my webpage. for that i compiled ffserver and ffmpeg in buildroot(buildsystem) and ported the root filesystem then i gave ffserver -f /etc/ffserver.conf & it shows as below ffserver version 0.8.12, Copyright (c) 2000-2011 the FFmpeg developers built on Oct 10 2013 11:04:38 with gcc 4.6.2 configuration: --enable-cross-compile --cross-prefix=/home/praveen/phytec_work/cosmic-am335x/buildroot-2013.05/output/host/usr/bin/arm-corten libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libswscale 2. 0. 0 / 2. 0. 0 Sat Jan 1 01:18:29 2000 FFserver started. Sat Jan 1 01:18:29 2000 Launch commandline: ffmpeg http://127.0.0.1:8090/feed1.ffm Sat Jan 1 01:18:30 2000 File '/feed1.ffm' not found Sat Jan 1 01:18:30 2000 127.0.0.1 - - [GET] "/feed1.ffm HTTP/1.1" 404 149 feed1.ffm: Pid 1267 exited with status 256 after 1 seconds Sat Jan 1 01:20:07 2000 192.168.1.7 - - [] " " 200 0 Sat Jan 1 01:20:18 2000 192.168.1.7 - - [] " " 200 0 after that i gave ffmpeg -f video4linux2 -s 640x480 -r 25 -i /dev/video0 ht tp://localhost:8090/feed1.ffm it shows like below ffmpeg version 0.8.12, Copyright (c) 2000-2011 the FFmpeg developers built on Oct 10 2013 11:04:38 with gcc 4.6.2 configuration: --enable-cross-compile --cross-prefix=/home/praveen/phytec_work/cosmic-am335x/buildroot-2013.05/output/host/usr/bin/arm-corten libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libswscale 2. 0. 0 / 2. 0. 0 [video4linux2 @ 0x2f410] The driver changed the time per frame from 1/25 to 1/30 [video4linux2 @ 0x2f410] Estimating duration from bitrate, this may be inaccurate Input #0, video4linux2, from '/dev/video0': Duration: N/A, start: 3122.719003, bitrate: 147456 kb/s Stream #0.0: Video: rawvideo, yuyv422, 640x480, 147456 kb/s, 30 tbr, 1000k tbn, 30 tbc Sat Jan 1 01:32:59 2000 File '/feed1.ffm' not found Sat Jan 1 01:32:59 2000 127.0.0.1 - - [GET] "/feed1.ffm HTTP/1.1" 404 149 [http @ 0x2f920] HTTP error 404 Not Found http://localhost:8090/feed1.ffm: Input/output error this is my ffserver.cong Port 8090 BindAddress 0.0.0.0 MaxHTTPConnections 2000 MaxClients 1000 MaxBandwidth 1000 CustomLog - NoDaemon File /tmp/feed1.ffm FileMaxSize 1G Launch ffmpeg ACL allow 192.168.1.7 192.168.255.255 Format mpeg Feed feed1.ffm VideoBitRate 64 VideoBufferSize 40 VideoFrameRate 3 VideoSize 640x480 can i get any help from you Thanks & Regards Praveen From mapandrei at gmail.com Fri Oct 11 15:08:48 2013 From: mapandrei at gmail.com (Andrei Petru Mura) Date: Fri, 11 Oct 2013 16:08:48 +0300 Subject: [FFmpeg-user] Issues with avi file In-Reply-To: References: Message-ID: Hi Carl, Have you fixed the problem related to GeoVision output video? I took your commit from the git repository, but it does not solve the problem. Thanks in advance! artaxerxe On Fri, Oct 11, 2013 at 11:05 AM, Carl Eugen Hoyos wrote: > Andrei Petru Mura gmail.com> writes: > > > I'm trying to convert an AVI file from a format that > > I don't have codecs for to a usable one using ffmpeg. > > Will be fixed in a few hours, thank you for the 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 Fri Oct 11 15:22:56 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 11 Oct 2013 13:22:56 +0000 (UTC) Subject: [FFmpeg-user] how to use ffserver+ ffmpeg References: Message-ID: praveen vattipalli gmail.com> writes: > ffserver version 0.8.12 This is ancient, please test current git head. Carl Eugen From cehoyos at ag.or.at Fri Oct 11 16:24:46 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 11 Oct 2013 14:24:46 +0000 (UTC) Subject: [FFmpeg-user] Issues with avi file References: Message-ID: Andrei Petru Mura gmail.com> writes: > Have you fixed the problem related to GeoVision output video? Yes, should be fixed now. Sorry for the delay, please avoid top-posting here, Carl Eugen From ricdur2 at gmail.com Fri Oct 11 17:38:55 2013 From: ricdur2 at gmail.com (Richard Duran) Date: Fri, 11 Oct 2013 09:38:55 -0600 Subject: [FFmpeg-user] appending image (jpg) to video (avi)? Message-ID: Hello, I have a sequence of images that I would like to use to create a movie. The issue I have is that the frame rate may be variable, e.g. 1 second gap between image001.jpg and image002.jpg, then a 2 second gap between image002.jpg and image003.jpg. I can use filesystem timestamps and/or the actual file name (which contains a timestamp) to determine the time gap, but I'm not sure how I could go about generating a movie. If there is no solution, my workaround will be to approximate a fixed frame rate by dividing the target length of my video by the number of images. Regards, -richard From lulebo at gmail.com Fri Oct 11 18:33:25 2013 From: lulebo at gmail.com (Carl Lindqvist) Date: Fri, 11 Oct 2013 18:33:25 +0200 Subject: [FFmpeg-user] Multiple outputs worse performance than multiple instances Message-ID: Hello I am trying to optimize my encodings. I have a single input file and transcode it to multiple bitrates with FFmpeg using x264 as codec. At the moment I am starting one instance of FFmpeg for each output. This seemed wasteful, so when I learned of the "multiple outputs" (https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs) in FFmpeg I wanted to try that. My results are very bad. Performance is much lower than starting multiple instances. I have tried putting -threads here and there in the command without getting good cpu usage. The problem is that cpu usage is hovering at 15-20% (machine has 12 cores, 24 threads with hyperthreading). When I run multiple instances I get 98-100% utilization and much higher encode speeds. I would think running one input (and therefore one read/decode/deinterlace) should be much faster. Does anyone have any experience with this? Is it a bug or am I simply doing it wrong? Best regards Carl Lindqvist From jrunta at gmail.com Fri Oct 11 18:44:10 2013 From: jrunta at gmail.com (Jason Runta) Date: Fri, 11 Oct 2013 09:44:10 -0700 Subject: [FFmpeg-user] Multiple outputs worse performance than multiple instances In-Reply-To: References: Message-ID: Can you post your ffmpeg commands? I am doing something similar and I'm not seeing a significant CPU increase. On Fri, Oct 11, 2013 at 9:33 AM, Carl Lindqvist wrote: > Hello > > I am trying to optimize my encodings. I have a single input file and > transcode it to multiple bitrates with FFmpeg using x264 as codec. At > the moment I am starting one instance of FFmpeg for each output. This > seemed wasteful, so when I learned of the "multiple outputs" > (https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs) in FFmpeg > I wanted to try that. My results are very bad. Performance is much > lower than starting multiple instances. I have tried putting -threads > here and there in the command without getting good cpu usage. > > The problem is that cpu usage is hovering at 15-20% (machine has 12 > cores, 24 threads with hyperthreading). When I run multiple instances > I get 98-100% utilization and much higher encode speeds. > > I would think running one input (and therefore one > read/decode/deinterlace) should be much faster. Does anyone have any > experience with this? Is it a bug or am I simply doing it wrong? > > Best regards > Carl Lindqvist > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- *-_-=Jason Runta=-_-* From mediastream at gmail.com Fri Oct 11 19:11:41 2013 From: mediastream at gmail.com (Dennis) Date: Fri, 11 Oct 2013 13:11:41 -0400 Subject: [FFmpeg-user] appending image (jpg) to video (avi)? In-Reply-To: References: Message-ID: Looks like your workaround is a correct approach. But you may try something I was using in overlay filter_complex, a duration variable ":enable=lte(t\,10)". So you can specify different durations for the concat sources, it might work. On Fri, Oct 11, 2013 at 11:38 AM, Richard Duran wrote: > Hello, > > I have a sequence of images that I would like to use to create a movie. The > issue I have is that the frame rate may be variable, e.g. 1 second gap > between image001.jpg and image002.jpg, then a 2 second gap between > image002.jpg and image003.jpg. I can use filesystem timestamps and/or the > actual file name (which contains a timestamp) to determine the time gap, > but I'm not sure how I could go about generating a movie. > > If there is no solution, my workaround will be to approximate a fixed frame > rate by dividing the target length of my video by the number of images. > > Regards, > -richard > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From mediastream at gmail.com Fri Oct 11 19:15:53 2013 From: mediastream at gmail.com (Dennis) Date: Fri, 11 Oct 2013 13:15:53 -0400 Subject: [FFmpeg-user] Correct way to get preview stream with no encoding? In-Reply-To: References: Message-ID: I think you do both in one shot. ffmpeg -i v4l -c:v libx264 -s 640x480 OUTPUT -c copy PREVIEW On Wed, Oct 9, 2013 at 5:35 PM, Jason Runta wrote: > I'm trying to make a program that behaves similar to FMLE where I need to > display a preview of the video from my webcam as well as a preview of the > encoded output. > > I was wondering if anyone could suggest the proper way of getting a preview > stream without any encoding being done to it. > > Do I need to make one ffmpeg call that uses -vcodec copy -acodec copy and > then redirect the output from that into subsequent ffmpeg calls? > > If anyone has a good idea of how to get the preview stream into a C# app > I'm all ears as well =) I'm pretty sure FMLE basically runs on top of > ffmpeg so if they're doing it, I should be able to do it... > > -- > *-_-=Jason Runta=-_-* > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From mediastream at gmail.com Fri Oct 11 19:26:29 2013 From: mediastream at gmail.com (Dennis) Date: Fri, 11 Oct 2013 13:26:29 -0400 Subject: [FFmpeg-user] Required script for 3GP (h.263) In-Reply-To: <20458242f61541d885255beef56bd92e@unisysinfo.in> References: <20458242f61541d885255beef56bd92e@unisysinfo.in> Message-ID: ffmpeg -i source.mp4 -c:v h263 -s 352x288 -b:v 150k -r 25 -b:a 64k -ar 44100 -c:a libfaac output.3gp should work... On Sat, Oct 5, 2013 at 7:09 AM, wrote: > > > Dear All Team > > please suggest me for transcode mp4 to 3gp in > ffmpeg with h.263 codec > > requirements are : > > 1. video codec 4.263 > > 2. > video frame rate 25 > > 3. video bitrate - 150 > > 4. resolution - 320x240 > > > 5. audio bitrate -64k > > 6. sample rate 44100 > > Thanks & Regards > > > Harneet Virk. > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From jrunta at gmail.com Fri Oct 11 19:37:04 2013 From: jrunta at gmail.com (Jason Runta) Date: Fri, 11 Oct 2013 10:37:04 -0700 Subject: [FFmpeg-user] Correct way to get preview stream with no encoding? In-Reply-To: References: Message-ID: Not sure if this is the best way, but here's what I ended up doing: ffmpeg -y -f dshow -threads 3 -s 640x480 -i video="Microsoft LifeCam Cinema":audio="Desktop Microphone (Cinema - Mi" -c:v copy -an -f rawvideo udp://localhost:1234 -f flv -c:v libx264 -tune zerolatency -preset ultrafast -b:v 400k -pix_fmt yuv420p -c:a libmp3lame -q:a 2 -b:a 56k -maxrate 750k "myMovie.flv" and then viewing my video from ffplay: ffplay -s 640x480 -pix_fmt yuyv422 -f rawvideo -i udp://localhost:1234 I'm processing everything off one ffmpeg call here instead of doing subsequent calls to ffmpeg, and specifying to use up to 3 threads to accomplish the raw video output (no audio) and then also encoding and saving the file out. Is there a better way? How about getting my stream to ffplay, is streaming point-to-point over UDP the best option? On Fri, Oct 11, 2013 at 10:15 AM, Dennis wrote: > I think you do both in one shot. > ffmpeg -i v4l -c:v libx264 -s 640x480 OUTPUT -c copy PREVIEW > > > On Wed, Oct 9, 2013 at 5:35 PM, Jason Runta wrote: > > > I'm trying to make a program that behaves similar to FMLE where I need to > > display a preview of the video from my webcam as well as a preview of the > > encoded output. > > > > I was wondering if anyone could suggest the proper way of getting a > preview > > stream without any encoding being done to it. > > > > Do I need to make one ffmpeg call that uses -vcodec copy -acodec copy and > > then redirect the output from that into subsequent ffmpeg calls? > > > > If anyone has a good idea of how to get the preview stream into a C# app > > I'm all ears as well =) I'm pretty sure FMLE basically runs on top of > > ffmpeg so if they're doing it, I should be able to do it... > > _______________________________________________ > > 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 mediastream at gmail.com Fri Oct 11 21:08:43 2013 From: mediastream at gmail.com (Dennis) Date: Fri, 11 Oct 2013 15:08:43 -0400 Subject: [FFmpeg-user] Correct way to get preview stream with no encoding? In-Reply-To: References: Message-ID: I'd go with VLC live encoder and playback, vlc forum is a good place to start. UDP would be least overhead. On Fri, Oct 11, 2013 at 1:37 PM, Jason Runta wrote: > Not sure if this is the best way, but here's what I ended up doing: > > ffmpeg -y -f dshow -threads 3 -s 640x480 -i video="Microsoft LifeCam > Cinema":audio="Desktop Microphone (Cinema - Mi" -c:v copy -an -f rawvideo > udp://localhost:1234 -f flv -c:v libx264 -tune zerolatency -preset > ultrafast -b:v 400k -pix_fmt yuv420p -c:a libmp3lame -q:a 2 -b:a 56k > -maxrate 750k "myMovie.flv" > > and then viewing my video from ffplay: > ffplay -s 640x480 -pix_fmt yuyv422 -f rawvideo -i udp://localhost:1234 > > I'm processing everything off one ffmpeg call here instead of doing > subsequent calls to ffmpeg, and specifying to use up to 3 threads to > accomplish the raw video output (no audio) and then also encoding and > saving the file out. > > Is there a better way? How about getting my stream to ffplay, is streaming > point-to-point over UDP the best option? > > > On Fri, Oct 11, 2013 at 10:15 AM, Dennis wrote: > > > I think you do both in one shot. > > ffmpeg -i v4l -c:v libx264 -s 640x480 OUTPUT -c copy PREVIEW > > > > > > On Wed, Oct 9, 2013 at 5:35 PM, Jason Runta wrote: > > > > > I'm trying to make a program that behaves similar to FMLE where I need > to > > > display a preview of the video from my webcam as well as a preview of > the > > > encoded output. > > > > > > I was wondering if anyone could suggest the proper way of getting a > > preview > > > stream without any encoding being done to it. > > > > > > Do I need to make one ffmpeg call that uses -vcodec copy -acodec copy > and > > > then redirect the output from that into subsequent ffmpeg calls? > > > > > > If anyone has a good idea of how to get the preview stream into a C# > app > > > I'm all ears as well =) I'm pretty sure FMLE basically runs on top of > > > ffmpeg so if they're doing it, I should be able to do it... > > > _______________________________________________ > > > 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-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ckoeber at gmail.com Fri Oct 11 21:25:24 2013 From: ckoeber at gmail.com (Christopher Koeber) Date: Fri, 11 Oct 2013 15:25:24 -0400 Subject: [FFmpeg-user] Request for Advice - Live Streaming with Interweaving of PPT and Computer Files/Etc... Message-ID: Hello,**** I realize this may be a little off topic for the FFMpeg-User list but I thought I would ask for advice on something multimedia related. (It was suggested that I come here by other folks so there's that as well.) I work at an organization which would like to utilize live streaming for some of their events so I am pulling together all of the different items I would need to do this.**** The requirements for the live streaming/broadcasting are these:**** 1. We would like to stream in HD when possible but at a bare minimum we need to stream at 480p to our viewers.**** 2. We need to be able to support up to 1000 viewers. We estimate that we may not reach the amount in the near-term but would need that capability. We will look into other options as needed. 3. We need to be able to interweave video as well as Microsoft PowerPoint slides and other multimedia within the live stream as it happens. **** Here are my notes on what we need so far based on these requirements and what I have noted as what we need:**** Camera >From our requirements we need a camera that has: **** - HQ Audio external microphone AND/OR **** - Ability to have input for external Microphone or sound system. **** - Built in hardware MPEG2 encoder at a minimum. If MPEG4 hardware encoder capable that would be a plus. Based on this I found the Canon XA10. Will this work? **** Link: http://www.usa.canon.com/cusa/professional/products/ professional_cameras/hd_video_cameras/xa10#Features Bandwidth We are currently using a T3 on campus with about 20-40% utilization. Would this be enough to support the 1000 viewers (max)? Software for Recording Live Streaming I have no idea what software we need on this front, and I need advice in this area. I know we need an application or setup that: - Supports Live Streaming of RAW MPEG2/MP4 video.**** - Interweaving of Microsoft PowerPoint Presentations and other media.**** - Saving of data in native (RAW) format to preserve quality. Storage I am probably oversizing here (but you can never go wrong with more) so I am estimating we need a minimum of 5TB of near-line storage available for video, especially for events that take a long time. Offline storage needs can be determined after recording has taken place but we are estimating at least 2-3 times the amount above (making the total offline storage around 20TB for the video and other support files. Internal/External Hosting and Delivery Application/Web Plugin - Depending on demand we can host externally; at this point we would host internally on our website via a module until demand requires us to host externally. - Additional research is needed on the exact application to be used to stream the video. - It was suggested that I should check out YouTube streaming but I don't know how that will play out with trying to link in multimedia from a computer as well as the live stream. - Again, soliciting advice on this item. I guess I put a lot out there to digest; I hope that I can receive assistance of any kind. ** Thank you in advance for any advice given. Regards, Christopher Koeber From rfg at tristatelogic.com Fri Oct 11 21:58:55 2013 From: rfg at tristatelogic.com (Ronald F. Guilmette) Date: Fri, 11 Oct 2013 12:58:55 -0700 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: <068B1B8693FF41FC93C7864BCE24927B@HPKANTOOR> Message-ID: <5752.1381521535@server1.tristatelogic.com> In message <068B1B8693FF41FC93C7864BCE24927B at HPKANTOOR>, "Bouke \(VideoToolShed\)" wrote: > >----- Original Message ----- >From: "Ronald F. Guilmette" >> >> So, just to be sure now, please confirm that it _is_ indeed the case that >> "-vcodec copy" is incompatible with the "-vf " option, yes? > >Perhaps i'm missing the point, but -vcodec copy -aspect 16:9 does work fine. >No need for -vf if you just want to change DAR. I agree that that combination _does_ in fact work for some subset of all of the video formats that ffmpeg handles... although it appeared to me that it does not work for all. I may be wrong about that, but at this moment I don't think so. I will need to do more experiments, I guess, and report hard facts and findings. As regards to the point which you may or may not be missing, as I said, as of 2.0.1, ffmpeg appears to allow both -vcodec copy and also -vf options to be used together, and it just silently ignores one of those, which in my opinion is rather entirely sub-optimal. Also, I believe that there may be cases where using -vf and setsar or setdar may cause unexpected effects, specifically on the aspect ratio that is _not_ being changed. But I have to get in and dig around on the mediainfo sources and try to find out if some of the things it is telling me may be slightly less than true, at least for some video formats. If so, then it is really mediainfo's fault for printing misleading output. Regards, rfg From thierry at lelegard.fr Sat Oct 12 00:08:07 2013 From: thierry at lelegard.fr (Thierry Lelegard) Date: Sat, 12 Oct 2013 00:08:07 +0200 (CEST) Subject: [FFmpeg-user] Announcing QtlMovie, a specialized Qt front-end to FFmpeg Message-ID: <1884560103.71733923.1381529287290.JavaMail.root@spooler3-g27.priv.proxad.net> Dear FFmpeg users, I am pleased to announce the public availability of QtlMovie, a simple specialized graphical front end to FFmpeg using the Qt framework. Why another front-end to FFmpeg? So many different ones already exist. QtlMovie is NOT a general-purpose GUI for FFmpeg, interfacing its rich set of options and filters. Instead, QtlMovie only performs a few repetitive specialized tasks which proved to be difficult or boring with other tools. In short, I developed QtlMovie primarily for my own usage to automate tasks which took me too long and I now share it. So, what is QtlMovie for? It is mostly the answer to the following needs: * I am a movie fan and want to watch movies exclusively in original audio version with subtitles when necessary. * I record many movies from TV (digital TV and MPEG-converted analog recordings) as well as collect other movie files and I want to create DVD's out of them. * I own an iPad and many DVD's and want to watch those DVD's on the iPad. Sounds reasonable? Yes. Or at least I thought so. Sounds simple? Not so simple in fact. Before developing QtlMovie, I needed to use a dozen different tools depending on the type of input and output files: MediaInfo (always a good starting point), AviDemux, ProjectX, VirtualDub, MediaCoder, DVD Decrypter, VOB Merge, DeeVeeDee, Nero, several more or less functional subtitle conversion tools and, for desperate cases, a good old long ffmpeg command line. None of these tools could be removed from the toolbox. There was always a specific case (mostly because of the subtitle formats hell) where one of them was necessary. See some more on that below. Note that I only mention free tools. There may be some magic and expensive tools which do what I want but I am simply not interested. This is why I deciding to unify all of them behind a common GUI which interfaces (but does not hide) ffmpeg and other command line tools. FFmpeg is the key tool which does most of the work. But additional tools are added to extract Teletext subtitles or create DVD file systems and media. A log window shows the generated commands and their output. To understand why QtlMovie can be useful, the log window shows no less than 10 successive commands to generate a DVD media from a TV recording containing Teletext subtitles. Using QtlMovie: Basically, the main workflow of QtlMovie is the following * Open a movie file of any type, including DVD file structures, with any combination and formats of audio, video and subtitles. * Five clicks: 1. select video track, 2. select audio track, 3. select subtitle track, 4. select output type, 5. start. All selections use simple radio buttons in one single window (no complex menus, no drop-down or combo boxes, etc.) * Everything is automated to create either a DVD (MPEG file, ISO image or burn the media, your choice) or an iPad movie file. The resulting output media is basic and simple: one video track with hardcoded subtitles, one audio track, that's all (no menu, no track selection). Why is this complicated ? Interestingly, although the most complex technical task, the video and audio transcoding was never a problem. Most tools handle that gracefully, mostly thanks to back-ends like FFmpeg and its libraries. Here is a list of some technical difficulties I had to face. No traditional tool can manage them all, I needed a combination of tools. And when a solution existed in a tool, I needed to select multiple options and make some calculation each time. I hate to repeat the same or (worse) similar operations when a technical solution could exist to automate them. * Video size, display and pixel aspect ratio. Example: Considering an input video size 1280x536 with pixel aspect ratio 1:1. How do you resize and pad it to obtain a DVD video with size 720x576 and display aspect ratio 16:9? Need some simple but boring math every time. * Identification of audio and subtitle language and properties (standard, forced, for hearing/visual impaired). VOB files from DVD do not carry this information. You have to analyze the .IFO file for that. With some tools, the properties are not clearly reported, making the selection decision more difficult. * Text subtitles. Which format: SRT, SSA, ASS. Which source: a stream in the input file or an external file. How to burn them in the video. AviDemux is mostly OK but unreliable, its support for SRT vs. ASS keeps changing with versions and I faced repeated and irritating crashes. * Teletext subtitles (common in DTTV and IP-TV). The only GUI which can extract them is ProjectX. But it works only on MPEG transport stream files and its GUI is complex and counter-intuitive. * Bitmap subtitles (DVD and DVB) position and size. The video and subtitle frames have sometimes distinct sizes and overlaying them needs some manual adjustments (after hours of nervous breakdown the first time, trying to figure out why those damn subtitles did not show up). * DVD subtitle colors. The VOB files from a DVD contains bitmap subtitles without any color information. The result is ugly and barely watchable subtitles in the video. You have to dig into the .IFO file in the DVD to extract (and convert) the color palette for the subtitle. * And other difficulties I have now forgotten. Well, enough is enough. I just wanted to open a file, 5 clicks, go for a coffee (or a "magret de canard") and later collect my DVD media or iPad movie. So I developed QtlMovie. QtlMovie is a not a sophisticated tool. It does not manipulate video and other complex bitstreams. It simply synchronizes the work of other excellent and complex tools such as FFmpeg. But "simply" is exactly the word that was missing and I hope that QtlMovie will bring it to you. QtlMovie is open source and released under the BSD license. It is developed in C++ using Qt 5 and should work on any platform supporting Qt 5, ffmpeg and the other media tools. QtlMovie is primarily developed on Windows but is also tested on Linux. QtlMovie is available on SourceForge at http://qtlmovie.sourceforge.net/ The source code is available both as one archive file per version and as a git repository. Binary installers for 32-bit and 64-bit Windows are available. These binary installers come pre-packaged with recent versions of ffmpeg, ffprobe, dvdauthor, telxcc, mkisofs and growisofs so that they are self-sufficient for end users. Please report problems using the ticket tracker on the project page at https://sourceforge.net/projects/qtlmovie/ A discussion forum is available. Anonymous postings are enabled but will be moderated first. Registered users of SourceForge may post without restriction. Acknowledgements: I would like to thank the authors of ffmpeg, dvdauthor, telxcc, mkisofs and growisofs. They developed great tools. QtlMovie is just providing the glue... -Thierry From ricdur2 at gmail.com Sat Oct 12 01:15:01 2013 From: ricdur2 at gmail.com (Richard Duran) Date: Fri, 11 Oct 2013 17:15:01 -0600 Subject: [FFmpeg-user] appending image (jpg) to video (avi)? In-Reply-To: References: Message-ID: Thanks for the tip. I'll take a look at that. I was reading about Dirac and libschroedinger, looking for a way to achieve a variable frame rate and also to store my JPEGs without adding more loss than is already there, but I couldn't find any way of actually changing the frame rate between frames. I couldn't even find much documentation besides references to it in General.html and Platform.html. On Fri, Oct 11, 2013 at 11:11 AM, Dennis wrote: > Looks like your workaround is a correct approach. But you may try something > I was using in overlay filter_complex, a duration variable > ":enable=lte(t\,10)". So you can specify different durations for the concat > sources, it might work. > > > On Fri, Oct 11, 2013 at 11:38 AM, Richard Duran wrote: > > > Hello, > > > > I have a sequence of images that I would like to use to create a movie. > The > > issue I have is that the frame rate may be variable, e.g. 1 second gap > > between image001.jpg and image002.jpg, then a 2 second gap between > > image002.jpg and image003.jpg. I can use filesystem timestamps and/or the > > actual file name (which contains a timestamp) to determine the time gap, > > but I'm not sure how I could go about generating a movie. > > > > If there is no solution, my workaround will be to approximate a fixed > frame > > rate by dividing the target length of my video by the number of images. > > > > Regards, > > -richard > > _______________________________________________ > > 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 wangxingchao2011 at gmail.com Sat Oct 12 04:07:40 2013 From: wangxingchao2011 at gmail.com (Wang Xingchao) Date: Sat, 12 Oct 2013 10:07:40 +0800 Subject: [FFmpeg-user] Mosaic issue when playing RMVB with ffmpeg Message-ID: Hi Community, I met an weird issue on Mosaic when playing RMVB video files with ffmpeg. It's 100% reproduced when seek forward/backward, the mosaic video picture would last for ~4 seconds and recover back to normal playback. The issue is reproduced with below environments: - ffplay version 0.8 - android + ffmpeg 1.13 - play some particular rmvb files, not all rmvb files please note this issue doesnot occur with ffplay 2.0 on desktop. The audio codec type is "AAC" inside the rmvb test video file. The only workround for me is to disable audio with "-an" option, this will remove mosaic, and that's why i feel so weird. Here's the log message i met when playing the video with "ffplay", you can see audio decoding fails: "[aac @ 0x7fe188000e20] channel element 0.0 is not allocated=0/2 f=0/0 [aac @ 0x7fe188000e20] SBR was found before the first channel element. [aac @ 0x7fe188000e20] channel element 0.13 is not allocated [aac @ 0x7fe188000e20] SBR was found before the first channel element. [aac @ 0x7fe188000e20] channel element 2.14 is not allocated [aac @ 0x7fe188000e20] channel element 2.12 is not allocated [rv40 @ 0x7fe188003260] First slice header is incorrect, ret=0, si.start 960 [rv40 @ 0x7fe188003260] Invalid decoder state: B-frame without reference data. [rv40 @ 0x7fe188003260] warning: first frame is no keyframe [aac @ 0x7fe188000e20] channel element 0.0 is not allocated=0/42 " Seems the audio packet damaged before sending to decoder, but how does that has impact on video playback? Another clue is about av_seek_frame(), it may seek to wrong positioin with audio enabled, that make video packets damaged too, but i'm not sure about that. I need more guide on such mosaic issue, any comments would be appreciated. thanks --xingchao From wangxingchao2011 at gmail.com Sat Oct 12 05:31:21 2013 From: wangxingchao2011 at gmail.com (Wang Xingchao) Date: Sat, 12 Oct 2013 11:31:21 +0800 Subject: [FFmpeg-user] Mosaic issue when playing RMVB with ffmpeg In-Reply-To: References: Message-ID: Hi All, i have uplaoded the test sample into upload.ffmpeg.org ftp: /incoming/rmvb-mosaic-samples/mother-rmvb-mosaic-test.txt /incoming/rmvb-mosaic-samples/mother-rmvb-mosaic-test.rmvb thanks --xingchao 2013/10/12 Wang Xingchao : > Hi Community, > > I met an weird issue on Mosaic when playing RMVB video files with ffmpeg. > It's 100% reproduced when seek forward/backward, the mosaic video > picture would last for ~4 seconds and recover back to normal playback. > > The issue is reproduced with below environments: > - ffplay version 0.8 > - android + ffmpeg 1.13 > - play some particular rmvb files, not all rmvb files > > please note this issue doesnot occur with ffplay 2.0 on desktop. > > The audio codec type is "AAC" inside the rmvb test video file. The > only workround for me is to disable audio with "-an" option, this will > remove mosaic, and that's why i feel so weird. > > Here's the log message i met when playing the video with "ffplay", you > can see audio decoding fails: > "[aac @ 0x7fe188000e20] channel element 0.0 is not allocated=0/2 f=0/0 > [aac @ 0x7fe188000e20] SBR was found before the first channel element. > [aac @ 0x7fe188000e20] channel element 0.13 is not allocated > [aac @ 0x7fe188000e20] SBR was found before the first channel element. > [aac @ 0x7fe188000e20] channel element 2.14 is not allocated > [aac @ 0x7fe188000e20] channel element 2.12 is not allocated > [rv40 @ 0x7fe188003260] First slice header is incorrect, ret=0, si.start 960 > [rv40 @ 0x7fe188003260] Invalid decoder state: B-frame without reference data. > [rv40 @ 0x7fe188003260] warning: first frame is no keyframe > [aac @ 0x7fe188000e20] channel element 0.0 is not allocated=0/42 > " > Seems the audio packet damaged before sending to decoder, but how does > that has impact on video playback? > > Another clue is about av_seek_frame(), it may seek to wrong positioin > with audio enabled, that make video packets damaged too, but i'm not > sure about that. > > I need more guide on such mosaic issue, any comments would be appreciated. > > thanks > --xingchao From cehoyos at ag.or.at Sat Oct 12 08:33:17 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 12 Oct 2013 06:33:17 +0000 (UTC) Subject: [FFmpeg-user] Mosaic issue when playing RMVB with ffmpeg References: Message-ID: Wang Xingchao gmail.com> writes: > The issue is reproduced with below environments: > - ffplay version 0.8 > please note this issue doesnot occur with ffplay 2.0 > on desktop. Don't you agree that there is nothing that we can do about this issue or do I miss something? Carl Eugen From cehoyos at ag.or.at Sat Oct 12 08:34:06 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 12 Oct 2013 06:34:06 +0000 (UTC) Subject: [FFmpeg-user] Multiple outputs worse performance than multiple instances References: Message-ID: Carl Lindqvist gmail.com> writes: > My results are very bad. Command line and complete, uncut console output missing. Carl Eugen From cehoyos at ag.or.at Sat Oct 12 08:35:43 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 12 Oct 2013 06:35:43 +0000 (UTC) Subject: [FFmpeg-user] Correct way to get preview stream with no encoding? References: Message-ID: Jason Runta gmail.com> writes: > I'm trying to make a program that behaves similar to FMLE > where I need to display a preview of the video from my > webcam as well as a preview of the encoded output. What's wrong with -f sdl ? Carl Eugen From cehoyos at ag.or.at Sat Oct 12 08:37:39 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 12 Oct 2013 06:37:39 +0000 (UTC) Subject: [FFmpeg-user] Use cropdetect only every xth frame References: <31395406.M0gE8IemH8@gentoo> <5377384.STYxkmvPO0@gentoo> <1635729.v0BkWSdRbW@gentoo> Message-ID: Gerion Entrup t-online.de> writes: > > You can use -skip_frame (see the documentation). > > Does not change that much: Depending on your use-case, this is impossible. (It of course cannot help for intra-only codecs) > $ time ffmpeg -skip_frame nokey -i test.mp4 -filter:v > select='not(mod(n\,10))',cropdetect > -filter:a volumedetect -map 0 -f null > /dev/null 2>/dev/null Complete, uncut console output missing. Carl Eugen From lingjiujianke at gmail.com Sat Oct 12 10:39:02 2013 From: lingjiujianke at gmail.com (=?GB2312?B?wfUg4ao=?=) Date: Sat, 12 Oct 2013 16:39:02 +0800 Subject: [FFmpeg-user] Mosaic issue when playing RMVB with ffmpeg In-Reply-To: References: Message-ID: <391A2135-6064-4903-885A-1375510464C2@gmail.com> ? 2013-10-12???2:33?Carl Eugen Hoyos ??? > Wang Xingchao gmail.com> writes: > >> The issue is reproduced with below environments: >> - ffplay version 0.8 > >> please note this issue doesnot occur with ffplay 2.0 >> on desktop. > > Don't you agree that there is nothing that we can do > about this issue or do I miss something? > Hi Carl, I can reproduce the problem, perhaps this is ffplay bug? Playback the rmvb video file with B-frame, And use the the full output is bellow: [StevenLiu at liudeMacBook-Pro ffmpeg]$ cat output.log ffplay version N-56531-g6d61a91 Copyright (c) 2003-2013 the FFmpeg developers built on Oct 9 2013 18:01:50 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) configuration: libavutil 52. 45.100 / 52. 45.100 libavcodec 55. 33.100 / 55. 33.100 libavformat 55. 18.102 / 55. 18.102 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 86.102 / 3. 86.102 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 [rm @ 0x7fd34180dc00] Invalid stream index 2 for index at pos 120266381 Input #0, rm, from '/Users/StevenLiu/Movies/Fuck_mosaic.rmvb': Metadata: title : unknown author : tiger-king-soft copyright : comment : ASMRuleBook : #($Bandwidth >= 0),Stream1Bandwidth = 96468, Stream0Bandwidth = 653532; Audiences : 750k Download (VBR); audioMode : music Creation Date : 1/5/2012 15:41:11 Generated By : Helix Producer SDK 10.0 for Windows, Build 10.0.0.545 Modification Date: 1/5/2012 15:41:11 videoMode : normal Duration: 00:20:59.45, start: 0.000000, bitrate: 763 kb/s Stream #0:0: Video: rv40 (RV40 / 0x30345652), yuv420p, 640x480, 662 kb/s, 25 fps, 25 tbr, 1k tbn, 1k tbc Stream #0:1: Audio: cook (cook / 0x6B6F6F63), 44100 Hz, stereo, fltp, 96 kb/s [rv40 @ 0x7fd3418f5000] Invalid decoder state: B-frame without reference data. [rv40 @ 0x7fd341912a00] Invalid decoder state: B-frame without reference data. [rv40 @ 0x7fd341930400] Invalid decoder state: B-frame without reference data. [rv40 @ 0x7fd3418f5000] Invalid decoder state: B-frame without reference data. [rv40 @ 0x7fd341912a00] Invalid decoder state: B-frame without reference data. [rv40 @ 0x7fd341930400] Invalid decoder state: B-frame without reference data. Truncating packet of size 990377780 to 117756985B sq= 0B f=0/0 [rm @ 0x7fd34180dc00] Impossibly sized packet [rv40 @ 0x7fd3418d8200] Changing dimensions to 160x120 [rv40 @ 0x7fd3418d8200] marking unfished frame as finished [rv40 @ 0x7fd3418d8200] concealing 79 DC, 79 AC, 79 MV errors in I frame [rv40 @ 0x7fd3418f5000] First slice header is incorrect [rv40 @ 0x7fd341912a00] Context scratch buffers could not be allocated due to unknown size. [rv40 @ 0x7fd341912a00] Invalid decoder state: B-frame without reference data. [rv40 @ 0x7fd341930400] Context scratch buffers could not be allocated due to unknown size. [rv40 @ 0x7fd341930400] Invalid decoder state: B-frame without reference data. [rv40 @ 0x7fd34194de00] Context scratch buffers could not be allocated due to unknown size. [rv40 @ 0x7fd34194de00] Invalid decoder state: B-frame without reference data. [rv40 @ 0x7fd3418f5000] concealing 80 DC, 80 AC, 80 MV errors in B frame [rv40 @ 0x7fd341912a00] concealing 80 DC, 80 AC, 80 MV errors in B frame [rv40 @ 0x7fd341930400] concealing 80 DC, 80 AC, 80 MV errors in B frame [rv40 @ 0x7fd341912a00] Dquant for B-frame 137KB sq= 0B f=0/0 Last message repeated 1 times [rv40 @ 0x7fd3418d8200] Changing dimensions to 640x480 0B f=0/0 [rv40 @ 0x7fd3418f5000] Invalid decoder state: B-frame without reference data. [rv40 @ 0x7fd341912a00] Context scratch buffers could not be allocated due to unknown size. Truncating packet of size 990377780 to 117756985B sq= 0B f=0/0 [rm @ 0x7fd34180dc00] Impossibly sized packet [rv40 @ 0x7fd3418d8200] Changing dimensions to 160x120 [rv40 @ 0x7fd3418d8200] marking unfished frame as finished [rv40 @ 0x7fd3418d8200] concealing 79 DC, 79 AC, 79 MV errors in I frame [rv40 @ 0x7fd3418f5000] First slice header is incorrect [rv40 @ 0x7fd341912a00] Context scratch buffers could not be allocated due to unknown size. [rv40 @ 0x7fd341912a00] Invalid decoder state: B-frame without reference data. [rv40 @ 0x7fd341930400] Context scratch buffers could not be allocated due to unknown size. [rv40 @ 0x7fd341930400] Invalid decoder state: B-frame without reference data. [rv40 @ 0x7fd34194de00] Context scratch buffers could not be allocated due to unknown size. [rv40 @ 0x7fd34194de00] Invalid decoder state: B-frame without reference data. [rv40 @ 0x7fd3418f5000] concealing 80 DC, 80 AC, 80 MV errors in B frame [rv40 @ 0x7fd341912a00] concealing 80 DC, 80 AC, 80 MV errors in B frame [rv40 @ 0x7fd341930400] concealing 80 DC, 80 AC, 80 MV errors in B frame [rv40 @ 0x7fd341912a00] Dquant for B-frame= 137KB sq= 0B f=0/0 Last message repeated 1 times [rv40 @ 0x7fd3418d8200] Changing dimensions to 640x480 0B f=0/0 [rv40 @ 0x7fd3418f5000] Invalid decoder state: B-frame without reference data. [rv40 @ 0x7fd341912a00] Context scratch buffers could not be allocated due to unknown size. 1254.33 A-V: -0.005 fd= 3 aq= 2KB vq= 78KB sq= 0B f=0/0 [StevenLiu at liudeMacBook-Pro ffmpeg]$ I just playing the rmvb file and press "Left key" . Thanks From gerion.entrup at t-online.de Sat Oct 12 12:00:28 2013 From: gerion.entrup at t-online.de (Gerion Entrup) Date: Sat, 12 Oct 2013 12:00:28 +0200 Subject: [FFmpeg-user] Use cropdetect only every xth frame In-Reply-To: References: <31395406.M0gE8IemH8@gentoo> <1635729.v0BkWSdRbW@gentoo> Message-ID: <5881800.tH0TMQkF6z@gentoo> Am Samstag, 12. Oktober 2013, 06:37:39 schrieb Carl Eugen Hoyos: > Gerion Entrup t-online.de> writes: > > > You can use -skip_frame (see the documentation). > > > > Does not change that much: > Depending on your use-case, this is impossible. > (It of course cannot help for intra-only codecs) Don't get what you mean. I was wondering, too, that the time was more less the same. > > > $ time ffmpeg -skip_frame nokey -i test.mp4 -filter:v > > select='not(mod(n\,10))',cropdetect > > -filter:a volumedetect -map 0 -f null > > /dev/null 2>/dev/null > > Complete, uncut console output missing. $ ffmpeg -skip_frame nokey -i test.mp4 -filter:v select='not(mod(n\,10))',cropdetect -filter:a volumedetect -map 0 -f null /dev/null 2>&1 | wc -l 9939 Very much ;). See http://pastebin.kde.org/phwvakpgq for the complete output or shall I include it in the mail? Gerion From cehoyos at ag.or.at Sat Oct 12 12:32:14 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 12 Oct 2013 10:32:14 +0000 (UTC) Subject: [FFmpeg-user] Use cropdetect only every xth frame References: <31395406.M0gE8IemH8@gentoo> <1635729.v0BkWSdRbW@gentoo> <5881800.tH0TMQkF6z@gentoo> Message-ID: Gerion Entrup t-online.de> writes: > > Complete, uncut console output missing. > $ ffmpeg -skip_frame nokey -i test.mp4 -filter:v > select='not(mod(n\,10))',cropdetect -filter:a volumedetect > -map 0 -f null /dev/null 2>&1 | wc -l > 9939 > > Very much ;). See http://pastebin.kde.org/phwvakpgq for > the complete output or shall I include it in the mail? Definitely. It is of course ok to trim the output. The problem is that at least the first 50 and the last 50 lines are needed, and that is what nearly everybody cuts, so I have to ask for "complete, uncut output" (but the "concealing" lines are of course less important if some of them are still visible. Consider testing again without the filters, I cannot rule out that the filters take longer than the decoding. After a quick look, I fear that a sample will be needed;-( While this is probably not the reason for your problem, you should really fix your configure line, or is that what Gentoo produces? It honestly looks broken... Carl Eugen From wangxingchao2011 at gmail.com Sat Oct 12 12:41:13 2013 From: wangxingchao2011 at gmail.com (Wang Xingchao) Date: Sat, 12 Oct 2013 18:41:13 +0800 Subject: [FFmpeg-user] Mosaic issue when playing RMVB with ffmpeg In-Reply-To: References: Message-ID: Hi Carl, 2013/10/12 Carl Eugen Hoyos : > Wang Xingchao gmail.com> writes: > >> The issue is reproduced with below environments: >> - ffplay version 0.8 > >> please note this issue doesnot occur with ffplay 2.0 >> on desktop. > > Don't you agree that there is nothing that we can do > about this issue or do I miss something? i wonder what caused this issue becuase i'm using ffmpeg(1.1.3) on android which met the problem. And seems this issue was fixed since V0.8, so knowing more background about this issue is helpful. thanks --xingchao > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From wangxingchao2011 at gmail.com Sat Oct 12 12:42:06 2013 From: wangxingchao2011 at gmail.com (Wang Xingchao) Date: Sat, 12 Oct 2013 18:42:06 +0800 Subject: [FFmpeg-user] Mosaic issue when playing RMVB with ffmpeg In-Reply-To: <391A2135-6064-4903-885A-1375510464C2@gmail.com> References: <391A2135-6064-4903-885A-1375510464C2@gmail.com> Message-ID: 2013/10/12 ? ? : > > ? 2013-10-12???2:33?Carl Eugen Hoyos ??? > > Wang Xingchao gmail.com> writes: > > The issue is reproduced with below environments: > - ffplay version 0.8 > > > please note this issue doesnot occur with ffplay 2.0 > on desktop. > > > Don't you agree that there is nothing that we can do > about this issue or do I miss something? > > > Hi Carl, > > I can reproduce the problem, perhaps this is ffplay bug? Thanks your test, Qi. --xingchao > Playback the rmvb video file with B-frame, And use the > the full output is bellow: > > [StevenLiu at liudeMacBook-Pro ffmpeg]$ cat output.log > ffplay version N-56531-g6d61a91 Copyright (c) 2003-2013 the FFmpeg > developers > built on Oct 9 2013 18:01:50 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) > configuration: > libavutil 52. 45.100 / 52. 45.100 > libavcodec 55. 33.100 / 55. 33.100 > libavformat 55. 18.102 / 55. 18.102 > libavdevice 55. 3.100 / 55. 3.100 > libavfilter 3. 86.102 / 3. 86.102 > libswscale 2. 5.100 / 2. 5.100 > libswresample 0. 17.103 / 0. 17.103 > [rm @ 0x7fd34180dc00] Invalid stream index 2 for index at pos 120266381 > Input #0, rm, from '/Users/StevenLiu/Movies/Fuck_mosaic.rmvb': > Metadata: > title : unknown > author : tiger-king-soft > copyright : > comment : > ASMRuleBook : #($Bandwidth >= 0),Stream1Bandwidth = 96468, > Stream0Bandwidth = 653532; > Audiences : 750k Download (VBR); > audioMode : music > Creation Date : 1/5/2012 15:41:11 > Generated By : Helix Producer SDK 10.0 for Windows, Build 10.0.0.545 > Modification Date: 1/5/2012 15:41:11 > videoMode : normal > Duration: 00:20:59.45, start: 0.000000, bitrate: 763 kb/s > Stream #0:0: Video: rv40 (RV40 / 0x30345652), yuv420p, 640x480, 662 > kb/s, 25 fps, 25 tbr, 1k tbn, 1k tbc > Stream #0:1: Audio: cook (cook / 0x6B6F6F63), 44100 Hz, stereo, fltp, 96 > kb/s > [rv40 @ 0x7fd3418f5000] Invalid decoder state: B-frame without reference > data. > [rv40 @ 0x7fd341912a00] Invalid decoder state: B-frame without reference > data. > [rv40 @ 0x7fd341930400] Invalid decoder state: B-frame without reference > data. > [rv40 @ 0x7fd3418f5000] Invalid decoder state: B-frame without reference > data. > [rv40 @ 0x7fd341912a00] Invalid decoder state: B-frame without reference > data. > [rv40 @ 0x7fd341930400] Invalid decoder state: B-frame without reference > data. > Truncating packet of size 990377780 to 117756985B sq= 0B f=0/0 > [rm @ 0x7fd34180dc00] Impossibly sized packet > [rv40 @ 0x7fd3418d8200] Changing dimensions to 160x120 > [rv40 @ 0x7fd3418d8200] marking unfished frame as finished > [rv40 @ 0x7fd3418d8200] concealing 79 DC, 79 AC, 79 MV errors in I frame > [rv40 @ 0x7fd3418f5000] First slice header is incorrect > [rv40 @ 0x7fd341912a00] Context scratch buffers could not be allocated due > to unknown size. > [rv40 @ 0x7fd341912a00] Invalid decoder state: B-frame without reference > data. > [rv40 @ 0x7fd341930400] Context scratch buffers could not be allocated due > to unknown size. > [rv40 @ 0x7fd341930400] Invalid decoder state: B-frame without reference > data. > [rv40 @ 0x7fd34194de00] Context scratch buffers could not be allocated due > to unknown size. > [rv40 @ 0x7fd34194de00] Invalid decoder state: B-frame without reference > data. > [rv40 @ 0x7fd3418f5000] concealing 80 DC, 80 AC, 80 MV errors in B frame > [rv40 @ 0x7fd341912a00] concealing 80 DC, 80 AC, 80 MV errors in B frame > [rv40 @ 0x7fd341930400] concealing 80 DC, 80 AC, 80 MV errors in B frame > [rv40 @ 0x7fd341912a00] Dquant for B-frame 137KB sq= 0B f=0/0 > Last message repeated 1 times > [rv40 @ 0x7fd3418d8200] Changing dimensions to 640x480 0B f=0/0 > [rv40 @ 0x7fd3418f5000] Invalid decoder state: B-frame without reference > data. > [rv40 @ 0x7fd341912a00] Context scratch buffers could not be allocated due > to unknown size. > Truncating packet of size 990377780 to 117756985B sq= 0B f=0/0 > [rm @ 0x7fd34180dc00] Impossibly sized packet > [rv40 @ 0x7fd3418d8200] Changing dimensions to 160x120 > [rv40 @ 0x7fd3418d8200] marking unfished frame as finished > [rv40 @ 0x7fd3418d8200] concealing 79 DC, 79 AC, 79 MV errors in I frame > [rv40 @ 0x7fd3418f5000] First slice header is incorrect > [rv40 @ 0x7fd341912a00] Context scratch buffers could not be allocated due > to unknown size. > [rv40 @ 0x7fd341912a00] Invalid decoder state: B-frame without reference > data. > [rv40 @ 0x7fd341930400] Context scratch buffers could not be allocated due > to unknown size. > [rv40 @ 0x7fd341930400] Invalid decoder state: B-frame without reference > data. > [rv40 @ 0x7fd34194de00] Context scratch buffers could not be allocated due > to unknown size. > [rv40 @ 0x7fd34194de00] Invalid decoder state: B-frame without reference > data. > [rv40 @ 0x7fd3418f5000] concealing 80 DC, 80 AC, 80 MV errors in B frame > [rv40 @ 0x7fd341912a00] concealing 80 DC, 80 AC, 80 MV errors in B frame > [rv40 @ 0x7fd341930400] concealing 80 DC, 80 AC, 80 MV errors in B frame > [rv40 @ 0x7fd341912a00] Dquant for B-frame= 137KB sq= 0B f=0/0 > Last message repeated 1 times > [rv40 @ 0x7fd3418d8200] Changing dimensions to 640x480 0B f=0/0 > [rv40 @ 0x7fd3418f5000] Invalid decoder state: B-frame without reference > data. > [rv40 @ 0x7fd341912a00] Context scratch buffers could not be allocated due > to unknown size. > 1254.33 A-V: -0.005 fd= 3 aq= 2KB vq= 78KB sq= 0B f=0/0 > [StevenLiu at liudeMacBook-Pro ffmpeg]$ > > > I just playing the rmvb file and press "Left key" . > > > > Thanks From gerion.entrup at t-online.de Sat Oct 12 12:58:48 2013 From: gerion.entrup at t-online.de (Gerion Entrup) Date: Sat, 12 Oct 2013 12:58:48 +0200 Subject: [FFmpeg-user] Use cropdetect only every xth frame In-Reply-To: References: <31395406.M0gE8IemH8@gentoo> <5881800.tH0TMQkF6z@gentoo> Message-ID: <3019311.rNPLnGjsfr@gentoo> Am Samstag, 12. Oktober 2013, 10:32:14 schrieb Carl Eugen Hoyos: > Gerion Entrup t-online.de> writes: > > > Complete, uncut console output missing. > > > > $ ffmpeg -skip_frame nokey -i test.mp4 -filter:v > > select='not(mod(n\,10))',cropdetect -filter:a volumedetect > > -map 0 -f null /dev/null 2>&1 | wc -l > > 9939 > > > > Very much ;). See http://pastebin.kde.org/phwvakpgq for > > the complete output or shall I include it in the mail? > > Definitely. > It is of course ok to trim the output. The problem is that > at least the first 50 and the last 50 lines are needed, > and that is what nearly everybody cuts, so I have to ask > for "complete, uncut output" (but the "concealing" lines > are of course less important if some of them are still > visible. Ok. I will try to paste meaningful :). > > Consider testing again without the filters, I cannot > rule out that the filters take longer than the decoding. I will do. > > After a quick look, I fear that a sample will be needed;-( I first try it with other videos and codecs. Maybe this helps. > While this is probably not the reason for your problem, > you should really fix your configure line, or is that > what Gentoo produces? It honestly looks broken... This is that, what Gentoo produces :). These are my activated useflags: X aac aacplus alsa bluray bzip2 cdio celt encode hardcoded-tables iconv jack ladspa mmx mmxext mp3 network opus quvi sdl speex ssse3 theora threads truetype vaapi vdpau vorbis vpx wavpack x264 xvid zlib If you want to go deeper into this, take a look at http://sources.gentoo.org/cgi-bin/viewvc.cgi/gentoo-x86/media-video/ffmpeg/ffmpeg-9999.ebuild?view=markup This is the ebuild, I use, to compile the Git Head. The configuration is done in src_configure (ebuilds are shellscripts more less). If you can say, what exactly look strange, I can also do this. The configuration seems to work so far. I didn't have any problems. From bouke at videotoolshed.com Sat Oct 12 13:38:48 2013 From: bouke at videotoolshed.com (Bouke (VideoToolShed)) Date: Sat, 12 Oct 2013 13:38:48 +0200 Subject: [FFmpeg-user] Quicktime track disabled? Message-ID: <07458DBFC58D469AA0D3CD221946360D@HPKANTOOR> boukes-Mac-Pro-2:~ bouke$ /Volumes/'code/Supersens/xtras/ffmpeg' -i /Volumes/fff2bouke/problems/prores/watch/untitled.mov -r 25 -vcodec copy -acodec copy -map 0:0 -map 0:1 -map 0:2 -aspect 16:9 /Volumes/'fff2bouke/out/test.mov' ffmpeg version 2.0-tessus Copyright (c) 2000-2013 the FFmpeg developers built on Jul 11 2013 00:54:32 with llvm-gcc 4.2.1 (LLVM build 2336.1.00) configuration: --prefix=/Users/tessus/data/ext/ffmpeg/sw --as=yasm --extra-version=tessus --disable-shared --enable-static --disable-ffplay --enable-gpl --enable-pthreads --enable-postproc --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libspeex --enable-bzlib --enable-zlib --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libxavs --enable-version3 --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvpx --enable-libgsm --enable-libopus --enable-fontconfig --enable-libfreetype --enable-libass --enable-filters --enable-runtime-cpudetect libavutil 52. 38.100 / 52. 38.100 libavcodec 55. 18.102 / 55. 18.102 libavformat 55. 12.100 / 55. 12.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 79.101 / 3. 79.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 Guessed Channel Layout for Input Stream #0.1 : stereo Guessed Channel Layout for Input Stream #0.2 : stereo Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Volumes/fff2bouke/problems/prores/watch/untitled.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2013-10-12 10:29:45 Duration: 00:00:08.84, start: 0.000000, bitrate: 71889 kb/s Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le, 720x576, 68812 kb/s, SAR 59:54 DAR 295:216, 25 fps, 25 tbr, 25 tbn, 25 tbc Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler timecode : 00:00:00:00 Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, s16, 1536 kb/s Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler Stream #0:2(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, s16, 1536 kb/s Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler Stream #0:3(eng): Data: none (tmcd / 0x64636D74) Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler timecode : 00:00:00:00 File '/Volumes/fff2bouke/out/test.mov' already exists. Overwrite ? [y/N] y Overriding aspect ratio with stream copy may produce invalid files Output #0, mov, to '/Volumes/fff2bouke/out/test.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt encoder : Lavf55.12.100 Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 68812 kb/s, 25 fps, 12800 tbn, 25 tbc Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler timecode : 00:00:00:00 Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, 1536 kb/s Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler Stream #0:2(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, 1536 kb/s Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Stream #0:2 -> #0:2 (copy) Press [q] to stop, [?] for help frame= 221 fps=0.0 q=-1.0 Lsize= 77578kB time=00:00:08.84 bitrate=71891.6kbits/s video:74256kB audio:3315kB subtitle:0 global headers:0kB muxing overhead 0.009485% boukes-Mac-Pro-2:~ bouke$ /Volumes/'code/Supersens/xtras/ffmpeg' -i /Volumes/fff2bouke/problems/prores/watch/untitled.mov -r 25 -vcodec copy -acodec copy -map 0:0 -map 0:1 -map 0:2 /Volumes/'fff2bouke/out/test.mov' ffmpeg version 2.0-tessus Copyright (c) 2000-2013 the FFmpeg developers built on Jul 11 2013 00:54:32 with llvm-gcc 4.2.1 (LLVM build 2336.1.00) configuration: --prefix=/Users/tessus/data/ext/ffmpeg/sw --as=yasm --extra-version=tessus --disable-shared --enable-static --disable-ffplay --enable-gpl --enable-pthreads --enable-postproc --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libspeex --enable-bzlib --enable-zlib --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libxavs --enable-version3 --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvpx --enable-libgsm --enable-libopus --enable-fontconfig --enable-libfreetype --enable-libass --enable-filters --enable-runtime-cpudetect libavutil 52. 38.100 / 52. 38.100 libavcodec 55. 18.102 / 55. 18.102 libavformat 55. 12.100 / 55. 12.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 79.101 / 3. 79.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 Guessed Channel Layout for Input Stream #0.1 : stereo Guessed Channel Layout for Input Stream #0.2 : stereo Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Volumes/fff2bouke/problems/prores/watch/untitled.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2013-10-12 10:29:45 Duration: 00:00:08.84, start: 0.000000, bitrate: 71889 kb/s Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le, 720x576, 68812 kb/s, SAR 59:54 DAR 295:216, 25 fps, 25 tbr, 25 tbn, 25 tbc Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler timecode : 00:00:00:00 Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, s16, 1536 kb/s Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler Stream #0:2(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, s16, 1536 kb/s Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler Stream #0:3(eng): Data: none (tmcd / 0x64636D74) Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler timecode : 00:00:00:00 File '/Volumes/fff2bouke/out/test.mov' already exists. Overwrite ? [y/N] y Output #0, mov, to '/Volumes/fff2bouke/out/test.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt encoder : Lavf55.12.100 Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le, 720x576 [SAR 59:54 DAR 295:216], q=2-31, 68812 kb/s, 25 fps, 12800 tbn, 25 tbc Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler timecode : 00:00:00:00 Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, 1536 kb/s Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler Stream #0:2(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, 1536 kb/s Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Stream #0:2 -> #0:2 (copy) Press [q] to stop, [?] for help frame= 221 fps= 83 q=-1.0 Lsize= 77578kB time=00:00:08.84 bitrate=71891.6kbits/s video:74256kB audio:3315kB subtitle:0 global headers:0kB muxing overhead 0.009485% boukes-Mac-Pro-2:~ bouke$ clear boukes-Mac-Pro-2:~ bouke$ /Volumes/'code/Supersens/xtras/ffmpeg' -i /Volumes/fff2bouke/problems/prores/watch/untitled.mov -r 25 -vcodec copy -acodec copy -map 0:0 -map 0:1 -map 0:2 -aspect 16:9 /Volumes/'fff2bouke/out/test.mov' ffmpeg version 2.0-tessus Copyright (c) 2000-2013 the FFmpeg developers built on Jul 11 2013 00:54:32 with llvm-gcc 4.2.1 (LLVM build 2336.1.00) configuration: --prefix=/Users/tessus/data/ext/ffmpeg/sw --as=yasm --extra-version=tessus --disable-shared --enable-static --disable-ffplay --enable-gpl --enable-pthreads --enable-postproc --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libspeex --enable-bzlib --enable-zlib --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libxavs --enable-version3 --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvpx --enable-libgsm --enable-libopus --enable-fontconfig --enable-libfreetype --enable-libass --enable-filters --enable-runtime-cpudetect libavutil 52. 38.100 / 52. 38.100 libavcodec 55. 18.102 / 55. 18.102 libavformat 55. 12.100 / 55. 12.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 79.101 / 3. 79.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 Guessed Channel Layout for Input Stream #0.1 : stereo Guessed Channel Layout for Input Stream #0.2 : stereo Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Volumes/fff2bouke/problems/prores/watch/untitled.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2013-10-12 10:29:45 Duration: 00:00:08.84, start: 0.000000, bitrate: 71889 kb/s Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le, 720x576, 68812 kb/s, SAR 59:54 DAR 295:216, 25 fps, 25 tbr, 25 tbn, 25 tbc Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler timecode : 00:00:00:00 Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, s16, 1536 kb/s Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler Stream #0:2(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, s16, 1536 kb/s Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler Stream #0:3(eng): Data: none (tmcd / 0x64636D74) Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler timecode : 00:00:00:00 File '/Volumes/fff2bouke/out/test.mov' already exists. Overwrite ? [y/N] y Overriding aspect ratio with stream copy may produce invalid files Output #0, mov, to '/Volumes/fff2bouke/out/test.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt encoder : Lavf55.12.100 Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 68812 kb/s, 25 fps, 12800 tbn, 25 tbc Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler timecode : 00:00:00:00 Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, 1536 kb/s Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler Stream #0:2(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, 1536 kb/s Metadata: creation_time : 2013-10-12 10:29:45 handler_name : Apple Alias Data Handler Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Stream #0:2 -> #0:2 (copy) Press [q] to stop, [?] for help frame= 221 fps= 84 q=-1.0 Lsize= 77578kB time=00:00:08.84 bitrate=71891.6kbits/s video:74256kB audio:3315kB subtitle:0 global headers:0kB muxing overhead 0.009485% From mapandrei at gmail.com Sat Oct 12 17:15:38 2013 From: mapandrei at gmail.com (Andrei Petru Mura) Date: Sat, 12 Oct 2013 18:15:38 +0300 Subject: [FFmpeg-user] Issues with compiling ffmpeg git project in Eclipse, LinuxOS Message-ID: Hello, I got FFMpeg from repository and tried to compile it in eclipse, but I have this error that I can't get rid of it: ?F_NOCACHE? undeclared (first use in this function) >From what I read on Internet this is a fcntl variable available for BSD systems. I'm running the compile process in a Linux OS (Ubuntu 13). Can anybody give me a hint about this issue and how to solve it? Thanks in advance. Andrei M. From cehoyos at ag.or.at Sat Oct 12 19:16:33 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 12 Oct 2013 17:16:33 +0000 (UTC) Subject: [FFmpeg-user] Issues with compiling ffmpeg git project in Eclipse, LinuxOS References: Message-ID: Andrei Petru Mura gmail.com> writes: > I got FFMpeg from repository and tried to compile it in eclipse, I may misunderstand but FFmpeg is generally compiled by running the configure script from a shell and running make when configure succeeded. Which configure line did you use? > but I have this error that I can't get rid of it: > > ?F_NOCACHE? undeclared (first use in this function) This looks incomplete... Carl Eugen From gerion.entrup at t-online.de Sat Oct 12 20:06:34 2013 From: gerion.entrup at t-online.de (Gerion Entrup) Date: Sat, 12 Oct 2013 20:06:34 +0200 Subject: [FFmpeg-user] Use cropdetect only every xth frame In-Reply-To: <3019311.rNPLnGjsfr@gentoo> References: <31395406.M0gE8IemH8@gentoo> <3019311.rNPLnGjsfr@gentoo> Message-ID: <3653663.LocUtV70gz@gentoo> Am Samstag, 12. Oktober 2013, 12:58:48 schrieb Gerion Entrup: > Am Samstag, 12. Oktober 2013, 10:32:14 schrieb Carl Eugen Hoyos: > > Consider testing again without the filters, I cannot > > rule out that the filters take longer than the decoding. > > I will do. > > > After a quick look, I fear that a sample will be needed;-( > > I first try it with other videos and codecs. Maybe this helps. I took the Big Bug Bunnys HD encodes as samples and there it works as expected: $ for i in *; do ffmpeg -i "$i"; echo "without skip, with filters"; time ffmpeg - i "$i" -filter:v cropdetect -filter:a volumedetect -map 0:a -map 0:v -f null /dev/null 2>/dev/null; echo "with skip, with filters"; time ffmpeg -skip_frame nokey -i "$i" -filter:v cropdetect -filter:a volumedetect -map 0:a -map 0:v -f null /dev/null 2>/dev/null; echo "with skip, without filters"; time ffmpeg - skip_frame nokey -i "$i" -map 0:a -map 0:v -f null /dev/null 2>/dev/null; echo "without all"; time ffmpeg -i "$i" -map 0:a -map 0:v -f null /dev/null 2>/dev/null; echo ------------; done ffmpeg version N-57057-g024bf3a Copyright (c) 2000-2013 the FFmpeg developers built on Oct 12 2013 01:53:36 with gcc 4.6.3 (Gentoo 4.6.3 p1.13, pie-0.5.2) configuration: --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='-O2 -pipe - march=native -fomit-frame-pointer' --extra-cflags='-O2 -pipe -march=native - fomit-frame-pointer' --extra-cxxflags='-O2 -pipe -march=native -fomit-frame- pointer' --disable-static --enable-gpl --enable-postproc --enable-avfilter -- enable-avresample --disable-stripping --enable-version3 --enable-nonfree -- disable-indev=v4l2 --disable-outdev=v4l2 --disable-indev=oss --disable- outdev=oss --enable-bzlib --disable-runtime-cpudetect --disable-debug -- disable-doc --disable-gnutls --enable-hardcoded-tables --enable-iconv -- enable-network --disable-openssl --enable-ffplay --enable-vaapi --enable-vdpau --enable-zlib --enable-libvo-aacenc --disable-libvo-amrwbenc --enable- libmp3lame --enable-libaacplus --disable-libfaac --enable-libtheora --disable- libtwolame --enable-libwavpack --enable-libx264 --enable-libxvid --enable- libcdio --disable-libiec61883 --disable-libdc1394 --disable-libcaca --disable- openal --disable-libv4l2 --disable-libpulse --enable-x11grab --disable-libflite --disable-frei0r --disable-fontconfig --enable-ladspa --disable-libass -- enable-libfreetype --disable-libsoxr --enable-pthreads --disable-libopencore- amrwb --disable-libopencore-amrnb --disable-libfdk-aac --disable-libopenjpeg --enable-libbluray --enable-libcelt --disable-libgme --disable-libgsm -- disable-libmodplug --enable-libopus --enable-libquvi --disable-librtmp -- disable-libschroedinger --enable-libspeex --enable-libvorbis --enable-libvpx --disable-libzvbi --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-avx --disable-vis --disable-neon --cpu=host libavutil 52. 46.101 / 52. 46.101 libavcodec 55. 35.100 / 55. 35.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.101 / 3. 88.101 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'big_buck_bunny_1080p_h264.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2008-05-27 18:40:35 timecode : 00:00:00:00 Duration: 00:09:56.46, start: 0.000000, bitrate: 9725 kb/s Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080, 9282 kb/s, 24 fps, 24 tbr, 2400 tbn, 4800 tbc (default) Metadata: creation_time : 2008-05-27 18:40:35 handler_name : Apple Alias Data Handler Stream #0:1(eng): Data: none (tmcd / 0x64636D74) (default) Metadata: creation_time : 2008-05-27 18:40:35 handler_name : Apple Alias Data Handler timecode : 00:00:00:00 Stream #0:2(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 437 kb/s (default) Metadata: creation_time : 2008-05-27 18:40:35 handler_name : Apple Alias Data Handler At least one output file must be specified without skip, with filters real 0m50.257s user 2m38.956s sys 0m1.451s with skip, with filters real 0m34.265s user 1m13.954s sys 0m1.738s with skip, without filters real 0m32.982s user 1m13.441s sys 0m1.812s without all real 0m50.799s user 2m41.453s sys 0m1.593s ------------ ffmpeg version N-57057-g024bf3a Copyright (c) 2000-2013 the FFmpeg developers built on Oct 12 2013 01:53:36 with gcc 4.6.3 (Gentoo 4.6.3 p1.13, pie-0.5.2) configuration: --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='-O2 -pipe - march=native -fomit-frame-pointer' --extra-cflags='-O2 -pipe -march=native - fomit-frame-pointer' --extra-cxxflags='-O2 -pipe -march=native -fomit-frame- pointer' --disable-static --enable-gpl --enable-postproc --enable-avfilter -- enable-avresample --disable-stripping --enable-version3 --enable-nonfree -- disable-indev=v4l2 --disable-outdev=v4l2 --disable-indev=oss --disable- outdev=oss --enable-bzlib --disable-runtime-cpudetect --disable-debug -- disable-doc --disable-gnutls --enable-hardcoded-tables --enable-iconv -- enable-network --disable-openssl --enable-ffplay --enable-vaapi --enable-vdpau --enable-zlib --enable-libvo-aacenc --disable-libvo-amrwbenc --enable- libmp3lame --enable-libaacplus --disable-libfaac --enable-libtheora --disable- libtwolame --enable-libwavpack --enable-libx264 --enable-libxvid --enable- libcdio --disable-libiec61883 --disable-libdc1394 --disable-libcaca --disable- openal --disable-libv4l2 --disable-libpulse --enable-x11grab --disable-libflite --disable-frei0r --disable-fontconfig --enable-ladspa --disable-libass -- enable-libfreetype --disable-libsoxr --enable-pthreads --disable-libopencore- amrwb --disable-libopencore-amrnb --disable-libfdk-aac --disable-libopenjpeg --enable-libbluray --enable-libcelt --disable-libgme --disable-libgsm -- disable-libmodplug --enable-libopus --enable-libquvi --disable-librtmp -- disable-libschroedinger --enable-libspeex --enable-libvorbis --enable-libvpx --disable-libzvbi --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-avx --disable-vis --disable-neon --cpu=host libavutil 52. 46.101 / 52. 46.101 libavcodec 55. 35.100 / 55. 35.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.101 / 3. 88.101 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, avi, from 'big_buck_bunny_1080p_stereo.avi': Metadata: encoder : MEncoder 2:1.0~rc2-0ubuntu13 Duration: 00:09:56.46, start: 0.000000, bitrate: 9586 kb/s Stream #0:0: Video: msmpeg4v2 (MP42 / 0x3234504D), yuv420p, 1920x1080, 24 tbr, 24 tbn, 24 tbc Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 48000 Hz, stereo, s16p, 245 kb/s At least one output file must be specified without skip, with filters real 1m3.237s user 1m2.493s sys 0m0.363s with skip, with filters real 0m3.095s user 0m2.889s sys 0m0.189s with skip, without filters real 0m2.844s user 0m2.644s sys 0m0.174s without all real 1m2.768s user 1m2.070s sys 0m0.315s ------------ ffmpeg version N-57057-g024bf3a Copyright (c) 2000-2013 the FFmpeg developers built on Oct 12 2013 01:53:36 with gcc 4.6.3 (Gentoo 4.6.3 p1.13, pie-0.5.2) configuration: --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='-O2 -pipe - march=native -fomit-frame-pointer' --extra-cflags='-O2 -pipe -march=native - fomit-frame-pointer' --extra-cxxflags='-O2 -pipe -march=native -fomit-frame- pointer' --disable-static --enable-gpl --enable-postproc --enable-avfilter -- enable-avresample --disable-stripping --enable-version3 --enable-nonfree -- disable-indev=v4l2 --disable-outdev=v4l2 --disable-indev=oss --disable- outdev=oss --enable-bzlib --disable-runtime-cpudetect --disable-debug -- disable-doc --disable-gnutls --enable-hardcoded-tables --enable-iconv -- enable-network --disable-openssl --enable-ffplay --enable-vaapi --enable-vdpau --enable-zlib --enable-libvo-aacenc --disable-libvo-amrwbenc --enable- libmp3lame --enable-libaacplus --disable-libfaac --enable-libtheora --disable- libtwolame --enable-libwavpack --enable-libx264 --enable-libxvid --enable- libcdio --disable-libiec61883 --disable-libdc1394 --disable-libcaca --disable- openal --disable-libv4l2 --disable-libpulse --enable-x11grab --disable-libflite --disable-frei0r --disable-fontconfig --enable-ladspa --disable-libass -- enable-libfreetype --disable-libsoxr --enable-pthreads --disable-libopencore- amrwb --disable-libopencore-amrnb --disable-libfdk-aac --disable-libopenjpeg --enable-libbluray --enable-libcelt --disable-libgme --disable-libgsm -- disable-libmodplug --enable-libopus --enable-libquvi --disable-librtmp -- disable-libschroedinger --enable-libspeex --enable-libvorbis --enable-libvpx --disable-libzvbi --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-avx --disable-vis --disable-neon --cpu=host libavutil 52. 46.101 / 52. 46.101 libavcodec 55. 35.100 / 55. 35.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.101 / 3. 88.101 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 [theora @ 0x1b33d00] 7 bits left in packet 82 [ogg @ 0x1b32fe0] Broken file, keyframe not correctly marked. Input #0, ogg, from 'big_buck_bunny_1080p_stereo.ogg': Duration: 00:09:56.46, start: 0.000000, bitrate: 12191 kb/s Stream #0:0: Video: theora, yuv420p, 1920x1080, 24 tbr, 24 tbn, 24 tbc Stream #0:1: Audio: vorbis, 48000 Hz, stereo, fltp, 192 kb/s At least one output file must be specified without skip, with filters real 0m47.223s user 2m31.742s sys 0m1.204s with skip, with filters real 0m2.360s user 0m2.096s sys 0m0.247s with skip, without filters real 0m2.169s user 0m1.875s sys 0m0.273s without all real 0m49.617s user 2m41.024s sys 0m1.237s ------------ ffmpeg version N-57057-g024bf3a Copyright (c) 2000-2013 the FFmpeg developers built on Oct 12 2013 01:53:36 with gcc 4.6.3 (Gentoo 4.6.3 p1.13, pie-0.5.2) configuration: --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='-O2 -pipe - march=native -fomit-frame-pointer' --extra-cflags='-O2 -pipe -march=native - fomit-frame-pointer' --extra-cxxflags='-O2 -pipe -march=native -fomit-frame- pointer' --disable-static --enable-gpl --enable-postproc --enable-avfilter -- enable-avresample --disable-stripping --enable-version3 --enable-nonfree -- disable-indev=v4l2 --disable-outdev=v4l2 --disable-indev=oss --disable- outdev=oss --enable-bzlib --disable-runtime-cpudetect --disable-debug -- disable-doc --disable-gnutls --enable-hardcoded-tables --enable-iconv -- enable-network --disable-openssl --enable-ffplay --enable-vaapi --enable-vdpau --enable-zlib --enable-libvo-aacenc --disable-libvo-amrwbenc --enable- libmp3lame --enable-libaacplus --disable-libfaac --enable-libtheora --disable- libtwolame --enable-libwavpack --enable-libx264 --enable-libxvid --enable- libcdio --disable-libiec61883 --disable-libdc1394 --disable-libcaca --disable- openal --disable-libv4l2 --disable-libpulse --enable-x11grab --disable-libflite --disable-frei0r --disable-fontconfig --enable-ladspa --disable-libass -- enable-libfreetype --disable-libsoxr --enable-pthreads --disable-libopencore- amrwb --disable-libopencore-amrnb --disable-libfdk-aac --disable-libopenjpeg --enable-libbluray --enable-libcelt --disable-libgme --disable-libgsm -- disable-libmodplug --enable-libopus --enable-libquvi --disable-librtmp -- disable-libschroedinger --enable-libspeex --enable-libvorbis --enable-libvpx --disable-libzvbi --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-avx --disable-vis --disable-neon --cpu=host libavutil 52. 46.101 / 52. 46.101 libavcodec 55. 35.100 / 55. 35.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.101 / 3. 88.101 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, avi, from 'big_buck_bunny_1080p_surround.avi': Metadata: encoder : AVI-Mux GUI 1.17.7, Aug 8 2006 20:59:17 JUNK : Duration: 00:09:56.46, start: 0.000000, bitrate: 12455 kb/s Stream #0:0: Video: mpeg4 (Simple Profile) (FMP4 / 0x34504D46), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 24 tbr, 24 tbn, 24 tbc Stream #0:1: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, 5.1(side), fltp, 448 kb/s At least one output file must be specified without skip, with filters real 0m36.299s user 1m52.853s sys 0m1.996s with skip, with filters real 0m4.720s user 0m4.804s sys 0m0.422s with skip, without filters real 0m4.061s user 0m4.227s sys 0m0.360s without all real 0m35.177s user 1m51.405s sys 0m1.869s ------------ ffmpeg version N-57057-g024bf3a Copyright (c) 2000-2013 the FFmpeg developers built on Oct 12 2013 01:53:36 with gcc 4.6.3 (Gentoo 4.6.3 p1.13, pie-0.5.2) configuration: --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='-O2 -pipe - march=native -fomit-frame-pointer' --extra-cflags='-O2 -pipe -march=native - fomit-frame-pointer' --extra-cxxflags='-O2 -pipe -march=native -fomit-frame- pointer' --disable-static --enable-gpl --enable-postproc --enable-avfilter -- enable-avresample --disable-stripping --enable-version3 --enable-nonfree -- disable-indev=v4l2 --disable-outdev=v4l2 --disable-indev=oss --disable- outdev=oss --enable-bzlib --disable-runtime-cpudetect --disable-debug -- disable-doc --disable-gnutls --enable-hardcoded-tables --enable-iconv -- enable-network --disable-openssl --enable-ffplay --enable-vaapi --enable-vdpau --enable-zlib --enable-libvo-aacenc --disable-libvo-amrwbenc --enable- libmp3lame --enable-libaacplus --disable-libfaac --enable-libtheora --disable- libtwolame --enable-libwavpack --enable-libx264 --enable-libxvid --enable- libcdio --disable-libiec61883 --disable-libdc1394 --disable-libcaca --disable- openal --disable-libv4l2 --disable-libpulse --enable-x11grab --disable-libflite --disable-frei0r --disable-fontconfig --enable-ladspa --disable-libass -- enable-libfreetype --disable-libsoxr --enable-pthreads --disable-libopencore- amrwb --disable-libopencore-amrnb --disable-libfdk-aac --disable-libopenjpeg --enable-libbluray --enable-libcelt --disable-libgme --disable-libgsm -- disable-libmodplug --enable-libopus --enable-libquvi --disable-librtmp -- disable-libschroedinger --enable-libspeex --enable-libvorbis --enable-libvpx --disable-libzvbi --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-avx --disable-vis --disable-neon --cpu=host libavutil 52. 46.101 / 52. 46.101 libavcodec 55. 35.100 / 55. 35.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.101 / 3. 88.101 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, matroska,webm, from 'test.mkv': Metadata: JUNK : ENCODER : Lavf55.19.100 Duration: 00:09:56.48, start: 0.000000, bitrate: 12451 kb/s Stream #0:0: Video: mpeg4 (Simple Profile), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 24 fps, 24 tbr, 1k tbn, 24 tbc (default) Stream #0:1: Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s (default) At least one output file must be specified without skip, with filters real 0m36.583s user 1m52.900s sys 0m2.053s with skip, with filters real 0m4.750s user 0m5.065s sys 0m0.449s with skip, without filters real 0m4.225s user 0m4.612s sys 0m0.395s without all real 0m35.786s user 1m52.440s sys 0m1.922s ------------ The last file (test.mkv) I get with an remux of the surround avi (want to test Matroska): ffmpeg -i big_buck_bunny_1080p_surround.avi -c copy -map 0 test.mkv I tried the same with the original testfile: $ for i in test.mp4; do ffmpeg -i "$i"; echo "without skip, with filters"; time ffmpeg -i "$i" -filter:v cropdetect -filter:a volumedetect -map 0:a -map 0:v -f null /dev/null 2>/dev/null; echo "with skip, with filters"; time ffmpeg - skip_frame nokey -i "$i" -filter:v cropdetect -filter:a volumedetect -map 0:a - map 0:v -f null /dev/null 2>/dev/null; echo "with skip, without filters"; time ffmpeg -skip_frame nokey -i "$i" -map 0:a -map 0:v -f null /dev/null 2>/dev/null; echo "without all"; time ffmpeg -i "$i" -map 0:a -map 0:v -f null /dev/null 2>/dev/null; echo ------------; done ffmpeg version N-57057-g024bf3a Copyright (c) 2000-2013 the FFmpeg developers built on Oct 12 2013 01:53:36 with gcc 4.6.3 (Gentoo 4.6.3 p1.13, pie-0.5.2) configuration: --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='-O2 -pipe - march=native -fomit-frame-pointer' --extra-cflags='-O2 -pipe -march=native - fomit-frame-pointer' --extra-cxxflags='-O2 -pipe -march=native -fomit-frame- pointer' --disable-static --enable-gpl --enable-postproc --enable-avfilter -- enable-avresample --disable-stripping --enable-version3 --enable-nonfree -- disable-indev=v4l2 --disable-outdev=v4l2 --disable-indev=oss --disable- outdev=oss --enable-bzlib --disable-runtime-cpudetect --disable-debug -- disable-doc --disable-gnutls --enable-hardcoded-tables --enable-iconv -- enable-network --disable-openssl --enable-ffplay --enable-vaapi --enable-vdpau --enable-zlib --enable-libvo-aacenc --disable-libvo-amrwbenc --enable- libmp3lame --enable-libaacplus --disable-libfaac --enable-libtheora --disable- libtwolame --enable-libwavpack --enable-libx264 --enable-libxvid --enable- libcdio --disable-libiec61883 --disable-libdc1394 --disable-libcaca --disable- openal --disable-libv4l2 --disable-libpulse --enable-x11grab --disable-libflite --disable-frei0r --disable-fontconfig --enable-ladspa --disable-libass -- enable-libfreetype --disable-libsoxr --enable-pthreads --disable-libopencore- amrwb --disable-libopencore-amrnb --disable-libfdk-aac --disable-libopenjpeg --enable-libbluray --enable-libcelt --disable-libgme --disable-libgsm -- disable-libmodplug --enable-libopus --enable-libquvi --disable-librtmp -- disable-libschroedinger --enable-libspeex --enable-libvorbis --enable-libvpx --disable-libzvbi --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-avx --disable-vis --disable-neon --cpu=host libavutil 52. 46.101 / 52. 46.101 libavcodec 55. 35.100 / 55. 35.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.101 / 3. 88.101 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2013-04-23 18:01:09 Duration: 00:03:13.17, start: 0.000000, bitrate: 4532 kb/s Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080, 4338 kb/s, 25 fps, 25 tbr, 50 tbn, 50 tbc (default) Metadata: handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 191 kb/s (default) Metadata: creation_time : 2013-04-23 18:01:18 handler_name : IsoMedia File Produced by Google, 5-11-2011 At least one output file must be specified without skip, with filters real 0m20.888s user 1m10.111s sys 0m0.534s with skip, with filters real 0m18.642s user 0m56.650s sys 0m1.407s with skip, without filters real 0m16.847s user 0m57.600s sys 0m1.631s without all real 0m21.501s user 1m9.793s sys 0m0.506s ------------ test.mp4 is from youtube, the 1080p-Verions of http://www.youtube.com/watch?v=7dPdyqtgHo8. I can upload it anywhere else, if you want it. Gerion From rieger at ari.uni-heidelberg.de Sun Oct 13 00:16:15 2013 From: rieger at ari.uni-heidelberg.de (rieger at ari.uni-heidelberg.de) Date: Sun, 13 Oct 2013 00:16:15 +0200 Subject: [FFmpeg-user] Following UbuntuCompilationGuide running into trouble with the x264 part Message-ID: <20131013001615.t8vhlkbk004480o0@wwwmail.urz.uni-heidelberg.de> Hello, I use Kubuntu 12.04 and I follow strictly this guide: http://trac.ffmpeg.org/wiki/UbuntuCompilationGuide Everything went right up to "make" in the section "x264". There I get following error message: ... yasm -f elf -m amd64 -DHAVE_ALIGNED_STACK=1 -DHIGH_BIT_DEPTH=0 -DBIT_DEPTH=8 -DARCH_X86_64=1 -I./common/x86/ -o common/x86/trellis-64.o common/x86/trellis-64.asm rm -f libx264.a ar rc libx264.a ... gcc -o x264 x264.o ... /usr/local/lib/libavcodec.a(libx264.o): In function `X264_init': /usr/src/ffmpeg/libavcodec/libx264.c:557: undefined reference to `x264_encoder_open_135' collect2: ld gab 1 als Ende-Status zur?ck make: *** [x264] Fehler 1 Before I have downloaded the "stable" Version of x264 (2013-08-23). Then: claus at Laptop:~/ffmpeg_sources/x264$ ./configure --prefix="$HOME/ffmpeg_build" --bindir="$HOME/bin" --enable-static fatal: Not a git repository (or any parent up to mount parent ) Stopping at filesystem boundary (GIT_DISCOVERY_ACROSS_FILESYSTEM not set). platform: X86_64 system: LINUX cli: yes libx264: internal shared: no static: yes asm: yes interlaced: yes avs: avxsynth lavf: yes ffms: no gpac: yes gpl: yes thread: posix opencl: yes filters: resize crop select_every debug: no gprof: no strip: no PIC: no visualize: no bit depth: 8 chroma format: all You can run 'make' or 'make fprofiled' now. The following "make" results in the error described before. Any hint where I could proceed to find the error would be great! Thanks a lot, Claus From raytiley at gmail.com Sun Oct 13 05:35:24 2013 From: raytiley at gmail.com (Ray Tiley) Date: Sat, 12 Oct 2013 23:35:24 -0400 Subject: [FFmpeg-user] Directshow Crossbar Support Message-ID: Hi, I'm trying to record some live video using an Osprey-230. I can't seem to get ffmpeg to work at all. After some googling it seems like it might have to do with the device requiring crossbar support. Is it still the case that ffmpeg lacks this? Here is a command that is not working along with the output. Any help greatly appreciated. -ray D:\ffmpeg-32\bin>ffmpeg -loglevel debug -f dshow -s 640x480 -r 29.97 -pixel_form at bgra -i video="Osprey-230 Video Device 1":audio="Balanced 1 (Osprey-2X0)" out .mp4 ffmpeg version N-57057-g024bf3a Copyright (c) 2000-2013 the FFmpeg developers built on Oct 11 2013 18:01:59 with gcc 4.8.1 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 46.101 / 52. 46.101 libavcodec 55. 35.100 / 55. 35.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.101 / 3. 88.101 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Splitting the commandline. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'. Reading option '-f' ... matched as option 'f' (force format) with argument 'dsho w'. Reading option '-s' ... matched as option 's' (set frame size (WxH or abbreviati on)) with argument '640x480'. Reading option '-r' ... matched as option 'r' (set frame rate (Hz value, fractio n or abbreviation)) with argument '29.97'. Reading option '-pixel_format' ... matched as AVOption 'pixel_format' with argum ent 'bgra'. Reading option '-i' ... matched as input file with argument 'video=Osprey-230 Vi deo Device 1:audio=Balanced 1 (Osprey-2X0)'. Reading option 'out.mp4' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option loglevel (set logging level) with argument debug. Successfully parsed a group of options. Parsing a group of options: input file video=Osprey-230 Video Device 1:audio=Bal anced 1 (Osprey-2X0). Applying option f (force format) with argument dshow. Applying option s (set frame size (WxH or abbreviation)) with argument 640x480. Applying option r (set frame rate (Hz value, fraction or abbreviation)) with arg ument 29.97. Successfully parsed a group of options. Opening an input file: video=Osprey-230 Video Device 1:audio=Balanced 1 (Osprey- 2X0). [dshow @ 025597e0] Could not connect pins video=Osprey-230 Video Device 1:audio=Balanced 1 (Osprey-2X0): Input/output error From anatol2002 at gmail.com Sun Oct 13 05:47:07 2013 From: anatol2002 at gmail.com (Anatol) Date: Sun, 13 Oct 2013 06:47:07 +0300 Subject: [FFmpeg-user] Different conversion results on similar conversion runs. Message-ID: Hi, I have came across a following strange ffmpeg behavior - The same conversion session executed consecutively gives different results. "same conversion session" means: - same machine - same source - same ffmpeg - same command line The difference is of 2-3 kbps. Command line + complete console output: ffmpeg -threads 1 -i sourcefile.mp4 -c:v libx264 -subq 5 -qcomp 0.6 -qmin 10 -qmax 50 -qdiff 4 -coder 1 -refs 2 -vprofile main -force_key_frames 0,2.002,4.004,6.006,8.008,10.01,12.012,14.014,16.016,18.018,20.02,22.022,24.024,26.026,28.028,30.03,32.032,34.034,36.036,38.038,40.04,42.042,44.044,46.046,48.048,50.0501,52.0521,54.0541,56.0561,58.0581,60.0601,62.0621,64.0641,66.0661,68.0681 -pix_fmt yuv420p -b:v 1200k -s 848x480 -r 29.97 -g 60 -c:a libfaac -b:a 64k -ar 44100 -ac 2 -map_chapters -1 -map_metadata -1 -f mp4 -flags +loop+mv4 -cmp 256 -partitions +parti4x4+partp8x8+partb8x8 -trellis 1 -refs 1 -me_range 16 -keyint_min 20 -sc_threshold 40 -i_qfactor 0.71 -bt 400k -maxrate 1200k -bufsize 1200k -rc_eq 'blurCplx^(1-qComp)' -level 30 -async 2 -vsync 1 -y outputfile.mp4 ffmpeg version 1.1.1 Copyright (c) 2000-2013 the FFmpeg developers built on Feb 26 2013 16:35:13 with gcc 4.1.2 (GCC) 20080704 (Red Hat 4.1.2-52) configuration: --extra-cflags=-O2 --enable-bzlib --disable-devices --enable-libfaac --enable-libfdk-aac --enable-libaacplus --enable-libgsm --enable-libmp3lame --enable-libschroedinger --enabl e-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-filter=movie --enable-avfilter --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libvp x --enable-libspeex --enable-postproc --enable-pthreads --enable-libass --enable-libfreetype --enable-fontconfig --disable-static --enable-shared --enable-gpl --disable-debug --disable-optimizat ions --disable-stripping --extra-cflags=-fPIC --extra-ldflags=-fPIC --enable-nonfree --enable-version3 --libdir=/usr/local/lib libavutil 52. 13.100 / 52. 13.100 libavcodec 54. 86.100 / 54. 86.100 libavformat 54. 59.106 / 54. 59.106 libavdevice 54. 3.102 / 54. 3.102 libavfilter 3. 32.100 / 3. 32.100 libswscale 2. 1.103 / 2. 1.103 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'sourcefile.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf54.59.106 Duration: 00:01:10.78, start: 0.023220, bitrate: 1129 kb/s Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1024x576, 1046 kb/s, 29.97 fps, 29.97 tbr, 11988 tbn, 59.94 tbc Metadata: handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 74 kb/s Metadata: handler_name : SoundHandler -async is forwarded to lavfi similarly to -af aresample=async=2:min_hard_comp=0.100000. [libx264 @ 0x13f52e0] MB rate (47652) > level limit (40500) [libx264 @ 0x13f52e0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX [libx264 @ 0x13f52e0] profile Main, level 3.0 [libx264 @ 0x13f52e0] 264 - core 129 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - - options: cabac=1 ref=1 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=5 psy =1 psy_rd=1.00:0.00 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=48 lookahead_threads=7 sliced_threads=0 nr=0 decimate=1 in terlaced=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=60 keyint_min=20 scenecut=40 intra_refresh=0 rc_lookahead=4 0 rc=cbr mbtree=1 bitrate=1200 ratetol=1.0 qcomp=0.60 qpmin=10 qpmax=50 qpstep=4 vbv_maxrate=1200 vbv_bufsize=1200 nal_hrd=none ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'outputfile.mp4': Metadata: encoder : Lavf54.59.106 Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 848x480, q=10-50, 1200 kb/s, 11988 tbn, 29.97 tbc Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0:0 -> #0:0 (h264 -> libx264) Stream #0:1 -> #0:1 (aac -> libfaac) Press [q] to stop, [?] for help frame= 537 fps=133 q=19.0 size= 1768kB time=00:00:17.80 bitrate= 813.5kbits/s dup=1 drop=0 frame= 1169 fps=145 q=16.0 size= 3940kB time=00:00:38.89 bitrate= 843.6kbits/s dup=1 drop=0 frame= 1685 fps=139 q=19.0 size= 5929kB time=00:00:53.80 bitrate= 867.6kbits/s dup=1 drop=0 frame= 2121 fps=140 q=-1.0 Lsize= 8071kB time=00:01:10.77 bitrate= 934.2kbits/s dup=1 drop=0 video:7533kB audio:464kB subtitle:0 global headers:0kB muxing overhead 0.927112% [libx264 @ 0x13f52e0] frame I:43 Avg QP:17.97 size: 58212 [libx264 @ 0x13f52e0] frame P:774 Avg QP:20.93 size: 5918 [libx264 @ 0x13f52e0] frame B:1304 Avg QP:19.52 size: 483 [libx264 @ 0x13f52e0] consecutive B-frames: 16.9% 1.9% 4.2% 76.9% [libx264 @ 0x13f52e0] mb I I16..4: 32.8% 0.0% 67.2% [libx264 @ 0x13f52e0] mb P I16..4: 13.5% 0.0% 4.9% P16..4: 28.7% 10.1% 5.1% 0.0% 0.0% skip:37.6% [libx264 @ 0x13f52e0] mb B I16..4: 0.2% 0.0% 0.0% B16..8: 5.6% 1.6% 0.1% direct: 4.3% skip:88.1% L0:34.4% L1:58.4% BI: 7.2% [libx264 @ 0x13f52e0] coded y,uvDC,uvAC intra: 34.7% 63.6% 31.7% inter: 5.6% 15.0% 2.2% [libx264 @ 0x13f52e0] i16 v,h,dc,p: 31% 34% 19% 16% [libx264 @ 0x13f52e0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 25% 21% 3% 8% 10% 6% 4% 4% [libx264 @ 0x13f52e0] i8c dc,h,v,p: 47% 27% 18% 8% [libx264 @ 0x13f52e0] Weighted P-Frames: Y:3.7% UV:3.2% [libx264 @ 0x13f52e0] kb/s:871.88 thanks, Anatol From 690271929 at qq.com Sat Oct 12 06:53:47 2013 From: 690271929 at qq.com (=?gb18030?B?SmFja3k=?=) Date: Sat, 12 Oct 2013 12:53:47 +0800 Subject: [FFmpeg-user] Extract Audio from Video file using ffmpeg Message-ID: hi? I have a m2ts file?the info : ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers built on Jul 31 2013 10:34:48 with gcc 4.4.6 (GCC) 20110731 (Red Hat 4.4.6-3) configuration: --prefix=/data/test/jh/ffmpeg --extra-cflags=-I/data/test/jh/src/ffmpeg/include --extra-ldflags=-L/data/test/jh/src/ffmpeg/lib --enable-gpl --enable-version3 --enable-nonfree --enable-debug --enable-memalign-hack --enable-libfaac --enable-libaacplus --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid libavutil 52. 38.100 / 52. 38.100 libavcodec 55. 18.102 / 55. 18.102 libavformat 55. 12.100 / 55. 12.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 79.101 / 3. 79.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 H264 encode type: Frame @@type: 0 [mpegts @ 0x317e860] Stream #6: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x317e860] Stream #7: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x317e860] Stream #8: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x317e860] Stream #9: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x317e860] Stream #10: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x317e860] Stream #11: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x317e860] Stream #12: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x317e860] Stream #13: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x317e860] Stream #14: not enough frames to estimate rate; consider increasing probesize [NULL @ 0x3185fe0] start time is not set in estimate_timings_from_pts [NULL @ 0x31869e0] start time is not set in estimate_timings_from_pts [NULL @ 0x31a1500] start time is not set in estimate_timings_from_pts [NULL @ 0x31a1ec0] start time is not set in estimate_timings_from_pts [NULL @ 0x31a27c0] start time is not set in estimate_timings_from_pts [NULL @ 0x31a31e0] start time is not set in estimate_timings_from_pts [NULL @ 0x31a3ae0] start time is not set in estimate_timings_from_pts [NULL @ 0x31a44c0] start time is not set in estimate_timings_from_pts [NULL @ 0x31a4da0] start time is not set in estimate_timings_from_pts [mpegts @ 0x317e860] Could not find codec parameters for stream 6 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x317e860] Could not find codec parameters for stream 7 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x317e860] Could not find codec parameters for stream 8 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x317e860] Could not find codec parameters for stream 9 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x317e860] Could not find codec parameters for stream 10 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x317e860] Could not find codec parameters for stream 11 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x317e860] Could not find codec parameters for stream 12 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x317e860] Could not find codec parameters for stream 13 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x317e860] Could not find codec parameters for stream 14 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, mpegts, from '/data/test/media/2270059.m2ts': Duration: 02:13:18.03, start: 11.608967, bitrate: 30389 kb/s Program 1 Stream #0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc Stream #0:1[0x1100]: Audio: truehd (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), s32 Stream #0:2[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s Stream #0:3[0x1101]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s Stream #0:4[0x1102]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s Stream #0:5[0x1103]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 192 kb/s Stream #0:6[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:7[0x1201]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:8[0x1202]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:9[0x1203]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:10[0x1204]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:11[0x1205]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:12[0x1206]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:13[0x1207]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:14[0x1208]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) [mpegts @ 0x31a64c0] frame size not set Output #0, mpegts, to 'audio.ts': Metadata: encoder : Lavf55.12.100 Stream #0:0: Audio: truehd (AC-3 / 0x332D4341), 48000 Hz, 5.1(side) Stream mapping: Stream #0:1 -> #0:0 (copy) Press [q] to stop, [?] for help size= 29036kB time=00:01:28.63 bitrate=2683.6kbits/s size= 65622kB time=00:02:53.32 bitrate=3101.5kbits/s size= 102779kB time=00:04:17.11 bitrate=3274.6kbits/s size= 130506kB time=00:05:57.78 bitrate=2988.1kbits/s size= 143709kB time=00:06:51.52 bitrate=2860.8kbits/s size= 158099kB time=00:07:41.40 bitrate=2806.9kbits/s size= 183188kB time=00:08:45.08 bitrate=2858.0kbits/s size= 201836kB time=00:09:47.06 bitrate=2816.5kbits/s size= 217184kB time=00:10:32.65 bitrate=2812.3kbits/s size= 231522kB time=00:11:10.94 bitrate=2826.8kbits/s size= 245563kB time=00:11:49.28 bitrate=2836.2kbits/s size= 262711kB time=00:12:39.26 bitrate=2834.5kbits/s size= 272760kB time=00:13:15.27 bitrate=2809.7kbits/s size= 286700kB time=00:14:08.80 bitrate=2767.0kbits/s size= 304315kB time=00:15:21.15 bitrate=2706.3kbits/s size= 325577kB time=00:16:25.98 bitrate=2705.0kbits/s size= 340834kB time=00:17:31.23 bitrate=2656.0kbits/s size= 369857kB time=00:18:51.52 bitrate=2677.7kbits/s size= 400261kB time=00:20:03.23 bitrate=2725.1kbits/s size= 411097kB time=00:20:27.93 bitrate=2742.6kbits/s size= 420947kB time=00:21:04.28 bitrate=2727.5kbits/s size= 440434kB time=00:21:54.75 bitrate=2744.3kbits/s size= 461979kB time=00:22:45.89 bitrate=2770.7kbits/s size= 485006kB time=00:23:47.64 bitrate=2783.0kbits/s size= 502782kB time=00:24:56.64 bitrate=2752.0kbits/s size= 517697kB time=00:26:05.50 bitrate=2709.0kbits/s size= 538650kB time=00:27:29.48 bitrate=2675.2kbits/s size= 562143kB time=00:28:42.18 bitrate=2674.0kbits/s size= 589098kB time=00:29:47.87 bitrate=2699.2kbits/s size= 601802kB time=00:30:18.20 bitrate=2711.4kbits/s size= 612701kB time=00:31:02.81 bitrate=2694.4kbits/s size= 626849kB time=00:32:07.20 bitrate=2664.6kbits/s size= 645453kB time=00:33:16.76 bitrate=2648.1kbits/s size= 665176kB time=00:34:21.45 bitrate=2643.3kbits/s size= 685750kB time=00:35:21.63 bitrate=2647.8kbits/s size= 709688kB time=00:36:29.79 bitrate=2654.9kbits/s size= 728084kB time=00:37:33.30 bitrate=2647.0kbits/s size= 749517kB time=00:38:28.62 bitrate=2659.6kbits/s size= 773498kB time=00:39:27.77 bitrate=2676.1kbits/s size= 799351kB time=00:40:31.41 bitrate=2693.2kbits/s size= 823039kB time=00:41:35.29 bitrate=2702.0kbits/s size= 837203kB time=00:42:25.81 bitrate=2694.0kbits/s size= 859126kB time=00:43:21.31 bitrate=2705.5kbits/s size= 888626kB time=00:44:36.64 bitrate=2719.7kbits/s size= 908233kB time=00:45:45.20 bitrate=2710.3kbits/s size= 919846kB time=00:46:27.78 bitrate=2703.0kbits/s size= 933270kB time=00:47:13.21 bitrate=2698.5kbits/s size= 945426kB time=00:47:51.13 bitrate=2697.5kbits/s size= 955306kB time=00:48:21.24 bitrate=2697.4kbits/s size= 968168kB time=00:49:06.33 bitrate=2691.9kbits/s size= 987251kB time=00:50:04.99 bitrate=2691.4kbits/s size= 1006855kB time=00:51:09.53 bitrate=2687.1kbits/s size= 1032854kB time=00:52:33.02 bitrate=2683.5kbits/s size= 1055804kB time=00:53:43.61 bitrate=2683.1kbits/s size= 1083188kB time=00:54:44.18 bitrate=2701.9kbits/s size= 1107507kB time=00:55:36.38 bitrate=2719.3kbits/s size= 1132957kB time=00:56:43.86 bitrate=2726.7kbits/s size= 1145117kB time=00:57:20.44 bitrate=2726.6kbits/s size= 1158918kB time=00:58:13.72 bitrate=2717.4kbits/s size= 1184437kB time=00:59:25.69 bitrate=2721.2kbits/s size= 1212273kB time=01:00:28.85 bitrate=2736.7kbits/s size= 1232305kB time=01:01:25.08 bitrate=2739.4kbits/s size= 1246631kB time=01:02:20.80 bitrate=2730.0kbits/s size= 1260744kB time=01:03:07.64 bitrate=2726.8kbits/s size= 1265837kB time=01:03:22.27 bitrate=2727.2kbits/s size= 1266410kB time=01:03:24.03 bitrate=2727.2kbits/s size= 1268324kB time=01:03:29.98 bitrate=2727.1kbits/s size= 1278935kB time=01:04:04.57 bitrate=2725.1kbits/s size= 1291253kB time=01:04:44.80 bitrate=2722.9kbits/s size= 1304270kB time=01:05:20.79 bitrate=2725.1kbits/s size= 1327996kB time=01:06:13.63 bitrate=2737.8kbits/s size= 1355447kB time=01:07:10.62 bitrate=2754.9kbits/s size= 1377053kB time=01:08:03.68 bitrate=2762.4kbits/s size= 1397095kB time=01:09:11.00 bitrate=2757.2kbits/s size= 1417907kB time=01:10:13.66 bitrate=2756.6kbits/s size= 1442252kB time=01:11:16.15 bitrate=2763.0kbits/s size= 1468647kB time=01:12:24.55 bitrate=2769.2kbits/s size= 1489018kB time=01:13:25.38 bitrate=2768.9kbits/s size= 1490379kB time=01:13:29.50 bitrate=2768.8kbits/s size= 1508269kB time=01:14:22.56 bitrate=2768.8kbits/s size= 1535204kB time=01:15:36.77 bitrate=2772.1kbits/s size= 1559398kB time=01:16:46.78 bitrate=2773.0kbits/s size= 1575418kB time=01:17:49.60 bitrate=2763.8kbits/s size= 1596503kB time=01:19:01.76 bitrate=2758.2kbits/s size= 1614959kB time=01:20:06.72 bitrate=2752.3kbits/s size= 1632625kB time=01:20:58.46 bitrate=2752.8kbits/s size= 1659674kB time=01:22:01.72 bitrate=2762.5kbits/s size= 1683620kB time=01:23:00.12 bitrate=2769.4kbits/s size= 1706544kB time=01:23:58.17 bitrate=2774.8kbits/s size= 1722191kB time=01:25:00.28 bitrate=2766.2kbits/s size= 1739554kB time=01:26:21.18 bitrate=2750.4kbits/s size= 1761930kB time=01:27:22.52 bitrate=2753.2kbits/s size= 1787518kB time=01:28:21.82 bitrate=2761.9kbits/s size= 1812308kB time=01:29:13.55 bitrate=2773.2kbits/s size= 1835432kB time=01:30:26.02 bitrate=2771.1kbits/s size= 1848247kB time=01:31:18.41 bitrate=2763.7kbits/s size= 1870552kB time=01:32:26.30 bitrate=2762.8kbits/s size= 1894665kB time=01:33:31.34 bitrate=2766.0kbits/s size= 1914840kB time=01:34:43.38 bitrate=2760.0kbits/s size= 1937360kB time=01:35:53.24 bitrate=2758.6kbits/s size= 1962925kB time=01:37:01.92 bitrate=2762.0kbits/s size= 1989550kB time=01:38:09.68 bitrate=2767.3kbits/s size= 2005340kB time=01:39:03.55 bitrate=2764.0kbits/s size= 2027040kB time=01:39:56.25 bitrate=2769.3kbits/s size= 2048703kB time=01:40:49.18 bitrate=2774.4kbits/s size= 2070285kB time=01:41:36.22 bitrate=2782.0kbits/s size= 2096063kB time=01:42:32.67 bitrate=2790.8kbits/s size= 2115441kB time=01:43:24.16 bitrate=2793.2kbits/s size= 2130383kB time=01:44:13.89 bitrate=2790.6kbits/s size= 2143276kB time=01:45:11.10 bitrate=2782.0kbits/s size= 2159125kB time=01:46:13.47 bitrate=2775.2kbits/s size= 2180030kB time=01:47:23.71 bitrate=2771.5kbits/s size= 2204543kB time=01:48:27.36 bitrate=2775.3kbits/s size= 2232982kB time=01:49:36.51 bitrate=2781.5kbits/s size= 2249450kB time=01:50:24.80 bitrate=2781.6kbits/s size= 2267357kB time=01:51:14.72 bitrate=2782.8kbits/s size= 2286033kB time=01:52:18.97 bitrate=2778.9kbits/s size= 2302163kB time=01:53:06.04 bitrate=2779.1kbits/s size= 2321230kB time=01:54:05.28 bitrate=2777.9kbits/s size= 2337307kB time=01:54:58.38 bitrate=2775.6kbits/s size= 2350190kB time=01:55:51.13 bitrate=2769.7kbits/s size= 2371482kB time=01:57:03.20 bitrate=2766.1kbits/s size= 2396044kB time=01:58:05.66 bitrate=2770.2kbits/s size= 2422638kB time=01:59:04.81 bitrate=2777.7kbits/s size= 2436868kB time=01:59:49.69 bitrate=2776.6kbits/s size= 2448027kB time=02:00:19.29 bitrate=2777.9kbits/s size= 2449953kB time=02:00:23.77 bitrate=2778.3kbits/s size= 2452549kB time=02:00:29.79 bitrate=2779.0kbits/s size= 2463361kB time=02:00:55.07 bitrate=2781.5kbits/s size= 2475274kB time=02:01:21.28 bitrate=2784.9kbits/s size= 2484788kB time=02:01:42.75 bitrate=2787.4kbits/s size= 2495388kB time=02:02:13.53 bitrate=2787.5kbits/s size= 2502169kB time=02:02:41.37 bitrate=2784.5kbits/s size= 2511370kB time=02:03:13.12 bitrate=2782.7kbits/s size= 2515862kB time=02:03:25.72 bitrate=2783.0kbits/s size= 2525416kB time=02:03:50.75 bitrate=2784.1kbits/s size= 2528206kB time=02:03:58.56 bitrate=2784.3kbits/s video:0kB audio:2259932kB subtitle:0 global headers:0kB muxing overhead 11.870920% frame=1366679008 but when i do this: ./ffmpeg-0 -y -i audio.ts ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers built on Jul 31 2013 10:34:48 with gcc 4.4.6 (GCC) 20110731 (Red Hat 4.4.6-3) configuration: --prefix=/data/test/jh/ffmpeg --extra-cflags=-I/data/test/jh/src/ffmpeg/include --extra-ldflags=-L/data/test/jh/src/ffmpeg/lib --enable-gpl --enable-version3 --enable-nonfree --enable-debug --enable-memalign-hack --enable-libfaac --enable-libaacplus --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid libavutil 52. 38.100 / 52. 38.100 libavcodec 55. 18.102 / 55. 18.102 libavformat 55. 12.100 / 55. 12.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 79.101 / 3. 79.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 [mpegts @ 0x22d6020] probed stream 0 failed [mpegts @ 0x22d6020] Could not find codec parameters for stream 0 (Unknown: none ([6][0][0][0] / 0x0006)): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options audio.ts: could not find codec parameters I have serveral this file?what can i do? thank you From andrey.aleksandrovich at googlemail.com Sun Oct 13 08:55:50 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Sun, 13 Oct 2013 09:55:50 +0300 Subject: [FFmpeg-user] Extract Audio from Video file using ffmpeg In-Reply-To: References: Message-ID: Try to change output file extension from .ts to .ac3 On 10/12/13, Jacky <690271929 at qq.com> wrote: > hi? > I have a m2ts file?the info : > ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers > built on Jul 31 2013 10:34:48 with gcc 4.4.6 (GCC) 20110731 (Red Hat > 4.4.6-3) > configuration: --prefix=/data/test/jh/ffmpeg > --extra-cflags=-I/data/test/jh/src/ffmpeg/include > --extra-ldflags=-L/data/test/jh/src/ffmpeg/lib --enable-gpl > --enable-version3 --enable-nonfree --enable-debug --enable-memalign-hack > --enable-libfaac --enable-libaacplus --enable-libfreetype > --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 > --enable-libxvid > libavutil 52. 38.100 / 52. 38.100 > libavcodec 55. 18.102 / 55. 18.102 > libavformat 55. 12.100 / 55. 12.100 > libavdevice 55. 3.100 / 55. 3.100 > libavfilter 3. 79.101 / 3. 79.101 > libswscale 2. 3.100 / 2. 3.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 3.100 / 52. 3.100 > H264 encode type: Frame > @@type: 0 > [mpegts @ 0x317e860] Stream #6: not enough frames to estimate rate; consider > increasing probesize > [mpegts @ 0x317e860] Stream #7: not enough frames to estimate rate; consider > increasing probesize > [mpegts @ 0x317e860] Stream #8: not enough frames to estimate rate; consider > increasing probesize > [mpegts @ 0x317e860] Stream #9: not enough frames to estimate rate; consider > increasing probesize > [mpegts @ 0x317e860] Stream #10: not enough frames to estimate rate; > consider increasing probesize > [mpegts @ 0x317e860] Stream #11: not enough frames to estimate rate; > consider increasing probesize > [mpegts @ 0x317e860] Stream #12: not enough frames to estimate rate; > consider increasing probesize > [mpegts @ 0x317e860] Stream #13: not enough frames to estimate rate; > consider increasing probesize > [mpegts @ 0x317e860] Stream #14: not enough frames to estimate rate; > consider increasing probesize > [NULL @ 0x3185fe0] start time is not set in estimate_timings_from_pts > [NULL @ 0x31869e0] start time is not set in estimate_timings_from_pts > [NULL @ 0x31a1500] start time is not set in estimate_timings_from_pts > [NULL @ 0x31a1ec0] start time is not set in estimate_timings_from_pts > [NULL @ 0x31a27c0] start time is not set in estimate_timings_from_pts > [NULL @ 0x31a31e0] start time is not set in estimate_timings_from_pts > [NULL @ 0x31a3ae0] start time is not set in estimate_timings_from_pts > [NULL @ 0x31a44c0] start time is not set in estimate_timings_from_pts > [NULL @ 0x31a4da0] start time is not set in estimate_timings_from_pts > [mpegts @ 0x317e860] Could not find codec parameters for stream 6 (Subtitle: > hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' > options > [mpegts @ 0x317e860] Could not find codec parameters for stream 7 (Subtitle: > hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' > options > [mpegts @ 0x317e860] Could not find codec parameters for stream 8 (Subtitle: > hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' > options > [mpegts @ 0x317e860] Could not find codec parameters for stream 9 (Subtitle: > hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' > options > [mpegts @ 0x317e860] Could not find codec parameters for stream 10 > (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' > options > [mpegts @ 0x317e860] Could not find codec parameters for stream 11 > (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' > options > [mpegts @ 0x317e860] Could not find codec parameters for stream 12 > (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' > options > [mpegts @ 0x317e860] Could not find codec parameters for stream 13 > (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' > options > [mpegts @ 0x317e860] Could not find codec parameters for stream 14 > (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' > options > Input #0, mpegts, from '/data/test/media/2270059.m2ts': > Duration: 02:13:18.03, start: 11.608967, bitrate: 30389 kb/s > Program 1 > Stream #0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p, > 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc > Stream #0:1[0x1100]: Audio: truehd (AC-3 / 0x332D4341), 48000 Hz, > 5.1(side), s32 > Stream #0:2[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, > 5.1(side), fltp, 448 kb/s > Stream #0:3[0x1101]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, > 5.1(side), fltp, 448 kb/s > Stream #0:4[0x1102]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, > 5.1(side), fltp, 448 kb/s > Stream #0:5[0x1103]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, > fltp, 192 kb/s > Stream #0:6[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / > 0x0090) > Stream #0:7[0x1201]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / > 0x0090) > Stream #0:8[0x1202]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / > 0x0090) > Stream #0:9[0x1203]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / > 0x0090) > Stream #0:10[0x1204]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / > 0x0090) > Stream #0:11[0x1205]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / > 0x0090) > Stream #0:12[0x1206]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / > 0x0090) > Stream #0:13[0x1207]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / > 0x0090) > Stream #0:14[0x1208]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / > 0x0090) > [mpegts @ 0x31a64c0] frame size not set > Output #0, mpegts, to 'audio.ts': > Metadata: > encoder : Lavf55.12.100 > Stream #0:0: Audio: truehd (AC-3 / 0x332D4341), 48000 Hz, 5.1(side) > Stream mapping: > Stream #0:1 -> #0:0 (copy) > Press [q] to stop, [?] for help > size= 29036kB time=00:01:28.63 bitrate=2683.6kbits/s > size= 65622kB time=00:02:53.32 bitrate=3101.5kbits/s > size= 102779kB time=00:04:17.11 bitrate=3274.6kbits/s > size= 130506kB time=00:05:57.78 bitrate=2988.1kbits/s > size= 143709kB time=00:06:51.52 bitrate=2860.8kbits/s > size= 158099kB time=00:07:41.40 bitrate=2806.9kbits/s > size= 183188kB time=00:08:45.08 bitrate=2858.0kbits/s > size= 201836kB time=00:09:47.06 bitrate=2816.5kbits/s > size= 217184kB time=00:10:32.65 bitrate=2812.3kbits/s > size= 231522kB time=00:11:10.94 bitrate=2826.8kbits/s > size= 245563kB time=00:11:49.28 bitrate=2836.2kbits/s > size= 262711kB time=00:12:39.26 bitrate=2834.5kbits/s > size= 272760kB time=00:13:15.27 bitrate=2809.7kbits/s > size= 286700kB time=00:14:08.80 bitrate=2767.0kbits/s > size= 304315kB time=00:15:21.15 bitrate=2706.3kbits/s > size= 325577kB time=00:16:25.98 bitrate=2705.0kbits/s > size= 340834kB time=00:17:31.23 bitrate=2656.0kbits/s > size= 369857kB time=00:18:51.52 bitrate=2677.7kbits/s > size= 400261kB time=00:20:03.23 bitrate=2725.1kbits/s > size= 411097kB time=00:20:27.93 bitrate=2742.6kbits/s > size= 420947kB time=00:21:04.28 bitrate=2727.5kbits/s > size= 440434kB time=00:21:54.75 bitrate=2744.3kbits/s > size= 461979kB time=00:22:45.89 bitrate=2770.7kbits/s > size= 485006kB time=00:23:47.64 bitrate=2783.0kbits/s > size= 502782kB time=00:24:56.64 bitrate=2752.0kbits/s > size= 517697kB time=00:26:05.50 bitrate=2709.0kbits/s > size= 538650kB time=00:27:29.48 bitrate=2675.2kbits/s > size= 562143kB time=00:28:42.18 bitrate=2674.0kbits/s > size= 589098kB time=00:29:47.87 bitrate=2699.2kbits/s > size= 601802kB time=00:30:18.20 bitrate=2711.4kbits/s > size= 612701kB time=00:31:02.81 bitrate=2694.4kbits/s > size= 626849kB time=00:32:07.20 bitrate=2664.6kbits/s > size= 645453kB time=00:33:16.76 bitrate=2648.1kbits/s > size= 665176kB time=00:34:21.45 bitrate=2643.3kbits/s > size= 685750kB time=00:35:21.63 bitrate=2647.8kbits/s > size= 709688kB time=00:36:29.79 bitrate=2654.9kbits/s > size= 728084kB time=00:37:33.30 bitrate=2647.0kbits/s > size= 749517kB time=00:38:28.62 bitrate=2659.6kbits/s > size= 773498kB time=00:39:27.77 bitrate=2676.1kbits/s > size= 799351kB time=00:40:31.41 bitrate=2693.2kbits/s > size= 823039kB time=00:41:35.29 bitrate=2702.0kbits/s > size= 837203kB time=00:42:25.81 bitrate=2694.0kbits/s > size= 859126kB time=00:43:21.31 bitrate=2705.5kbits/s > size= 888626kB time=00:44:36.64 bitrate=2719.7kbits/s > size= 908233kB time=00:45:45.20 bitrate=2710.3kbits/s > size= 919846kB time=00:46:27.78 bitrate=2703.0kbits/s > size= 933270kB time=00:47:13.21 bitrate=2698.5kbits/s > size= 945426kB time=00:47:51.13 bitrate=2697.5kbits/s > size= 955306kB time=00:48:21.24 bitrate=2697.4kbits/s > size= 968168kB time=00:49:06.33 bitrate=2691.9kbits/s > size= 987251kB time=00:50:04.99 bitrate=2691.4kbits/s > size= 1006855kB time=00:51:09.53 bitrate=2687.1kbits/s > size= 1032854kB time=00:52:33.02 bitrate=2683.5kbits/s > size= 1055804kB time=00:53:43.61 bitrate=2683.1kbits/s > size= 1083188kB time=00:54:44.18 bitrate=2701.9kbits/s > size= 1107507kB time=00:55:36.38 bitrate=2719.3kbits/s > size= 1132957kB time=00:56:43.86 bitrate=2726.7kbits/s > size= 1145117kB time=00:57:20.44 bitrate=2726.6kbits/s > size= 1158918kB time=00:58:13.72 bitrate=2717.4kbits/s > size= 1184437kB time=00:59:25.69 bitrate=2721.2kbits/s > size= 1212273kB time=01:00:28.85 bitrate=2736.7kbits/s > size= 1232305kB time=01:01:25.08 bitrate=2739.4kbits/s > size= 1246631kB time=01:02:20.80 bitrate=2730.0kbits/s > size= 1260744kB time=01:03:07.64 bitrate=2726.8kbits/s > size= 1265837kB time=01:03:22.27 bitrate=2727.2kbits/s > size= 1266410kB time=01:03:24.03 bitrate=2727.2kbits/s > size= 1268324kB time=01:03:29.98 bitrate=2727.1kbits/s > size= 1278935kB time=01:04:04.57 bitrate=2725.1kbits/s > size= 1291253kB time=01:04:44.80 bitrate=2722.9kbits/s > size= 1304270kB time=01:05:20.79 bitrate=2725.1kbits/s > size= 1327996kB time=01:06:13.63 bitrate=2737.8kbits/s > size= 1355447kB time=01:07:10.62 bitrate=2754.9kbits/s > size= 1377053kB time=01:08:03.68 bitrate=2762.4kbits/s > size= 1397095kB time=01:09:11.00 bitrate=2757.2kbits/s > size= 1417907kB time=01:10:13.66 bitrate=2756.6kbits/s > size= 1442252kB time=01:11:16.15 bitrate=2763.0kbits/s > size= 1468647kB time=01:12:24.55 bitrate=2769.2kbits/s > size= 1489018kB time=01:13:25.38 bitrate=2768.9kbits/s > size= 1490379kB time=01:13:29.50 bitrate=2768.8kbits/s > size= 1508269kB time=01:14:22.56 bitrate=2768.8kbits/s > size= 1535204kB time=01:15:36.77 bitrate=2772.1kbits/s > size= 1559398kB time=01:16:46.78 bitrate=2773.0kbits/s > size= 1575418kB time=01:17:49.60 bitrate=2763.8kbits/s > size= 1596503kB time=01:19:01.76 bitrate=2758.2kbits/s > size= 1614959kB time=01:20:06.72 bitrate=2752.3kbits/s > size= 1632625kB time=01:20:58.46 bitrate=2752.8kbits/s > size= 1659674kB time=01:22:01.72 bitrate=2762.5kbits/s > size= 1683620kB time=01:23:00.12 bitrate=2769.4kbits/s > size= 1706544kB time=01:23:58.17 bitrate=2774.8kbits/s > size= 1722191kB time=01:25:00.28 bitrate=2766.2kbits/s > size= 1739554kB time=01:26:21.18 bitrate=2750.4kbits/s > size= 1761930kB time=01:27:22.52 bitrate=2753.2kbits/s > size= 1787518kB time=01:28:21.82 bitrate=2761.9kbits/s > size= 1812308kB time=01:29:13.55 bitrate=2773.2kbits/s > size= 1835432kB time=01:30:26.02 bitrate=2771.1kbits/s > size= 1848247kB time=01:31:18.41 bitrate=2763.7kbits/s > size= 1870552kB time=01:32:26.30 bitrate=2762.8kbits/s > size= 1894665kB time=01:33:31.34 bitrate=2766.0kbits/s > size= 1914840kB time=01:34:43.38 bitrate=2760.0kbits/s > size= 1937360kB time=01:35:53.24 bitrate=2758.6kbits/s > size= 1962925kB time=01:37:01.92 bitrate=2762.0kbits/s > size= 1989550kB time=01:38:09.68 bitrate=2767.3kbits/s > size= 2005340kB time=01:39:03.55 bitrate=2764.0kbits/s > size= 2027040kB time=01:39:56.25 bitrate=2769.3kbits/s > size= 2048703kB time=01:40:49.18 bitrate=2774.4kbits/s > size= 2070285kB time=01:41:36.22 bitrate=2782.0kbits/s > size= 2096063kB time=01:42:32.67 bitrate=2790.8kbits/s > size= 2115441kB time=01:43:24.16 bitrate=2793.2kbits/s > size= 2130383kB time=01:44:13.89 bitrate=2790.6kbits/s > size= 2143276kB time=01:45:11.10 bitrate=2782.0kbits/s > size= 2159125kB time=01:46:13.47 bitrate=2775.2kbits/s > size= 2180030kB time=01:47:23.71 bitrate=2771.5kbits/s > size= 2204543kB time=01:48:27.36 bitrate=2775.3kbits/s > size= 2232982kB time=01:49:36.51 bitrate=2781.5kbits/s > size= 2249450kB time=01:50:24.80 bitrate=2781.6kbits/s > size= 2267357kB time=01:51:14.72 bitrate=2782.8kbits/s > size= 2286033kB time=01:52:18.97 bitrate=2778.9kbits/s > size= 2302163kB time=01:53:06.04 bitrate=2779.1kbits/s > size= 2321230kB time=01:54:05.28 bitrate=2777.9kbits/s > size= 2337307kB time=01:54:58.38 bitrate=2775.6kbits/s > size= 2350190kB time=01:55:51.13 bitrate=2769.7kbits/s > size= 2371482kB time=01:57:03.20 bitrate=2766.1kbits/s > size= 2396044kB time=01:58:05.66 bitrate=2770.2kbits/s > size= 2422638kB time=01:59:04.81 bitrate=2777.7kbits/s > size= 2436868kB time=01:59:49.69 bitrate=2776.6kbits/s > size= 2448027kB time=02:00:19.29 bitrate=2777.9kbits/s > size= 2449953kB time=02:00:23.77 bitrate=2778.3kbits/s > size= 2452549kB time=02:00:29.79 bitrate=2779.0kbits/s > size= 2463361kB time=02:00:55.07 bitrate=2781.5kbits/s > size= 2475274kB time=02:01:21.28 bitrate=2784.9kbits/s > size= 2484788kB time=02:01:42.75 bitrate=2787.4kbits/s > size= 2495388kB time=02:02:13.53 bitrate=2787.5kbits/s > size= 2502169kB time=02:02:41.37 bitrate=2784.5kbits/s > size= 2511370kB time=02:03:13.12 bitrate=2782.7kbits/s > size= 2515862kB time=02:03:25.72 bitrate=2783.0kbits/s > size= 2525416kB time=02:03:50.75 bitrate=2784.1kbits/s > size= 2528206kB time=02:03:58.56 bitrate=2784.3kbits/s > > > video:0kB audio:2259932kB subtitle:0 global headers:0kB muxing overhead > 11.870920% > > > frame=1366679008 > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > but when i do this: > ./ffmpeg-0 -y -i audio.ts > > > ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers > built on Jul 31 2013 10:34:48 with gcc 4.4.6 (GCC) 20110731 (Red Hat > 4.4.6-3) > configuration: --prefix=/data/test/jh/ffmpeg > --extra-cflags=-I/data/test/jh/src/ffmpeg/include > --extra-ldflags=-L/data/test/jh/src/ffmpeg/lib --enable-gpl > --enable-version3 --enable-nonfree --enable-debug --enable-memalign-hack > --enable-libfaac --enable-libaacplus --enable-libfreetype > --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 > --enable-libxvid > libavutil 52. 38.100 / 52. 38.100 > libavcodec 55. 18.102 / 55. 18.102 > libavformat 55. 12.100 / 55. 12.100 > libavdevice 55. 3.100 / 55. 3.100 > libavfilter 3. 79.101 / 3. 79.101 > libswscale 2. 3.100 / 2. 3.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 3.100 / 52. 3.100 > [mpegts @ 0x22d6020] probed stream 0 failed > [mpegts @ 0x22d6020] Could not find codec parameters for stream 0 (Unknown: > none ([6][0][0][0] / 0x0006)): unknown codec > Consider increasing the value for the 'analyzeduration' and 'probesize' > options > audio.ts: could not find codec parameters > > > > > > > I have serveral this file?what can i do? > > > > > > thank you > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Sun Oct 13 10:01:40 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 13 Oct 2013 08:01:40 +0000 (UTC) Subject: [FFmpeg-user] Extract Audio from Video file using ffmpeg References: Message-ID: Jacky <690271929 qq.com> writes: > Output #0, mpegts, to 'audio.ts': > Metadata: > encoder : Lavf55.12.100 > Stream #0:0: Audio: truehd (AC-3 / 0x332D4341), 48000 Hz, 5.1(side) > Stream mapping: > Stream #0:1 -> #0:0 (copy) (Adding the console output is of course good, but please also add the command line next time.) Muxing truehd in mpegts is not supported, patch welcome! Carl Eugen From cehoyos at ag.or.at Sun Oct 13 10:03:00 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 13 Oct 2013 08:03:00 +0000 (UTC) Subject: [FFmpeg-user] Different conversion results on similar conversion runs. References: Message-ID: Anatol gmail.com> writes: > "same conversion session" means: > - same machine > - same source > - same ffmpeg > - same command line Did you try a native encoder? -threads 1? Carl Eugen From anatol2002 at gmail.com Sun Oct 13 10:36:53 2013 From: anatol2002 at gmail.com (Anatol) Date: Sun, 13 Oct 2013 11:36:53 +0300 Subject: [FFmpeg-user] Different conversion results on similar conversion runs. In-Reply-To: References: Message-ID: Hi Carl, Native encoder does the same - couple of kbps difference. '-thread 1' - there are some source formats (mpeg-ts if i recall it correctly), that occasionally get crazy during decoding ("occasionally" - some server hw setups, running certain Linux versions ... it was relevant for ffmpeg 0.10). I tried the same command line w/out "thread 1" - same results. On Sun, Oct 13, 2013 at 11:03 AM, Carl Eugen Hoyos wrote: > Anatol gmail.com> writes: > > > "same conversion session" means: > > - same machine > > - same source > > - same ffmpeg > > - same command line > > Did you try a native encoder? > -threads 1? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Sun Oct 13 13:27:05 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 13 Oct 2013 11:27:05 +0000 (UTC) Subject: [FFmpeg-user] Different conversion results on similar conversion runs. References: Message-ID: Anatol gmail.com> writes: > Native encoder does the same - couple of kbps difference. Please post a command line with a native encoder and -threads 1, I would like to test. Please do not top-post here, Carl Eugen From anatol2002 at gmail.com Sun Oct 13 13:50:54 2013 From: anatol2002 at gmail.com (Anatol) Date: Sun, 13 Oct 2013 14:50:54 +0300 Subject: [FFmpeg-user] Different conversion results on similar conversion runs. In-Reply-To: References: Message-ID: First the command line u asked for: ffmpeg -i input.mp4 -c:v h264 -subq 5 -qcomp 0.6 -qmin 10 -qmax 50 -qdiff 4 -coder 1 -refs 2 -vprofile main -force_key_frames 0,2.002,4.004,6.006,8.008,10.01,12.012,14.014,16.016,18.018,20.02,22.022,24.024,26.026,28.028,30.03,32.032,34.034,36.036,38.038,40.04,42.042,44.044,46.046,48.048,50.0501,52.0521,54.0541,56.0561,58.0581,60.0601,62.0621,64.0641,66.0661,68.0681 -pix_fmt yuv420p -b:v 1200k -s 848x480 -r 29.97 -g 60 -c:a libfaac -b:a 64k -ar 44100 -ac 2 -map_chapters -1 -map_metadata -1 -f mp4 -flags +loop+mv4 -cmp 256 -partitions +parti4x4+partp8x8+partb8x8 -trellis 1 -refs 1 -me_range 16 -keyint_min 20 -sc_threshold 40 -i_qfactor 0.71 -bt 400k -maxrate 1200k -bufsize 1200k -rc_eq 'blurCplx^(1-qComp)' -level 30 -async 2 -vsync 1 -y output.mp4 I got a hint from the FFmpeg Support Forum. It is about setting encoding side 'threads' value (rather than decoding side in my original post). When the ENCODING threads are constrained to 1 - the results are the consistent. When they are no encoding 'threads' setting (or it is set 0), ffmpeg allocates uses all machine cores. In my case - 48. When I set the threads to 4 - the result is still consistent, For 10 it became inconsistent again. I understand why there are differences when the command line is executed with different threads settings. I don't understand why execution of exactly the same command line generates different files on consecutive runs. Anatol From cehoyos at ag.or.at Sun Oct 13 14:01:46 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 13 Oct 2013 12:01:46 +0000 (UTC) Subject: [FFmpeg-user] Different conversion results on similar conversion runs. References: Message-ID: Anatol gmail.com> writes: > > First the command line u asked for: > ffmpeg -i input.mp4 -c:v h264 I asked for a command line that does not use external libraries (like x264). The reason is that I know the FFmpeg version you are using (and I can test the exact version) if necessary but I cannot know your x264 version (and it takes a magnitude more effort to use the same version here). Since you said the problem is reproducible without external libraries, I'd like to test that. > I got a hint from the FFmpeg Support Forum. > It is about setting encoding side 'threads' > value (rather than decoding side in my > original post). Yes, this is what I wrote this morning. (It would be a bug if setting the decoder thread count to a different value made a difference.) > I understand why there are differences when the > command line is executed with different threads > settings. Good. > I don't understand why execution of exactly the > same command line generates different files on > consecutive runs. Then you apparently do not understand above... (Or I don't but I suspect FFmpeg cannot know how much work your whole system has to do at a given moment, it cannot be multi-threaded and deterministic at the same time.) Carl Eugen From anatol2002 at gmail.com Sun Oct 13 14:44:19 2013 From: anatol2002 at gmail.com (Anatol) Date: Sun, 13 Oct 2013 15:44:19 +0300 Subject: [FFmpeg-user] Different conversion results on similar conversion runs. In-Reply-To: References: Message-ID: I thought that setting the "c:v h264" will bring in the ffmpeg native h264 encoder, but seems that it does not. How do I run native h264? I use libx264 core 129, it is in the uncut ffmpeg printout... In all "auto-multi-threaded" runs, ffmpeg used all 48 cores, therefore, IMHO, the renditions should have been consistent. But they are not. From onemda at gmail.com Sun Oct 13 14:46:22 2013 From: onemda at gmail.com (Paul B Mahol) Date: Sun, 13 Oct 2013 12:46:22 +0000 Subject: [FFmpeg-user] Different conversion results on similar conversion runs. In-Reply-To: References: Message-ID: On 10/13/13, Anatol wrote: > I thought that setting the "c:v h264" will bring in the ffmpeg native h264 > encoder, but seems that it does not. > How do I run native h264? > I use libx264 core 129, it is in the uncut ffmpeg printout... There is no native h264 encoder. > > In all "auto-multi-threaded" runs, ffmpeg used all 48 cores, therefore, > IMHO, the renditions should have been consistent. > But they are not. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From andrzej.nikitorowicz at gmail.com Sun Oct 13 15:41:21 2013 From: andrzej.nikitorowicz at gmail.com (Andrzej Nikitorowicz) Date: Sun, 13 Oct 2013 15:41:21 +0200 Subject: [FFmpeg-user] Broken metadata handling with ogg vorbis? Message-ID: <20131013154121.43e00a18@icore7.at.andyn> andynek_15:26:37 z $ ffmpeg -i J.Fijor-2013.10.09.Wed-21.00-1.ogg -frames 1 -f ffmetadata metadata.txt ffmpeg version N-56841-g4d5d905 Copyright (c) 2000-2013 the FFmpeg developers built on Oct 3 2013 18:40:50 with gcc 4.7.2 (Gentoo 4.7.2-r1 p1.4, pie-0.5.5) configuration: --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='-O2 -pipe -march=native ' --extra-cflags='-O2 -pipe -march=native ' --extra-cxxflags='-O2 -pipe -march=native ' --disable-static --enable-gpl --enable-postproc --enable-avfilter --enable-avresample --disable-stripping --enable-nonfree --enable-version3 --enable- nonfree --enable-version3 --enable-bzlib --disable-runtime-cpudetect --disable-debug --enable-doc --enable-gnutls --enable-hardcoded-tables --enable-iconv --enable-network --enable-openssl --enable-ffplay --enable-vaapi --enable-vdpau -- enable-zlib --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libmp3lame --enable-libfdk-aac --enable-libaacplus --enable-libfaac --enable-libtheora --enable-libtwolame --enable-libwavpack --enable-libx264 --enable-libxvid --disable -libcdio --disable-libiec61883 --disable-libdc1394 --enable-libcaca --enable-openal --enable-libv4l2 --enable-libpulse --enable-x11grab --enable-libflite --enable-frei0r --enable-fontconfig --enable-libass --enable-libfreetype --enable-l ibsoxr --enable-pthreads --enable-libopencore-amrwb --enable-libopencore-amrnb --enable-libopenjpeg --disable-libbluray --disable-libcelt --enable-libgme --enable-libgsm --enable-libmodplug --enable-libopus --enable-libquvi --enable-libr tmp --enable-libschroedinger --enable-libspeex --enable-libvorbis --enable-libvpx --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-vis --disable-neon --enable-pic --cpu=host libavutil 52. 46.100 / 52. 46.100 libavcodec 55. 34.100 / 55. 34.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 88.101 / 3. 88.101 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, ogg, from 'J.Fijor-2013.10.09.Wed-21.00-1.ogg': Duration: 00:36:51.50, start: 0.000000, bitrate: 33 kb/s Stream #0:0: Audio: vorbis, 44100 Hz, stereo, fltp, 48 kb/s Metadata: TITLE : Janek Fijor - 2013.10.09.Wed-21.00-1 ARTIST : Radio Kontestacja ALBUM : Janek Fijor DATE : 2013 GENRE : Polityka track : 1 COMMENT : Blacha wie co m?wi Output #0, ffmetadata, to 'metadata.txt': Stream mapping: Press [q] to stop, [?] for help size= 0kB time=-577014:-32:-22.-77 bitrate=N/A video:0kB audio:0kB subtitle:0 global headers:0kB muxing overhead inf% Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used) andynek_15:26:39 z $ andynek_15:26:41 z $ cat metadata.txt ;FFMETADATA1 Another question, howto copy metadata from one file to anther? andynek_15:36:58 z $ ffmpeg -i "concat:J.Fijor-2013.10.09.Wed-20.57.30-1.ogg|J.Fijor-2013.10.09.Wed-21.00-1.ogg" \ -i J.Fijor-2013.10.09.Wed-21.00-1.ogg -acodec copy J.Fijor-2013.10.09.Wed-20.57.30-11.ogg -map_metadata 0:1 ffmpeg version N-56841-g4d5d905 Copyright (c) 2000-2013 the FFmpeg developers built on Oct 3 2013 18:40:50 with gcc 4.7.2 (Gentoo 4.7.2-r1 p1.4, pie-0.5.5) configuration: --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='-O2 -pipe -march=native ' --extra-cflags='-O2 -pipe -march=native ' --extra-cxxflags='-O2 -pipe -march=native ' --disable-static --enable-gpl --enable-postproc --enable-avfilter --enable-avresample --disable-stripping --enable-nonfree --enable-version3 --enable- nonfree --enable-version3 --enable-bzlib --disable-runtime-cpudetect --disable-debug --enable-doc --enable-gnutls --enable-hardcoded-tables --enable-iconv --enable-network --enable-openssl --enable-ffplay --enable-vaapi --enable-vdpau -- enable-zlib --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libmp3lame --enable-libfdk-aac --enable-libaacplus --enable-libfaac --enable-libtheora --enable-libtwolame --enable-libwavpack --enable-libx264 --enable-libxvid --disable -libcdio --disable-libiec61883 --disable-libdc1394 --enable-libcaca --enable-openal --enable-libv4l2 --enable-libpulse --enable-x11grab --enable-libflite --enable-frei0r --enable-fontconfig --enable-libass --enable-libfreetype --enable-l ibsoxr --enable-pthreads --enable-libopencore-amrwb --enable-libopencore-amrnb --enable-libopenjpeg --disable-libbluray --disable-libcelt --enable-libgme --enable-libgsm --enable-libmodplug --enable-libopus --enable-libquvi --enable-libr tmp --enable-libschroedinger --enable-libspeex --enable-libvorbis --enable-libvpx --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-vis --disable-neon --enable-pic --cpu=host libavutil 52. 46.100 / 52. 46.100 libavcodec 55. 34.100 / 55. 34.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 88.101 / 3. 88.101 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Trailing options were found on the commandline. [ogg @ 0x667620] Cannot identify new stream [ogg @ 0x667620] failed to create or replace stream Input #0, ogg, from 'concat:J.Fijor-2013.10.09.Wed-20.57.30-1.ogg|J.Fijor-2013.10.09.Wed-21.00-1.ogg': Duration: 159957:47:02.63, start: 0.019252, bitrate: N/A Stream #0:0: Audio: vorbis, 44100 Hz, stereo, fltp, 48 kb/s Metadata: ALBUM : Kontestacja.com ARTIST : Jan Fijor Kamil Cebulski TITLE : Jan Fijor - Blacha wie co m?wi ENCODER : Lavf55.19.100 Input #1, ogg, from 'J.Fijor-2013.10.09.Wed-21.00-1.ogg': Duration: 00:36:51.50, start: 0.000000, bitrate: 33 kb/s Stream #1:0: Audio: vorbis, 44100 Hz, stereo, fltp, 48 kb/s Metadata: TITLE : Janek Fijor - 2013.10.09.Wed-21.00-1 ARTIST : Radio Kontestacja ALBUM : Janek Fijor DATE : 2013 GENRE : Polityka track : 1 COMMENT : Blacha wie co m?wi File 'J.Fijor-2013.10.09.Wed-20.57.30-11.ogg' already exists. Overwrite ? [y/N] y Output #0, ogg, to 'J.Fijor-2013.10.09.Wed-20.57.30-11.ogg': Metadata: encoder : Lavf55.19.100 Stream #0:0: Audio: vorbis, 44100 Hz, stereo, 48 kb/s Metadata: ALBUM : Kontestacja.com ARTIST : Jan Fijor Kamil Cebulski TITLE : Jan Fijor - Blacha wie co m?wi ENCODER : Lavf55.19.100 Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help [NULL @ 0x630e40] Invalid packet bitrate= 40.8kbits/s Last message repeated 2 times [ogg @ 0x63b2c0] Non-monotonous DTS in output stream 0:0; previous: 6834690, current: 6834688; changing to 6834691. This may result in incorrect timestamps in the output file. size= 9895kB time=00:39:26.46 bitrate= 34.3kbits/s video:0kB audio:9775kB subtitle:0 global headers:0kB muxing overhead 1.227961% andynek_15:37:37 z $ Result is copy metadata from J.Fijor-2013.10.09.Wed-20.57.30-1.ogg, but I want metadata from J.Fijor-2013.10.09.Wed-21.00-1.ogg From andrzej.nikitorowicz at gmail.com Sun Oct 13 15:46:06 2013 From: andrzej.nikitorowicz at gmail.com (Andrzej Nikitorowicz) Date: Sun, 13 Oct 2013 15:46:06 +0200 Subject: [FFmpeg-user] Broken metadata handling with ogg vorbis? Message-ID: <20131013154606.462dd86a@icore7.at.andyn> andynek_15:26:37 z $ ffmpeg -i J.Fijor-2013.10.09.Wed-21.00-1.ogg -frames 1 -f ffmetadata metadata.txt ffmpeg version N-56841-g4d5d905 Copyright (c) 2000-2013 the FFmpeg developers built on Oct 3 2013 18:40:50 with gcc 4.7.2 (Gentoo 4.7.2-r1 p1.4, pie-0.5.5) configuration: --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='-O2 -pipe -march=native ' --extra-cflags='-O2 -pipe -march=native ' --extra-cxxflags='-O2 -pipe -march=native ' --disable-static --enable-gpl --enable-postproc --enable-avfilter --enable-avresample --disable-stripping --enable-nonfree --enable-version3 --enable- nonfree --enable-version3 --enable-bzlib --disable-runtime-cpudetect --disable-debug --enable-doc --enable-gnutls --enable-hardcoded-tables --enable-iconv --enable-network --enable-openssl --enable-ffplay --enable-vaapi --enable-vdpau -- enable-zlib --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libmp3lame --enable-libfdk-aac --enable-libaacplus --enable-libfaac --enable-libtheora --enable-libtwolame --enable-libwavpack --enable-libx264 --enable-libxvid --disable -libcdio --disable-libiec61883 --disable-libdc1394 --enable-libcaca --enable-openal --enable-libv4l2 --enable-libpulse --enable-x11grab --enable-libflite --enable-frei0r --enable-fontconfig --enable-libass --enable-libfreetype --enable-l ibsoxr --enable-pthreads --enable-libopencore-amrwb --enable-libopencore-amrnb --enable-libopenjpeg --disable-libbluray --disable-libcelt --enable-libgme --enable-libgsm --enable-libmodplug --enable-libopus --enable-libquvi --enable-libr tmp --enable-libschroedinger --enable-libspeex --enable-libvorbis --enable-libvpx --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-vis --disable-neon --enable-pic --cpu=host libavutil 52. 46.100 / 52. 46.100 libavcodec 55. 34.100 / 55. 34.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 88.101 / 3. 88.101 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, ogg, from 'J.Fijor-2013.10.09.Wed-21.00-1.ogg': Duration: 00:36:51.50, start: 0.000000, bitrate: 33 kb/s Stream #0:0: Audio: vorbis, 44100 Hz, stereo, fltp, 48 kb/s Metadata: TITLE : Janek Fijor - 2013.10.09.Wed-21.00-1 ARTIST : Radio Kontestacja ALBUM : Janek Fijor DATE : 2013 GENRE : Polityka track : 1 COMMENT : Blacha wie co m?wi Output #0, ffmetadata, to 'metadata.txt': Stream mapping: Press [q] to stop, [?] for help size= 0kB time=-577014:-32:-22.-77 bitrate=N/A video:0kB audio:0kB subtitle:0 global headers:0kB muxing overhead inf% Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used) andynek_15:26:39 z $ andynek_15:26:41 z $ cat metadata.txt ;FFMETADATA1 Another question, howto copy metadata from one file to anther? andynek_15:36:58 z $ ffmpeg -i "concat:J.Fijor-2013.10.09.Wed-20.57.30-1.ogg|J.Fijor-2013.10.09.Wed-21.00-1.ogg" \ -i J.Fijor-2013.10.09.Wed-21.00-1.ogg -acodec copy J.Fijor-2013.10.09.Wed-20.57.30-11.ogg -map_metadata 0:1 ffmpeg version N-56841-g4d5d905 Copyright (c) 2000-2013 the FFmpeg developers built on Oct 3 2013 18:40:50 with gcc 4.7.2 (Gentoo 4.7.2-r1 p1.4, pie-0.5.5) configuration: --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='-O2 -pipe -march=native ' --extra-cflags='-O2 -pipe -march=native ' --extra-cxxflags='-O2 -pipe -march=native ' --disable-static --enable-gpl --enable-postproc --enable-avfilter --enable-avresample --disable-stripping --enable-nonfree --enable-version3 --enable- nonfree --enable-version3 --enable-bzlib --disable-runtime-cpudetect --disable-debug --enable-doc --enable-gnutls --enable-hardcoded-tables --enable-iconv --enable-network --enable-openssl --enable-ffplay --enable-vaapi --enable-vdpau -- enable-zlib --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libmp3lame --enable-libfdk-aac --enable-libaacplus --enable-libfaac --enable-libtheora --enable-libtwolame --enable-libwavpack --enable-libx264 --enable-libxvid --disable -libcdio --disable-libiec61883 --disable-libdc1394 --enable-libcaca --enable-openal --enable-libv4l2 --enable-libpulse --enable-x11grab --enable-libflite --enable-frei0r --enable-fontconfig --enable-libass --enable-libfreetype --enable-l ibsoxr --enable-pthreads --enable-libopencore-amrwb --enable-libopencore-amrnb --enable-libopenjpeg --disable-libbluray --disable-libcelt --enable-libgme --enable-libgsm --enable-libmodplug --enable-libopus --enable-libquvi --enable-libr tmp --enable-libschroedinger --enable-libspeex --enable-libvorbis --enable-libvpx --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-vis --disable-neon --enable-pic --cpu=host libavutil 52. 46.100 / 52. 46.100 libavcodec 55. 34.100 / 55. 34.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 88.101 / 3. 88.101 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Trailing options were found on the commandline. [ogg @ 0x667620] Cannot identify new stream [ogg @ 0x667620] failed to create or replace stream Input #0, ogg, from 'concat:J.Fijor-2013.10.09.Wed-20.57.30-1.ogg|J.Fijor-2013.10.09.Wed-21.00-1.ogg': Duration: 159957:47:02.63, start: 0.019252, bitrate: N/A Stream #0:0: Audio: vorbis, 44100 Hz, stereo, fltp, 48 kb/s Metadata: ALBUM : Kontestacja.com ARTIST : Jan Fijor Kamil Cebulski TITLE : Jan Fijor - Blacha wie co m?wi ENCODER : Lavf55.19.100 Input #1, ogg, from 'J.Fijor-2013.10.09.Wed-21.00-1.ogg': Duration: 00:36:51.50, start: 0.000000, bitrate: 33 kb/s Stream #1:0: Audio: vorbis, 44100 Hz, stereo, fltp, 48 kb/s Metadata: TITLE : Janek Fijor - 2013.10.09.Wed-21.00-1 ARTIST : Radio Kontestacja ALBUM : Janek Fijor DATE : 2013 GENRE : Polityka track : 1 COMMENT : Blacha wie co m?wi File 'J.Fijor-2013.10.09.Wed-20.57.30-11.ogg' already exists. Overwrite ? [y/N] y Output #0, ogg, to 'J.Fijor-2013.10.09.Wed-20.57.30-11.ogg': Metadata: encoder : Lavf55.19.100 Stream #0:0: Audio: vorbis, 44100 Hz, stereo, 48 kb/s Metadata: ALBUM : Kontestacja.com ARTIST : Jan Fijor Kamil Cebulski TITLE : Jan Fijor - Blacha wie co m?wi ENCODER : Lavf55.19.100 Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help [NULL @ 0x630e40] Invalid packet bitrate= 40.8kbits/s Last message repeated 2 times [ogg @ 0x63b2c0] Non-monotonous DTS in output stream 0:0; previous: 6834690, current: 6834688; changing to 6834691. This may result in incorrect timestamps in the output file. size= 9895kB time=00:39:26.46 bitrate= 34.3kbits/s video:0kB audio:9775kB subtitle:0 global headers:0kB muxing overhead 1.227961% andynek_15:37:37 z $ Result is copy metadata from J.Fijor-2013.10.09.Wed-20.57.30-1.ogg, but I want metadata from J.Fijor-2013.10.09.Wed-21.00-1.ogg From junjiepang at gmail.com Sun Oct 13 16:50:26 2013 From: junjiepang at gmail.com (Jun Jie Pang) Date: Sun, 13 Oct 2013 22:50:26 +0800 Subject: [FFmpeg-user] Qns regarding compiling c++ code with ffmpeg Message-ID: Hi, I have installed ffmpeg on centOS according to https://trac.ffmpeg.org/wiki/CentosCompilationGuide. However, when I try to compile the code, it gives me multiple " undefined reference to ....". I have looked up on this problem, already tried and implemented a few things: 1) Adding: extern "C" { #includes ...} 2) Compiling with path to lib: $gcc prog prog.cpp -L/path/to/lib -lavcodec -lavformat -lavutil However, I still couldn't compile the example code. I'm at a lost here. Could any kinds souls help look into my problem ? Thank you, Pang Jun Jie From cehoyos at ag.or.at Sun Oct 13 23:31:00 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 13 Oct 2013 21:31:00 +0000 (UTC) Subject: [FFmpeg-user] Qns regarding compiling c++ code with ffmpeg References: Message-ID: Jun Jie Pang gmail.com> writes: > I have installed ffmpeg on centOS according to > https://trac.ffmpeg.org/wiki/CentosCompilationGuide. > However, when I try to compile the code, it gives > me multiple " undefined reference to ....". Unfortunately, this is not enough information;-( What are you trying to compile, how and what output does the linker print? Carl Eugen From lou at lrcd.com Sun Oct 13 23:45:51 2013 From: lou at lrcd.com (Lou) Date: Sun, 13 Oct 2013 13:45:51 -0800 Subject: [FFmpeg-user] Following UbuntuCompilationGuide running into trouble with the x264 part In-Reply-To: <20131013001615.t8vhlkbk004480o0@wwwmail.urz.uni-heidelberg.de> References: <20131013001615.t8vhlkbk004480o0@wwwmail.urz.uni-heidelberg.de> Message-ID: <20131013134551.71422353@lrcd.com> On Sun, 13 Oct 2013 00:16:15 +0200 rieger at ari.uni-heidelberg.de wrote: > /usr/local/lib/libavcodec.a(libx264.o): In function `X264_init': > /usr/src/ffmpeg/libavcodec/libx264.c:557: undefined reference to > `x264_encoder_open_135' Check to see if you're trying to to use ffmpeg (for lavfi support in x264) that is already linked to another installed version of x264. From lou at lrcd.com Sun Oct 13 23:49:08 2013 From: lou at lrcd.com (Lou) Date: Sun, 13 Oct 2013 13:49:08 -0800 Subject: [FFmpeg-user] Following UbuntuCompilationGuide running into trouble with the x264 part In-Reply-To: <20131013134551.71422353@lrcd.com> References: <20131013001615.t8vhlkbk004480o0@wwwmail.urz.uni-heidelberg.de> <20131013134551.71422353@lrcd.com> Message-ID: <20131013134908.742a7641@lrcd.com> On Sun, 13 Oct 2013 13:45:51 -0800 Lou wrote: > Check to see if you're trying to to use ffmpeg (for lavfi support in > x264) that is already linked to another installed version of x264. s/lavfi/lavf (aka libavformat) From lingjiujianke at gmail.com Mon Oct 14 01:35:33 2013 From: lingjiujianke at gmail.com (=?GB2312?B?wfUg4ao=?=) Date: Mon, 14 Oct 2013 07:35:33 +0800 Subject: [FFmpeg-user] Qns regarding compiling c++ code with ffmpeg In-Reply-To: References: Message-ID: <5F5A0287-51D7-457F-8458-66AFE74E0186@gmail.com> ? 2013-10-14???5:31?Carl Eugen Hoyos ??? > Jun Jie Pang gmail.com> writes: > >> I have installed ffmpeg on centOS according to >> https://trac.ffmpeg.org/wiki/CentosCompilationGuide. >> However, when I try to compile the code, it gives >> me multiple " undefined reference to ....". > > Unfortunately, this is not enough information;-( > What are you trying to compile, how and what > output does the linker print? > Junjie Pang, Please give us the enough information, Steven From soho123.2012 at gmail.com Mon Oct 14 07:25:14 2013 From: soho123.2012 at gmail.com (Huang Soho) Date: Mon, 14 Oct 2013 13:25:14 +0800 Subject: [FFmpeg-user] ffmpeg how to output multiple file In-Reply-To: References: Message-ID: In advance, can i limit the size of output file? If I would like to output to a file and each file have the same file size , is it possible? 2013/10/10 Jason Runta > It can. See this article for more information: > http://ffmpeg.org/trac/ffmpeg/wiki/Creating%20multiple%20outputs > > The important part is the tee command and getting the mappings correct. > Here's an example I was using to test: > > ffmpeg -y -f dshow -s 640x480 -r 29.97 -i video="Microsoft LifeCam > Cinema":audio="Desktop Microphone (Cinema - Mi" -c:v libx264 -preset > ultrafast -b:v 800k -c:a aac -strict experimental -ar 44100 -b:a 56k -f tee > -map 0:0 -map 0:1 "[f=rtsp] -rtscp_transport tcp > rtsp://localhost:1234/live.sdp|C:\\users\\jasonr\\mymovie.mkv" > > and then using ffplay I could watch my stream using: > ffplay -rtsp_flags listen rtsp://localhost:1234/live.sdp > > > On Wed, Oct 9, 2013 at 6:41 AM, Huang Soho wrote: > > > Hi All, > > > > Can ffmpeg output to multiple file? > > for example: > > if the command line is : > > ffmpeg -sn -f video4linux2 -r 30 -s 1280x720 -input_format h264 -i > > /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy > -map > > 0:0 -map 1:0 http://localhost:8090/feed2.ffm > > > > I can capture video and audio from device, then output to feed2.ffm, > > Can ffmpeg output the video and audio data to local file? for example > .avi > > or others? > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > -- > *-_-=Jason Runta=-_-* > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From soho123.2012 at gmail.com Mon Oct 14 07:29:30 2013 From: soho123.2012 at gmail.com (Huang Soho) Date: Mon, 14 Oct 2013 13:29:30 +0800 Subject: [FFmpeg-user] how to limit the size or time duration when using ffmpeg output to a file? Message-ID: Hi All, Can I use ffmpeg to capture video data from USB webcam and save to local SD card by every 100MB? for example: ffmpeg -sn -f video4linux2 -r 30 -s 1280x720 -input_format h264 -i /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy -map 0:0 -map 1:0 http://localhost:8090/feed2.ffm I can ouput video + audio to feed file. And I can save to loacal, But how I can save video+audio to SD card by every 100MB or every 5 minutes? From zuwei at imrsv.com Mon Oct 14 08:47:55 2013 From: zuwei at imrsv.com (Zuwei Li) Date: Mon, 14 Oct 2013 14:47:55 +0800 Subject: [FFmpeg-user] Does FFmpeg support RTMP dynamic streaming Message-ID: Does FFMpeg support RTMP dynamic streaming using SMIL manifest or smiliar? Is there a mechanism in ffmpeg to notify the server to request to change streams described in the following article: http://www.streamingmedia.com/Articles/Editorial/Featured-Articles/How-to-do-Dynamic-Streaming-with-Flash-Media-Server---66199.aspx -- Li Zuwei From soho123.2012 at gmail.com Mon Oct 14 10:33:41 2013 From: soho123.2012 at gmail.com (Huang Soho) Date: Mon, 14 Oct 2013 16:33:41 +0800 Subject: [FFmpeg-user] How to output to multiple file "DYNAMICALLY"? Message-ID: Hi All, Can ffmpeg output to multiple file dynamically? for example: if the command line is : ffmpeg -sn -f video4linux2 -r 30 -s 1280x720 -input_format h264 -i /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy -map 0:0 -map 1:0 http://localhost:8090/feed2.ffm I can capture video and audio from device, then output to feed2.ffm, Can ffmpeg output the video and audio data to local file if there is the specific event take place? for example : if the current time is match that user setup, or motion detection is occur, ....etc.... From soho123.2012 at gmail.com Mon Oct 14 11:10:04 2013 From: soho123.2012 at gmail.com (Huang Soho) Date: Mon, 14 Oct 2013 17:10:04 +0800 Subject: [FFmpeg-user] ffmpeg how to output multiple file In-Reply-To: References: Message-ID: 2013/10/10 Jason Runta > It can. See this article for more information: > http://ffmpeg.org/trac/ffmpeg/wiki/Creating%20multiple%20outputs > > The important part is the tee command and getting the mappings correct. > Here's an example I was using to test: > > ffmpeg -y -f dshow -s 640x480 -r 29.97 -i video="Microsoft LifeCam > Cinema":audio="Desktop Microphone (Cinema - Mi" -c:v libx264 -preset > ultrafast -b:v 800k -c:a aac -strict experimental -ar 44100 -b:a 56k -f tee > -map 0:0 -map 0:1 "[f=rtsp] -rtscp_transport tcp > rtsp://localhost:1234/live.sdp|C:\\users\\jasonr\\mymovie.mkv" > > and then using ffplay I could watch my stream using: > ffplay -rtsp_flags listen rtsp://localhost:1234/live.sdp > > ffmpeg -sn -f video4linux2 -r 30 -s 1280x720 -input_format h264 -i /dev/video1 -vcodec copy http://localhost:8090/feed2.ffm -vcodec copy /var/tmp/usb/sda1/test.ts the command I used to test, I have enable mpegts muxer when I make ffmpeg, but the output file test.ts can not be play when I try to VLC to open test.ts. Do you have any idea? > > > -- > *-_-=Jason Runta=-_-* > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From soho123.2012 at gmail.com Mon Oct 14 11:30:17 2013 From: soho123.2012 at gmail.com (Huang Soho) Date: Mon, 14 Oct 2013 17:30:17 +0800 Subject: [FFmpeg-user] how can I set limit_filesize or recording_time for the specific output dynamically? Message-ID: Hi All, how can I set limit_filesize or recording_time for the specific output? Is it possible to set by "dynamically"? for example: if the command line is : ffmpeg -sn -f video4linux2 -r 30 -s 1280x720 -input_format h264 -i /dev/video1 -vcodec copy http://localhost:8090/feed2.ffm -vcodec copy /var/tmp/usb/sda1/test1.ts I can capture video from usb device, then output to feed2.ffm, Can ffmpeg output the video and audio data to local file if there is the specific event take place? for example : if the current time is match that user setup, or motion detection is occur, ....etc.... and Can I set the limit file size or time duration for the specific output file ? From cehoyos at ag.or.at Mon Oct 14 16:14:45 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 14 Oct 2013 14:14:45 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?how_can_I_set_limit=5Ffilesize_or_recordi?= =?utf-8?q?ng=5Ftime_for_the_specific_output_dynamically=3F?= References: Message-ID: Huang Soho gmail.com> writes: > how can I set limit_filesize or recording_time for the specific output? > Is it possible to set by "dynamically"? > for example: > if the command line is : > ffmpeg -sn -f video4linux2 -r 30 -s 1280x720 -input_format h264 -i > /dev/video1 -vcodec copy http://localhost:8090/feed2.ffm -vcodec copy > /var/tmp/usb/sda1/test1.ts (Complete, uncut console output missing.) There is the segment muxer, see 4.15 in http://ffmpeg.org/ffmpeg-formats.html Carl Eugen From soho123.2012 at gmail.com Mon Oct 14 18:32:25 2013 From: soho123.2012 at gmail.com (Huang Soho) Date: Tue, 15 Oct 2013 00:32:25 +0800 Subject: [FFmpeg-user] how can I set limit_filesize or recording_time for the specific output dynamically? In-Reply-To: References: Message-ID: 2013/10/14 Carl Eugen Hoyos > Huang Soho gmail.com> writes: > > > how can I set limit_filesize or recording_time for the specific output? > > Is it possible to set by "dynamically"? > > for example: > > if the command line is : > > ffmpeg -sn -f video4linux2 -r 30 -s 1280x720 -input_format h264 -i > > /dev/video1 -vcodec copy http://localhost:8090/feed2.ffm -vcodec copy > > /var/tmp/usb/sda1/test1.ts > > (Complete, uncut console output missing.) > > There is the segment muxer, see 4.15 in > http://ffmpeg.org/ffmpeg-formats.html > > I do not know about how to show the output from the example command line. But the "segment" seems does not meet my requirement. I would like to save multiple files that each file hase fixed size. for example: ffmpeg -sn -f video4linux2 -r 30 -s 1280x720 -input_format h264 -i /dev/video1 -vcodec copy http://localhost:8090/feed2.ffm -vcodec copy /var/tmp/usb/sda1/test1.ts the command can capture live video data from /dev/video1 and send to feed2.ffm for ffserver , then ffserver can output to remote side for preview, and can write video data to test1.ts, continuely. the command can not set the file size while the file "test1.ts" is get bigger. I would like to :let ffserver output the live stream to remote side for preview and save the live video stream to local SD card (multiple files)by fixed size. if the segment you mentioned can get the purpose that I want, Could you kindly help to modify the command I write above for show me the example? every input is very appreciated!!!! thanks a lot! From junjiepang at gmail.com Mon Oct 14 18:44:19 2013 From: junjiepang at gmail.com (Jun Jie Pang) Date: Tue, 15 Oct 2013 00:44:19 +0800 Subject: [FFmpeg-user] Qns regarding compiling c++ code with ffmpeg In-Reply-To: <5F5A0287-51D7-457F-8458-66AFE74E0186@gmail.com> References: <5F5A0287-51D7-457F-8458-66AFE74E0186@gmail.com> Message-ID: Hi, Thank you guys for your input ! Sorry for not providing sufficient information. The command ran: gcc -o sample_vid_encoder sample_vid_encoder.cpp -L/$HOME/ffmpeg_build/lib -lavcodec -lavformat -lavutil The error message: //home/Pang/ffmpeg_build/lib/libavcodec.a(amrnbdec.o): In function `amrnb_decode_frame': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/amrnbdec.c:1035: undefined reference to `truncf' //home/Pang/ffmpeg_build/lib/libavcodec.a(amrwbdec.o): In function `amrwb_decode_frame': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/amrwbdec.c:1204: undefined reference to `truncf' //home/Pang/ffmpeg_build/lib/libavcodec.a(atrac3.o): In function `init_imdct_window': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/atrac3.c:181: undefined reference to `sin' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/atrac3.c:182: undefined reference to `sin' //home/Pang/ffmpeg_build/lib/libavcodec.a(binkaudio.o): In function `decode_init': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/binkaudio.c:116: undefined reference to `expf' //home/Pang/ffmpeg_build/lib/libavcodec.a(cngdec.o): In function `cng_decode_frame': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/cngdec.c:115: undefined reference to `pow' //home/Pang/ffmpeg_build/lib/libavcodec.a(cngenc.o): In function `cng_encode_frame': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/cngenc.c:89: undefined reference to `log10' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/cngenc.c:90: undefined reference to `floor' //home/Pang/ffmpeg_build/lib/libavcodec.a(cook.o): In function `init_pow2table': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/cook.c:169: undefined reference to `pow' //home/Pang/ffmpeg_build/lib/libavcodec.a(cook.o): In function `init_gain_table': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/cook.c:180: undefined reference to `pow' //home/Pang/ffmpeg_build/lib/libavcodec.a(cscd.o): In function `decode_frame': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/cscd.c:91: undefined reference to `uncompress' //home/Pang/ffmpeg_build/lib/libavcodec.a(dct.o): In function `ff_dct_init': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/dct.c:201: undefined reference to `sin' //home/Pang/ffmpeg_build/lib/libavcodec.a(dxa.o): In function `decode_frame': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/dxa.c:243: undefined reference to `uncompress' //home/Pang/ffmpeg_build/lib/libavcodec.a(evrcdec.o): In function `frame_erasure': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/evrcdec.c:679: undefined reference to `pow' //home/Pang/ffmpeg_build/lib/libavcodec.a(evrcdec.o): In function `evrc_decode_frame': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/evrcdec.c:831: undefined reference to `pow' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/evrcdec.c:831: undefined reference to `pow' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/evrcdec.c:831: undefined reference to `pow' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/evrcdec.c:855: undefined reference to `exp' //home/Pang/ffmpeg_build/lib/libavcodec.a(evrcdec.o): In function `evrc_decode_init': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/evrcdec.c:259: undefined reference to `cos' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/evrcdec.c:260: undefined reference to `sin' //home/Pang/ffmpeg_build/lib/libavcodec.a(exr.o): In function `zip_uncompress': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/exr.c:225: undefined reference to `uncompress' //home/Pang/ffmpeg_build/lib/libavcodec.a(exr.o): In function `pxr24_uncompress': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/exr.c:289: undefined reference to `uncompress' //home/Pang/ffmpeg_build/lib/libavcodec.a(fft_float.o): In function `ff_init_ff_cos_tabs': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/fft.c:96: undefined reference to `cos' //home/Pang/ffmpeg_build/lib/libavcodec.a(ffv1enc.o): In function `sort_stt': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/ffv1enc.c:629: undefined reference to `log2' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/ffv1enc.c:629: undefined reference to `log2' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/ffv1enc.c:629: undefined reference to `log2' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/ffv1enc.c:629: undefined reference to `log2' //home/Pang/ffmpeg_build/lib/libavcodec.a(ffv1enc.o): In function `find_best_state': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/ffv1enc.c:145: undefined reference to `log2' //home/Pang/ffmpeg_build/lib/libavcodec.a(ffv1enc.o): In function `encode_init': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/ffv1enc.c:903: undefined reference to `round' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/ffv1enc.c:914: undefined reference to `round' //home/Pang/ffmpeg_build/lib/libavcodec.a(ffwavesynth.o): In function `wavesynth_init': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/ffwavesynth.c:338: undefined reference to `sin' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/ffwavesynth.c:338: undefined reference to `floor' //home/Pang/ffmpeg_build/lib/libavcodec.a(flashsv.o): In function `flashsv_decode_block': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv.c:173: undefined reference to `inflateReset' //home/Pang/ffmpeg_build/lib/libavcodec.a(flashsv.o): In function `flashsv2_prime': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv.c:140: undefined reference to `inflate' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv.c:142: undefined reference to `deflateInit_' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv.c:148: undefined reference to `deflate' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv.c:149: undefined reference to `deflateEnd' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv.c:151: undefined reference to `inflateReset' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv.c:160: undefined reference to `inflate' //home/Pang/ffmpeg_build/lib/libavcodec.a(flashsv.o): In function `flashsv_decode_block': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv.c:189: undefined reference to `inflate' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv.c:192: undefined reference to `inflateSync' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv.c:193: undefined reference to `inflate' //home/Pang/ffmpeg_build/lib/libavcodec.a(flashsv.o): In function `calc_deflate_block_size': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv.c:233: undefined reference to `deflateInit_' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv.c:235: undefined reference to `deflateBound' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv.c:236: undefined reference to `deflateEnd' //home/Pang/ffmpeg_build/lib/libavcodec.a(flashsv.o): In function `flashsv_decode_end': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv.c:468: undefined reference to `inflateEnd' //home/Pang/ffmpeg_build/lib/libavcodec.a(flashsv.o): In function `flashsv_decode_init': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv.c:112: undefined reference to `inflateInit_' //home/Pang/ffmpeg_build/lib/libavcodec.a(flashsv2enc.o): In function `encode_zlib': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv2enc.c:355: undefined reference to `compress2' //home/Pang/ffmpeg_build/lib/libavcodec.a(flashsv2enc.o): In function `encode_zlibprime': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv2enc.c:367: undefined reference to `deflateInit_' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv2enc.c:376: undefined reference to `deflate' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv2enc.c:385: undefined reference to `deflate' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsv2enc.c:386: undefined reference to `deflateEnd' //home/Pang/ffmpeg_build/lib/libavcodec.a(flashsvenc.o): In function `encode_bitstream': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsvenc.c:170: undefined reference to `compress2' //home/Pang/ffmpeg_build/lib/libavcodec.a(flashsvenc.o): In function `flashsv_encode_end': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/flashsvenc.c:269: undefined reference to `deflateEnd' //home/Pang/ffmpeg_build/lib/libavcodec.a(g2meet.o): In function `kempf_decode_tile': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/g2meet.c:395: undefined reference to `uncompress' //home/Pang/ffmpeg_build/lib/libavcodec.a(imc.o): In function `freq2bark': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/imc.c:123: undefined reference to `atan' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/imc.c:123: undefined reference to `atan' //home/Pang/ffmpeg_build/lib/libavcodec.a(imc.o): In function `imc_decode_level_coefficients_raw': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/imc.c:414: undefined reference to `pow' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/imc.c:415: undefined reference to `log2f' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/imc.c:424: undefined reference to `powf' //home/Pang/ffmpeg_build/lib/libavcodec.a(imc.o): In function `imc_decode_level_coefficients': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/imc.c:360: undefined reference to `exp2' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/imc.c:361: undefined reference to `log2f' //home/Pang/ffmpeg_build/lib/libavcodec.a(imc.o): In function `bit_allocation': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/imc.c:459: undefined reference to `log2f' //home/Pang/ffmpeg_build/lib/libavcodec.a(imc.o): In function `iac_generate_tabs': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/imc.c:140: undefined reference to `pow' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/imc.c:141: undefined reference to `pow' //home/Pang/ffmpeg_build/lib/libavcodec.a(imc.o): In function `imc_decode_init': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/imc.c:206: undefined reference to `sincos' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/imc.c:207: undefined reference to `sincos' //home/Pang/ffmpeg_build/lib/libavcodec.a(lcldec.o): In function `zlib_decomp': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/lcldec.c:137: undefined reference to `inflateReset' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/lcldec.c:146: undefined reference to `inflate' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/lcldec.c:137: undefined reference to `inflateReset' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/lcldec.c:146: undefined reference to `inflate' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/lcldec.c:137: undefined reference to `inflateReset' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/lcldec.c:146: undefined reference to `inflate' //home/Pang/ffmpeg_build/lib/libavcodec.a(lcldec.o): In function `decode_end': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/lcldec.c:645: undefined reference to `inflateEnd' //home/Pang/ffmpeg_build/lib/libavcodec.a(lcldec.o): In function `decode_init': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/lcldec.c:621: undefined reference to `inflateInit_' //home/Pang/ffmpeg_build/lib/libavcodec.a(lclenc.o): In function `encode_frame': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/lclenc.c:80: undefined reference to `deflateBound' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/lclenc.c:90: undefined reference to `deflateReset' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/lclenc.c:101: undefined reference to `deflate' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/lclenc.c:107: undefined reference to `deflate' //home/Pang/ffmpeg_build/lib/libavcodec.a(lclenc.o): In function `encode_end': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/lclenc.c:177: undefined reference to `deflateEnd' //home/Pang/ffmpeg_build/lib/libavcodec.a(lclenc.o): In function `encode_init': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/lclenc.c:158: undefined reference to `deflateInit_' //home/Pang/ffmpeg_build/lib/libavcodec.a(libfdk-aacdec.o): In function `fdk_aac_decode_frame': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacdec.c:224: undefined reference to `aacDecoder_Fill' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacdec.c:246: undefined reference to `aacDecoder_DecodeFrame' //home/Pang/ffmpeg_build/lib/libavcodec.a(libfdk-aacdec.o): In function `get_stream_info': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacdec.c:63: undefined reference to `aacDecoder_GetStreamInfo' //home/Pang/ffmpeg_build/lib/libavcodec.a(libfdk-aacdec.o): In function `fdk_aac_decode_flush': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacdec.c:292: undefined reference to `aacDecoder_SetParam' //home/Pang/ffmpeg_build/lib/libavcodec.a(libfdk-aacdec.o): In function `fdk_aac_decode_close': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacdec.c:176: undefined reference to `aacDecoder_Close' //home/Pang/ffmpeg_build/lib/libavcodec.a(libfdk-aacdec.o): In function `fdk_aac_decode_init': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacdec.c:186: undefined reference to `aacDecoder_Open' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacdec.c:193: undefined reference to `aacDecoder_ConfigRaw' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacdec.c:201: undefined reference to `aacDecoder_SetParam' //home/Pang/ffmpeg_build/lib/libavcodec.a(libfdk-aacenc.o): In function `aac_encode_close': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacenc.c:99: undefined reference to `aacEncClose' //home/Pang/ffmpeg_build/lib/libavcodec.a(libfdk-aacenc.o): In function `aac_encode_frame': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacenc.c:345: undefined reference to `aacEncEncode' //home/Pang/ffmpeg_build/lib/libavcodec.a(libfdk-aacenc.o): In function `aac_encode_init': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacenc.c:116: undefined reference to `aacEncOpen' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacenc.c:125: undefined reference to `aacEncoder_SetParam' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacenc.c:132: undefined reference to `aacEncoder_SetParam' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacenc.c:140: undefined reference to `aacEncoder_SetParam' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacenc.c:160: undefined reference to `aacEncoder_SetParam' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacenc.c:167: undefined reference to `aacEncoder_SetParam' //home/Pang/ffmpeg_build/lib/libavcodec.a(libfdk-aacenc.o):/home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacenc.c:185: more undefined references to `aacEncoder_SetParam' follow //home/Pang/ffmpeg_build/lib/libavcodec.a(libfdk-aacenc.o): In function `aac_encode_init': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacenc.c:264: undefined reference to `aacEncEncode' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libfdk-aacenc.c:270: undefined reference to `aacEncInfo' //home/Pang/ffmpeg_build/lib/libavcodec.a(libmp3lame.o): In function `mp3lame_encode_frame': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libmp3lame.c:210: undefined reference to `lame_encode_flush' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libmp3lame.c:188: undefined reference to `lame_encode_buffer' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libmp3lame.c:191: undefined reference to `lame_encode_buffer_int' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libmp3lame.c:204: undefined reference to `lame_encode_buffer_float' //home/Pang/ffmpeg_build/lib/libavcodec.a(libmp3lame.o): In function `mp3lame_encode_close': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libmp3lame.c:88: undefined reference to `lame_close' //home/Pang/ffmpeg_build/lib/libavcodec.a(libmp3lame.o): In function `mp3lame_encode_init': /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libmp3lame.c:100: undefined reference to `lame_init' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libmp3lame.c:104: undefined reference to `lame_set_num_channels' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libmp3lame.c:105: undefined reference to `lame_set_mode' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libmp3lame.c:108: undefined reference to `lame_set_in_samplerate' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libmp3lame.c:109: undefined reference to `lame_set_out_samplerate' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libmp3lame.c:113: undefined reference to `lame_set_quality' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libmp3lame.c:115: undefined reference to `lame_set_quality' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libmp3lame.c:119: undefined reference to `lame_set_VBR' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libmp3lame.c:120: undefined reference to `lame_set_VBR_quality' /home/Pang/ffmpeg_sources/ffmpeg/libavcodec/libmp3lame.c:123: undefined reference to `lame_set_brate' and the error goes on and on.... ------------------------------------------------------------------------------------------------------- Running ffmpeg on terminal: ffmpeg version N-39064-gfa7e9f9 Copyright (c) 2000-2013 the FFmpeg developers built on Oct 12 2013 11:01:14 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-3) configuration: --prefix=/home/Pang/ffmpeg_build --extra-cflags=-I/home/Pang/ffmpeg_build/include --extra-ldflags=-L/home/Pang/ffmpeg_build/lib --bindir=/home/Pang/bin --extra-libs=-ldl --enable-gpl --enable-nonfree --enable-libfdk_aac --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 libavutil 52. 46.101 / 52. 46.101 libavcodec 55. 35.100 / 55. 35.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.101 / 3. 88.101 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Hyper fast Audio and Video encoder usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}... Use -h to get full help or, even better, run 'man ffmpeg' -------------------------------------------------------------------------------------------------------- I am not sure whether I have provided sufficient information. Really appreciate you guys for help me look into it ! =D Thank you Pang Jun Jie On Mon, Oct 14, 2013 at 7:35 AM, ? ? wrote: > > ? 2013-10-14???5:31?Carl Eugen Hoyos ??? > > > Jun Jie Pang gmail.com> writes: > > > >> I have installed ffmpeg on centOS according to > >> https://trac.ffmpeg.org/wiki/CentosCompilationGuide. > >> However, when I try to compile the code, it gives > >> me multiple " undefined reference to ....". > > > > Unfortunately, this is not enough information;-( > > What are you trying to compile, how and what > > output does the linker print? > > > > Junjie Pang, > > Please give us the enough information, > > > Steven > > From 690271929 at qq.com Mon Oct 14 03:06:30 2013 From: 690271929 at qq.com (=?gb18030?B?SmFja3k=?=) Date: Mon, 14 Oct 2013 09:06:30 +0800 Subject: [FFmpeg-user] Extract Audio from Video file using ffmpeg Message-ID: hi Carl : I use the command: ffmpeg-0 -i /data/test/media/2270059.m2ts -vn -acodec copy audio.m2ts to extract Audio from Video file,i get the info, [root at cdn 2269968]# ./ffmpeg-0 -i /data/test/media/2270059.m2ts -vn -acodec copy audio.m2ts ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers built on Jul 31 2013 10:34:48 with gcc 4.4.6 (GCC) 20110731 (Red Hat 4.4.6-3) configuration: --prefix=/data/test/jh/ffmpeg --extra-cflags=-I/data/test/jh/src/ffmpeg/include --extra-ldflags=-L/data/test/jh/src/ffmpeg/lib --enable-gpl --enable-version3 --enable-nonfree --enable-debug --enable-memalign-hack --enable-libfaac --enable-libaacplus --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid libavutil 52. 38.100 / 52. 38.100 libavcodec 55. 18.102 / 55. 18.102 libavformat 55. 12.100 / 55. 12.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 79.101 / 3. 79.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 H264 encode type: Frame @@type: 0 [mpegts @ 0x254a820] Stream #6: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x254a820] Stream #7: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x254a820] Stream #8: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x254a820] Stream #9: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x254a820] Stream #10: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x254a820] Stream #11: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x254a820] Stream #12: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x254a820] Stream #13: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0x254a820] Stream #14: not enough frames to estimate rate; consider increasing probesize [NULL @ 0x2551fa0] start time is not set in estimate_timings_from_pts [NULL @ 0x25529a0] start time is not set in estimate_timings_from_pts [NULL @ 0x256d4c0] start time is not set in estimate_timings_from_pts [NULL @ 0x256de80] start time is not set in estimate_timings_from_pts [NULL @ 0x256e780] start time is not set in estimate_timings_from_pts [NULL @ 0x256f1a0] start time is not set in estimate_timings_from_pts [NULL @ 0x256faa0] start time is not set in estimate_timings_from_pts [NULL @ 0x2570480] start time is not set in estimate_timings_from_pts [NULL @ 0x2570d60] start time is not set in estimate_timings_from_pts [mpegts @ 0x254a820] Could not find codec parameters for stream 6 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x254a820] Could not find codec parameters for stream 7 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x254a820] Could not find codec parameters for stream 8 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x254a820] Could not find codec parameters for stream 9 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x254a820] Could not find codec parameters for stream 10 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x254a820] Could not find codec parameters for stream 11 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x254a820] Could not find codec parameters for stream 12 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x254a820] Could not find codec parameters for stream 13 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [mpegts @ 0x254a820] Could not find codec parameters for stream 14 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, mpegts, from '/data/test/media/2270059.m2ts': Duration: 02:13:18.03, start: 11.608967, bitrate: 30389 kb/s Program 1 Stream #0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc Stream #0:1[0x1100]: Audio: truehd (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), s32 Stream #0:2[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s Stream #0:3[0x1101]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s Stream #0:4[0x1102]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s Stream #0:5[0x1103]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 192 kb/s Stream #0:6[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:7[0x1201]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:8[0x1202]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:9[0x1203]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:10[0x1204]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:11[0x1205]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:12[0x1206]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:13[0x1207]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) Stream #0:14[0x1208]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) [mpegts @ 0x25f2b20] frame size not set Output #0, mpegts, to 'audio.m2ts': Metadata: encoder : Lavf55.12.100 Stream #0:0: Audio: truehd (AC-3 / 0x332D4341), 48000 Hz, 5.1(side) Stream mapping: Stream #0:1 -> #0:0 (copy) Press [q] to stop, [?] for help size= 2828405kB time=02:13:17.98 bitrate=2897.0kbits/s video:0kB audio:2475415kB subtitle:0 global headers:0kB muxing overhead 14.259822% and i do this: [root at cdn 2269968]# ./ffmpeg-0 -i audio.m2ts i get info: ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers built on Jul 31 2013 10:34:48 with gcc 4.4.6 (GCC) 20110731 (Red Hat 4.4.6-3) configuration: --prefix=/data/test/jh/ffmpeg --extra-cflags=-I/data/test/jh/src/ffmpeg/include --extra-ldflags=-L/data/test/jh/src/ffmpeg/lib --enable-gpl --enable-version3 --enable-nonfree --enable-debug --enable-memalign-hack --enable-libfaac --enable-libaacplus --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid libavutil 52. 38.100 / 52. 38.100 libavcodec 55. 18.102 / 55. 18.102 libavformat 55. 12.100 / 55. 12.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 79.101 / 3. 79.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 [mpegts @ 0x2b95fe0] probed stream 0 failed [mpegts @ 0x2b95fe0] Could not find codec parameters for stream 0 (Unknown: none ([6][0][0][0] / 0x0006)): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options audio.m2ts: could not find codec parameters Muxing truehd in m2ts is not supported too? if it is, then how can the source file 2270059.m2ts is created? this wiki said m2ts and mpegts is support ac-3.http://en.wikipedia.org/wiki/Comparison_of_container_formats thank you very much. From 690271929 at qq.com Mon Oct 14 04:19:22 2013 From: 690271929 at qq.com (=?ISO-8859-1?B?SmFja3k=?=) Date: Mon, 14 Oct 2013 10:19:22 +0800 Subject: [FFmpeg-user] ffmpeg get video from video file of avi Message-ID: hi: i use the command:" ffmpeg -t 20 -i /data/test/media/1789035.avi -an -vcodec copy video.avi " to get the video data. i get the ffmpeg info is: [root at cdn 2269968]# ./ffmpeg -i audio.avi -i video.avi -c copy av.avi ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers built on Jul 31 2013 10:34:48 with gcc 4.4.6 (GCC) 20110731 (Red Hat 4.4.6-3) configuration: --prefix=/data/test/jh/ffmpeg --extra-cflags=-I/data/test/jh/src/ffmpeg/include --extra-ldflags=-L/data/test/jh/src/ffmpeg/lib --enable-gpl --enable-version3 --enable-nonfree --enable-debug --enable-memalign-hack --enable-libfaac --enable-libaacplus --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid libavutil 52. 38.100 / 52. 38.100 libavcodec 55. 18.102 / 55. 18.102 libavformat 55. 12.100 / 55. 12.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 79.101 / 3. 79.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 Input #0, avi, from 'audio.avi': Metadata: encoder : Lavf55.12.100 Duration: 01:49:24.68, start: 0.000000, bitrate: 264 kb/s Stream #0:0: Audio: aac ([255][0][0][0] / 0x00FF), 44100 Hz, stereo, fltp, 256 kb/s Input #1, avi, from 'video.avi': Metadata: encoder : Lavf55.12.100 Duration: 00:00:20.04, start: 0.000000, bitrate: 17246 kb/s Stream #1:0: Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1920x1080, 24 fps, 24 tbr, 24 tbn, 48 tbc Output #0, avi, to 'av.avi': Metadata: ISFT : Lavf55.12.100 Stream #0:0: Video: h264 (avc1 / 0x31637661), yuv420p, 1920x1080, q=2-31, 24 fps, 24 tbn, 24 tbc Stream #0:1: Audio: aac ([255][0][0][0] / 0x00FF), 44100 Hz, stereo, 256 kb/s Stream mapping: Stream #1:0 -> #0:0 (copy) Stream #0:0 -> #0:1 (copy) Press [q] to stop, [?] for help frame= 481 fps= 92 q=-1.0 Lsize= 254107kB time=01:49:24.67 bitrate= 317.1kbits/s video:42175kB audio:205146kB subtitle:0 global headers:0kB muxing overhead 2.743437% and i test the video.avi use the command: /ffmpeg -i video.avi i get: ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers built on Jul 31 2013 10:34:48 with gcc 4.4.6 (GCC) 20110731 (Red Hat 4.4.6-3) configuration: --prefix=/data/test/jh/ffmpeg --extra-cflags=-I/data/test/jh/src/ffmpeg/include --extra-ldflags=-L/data/test/jh/src/ffmpeg/lib --enable-gpl --enable-version3 --enable-nonfree --enable-debug --enable-memalign-hack --enable-libfaac --enable-libaacplus --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid libavutil 52. 38.100 / 52. 38.100 libavcodec 55. 18.102 / 55. 18.102 libavformat 55. 12.100 / 55. 12.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 79.101 / 3. 79.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 Input #0, avi, from 'video.avi': Metadata: encoder : Lavf55.12.100 Duration: 00:00:20.04, start: 0.000000, bitrate: 17246 kb/s Stream #0:0: Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1920x1080, 24 fps, 24 tbr, 24 tbn, 48 tbc At least one output file must be specified and i use mediainfo : mediainfo video.avi General Complete name : video.avi Format : AVI Format/Info : Audio Video Interleave File size : 41.2 MiB Duration : 20s 42ms Overall bit rate : 17.2 Mbps Writing application : Lavf55.12.100 Video ID : 0 Format : AVC Format/Info : Advanced Video Codec Format profile : Main at L4.1 Format settings, CABAC : No Format settings, ReFrames : 2 frames Codec ID : avc1 Duration : 20s 42ms Bit rate : 17.2 Mbps Width : 1 920 pixels Height : 1 080 pixels Display aspect ratio : 16:9 Frame rate : 24.000 fps Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Progressive Bits/(Pixel*Frame) : 0.346 Stream size : 41.2 MiB (100%) Color primaries : BT.709 Transfer characteristics : BT.709 Matrix coefficients : BT.709 it seem no problem of this video.avi, but when i play this file, it has no picture. i try more than one player, but no one can play it. thank you very much. From rieger at ari.uni-heidelberg.de Mon Oct 14 22:33:29 2013 From: rieger at ari.uni-heidelberg.de (rieger at ari.uni-heidelberg.de) Date: Mon, 14 Oct 2013 22:33:29 +0200 Subject: [FFmpeg-user] Following UbuntuCompilationGuide running into trouble with the x264 part In-Reply-To: <20131013134908.742a7641@lrcd.com> References: <20131013001615.t8vhlkbk004480o0@wwwmail.urz.uni-heidelberg.de> <20131013134551.71422353@lrcd.com> <20131013134908.742a7641@lrcd.com> Message-ID: <20131014223329.6xx14r3iaogcw8kk@wwwmail.urz.uni-heidelberg.de> Hi Lou, >> Check to see if you're trying to to use ffmpeg (for lavfi support in >> x264) that is already linked to another installed version of x264. > > s/lavfi/lavf (aka libavformat) you made my day: - I have libavformat.so.53 installed - I found the following files, too: - /usr/lib/liblavfile-1.9.so.0 - /usr/lib/liblavfile-1.9.so.0.0.0 Because I don't want deinstall libavformat (used by many other packages) I have tried to use the option "-disable-lavf": cd ~/ffmpeg_sources cd x264 ./configure --prefix="$HOME/ffmpeg_build" --bindir="$HOME/bin" --enable-static --disable-lavf make make install make distclean It works without error! Thanks for the help to find a solution! Claus From cehoyos at ag.or.at Tue Oct 15 01:27:35 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 14 Oct 2013 23:27:35 +0000 (UTC) Subject: [FFmpeg-user] Qns regarding compiling c++ code with ffmpeg References: <5F5A0287-51D7-457F-8458-66AFE74E0186@gmail.com> Message-ID: Jun Jie Pang gmail.com> writes: > undefined reference to `truncf' -lm > undefined reference to `uncompress' -lz For libfdk-aac and libmp3lame you will have to link the appropriate libraries. Carl Eugen From lingjiujianke at gmail.com Tue Oct 15 01:50:03 2013 From: lingjiujianke at gmail.com (=?gb18030?B?wfUg4ao=?=) Date: Tue, 15 Oct 2013 07:50:03 +0800 Subject: [FFmpeg-user] Extract Audio from Video file using ffmpeg In-Reply-To: References: Message-ID: <3E3402C3-0EC4-46F6-B8A6-65355C86B10E@gmail.com> ? 2013-10-14???9:06?"Jacky" <690271929 at qq.com> ??? > hi Carl : > I use the command: > ffmpeg-0 -i /data/test/media/2270059.m2ts -vn -acodec copy audio.m2ts to extract Audio from Video file,i get the info, > > > [root at cdn 2269968]# ./ffmpeg-0 -i /data/test/media/2270059.m2ts -vn -acodec copy audio.m2ts > ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers > built on Jul 31 2013 10:34:48 with gcc 4.4.6 (GCC) 20110731 (Red Hat 4.4.6-3) > configuration: --prefix=/data/test/jh/ffmpeg --extra-cflags=-I/data/test/jh/src/ffmpeg/include --extra-ldflags=-L/data/test/jh/src/ffmpeg/lib --enable-gpl --enable-version3 --enable-nonfree --enable-debug --enable-memalign-hack --enable-libfaac --enable-libaacplus --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid > libavutil 52. 38.100 / 52. 38.100 > libavcodec 55. 18.102 / 55. 18.102 > libavformat 55. 12.100 / 55. 12.100 > libavdevice 55. 3.100 / 55. 3.100 > libavfilter 3. 79.101 / 3. 79.101 > libswscale 2. 3.100 / 2. 3.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 3.100 / 52. 3.100 > H264 encode type: Frame > @@type: 0 > [mpegts @ 0x254a820] Stream #6: not enough frames to estimate rate; consider increasing probesize > [mpegts @ 0x254a820] Stream #7: not enough frames to estimate rate; consider increasing probesize > [mpegts @ 0x254a820] Stream #8: not enough frames to estimate rate; consider increasing probesize > [mpegts @ 0x254a820] Stream #9: not enough frames to estimate rate; consider increasing probesize > [mpegts @ 0x254a820] Stream #10: not enough frames to estimate rate; consider increasing probesize > [mpegts @ 0x254a820] Stream #11: not enough frames to estimate rate; consider increasing probesize > [mpegts @ 0x254a820] Stream #12: not enough frames to estimate rate; consider increasing probesize > [mpegts @ 0x254a820] Stream #13: not enough frames to estimate rate; consider increasing probesize > [mpegts @ 0x254a820] Stream #14: not enough frames to estimate rate; consider increasing probesize > [NULL @ 0x2551fa0] start time is not set in estimate_timings_from_pts > [NULL @ 0x25529a0] start time is not set in estimate_timings_from_pts > [NULL @ 0x256d4c0] start time is not set in estimate_timings_from_pts > [NULL @ 0x256de80] start time is not set in estimate_timings_from_pts > [NULL @ 0x256e780] start time is not set in estimate_timings_from_pts > [NULL @ 0x256f1a0] start time is not set in estimate_timings_from_pts > [NULL @ 0x256faa0] start time is not set in estimate_timings_from_pts > [NULL @ 0x2570480] start time is not set in estimate_timings_from_pts > [NULL @ 0x2570d60] start time is not set in estimate_timings_from_pts > [mpegts @ 0x254a820] Could not find codec parameters for stream 6 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' options > [mpegts @ 0x254a820] Could not find codec parameters for stream 7 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' options > [mpegts @ 0x254a820] Could not find codec parameters for stream 8 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' options > [mpegts @ 0x254a820] Could not find codec parameters for stream 9 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' options > [mpegts @ 0x254a820] Could not find codec parameters for stream 10 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' options > [mpegts @ 0x254a820] Could not find codec parameters for stream 11 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' options > [mpegts @ 0x254a820] Could not find codec parameters for stream 12 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' options > [mpegts @ 0x254a820] Could not find codec parameters for stream 13 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' options > [mpegts @ 0x254a820] Could not find codec parameters for stream 14 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' options > Input #0, mpegts, from '/data/test/media/2270059.m2ts': > Duration: 02:13:18.03, start: 11.608967, bitrate: 30389 kb/s > Program 1 > Stream #0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc > Stream #0:1[0x1100]: Audio: truehd (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), s32 > Stream #0:2[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s > Stream #0:3[0x1101]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s > Stream #0:4[0x1102]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s > Stream #0:5[0x1103]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 192 kb/s > Stream #0:6[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) > Stream #0:7[0x1201]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) > Stream #0:8[0x1202]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) > Stream #0:9[0x1203]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) > Stream #0:10[0x1204]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) > Stream #0:11[0x1205]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) > Stream #0:12[0x1206]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) > Stream #0:13[0x1207]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) > Stream #0:14[0x1208]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) > [mpegts @ 0x25f2b20] frame size not set > Output #0, mpegts, to 'audio.m2ts': > Metadata: > encoder : Lavf55.12.100 > Stream #0:0: Audio: truehd (AC-3 / 0x332D4341), 48000 Hz, 5.1(side) > Stream mapping: > Stream #0:1 -> #0:0 (copy) > Press [q] to stop, [?] for help > size= 2828405kB time=02:13:17.98 bitrate=2897.0kbits/s > video:0kB audio:2475415kB subtitle:0 global headers:0kB muxing overhead 14.259822% > > > and i do this: > [root at cdn 2269968]# ./ffmpeg-0 -i audio.m2ts > > i get info: > ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers > built on Jul 31 2013 10:34:48 with gcc 4.4.6 (GCC) 20110731 (Red Hat 4.4.6-3) > configuration: --prefix=/data/test/jh/ffmpeg --extra-cflags=-I/data/test/jh/src/ffmpeg/include --extra-ldflags=-L/data/test/jh/src/ffmpeg/lib --enable-gpl --enable-version3 --enable-nonfree --enable-debug --enable-memalign-hack --enable-libfaac --enable-libaacplus --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid > libavutil 52. 38.100 / 52. 38.100 > libavcodec 55. 18.102 / 55. 18.102 > libavformat 55. 12.100 / 55. 12.100 > libavdevice 55. 3.100 / 55. 3.100 > libavfilter 3. 79.101 / 3. 79.101 > libswscale 2. 3.100 / 2. 3.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 3.100 / 52. 3.100 > [mpegts @ 0x2b95fe0] probed stream 0 failed > [mpegts @ 0x2b95fe0] Could not find codec parameters for stream 0 (Unknown: none ([6][0][0][0] / 0x0006)): unknown codec > Consider increasing the value for the 'analyzeduration' and 'probesize' options > audio.m2ts: could not find codec parameters > > > > > > Muxing truehd in m2ts is not supported too? if it is, then how can the source file 2270059.m2ts is created? this wiki said m2ts and mpegts is support ac-3.http://en.wikipedia.org/wiki/Comparison_of_container_formats Jacky, Please upload your test file to ftp://upload.ffmpeg.org/incoming/ From cehoyos at ag.or.at Tue Oct 15 01:59:45 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 14 Oct 2013 23:59:45 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg get video from video file of avi References: Message-ID: Jacky <690271929 qq.com> writes: > it seem no problem of this video.avi, but when i > play this file, it has no picture. i try more than > one player, but no one can play it. It neither plays with MPlayer nor ffplay? Carl Eugen From cehoyos at ag.or.at Tue Oct 15 02:01:48 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 15 Oct 2013 00:01:48 +0000 (UTC) Subject: [FFmpeg-user] Extract Audio from Video file using ffmpeg References: Message-ID: Jacky <690271929 qq.com> writes: > Muxing truehd in m2ts is not supported too? Muxing TrueHD in transport streams is not supported by FFmpeg currently, patch welcome. You don't have to upload a sample, we have a few TrueHD files;-) Carl Eugen From cehoyos at ag.or.at Tue Oct 15 02:04:34 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 15 Oct 2013 00:04:34 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?how_can_I_set_limit=5Ffilesize_or_recordi?= =?utf-8?q?ng=5Ftime_for_the_specific_output_dynamically=3F?= References: Message-ID: Huang Soho gmail.com> writes: > > (Complete, uncut console output missing.) > > > > There is the segment muxer, see 4.15 in > > http://ffmpeg.org/ffmpeg-formats.html > > I do not know about how to show the output from > the example command line. Just paste whatever FFmpeg prints on the console into an email. > But the "segment" seems does not meet my requirement. > I would like to save multiple files that each file > hase fixed size. You wrote before: [quote] how can I set limit_filesize or recording_time [/quote] Setting an exact file size is not possible for real codecs, you can set the recording time with the segment muxer. Carl Eugen From soho123.2012 at gmail.com Tue Oct 15 05:30:47 2013 From: soho123.2012 at gmail.com (Huang Soho) Date: Tue, 15 Oct 2013 11:30:47 +0800 Subject: [FFmpeg-user] how can I set limit_filesize or recording_time for the specific output dynamically? In-Reply-To: References: Message-ID: > > But the "segment" seems does not meet my requirement. > > I would like to save multiple files that each file > > hase fixed size. > > You wrote before: > [quote] > how can I set limit_filesize or recording_time > [/quote] > Setting an exact file size is not possible for real > codecs, you can set the recording time with the > segment muxer. > > I try to use segment muxer, but there is a error when the command execue. the log : ffmpeg -sn -f video4linux2 -r 30 -s 1280x720 -input_format h264 -i /dev/video1 -vcodec copy http://localhost:8090/feed2.ffm -vcodec copy -f segment -segment_ time 3 -segment_format mpegts /var/stream%05d.ts ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers built on Oct 15 2013 11:13:34 with gcc 4.4.7 configuration: --enable-cross-compile --cross-prefix=sdk-linux- --arch=mips --target-os=linux --disable-doc --disable-htmlpages --disable-manpages --disable-podpages --disable-txtpages --disable-mips32r2 --enable-small --disable-ffprobe --disable-ffplay --disable-postproc --disable-runtime-cpudetect --disable-swscale-alpha --disable-mipsdspr1 --disable-mipsdspr2 --disable-mipsfpu --enable-small --prefix=/ffmpeg/romfs --bindir=/ffmpeg/romfs --disable-bsfs --disable-filters --enable-gpl --enable-libx264 --extra-cflags=-fPIC --enable-filter='aformat,aresample,anull,copy,format,fps,framestep,resample' --disable-encoders --enable-encoder='mjpeg,h264,libx264,libx264rgb,yuv4,pcm_s16le,pcm_s16be,pcm_mulaw,wmav2,wmav1' --disable-decoders --enable-decoder='mjpeg,h264,yuv4' --disable-hwaccels --disable-muxers --enable-muxer='ffm,asf,asf_stream,rtsp,mjpeg,h264,wav,mpegts,se libavutil 52. 18.100 / 52. 18.100 libavcodec 54. 92.100 / 54. 92.100 libavformat 54. 63.104 / 54. 63.104 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 42.103 / 3. 42.103 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 [video4linux2,v4l2 @ 0x709350] Estimating duration from bitrate, this may be inaccurate Input #0, video4linux2,v4l2, from '/dev/video1': Duration: N/A, start: 57.420000, bitrate: N/A Stream #0:0: Video: h264, yuv420p, 1280x720, -5 kb/s, 30 fps, 30 tbr, 1000k tbn, 2000k tbc Tue Oct 15 11:22:50 2013 0.0.0.0 - - [GET] "/feed2.ffm HTTP/1.1" 200 32847 Output #1, segment, to '/var/stream%05d.ts': Output file #1 does not contain any stream Tue Oct 15 11:22:50 2013 0.0.0.0 - - [POST] "/feed2.ffm HTTP/1.1" 200 0 # From 690271929 at qq.com Tue Oct 15 05:37:49 2013 From: 690271929 at qq.com (=?gb18030?B?SmFja3k=?=) Date: Tue, 15 Oct 2013 11:37:49 +0800 Subject: [FFmpeg-user] how can I set limit_filesize or recording_timefor the specific output dynamically? In-Reply-To: References: Message-ID: you need add -map 0?VideoTrack ------------------ Original ------------------ From: "Huang Soho";; Date: Tue, Oct 15, 2013 11:30 AM To: "FFmpeg user questions"; Subject: Re: [FFmpeg-user] how can I set limit_filesize or recording_timefor the specific output dynamically? > > But the "segment" seems does not meet my requirement. > > I would like to save multiple files that each file > > hase fixed size. > > You wrote before: > [quote] > how can I set limit_filesize or recording_time > [/quote] > Setting an exact file size is not possible for real > codecs, you can set the recording time with the > segment muxer. > > I try to use segment muxer, but there is a error when the command execue. the log : ffmpeg -sn -f video4linux2 -r 30 -s 1280x720 -input_format h264 -i /dev/video1 -vcodec copy http://localhost:8090/feed2.ffm -vcodec copy -f segment -segment_ time 3 -segment_format mpegts /var/stream%05d.ts ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers built on Oct 15 2013 11:13:34 with gcc 4.4.7 configuration: --enable-cross-compile --cross-prefix=sdk-linux- --arch=mips --target-os=linux --disable-doc --disable-htmlpages --disable-manpages --disable-podpages --disable-txtpages --disable-mips32r2 --enable-small --disable-ffprobe --disable-ffplay --disable-postproc --disable-runtime-cpudetect --disable-swscale-alpha --disable-mipsdspr1 --disable-mipsdspr2 --disable-mipsfpu --enable-small --prefix=/ffmpeg/romfs --bindir=/ffmpeg/romfs --disable-bsfs --disable-filters --enable-gpl --enable-libx264 --extra-cflags=-fPIC --enable-filter='aformat,aresample,anull,copy,format,fps,framestep,resample' --disable-encoders --enable-encoder='mjpeg,h264,libx264,libx264rgb,yuv4,pcm_s16le,pcm_s16be,pcm_mulaw,wmav2,wmav1' --disable-decoders --enable-decoder='mjpeg,h264,yuv4' --disable-hwaccels --disable-muxers --enable-muxer='ffm,asf,asf_stream,rtsp,mjpeg,h264,wav,mpegts,se libavutil 52. 18.100 / 52. 18.100 libavcodec 54. 92.100 / 54. 92.100 libavformat 54. 63.104 / 54. 63.104 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 42.103 / 3. 42.103 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 [video4linux2,v4l2 @ 0x709350] Estimating duration from bitrate, this may be inaccurate Input #0, video4linux2,v4l2, from '/dev/video1': Duration: N/A, start: 57.420000, bitrate: N/A Stream #0:0: Video: h264, yuv420p, 1280x720, -5 kb/s, 30 fps, 30 tbr, 1000k tbn, 2000k tbc Tue Oct 15 11:22:50 2013 0.0.0.0 - - [GET] "/feed2.ffm HTTP/1.1" 200 32847 Output #1, segment, to '/var/stream%05d.ts': Output file #1 does not contain any stream Tue Oct 15 11:22:50 2013 0.0.0.0 - - [POST] "/feed2.ffm HTTP/1.1" 200 0 # _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user . From soho123.2012 at gmail.com Tue Oct 15 07:42:07 2013 From: soho123.2012 at gmail.com (Huang Soho) Date: Tue, 15 Oct 2013 13:42:07 +0800 Subject: [FFmpeg-user] how can I set limit_filesize or recording_timefor the specific output dynamically? In-Reply-To: References: Message-ID: 2013/10/15 Jacky <690271929 at qq.com> > you need add -map 0?VideoTrack > > hi yes ,that is right ! but the option "segment_time" seems not have any function. when I set -segment_time 3, ffmpeg does not split multiple files by each 3 seconds. the log : ffmpeg -sn -f video4linux2 -r 30 -s 1280x720 -input_format h264 -i /dev/video1 -vcodec copy http://localhost:8090/feed2.ffm -vcodec copy -map 0 -f segment -s egment_time 3 -segment_format mpegts /var/stream%05d.ts ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers built on Oct 15 2013 11:13:34 with gcc 4.4.7 configuration: --enable-cross-compile --cross-prefix=sdk-linux- --arch=mips --target-os=linux --disable-doc --disable-htmlpages --disable-manpages --disable-podpages --disable-txtpages --disable-mips32r2 --enable-small --disable-ffprobe --disable-ffplay --disable-postproc --disable-runtime-cpudetect --disable-swscale-alpha --disable-mipsdspr1 --disable-mipsdspr2 --disable-mipsfpu --enable-small --prefix=/ffmpeg/romfs --bindir=/ffmpeg/romfs --disable-bsfs --disable-filters --enable-gpl --enable-libx264 --extra-cflags=-fPIC --enable-filter='aformat,aresample,anull,copy,format,fps,framestep,resample' --disable-encoders --enable-encoder='mjpeg,h264,libx264,libx264rgb,yuv4,pcm_s16le,pcm_s16be,pcm_mulaw,wmav2,wmav1' --disable-decoders --enable-decoder='mjpeg,h264,yuv4' --disable-hwaccels --disable-muxers --enable-muxer='ffm,asf,asf_stream,rtsp,mjpeg,h264,wav,mpegts,se libavutil 52. 18.100 / 52. 18.100 libavcodec 54. 92.100 / 54. 92.100 libavformat 54. 63.104 / 54. 63.104 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 42.103 / 3. 42.103 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 [video4linux2,v4l2 @ 0x709350] Estimating duration from bitrate, this may be inaccurate Input #0, video4linux2,v4l2, from '/dev/video1': Duration: N/A, start: 374.000000, bitrate: N/A Stream #0:0: Video: h264, yuv420p, 1280x720, -5 kb/s, 30 fps, 30 tbr, 1000k tbn, 2000k tbc Tue Oct 15 11:28:07 2013 0.0.0.0 - - [GET] "/feed2.ffm HTTP/1.1" 200 32847 Output #0, ffm, to 'http://localhost:8090/feed2.ffm': Metadata: creation_time : now encoder : Lavf54.63.104 Stream #0:0: Video: h264, yuv420p, 1280x720, q=2-31, -5 kb/s, 30 fps, 1000k tbn, 1000k tbc Output #1, segment, to '/var/stream%05d.ts': Metadata: encoder : Lavf54.63.104 Stream #1:0: Video: h264, yuv420p, 1280x720, q=2-31, -5 kb/s, 30 fps, 90k tbn, 1000k tbc Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:0 -> #1:0 (copy) Press [q] to stop, [?] for help # ls /var/ -al -rw-r--r-- 1 root root 7128396 Oct 15 11:28 stream00000.ts From mapandrei at gmail.com Tue Oct 15 13:55:55 2013 From: mapandrei at gmail.com (Andrei Petru Mura) Date: Tue, 15 Oct 2013 14:55:55 +0300 Subject: [FFmpeg-user] Convert avi to swf Message-ID: Hello, I'm very new to ffmpeg. Can anybody help me on converting an avi file to swf? Here is the command that I used. It works, but it shortens the swf file. ffmpeg -i input.avi -acodec copy -vcodec flv output.swf From mike at redtux.org.uk Tue Oct 15 14:16:10 2013 From: mike at redtux.org.uk (Mike Martin) Date: Tue, 15 Oct 2013 13:16:10 +0100 Subject: [FFmpeg-user] Changing DAR ? In-Reply-To: <5752.1381521535@server1.tristatelogic.com> References: <068B1B8693FF41FC93C7864BCE24927B@HPKANTOOR> <5752.1381521535@server1.tristatelogic.com> Message-ID: This is what I use (retaining original AR) for mp4 -c:v libx264 -preset fast -crf 22 -g 20 -c:a aac -b:a 32000 -threads 2 -af volume=2.5 -vf scale="352:trunc(ow/dar/2)*2",fps="fps= 20",setsar="1/1" -strict -2 -y It also fixes SAR to be square opts are libx264 crf - values between 18-25 for good quality scale - I resize to 352 width and adjust height to fit AR fps - 20 audio - aac adjusted to 2.5 times output with bitrate of 32k -g - sets GOP size to 1 second hope this helps On 11 October 2013 20:58, Ronald F. Guilmette wrote: > > In message <068B1B8693FF41FC93C7864BCE24927B at HPKANTOOR>, > "Bouke \(VideoToolShed\)" wrote: > > > > >----- Original Message ----- > >From: "Ronald F. Guilmette" > >> > >> So, just to be sure now, please confirm that it _is_ indeed the case > that > >> "-vcodec copy" is incompatible with the "-vf " option, yes? > > > >Perhaps i'm missing the point, but -vcodec copy -aspect 16:9 does work > fine. > >No need for -vf if you just want to change DAR. > > I agree that that combination _does_ in fact work for some subset of > all of the video formats that ffmpeg handles... although it appeared > to me that it does not work for all. I may be wrong about that, but > at this moment I don't think so. I will need to do more experiments, > I guess, and report hard facts and findings. > > As regards to the point which you may or may not be missing, as I said, > as of 2.0.1, ffmpeg appears to allow both -vcodec copy and also -vf options > to be used together, and it just silently ignores one of those, which in > my opinion is rather entirely sub-optimal. > > Also, I believe that there may be cases where using -vf and setsar or > setdar may cause unexpected effects, specifically on the aspect ratio > that is _not_ being changed. But I have to get in and dig around on the > mediainfo sources and try to find out if some of the things it is telling > me may be slightly less than true, at least for some video formats. If > so, then it is really mediainfo's fault for printing misleading output. > > > Regards, > rfg > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From mike at redtux.org.uk Tue Oct 15 14:32:54 2013 From: mike at redtux.org.uk (Mike Martin) Date: Tue, 15 Oct 2013 13:32:54 +0100 Subject: [FFmpeg-user] split audio/video and merge later gets async In-Reply-To: <20130902154108.1f354e9e@archthink.fritz.box> References: <20130901144031.746d794d@archthink.fritz.box> <20130902154108.1f354e9e@archthink.fritz.box> Message-ID: Someone can correct me, but at least with async (deprecated I know) you have to place it before input ie ffmpeg -async 24 -i On 2 September 2013 14:41, Frank Tetzel wrote: > > > Even if i don't process the audio file (except convert to wav). > > > Just splitting and merging gets out of sync. > > > $ ffmpeg -i grab.mkv -vn -c:a pcm_s16le grab-aud.wav > > > $ ffmpeg -i grab.mkv -i grab-aud.wav -map 0:0 -map 1:0 -c copy > > > merge.mkv > > > > Complete, uncut console output missing. > > See end of message. > > > Audio in mkv probably has a delay which cannot be saved in wav, > > I don't think anything except finding out the delay and using > > it when remuxing (itsoffset) is possible. > > The delay isn't constant. The gap seems to get bigger over time, audio > before video. > > Besides, how do i find out the delay? ffprobe (-show_streams) doesn't > help me here (see end of message). > > > Which audio filter is missing in FFmpeg? > > A filter for noise reduction would be handy. sox noisered does a good > job, similar to audacity. > > Regards, > Frank. > > > (I rebuilt ffmpeg recently to enable libfdk-aac and libaacplus. The > recording was done without it. The problem was already there.) > > $ ffmpeg -i grab.mkv -vn -c:a pcm_s16le grab-aud.wav > ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers > built on Sep 1 2013 19:33:37 with gcc 4.8.1 (GCC) 20130725 (prerelease) > configuration: --prefix=/usr --disable-debug --disable-static > --enable-avresample --enable-dxva2 --enable-fontconfig --enable-gpl > --enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm > --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb > --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus > --enable-libpulse --enable-librtmp --enable-libschroedinger > --enable-libspeex --enable-libtheora --enable-libv4l2 --enable-libvorbis > --enable-libvpx --enable-libx264 --enable-libxvid --enable-pic > --enable-postproc --enable-runtime-cpudetect --enable-shared > --enable-swresample --enable-vdpau --enable-version3 --enable-x11grab > --enable-libaacplus --enable-libfdk-aac --enable-nonfree > libavutil 52. 38.100 / 52. 38.100 > libavcodec 55. 18.102 / 55. 18.102 > libavformat 55. 12.100 / 55. 12.100 > libavdevice 55. 3.100 / 55. 3.100 > libavfilter 3. 79.101 / 3. 79.101 > libavresample 1. 1. 0 / 1. 1. 0 > libswscale 2. 3.100 / 2. 3.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 3.100 / 52. 3.100 > Input #0, matroska,webm, from 'grab.mkv': > Metadata: > ENCODER : Lavf55.12.100 > Duration: 01:05:21.04, start: 0.000000, bitrate: 754 kb/s > Stream #0:0: Video: h264 (High 4:4:4 Predictive), yuv444p, 798x438, > SAR 1:1 DAR 133:73, 25 fps, 25 tbr, 1k tbn, 50 tbc (default) > Stream #0:1: Audio: flac, 44100 Hz, mono, s16 (default) > Output #0, wav, to 'grab-aud.wav': > Metadata: > ISFT : Lavf55.12.100 > Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, > s16, 705 kb/s (default) > Stream mapping: > Stream #0:1 -> #0:0 (flac -> pcm_s16le) > Press [q] to stop, [?] for help > size= 337437kB time=01:05:20.94 bitrate= 705.0kbits/s > video:0kB audio:337437kB subtitle:0 global headers:0kB muxing overhead > 0.000023% > > $ ffmpeg -i grab.mkv -i grab-aud.wav -map 0:0,0:0 -map 1:0,0:0 -c copy > merge.mkv > ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers > built on Sep 1 2013 19:33:37 with gcc 4.8.1 (GCC) 20130725 (prerelease) > configuration: --prefix=/usr --disable-debug --disable-static > --enable-avresample --enable-dxva2 --enable-fontconfig --enable-gpl > --enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm > --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb > --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus > --enable-libpulse --enable-librtmp --enable-libschroedinger > --enable-libspeex --enable-libtheora --enable-libv4l2 --enable-libvorbis > --enable-libvpx --enable-libx264 --enable-libxvid --enable-pic > --enable-postproc --enable-runtime-cpudetect --enable-shared > --enable-swresample --enable-vdpau --enable-version3 --enable-x11grab > --enable-libaacplus --enable-libfdk-aac --enable-nonfree > libavutil 52. 38.100 / 52. 38.100 > libavcodec 55. 18.102 / 55. 18.102 > libavformat 55. 12.100 / 55. 12.100 > libavdevice 55. 3.100 / 55. 3.100 > libavfilter 3. 79.101 / 3. 79.101 > libavresample 1. 1. 0 / 1. 1. 0 > libswscale 2. 3.100 / 2. 3.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 3.100 / 52. 3.100 > Input #0, matroska,webm, from 'grab.mkv': > Metadata: > ENCODER : Lavf55.12.100 > Duration: 01:05:21.04, start: 0.000000, bitrate: 754 kb/s > Stream #0:0: Video: h264 (High 4:4:4 Predictive), yuv444p, 798x438, > SAR 1:1 DAR 133:73, 25 fps, 25 tbr, 1k tbn, 50 tbc (default) > Stream #0:1: Audio: flac, 44100 Hz, mono, s16 (default) > Guessed Channel Layout for Input Stream #1.0 : mono > Input #1, wav, from 'grab-aud.wav': > Metadata: > encoder : Lavf55.12.100 > Duration: 01:05:17.64, bitrate: 705 kb/s > Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, > s16, 705 kb/s > Output #0, matroska, to 'merge.mkv': > Metadata: > encoder : Lavf55.12.100 > Stream #0:0: Video: h264 (H264 / 0x34363248), yuv444p, 798x438 [SAR > 1:1 DAR 133:73], q=2-31, 25 fps, 1k tbn, 1k tbc (default) > Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, > 705 kb/s > Stream mapping: > Stream #0:0 -> #0:0 (copy) > Stream #1:0 -> #0:1 (copy) > Press [q] to stop, [?] for help > frame=98026 fps=52546 q=-1.0 Lsize= 508544kB time=01:05:21.00 > bitrate=1062.5kbits/s > video:169839kB audio:337437kB subtitle:0 global headers:0kB muxing > overhead 0.249911% > > $ ffprobe -show_streams grab.mkv > ffprobe version 2.0.1 Copyright (c) 2007-2013 the FFmpeg developers > built on Sep 1 2013 19:33:37 with gcc 4.8.1 (GCC) 20130725 (prerelease) > configuration: --prefix=/usr --disable-debug --disable-static > --enable-avresample --enable-dxva2 --enable-fontconfig --enable-gpl > --enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm > --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb > --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus > --enable-libpulse --enable-librtmp --enable-libschroedinger > --enable-libspeex --enable-libtheora --enable-libv4l2 --enable-libvorbis > --enable-libvpx --enable-libx264 --enable-libxvid --enable-pic > --enable-postproc --enable-runtime-cpudetect --enable-shared > --enable-swresample --enable-vdpau --enable-version3 --enable-x11grab > --enable-libaacplus --enable-libfdk-aac --enable-nonfree > libavutil 52. 38.100 / 52. 38.100 > libavcodec 55. 18.102 / 55. 18.102 > libavformat 55. 12.100 / 55. 12.100 > libavdevice 55. 3.100 / 55. 3.100 > libavfilter 3. 79.101 / 3. 79.101 > libavresample 1. 1. 0 / 1. 1. 0 > libswscale 2. 3.100 / 2. 3.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 3.100 / 52. 3.100 > Input #0, matroska,webm, from 'grab.mkv': > Metadata: > ENCODER : Lavf55.12.100 > Duration: 01:05:21.04, start: 0.000000, bitrate: 754 kb/s > Stream #0:0: Video: h264 (High 4:4:4 Predictive), yuv444p, 798x438, > SAR 1:1 DAR 133:73, 25 fps, 25 tbr, 1k tbn, 50 tbc (default) > Stream #0:1: Audio: flac, 44100 Hz, mono, s16 (default) > [STREAM] > index=0 > codec_name=h264 > codec_long_name=H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 > profile=High 4:4:4 Predictive > codec_type=video > codec_time_base=1/50 > codec_tag_string=[0][0][0][0] > codec_tag=0x0000 > width=798 > height=438 > has_b_frames=0 > sample_aspect_ratio=1:1 > display_aspect_ratio=133:73 > pix_fmt=yuv444p > level=30 > timecode=N/A > id=N/A > r_frame_rate=25/1 > avg_frame_rate=25/1 > time_base=1/1000 > start_pts=0 > start_time=0.000000 > duration_ts=N/A > duration=N/A > bit_rate=N/A > nb_frames=N/A > nb_read_frames=N/A > nb_read_packets=N/A > DISPOSITION:default=1 > DISPOSITION:dub=0 > DISPOSITION:original=0 > DISPOSITION:comment=0 > DISPOSITION:lyrics=0 > DISPOSITION:karaoke=0 > DISPOSITION:forced=0 > DISPOSITION:hearing_impaired=0 > DISPOSITION:visual_impaired=0 > DISPOSITION:clean_effects=0 > DISPOSITION:attached_pic=0 > [/STREAM] > [STREAM] > index=1 > codec_name=flac > codec_long_name=FLAC (Free Lossless Audio Codec) > profile=unknown > codec_type=audio > codec_time_base=1/44100 > codec_tag_string=[0][0][0][0] > codec_tag=0x0000 > sample_fmt=s16 > sample_rate=44100 > channels=1 > bits_per_sample=0 > id=N/A > r_frame_rate=0/0 > avg_frame_rate=0/0 > time_base=1/1000 > start_pts=0 > start_time=0.000000 > duration_ts=N/A > duration=N/A > bit_rate=N/A > nb_frames=N/A > nb_read_frames=N/A > nb_read_packets=N/A > DISPOSITION:default=1 > DISPOSITION:dub=0 > DISPOSITION:original=0 > DISPOSITION:comment=0 > DISPOSITION:lyrics=0 > DISPOSITION:karaoke=0 > DISPOSITION:forced=0 > DISPOSITION:hearing_impaired=0 > DISPOSITION:visual_impaired=0 > DISPOSITION:clean_effects=0 > DISPOSITION:attached_pic=0 > [/STREAM] > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From lingjiujianke at gmail.com Tue Oct 15 14:37:10 2013 From: lingjiujianke at gmail.com (=?GB2312?B?wfUg4ao=?=) Date: Tue, 15 Oct 2013 20:37:10 +0800 Subject: [FFmpeg-user] ffmpeg get video from video file of avi In-Reply-To: References: Message-ID: ? 2013-10-15???7:59?Carl Eugen Hoyos ??? > Jacky <690271929 qq.com> writes: > >> it seem no problem of this video.avi, but when i >> play this file, it has no picture. i try more than >> one player, but no one can play it. > > It neither plays with MPlayer nor ffplay? > Carl, Jacky's test file is here: http://blog.fs-linux.org/video.avi I have try to play and ffprobe show_packtes?but it just quit near at 00:06:00 Steven From lingjiujianke at gmail.com Tue Oct 15 14:42:58 2013 From: lingjiujianke at gmail.com (=?GB2312?B?wfUg4ao=?=) Date: Tue, 15 Oct 2013 20:42:58 +0800 Subject: [FFmpeg-user] how can I set limit_filesize or recording_timefor the specific output dynamically? In-Reply-To: References: Message-ID: <8F721E1F-442D-4034-9BC1-71082FE15587@gmail.com> perhaps you need make the keyframe at every 3 second. For example: ffmpeg -i input -force_key_frames 3,6,9,12,15,18,21,24 -vcodec libx264 -vprofile baseline -vlevel 1.0 -s 640x480 -b:v 800k -r 15 -pix_fmt yuv420p -acodec copy -strict -2 -f segment -segment_format mpegts -segment_time 3 -segment_list output.m3u8 -segment_list_flags live -map 0 -flags -global_header video-%d.ts here are split at 00:00:03, 00:00:06??..00:00:24 you can wite a program to make the time to limit, and duration on three seconds. Steven ? 2013-10-15???1:42?Huang Soho ??? > 2013/10/15 Jacky <690271929 at qq.com> > >> you need add -map 0?VideoTrack >> >> > > > hi > yes ,that is right ! > but the option "segment_time" seems not have any function. > when I set -segment_time 3, ffmpeg does not split multiple files by each 3 > seconds. > the log : > > ffmpeg -sn -f video4linux2 -r 30 -s 1280x720 -input_format h264 -i > /dev/video1 > -vcodec copy http://localhost:8090/feed2.ffm -vcodec copy -map 0 -f > segment -s > egment_time 3 -segment_format mpegts /var/stream%05d.ts > ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers > built on Oct 15 2013 11:13:34 with gcc 4.4.7 > configuration: --enable-cross-compile --cross-prefix=sdk-linux- > --arch=mips --target-os=linux --disable-doc --disable-htmlpages > --disable-manpages --disable-podpages --disable-txtpages --disable-mips32r2 > --enable-small --disable-ffprobe --disable-ffplay --disable-postproc > --disable-runtime-cpudetect --disable-swscale-alpha --disable-mipsdspr1 > --disable-mipsdspr2 --disable-mipsfpu --enable-small --prefix=/ffmpeg/romfs > --bindir=/ffmpeg/romfs --disable-bsfs --disable-filters --enable-gpl > --enable-libx264 --extra-cflags=-fPIC > --enable-filter='aformat,aresample,anull,copy,format,fps,framestep,resample' > --disable-encoders > --enable-encoder='mjpeg,h264,libx264,libx264rgb,yuv4,pcm_s16le,pcm_s16be,pcm_mulaw,wmav2,wmav1' > --disable-decoders --enable-decoder='mjpeg,h264,yuv4' --disable-hwaccels > --disable-muxers > --enable-muxer='ffm,asf,asf_stream,rtsp,mjpeg,h264,wav,mpegts,se > libavutil 52. 18.100 / 52. 18.100 > libavcodec 54. 92.100 / 54. 92.100 > libavformat 54. 63.104 / 54. 63.104 > libavdevice 54. 3.103 / 54. 3.103 > libavfilter 3. 42.103 / 3. 42.103 > libswscale 2. 2.100 / 2. 2.100 > libswresample 0. 17.102 / 0. 17.102 > [video4linux2,v4l2 @ 0x709350] Estimating duration from bitrate, this may > be inaccurate > Input #0, video4linux2,v4l2, from '/dev/video1': > Duration: N/A, start: 374.000000, bitrate: N/A > Stream #0:0: Video: h264, yuv420p, 1280x720, -5 kb/s, 30 fps, 30 tbr, > 1000k tbn, 2000k tbc > Tue Oct 15 11:28:07 2013 0.0.0.0 - - [GET] "/feed2.ffm HTTP/1.1" 200 32847 > Output #0, ffm, to 'http://localhost:8090/feed2.ffm': > Metadata: > creation_time : now > encoder : Lavf54.63.104 > Stream #0:0: Video: h264, yuv420p, 1280x720, q=2-31, -5 kb/s, 30 fps, > 1000k tbn, 1000k tbc > Output #1, segment, to '/var/stream%05d.ts': > Metadata: > encoder : Lavf54.63.104 > Stream #1:0: Video: h264, yuv420p, 1280x720, q=2-31, -5 kb/s, 30 fps, > 90k tbn, 1000k tbc > Stream mapping: > Stream #0:0 -> #0:0 (copy) > Stream #0:0 -> #1:0 (copy) > Press [q] to stop, [?] for help > > > # ls /var/ -al > -rw-r--r-- 1 root root 7128396 Oct 15 11:28 stream00000.ts > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From soho123.2012 at gmail.com Tue Oct 15 15:08:34 2013 From: soho123.2012 at gmail.com (Huang Soho) Date: Tue, 15 Oct 2013 21:08:34 +0800 Subject: [FFmpeg-user] how can I set limit_filesize or recording_timefor the specific output dynamically? In-Reply-To: <8F721E1F-442D-4034-9BC1-71082FE15587@gmail.com> References: <8F721E1F-442D-4034-9BC1-71082FE15587@gmail.com> Message-ID: perhaps you need make the keyframe at every 3 second. > It is impossible to set key frame since the video frame is capture from usb webcam and do stream copy. Is there any other idea ? it seems many limitation about "segment muxer" . why it can not used when capture from usb webcam and stream copy? From cehoyos at ag.or.at Tue Oct 15 15:56:22 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 15 Oct 2013 13:56:22 +0000 (UTC) Subject: [FFmpeg-user] Convert avi to swf References: Message-ID: Andrei Petru Mura gmail.com> writes: > It works, but it shortens the swf file. > > ffmpeg -i input.avi -acodec copy -vcodec flv output.swf Complete, uncut console output missing. Carl Eugen From cehoyos at ag.or.at Tue Oct 15 15:57:50 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 15 Oct 2013 13:57:50 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?how_can_I_set_limit=5Ffilesize_or_recordi?= =?utf-8?q?ng=5Ftime_for_the_specific_output_dynamically=3F?= References: Message-ID: Huang Soho gmail.com> writes: > ffmpeg version 1.2 Please test current git head. Carl Eugen From mapandrei at gmail.com Tue Oct 15 16:01:28 2013 From: mapandrei at gmail.com (Andrei Petru Mura) Date: Tue, 15 Oct 2013 17:01:28 +0300 Subject: [FFmpeg-user] Convert avi to swf In-Reply-To: References: Message-ID: On Tue, Oct 15, 2013 at 4:56 PM, Carl Eugen Hoyos wrote: > Andrei Petru Mura gmail.com> writes: > > > It works, but it shortens the swf file. > > > > ffmpeg -i input.avi -acodec copy -vcodec flv output.swf > > Complete, uncut console output missing. > /home/netnfork/bin/ffmpeg -i ./Event20131015123035001.avi -acodec copy -vcodec flv out.swf ffmpeg version git-2013-10-15-c35d29a Copyright (c) 2000-2013 the FFmpeg developers built on Oct 15 2013 11:56:38 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-3) configuration: --prefix=/home/netnfork/ffmpeg_build --extra-cflags=-I/home/netnfork/ffmpeg_build/include --extra-ldflags=-L/home/netnfork/ffmpeg_build/lib --bindir=/home/netnfork/bin --extra-libs=-ldl --enable-gpl --enable-nonfree --enable-libfdk_aac --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libfreetype --enable-libspeex --enable-libtheora libavutil 52. 46.101 / 52. 46.101 libavcodec 55. 36.100 / 55. 36.100 libavformat 55. 19.102 / 55. 19.102 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.101 / 3. 88.101 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 [avi @ 0xada5d60] probed stream 1 failed [avi @ 0xada5d60] Could not find codec parameters for stream 1 (Subtitle: none): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options [avi @ 0xada5d60] Could not find codec parameters for stream 2 (Subtitle: none): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, avi, from './Event20131015123035001.avi': Duration: 00:00:36.50, start: 0.000000, bitrate: 6799 kb/s Stream #0:0: Video: h264 (High) (X264 / 0x34363258), yuvj420p(pc), 1920x1080, 30 fps, 30 tbr, 30 tbn, 30 tbc Stream #0:1: Subtitle: none Stream #0:2: Subtitle: none [swscaler @ 0xad94020] deprecated pixel format used, make sure you did set range correctly Output #0, swf, to 'out.swf': Metadata: encoder : Lavf55.19.102 Stream #0:0: Video: flv1 (flv), yuv420p, 1920x1080, q=2-31, 200 kb/s, 90k tbn, 30 tbc Stream mapping: Stream #0:0 -> #0:0 (h264 -> flv) Press [q] to stop, [?] for help frame= 1095 fps= 37 q=31.0 Lsize= 44798kB time=00:00:36.50 bitrate=10054.3kbits/s video:44777kB audio:0kB subtitle:0 global headers:0kB muxing overhead 0.045488% > Carl Eugen > From cehoyos at ag.or.at Tue Oct 15 17:15:24 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 15 Oct 2013 15:15:24 +0000 (UTC) Subject: [FFmpeg-user] Convert avi to swf References: Message-ID: Andrei Petru Mura gmail.com> writes: > > > It works, but it shortens the swf file. > Input #0, avi, from './Event20131015123035001.avi': > Duration: 00:00:36.50, start: 0.000000, bitrate: 6799 kb/s > frame= 1095 fps= 37 q=31.0 Lsize= 44798kB time=00:00:36.50 It appears that the avi header reports a 36 seconds sample and that the encoded swf file is 36 seconds long. What is the problem with the output file? Consider providing the input sample if you believe there is a problem. Carl Eugen From cehoyos at ag.or.at Tue Oct 15 17:18:00 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 15 Oct 2013 15:18:00 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg get video from video file of avi References: Message-ID: ? ? gmail.com> writes: > Jacky's test file is here: > http://blog.fs-linux.org/video.avi > > I have try to play and ffprobe show_packtes?but it > just quit near at 00:06:00 It quits after ~6:30 because there is not more audio and video data in the file. Carl Eugen From soho123.2012 at gmail.com Tue Oct 15 18:14:55 2013 From: soho123.2012 at gmail.com (Huang Soho) Date: Wed, 16 Oct 2013 00:14:55 +0800 Subject: [FFmpeg-user] how can I set limit_filesize or recording_time for the specific output dynamically? In-Reply-To: References: Message-ID: > ffmpeg version 1.2 > > Please test current git head. > I have tested head version. I got the same result. "-segment_time 3" still does not function. From kabelbrand at gmail.com Tue Oct 15 18:20:44 2013 From: kabelbrand at gmail.com (Gunnar) Date: Tue, 15 Oct 2013 18:20:44 +0200 Subject: [FFmpeg-user] ffmpeg option/filter similar to AviSynth Limiter Message-ID: Hi list, is there a way to limit levels to TV (16-235) in ffmpeg similar to AviSynth's Limiter or Tweak(bright=0) functions. I tried -vf scale with in_range/out_range which wasn't recorgnized and the color_range option which seems to do nothing... I do not want to convert from PC levels to TV levels. I just want to enforce TV levels by removing values outside the TV range. Thank you. Best regards, Gunnar From jamesp at northernwholesale.com Tue Oct 15 18:50:34 2013 From: jamesp at northernwholesale.com (James Peterson) Date: Tue, 15 Oct 2013 16:50:34 +0000 Subject: [FFmpeg-user] Compiler error Message-ID: Hello friends, I m having difficulties installing ffmpeg from source. It worked fine with I just took all the defaults ./configure But then realized that I needed some more libraries included. Here is my script I was wondering if ou seen anything obviously wrong: ./configure --prefix="/usr/bin/ffmpeg" \ --extra-cflags="/usr/bin/ffmpeg/includes" --extra-ldflags="/usr/bin/ffmpeg/lib" --bindir="/usr/bin/ffmpeg/bin" \ --extra-libs="-lldl" \ --enable-gpl \ --enable-libass \ --enable-libfdk-aac \ --enable-libvpx \ --enable-libvorbis \ --enable-libtheora \ --enable-libx264 Here is my error: --------------------------------------------- gcc is unable to create an executable file. If gcc is a cross-compiler, use the --enable-cross-compile option. Only do this if you know what cross compiling means. 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. --------------------------------------------- I am running Fedora 14 Linux (x64) Gcc is not set to cross-compile And from what I can tell all my dependancies are up to dare Any advise you can offer would be greatly appreciated. Thank you in advance James From mathieu.multicam+ffmpeg at gmail.com Mon Oct 14 16:55:41 2013 From: mathieu.multicam+ffmpeg at gmail.com (MatCam) Date: Mon, 14 Oct 2013 07:55:41 -0700 (PDT) Subject: [FFmpeg-user] Generate Video from picture Message-ID: <1381762540967-4661811.post@n4.nabble.com> Hi I'm trying to generate a video from a picture. I'm Following FAQ : https://trac.ffmpeg.org/wiki/Create%20a%20video%20slideshow%20from% Unfortunately, I have an error. Here my cmd and logs : D:\test>ffmpeg.exe -loop 1 -i "D:\Records\2013-10-11_110403\PublishingDirectory\ImportedMedias\High Definition Wallpapers (14).jpg" -t 5.00 -pix_fmt yuv420p -y "D:\Records\2013-10-11_110403\Publishin gDirectory\Temporary1.mpeg" ffmpeg version N-55515-gbbbd959 Copyright (c) 2000-2013 the FFmpeg developers built on Aug 13 2013 18:06:32 with gcc 4.7.3 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libblu ray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-lib vpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 42.100 / 52. 42.100 libavcodec 55. 27.100 / 55. 27.100 libavformat 55. 13.102 / 55. 13.102 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 82.100 / 3. 82.100 libswscale 2. 4.100 / 2. 4.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, image2, from 'D:\Records\2013-10-11_110403\PublishingDirectory\ImportedMedias\High Definition Wallpapers (14).jpg': Duration: 00:00:00.04, start: 0.000000, bitrate: N/A Stream #0:0: Video: mjpeg, yuvj444p, 1920x1080 [SAR 300:300 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 25 tbc [swscaler @ 000000000252ba00] deprecated pixel format used, make sure you did set range correctly [mpeg @ 000000000268f5e0] VBV buffer size not set, muxing may fail Output #0, mpeg, to 'D:\Records\2013-10-11_110403\PublishingDirectory\Temporary1.mpeg': Metadata: encoder : Lavf55.13.102 Stream #0:0: Video: mpeg1video, yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0:0 -> #0:0 (mjpeg -> mpeg1video) Press [q] to stop, [?] for help [mpeg @ 000000000268f5e0] buffer underflow i=0 bufi=234442 size=424447 [mpeg @ 000000000268f5e0] packet too large, ignoring buffer limits to mux it [mpeg @ 000000000268f5e0] buffer underflow i=0 bufi=234442 size=424447 [mpeg @ 000000000268f5e0] buffer underflow i=0 bufi=236471 size=424447 [mpeg @ 000000000268f5e0] packet too large, ignoring buffer limits to mux it [mpeg @ 000000000268f5e0] buffer underflow i=0 bufi=236471 size=424447 [mpeg @ 000000000268f5e0] buffer underflow i=0 bufi=238512 size=424447 [mpeg @ 000000000268f5e0] packet too large, ignoring buffer limits to mux it [mpeg @ 000000000268f5e0] buffer underflow i=0 bufi=238512 size=424447 [mpeg @ 000000000268f5e0] buffer underflow i=0 bufi=240553 size=424447 [mpeg @ 000000000268f5e0] packet too large, ignoring buffer limits to mux it [mpeg @ 000000000268f5e0] buffer underflow i=0 bufi=240553 size=424447 [mpeg @ 000000000268f5e0] buffer underflow i=0 bufi=242594 size=424447 .... .... .... .... [mpeg @ 000000000268f5e0] buffer underflow i=0 bufi=424243 size=424447 [mpeg @ 000000000268f5e0] packet too large, ignoring buffer limits to mux itbits/s [mpeg @ 000000000268f5e0] buffer underflow i=0 bufi=424243 size=424447 frame= 125 fps= 37 q=31.0 Lsize= 1330kB time=00:00:04.96 bitrate=2196.6kbits/s video:1324kB audio:0kB subtitle:0 global headers:0kB muxing overhead 0.470587% When i try to read my video, Nothing happen. My picture infos : 635ko, 1920*1080 Thanks ! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Generate-Video-from-picture-tp4661811.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From dimytch at mail.ru Tue Oct 15 21:26:25 2013 From: dimytch at mail.ru (=?UTF-8?B?0JTQvNC40YLRgNC40Lkg0J/QtdC60LDRgNC+0LLRgdC60LjQuQ==?=) Date: Tue, 15 Oct 2013 22:26:25 +0300 Subject: [FFmpeg-user] setting timeout with rtsp connection Message-ID: <20131015222625.a8230840e3b04bf7ea8f8d3a@mail.ru> Hi, all I want to set a small timeout for testing my ipcams with ffprobe. I found here http://ffmpeg.org/ffprobe-all.html that -rtsp_flags with stimeout is what I need, but any variations of ffprobe -show_streams -rtsp_flags stimeout=100 'rtsp://ipcam' ffprobe -show_streams -rtsp_flags stimeout:100 'rtsp://ipcam' ffprobe -show_streams -rtsp_flags timeout=100 'rtsp://ipcam' etc shows only [RTSP demuxer @ 0x20aca80] [Eval @ 0x7fffef479130] Undefined constant or missing '(' in 'stimeout=100' [RTSP demuxer @ 0x20aca80] Unable to parse option value "stimeout=100" [RTSP demuxer @ 0x20aca80] Error setting option rtsp_flags to value stimeout=100. What is right syntax for timeouts? -- //DP From belcampo at zonnet.nl Tue Oct 15 22:12:48 2013 From: belcampo at zonnet.nl (Henk D. Schoneveld) Date: Tue, 15 Oct 2013 22:12:48 +0200 Subject: [FFmpeg-user] how can I set limit_filesize or recording_time for the specific output dynamically? In-Reply-To: References: Message-ID: AFAIK cuts can ONLY be done at I-Frame Keyframe points. A new part has to start with a I-frame. Henk On Oct 15, 2013, at 6:14 PM, Huang Soho wrote: >> ffmpeg version 1.2 >> >> Please test current git head. >> > > > I have tested head version. > I got the same result. > "-segment_time 3" still does not function. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From onemda at gmail.com Tue Oct 15 23:14:27 2013 From: onemda at gmail.com (Paul B Mahol) Date: Tue, 15 Oct 2013 21:14:27 +0000 Subject: [FFmpeg-user] ffmpeg option/filter similar to AviSynth Limiter In-Reply-To: References: Message-ID: On 10/15/13, Gunnar wrote: > Hi list, > > is there a way to limit levels to TV (16-235) in ffmpeg similar to > AviSynth's Limiter or Tweak(bright=0) functions. > I tried -vf scale with in_range/out_range which wasn't recorgnized and the > color_range option which seems to do nothing... > > I do not want to convert from PC levels to TV levels. I just want to > enforce TV levels by removing values outside the TV range. There is lut filter. > > Thank you. Best regards, > Gunnar > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From thierry.lelegard at free.fr Tue Oct 15 23:15:12 2013 From: thierry.lelegard at free.fr (=?ISO-8859-1?Q?Thierry_Lel=E9gard?=) Date: Tue, 15 Oct 2013 23:15:12 +0200 Subject: [FFmpeg-user] Weird characters in created passlogfile on Windows Message-ID: <525DB060.8060503@free.fr> Hello, I found a strange problem with ffmpeg on Windows (not tested yet on Linux) with the -passlogfile option. Consider a file path containing non ASCII characters like ? (e grave) used in all file specifications (in a directory name for instance). There are three different use cases: input file, output file, pass log file. In the first two cases, the non-ASCII characters in file names are processed correctly. But in the case of the pass log file, the file name is transformed into some weird sequence (typical UTF-8 sequence). The strange thing is that ffmpeg creates two files, the pass log file and the output file, but processes the characters in the file name quite differently. The way the two files are created must be completely different. See the following example: D:\Public\Videos\TEMP>ls -l h* -rw-rw-rw- 1 user group 18800000 Oct 15 22:50 h?.ts There is one single file. Let's transcode it that way; D:\Public\Videos\TEMP>ffmpeg -i h?.ts -pass 1 -passlogfile h? h?.mpg ffmpeg version N-56060-gbcd1c20 Copyright (c) 2000-2013 the FFmpeg developers built on Sep 6 2013 00:49:05 with gcc 4.7.3 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 43.100 / 52. 43.100 libavcodec 55. 31.101 / 55. 31.101 libavformat 55. 16.101 / 55. 16.101 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 83.104 / 3. 83.104 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 [....] Now look at the result: D:\Public\Videos\TEMP>ls -l h* -rw-rw-rw- 1 user group 258179 Oct 15 22:50 h??-0.log -rw-rw-rw- 1 user group 5261312 Oct 15 22:50 h?.mpg -rw-rw-rw- 1 user group 18800000 Oct 15 22:50 h?.ts The command contains "-passlogfile h? h?.mpg" but the "h?" in the two created files resulted in two different encodings. The problem is that the specified file name on the command line does NOT correspond to the actual file name in the file system. One may say that we don't care since this is a temporary log file. But there is a real problem when the non-ASCII character is in the name of the directory for all files. In that case, ffmpeg transforms the directory name and the resulting path is invalid since the transformed path uses a non-existent directory. Error resulting from option -passlogfile D:\Public\Videos\DVD\h?\fflog (assuming that the directory D:\Public\Videos\DVD\h? does exist and contains all files); Cannot write log file 'D:\Public\Videos\DVD\h?\fflog-0.log' for pass-1 encoding: No such file or directory Interestingly, the characters in the error message are correct. So, you do not understand why this failed. This is only after a test with an accent in the file name but none in the directory name that I discovered the transformation. Is this known? Shall I open a bug report for that? Best regards, -Thierry From lou at lrcd.com Tue Oct 15 23:31:21 2013 From: lou at lrcd.com (Lou) Date: Tue, 15 Oct 2013 13:31:21 -0800 Subject: [FFmpeg-user] Compiler error In-Reply-To: References: Message-ID: <20131015133121.5f4f42e8@lrcd.com> On Tue, 15 Oct 2013 16:50:34 +0000 James Peterson wrote: > --extra-libs="-lldl" \ Looks like a typo. Try: --extra-libs="-ldl" From lingjiujianke at gmail.com Wed Oct 16 01:31:24 2013 From: lingjiujianke at gmail.com (=?GB2312?B?wfUg4ao=?=) Date: Wed, 16 Oct 2013 07:31:24 +0800 Subject: [FFmpeg-user] how can I set limit_filesize or recording_time for the specific output dynamically? In-Reply-To: References: Message-ID: <3EE5B2C9-5A11-49DC-AC8F-DA714781CE68@gmail.com> ? 2013-10-16???12:14?Huang Soho ??? >> ffmpeg version 1.2 >> >> Please test current git head. >> > > > I have tested head version. > I got the same result. > "-segment_time 3" still does not function. Hey, Please try it , it can work when i used my camera [root at CM ~]# ffmpeg -f video4linux2 -s 1280x720 -i /dev/video0 -preset ultrafast -force_key_frames 3,6,9,12,15,18,21,24 -vcodec libx264 -vprofile baseline -vlevel 1.0 -s 640x480 -b:v 800k -r 15 -pix_fmt yuv420p -map 0:0 -f segment -segment_format mpegts -segment_time 3 -segment_list a.m3u8 -segment_list_flags live -map 0 -flags -global_header video-%d.ts ffmpeg version N-54918-g11cb697 Copyright (c) 2000-2013 the FFmpeg developers built on Jun 19 2013 08:54:07 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-3) configuration: --disable-yasm --enable-libx264 --enable-gpl libavutil 52. 40.100 / 52. 40.100 libavcodec 55. 19.100 / 55. 19.100 libavformat 55. 12.102 / 55. 12.102 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 81.102 / 3. 81.102 libswscale 2. 4.100 / 2. 4.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, video4linux2,v4l2, from '/dev/video0': Duration: N/A, start: 404985.663701, bitrate: 147456 kb/s Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 1280x720, 147456 kb/s, 10 fps, 10 tbr, 1000k tbn, 1000k tbc [libx264 @ 0x289c4a0] frame MB size (40x30) > level limit (99) [libx264 @ 0x289c4a0] DPB size (1 frames, 1200 mbs) > level limit (0 frames, 396 mbs) [libx264 @ 0x289c4a0] MB rate (18000) > level limit (1485) [libx264 @ 0x289c4a0] using cpu capabilities: none! [libx264 @ 0x289c4a0] profile Constrained Baseline, level 1.0 [libx264 @ 0x289d160] frame MB size (40x30) > level limit (99) [libx264 @ 0x289d160] DPB size (1 frames, 1200 mbs) > level limit (0 frames, 396 mbs) [libx264 @ 0x289d160] MB rate (18000) > level limit (1485) [libx264 @ 0x289d160] using cpu capabilities: none! [libx264 @ 0x289d160] profile Constrained Baseline, level 1.0 [segment @ 0x289ba00] Codec for stream 0 does not use global headers but container format requires global headers [segment @ 0x289ba00] Codec for stream 1 does not use global headers but container format requires global headers Output #0, segment, to 'video-%d.ts': Metadata: encoder : Lavf55.12.102 Stream #0:0: Video: h264 (libx264), yuv420p, 640x480, q=-1--1, 800 kb/s, 90k tbn, 15 tbc Stream #0:1: Video: h264 (libx264), yuv420p, 640x480, q=-1--1, 800 kb/s, 90k tbn, 15 tbc Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> libx264) Stream #0:0 -> #0:1 (rawvideo -> libx264) Press [q] to stop, [?] for help frame= 472 fps= 15 q=-1.0 Lq=-1.0 size=N/A time=00:00:31.46 bitrate=N/A dup=346 drop=0 video:6195kB audio:0kB subtitle:0 global headers:0kB muxing overhead -100.000347% [libx264 @ 0x289c4a0] frame I:9 Avg QP:24.56 size: 33423 [libx264 @ 0x289c4a0] frame P:463 Avg QP:27.58 size: 6201 [libx264 @ 0x289c4a0] mb I I16..4: 100.0% 0.0% 0.0% [libx264 @ 0x289c4a0] mb P I16..4: 12.1% 0.0% 0.0% P16..4: 62.3% 0.0% 0.0% 0.0% 0.0% skip:25.6% [libx264 @ 0x289c4a0] final ratefactor: 27.96 [libx264 @ 0x289c4a0] coded y,uvDC,uvAC intra: 66.3% 68.2% 40.5% inter: 31.0% 29.2% 3.2% [libx264 @ 0x289c4a0] i16 v,h,dc,p: 18% 21% 44% 17% [libx264 @ 0x289c4a0] i8c dc,h,v,p: 61% 19% 15% 4% [libx264 @ 0x289c4a0] kb/s:806.38 [libx264 @ 0x289d160] frame I:9 Avg QP:24.56 size: 33423 [libx264 @ 0x289d160] frame P:463 Avg QP:27.58 size: 6201 [libx264 @ 0x289d160] mb I I16..4: 100.0% 0.0% 0.0% [libx264 @ 0x289d160] mb P I16..4: 12.1% 0.0% 0.0% P16..4: 62.3% 0.0% 0.0% 0.0% 0.0% skip:25.6% [libx264 @ 0x289d160] final ratefactor: 27.96 [libx264 @ 0x289d160] coded y,uvDC,uvAC intra: 66.3% 68.2% 40.5% inter: 31.0% 29.2% 3.2% [libx264 @ 0x289d160] i16 v,h,dc,p: 18% 21% 44% 17% [libx264 @ 0x289d160] i8c dc,h,v,p: 61% 19% 15% 4% [libx264 @ 0x289d160] kb/s:806.38 [root at CM ~]# [root at CM ~]# ffmpeg -i video-0.ts -i video-1.ts -i video-2.ts -i video-3.ts -i video-4.ts -i video-5.ts ffmpeg version N-54918-g11cb697 Copyright (c) 2000-2013 the FFmpeg developers built on Jun 19 2013 08:54:07 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-3) configuration: --disable-yasm --enable-libx264 --enable-gpl libavutil 52. 40.100 / 52. 40.100 libavcodec 55. 19.100 / 55. 19.100 libavformat 55. 12.102 / 55. 12.102 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 81.102 / 3. 81.102 libswscale 2. 4.100 / 2. 4.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, mpegts, from 'video-0.ts': Duration: 00:00:02.93, start: 0.000000, bitrate: 2077 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 640x480, 15 fps, 15 tbr, 90k tbn, 30 tbc Stream #0:1[0x101]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 640x480, 15 fps, 15 tbr, 90k tbn, 30 tbc Input #1, mpegts, from 'video-1.ts': Duration: 00:00:02.93, start: 3.000000, bitrate: 1665 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #1:0[0x100]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 640x480, 15 fps, 15 tbr, 90k tbn, 30 tbc Stream #1:1[0x101]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 640x480, 15 fps, 15 tbr, 90k tbn, 30 tbc Input #2, mpegts, from 'video-2.ts': Duration: 00:00:02.93, start: 6.000000, bitrate: 1665 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #2:0[0x100]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 640x480, 15 fps, 15 tbr, 90k tbn, 30 tbc Stream #2:1[0x101]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 640x480, 15 fps, 15 tbr, 90k tbn, 30 tbc Input #3, mpegts, from 'video-3.ts': Duration: 00:00:02.93, start: 9.000000, bitrate: 1810 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #3:0[0x100]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 640x480, 15 fps, 15 tbr, 90k tbn, 30 tbc Stream #3:1[0x101]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 640x480, 15 fps, 15 tbr, 90k tbn, 30 tbc Input #4, mpegts, from 'video-4.ts': Duration: 00:00:02.93, start: 12.000000, bitrate: 1787 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #4:0[0x100]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 640x480, 15 fps, 15 tbr, 90k tbn, 30 tbc Stream #4:1[0x101]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 640x480, 15 fps, 15 tbr, 90k tbn, 30 tbc Input #5, mpegts, from 'video-5.ts': Duration: 00:00:02.93, start: 15.000000, bitrate: 1839 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #5:0[0x100]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 640x480, 15 fps, 15 tbr, 90k tbn, 30 tbc Stream #5:1[0x101]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 640x480, 15 fps, 15 tbr, 90k tbn, 30 tbc At least one output file must be specified you can write a bash or other scripts, to compute the keyframe point from 0 ~ unlimited, and set the key frame at every 3s for example: I want to cat 24hours: compute : #define LIMIT 20 * 60 * 60; int split_point = 0; char buffer[8]; char keyframe_point_set[n]; for (split_point = 0; split_point < LIMIT; split_point += 3) { memset(buffer, 0, strlen(buffer)) snprintf(buffer, sizeof(buffer), ",%s", split_point); strncat(keyframe_point_set, buffer, strlen(buffer)); } if you don't want do like this, perhaps you can set the *gopsize* as 1 :) Carl, Is there have some args can set the gop=1? Thanks, From amos.kittelson at sidewalktech.com Wed Oct 16 00:14:57 2013 From: amos.kittelson at sidewalktech.com (Sidewalk_Tech) Date: Tue, 15 Oct 2013 15:14:57 -0700 (PDT) Subject: [FFmpeg-user] How do I make ffplay play without high latency? In-Reply-To: <50B46910.2090909@gmail.com> References: <50B46910.2090909@gmail.com> Message-ID: <1381875297852-4661841.post@n4.nabble.com> I want to thank you for the Firefox frame lag reference! I've been trying for days to get a lower lag rate from my Panasonic WV-SC385 inside Chrome on Linux using h264 and rtsp. Like you I've been able to get near instantaneous video using RTSP in ffplay and vlc, and in Internet Exploder but not inside a linux browser. Trying it in Firefox made a HUGE difference. I still would be nice to be able to use Chrome and PNaCl to embed a ffmpeg frame for playing and RSTP stream for better frame rates. Did you ever figure out a solution to your lag issue? That might help me out in the browser. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-do-I-make-ffplay-play-without-high-latency-tp4655272p4661841.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From 690271929 at qq.com Wed Oct 16 02:28:01 2013 From: 690271929 at qq.com (=?gb18030?B?SmFja3k=?=) Date: Wed, 16 Oct 2013 08:28:01 +0800 Subject: [FFmpeg-user] =?gb18030?b?u9i4tKO6ICBmZm1wZWcgZ2V0IHZpZGVvIGZy?= =?gb18030?q?om_video_file_of_avi?= Message-ID: carl? i upload the file just a part?the whole file is 16G?when i get video from the file?the VLC and other players didn`t work with the video file.but ffplayer can do. what`s wrong? ------------------ ???? ------------------ ???: "Carl Eugen Hoyos";; ????: 2013?10?15?(???) ??11:18 ???: "ffmpeg-user"; ??: Re: [FFmpeg-user] ffmpeg get video from video file of avi ? ? gmail.com> writes: > Jacky's test file is here: > http://blog.fs-linux.org/video.avi > > I have try to play and ffprobe show_packtes?but it > just quit near at 00:06:00 It quits after ~6:30 because there is not more audio and video data in the file. Carl Eugen _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From lingjiujianke at gmail.com Wed Oct 16 04:22:32 2013 From: lingjiujianke at gmail.com (=?GB2312?B?wfUg4ao=?=) Date: Wed, 16 Oct 2013 10:22:32 +0800 Subject: [FFmpeg-user] Extract Audio from Video file using ffmpeg In-Reply-To: References: Message-ID: <25DBC416-CEBB-465F-B1EE-F9DA206A7F5E@gmail.com> Carl, I have upload Jacky's file here http://blog.fs-linux.org/truehd.m2ts ? 2013-10-15???8:01?Carl Eugen Hoyos ??? > Jacky <690271929 qq.com> writes: > >> Muxing truehd in m2ts is not supported too? > > Muxing TrueHD in transport streams is not supported > by FFmpeg currently, patch welcome. > > You don't have to upload a sample, we have a few > TrueHD files;-) > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From sayaprashantha at gmail.com Wed Oct 16 06:35:50 2013 From: sayaprashantha at gmail.com (prashantha S) Date: Wed, 16 Oct 2013 10:05:50 +0530 Subject: [FFmpeg-user] stream RTP over UDP from specific source to destination (negotiated using SIP SDP) using ffmpeg Message-ID: Hi, I am trying to stream RTP over UDP from specific source to destination (negotiated using SIP SDP) using ffmpeg I would like to open the source port and send the packets to destination port from this port This is because 1. my destination will always check from where the RTP is coming from (source verification) 2. Another usage by opening the source port is - receive packets sent from destination, otherwise destination will complain "ICMP unreachable" SIP session INVITE source SDP c=IN IP4 172.27.6.45 t=0 0 m=video 20002 RTP/AVP 120 b=AS:768 a=rtpmap:120 VP8/90000 a=sendrecv SIP session INVITE destination SDP m=video 40154 RTP/AVP 120 c=IN IP4 10.211.5.13 b=AS:768 a=rtpmap:120 VP8/90000 a=sendrecv So i have tried the below in source ffmpeg -re -i HD.webm -t 00:01:00 -an -vcodec libvpx -s 352x288 -b:v 768k -r 30 -f rtp rtp://10.211.5.13:40154?localaddr=172.27.6.45&localport=20002 But using ffmpeg i am not able to achieve this 1. FFMPEG does not stream from source IP&port i.e. localaddr=172.27.6.45&localport=20002. It uses random ports Does the usage of "?localaddr=172.27.6.45&localport=20002" is correct? 2. Destination complain that "ICMP unreachable" for 172.27.6.45:20002 Is there any solution for these 2 issues? Thank you Psaya From lingjiujianke at gmail.com Wed Oct 16 07:23:42 2013 From: lingjiujianke at gmail.com (=?GB2312?B?wfUg4ao=?=) Date: Wed, 16 Oct 2013 13:23:42 +0800 Subject: [FFmpeg-user] how can I set limit_filesize or recording_time for the specific output dynamically? In-Reply-To: References: <3EE5B2C9-5A11-49DC-AC8F-DA714781CE68@gmail.com> Message-ID: <19E22579-46BB-4BC6-812D-E09AC32E1E60@gmail.com> ? 2013-10-16???10:14?Huang Soho ??? > > > > > > Hey, > > Please try it , it can work when i used my camera > > > [root at CM ~]# ffmpeg -f video4linux2 -s 1280x720 -i /dev/video0 -preset ultrafast -force_key_frames 3,6,9,12,15,18,21,24 -vcodec libx264 -vprofile baseline -vlevel 1.0 -s 640x480 -b:v 800k -r 15 -pix_fmt yuv420p -map 0:0 -f segment -segment_format mpegts -segment_time 3 -segment_list a.m3u8 -segment_list_flags live -map 0 -flags -global_header video-%d.ts > > > hi, > > ffmpeg does not get Key frame information when ffmpeg do streamcopy, right? Right > > because the check is FALSE always: > "pkt->flags & AV_PKT_FLAG_KEY ", > > I do not know where is the code will set the pkt->flags to enable AV_PKT_FLAG_KEY when ffmpeg do streamcopy. > if ffmpeg does not do transcode, then ffmpeg does not know the content of each video frame, right? Right > then ffmpeg how to know which video frame is Key frame? if you want to set the keyframe into the AVFrame, maybe you must transcode:-), I'm not sure, Please ask in maillist. And CC Carl Because I'm not always right. Steven From mapandrei at gmail.com Wed Oct 16 07:43:27 2013 From: mapandrei at gmail.com (Andrei Petru Mura) Date: Wed, 16 Oct 2013 08:43:27 +0300 Subject: [FFmpeg-user] Convert avi to swf In-Reply-To: References: Message-ID: On Tue, Oct 15, 2013 at 6:15 PM, Carl Eugen Hoyos wrote: > Andrei Petru Mura gmail.com> writes: > > > > > It works, but it shortens the swf file. > > > Input #0, avi, from './Event20131015123035001.avi': > > Duration: 00:00:36.50, start: 0.000000, bitrate: 6799 kb/s > > > frame= 1095 fps= 37 q=31.0 Lsize= 44798kB time=00:00:36.50 > > It appears that the avi header reports a 36 seconds sample > and that the encoded swf file is 36 seconds long. > > Sorry... I twisted a little bit my files. Problem solved. Andrei M. From cehoyos at ag.or.at Wed Oct 16 10:56:18 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Oct 2013 08:56:18 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?how_can_I_set_limit=5Ffilesize_or_recordi?= =?utf-8?q?ng=5Ftime_for_the_specific_output_dynamically=3F?= References: <3EE5B2C9-5A11-49DC-AC8F-DA714781CE68@gmail.com> <19E22579-46BB-4BC6-812D-E09AC32E1E60@gmail.com> Message-ID: ? ? gmail.com> writes: > And CC Carl Please don't, it reduces the chance that you get an answer. Carl Eugen From cehoyos at ag.or.at Wed Oct 16 10:57:38 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Oct 2013 08:57:38 +0000 (UTC) Subject: [FFmpeg-user] Convert avi to swf References: Message-ID: Andrei Petru Mura gmail.com> writes: > Input #0, avi, from './Event20131015123035001.avi': > Duration: 00:00:36.50, start: 0.000000, bitrate: 6799 kb/s > Stream #0:0: Video: h264 (High) (X264 / 0x34363258), yuvj420p(pc), > 1920x1080, 30 fps, 30 tbr, 30 tbn, 30 tbc > Stream #0:1: Subtitle: none > Stream #0:2: Subtitle: none Is there any application that can do something with the two streams 0:1 and 0:2 (not matter if open source or not)? If yes, it would be great if you could provide a sample. Carl Eugen From tevans.uk at googlemail.com Wed Oct 16 11:06:52 2013 From: tevans.uk at googlemail.com (Tom Evans) Date: Wed, 16 Oct 2013 10:06:52 +0100 Subject: [FFmpeg-user] Weird characters in created passlogfile on Windows In-Reply-To: <525DB060.8060503@free.fr> References: <525DB060.8060503@free.fr> Message-ID: On Tue, Oct 15, 2013 at 10:15 PM, Thierry Lel?gard wrote: > Hello, > > I found a strange problem with ffmpeg on Windows (not tested yet on Linux) > with the -passlogfile option. > > Consider a file path containing non ASCII characters like ? (e grave) used > in all file specifications (in a directory name for instance). There are > three different use cases: input file, output file, pass log file. All three work correctly under FreeBSD: > $ ffmpeg -y -i ../h?.mkv -t 60 -pass 1 -passlogfile h? h?.ts >/dev/null 2>&1 && ls -l total 1 -rw-r--r-- 1 tom wheel 202078 16 Oct 09:05 h?-0.log -rw-r--r-- 1 tom wheel 5054756 16 Oct 09:05 h?.ts Cheers Tom From mapandrei at gmail.com Wed Oct 16 11:27:03 2013 From: mapandrei at gmail.com (Andrei Petru Mura) Date: Wed, 16 Oct 2013 12:27:03 +0300 Subject: [FFmpeg-user] Convert avi to swf In-Reply-To: References: Message-ID: On Wed, Oct 16, 2013 at 11:57 AM, Carl Eugen Hoyos wrote: > Andrei Petru Mura gmail.com> writes: > > > Input #0, avi, from './Event20131015123035001.avi': > > Duration: 00:00:36.50, start: 0.000000, bitrate: 6799 kb/s > > Stream #0:0: Video: h264 (High) (X264 / 0x34363258), yuvj420p(pc), > > 1920x1080, 30 fps, 30 tbr, 30 tbn, 30 tbc > > > Stream #0:1: Subtitle: none > > Stream #0:2: Subtitle: none > > Is there any application that can do something with > the two streams 0:1 and 0:2 (not matter if open > source or not)? > > If yes, it would be great if you could provide a > sample. To be honest. I'm very new to video processing, to ffmpeg, and related stuff. All commands that I put here are taken from Internet (so , I can't understand to much about the commands that I use). The commands that I use are working with a kind of video encoded with GeoVision codecs or something similar to this. All what I'm doing is to convert this videos at a standard video format visible on any platform. I can provide you with this kind of videos (I provided a sample some time ago). But what the specified streams are doing, I don't know. If I can help you in other way, please be more explicit. Andrei M. From cehoyos at ag.or.at Wed Oct 16 11:34:42 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Oct 2013 09:34:42 +0000 (UTC) Subject: [FFmpeg-user] Convert avi to swf References: Message-ID: Andrei Petru Mura gmail.com> writes: > If I can help you in other way, please be more explicit. The console output you kindly posted implies that FFmpeg found two subtitle streams in the file you used that it could not decode. If you want to help, please upload the file so developers can have a look if there is a missing feature in FFmpeg. Either use http://www.datafilehost.com/ or read http://ffmpeg.org/bugreports.html (there is no hard filesize limit). Thank you, Carl Eugen From mapandrei at gmail.com Wed Oct 16 12:10:59 2013 From: mapandrei at gmail.com (Andrei Petru Mura) Date: Wed, 16 Oct 2013 13:10:59 +0300 Subject: [FFmpeg-user] Convert avi to swf In-Reply-To: References: Message-ID: On Wed, Oct 16, 2013 at 12:34 PM, Carl Eugen Hoyos wrote: > Andrei Petru Mura gmail.com> writes: > > > If I can help you in other way, please be more explicit. > > The console output you kindly posted implies that FFmpeg > found two subtitle streams in the file you used that it > could not decode. > If you want to help, please upload the file so developers > can have a look if there is a missing feature in FFmpeg. > Either use http://www.datafilehost.com/ or [...] > This file is a video recording with a GeoVision camera. You can see this kind of files only if you use their codecs. This one is a video recorded with a GeoVision camera, but converted to a standard format (by GeoVision software - it is a proprietary software. Don't ask me how they do this. It is their management application installed on the NVR - network video recorder), so that you can use your codecs. For any other details, just ask me and I'll tell you what I know/can. In the case of bug fixes, is there a way to know about them, other then looking for new commits in the git repo? Or should I report this as a bug? (Sorry for noob questions if present :) ) Andrei M. From renaux.jacky at orange.fr Wed Oct 16 12:15:05 2013 From: renaux.jacky at orange.fr (jacky) Date: Wed, 16 Oct 2013 12:15:05 +0200 Subject: [FFmpeg-user] Weird characters in created passlogfile on Windows In-Reply-To: References: <525DB060.8060503@free.fr> Message-ID: <525E6729.40204@orange.fr> Le 16/10/2013 11:06, Tom Evans a ?crit : > On Tue, Oct 15, 2013 at 10:15 PM, Thierry Lel?gard > wrote: >> Hello, >> >> I found a strange problem with ffmpeg on Windows (not tested yet on Linux) >> with the -passlogfile option. >> >> Consider a file path containing non ASCII characters like ? (e grave) used >> in all file specifications (in a directory name for instance). There are >> three different use cases: input file, output file, pass log file. > All three work correctly under FreeBSD: > >> $ ffmpeg -y -i ../h?.mkv -t 60 -pass 1 -passlogfile h? h?.ts >/dev/null 2>&1 && ls -l > total 1 > -rw-r--r-- 1 tom wheel 202078 16 Oct 09:05 h?-0.log > -rw-r--r-- 1 tom wheel 5054756 16 Oct 09:05 h?.ts > > Cheers > > Tom > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Hi I havent tested with your particular case but it seems to be related to windows dos page (characters dos convertion page : chcp) I had same problem running batch file " ffmpeg -list_devices true -f dshow -i dummy" i solved it that way on dos screen undestand which active page (characters table) type chcp and note the given value (mine is 850 ) type set chcp 1252 (this enlarge ascci characters page to latin ) go back to the default page when you end up the full session hope this help regards jacky From cehoyos at ag.or.at Wed Oct 16 12:29:15 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Oct 2013 10:29:15 +0000 (UTC) Subject: [FFmpeg-user] Convert avi to swf References: Message-ID: Andrei Petru Mura gmail.com> writes: > This file > is a video recording with a GeoVision camera. You > can see this kind of files only if you use their > codecs. > This one > is a video recorded with a GeoVision camera, but > converted to a standard format Was the second (converted) sample you uploaded made from the first one? The reason I ask is that the first one plays fine here with FFmpeg (and MPlayer) but the content is quite monotonous - contrary to the second sample... If you find a bug, consider reporting it, so far I was unable to reproduce one (I would open a ticket if I find one, it is sufficient to report them here). Thank you, Carl Eugen From cehoyos at ag.or.at Wed Oct 16 12:30:26 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Oct 2013 10:30:26 +0000 (UTC) Subject: [FFmpeg-user] Convert avi to swf References: Message-ID: Andrei Petru Mura gmail.com> writes: > This file is a > video recording with a GeoVision camera. One more question: Is the file supposed to contains audio? Ie, do their codecs play audio? Or do you see any kind of subtitles, for example timecode? Carl Eugen From mapandrei at gmail.com Wed Oct 16 12:55:36 2013 From: mapandrei at gmail.com (Andrei Petru Mura) Date: Wed, 16 Oct 2013 13:55:36 +0300 Subject: [FFmpeg-user] Convert avi to swf In-Reply-To: References: Message-ID: On Wed, Oct 16, 2013 at 1:29 PM, Carl Eugen Hoyos wrote: > Andrei Petru Mura gmail.com> writes: > > > This file > > is a video recording with a GeoVision camera. You > > can see this kind of files only if you use their > > codecs. > > This one > > is a video recorded with a GeoVision camera, but > > converted to a standard format > > Was the second (converted) sample you uploaded made > from the first one? No. they are different > The reason I ask is that the > first one plays fine here with FFmpeg (and MPlayer) > I don't know how to play with ffmpeg (probably you mean ffplay). To make mplayer use ffplay I don't know > but the content is quite monotonous - contrary to > the second sample... > That should be. It's good. Anyway, I can't play the first one in Linux. (I'm using Ubuntu). But when I play movies, I play them with something else then ffplay. > > If you find a bug, consider reporting it, so far I > was unable to reproduce one (I would open a ticket > if I find one, it is sufficient to report them here). > It is likely I'll come back to this issue some time in the future. Thanks. Andrei M. From mapandrei at gmail.com Wed Oct 16 13:01:51 2013 From: mapandrei at gmail.com (Andrei Petru Mura) Date: Wed, 16 Oct 2013 14:01:51 +0300 Subject: [FFmpeg-user] Convert avi to swf In-Reply-To: References: Message-ID: On Wed, Oct 16, 2013 at 1:30 PM, Carl Eugen Hoyos wrote: > Andrei Petru Mura gmail.com> writes: > > > This file is a > > video recording with a GeoVision camera. > > One more question: Is the file supposed to contains audio? Ie, do their codecs play audio? > No. Just video. > Or do you see any kind of subtitles, for example timecode? > No. From kabelbrand at gmail.com Wed Oct 16 13:15:26 2013 From: kabelbrand at gmail.com (Gunnar) Date: Wed, 16 Oct 2013 13:15:26 +0200 Subject: [FFmpeg-user] ffmpeg option/filter similar to AviSynth Limiter In-Reply-To: References: Message-ID: Thanks Paul. But the lut filter is only for chroma, right? What about luma values? Also this filter might be a bit overkill for this purpose. On Tue, Oct 15, 2013 at 11:14 PM, Paul B Mahol wrote: > On 10/15/13, Gunnar wrote: > > Hi list, > > > > is there a way to limit levels to TV (16-235) in ffmpeg similar to > > AviSynth's Limiter or Tweak(bright=0) functions. > > I tried -vf scale with in_range/out_range which wasn't recorgnized and > the > > color_range option which seems to do nothing... > > > > I do not want to convert from PC levels to TV levels. I just want to > > enforce TV levels by removing values outside the TV range. > > There is lut filter. > From cehoyos at ag.or.at Wed Oct 16 13:23:25 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Oct 2013 11:23:25 +0000 (UTC) Subject: [FFmpeg-user] Convert avi to swf References: Message-ID: Andrei Petru Mura gmail.com> writes: > I don't know how to play with ffmpeg (probably you mean ffplay). I meant FFmpeg the project which also contains ffplay. But the sample also plays fine with the following here: $ ffmpeg -i geo_vision_format.avi -f sdl -pix_fmt yuv420p 0 > To make mplayer use ffplay I don't know The following works fine here: $ mplayer geo_vision_format.avi (ffplay only supports sdl which is a bad choice, MPlayer supports OpenGl, XVideo and VDPAU which all should perform better.) > I can't play the first one in Linux. Please provide your command line together with complete, uncut console output (current git head). Carl Eugen From mapandrei at gmail.com Wed Oct 16 14:31:47 2013 From: mapandrei at gmail.com (Andrei Petru Mura) Date: Wed, 16 Oct 2013 15:31:47 +0300 Subject: [FFmpeg-user] Convert avi to swf In-Reply-To: References: Message-ID: On Wed, Oct 16, 2013 at 2:23 PM, Carl Eugen Hoyos wrote: > Andrei Petru Mura gmail.com> writes: > > > I don't know how to play with ffmpeg (probably you mean ffplay). > > I meant FFmpeg the project which also contains ffplay. > But the sample also plays fine with the following here: > $ ffmpeg -i geo_vision_format.avi -f sdl -pix_fmt yuv420p 0 > $ ./bin/ffmpeg -i /media/artaxerxe/AC1B-69A3/avi_from_server/geo_vision_format.avi -f sdl -pix_fmt yuv420p 0 ffmpeg version git-2013-10-16-10c6d1b Copyright (c) 2000-2013 the FFmpeg developers built on Oct 16 2013 14:56:56 with gcc 4.7 (Ubuntu/Linaro 4.7.3-1ubuntu1) configuration: --prefix=/home/artaxerxe/ffmpeg_build --extra-cflags=-I/home/artaxerxe/ffmpeg_build/include --extra-ldflags=-L/home/artaxerxe/ffmpeg_build/lib --bindir=/home/artaxerxe/bin --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-x11grab libavutil 52. 46.101 / 52. 46.101 libavcodec 55. 37.100 / 55. 37.100 libavformat 55. 19.102 / 55. 19.102 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.101 / 3. 88.101 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 [avi @ 0x9e2ad60] probed stream 1 failed [avi @ 0x9e2ad60] Could not find codec parameters for stream 1 (Subtitle: none): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options [avi @ 0x9e2ad60] Could not find codec parameters for stream 2 (Subtitle: none): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, avi, from '/media/artaxerxe/AC1B-69A3/avi_from_server/geo_vision_format.avi': Duration: 00:00:42.13, start: 0.000000, bitrate: 8301 kb/s Stream #0:0: Video: h264 (High) (GAVC / 0x43564147), yuvj420p(pc), 1920x1080, 30 fps, 30 tbr, 30 tbn, 30 tbc Stream #0:1: Subtitle: none Stream #0:2: Subtitle: none [swscaler @ 0x9e19020] deprecated pixel format used, make sure you did set range correctly Output #0, sdl, to '0': Metadata: encoder : Lavf55.19.102 Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 1920x1080, q=2-31, 200 kb/s, 90k tbn, 30 tbc Stream mapping: Stream #0:0 -> #0:0 (h264 -> rawvideo) Press [q] to stop, [?] for help frame= 49 fps=8.2 q=0.0 Lsize=N/A time=00:00:01.63 bitrate=N/A video:148838kB audio:0kB subtitle:0 global headers:0kB muxing overhead -100.000014% Received signal 2: terminating. So, it seems that something is missing (Stream #0:1 and #0:2). But it is the same with the video played with GeoVision's codecs. > > > To make mplayer use ffplay I don't know > > The following works fine here: > $ mplayer geo_vision_format.avi > (ffplay only supports sdl which is a bad choice, MPlayer > supports OpenGl, XVideo and VDPAU which all should perform > better.) > $ mplayer /media/artaxerxe/AC1B-69A3/avi_from_server/geo_vision_format.avi MPlayer svn r34540 (Ubuntu), built with gcc-4.7 (C) 2000-2012 MPlayer Team mplayer: could not connect to socket mplayer: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing /media/artaxerxe/AC1B-69A3/avi_from_server/geo_vision_format.avi. libavformat version 53.21.1 (external) Mismatching header version 53.19.0 AVI file format detected. [aviheader] Video stream found, -vid 0 AVI: No audio stream found -> no sound. VIDEO: [GAVC] 1920x1080 24bpp 30.000 fps 8280.1 kbps (1010.8 kbyte/s) Load subtitles in /media/artaxerxe/AC1B-69A3/avi_from_server/ [vdpau] Error when calling vdp_device_create_x11: 1 ========================================================================== Cannot find codec matching selected -vo and video format 0x43564147. ========================================================================== Exiting... (End of file) Andrei M. From thierry at lelegard.fr Wed Oct 16 14:51:02 2013 From: thierry at lelegard.fr (Thierry Lelegard) Date: Wed, 16 Oct 2013 14:51:02 +0200 (CEST) Subject: [FFmpeg-user] Weird characters in created passlogfile on Windows In-Reply-To: <525E6729.40204@orange.fr> Message-ID: <564876467.85126020.1381927862644.JavaMail.root@spooler3-g27.priv.proxad.net> > De: "jacky" > > I havent tested with your particular case but it seems to be related > to windows dos page (characters dos convertion page : chcp) I understand what you mean but this cannot be the right explanation. I wrote that the same sequence of characters was used three times on the command line: ffmpeg -i h?.ts -pass 1 -passlogfile h? h?.mpg However, only the sequence for -passlogfile is interpreted differently: -rw-rw-rw- 1 user group 258179 Oct 15 22:50 h??-0.log -rw-rw-rw- 1 user group 5261312 Oct 15 22:50 h?.mpg -rw-rw-rw- 1 user group 18800000 Oct 15 22:50 h?.ts If your explanation were right, the output file h?.mpg would also have been created with the same gotcha. And the input file h?.ts would not even have been found in the first place. So, ffmpeg treats command line arguments and code pages quite properly. There is something specific in the creation of the passlogfile. -Thierry From cehoyos at ag.or.at Wed Oct 16 14:56:02 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Oct 2013 12:56:02 +0000 (UTC) Subject: [FFmpeg-user] Convert avi to swf References: Message-ID: Andrei Petru Mura gmail.com> writes: > $ mplayer /media/artaxerxe/AC1B-69A3/avi_from_server/geo_vision_format.avi > MPlayer svn r34540 (Ubuntu), built with gcc-4.7 (C) 2000-2012 MPlayer Team > mplayer: could not connect to socket > mplayer: No such file or directory > Failed to open LIRC support. You will not be able to use your remote > control. > > Playing /media/artaxerxe/AC1B-69A3/avi_from_server/geo_vision_format.avi. > libavformat version 53.21.1 (external) This is an intentionally broken version of MPlayer that we cannot support (neither here nor on mplayer-users). The version is said to be exploitable (I don't know if this is actually true). The current supported version from mplayerhq.hu plays your file here. Sorry, Carl Eugen From rodney.baker at iinet.net.au Wed Oct 16 15:15:53 2013 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Wed, 16 Oct 2013 23:45:53 +1030 Subject: [FFmpeg-user] Weird characters in created passlogfile on Windows In-Reply-To: <564876467.85126020.1381927862644.JavaMail.root@spooler3-g27.priv.proxad.net> References: <564876467.85126020.1381927862644.JavaMail.root@spooler3-g27.priv.proxad.net> Message-ID: <3340993.jZBHLO5x3E@mako> On Wed, 16 Oct 2013 14:51:02 Thierry Lelegard wrote: > > De: "jacky" > > > > I havent tested with your particular case but it seems to be related > > to windows dos page (characters dos convertion page : chcp) > > I understand what you mean but this cannot be the right explanation. > > I wrote that the same sequence of characters was used three times > on the command line: ffmpeg -i h?.ts -pass 1 -passlogfile h? h?.mpg > > However, only the sequence for -passlogfile is interpreted differently: > > -rw-rw-rw- 1 user group 258179 Oct 15 22:50 h??-0.log > -rw-rw-rw- 1 user group 5261312 Oct 15 22:50 h?.mpg > -rw-rw-rw- 1 user group 18800000 Oct 15 22:50 h?.ts > > If your explanation were right, the output file h?.mpg would also have > been created with the same gotcha. And the input file h?.ts would not > even have been found in the first place. > > So, ffmpeg treats command line arguments and code pages quite properly. > There is something specific in the creation of the passlogfile. > > -Thierry > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Thierry, It might be worth asking this question on the ffmpeg-devel mailing list. If you do, make sure you include the full command line and full, uncut console output, plus as much detail as possible to reproduce the problem (the way you've described it here is probably OK). Note that most developers use Linux though, so be prepared to assist with testing any fixes and provide additional feedback. If you know enough to propose a patch it would be welcomed I'm sure. You can submit the proposed patch to the ffmpeg-devel mailing list. Rodney. -- ============================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au ============================================================== From onemda at gmail.com Wed Oct 16 16:31:42 2013 From: onemda at gmail.com (Paul B Mahol) Date: Wed, 16 Oct 2013 14:31:42 +0000 Subject: [FFmpeg-user] ffmpeg option/filter similar to AviSynth Limiter In-Reply-To: References: Message-ID: On 10/16/13, Gunnar wrote: > Thanks Paul. > > But the lut filter is only for chroma, right? What about luma values? Have you actually read documentation? lut filter can change only luma component just fine. > Also this filter might be a bit overkill for this purpose. Huh? > > > On Tue, Oct 15, 2013 at 11:14 PM, Paul B Mahol wrote: > >> On 10/15/13, Gunnar wrote: >> > Hi list, >> > >> > is there a way to limit levels to TV (16-235) in ffmpeg similar to >> > AviSynth's Limiter or Tweak(bright=0) functions. >> > I tried -vf scale with in_range/out_range which wasn't recorgnized and >> the >> > color_range option which seems to do nothing... >> > >> > I do not want to convert from PC levels to TV levels. I just want to >> > enforce TV levels by removing values outside the TV range. >> >> There is lut filter. >> > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Wed Oct 16 16:46:17 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Oct 2013 14:46:17 +0000 (UTC) Subject: [FFmpeg-user] Weird characters in created passlogfile on Windows References: <564876467.85126020.1381927862644.JavaMail.root@spooler3-g27.priv.proxad.net> <3340993.jZBHLO5x3E@mako> Message-ID: Rodney Baker iinet.net.au> writes: > It might be worth asking this question on the > ffmpeg-devel mailing list. (I did not really read this thread.) Please only do that if you plan to work on a fix yourself, bug reports should never be sent to ffmpeg-devel. Carl Eugen From kabelbrand at gmail.com Wed Oct 16 16:55:46 2013 From: kabelbrand at gmail.com (Gunnar) Date: Wed, 16 Oct 2013 16:55:46 +0200 Subject: [FFmpeg-user] ffmpeg option/filter similar to AviSynth Limiter In-Reply-To: References: Message-ID: Of course, thanks. I had mistaken lut with the haldclut filter. On Wed, Oct 16, 2013 at 4:31 PM, Paul B Mahol wrote: > On 10/16/13, Gunnar wrote: > > But the lut filter is only for chroma, right? What about luma values? > > Have you actually read documentation? lut filter can change only luma > component just > fine. > > > Also this filter might be a bit overkill for this purpose. > > Huh? > > > On Tue, Oct 15, 2013 at 11:14 PM, Paul B Mahol wrote: > > > >> On 10/15/13, Gunnar wrote: > >> > Hi list, > >> > > >> > is there a way to limit levels to TV (16-235) in ffmpeg similar to > >> > AviSynth's Limiter or Tweak(bright=0) functions. > >> > I tried -vf scale with in_range/out_range which wasn't recorgnized and > >> the > >> > color_range option which seems to do nothing... > >> > > >> > I do not want to convert from PC levels to TV levels. I just want to > >> > enforce TV levels by removing values outside the TV range. > >> > >> There is lut filter. > From thierry at lelegard.fr Wed Oct 16 16:57:24 2013 From: thierry at lelegard.fr (Thierry Lelegard) Date: Wed, 16 Oct 2013 16:57:24 +0200 (CEST) Subject: [FFmpeg-user] Weird characters in created passlogfile on Windows In-Reply-To: Message-ID: <1398668576.85483548.1381935444062.JavaMail.root@spooler3-g27.priv.proxad.net> > De: "Carl Eugen Hoyos" > > > It might be worth asking this question on the > > ffmpeg-devel mailing list. > > (I did not really read this thread.) > Please only do that if you plan to work on a fix yourself, > bug reports should never be sent to ffmpeg-devel. You are right. I will submit a bug report to FFmpeg Trac system when I have time. -Thierry From liuzx.soho at gmail.com Wed Oct 16 11:33:04 2013 From: liuzx.soho at gmail.com (liuzx.soho) Date: Wed, 16 Oct 2013 17:33:04 +0800 Subject: [FFmpeg-user] configure make a mistake with the lastest version Message-ID: <201310161733010584161@gmail.com> hello , I have tested the version FFmpeg-master and ffmpeg-2.0.2 with command "./config.sh",while it couldn't compile successfully . The following is the log info : ./config.sh: line 9: --enable-shared: command not found sed????? config.h?No such file or directory sed????? config.h?No such file or directory sed????? config.h?No such file or directory sed????? config.h?No such file or directory sed????? config.h?No such file or directory sed????? config.h?No such file or directory sed????? config.h?No such file or directory what's the problem ? Thank you liuzx.soho From lou at lrcd.com Wed Oct 16 20:56:34 2013 From: lou at lrcd.com (Lou) Date: Wed, 16 Oct 2013 10:56:34 -0800 Subject: [FFmpeg-user] configure make a mistake with the lastest version In-Reply-To: <201310161733010584161@gmail.com> References: <201310161733010584161@gmail.com> Message-ID: <20131016105634.1266a66d@lrcd.com> On Wed, 16 Oct 2013 17:33:04 +0800 liuzx.soho wrote: > ./config.sh: line 9: --enable-shared: command not found FFmpeg does not come with a config.sh file. Perhaps you meant "./configure". From elliottbalsley at gmail.com Wed Oct 16 21:28:17 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Wed, 16 Oct 2013 12:28:17 -0700 Subject: [FFmpeg-user] Scaling 1080 to 720 Message-ID: What is the difference between using "-s hd720" and "-vf scale=hd720" ? From onemda at gmail.com Wed Oct 16 21:50:31 2013 From: onemda at gmail.com (Paul B Mahol) Date: Wed, 16 Oct 2013 19:50:31 +0000 Subject: [FFmpeg-user] Scaling 1080 to 720 In-Reply-To: References: Message-ID: On 10/16/13, Elliott Balsley wrote: > What is the difference between using "-s hd720" and "-vf scale=hd720" ? > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > That first is applied last and other somewhere in filtergraph where filter is actually positioned. From elliottbalsley at gmail.com Wed Oct 16 23:25:49 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Wed, 16 Oct 2013 14:25:49 -0700 Subject: [FFmpeg-user] error handling for map_channel Message-ID: <959BCDFB-4736-4F18-9A22-D9B099A02785@gmail.com> I've been using ffmpeg to batch process lots of files by putting it in a find -execdir command. The source clips I'm encoding have varying numbers of audio tracks, and I always want to encode only the first two. Below is my command, which is working beautifully... find $source_dir -maxdepth 1 -iname "*.mov" -execdir ffmpeg -i "{}" -pix_fmt yuv420p -vf scale=hd720 -acodec libfdk_aac -b:a 128k -map_channel 0.1.0 -map_channel 0.1.1 -vcodec libx264 -preset veryfast -crf 18 -level 31 H.264/"{}" \; But when it comes to a clip that has fewer than two audio tracks, it fails with this error: mapchan: stream #0.1 is not an audio stream. In this case, I want the encode to have the same number of channels as the source (either 0 or 1). Is there some way to make ffmpeg handle this? From elliottbalsley at gmail.com Wed Oct 16 23:27:42 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Wed, 16 Oct 2013 14:27:42 -0700 Subject: [FFmpeg-user] Scaling 1080 to 720 In-Reply-To: References: Message-ID: <3848B59B-CBDC-4BFA-875C-2BA505DF496D@gmail.com> On Oct 16, 2013, at 12:50 PM, Paul B Mahol wrote: > On 10/16/13, Elliott Balsley wrote: >> What is the difference between using "-s hd720" and "-vf scale=hd720" ? >> > That first is applied last and other somewhere in filtergraph where > filter is actually positioned. So if that's the only video filter being used, there would be no difference? From dashing.meng at gmail.com Thu Oct 17 04:03:11 2013 From: dashing.meng at gmail.com (littlebat) Date: Thu, 17 Oct 2013 10:03:11 +0800 Subject: [FFmpeg-user] error handling for map_channel In-Reply-To: <959BCDFB-4736-4F18-9A22-D9B099A02785@gmail.com> References: <959BCDFB-4736-4F18-9A22-D9B099A02785@gmail.com> Message-ID: <20131017100311.8d4be33540bba4026d3cc1f9@gmail.com> On Wed, 16 Oct 2013 14:25:49 -0700 Elliott Balsley wrote: > I've been using ffmpeg to batch process lots of files by putting it > in a find -execdir command. The source clips I'm encoding have > varying numbers of audio tracks, and I always want to encode only the > first two. Below is my command, which is working beautifully... > > find $source_dir -maxdepth 1 -iname "*.mov" -execdir ffmpeg -i "{}" > -pix_fmt yuv420p -vf scale=hd720 -acodec libfdk_aac -b:a 128k > -map_channel 0.1.0 -map_channel 0.1.1 -vcodec libx264 -preset > veryfast -crf 18 -level 31 H.264/"{}" \; > > But when it comes to a clip that has fewer than two audio tracks, it > fails with this error: mapchan: stream #0.1 is not an audio stream. > > In this case, I want the encode to have the same number of channels > as the source (either 0 or 1). Is there some way to make ffmpeg > handle this? _______________________________________________ You can write a script, detect audio channel information using ffmpeg (ffprobe) first, then deal with the audio mapping according the information. From 690271929 at qq.com Thu Oct 17 10:01:59 2013 From: 690271929 at qq.com (=?gb18030?B?SmFja3k=?=) Date: Thu, 17 Oct 2013 16:01:59 +0800 Subject: [FFmpeg-user] m2ts file with TRUEHD audio Message-ID: carl: i have upload the m2ts file here: http://blog.fs-linux.org/truehd.m2ts, i debug /libavformat/mpegts.c and /libavformat/mpegenc.c, it seems demux suppport truehd, but mux don't support. we have to add something in function mpegts_write_pmt() of /libavformat/mpegenc.c? do you have some infomation of truehd? switch(st->codec->codec_type) { case AVMEDIA_TYPE_AUDIO: if(st->codec->codec_id==AV_CODEC_ID_EAC3){ *q++=0x7a; // EAC3 descriptor see A038 DVB SI *q++=1; // 1 byte, all flags sets to 0 *q++=0; // omit all fields... } if(st->codec->codec_id==AV_CODEC_ID_S302M){ *q++ = 0x05; /* MPEG-2 registration descriptor*/ *q++ = 4; *q++ = 'B'; *q++ = 'S'; *q++ = 'S'; *q++ = 'D'; } if(st->codec->codec_id==AV_CODEC_ID_TRUEHD){ ???????????? } .... } thank you very much. From onemda at gmail.com Thu Oct 17 10:28:45 2013 From: onemda at gmail.com (Paul B Mahol) Date: Thu, 17 Oct 2013 08:28:45 +0000 Subject: [FFmpeg-user] Scaling 1080 to 720 In-Reply-To: <3848B59B-CBDC-4BFA-875C-2BA505DF496D@gmail.com> References: <3848B59B-CBDC-4BFA-875C-2BA505DF496D@gmail.com> Message-ID: On 10/16/13, Elliott Balsley wrote: > On Oct 16, 2013, at 12:50 PM, Paul B Mahol wrote: > >> On 10/16/13, Elliott Balsley wrote: >>> What is the difference between using "-s hd720" and "-vf scale=hd720" ? >>> >> That first is applied last and other somewhere in filtergraph where >> filter is actually positioned. > > So if that's the only video filter being used, there would be no > difference? Maybe, because defaults args for -vf scale may be different than -s .... Depends how -s is done in ffmpeg.... > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ramitbhalla at gmail.com Thu Oct 17 20:02:35 2013 From: ramitbhalla at gmail.com (Ramit Bhalla) Date: Thu, 17 Oct 2013 14:02:35 -0400 Subject: [FFmpeg-user] Understanding the ffmpeg output for start Message-ID: Does anyone know what Start represents in the ffmpeg output? Duration: 01:55:44.37, start: 573.203731, bitrate: 17727 kb/s From liuzx.soho at gmail.com Thu Oct 17 11:05:56 2013 From: liuzx.soho at gmail.com (liuzx.soho) Date: Thu, 17 Oct 2013 17:05:56 +0800 Subject: [FFmpeg-user] configure make a warning with the lastest version Message-ID: <201310171705529416922@gmail.com> To configure the enviroment of ffmpeg , I solved one by one problem . At last only one is left ,simply it is a warning . I wonder if the arm-linux-androideabi-pkg-config is necessary . perhaps i have found the answer in "http://www.16kan.com/question/detail/109220.html", while I still want to get a confirm . The problem info is as follows : WARNING: /cygdrive/e/Android_Home/Software/cygwin/home/js/android-ndk-r8/toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows/bin/arm-linux-androideabi-pkg-c onfig not found, library detection may fail.WARNING: Compiler does not indicate floating-point ABI, guessing vfp. Thank you ! liuzx.soho From ranyang1005 at gmail.com Thu Oct 17 14:56:45 2013 From: ranyang1005 at gmail.com (Ran Yang) Date: Thu, 17 Oct 2013 20:56:45 +0800 Subject: [FFmpeg-user] About FFmpeg 0.5.13 Message-ID: I have noticed that FFmpeg 0.5.13 was released this year. I have two questions it. Firstly, was all known CVE issue solved in this release? Secondly, Changes after 0.5.10 cannot found in Changelog, what's that? Would you please help me? Thanks in advance. From luthlee at gmail.com Tue Oct 15 14:31:19 2013 From: luthlee at gmail.com (vivienlwt) Date: Tue, 15 Oct 2013 05:31:19 -0700 (PDT) Subject: [FFmpeg-user] How to get raw frame data from AVFrame.data[] and AVFrame.linesize[] without specifying the pixel format? Message-ID: I get the general idea that the frame.data[] is interpreted depending on which pixel format is the video (RGB or YUV). But is there any general way to get all the pixel data from the frame? I just want to compute the hash of the frame data, without interpret it to display the image. According to AVFrame.h: uint8_t* AVFrame::data[AV_NUM_DATA_POINTERS] pointer to the picture/channel planes. int AVFrame::linesize[AV_NUM_DATA_POINTERS] For video, size in bytes of each picture line. Does this mean that if I just extract from data[i] for linesize[i] bytes then I get the full pixel information about the frame? But according to some rendering code of RGB it seems that for one channel (channel 0), it contains more data than linesize[0] suggested: for (int y = 0; y < frameHeight; y++) memcpy(img.scanLine(y), frameRGB->data[0] + y*frameRGB->linesize[0], frameWidth*3); the code suggests that frameRGB->data[0] actually contains frameHeight * frameWidth * 3 bytes of data, where frameRGB->linesize[0] = frameWidth*3 So what should be the right way to get all the pixel information from frame->data[]?? Thanks. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-to-get-raw-frame-data-from-AVFrame-data-and-AVFrame-linesize-without-specifying-the-pixel-format-tp4661827.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From lou at lrcd.com Thu Oct 17 20:35:10 2013 From: lou at lrcd.com (Lou) Date: Thu, 17 Oct 2013 10:35:10 -0800 Subject: [FFmpeg-user] Scaling 1080 to 720 In-Reply-To: References: <3848B59B-CBDC-4BFA-875C-2BA505DF496D@gmail.com> Message-ID: <20131017103510.6e29e06d@lrcd.com> On Thu, 17 Oct 2013 08:28:45 +0000 Paul B Mahol wrote: > Maybe, because defaults args for -vf scale may be different than -s .... > > Depends how -s is done in ffmpeg.... At least the outputs, in this lazy test, seem the same: $ ffmpeg -f lavfi -i testsrc=s=hd1080:d=3 -vf scale=hd720 -f md5 - MD5=59fee7d8eeeec0bc4d6f59864721feba $ ffmpeg -f lavfi -i testsrc=s=hd1080:d=3 -s hd720 -f md5 - MD5=59fee7d8eeeec0bc4d6f59864721feba From vincent at up4.com Thu Oct 17 20:56:10 2013 From: vincent at up4.com (Vincent Olivier) Date: Thu, 17 Oct 2013 14:56:10 -0400 Subject: [FFmpeg-user] FCPX-remuxed XAVC Message-ID: <83B55B4E-8560-47A7-B532-26E144A467C1@up4.com> Hi, Final Cut Pro X remixes Sony's XAVC material from the original MXF file to a MOV Quicktime container. The bitstream is (supposedly) untouched, but while FFMPEG can parse the original MXF file correctly, it refuses to recognize the AVC stream and pass it to x264. I think Apple is using a new FOURCC identifier (aivx) that is not recognized as h264 but should be? Here are the ffprobe's outputs for both files: MOV file: Stream #0:0(und): Video: none (aivx / 0x78766961), 4096x2160, 231409 kb/s, SAR 1:1 DAR 256:135, 23.98 fps, 23.98 tbr, 24k tbn, 24k tbc (default) MXF file: Stream #0:0: Video: h264 (High 4:2:2 Intra), yuv422p10le, 4096x2160 [SAR 1:1 DAR 256:135], 23.98 fps, 23.98 tbr, 23.98 tbn, 47.95 tbc Is there a way to force x264 decoding anyways through the command line or programmatically (I also wrote a program using libavcodec as libraries? Here is the FFMPEG version I'm using: ffmpeg-devel-20131009_0+gpl2+nonfree.darwin_12.x86_64.tbz2 from http://lil.fr.packages.macports.org/ffmpeg-devel Regards, and thanks. Vincent -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4879 bytes Desc: not available URL: From ffmpeg at theindianmaiden.com Thu Oct 17 17:23:25 2013 From: ffmpeg at theindianmaiden.com (imaiden) Date: Thu, 17 Oct 2013 08:23:25 -0700 (PDT) Subject: [FFmpeg-user] [mov, mp4, m4a, 3gp, 3g2, mj2 @ 0x2684100] error, moov atom not found, file broken Message-ID: <1382023405375-4661884.post@n4.nabble.com> I have a fresh install on a centos 6.4 machine and keep getting this error for prores HQ files. The source is not damaged in nay way and reads fine with ffmpeg on os x. I have tried everything I can find on the web to assist. I also get the same error with ffmbc. Does anybody know what it is I am doing wrong? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/mov-mp4-m4a-3gp-3g2-mj2-0x2684100-error-moov-atom-not-found-file-broken-tp4661884.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From cehoyos at ag.or.at Thu Oct 17 23:09:50 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 17 Oct 2013 21:09:50 +0000 (UTC) Subject: [FFmpeg-user] [mov, mp4, m4a, 3gp, 3g2, mj2 0x2684100] error, moov atom not found, file broken References: <1382023405375-4661884.post@n4.nabble.com> Message-ID: imaiden theindianmaiden.com> writes: > I have a fresh install on a centos 6.4 machine and > keep getting this error for prores HQ files. The > source is not damaged in nay way and reads fine > with ffmpeg on os x. Please provide the complete, uncut output of ffmpeg -i yourfile for both installations. Carl Eugen From cehoyos at ag.or.at Thu Oct 17 23:12:54 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 17 Oct 2013 21:12:54 +0000 (UTC) Subject: [FFmpeg-user] How to get raw frame data from AVFrame.data[] and AVFrame.linesize[] without specifying the pixel format? References: Message-ID: vivienlwt gmail.com> writes: > According to AVFrame.h: > > uint8_t* AVFrame::data[AV_NUM_DATA_POINTERS] > > pointer to the picture/channel planes. > > int AVFrame::linesize[AV_NUM_DATA_POINTERS] > > For video, size in bytes of each picture line. > > Does this mean that if I just extract from data[i] > for linesize[i] bytes then I get the full pixel > information about the frame? No, it contains the information for one (horizontal) line of the frame. Carl Eugen From cehoyos at ag.or.at Thu Oct 17 23:10:38 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 17 Oct 2013 21:10:38 +0000 (UTC) Subject: [FFmpeg-user] FCPX-remuxed XAVC References: <83B55B4E-8560-47A7-B532-26E144A467C1@up4.com> Message-ID: Vincent Olivier up4.com> writes: > MOV file: Stream #0:0(und): Video: none Please provide the sample. For future questions: Please always provide the complete, uncut console output, do not post excerpts. Carl Eugen From belcampo at zonnet.nl Thu Oct 17 23:23:38 2013 From: belcampo at zonnet.nl (Henk D. Schoneveld) Date: Thu, 17 Oct 2013 23:23:38 +0200 Subject: [FFmpeg-user] Understanding the ffmpeg output for start In-Reply-To: References: Message-ID: <0CE1302B-3B97-402C-9803-703D7095FE72@zonnet.nl> On Oct 17, 2013, at 8:02 PM, Ramit Bhalla wrote: > Does anyone know what Start represents in the ffmpeg output? AFAIK, Recordings from TV have an ongoing timer running. Say you record from 9:00 AM till 10:00 AM, start could be somewhere around well 9*60*60 = 32400. But if the broadcaster started at 07:00 it could be 2*60*60 = 7200. It mostly depends on when broadcasters 'starts' counting. Henk > > Duration: 01:55:44.37, start: 573.203731, bitrate: 17727 kb/s > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From cehoyos at ag.or.at Thu Oct 17 23:13:50 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 17 Oct 2013 21:13:50 +0000 (UTC) Subject: [FFmpeg-user] Understanding the ffmpeg output for start References: Message-ID: Ramit Bhalla gmail.com> writes: > Does anyone know what Start represents in the ffmpeg output? > > Duration: 01:55:44.37, start: 573.203731, bitrate: 17727 kb/s It is the timestamp of the first frame. Carl Eugen From bouke at videotoolshed.com Thu Oct 17 23:29:37 2013 From: bouke at videotoolshed.com (Bouke (VideoToolShed)) Date: Thu, 17 Oct 2013 23:29:37 +0200 Subject: [FFmpeg-user] Understanding the ffmpeg output for start References: Message-ID: <7E50CE8D044E45F495FA8688AB425BAA@HPKANTOOR> ----- Original Message ----- From: "Carl Eugen Hoyos" To: Sent: Thursday, October 17, 2013 11:13 PM Subject: Re: [FFmpeg-user] Understanding the ffmpeg output for start > Ramit Bhalla gmail.com> writes: > >> Does anyone know what Start represents in the ffmpeg output? >> >> Duration: 01:55:44.37, start: 573.203731, bitrate: 17727 kb/s > > It is the timestamp of the first frame. At the risk of sounding more stupid than i am: What exactly is a timestamp in this context? How to interpret it? Bouke > 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 Oct 17 23:11:29 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 17 Oct 2013 21:11:29 +0000 (UTC) Subject: [FFmpeg-user] About FFmpeg 0.5.13 References: Message-ID: Ran Yang gmail.com> writes: > I have noticed that FFmpeg 0.5.13 was released this year. Please try to update to current git head, if this is impossible, use 2.0 Carl Eugen From bouke at videotoolshed.com Thu Oct 17 23:32:08 2013 From: bouke at videotoolshed.com (Bouke (VideoToolShed)) Date: Thu, 17 Oct 2013 23:32:08 +0200 Subject: [FFmpeg-user] Understanding the ffmpeg / DOH References: <7E50CE8D044E45F495FA8688AB425BAA@HPKANTOOR> Message-ID: <0EFAF80AAF6D4916A680965018F596F9@HPKANTOOR> Ignore my previous message... ----- Original Message ----- From: "Bouke (VideoToolShed)" To: "FFmpeg user questions" Sent: Thursday, October 17, 2013 11:29 PM Subject: Re: [FFmpeg-user] Understanding the ffmpeg output for start > ----- Original Message ----- > From: "Carl Eugen Hoyos" > To: > Sent: Thursday, October 17, 2013 11:13 PM > Subject: Re: [FFmpeg-user] Understanding the ffmpeg output for start > > >> Ramit Bhalla gmail.com> writes: >> >>> Does anyone know what Start represents in the ffmpeg output? >>> >>> Duration: 01:55:44.37, start: 573.203731, bitrate: 17727 kb/s >> >> It is the timestamp of the first frame. > > At the risk of sounding more stupid than i am: > What exactly is a timestamp in this context? How to interpret it? Note to self, read back a bit.... Bouke > Bouke > >> 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 vincent at up4.com Thu Oct 17 23:52:36 2013 From: vincent at up4.com (Vincent Olivier) Date: Thu, 17 Oct 2013 17:52:36 -0400 Subject: [FFmpeg-user] FCPX-remuxed XAVC In-Reply-To: References: <83B55B4E-8560-47A7-B532-26E144A467C1@up4.com> Message-ID: <6942E6FB-F743-450B-98FD-060CBD8E1D8D@up4.com> Sample: http://j.mp/1czaPPe Full log: vincent$ ffprobe -i 2013-03-31\ 13_20_46\ \(id\).mov ffprobe version 2.0-3aa5765 Copyright (c) 2007-2013 the FFmpeg developers built on Oct 17 2013 14:52:20 with Apple LLVM version 5.0 (clang-500.2.75) (based on LLVM 3.3svn) configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libass --enable-libbluray --enable-gnutls --enable-libfreetype --disable-outdev=xv --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --arch=x86_64 --enable-yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid --enable-nonfree --enable-libfaac libavutil 52. 46.101 / 52. 46.101 libavcodec 55. 35.100 / 55. 35.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.101 / 3. 88.101 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7facd285d400] Stream #1: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7facd285d400] Stream #2: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7facd285d400] Stream #3: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7facd285d400] Stream #4: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7facd285d400] Stream #5: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7facd285d400] Stream #6: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7facd285d400] Stream #7: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7facd285d400] Stream #8: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7facd285d400] Stream #9: not enough frames to estimate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7facd285d400] Could not find codec parameters for stream 0 (Video: none (aivx / 0x78766961), 4096x2160, 231333 kb/s): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '2013-03-31 13_20_46 (id).mov': Metadata: major_brand : qt minor_version : 0 compatible_brands: qt creation_time : 2013-03-31 23:33:12 Duration: 00:00:06.67, start: 0.000000, bitrate: 244137 kb/s Stream #0:0(und): Video: none (aivx / 0x78766961), 4096x2160, 231333 kb/s, SAR 1:1 DAR 256:135, 23.98 fps, 23.98 tbr, 24k tbn, 24k tbc (default) Metadata: creation_time : 2013-03-31 23:33:12 handler_name : Core Media Data Handler timecode : 05:11:04:20 Stream #0:1(und): Audio: pcm_f32le (fl32 / 0x32336C66), 48000 Hz, 1 channels, flt, 1536 kb/s (default) Metadata: creation_time : 2013-03-31 23:33:12 handler_name : Core Media Data Handler Stream #0:2(und): Audio: pcm_f32le (fl32 / 0x32336C66), 48000 Hz, 1 channels, flt, 1536 kb/s (default) Metadata: creation_time : 2013-03-31 23:33:12 handler_name : Core Media Data Handler Stream #0:3(und): Audio: pcm_f32le (fl32 / 0x32336C66), 48000 Hz, 1 channels, flt, 1536 kb/s (default) Metadata: creation_time : 2013-03-31 23:33:12 handler_name : Core Media Data Handler Stream #0:4(und): Audio: pcm_f32le (fl32 / 0x32336C66), 48000 Hz, 1 channels, flt, 1536 kb/s (default) Metadata: creation_time : 2013-03-31 23:33:12 handler_name : Core Media Data Handler Stream #0:5(und): Audio: pcm_f32le (fl32 / 0x32336C66), 48000 Hz, 1 channels, flt, 1536 kb/s (default) Metadata: creation_time : 2013-03-31 23:33:12 handler_name : Core Media Data Handler Stream #0:6(und): Audio: pcm_f32le (fl32 / 0x32336C66), 48000 Hz, 1 channels, flt, 1536 kb/s (default) Metadata: creation_time : 2013-03-31 23:33:12 handler_name : Core Media Data Handler Stream #0:7(und): Audio: pcm_f32le (fl32 / 0x32336C66), 48000 Hz, 1 channels, flt, 1536 kb/s (default) Metadata: creation_time : 2013-03-31 23:33:12 handler_name : Core Media Data Handler Stream #0:8(und): Audio: pcm_f32le (fl32 / 0x32336C66), 48000 Hz, 1 channels, flt, 1536 kb/s (default) Metadata: creation_time : 2013-03-31 23:33:12 handler_name : Core Media Data Handler Stream #0:9(und): Data: none (tmcd / 0x64636D74), 0 kb/s Metadata: creation_time : 2013-03-31 23:33:12 handler_name : Core Media Data Handler timecode : 05:11:04:20 Unsupported codec with id 0 for input stream 0 Unsupported codec with id 0 for input stream 9 On 2013-10-17, at 17:10, Carl Eugen Hoyos wrote: > Vincent Olivier up4.com> writes: > >> MOV file: Stream #0:0(und): Video: none > > Please provide the sample. > > For future questions: Please always provide the complete, > uncut console output, do not post excerpts. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4879 bytes Desc: not available URL: From peter.zhukov at gmail.com Fri Oct 18 11:13:24 2013 From: peter.zhukov at gmail.com (=?UTF-8?B?0J/QtdGC0YAg0JbRg9C60L7Qsg==?=) Date: Fri, 18 Oct 2013 13:13:24 +0400 Subject: [FFmpeg-user] Concat image & video Message-ID: Hello, my work needs (me) to find a solution to question: 1. How to concat image & video -> to result video. To create a video from image I used command: ffmpeg -loop 1 -i poster.png -c:v libx264 -t 30 3.mp4 And to concat 2 videos I used command: ffmpeg -i 1.mp4 -i 2.mp4 -filter_complex "concat=n=2:v=1:a=1" 3.mp4 but command like this does not work: ffmpeg -loop 1 -i poster.png -t 30 -i 1.mp4 -filter_complex "concat=n=2:v=1:a=1" d:\3.mp4 How to concat image, for 5 secs & video next right after, in one command? Is it really possible, does someone know this? ... And, may be this can clear something: ffmpeg -i input1.mp4 -i input2.webm -filter_complex "[0:0] [0:1] [1:0] [1:1] concat=n=2:v=1:a=1 [v] [a]" -map "[v]" -map "[a]" 3.mp4 this concats 2 video files using streams from them: v = [0:0] & [1:0], a=[0:1] & [1:1] and [v] [a] is output streams, and -map - maps them to the streams in output file. Thanks. From mailinglist at podiumbv.nl Fri Oct 18 11:18:49 2013 From: mailinglist at podiumbv.nl (Podium B.V.) Date: Fri, 18 Oct 2013 11:18:49 +0200 Subject: [FFmpeg-user] ProRes -> 1stream (pcm_s24le) 16ch audio -> invalid channel layout Message-ID: <5260FCF9.3030206@podiumbv.nl> Hi All, first of all, and I can't say it enough, what is FFMpeg a GREAT tool! I work on the IT-department of a Video Broadcast Company, and where the "video-people" want to buy all kinds of expensive (windows) software. We almost everytime can script it cheaper, better and much faster!! (on a good machine FFMpeg is FAST!) then the proprietary software. But there is one little problem, wich could us save much time in our workflows, where I can't find a good solution for... In the broadcast video world (not ENG) it is very default to asume that when there is one video signaal there could be 16 (mono) audio channels. This is because of the spefication of the SDI protocol we use between equiment. And the way these mov-containers most of the time are recored is: Stream# 0.0 Video Stream# 0.1 Audio -> with 16 diffrent (mono) channels! Stream# 0.2 Data With a lot of testing I did find out, when there are more than 8 channels (7.1 surround) it's not longer possible to extract the audio or split channels to diffrent stream. Not with: -map / -map_channel / -filter_complex channelspilt / pan / etc... Also not with the latest build (yesterday, 17th), the build of 1st of July or the latest static build (2.0.1)... Our workarround for now is first open the recored file in a proprietary broadcast application. Extract 8 audio channels, save the file without a render (which still can take up to a hour!?!) and start the desired FFMpeg script... A quote from the creativecow forum (http://forums.creativecow.net/thread/291/978) "/I think your right. I created a test file with a video and a 16 channel audio stream.// //I created it with ffmpeg btw. It can _create_ the 16 channel audio stream, but the// //filter subsystem doesn't like it when I feed it back in to try to split the channels out.// //Gives errors about invalid channel layout./" My question: * Do I maybe overlooked something? * Or is this a (new) feature not (yet) implemented? * Or is this a "problem" which is not easy to fix cause of the way ffmpeg is build? I am not a programmer, so I really have no idea how to proceed... ...even if I could get my boss to hire a programmer to help solve this problem for everybody out there, I don't know if it is even possible... Kind regards, Ed From cehoyos at ag.or.at Fri Oct 18 11:47:34 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 18 Oct 2013 09:47:34 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?ProRes_-=3E_1stream_=28pcm=5Fs24le=29_16c?= =?utf-8?q?h_audio_-=3E_invalid_channel_layout?= References: <5260FCF9.3030206@podiumbv.nl> Message-ID: Podium B.V. podiumbv.nl> writes: > With a lot of testing I did find out, when there are > more than 8 channels (7.1 surround) it's not longer > possible to extract the audio or split channels to > diffrent stream. Please test current git head and provide your failing command line together with the complete, uncut console output. Carl Eugen From mailinglist at podiumbv.nl Fri Oct 18 11:56:10 2013 From: mailinglist at podiumbv.nl (Podium B.V.) Date: Fri, 18 Oct 2013 11:56:10 +0200 Subject: [FFmpeg-user] ProRes -> 1stream (pcm_s24le) 16ch audio -> invalid channel layout In-Reply-To: References: <5260FCF9.3030206@podiumbv.nl> Message-ID: <526105BA.4040506@podiumbv.nl> On 18-10-13 11:47, Carl Eugen Hoyos wrote: > Podium B.V. podiumbv.nl> writes: >> With a lot of testing I did find out, when there are >> more than 8 channels (7.1 surround) it's not longer >> possible to extract the audio or split channels to >> diffrent stream. > Please test current git head and provide your failing > command line together with the complete, uncut > console output. > > Carl Eugen Dear Carl, "Please test current git head" Can I use the latest buid from: http://ffmpeg.gusari.org/static/64bit/ Or do really mean to checkout the code and build it myself ? "provide your failing command line together with the complete, uncut console output." I will! From cehoyos at ag.or.at Fri Oct 18 11:58:27 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 18 Oct 2013 09:58:27 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?ProRes_-=3E_1stream_=28pcm=5Fs24le=29_16c?= =?utf-8?q?h_audio_-=3E_invalid_channel_layout?= References: <5260FCF9.3030206@podiumbv.nl> <526105BA.4040506@podiumbv.nl> Message-ID: Podium B.V. podiumbv.nl> writes: > Can I use the latest buid from: http://ffmpeg.gusari.org/static/64bit/ > Or do really mean to checkout the code and build it myself ? Whatever seems more appropriate to you. (Since my next answer will be: "Please test b1ef4dc", you should really consider to compile yourself, it does not hurt.) Carl Eugen From mailinglist at podiumbv.nl Fri Oct 18 12:03:26 2013 From: mailinglist at podiumbv.nl (Podium B.V.) Date: Fri, 18 Oct 2013 12:03:26 +0200 Subject: [FFmpeg-user] ProRes -> 1stream (pcm_s24le) 16ch audio -> invalid channel layout In-Reply-To: References: <5260FCF9.3030206@podiumbv.nl> <526105BA.4040506@podiumbv.nl> Message-ID: <5261076E.9030409@podiumbv.nl> On 18-10-13 11:58, Carl Eugen Hoyos wrote: > Podium B.V. podiumbv.nl> writes: > >> Can I use the latest buid from: http://ffmpeg.gusari.org/static/64bit/ >> Or do really mean to checkout the code and build it myself ? > Whatever seems more appropriate to you. > (Since my next answer will be: "Please test b1ef4dc", > you should really consider to compile yourself, it > does not hurt.) > > Carl Eugen > Thanks, have a nice weekend and next week I'll reply with all the (for now) asked info :-) From cehoyos at ag.or.at Fri Oct 18 12:06:26 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 18 Oct 2013 10:06:26 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?ProRes_-=3E_1stream_=28pcm=5Fs24le=29_16c?= =?utf-8?q?h_audio_-=3E_invalid_channel_layout?= References: <5260FCF9.3030206@podiumbv.nl> <526105BA.4040506@podiumbv.nl> <5261076E.9030409@podiumbv.nl> Message-ID: Podium B.V. podiumbv.nl> writes: > have a nice weekend In case you don't want to wait: I suspect this is ticket #2899 where a workaround is described and as said, you could consider to use b1ef4dc (or to revert the offensive patch locally). A report here will still be very welcome! Carl Eugen From mailinglist at podiumbv.nl Fri Oct 18 12:24:51 2013 From: mailinglist at podiumbv.nl (Podium B.V.) Date: Fri, 18 Oct 2013 12:24:51 +0200 Subject: [FFmpeg-user] ProRes -> 1stream (pcm_s24le) 16ch audio -> invalid channel layout In-Reply-To: References: <5260FCF9.3030206@podiumbv.nl> <526105BA.4040506@podiumbv.nl> <5261076E.9030409@podiumbv.nl> Message-ID: <52610C73.602@podiumbv.nl> On 18-10-13 12:06, Carl Eugen Hoyos wrote: > > In case you don't want to wait: > I suspect this is ticket #2899 where a workaround > is described and as said, you could consider to > use b1ef4dc (or to revert the offensive patch > locally). > > A report here will still be very welcome! > > Carl Eugen > ticket #2899 has indeed similarities to my problem! I'll give this all a try and will report it back... From cehoyos at ag.or.at Fri Oct 18 16:33:49 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 18 Oct 2013 14:33:49 +0000 (UTC) Subject: [FFmpeg-user] FCPX-remuxed XAVC References: <83B55B4E-8560-47A7-B532-26E144A467C1@up4.com> Message-ID: Vincent Olivier up4.com> writes: > MOV file: Stream #0:0(und): Video: none Should be fixed, thank you for the report and the sample! Carl Eugen From vincent at up4.com Fri Oct 18 17:37:11 2013 From: vincent at up4.com (Vincent Olivier) Date: Fri, 18 Oct 2013 11:37:11 -0400 Subject: [FFmpeg-user] FCPX-remuxed XAVC In-Reply-To: References: <83B55B4E-8560-47A7-B532-26E144A467C1@up4.com> Message-ID: <5480C666-6A2B-41C9-925B-16E1973A7596@up4.com> Thanks Carl! However: I looked at your patch. The MKTAG('a', 'i', 'v', 'x') refers not only to AVC-Intra 200M 4K 24p, but also to all XAVC supported resolutions, framerates and bitrates. That's just the comment that is incorrect. The only constant is that it is (for now) 4:2:2 10-bit. But it ranges from 23.98fps 1920x1080p to 60fps 4K at 600Mbps. Thanks again for the patch. On 2013-10-18, at 10:33, Carl Eugen Hoyos wrote: > Vincent Olivier up4.com> writes: > >> MOV file: Stream #0:0(und): Video: none > > Should be fixed, thank you for the report and the sample! > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4879 bytes Desc: not available URL: From cehoyos at ag.or.at Fri Oct 18 19:38:11 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 18 Oct 2013 17:38:11 +0000 (UTC) Subject: [FFmpeg-user] FCPX-remuxed XAVC References: <83B55B4E-8560-47A7-B532-26E144A467C1@up4.com> <5480C666-6A2B-41C9-925B-16E1973A7596@up4.com> Message-ID: Vincent Olivier up4.com> writes: > The MKTAG('a', 'i', 'v', 'x') refers not only to > AVC-Intra 200M 4K 24p, but also to all XAVC > supported resolutions, framerates and bitrates. Thank you for correcting me! Should be fixed, Carl Eugen From dev at rarevision.com Fri Oct 18 20:49:13 2013 From: dev at rarevision.com (Thomas Worth) Date: Fri, 18 Oct 2013 11:49:13 -0700 Subject: [FFmpeg-user] deprecated pixel format used, make sure you did set range correctly Message-ID: When using the yuvj420p pixel format, I get the following from ffmpeg: "deprecated pixel format used, make sure you did set range correctly" However, using yuv420p along with -color_range, as instructed by pixfmt.h: AV_PIX_FMT_YUVJ420P, ///< planar YUV 4:2:0, 12bpp, full scale (JPEG), deprecated in favor of PIX_FMT_YUV420P and setting color_range Changes absolutely nothing. Of course, no ffmpeg email would be worth reading without the complete, uncut console output so without further ado, here it is: MacPro:test user$ ffmpeg -i my_awesome_movie.mov -an -vcodec libx264 -tune film -vb 12000k -pix_fmt yuv420p -color_range 0 -x264opts bframes=3:keyint=72:8x8dct=1 -y out.mkv ffmpeg version 2.0-8d9c1b3 Copyright (c) 2000-2013 the FFmpeg developers built on Jul 15 2013 21:56:22 with gcc 4.8.1 (GCC) configuration: --prefix=/Users/user/Documents/SOURCE/ffinstall --disable-doc --enable-static --disable-shared --enable-gpl --enable-nonfree --enable-version3 --enable-libfaac --enable-libfdk-aac --enable-libvpx --enable-libmp3lame --enable-libvorbis --enable-libx264 --cc=/Users/user/usr/bin/gcc --disable-vda --enable-pthreads --arch=x86_64 --target-os=darwin --extra-cflags='-I/Users/user/Documents/SOURCE/ffinstall/include -m64' --extra-ldflags='-L/Users/user/Documents/SOURCE/ffinstall/lib -static-libgcc -Wl,-arch,x86_64' libavutil 52. 39.100 / 52. 39.100 libavcodec 55. 18.102 / 55. 18.102 libavformat 55. 12.102 / 55. 12.102 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 81.101 / 3. 81.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'my_awesome_movie.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2013-10-18 17:57:28 Duration: 00:00:20.02, start: 0.000000, bitrate: 23996 kb/s Stream #0:0(eng): Video: qtrle (rle / 0x20656C72), rgb24, 1920x1080, 22457 kb/s, SAR 1920:1920 DAR 16:9, 23.98 fps, 23.98 tbr, 24k tbn, 24k tbc Metadata: creation_time : 2013-10-18 17:57:28 handler_name : Apple Alias Data Handler timecode : 00:00:00:00 Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, s16, 1536 kb/s Metadata: creation_time : 2013-10-18 17:57:28 handler_name : Apple Alias Data Handler timecode : 00:00:00:00 Stream #0:2(eng): Data: none (tmcd / 0x64636D74), 0 kb/s Metadata: creation_time : 2013-10-18 17:57:47 handler_name : Apple Alias Data Handler timecode : 00:00:00:00 [libx264 @ 0x7fd32101f200] using SAR=1/1 [libx264 @ 0x7fd32101f200] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.1 Cache64 [libx264 @ 0x7fd32101f200] profile High, level 4.0 [libx264 @ 0x7fd32101f200] 264 - core 135 - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:-1:-1 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.15 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-3 threads=12 lookahead_threads=2 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=72 keyint_min=7 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=12000 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, matroska, to 'out.mkv': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt encoder : Lavf55.12.102 Stream #0:0(eng): Video: h264 (libx264) (H264 / 0x34363248), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 12000 kb/s, 1k tbn, 23.98 tbc Metadata: creation_time : 2013-10-18 17:57:28 handler_name : Apple Alias Data Handler timecode : 00:00:00:00 Stream mapping: Stream #0:0 -> #0:0 (qtrle -> libx264) Press [q] to stop, [?] for help frame= 480 fps= 42 q=-1.0 Lsize= 472kB time=00:00:19.93 bitrate= 193.8kbits/s video:467kB audio:0kB subtitle:0 global headers:0kB muxing overhead 0.883026% [libx264 @ 0x7fd32101f200] frame I:7 Avg QP: 0.42 size: 40739 [libx264 @ 0x7fd32101f200] frame P:120 Avg QP: 0.31 size: 1090 [libx264 @ 0x7fd32101f200] frame B:353 Avg QP: 1.87 size: 176 [libx264 @ 0x7fd32101f200] consecutive B-frames: 1.5% 0.0% 4.4% 94.2% [libx264 @ 0x7fd32101f200] mb I I16..4: 92.8% 0.5% 6.6% [libx264 @ 0x7fd32101f200] mb P I16..4: 0.0% 0.0% 0.0% P16..4: 1.8% 0.0% 0.0% 0.0% 0.0% skip:98.2% [libx264 @ 0x7fd32101f200] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 0.5% 0.0% 0.0% direct: 0.1% skip:99.4% L0:61.3% L1:38.7% BI: 0.0% [libx264 @ 0x7fd32101f200] final ratefactor: -34.58 [libx264 @ 0x7fd32101f200] 8x8 transform intra:0.5% inter:41.0% [libx264 @ 0x7fd32101f200] coded y,uvDC,uvAC intra: 3.7% 0.0% 0.0% inter: 0.2% 0.0% 0.0% [libx264 @ 0x7fd32101f200] i16 v,h,dc,p: 90% 9% 2% 0% [libx264 @ 0x7fd32101f200] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 68% 28% 5% 0% 0% 0% 0% 0% 0% [libx264 @ 0x7fd32101f200] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 36% 31% 17% 4% 2% 3% 2% 3% 2% [libx264 @ 0x7fd32101f200] i8c dc,h,v,p: 100% 0% 0% 0% [libx264 @ 0x7fd32101f200] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0x7fd32101f200] ref P L0: 95.0% 0.0% 4.9% 0.1% [libx264 @ 0x7fd32101f200] ref B L0: 87.6% 12.3% 0.1% [libx264 @ 0x7fd32101f200] ref B L1: 97.8% 2.2% [libx264 @ 0x7fd32101f200] kb/s:190.99 From battistel at gmail.com Fri Oct 18 21:04:38 2013 From: battistel at gmail.com (Massimo Battistel) Date: Fri, 18 Oct 2013 21:04:38 +0200 Subject: [FFmpeg-user] Suspect memory leak in NUT muxer Message-ID: hello, with the following command line: ffmpeg -i any.mpg -vcodec rawvideo -acodec pcm_s16le -f nut - 2> NUL | ffplay - for any video file, you can experience a substantial memory leak. Same problem using directshow capture source (decklink). In about 12 hours ffmpeg passed from 40mb to 550mb and ffplay from 25mb to 380mb. I suspect the problem is the nut muxer. Using ASF (-f asf) as muxer everything is fine and no mem leak is detected. I have not tested other muxers. ffmpeg/ffplay version used for these tests: "N-55020-g768e40b" 64bit on Windows 7 64bit. "N-54960-gf3f4e13" 32bit on Windows XP 32 bit. thanks, MB From onemda at gmail.com Fri Oct 18 21:28:48 2013 From: onemda at gmail.com (Paul B Mahol) Date: Fri, 18 Oct 2013 19:28:48 +0000 Subject: [FFmpeg-user] Suspect memory leak in NUT muxer In-Reply-To: References: Message-ID: On 10/18/13, Massimo Battistel wrote: > hello, > with the following command line: > > ffmpeg -i any.mpg -vcodec rawvideo -acodec pcm_s16le -f nut - 2> NUL | > ffplay - > > for any video file, you can experience a substantial memory leak. Same > problem using directshow capture source (decklink). In about 12 hours > ffmpeg passed from 40mb to 550mb and ffplay from 25mb to 380mb. > > I suspect the problem is the nut muxer. > Using ASF (-f asf) as muxer everything is fine and no mem leak is detected. > I have not tested other muxers. > > ffmpeg/ffplay version used for these tests: > "N-55020-g768e40b" 64bit on Windows 7 64bit. > "N-54960-gf3f4e13" 32bit on Windows XP 32 bit. > > thanks, > MB > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > nut allocates memory for sync points, so more frames more memory is needed. From battistel at gmail.com Fri Oct 18 21:35:28 2013 From: battistel at gmail.com (Massimo Battistel) Date: Fri, 18 Oct 2013 21:35:28 +0200 Subject: [FFmpeg-user] Suspect memory leak in NUT muxer In-Reply-To: References: Message-ID: 2013/10/18 Paul B Mahol > On 10/18/13, Massimo Battistel wrote: > > hello, > > with the following command line: > > > > ffmpeg -i any.mpg -vcodec rawvideo -acodec pcm_s16le -f nut - 2> NUL | > > ffplay - > > > > for any video file, you can experience a substantial memory leak. Same > > problem using directshow capture source (decklink). In about 12 hours > > ffmpeg passed from 40mb to 550mb and ffplay from 25mb to 380mb. > > > > I suspect the problem is the nut muxer. > > Using ASF (-f asf) as muxer everything is fine and no mem leak is > detected. > > I have not tested other muxers. > > > > ffmpeg/ffplay version used for these tests: > > "N-55020-g768e40b" 64bit on Windows 7 64bit. > > "N-54960-gf3f4e13" 32bit on Windows XP 32 bit. > > > > thanks, > > MB > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > nut allocates memory for sync points, so more frames more memory is needed. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Do you mean that there is no mem leak and what i've experienced is the normal behaviour? Ok, not a good container for 24/7 streaming :-) So what is the advantage of using nut over other containers? thanks From onemda at gmail.com Fri Oct 18 21:50:02 2013 From: onemda at gmail.com (Paul B Mahol) Date: Fri, 18 Oct 2013 19:50:02 +0000 Subject: [FFmpeg-user] Suspect memory leak in NUT muxer In-Reply-To: References: Message-ID: On 10/18/13, Massimo Battistel wrote: > 2013/10/18 Paul B Mahol > >> On 10/18/13, Massimo Battistel wrote: >> > hello, >> > with the following command line: >> > >> > ffmpeg -i any.mpg -vcodec rawvideo -acodec pcm_s16le -f nut - 2> NUL | >> > ffplay - >> > >> > for any video file, you can experience a substantial memory leak. Same >> > problem using directshow capture source (decklink). In about 12 hours >> > ffmpeg passed from 40mb to 550mb and ffplay from 25mb to 380mb. >> > >> > I suspect the problem is the nut muxer. >> > Using ASF (-f asf) as muxer everything is fine and no mem leak is >> detected. >> > I have not tested other muxers. >> > >> > ffmpeg/ffplay version used for these tests: >> > "N-55020-g768e40b" 64bit on Windows 7 64bit. >> > "N-54960-gf3f4e13" 32bit on Windows XP 32 bit. >> > >> > thanks, >> > MB >> > _______________________________________________ >> > ffmpeg-user mailing list >> > ffmpeg-user at ffmpeg.org >> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > >> >> nut allocates memory for sync points, so more frames more memory is >> needed. >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > > Do you mean that there is no mem leak and what i've experienced is the > normal behaviour? > Ok, not a good container for 24/7 streaming :-) I guess those stuff that allocates memory could be disabled as index seek table is not very useful in streaming, and index table iirc is optional. So it should be relatively trivial to add such option to muxer, it just nobody come with idea of using nut for 24/7 streaming until now. (Better ever then never) > > So what is the advantage of using nut over other containers? > > thanks > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From vincent at up4.com Fri Oct 18 21:58:00 2013 From: vincent at up4.com (Vincent Olivier) Date: Fri, 18 Oct 2013 15:58:00 -0400 Subject: [FFmpeg-user] FCPX-remuxed XAVC In-Reply-To: References: <83B55B4E-8560-47A7-B532-26E144A467C1@up4.com> <5480C666-6A2B-41C9-925B-16E1973A7596@up4.com> Message-ID: Thanks! The fix indeed works. Vincent On 2013-10-18, at 13:38, Carl Eugen Hoyos wrote: > Vincent Olivier up4.com> writes: > >> The MKTAG('a', 'i', 'v', 'x') refers not only to >> AVC-Intra 200M 4K 24p, but also to all XAVC >> supported resolutions, framerates and bitrates. > > Thank you for correcting me! > > Should be fixed, Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4879 bytes Desc: not available URL: From battistel at gmail.com Fri Oct 18 22:26:43 2013 From: battistel at gmail.com (Massimo Battistel) Date: Fri, 18 Oct 2013 22:26:43 +0200 Subject: [FFmpeg-user] Suspect memory leak in NUT muxer In-Reply-To: References: Message-ID: I was joking when speaking about streaming, but not completely. My real need is sending both audio and video through pipes (and this is not very different from streaming in the end...). At present nut is the only container that allows me to pass audio with no problems. If you have any suggestion about a good container... My final goal is sending ffmpeg output to decklink like that: ffmpeg -i .... -f nut(or somewhat) - | bmdplay ... thanks 2013/10/18 Paul B Mahol > On 10/18/13, Massimo Battistel wrote: > > 2013/10/18 Paul B Mahol > > > >> On 10/18/13, Massimo Battistel wrote: > >> > hello, > >> > with the following command line: > >> > > >> > ffmpeg -i any.mpg -vcodec rawvideo -acodec pcm_s16le -f nut - 2> NUL | > >> > ffplay - > >> > > >> > for any video file, you can experience a substantial memory leak. Same > >> > problem using directshow capture source (decklink). In about 12 hours > >> > ffmpeg passed from 40mb to 550mb and ffplay from 25mb to 380mb. > >> > > >> > I suspect the problem is the nut muxer. > >> > Using ASF (-f asf) as muxer everything is fine and no mem leak is > >> detected. > >> > I have not tested other muxers. > >> > > >> > ffmpeg/ffplay version used for these tests: > >> > "N-55020-g768e40b" 64bit on Windows 7 64bit. > >> > "N-54960-gf3f4e13" 32bit on Windows XP 32 bit. > >> > > >> > thanks, > >> > MB > >> > _______________________________________________ > >> > ffmpeg-user mailing list > >> > ffmpeg-user at ffmpeg.org > >> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > >> > > >> > >> nut allocates memory for sync points, so more frames more memory is > >> needed. > >> _______________________________________________ > >> ffmpeg-user mailing list > >> ffmpeg-user at ffmpeg.org > >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > >> > > > > > > Do you mean that there is no mem leak and what i've experienced is the > > normal behaviour? > > Ok, not a good container for 24/7 streaming :-) > > I guess those stuff that allocates memory could be disabled > as index seek table is not very useful in streaming, and > index table iirc is optional. > > So it should be relatively trivial to add such option to muxer, > it just nobody come with idea of using nut for 24/7 streaming until now. > (Better ever then never) > > > > > So what is the advantage of using nut over other containers? > > > > 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 > -- Massimo Battistel. From onemda at gmail.com Fri Oct 18 22:58:58 2013 From: onemda at gmail.com (Paul B Mahol) Date: Fri, 18 Oct 2013 20:58:58 +0000 Subject: [FFmpeg-user] Suspect memory leak in NUT muxer In-Reply-To: References: Message-ID: On 10/18/13, Massimo Battistel wrote: > I was joking when speaking about streaming, but not completely. > > My real need is sending both audio and video through pipes (and this is not > very different from streaming in the end...). At present nut is the only > container that allows me to pass audio with no problems. If you have any > suggestion about a good container... Just open feature request on bug tracker. > > My final goal is sending ffmpeg output to decklink like that: > > ffmpeg -i .... -f nut(or somewhat) - | bmdplay ... > > thanks > > > From elliottbalsley at gmail.com Sat Oct 19 02:27:30 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Fri, 18 Oct 2013 17:27:30 -0700 Subject: [FFmpeg-user] map_channel deprecated Message-ID: I've been using ffmpeg with -map_channel to extract two channels from a multi-channel audio stream. It works, but gives this warning: -map_channel is forwarded to lavfi similarly to -af pan=0x3:c0=c0:c1=c1. [pan @ 0x7f8e9b42bde0] This syntax is deprecated. Use '|' to separate the list items. I have a hard time finding good documentation on lavfi, and don't really understand how to use it. Is there a better (non-deprecated) way of doing this? My command is ffmpeg -i 1.mov -map_channel 0.1.0 -map_channel 0.1.1 -vn 1.mp4 and I pasted the full output at http://pastebin.com/NRsur5c3 From satyagowtham.k at gmail.com Sat Oct 19 05:42:39 2013 From: satyagowtham.k at gmail.com (satya gowtham kudupudi) Date: Sat, 19 Oct 2013 09:12:39 +0530 Subject: [FFmpeg-user] What is rightway to convert rawframe into jpeg image? Message-ID: Firtst I've created a raw frame using ffmpeg -f v4l2 -s 320x240 -i /dev/video0 -f rawvideo -vcodec rawvideo -r 1 -t 1 rawframe ffmpeg version N-38332-g0a4aea6 Copyright (c) 2000-2013 the FFmpeg developers built on Sep 30 2013 19:27:13 with gcc 4.7 (Ubuntu/Linaro 4.7.3-1ubuntu1) configuration: --prefix=/home/gowtham/NetBeansProjects/remotedevicecontroller/ffmpeg_build --extra-cflags=-I/home/gowtham/NetBeansProjects/remotedevicecontroller/ffmpeg_build/include --extra-ldflags=-L/home/gowtham/NetBeansProjects/remotedevicecontroller/ffmpeg_build/lib --bindir=/home/gowtham/NetBeansProjects/remotedevicecontroller/ffmpeg_build/bin --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree libavutil 52. 46.100 / 52. 46.100 libavcodec 55. 33.101 / 55. 33.101 libavformat 55. 18.104 / 55. 18.104 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 88.100 / 3. 88.100 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, video4linux2,v4l2, from '/dev/video0': Duration: N/A, start: 1335.083679, bitrate: 36864 kb/s Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 320x240, 36864 kb/s, 30 fps, 30 tbr, 1000k tbn, 1000k tbc Output #0, rawvideo, to 'rawframe': Metadata: encoder : Lavf55.18.104 Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 320x240, q=2-31, 200 kb/s, 90k tbn, 1 tbc Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> rawvideo) Press [q] to stop, [?] for help frame= 1 fps=0.0 q=0.0 Lsize= 150kB time=00:00:01.00 bitrate=1228.8kbits/s video:150kB audio:0kB subtitle:0 global headers:0kB muxing overhead 0.000000% I tried converting rawframe to jpeg image using $ ffmpeg -f rawvideo -vcodec rawvideo -s 320x240 -r 1 -pix_fmt yuv422p -i rawframe -updatefirst 1 rawframe.jpg ffmpeg version N-38332-g0a4aea6 Copyright (c) 2000-2013 the FFmpeg developers built on Sep 30 2013 19:27:13 with gcc 4.7 (Ubuntu/Linaro 4.7.3-1ubuntu1) configuration: --prefix=/home/gowtham/NetBeansProjects/remotedevicecontroller/ffmpeg_build --extra-cflags=-I/home/gowtham/NetBeansProjects/remotedevicecontroller/ffmpeg_build/include --extra-ldflags=-L/home/gowtham/NetBeansProjects/remotedevicecontroller/ffmpeg_build/lib --bindir=/home/gowtham/NetBeansProjects/remotedevicecontroller/ffmpeg_build/bin --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree libavutil 52. 46.100 / 52. 46.100 libavcodec 55. 33.101 / 55. 33.101 libavformat 55. 18.104 / 55. 18.104 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 88.100 / 3. 88.100 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 [rawvideo @ 0xa0a6120] Estimating duration from bitrate, this may be inaccurate Input #0, rawvideo, from 'rawframe': Duration: 00:00:01.00, start: 0.000000, bitrate: 1228 kb/s Stream #0:0: Video: rawvideo (Y42B / 0x42323459), yuv422p, 320x240, 1228 kb/s, 1 tbr, 1 tbn, 1 tbc [swscaler @ 0xa094060] deprecated pixel format used, make sure you did set range correctly Output #0, image2, to 'rawframe.jpg': Metadata: encoder : Lavf55.18.104 Stream #0:0: Video: mjpeg, yuvj422p, 320x240, q=2-31, 200 kb/s, 90k tbn, 1 tbc Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> mjpeg) Press [q] to stop, [?] for help [swscaler @ 0xa094060] Warning: data is not aligned! This can lead to a speedloss frame= 1 fps=0.0 q=0.0 Lsize=N/A time=00:00:01.00 bitrate=N/A But the rawframe.jpg is cluttered. I'm attaching the rawframe.jpg. https://lh3.googleusercontent.com/-qUIhUMjFJog/UmH_OjWPp9I/AAAAAAAAAvE/8onctMkRB10/w320-h240-no/rawframe.jpgis the link for rawframe.jpg if the attachment fails. What mistake I've done? What is the right way of converting raw frame to jpeg image. -- Gowtham -------------- next part -------------- A non-text attachment was scrubbed... Name: rawframe.jpg Type: image/jpeg Size: 50648 bytes Desc: not available URL: From satyagowtham.k at gmail.com Sat Oct 19 06:49:55 2013 From: satyagowtham.k at gmail.com (satya gowtham kudupudi) Date: Sat, 19 Oct 2013 10:19:55 +0530 Subject: [FFmpeg-user] What is rightway to convert rawframe into jpeg image? In-Reply-To: References: Message-ID: I'm sry I got confused between yuv422p and yuyv422 Thank you! -- *Gowtham* From junjiepang at gmail.com Sat Oct 19 06:52:54 2013 From: junjiepang at gmail.com (jjpang) Date: Fri, 18 Oct 2013 21:52:54 -0700 (PDT) Subject: [FFmpeg-user] Qns regarding compiling c++ code with ffmpeg In-Reply-To: References: <5F5A0287-51D7-457F-8458-66AFE74E0186@gmail.com> Message-ID: <1382158374087-4661921.post@n4.nabble.com> Thank you very much for your input ! =D Seems like I need to link a lot of libs... gcc -o sample_vid_encoder sample_vid_encoder.cpp -L/$HOME/ffmpeg_build/lib -lavformat -lavcodec -lavdevice -lavfilter -lfdk-aac -lmp3lame -lopus -lpostproc -lswresample -lswscale -lvorbis -lvorbisenc -lvorbisfile -logg -lvpx -lx264 -lyasm -lavutil -lm -lz -pthread -ldl -lstdc++ -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Qns-regarding-compiling-c-code-with-ffmpeg-tp4661799p4661921.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From junjiepang at gmail.com Sat Oct 19 07:24:58 2013 From: junjiepang at gmail.com (jjpang) Date: Fri, 18 Oct 2013 22:24:58 -0700 (PDT) Subject: [FFmpeg-user] Encode jpegs into .mp4 (xh264) Message-ID: <1382160298074-4661922.post@n4.nabble.com> Hi, I have been trying to find out how to convert a series of JPEGs into a .mp4 container with h264 encoding in C/C++. However, I could not find a "sample/tutorial" online. I've encountered some problems and wondering if anyone could help out this newbie =) 1) How do i prepare/load the image for encoding ? 2) I've tried changing the sample program: (https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/decoding_encoding.c) to pass AV_CODEC_ID_MPEG4 as codec... However the output file seems unplayable... -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Encode-jpegs-into-mp4-xh264-tp4661922.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From satyagowtham.k at gmail.com Sat Oct 19 14:01:28 2013 From: satyagowtham.k at gmail.com (satya gowtham kudupudi) Date: Sat, 19 Oct 2013 17:31:28 +0530 Subject: [FFmpeg-user] How to catch disk full error? Message-ID: What error does ffmpeg throws in linux on disk full? and how to catch it? I am writing a bash script in which it frees up the disk space if ffmpeg exits due to in adequate disk space. On what should I count on? Can I count on exit code? -- *Gowtham* From bostjan.strojan at gmail.com Sat Oct 19 14:23:39 2013 From: bostjan.strojan at gmail.com (=?UTF-8?Q?Bo=C5=A1tjan_Strojan?=) Date: Sat, 19 Oct 2013 14:23:39 +0200 Subject: [FFmpeg-user] Conversion to prores, close encode every N minutes, how? Message-ID: Converting a very long stream, 2 hours and more into more editable format like prores. My question is: Is it possible to close the encode every N minutes and start a new one (generating multiple files), without restarting the process, if yes, how? tia, b. From george at nsup.org Sun Oct 20 09:55:13 2013 From: george at nsup.org (Nicolas George) Date: Sun, 20 Oct 2013 09:55:13 +0200 Subject: [FFmpeg-user] map_channel deprecated In-Reply-To: References: Message-ID: <20131020075513.GA396@phare.normalesup.org> Le septidi 27 vend?miaire, an CCXXII, Elliott Balsley a ?crit?: > I've been using ffmpeg with -map_channel to extract two channels from a > multi-channel audio stream. It works, but gives this warning: > -map_channel is forwarded to lavfi similarly to -af pan=0x3:c0=c0:c1=c1. > [pan @ 0x7f8e9b42bde0] This syntax is deprecated. Use '|' to separate the > list items. Do not worry about it: it is just an artifact of the way map_channel is implemented internally that nobody has bothered to update until now, but it has no consequence. > I have a hard time finding good documentation on lavfi, and don't really > understand how to use it. What is wrong with http://ffmpeg.org/ffmpeg-filters.html ? Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 836 bytes Desc: Digital signature URL: From krueger at lesspain.de Sun Oct 20 13:06:29 2013 From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=) Date: Sun, 20 Oct 2013 13:06:29 +0200 Subject: [FFmpeg-user] Trac file size limit Message-ID: Hi, would it be a problem to extend that just a little? 2.5 Meg seems really small unless you never want video to be attached using that mechanism. Of course I will FTP my 5 Meg video sample if I have to but having the upload function would be a lot more convenient for users. If maybe for administration reasons, it is discouraged to use this function to upload sample files then it would be great to document this somewhere. Cheers, Robert From cehoyos at ag.or.at Sun Oct 20 13:55:42 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 20 Oct 2013 11:55:42 +0000 (UTC) Subject: [FFmpeg-user] Trac file size limit References: Message-ID: Robert Kr?ger lesspain.de> writes: > would it be a problem to extend that just a little? I am sorry but I am strictly against this. > 2.5 Meg seems really small unless you never want > video to be attached using that mechanism. This surprises me, nearly all non-desync related issues can be shown with smaller samples, issues related to A/V synchronisation will practically always need much more than 5MB. (Typically 50 - 100MB) > Of course I will FTP my 5 Meg video sample if I > have to but having the upload function would be > a lot more convenient for users. You may use http://www.datafilehost.com/ which may be slightly more convenient than ftp. > If maybe for administration reasons, it is > discouraged to use this function to upload sample > files then it would be great to document this > somewhere. It is definitely not discouraged, on the contrary. Carl Eugen From krueger at lesspain.de Sun Oct 20 14:10:37 2013 From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=) Date: Sun, 20 Oct 2013 14:10:37 +0200 Subject: [FFmpeg-user] Trac file size limit In-Reply-To: References: Message-ID: On Sun, Oct 20, 2013 at 1:55 PM, Carl Eugen Hoyos wrote: > Robert Kr?ger lesspain.de> writes: > >> would it be a problem to extend that just a little? > > I am sorry but I am strictly against this. > >> 2.5 Meg seems really small unless you never want >> video to be attached using that mechanism. > > This surprises me, nearly all non-desync related > issues can be shown with smaller samples, issues > related to A/V synchronisation will practically > always need much more than 5MB. > (Typically 50 - 100MB) > >> Of course I will FTP my 5 Meg video sample if I >> have to but having the upload function would be >> a lot more convenient for users. > > You may use http://www.datafilehost.com/ which may > be slightly more convenient than ftp. > >> If maybe for administration reasons, it is >> discouraged to use this function to upload sample >> files then it would be great to document this >> somewhere. > > It is definitely not discouraged, on the contrary. > > Carl Eugen I am not quite sure I understand your reasoning completely, but you're running the bug tracker so you decide. I will FTP files over 2.5 Meg then. From werner.robitza at gmail.com Sun Oct 20 21:08:07 2013 From: werner.robitza at gmail.com (Werner Robitza) Date: Sun, 20 Oct 2013 21:08:07 +0200 Subject: [FFmpeg-user] Conversion to prores, close encode every N minutes, how? In-Reply-To: References: Message-ID: On Sat, Oct 19, 2013 at 2:23 PM, Bo?tjan Strojan wrote: > > Converting a very long stream, 2 hours and more into more editable format > like prores. > > My question is: Is it possible to close the encode every N minutes and > start a new one (generating multiple files), without restarting the > process, if yes, how? You can achieve this with the segment muxer. Refer to the documentation for some examples: http://ffmpeg.org/ffmpeg-formats.html#segment_002c-stream_005fsegment_002c-ssegment From ronag89 at gmail.com Mon Oct 21 00:00:12 2013 From: ronag89 at gmail.com (Robert Nagy) Date: Mon, 21 Oct 2013 00:00:12 +0200 Subject: [FFmpeg-user] Issues with converting DNxHD MOV to IMX50 MXF D10 Message-ID: I have an DNxHD file in a mov container with a timecode track I want to convert into IMX50 mxf D10. I'm having some issues with copying over the timecode track from the source file, I get a " track 2: could not find essence container ul, codec not currently supported in container" error. Also, when encoding the video I keep getting "rc buffer underflow max bitrate possibly too small or try trellis with large lmax or increase qmax". What am I doing wrong? C:\Users\asd\Desktop\FFmbc-0.7rc8-win64>ffmpeg -i "M:\24\ 130610.mov" -map 0:1 -map 0:2 -map 0:0 -t 10 -r 25 -pix_fmt yuv422p -codec:v mpeg2video -minrate:v 50000k -maxrate:v 50000k -b:v 50000k -intra -top 1 -flags +ildct+low_delay -intra_vlc 1 -non_linear_quant 1 -dc 10 -ps 1 -qmin:v 1 -qmax: v 3 -bufsize:v 2000000 -rc_init_occupancy 2000000 -rc_buf_aggressivity 0.25 -cod ec:a pcm_s16le -ar 48000 -ac 8 -map_channel 0.2.0 -map_channel 0.2.1 -map_channe l 0.2.2 -map_channel 0.2.3 -map_channel 0.2.4 -map_channel 0.2.5 -map_channel 0. 2.6 -map_channel 0.2.7 -codec:d copy -f mxf_d10 "C:\24\130610.mxf" ffmpeg version N-57245-gf6b56b1 Copyright (c) 2000-2013 the FFmpeg developers built on Oct 18 2013 18:08:17 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 47.100 / 52. 47.100 libavcodec 55. 37.101 / 55. 37.101 libavformat 55. 19.102 / 55. 19.102 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.101 / 3. 88.101 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 [mov,mp4,m4a,3gp,3g2,mj2 @ 000000000254be00] Stream #0: not enough frames to est imate rate; consider increasing probesize [mov,mp4,m4a,3gp,3g2,mj2 @ 000000000254be00] Stream #2: not enough frames to est imate rate; consider increasing probesize Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'M:\24\130610.mov ': Metadata: creation_time : 2020-11-08 08:14:22 timecode : 11:26:52:03 Duration: 02:27:34.44, start: 0.000000, bitrate: 196915 kb/s Stream #0:0(eng): Data: none (tmcd / 0x64636D74) (default) Metadata: creation_time : 2020-11-08 08:14:22 handler_name : Apple Alias Data Handler timecode : 11:26:52:03 Stream #0:1(eng): Video: prores (apch / 0x68637061), yuv422p10le, 1920x1080, 178371 kb/s, SAR 1:1 DAR 16:9, 25 fps, 25 tbr, 2500 tbn, 2500 tbc (default) Metadata: creation_time : 2020-11-08 08:14:22 handler_name : Apple Alias Data Handler timecode : 11:26:52:03 Stream #0:2(eng): Audio: pcm_s24le (lpcm / 0x6D63706C), 48000 Hz, 16 channel s, s32, 18432 kb/s (default) Metadata: creation_time : 2020-11-08 08:14:22 handler_name : Apple Alias Data Handler timecode : 11:26:52:03 File 'C:\24\130610.mxf' already exists. Overwrite ? [y/N] y -map_channel is forwarded to lavfi similarly to -af pan=0x63f:c0=c0:c1=c1:c2=c2: c3=c3:c4=c4:c5=c5:c6=c6:c7=c7. [pan @ 00000000025a52e0] This syntax is deprecated. Use '|' to separate the list items. [pan @ 00000000025a52e0] Pure channel mapping detected: M M M M M M M M [mxf_d10 @ 0000000004b30420] track 2: could not find essence container ul, codec not currently supported in container Output #0, mxf_d10, to 'C:\24\130610.mxf': Metadata: timecode : 11:26:52:03 encoder : Lavf55.19.102 Stream #0:0(eng): Video: mpeg2video, yuv422p, 1920x1080 [SAR 1:1 DAR 16:9], q=1-3, 50000 kb/s, 25 tbn, 25 tbc (default) Metadata: creation_time : 2020-11-08 08:14:22 handler_name : Apple Alias Data Handler timecode : 11:26:52:03 Stream #0:1(eng): Audio: pcm_s16le, 48000 Hz, 7.1, s16, 6144 kb/s (default) Metadata: creation_time : 2020-11-08 08:14:22 handler_name : Apple Alias Data Handler timecode : 11:26:52:03 Stream #0:2(eng): Data: none (tmcd / 0x64636D74) (default) Metadata: creation_time : 2020-11-08 08:14:22 handler_name : Apple Alias Data Handler timecode : 11:26:52:03 Stream mapping: Stream #0:1 -> #0:0 (prores -> mpeg2video) Stream #0:2 -> #0:1 (pcm_s24le -> pcm_s16le) Stream #0:0 -> #0:2 (copy) Could not write header for output file #0 (incorrect codec parameters ?): Error number -1 occurred From lulebo at gmail.com Mon Oct 21 11:40:46 2013 From: lulebo at gmail.com (Carl Lindqvist) Date: Mon, 21 Oct 2013 11:40:46 +0200 Subject: [FFmpeg-user] Multiple outputs worse performance than multiple instances In-Reply-To: References: Message-ID: Hello. I have done some more testing and been trying to squeeze out more performance from my encodes. My main goal is to do reading/decoding/deinterlacing/cutting only once, and do resize and encode seperately. Here is the current command (put in a bat file): ffmpeg -i %1 -filter_complex "[0:0] yadif=0,select='-between(t,5,20)-between(t,40,52)',setpts=N/FRAME_RATE/TB,split=5 [a1] [b1] [c1] [d1] [e1];[0:1] aselect='-between(t,5,20)-between(t,40,52)',asetpts=N/SR/TB,asplit=5 [a2] [b2] [c2] [d2] [e2]" -y \ -map "[a1]" -map "[a2]" -s 128:72 -c:v libx264 -preset veryfast -x264opts keyint=75:min-keyint=75:no-scenecut -b:v 98k -minrate 98k -maxrate 98k -bufsize 196k -profile:v baseline -aspect 16:9 -c:a libvo_aacenc -b:a 32k -ac 2 %~n1130.mp4 \ -map "[b1]" -map "[b2]" -s 384:216 -c:v libx264 -preset veryfast -x264opts keyint=75:min-keyint=75:no-scenecut -b:v 252k -minrate 252k -maxrate 252k -bufsize 504k -profile:v baseline -aspect 16:9 -c:a libvo_aacenc -b:a 48k -ac 2 %~n1300.mp4 \ -map "[c1]" -map "[c2]" -s 640:360 -c:v libx264 -preset veryfast -x264opts keyint=75:min-keyint=75:no-scenecut -b:v 704k -minrate 704k -maxrate 704k -bufsize 1408k -profile:v baseline -aspect 16:9 -c:a libvo_aacenc -b:a 96k -ac 2 %~n1800.mp4 \ -map "[d1]" -map "[d2]" -s 768:432 -c:v libx264 -preset veryfast -x264opts keyint=75:min-keyint=75:no-scenecut -b:v 1404k -minrate 1404k -maxrate 1404k -bufsize 2808k -profile:v main -aspect 16:9 -c:a libvo_aacenc -b:a 96k -ac 2 %~n11500.mp4 \ -map "[e1]" -map "[e2]" -s 1024:576 -c:v libx264 -preset veryfast -x264opts keyint=75:min-keyint=75:no-scenecut -b:v 2244k -minrate 2244k -maxrate 2244k -bufsize 4488k -profile:v main -aspect 16:9 -c:a libvo_aacenc -b:a 256k -ac 2 %~n12500.mp4 Everything works perfectly, but cpu usage on the very fast server (16 cores) is quite bad. I can set -preset slow and get more cpu usage and get better cpu usage. This leads me to believe decoding is either the culprit or I am doing something really stupid somewhere. I must confess that I do not fully understand the filter_complex yet, but I think I have got a hang of it. I am just not sure how it is handled behind the scenes. What is a good way to test decoding speed/cpu usage independantly? And can I do something better in the command above? Best Regards Carl L 2013/10/12 Carl Eugen Hoyos : > Carl Lindqvist gmail.com> writes: > >> My results are very bad. > > 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 onemda at gmail.com Mon Oct 21 13:01:18 2013 From: onemda at gmail.com (Paul B Mahol) Date: Mon, 21 Oct 2013 11:01:18 +0000 Subject: [FFmpeg-user] Trac file size limit In-Reply-To: References: Message-ID: On 10/20/13, Robert Krueger wrote: > On Sun, Oct 20, 2013 at 1:55 PM, Carl Eugen Hoyos wrote: >> Robert Krueger lesspain.de> writes: >> >>> would it be a problem to extend that just a little? >> >> I am sorry but I am strictly against this. >> >>> 2.5 Meg seems really small unless you never want >>> video to be attached using that mechanism. >> >> This surprises me, nearly all non-desync related >> issues can be shown with smaller samples, issues >> related to A/V synchronisation will practically >> always need much more than 5MB. >> (Typically 50 - 100MB) >> >>> Of course I will FTP my 5 Meg video sample if I >>> have to but having the upload function would be >>> a lot more convenient for users. >> >> You may use http://www.datafilehost.com/ which may >> be slightly more convenient than ftp. >> >>> If maybe for administration reasons, it is >>> discouraged to use this function to upload sample >>> files then it would be great to document this >>> somewhere. >> >> It is definitely not discouraged, on the contrary. >> >> Carl Eugen > > I am not quite sure I understand your reasoning completely, but you're > running the bug tracker so you decide. I will FTP files over 2.5 Meg > then. Carl is not running the bug tracker, and even if he does, he is not the only one to decide about size limit. From lulebo at gmail.com Mon Oct 21 14:36:47 2013 From: lulebo at gmail.com (Carl Lindqvist) Date: Mon, 21 Oct 2013 14:36:47 +0200 Subject: [FFmpeg-user] Multiple outputs worse performance than multiple instances In-Reply-To: References: Message-ID: 2013/10/21 Carl Lindqvist : > Hello. I have done some more testing and been trying to squeeze out > more performance from my encodes. My main goal is to do > reading/decoding/deinterlacing/cutting only once, and do resize and > encode seperately. Here is the current command (put in a bat file): > > ffmpeg -i %1 -filter_complex "[0:0] > yadif=0,select='-between(t,5,20)-between(t,40,52)',setpts=N/FRAME_RATE/TB,split=5 > [a1] [b1] [c1] [d1] [e1];[0:1] > aselect='-between(t,5,20)-between(t,40,52)',asetpts=N/SR/TB,asplit=5 > [a2] [b2] [c2] [d2] [e2]" -y \ > -map "[a1]" -map "[a2]" -s 128:72 -c:v libx264 -preset veryfast > -x264opts keyint=75:min-keyint=75:no-scenecut -b:v 98k -minrate 98k > -maxrate 98k -bufsize 196k -profile:v baseline -aspect 16:9 -c:a > libvo_aacenc -b:a 32k -ac 2 %~n1130.mp4 \ > -map "[b1]" -map "[b2]" -s 384:216 -c:v libx264 -preset veryfast > -x264opts keyint=75:min-keyint=75:no-scenecut -b:v 252k -minrate 252k > -maxrate 252k -bufsize 504k -profile:v baseline -aspect 16:9 -c:a > libvo_aacenc -b:a 48k -ac 2 %~n1300.mp4 \ > -map "[c1]" -map "[c2]" -s 640:360 -c:v libx264 -preset veryfast > -x264opts keyint=75:min-keyint=75:no-scenecut -b:v 704k -minrate 704k > -maxrate 704k -bufsize 1408k -profile:v baseline -aspect 16:9 -c:a > libvo_aacenc -b:a 96k -ac 2 %~n1800.mp4 \ > -map "[d1]" -map "[d2]" -s 768:432 -c:v libx264 -preset veryfast > -x264opts keyint=75:min-keyint=75:no-scenecut -b:v 1404k -minrate > 1404k -maxrate 1404k -bufsize 2808k -profile:v main -aspect 16:9 -c:a > libvo_aacenc -b:a 96k -ac 2 %~n11500.mp4 \ > -map "[e1]" -map "[e2]" -s 1024:576 -c:v libx264 -preset veryfast > -x264opts keyint=75:min-keyint=75:no-scenecut -b:v 2244k -minrate > 2244k -maxrate 2244k -bufsize 4488k -profile:v main -aspect 16:9 -c:a > libvo_aacenc -b:a 256k -ac 2 %~n12500.mp4 > > Everything works perfectly, but cpu usage on the very fast server (16 > cores) is quite bad. I can set -preset slow and get more cpu usage and > get better cpu usage. > > This leads me to believe decoding is either the culprit or I am doing > something really stupid somewhere. I must confess that I do not fully > understand the filter_complex yet, but I think I have got a hang of > it. I am just not sure how it is handled behind the scenes. > > What is a good way to test decoding speed/cpu usage independantly? And > can I do something better in the command above? > > Best Regards > Carl L > > 2013/10/12 Carl Eugen Hoyos : >> Carl Lindqvist gmail.com> writes: >> >>> My results are very bad. >> >> 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 Sorry for the previous top post. Gmail is giving me a headache.. I did a small typo/brain fart, what I meant to say was: Everything works perfectly, but cpu usage on the very fast server (16 cores) is quite bad. I can set -preset slow and get the same fps encoding speed, but with higher cpu usage. Best regards Carl L From ib at wupperonline.de Mon Oct 21 16:59:04 2013 From: ib at wupperonline.de (=?ISO-8859-1?Q?Ingo=20Br=FCckl?=) Date: Mon, 21 Oct 2013 16:59:04 +0200 Subject: [FFmpeg-user] Blurry artefacts during playback Message-ID: <52654197.45d9da0e.bm001@wupperonline.de> After updating FFmpeg yesterday, I'm experiencing a huge amount of blurry artefacts during playback with scenes where there is some motion in it (as if the difference information from the last frame was corrupt). This affects several codecs like mpeg4/XVID and h264. Today I took pleasure into bisecting it down which was fun due to all the recent sse2 instruction for sse CPUs bugs which first of all caused my builds crashing mostly. Finally: face578d56c2d1375e40d5e2a28acc122132bc55 is the first bad commit commit face578d56c2d1375e40d5e2a28acc122132bc55 Author: Ronald S. Bultje Date: Fri Sep 20 08:01:19 2013 -0400 Rewrite emu_edge functions to have separate src/dst_stride arguments. This allows supporting files for which the image stride is smaller than the max. block size + number of subpel mc taps, e.g. a 64x64 VP9 file or a 16x16 VP8 file with -fflags +emu_edge. :040000 040000 887c5e644a8924a40c1e5e056ba20fa3495e5d18 e7d581f402be299bdf3fe53e878b5a1aaa8bd6ad M libavcodec Ingo From cehoyos at ag.or.at Mon Oct 21 17:04:45 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 21 Oct 2013 15:04:45 +0000 (UTC) Subject: [FFmpeg-user] Blurry artefacts during playback References: <52654197.45d9da0e.bm001@wupperonline.de> Message-ID: Ingo Br?ckl wupperonline.de> writes: > face578d56c2d1375e40d5e2a28acc122132bc55 is the first bad commit Thank you for doing the bisect, but if you don't intend to send a patch, please point to a sample. Carl Eugen From ib at wupperonline.de Mon Oct 21 17:53:09 2013 From: ib at wupperonline.de (=?ISO-8859-1?Q?Ingo=20Br=FCckl?=) Date: Mon, 21 Oct 2013 17:53:09 +0200 Subject: [FFmpeg-user] Blurry artefacts during playback In-Reply-To: Message-ID: <52654ebe.3ade4e96.bm000@wupperonline.de> Carl Eugen Hoyos wrote on Mon, 21 Oct 2013 15:04:45 +0000 (UTC): > Ingo Br?ckl wupperonline.de> writes: >> face578d56c2d1375e40d5e2a28acc122132bc55 is the first bad commit > Thank you for doing the bisect, but if you don't intend to > send a patch, please point to a sample. Get V-codecs/h264/hbc9.avi from mplayerhq samples and the screenshot artefacts.png which was taken at approx. 1:18 min from upload.mplayerhq.hu incoming. It seems that the artefacts are intruding from the edges and that they are the huger the faster the scene motion is. Ingo From ib at wupperonline.de Mon Oct 21 18:05:38 2013 From: ib at wupperonline.de (=?ISO-8859-1?Q?Ingo=20Br=FCckl?=) Date: Mon, 21 Oct 2013 18:05:38 +0200 Subject: [FFmpeg-user] strip doesn't work Message-ID: <52655121.5af7e7a4.bm000@wupperonline.de> CC libswscale/utils.o YASM libswscale/x86/input.o STRIP libswscale/x86/input.o /bin/sh: 1: -wN: not found Apparently commit e52567c2954f627d420b30f75f71af2f2e4afe80. Ingo From gasto5 at hotmail.com Mon Oct 21 18:20:35 2013 From: gasto5 at hotmail.com (carlos gabriel hasbun comandari) Date: Mon, 21 Oct 2013 10:20:35 -0600 Subject: [FFmpeg-user] MP4 video showing frame delay. Message-ID: Why does my video halt on a frame when playing it with VLC, while playing it with Media Player Classic it plays correctly but with a squashed resolution (aspect ration with the height diminished) These are the parameters of the video created with FFMPEG N-45739-g04bf2e7: General Complete name : C:\Users\C. Gabriel H.C\Videos\FinalGIFtoAVIenhanced.mp4 Format : MPEG-4 Format profile : Base Media Codec ID : isom File size : 35.2 MiB Duration : 7mn 42s Overall bit rate mode : Variable Overall bit rate : 638 Kbps Writing application : Lavf54.33.100 Video ID : 1 Format : AVC Format/Info : Advanced Video Codec Format profile : High 4:4:4 Predictive at L4.0 Format settings, CABAC : Yes Format settings, ReFrames : 4 frames Codec ID : avc1 Codec ID/Info : Advanced Video Coding Duration : 7mn 42s Bit rate : 512 Kbps Width : 1 680 pixels Height : 1 050 pixels Display aspect ratio : 1.600 Frame rate mode : Constant Frame rate : 5.000 fps Color space : YUV Chroma subsampling : 4:4:4 Bit depth : 8 bits Scan type : Progressive Bits/(Pixel*Frame) : 0.058 Stream size : 28.0 MiB (80%) Writing library : x264 core 128 r2216 198a7ea Encoding settings : 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=4 / threads=6 / lookahead_threads=1 / 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=5 / scenecut=40 / intra_refresh=0 / rc_lookahead=40 / rc=abr / mbtree=1 / bitrate=512 / ratetol=1.0 / qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 / ip_ratio=1.40 / aq=1:1.00 Audio ID : 2 Format : Ogg Codec ID : DD Duration : 7mn 42s Bit rate mode : Variable Bit rate : 123 Kbps Maximum bit rate : 128 Kbps Channel(s) : 2 channels Sampling rate : 48.0 KHz Stream size : 6.80 MiB (19%) The ffmpeg was called from the command line like so: C:\Users\C. Gabriel H.C\Videos>ffmpeg -r 5 -f image2 -i "J:\Screencasts\GIMP\GAP\GIFtoAVI\video\GifToAvi\BMP\GifToAviTut_%06d.bmp" -i "J:\Screencasts\GIMP\GAP\GIFtoAV I\audio\Export\GIMPGAP_GIFtoAVI_enhanced2.flac" -map 0 -map 1 -r 5 -b:v 512k -b:a 128k -acodec libvorbis -ac 1 -ar 48000 FinalGIFtoAVIenhanced.mp4 By the way, once uploaded to YouTube, it plays well(on the Flash, HTML5 and VLC players). These are the specs of the YouTube-encoded file: General Complete name : C:\Users\C. Gabriel H.C\Downloads\FinalGIFtoAVIenhanced (HD-720p).mp4 Format : MPEG-4 Format profile : Base Media / Version 2 Codec ID : mp42 File size : 41.5 MiB Duration : 7mn 42s Overall bit rate mode : Variable Overall bit rate : 752 Kbps Encoded date : UTC 2013-10-20 13:48:06 Tagged date : UTC 2013-10-20 13:48:06 gsst : 0 gstd : 462656 gssd : B4A7DA864MH1382366082235072 gshh : r14---sn-h5q7eney.c.youtube.com Video ID : 1 Format : AVC Format/Info : Advanced Video Codec Format profile : High at L3.1 Format settings, CABAC : Yes Format settings, ReFrames : 1 frame Format settings, GOP : M=1, N=30 Codec ID : avc1 Codec ID/Info : Advanced Video Coding Duration : 7mn 42s Bit rate : 607 Kbps Maximum bit rate : 3 905 Kbps Width : 1 152 pixels Height : 720 pixels Display aspect ratio : 1.600 Frame rate mode : Constant Frame rate : 6.000 fps Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Progressive Bits/(Pixel*Frame) : 0.122 Stream size : 33.4 MiB (81%) Tagged date : UTC 2013-10-20 13:48:07 Audio ID : 2 Format : AAC Format/Info : Advanced Audio Codec Format profile : LC Codec ID : 40 Duration : 7mn 42s Bit rate mode : Variable Bit rate : 144 Kbps Maximum bit rate : 150 Kbps Channel(s) : 1 channel Channel positions : Front: C Sampling rate : 44.1 KHz Compression mode : Lossy Stream size : 7.94 MiB (19%) Title : IsoMedia File Produced by Google, 5-11-2011 Encoded date : UTC 2013-10-20 13:48:06 Tagged date : UTC 2013-10-20 13:48:07 From cehoyos at ag.or.at Mon Oct 21 20:12:48 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 21 Oct 2013 18:12:48 +0000 (UTC) Subject: [FFmpeg-user] Blurry artefacts during playback References: <52654ebe.3ade4e96.bm000@wupperonline.de> Message-ID: Ingo Br?ckl wupperonline.de> writes: > > Thank you for doing the bisect, but if you don't intend to > > send a patch, please point to a sample. > > Get V-codecs/h264/hbc9.avi I tested the following with todays git head: ffplay hbc9.avi shows no artefacts here. ffmpeg -i hbc9.avi -an crc - shows identical output as FFmpeg 2.0. How can I reproduce the problem? Carl Eugen From ib at wupperonline.de Mon Oct 21 20:27:13 2013 From: ib at wupperonline.de (=?ISO-8859-1?Q?Ingo=20Br=FCckl?=) Date: Mon, 21 Oct 2013 20:27:13 +0200 Subject: [FFmpeg-user] Blurry artefacts during playback In-Reply-To: Message-ID: <5265733e.56683c3d.bm000@wupperonline.de> Carl Eugen Hoyos wrote on Mon, 21 Oct 2013 18:12:48 +0000 (UTC): > Ingo Br?ckl wupperonline.de> writes: >> > Thank you for doing the bisect, but if you don't intend to >> > send a patch, please point to a sample. >> >> Get V-codecs/h264/hbc9.avi > I tested the following with todays git head: > ffplay hbc9.avi shows no artefacts here. > ffmpeg -i hbc9.avi -an crc - shows identical > output as FFmpeg 2.0. > How can I reproduce the problem? So far, my problems with FFmpeg in the past (as well as my problems during the bisect) were because of my old CPU (AMD Athlon XP 2400+). Could this commit break something for SSE/MMX-only 32-bit CPUs? Ingo From cehoyos at ag.or.at Mon Oct 21 20:41:20 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 21 Oct 2013 18:41:20 +0000 (UTC) Subject: [FFmpeg-user] Blurry artefacts during playback References: <5265733e.56683c3d.bm000@wupperonline.de> Message-ID: Ingo Br?ckl wupperonline.de> writes: > So far, my problems with FFmpeg in the past (as > well as my problems during the bisect) were > because of my old CPU (AMD Athlon XP 2400+). Could you test the following? $ ffplay -cpuflags 0 hbc9.avi $ ffplay -cpuflags mmx hbc9.avi $ ffplay -cpuflags sse hbc9.avi $ ffplay -cpuflags 3dnow hbc9.avi $ ffplay -cpuflags 3dnow+3dnowext hbc9.avi Carl Eugen From ib at wupperonline.de Mon Oct 21 21:48:02 2013 From: ib at wupperonline.de (=?ISO-8859-1?Q?Ingo=20Br=FCckl?=) Date: Mon, 21 Oct 2013 21:48:02 +0200 Subject: [FFmpeg-user] Blurry artefacts during playback In-Reply-To: Message-ID: <526584f2.30ab263c.bm000@wupperonline.de> Carl Eugen Hoyos wrote on Mon, 21 Oct 2013 18:41:20 +0000 (UTC): > Ingo Br?ckl wupperonline.de> writes: >> So far, my problems with FFmpeg in the past (as >> well as my problems during the bisect) were >> because of my old CPU (AMD Athlon XP 2400+). > Could you test the following? > $ ffplay -cpuflags 0 hbc9.avi > $ ffplay -cpuflags mmx hbc9.avi > $ ffplay -cpuflags sse hbc9.avi > $ ffplay -cpuflags 3dnow hbc9.avi > $ ffplay -cpuflags 3dnow+3dnowext hbc9.avi Is there a possibility to test without ffplay? I would have to compile SDL (and possible depending stuff) in order to build it. Ingo From onemda at gmail.com Mon Oct 21 21:53:24 2013 From: onemda at gmail.com (Paul B Mahol) Date: Mon, 21 Oct 2013 19:53:24 +0000 Subject: [FFmpeg-user] Blurry artefacts during playback In-Reply-To: <526584f2.30ab263c.bm000@wupperonline.de> References: <526584f2.30ab263c.bm000@wupperonline.de> Message-ID: On 10/21/13, Ingo Brueckl wrote: > Carl Eugen Hoyos wrote on Mon, 21 Oct 2013 18:41:20 +0000 (UTC): > >> Ingo Brueckl wupperonline.de> writes: > >>> So far, my problems with FFmpeg in the past (as >>> well as my problems during the bisect) were >>> because of my old CPU (AMD Athlon XP 2400+). > >> Could you test the following? >> $ ffplay -cpuflags 0 hbc9.avi >> $ ffplay -cpuflags mmx hbc9.avi >> $ ffplay -cpuflags sse hbc9.avi >> $ ffplay -cpuflags 3dnow hbc9.avi >> $ ffplay -cpuflags 3dnow+3dnowext hbc9.avi > > Is there a possibility to test without ffplay? I would have to compile SDL > (and possible depending stuff) in order to build it. You can use same args for ffmpeg. > > Ingo > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ib at wupperonline.de Mon Oct 21 22:17:37 2013 From: ib at wupperonline.de (=?ISO-8859-1?Q?Ingo=20Br=FCckl?=) Date: Mon, 21 Oct 2013 22:17:37 +0200 Subject: [FFmpeg-user] Blurry artefacts during playback In-Reply-To: Message-ID: <52658bfb.492373de.bm000@wupperonline.de> Paul B Mahol wrote on Mon, 21 Oct 2013 19:53:24 +0000: > On 10/21/13, Ingo Brueckl wrote: >> Carl Eugen Hoyos wrote on Mon, 21 Oct 2013 18:41:20 +0000 (UTC): >> >>> Ingo Brueckl wupperonline.de> writes: >> >>>> So far, my problems with FFmpeg in the past (as >>>> well as my problems during the bisect) were >>>> because of my old CPU (AMD Athlon XP 2400+). >> >>> Could you test the following? >>> $ ffplay -cpuflags 0 hbc9.avi >>> $ ffplay -cpuflags mmx hbc9.avi >>> $ ffplay -cpuflags sse hbc9.avi >>> $ ffplay -cpuflags 3dnow hbc9.avi >>> $ ffplay -cpuflags 3dnow+3dnowext hbc9.avi >> >> Is there a possibility to test without ffplay? I would have to compile SDL >> (and possible depending stuff) in order to build it. > You can use same args for ffmpeg. I did copies of the video (h264 -> mpeg4)... ...without cpuflags: artefacts ...with -cpuflags sse: artefacts ...with any other cpuflags: ok! Ingo From pb at das-werkstatt.com Mon Oct 21 22:27:32 2013 From: pb at das-werkstatt.com (Peter B.) Date: Mon, 21 Oct 2013 22:27:32 +0200 Subject: [FFmpeg-user] FFV1 multi-pass: Could not allocate file buffer Message-ID: <52658E34.1060007@das-werkstatt.com> Hello, I've tried to run a 2-pass encoding with FFV1.3 on a 4h computer-generated video, but I'm getting the following error message when I try to run the 2nd pass: "Could not allocate file buffer Error reading log file 'ffv1_passlog-0.log' for pass-2 encoding" The file exists, has around 7.2 GiB and is readable by the user running the command. Could the filesize of the pass-logfile be the problem? I've never ran 2-pass tests on FFV1 videos of that size in one piece. Running the same 2 commands, but adding duration limit (-t 00:30:00.000), 2-pass encoding works fine (logfile is "only" 1.6 GiB). I'd be grateful for any hints about how to solve this, Pb Here's the commandline and full, uncut console output: ================================ PASS 1 ================================ $ ffmpeg -i 360000_frames.avi -an -c:v ffv1 -level 3 -coder 1 -context 1 -slices 24 -slicecrc 1 -pass 1 -passlogfile ffv1_passlog -threads 8 output_pass1.avi ffmpeg version N-57294-gab2bfb8 Copyright (c) 2000-2013 the FFmpeg developers built on Oct 21 2013 18:16:55 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) configuration: --prefix=/usr/local --enable-gpl --enable-nonfree --enable-version3 --enable-postproc --enable-swscale --enable-avfilter --enable-pthreads --enable-bzlib --enable-libmp3lame --enable-libvorbis --enable-libxvid --enable-zlib --enable-libopenjpeg --enable-decoder=png --enable-encoder=png --enable-libfreetype --enable-libschroedinger --enable-libvpx --enable-libvorbis --enable-libx264 --enable-libfaac libavutil 52. 47.101 / 52. 47.101 libavcodec 55. 37.102 / 55. 37.102 libavformat 55. 19.103 / 55. 19.103 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.102 / 3. 88.102 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Guessed Channel Layout for Input Stream #0.1 : stereo Input #0, avi, from '360000_frames.avi': Metadata: date : 2013-10-20T23:39 title : DVA Fidelity-Analyzer Testvideo (v2.0) encoder : Lavf55.19.100 encoded_by : ?sterreichische Mediathek Duration: 04:00:00.00, start: 0.000000, bitrate: 10896 kb/s Stream #0:0: Video: ffv1 (FFV1 / 0x31564646), yuv422p, 720x576, SAR 1:1 DAR 5:4, 25 tbr, 25 tbn, 25 tbc Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s Output #0, avi, to 'output_pass1.avi': Metadata: ICRD : 2013-10-20T23:39 INAM : DVA Fidelity-Analyzer Testvideo (v2.0) ITCH : ?sterreichische Mediathek ISFT : Lavf55.19.103 Stream #0:0: Video: ffv1 (FFV1 / 0x31564646), yuv422p, 720x576 [SAR 1:1 DAR 5:4], q=2-31, pass 1, 200 kb/s, 25 tbn, 25 tbc Stream mapping: Stream #0:0 -> #0:0 (ffv1 -> ffv1) Press [q] to stop, [?] for help frame=360000 fps=141 q=-1.0 Lsize= 7501157kB time=04:00:00.00 bitrate=4267.3kbits/s video:7494564kB audio:0kB subtitle:0 global headers:0kB muxing overhead 0.087971% ================================ PASS 2 ================================ $ ffmpeg -i 360000_frames.avi -c:a copy -c:v ffv1 -level 3 -coder 1 -context 1 -slices 24 -slicecrc 1 -pass 2 -passlogfile ffv1_passlog -threads 8 output_pass2.avi ffmpeg version N-57294-gab2bfb8 Copyright (c) 2000-2013 the FFmpeg developers built on Oct 21 2013 18:16:55 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) configuration: --prefix=/usr/local --enable-gpl --enable-nonfree --enable-version3 --enable-postproc --enable-swscale --enable-avfilter --enable-pthreads --enable-bzlib --enable-libmp3lame --enable-libvorbis --enable-libxvid --enable-zlib --enable-libopenjpeg --enable-decoder=png --enable-encoder=png --enable-libfreetype --enable-libschroedinger --enable-libvpx --enable-libvorbis --enable-libx264 --enable-libfaac libavutil 52. 47.101 / 52. 47.101 libavcodec 55. 37.102 / 55. 37.102 libavformat 55. 19.103 / 55. 19.103 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.102 / 3. 88.102 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Guessed Channel Layout for Input Stream #0.1 : stereo Input #0, avi, from '360000_frames.avi': Metadata: date : 2013-10-20T23:39 title : DVA Fidelity-Analyzer Testvideo (v2.0) encoder : Lavf55.19.100 encoded_by : ?sterreichische Mediathek Duration: 04:00:00.00, start: 0.000000, bitrate: 10896 kb/s Stream #0:0: Video: ffv1 (FFV1 / 0x31564646), yuv422p, 720x576, SAR 1:1 DAR 5:4, 25 tbr, 25 tbn, 25 tbc Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s Could not allocate file buffer Error reading log file 'ffv1_passlog-0.log' for pass-2 encoding From cehoyos at ag.or.at Mon Oct 21 23:36:10 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 21 Oct 2013 21:36:10 +0000 (UTC) Subject: [FFmpeg-user] FFV1 multi-pass: Could not allocate file buffer References: <52658E34.1060007@das-werkstatt.com> Message-ID: Peter B. das-werkstatt.com> writes: > I've tried to run a 2-pass encoding with FFV1.3 on a 4h > computer-generated video, but I'm getting the following > error message when I try to run the 2nd pass: > > "Could not allocate file buffer This is the relevant code: *size = ftell(f); ... *bufptr = av_malloc(*size + 1); if (!*bufptr) { av_log(NULL, AV_LOG_ERROR, "Could not allocate file buffer\n"); I wonder if there is anyrhing we can do... (But you could increase ram.) Allocating 1.6G should be possible with the code in libavutil/mem.c (at least from a quick look). What does free report on your system? Is this 32 or 64bit? Carl Eugen From onemda at gmail.com Mon Oct 21 23:39:42 2013 From: onemda at gmail.com (Paul B Mahol) Date: Mon, 21 Oct 2013 21:39:42 +0000 Subject: [FFmpeg-user] FFV1 multi-pass: Could not allocate file buffer In-Reply-To: References: <52658E34.1060007@das-werkstatt.com> Message-ID: On 10/21/13, Carl Eugen Hoyos wrote: > Peter B. das-werkstatt.com> writes: > >> I've tried to run a 2-pass encoding with FFV1.3 on a 4h >> computer-generated video, but I'm getting the following >> error message when I try to run the 2nd pass: >> >> "Could not allocate file buffer > > This is the relevant code: > *size = ftell(f); > ... > *bufptr = av_malloc(*size + 1); > if (!*bufptr) { > av_log(NULL, AV_LOG_ERROR, "Could not allocate file buffer\n"); > > I wonder if there is anyrhing we can do... > (But you could increase ram.) > > Allocating 1.6G should be possible with the code in > libavutil/mem.c (at least from a quick look). > > What does free report on your system? > Is this 32 or 64bit? Are you telling that whole file is kept in memory? > > 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 Oct 21 23:41:51 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 21 Oct 2013 21:41:51 +0000 (UTC) Subject: [FFmpeg-user] FFV1 multi-pass: Could not allocate file buffer References: <52658E34.1060007@das-werkstatt.com> Message-ID: Paul B Mahol gmail.com> writes: > >> "Could not allocate file buffer > > > > This is the relevant code: > > *size = ftell(f); > > ... > > *bufptr = av_malloc(*size + 1); > > if (!*bufptr) { > > av_log(NULL, AV_LOG_ERROR, "Could not allocate file buffer\n"); > Are you telling that whole file is kept in memory? This is how I interpret the code I posted but this may of course be completely wrong. Carl Eugen From elliottbalsley at gmail.com Tue Oct 22 00:01:10 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Mon, 21 Oct 2013 15:01:10 -0700 Subject: [FFmpeg-user] map_channel deprecated In-Reply-To: <20131020075513.GA396@phare.normalesup.org> References: <20131020075513.GA396@phare.normalesup.org> Message-ID: >> I have a hard time finding good documentation on lavfi, and don't really >> understand how to use it. > > What is wrong with > http://ffmpeg.org/ffmpeg-filters.html > ? I've read that page. It has a few references to lavfi, but no section that explains what it actually is, what the syntax is, and why it's better/different than not using it. > > Regards, > > -- > Nicolas George > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From bostjan.strojan at gmail.com Tue Oct 22 01:13:46 2013 From: bostjan.strojan at gmail.com (=?UTF-8?Q?Bo=C5=A1tjan_Strojan?=) Date: Tue, 22 Oct 2013 01:13:46 +0200 Subject: [FFmpeg-user] Conversion to prores, close encode every N minutes, how? In-Reply-To: References: Message-ID: On Sun, Oct 20, 2013 at 9:08 PM, Werner Robitza wrote: > > On Sat, Oct 19, 2013 at 2:23 PM, Bo?tjan Strojan > wrote: > > > > Converting a very long stream, 2 hours and more into more editable format > > like prores. > > > > My question is: Is it possible to close the encode every N minutes and > > start a new one (generating multiple files), without restarting the > > process, if yes, how? > > You can achieve this with the segment muxer. Refer to the > documentation for some examples: > > http://ffmpeg.org/ffmpeg-formats.html#segment_002c-stream_005fsegment_002c-ssegment Thanks, but getting various errors (freshly compiled ffmpeg from git) like: ffmpeg started on 2013-10-22 at 01:10:48 Report written to "ffmpeg-20131022-011048.log" Command line: /Volumes/raid0/ffdrop_dev/dev/ffdrop14dev/ffmpeg/ffmpeg -report -i /Volumes/raid0/mokrisca/ae/avto.mov -c:v prores -map 0 -f segment -segment_time 10 "%03d.mov" ffmpeg version N-57300-g2f9422d Copyright (c) 2000-2013 the FFmpeg developers built on Oct 21 2013 22:15:31 with Apple clang version 4.1 (tags/Apple/clang-421.11.66) (based on LLVM 3.1svn) configuration: --prefix=/Volumes/tempdisk/sw --enable-gpl --enable-libx264 --cc=clang --enable-runtime-cpudetect libavutil 52. 47.101 / 52. 47.101 libavcodec 55. 37.102 / 55. 37.102 libavformat 55. 19.103 / 55. 19.103 libavdevice 55. 4.100 / 55. 4.100 libavfilter 3. 88.102 / 3. 88.102 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Splitting the commandline. Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'. Reading option '-i' ... matched as input file with argument '/Volumes/raid0/mokrisca/ae/avto.mov'. Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'prores'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0'. Reading option '-f' ... matched as option 'f' (force format) with argument 'segment'. Reading option '-segment_time' ... matched as AVOption 'segment_time' with argument '10'. Reading option '%03d.mov' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option report (generate a report) with argument 1. Successfully parsed a group of options. Parsing a group of options: input file /Volumes/raid0/mokrisca/ae/avto.mov. Successfully parsed a group of options. Opening an input file: /Volumes/raid0/mokrisca/ae/avto.mov. [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f8a4201ec00] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f8a4201ec00] ISO: File Type Major Brand: qt [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f8a4201ec00] File position before avformat_find_stream_info() is 7713733 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f8a4201ec00] All info found [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f8a4201ec00] File position after avformat_find_stream_info() is 588048 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Volumes/raid0/mokrisca/ae/avto.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2013-09-21 19:20:22 Duration: 00:00:00.56, start: 0.000000, bitrate: 110196 kb/s Stream #0:0(eng), 1, 1/25: Video: prores (apcn / 0x6E637061), yuv422p10le, 1920x1080, 110093 kb/s, SAR 1920:1920 DAR 16:9, 25 fps, 25 tbr, 25 tbn, 25 tbc (default) Metadata: creation_time : 2013-09-21 19:20:22 handler_name : Apple Alias Data Handler timecode : 00:00:11:17 Stream #0:1(eng), 0, 1/25: Data: none (tmcd / 0x64636D74), 0 kb/s (default) Metadata: creation_time : 2013-09-21 19:20:27 handler_name : Apple Alias Data Handler timecode : 00:00:11:17 Successfully opened the file. Parsing a group of options: output file %03d.mov. Applying option c:v (codec name) with argument prores. Applying option map (set input stream mapping) with argument 0. Applying option f (force format) with argument segment. Successfully parsed a group of options. Opening an output file: %03d.mov. Data stream encoding not supported yet (only streamcopy) [AVIOContext @ 0x7f8a41c03ea0] Statistics: 627909 bytes read, 2 seeks any clues? tia, b. From soeren at zfaas.com Tue Oct 22 02:37:00 2013 From: soeren at zfaas.com (Soeren Balko) Date: Tue, 22 Oct 2013 10:37:00 +1000 Subject: [FFmpeg-user] Audio "bumps" in concatenated file Message-ID: Hi, I am currently trying to accomplish the following scenario: (1) Split an input file (say a MOV file) into same size segments, using the "segment" muxer: ffmpeg -i input.mov -codec copy -f segment -segment_time segment%05d.mov (2) Transcode the individual segments separately into MPEG transport stream segments using x264 and fdk-aac as video and audio codecs, respectively, e.g.: ffmpeg -i segment00001.mov -bsf:v h264_mp4toannexb -f mpegts -acodec libfdk_aac -b:a 160k -ar 44100 -ac 2 -vcodec libx264 -profile:v baseline -x264-params level=4:ref=3 -g 90 -s hd1080 -r 29.97 -aspect 16/9 -b:v 5400k segment00001.ts (3) And finally, I concatenate the transcoded segments into a single MP4 output file: ffmpeg -i concat:segment00000.ts|segment00001.ts -codec copy -bsf:a aac_adtstoasc output.mp4 This all works and the resulting file looks good. However at the "glue points" of the transcoded segments there are brief "bumps" in the audio. My take on this is that the individually transcoded segments "fade out" at the end, causing the aforementioned "bumps". Naturally, I want a smooth, continuous audio in the concatenated output file and was wondering what to do about it. I searched the fdk_aac documentation and couldn't find anything. Also there is no mentioning of this problem in the FFmpeg documentation with regards to the segment and concat (de)muxers. So I was for one wondering if this is a known problem and more importantly, how to work around it. One idea was to simply let the segments overlap (I couldn't use the segment muxer for that) and when concatenating the transcoded segments, let the successive one start before its predecessor has ended. Not sure how complex this was (suggestions???) with regards to precise timing. At any rate, I would probably need to splice out the audio and video streams and treat these separately. Another idea was a possible "smoothening" run over the concatenated output file, provided such a feature exists in FFmpeg. Any suggestions? Thanks heaps, Soeren From lou at lrcd.com Tue Oct 22 02:59:32 2013 From: lou at lrcd.com (Lou) Date: Mon, 21 Oct 2013 16:59:32 -0800 Subject: [FFmpeg-user] MP4 video showing frame delay. In-Reply-To: References: Message-ID: <20131021165932.31f2f438@lrcd.com> On Mon, 21 Oct 2013 10:20:35 -0600 carlos gabriel hasbun comandari wrote: > Why does my video halt on a frame when playing it with VLC, while playing it with Media Player > Classic it plays correctly but with a squashed resolution (aspect ration with the height diminished) > > These are the parameters of the video created with FFMPEG N-45739-g04bf2e7: > > C:\Users\C. Gabriel H.C\Videos>ffmpeg -r 5 -f image2 -i "J:\Screencasts\GIMP\GAP\GIFtoAVI\video\GifToAvi\BMP\GifToAviTut_%06d.bmp" -i "J:\Screencasts\GIMP\GAP\GIFtoAV > I\audio\Export\GIMPGAP_GIFtoAVI_enhanced2.flac" -map 0 -map 1 -r 5 -b:v 512k -b:a 128k -acodec libvorbis -ac 1 -ar 48000 FinalGIFtoAVIenhanced.mp4 Please include the complete ffmpeg console output. mediainfo output is generally not as useful. From cehoyos at ag.or.at Tue Oct 22 08:41:05 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 22 Oct 2013 06:41:05 +0000 (UTC) Subject: [FFmpeg-user] Audio "bumps" in concatenated file References: Message-ID: Soeren Balko zfaas.com> writes: > ffmpeg -i segment00001.mov -bsf:v h264_mp4toannexb -f mpegts -acodec > libfdk_aac -b:a 160k -ar 44100 -ac 2 -vcodec libx264 -profile:v baseline > -x264-params level=4:ref=3 -g 90 -s hd1080 -r 29.97 -aspect 16/9 -b:v 5400k > segment00001.ts (If you are doing this only for performance reasons, it is missing -threads 1, threads > 1 have an overhead.) Does it work for audio-only? Did you test other audio encoders (mp2, ac3, native aac)? Carl Eugen From cehoyos at ag.or.at Tue Oct 22 08:42:34 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 22 Oct 2013 06:42:34 +0000 (UTC) Subject: [FFmpeg-user] Conversion to prores, close encode every N minutes, how? References: Message-ID: Bo?tjan Strojan gmail.com> writes: > /Volumes/raid0/ffdrop_dev/dev/ffdrop14dev/ffmpeg/ffmpeg -report -i > /Volumes/raid0/mokrisca/ae/avto.mov -c:v prores -map 0 -f segment > -segment_time 10 "%03d.mov" Use map 0:0 instead of map 0 Carl Eugen From bostjan.strojan at gmail.com Tue Oct 22 08:56:21 2013 From: bostjan.strojan at gmail.com (=?UTF-8?Q?Bo=C5=A1tjan_Strojan?=) Date: Tue, 22 Oct 2013 08:56:21 +0200 Subject: [FFmpeg-user] Conversion to prores, close encode every N minutes, how? In-Reply-To: References: Message-ID: On Tue, Oct 22, 2013 at 8:42 AM, Carl Eugen Hoyos wrote: > Use map 0:0 instead of map 0 Wonderfull, some files are now being made, but they are borken, can I fix them somehow? a codec copy 2nd pass perhaps? tia, b. From dashing.meng at gmail.com Tue Oct 22 09:33:35 2013 From: dashing.meng at gmail.com (littlebat) Date: Tue, 22 Oct 2013 15:33:35 +0800 Subject: [FFmpeg-user] Multiple outputs worse performance than multiple instances In-Reply-To: References: Message-ID: <20131022153335.c72f338922c6fdce8eacd3a0@gmail.com> On Mon, 21 Oct 2013 14:36:47 +0200 Carl Lindqvist wrote: > Sorry for the previous top post. Gmail is giving me a headache.. > > I did a small typo/brain fart, what I meant to say was: > > Everything works perfectly, but cpu usage on the very fast server (16 > cores) is quite bad. I can set -preset slow and get the same fps > encoding speed, but with higher cpu usage. > Don't know more about multiple threads decoding or encoding, But I think you need understand some things: 1, CPU usage is low, if there are some other bottle neck? e.g., disk IO? 2, What kinds of codecs support multiple threads decoding or encoding? 3, If a codecs support multiple threads decoding or encoding, if these threads can occupy all the CPUs' usage? Just a clue. If we can't issue enough commands at same time to occupy all the CPU cores, and how to use the multiple decoding and encoding ability of ffmpeg to achieve this goal, I am also interest in this question :) From onemda at gmail.com Tue Oct 22 10:14:51 2013 From: onemda at gmail.com (Paul B Mahol) Date: Tue, 22 Oct 2013 08:14:51 +0000 Subject: [FFmpeg-user] strip doesn't work In-Reply-To: <52655121.5af7e7a4.bm000@wupperonline.de> References: <52655121.5af7e7a4.bm000@wupperonline.de> Message-ID: On 10/21/13, Ingo Brueckl wrote: > CC libswscale/utils.o > YASM libswscale/x86/input.o > STRIP libswscale/x86/input.o > /bin/sh: 1: -wN: not found > > Apparently commit e52567c2954f627d420b30f75f71af2f2e4afe80. Perhaps you have very old binutils. > > Ingo > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From wangxingchao2011 at gmail.com Tue Oct 22 10:56:02 2013 From: wangxingchao2011 at gmail.com (Wang Xingchao) Date: Tue, 22 Oct 2013 16:56:02 +0800 Subject: [FFmpeg-user] Mosaic issue when playing RMVB with ffmpeg In-Reply-To: References: Message-ID: Hi All, finally i found this commit fix the mosaic issue in my case: commit ee4b14322155b5808eeceb463f5edcd751eb3a98 Author: Marton Balint Date: Thu Mar 15 00:41:57 2012 +0100 ffplay: use frame count based queueing for audio queue This reduces the number of queued frames for audio data but also reduces the amount of A-V difference after changing the audio stream (because less frames are queued). Fixes bug #1035. Signed-off-by: Marton Balint And it was claimed to fix the bug here: https://trac.ffmpeg.org/ticket/1035 But i really donot know why it could also fix the mosaic issue. Any one know more background about this fix? thanks --xingchao 2013/10/12 Wang Xingchao : > Hi Carl, > > 2013/10/12 Carl Eugen Hoyos : >> Wang Xingchao gmail.com> writes: >> >>> The issue is reproduced with below environments: >>> - ffplay version 0.8 >> >>> please note this issue doesnot occur with ffplay 2.0 >>> on desktop. >> >> Don't you agree that there is nothing that we can do >> about this issue or do I miss something? > > i wonder what caused this issue becuase i'm using ffmpeg(1.1.3) on > android which met the problem. > And seems this issue was fixed since V0.8, so knowing more background > about this issue is helpful. > > thanks > --xingchao >> >> Carl Eugen >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user From soeren at zfaas.com Tue Oct 22 11:39:18 2013 From: soeren at zfaas.com (Soeren Balko) Date: Tue, 22 Oct 2013 19:39:18 +1000 Subject: [FFmpeg-user] Audio "bumps" in concatenated file In-Reply-To: References: Message-ID: I tested this for audio-only with the same result. Didn't check other audio codecs, though but will do and report on the outcome. On Tue, Oct 22, 2013 at 4:41 PM, Carl Eugen Hoyos wrote: > Soeren Balko zfaas.com> writes: > > > ffmpeg -i segment00001.mov -bsf:v h264_mp4toannexb -f mpegts -acodec > > libfdk_aac -b:a 160k -ar 44100 -ac 2 -vcodec libx264 -profile:v baseline > > -x264-params level=4:ref=3 -g 90 -s hd1080 -r 29.97 -aspect 16/9 -b:v > 5400k > > segment00001.ts > > (If you are doing this only for performance reasons, > it is missing -threads 1, threads > 1 have an overhead.) > > Does it work for audio-only? > Did you test other audio encoders (mp2, ac3, native aac)? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From tsinghal18 at gmail.com Tue Oct 22 13:35:37 2013 From: tsinghal18 at gmail.com (Tarun singhal) Date: Tue, 22 Oct 2013 17:05:37 +0530 Subject: [FFmpeg-user] Remux to a "brand mp42" MP4 file In-Reply-To: References: <66567361A4EC46149E5014E23C83D4CE@vasonote> <1373183970642-4659859.post@n4.nabble.com> Message-ID: On Sun, Jul 7, 2013 at 5:02 PM, Carl Eugen Hoyos wrote: > svnpenn gmail.com> writes: > > > > Could you explain your use-case, so we would > > > know what to test? > > > (Why is mp41 not godd enough?) > > > > Carl, I believe my issue is also caused by bad > > "atoms" being written by FFmpeg > > Command line and complete, uncut console output missing. > > Please test -f ipod > > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Dear Carl/All, I am responding in continuation this email chain- 1) Does ffmpeg supports mp4v2 or not? 2) Why do I need mp4v2, reason for that- It seems eac3 codec is not supported in mp4v1 container. Console Output: *ffmpeg version N-57147-ga06dcde Copyright (c) 2000-2013 the FFmpeg developers* * built on Oct 14 2013 18:09:04 with gcc 4.8.1 (GCC)* * configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --* *bcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libo* *le-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --ena* *-enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib* * libavutil 52. 46.101 / 52. 46.101* * libavcodec 55. 36.100 / 55. 36.100* * libavformat 55. 19.102 / 55. 19.102* * libavdevice 55. 4.100 / 55. 4.100* * libavfilter 3. 88.101 / 3. 88.101* * libswscale 2. 5.101 / 2. 5.101* * libswresample 0. 17.103 / 0. 17.103* * libpostproc 52. 3.100 / 52. 3.100* *Input #0, mpeg, from 'source.mpg':* * Duration: 00:00:09.08, start: 0.739989, bitrate: 8253 kb/s* * Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p(tv), 720x576 [SAR 64:45 DAR 16:9], max. 104857 kb/s, 25 fps, 25 tbr, 90k tbn, 50 t* * Stream #0:1[0x1c0]: Audio: mp2, 48000 Hz, stereo, s16p, 256 kb/s* *File 'out2.mp4' already exists. Overwrite ? [y/N] y* *[mp4 @ 0000000004d2b520] track 1: could not find tag, codec not currently supported in container* *Output #0, mp4, to 'out2.mp4':* * Metadata:* * encoder : Lavf55.19.102* * Stream #0:0: Video: mpeg4 ( [0][0][0] / 0x0020), yuv420p, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 12800 tbn, 25 tbc* * Stream #0:1: Audio: eac3, 48000 Hz, 2.1, fltp, 192 kb/s* *Stream mapping:* * Stream #0:0 -> #0:0 (mpeg2video -> mpeg4)* * Stream #0:1 -> #0:1 (mp2 -> eac3)* *Could not write header for output file #0 (incorrect codec parameters ?): Error number -1 occurred* But I downloaded one mp4 video from dolby http://download.dolby.com/Content_Download/channelcheck-cXp.mp4 This video contains eac3 codec, the only diff is its mp4v2. BTW, I tried with -f ipod also, But I get almost same output as of above. Thanks, Tarun From pb at das-werkstatt.com Tue Oct 22 13:41:36 2013 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 22 Oct 2013 13:41:36 +0200 Subject: [FFmpeg-user] FFV1 multi-pass: Could not allocate file buffer In-Reply-To: References: <52658E34.1060007@das-werkstatt.com> Message-ID: <20131022134136.86626n23jsi449dc@webmail.tuwien.ac.at> Quoting Carl Eugen Hoyos : > This is the relevant code: > *size = ftell(f); > ... > *bufptr = av_malloc(*size + 1); > if (!*bufptr) { > av_log(NULL, AV_LOG_ERROR, "Could not allocate file buffer\n"); Thanks! Happy to see that you also point to the that piece of code. I already looked at it when searching for more information about what throws this error message. Unfortunately, I'm not C-experienced enough to tell what "ftell()" actually returns in that case. It's not the filesize (which is what I first assumed), but in ftell's reference it says: "Get current position in stream Returns the current value of the position indicator of the stream. For binary streams, this is the number of bytes from the beginning of the file." > I wonder if there is anyrhing we can do... > (But you could increase ram.) Increasing the RAM on our machines (yes, more than one) would seem more like a workaround. Especially, considering that this video is rather "small" (18.2 GiB for 4h), compared to analog-captured VHS material with around 100 GiB for 4h. :( Maybe pass-logfiles of this size are just so rare that noone ever ran into allocation issues of large files? Could the allocation mode be changed? > Allocating 1.6G should be possible with the code in > libavutil/mem.c (at least from a quick look). The 1.6 GiB pass-logfile is the one that does work. It wouldn't surprise me, since the machine I tested that on has 8 GiB of RAM. > What does free report on your system? > Is this 32 or 64bit? All systems I tested this on were 64bit Xubuntu GNU/Linux. When should I execute "free" for this: Before, after or during execution of the 2nd pass? Thank you very much for your help, Pb From cehoyos at ag.or.at Tue Oct 22 14:03:04 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 22 Oct 2013 12:03:04 +0000 (UTC) Subject: [FFmpeg-user] FFV1 multi-pass: Could not allocate file buffer References: <52658E34.1060007@das-werkstatt.com> <20131022134136.86626n23jsi449dc@webmail.tuwien.ac.at> Message-ID: Peter B. das-werkstatt.com> writes: > > Allocating 1.6G should be possible with the code in > > libavutil/mem.c (at least from a quick look). > > The 1.6 GiB pass-logfile is the one that does work. Then the problem is likely that the larger files are so large that a check in libavutil/mem.c prohibits allocation. You could remove that check but I start to realize that this is indeed something that has to be changed. How large is the passlog-file that does not work? Carl Eugen From pb at das-werkstatt.com Tue Oct 22 14:13:35 2013 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 22 Oct 2013 14:13:35 +0200 Subject: [FFmpeg-user] FFV1 multi-pass: Could not allocate file buffer In-Reply-To: References: <52658E34.1060007@das-werkstatt.com> <20131022134136.86626n23jsi449dc@webmail.tuwien.ac.at> Message-ID: <20131022141335.11347dspn2ou9drj@webmail.tuwien.ac.at> Quoting Carl Eugen Hoyos : > Then the problem is likely that the larger files are > so large that a check in libavutil/mem.c prohibits > allocation. > You could remove that check but I start to realize > that this is indeed something that has to be changed. I'll give that a try! Thanks for the tip. If I understood you correctly, then disabling the check "might" allow to use large files like that. But I guess there's a reason for that check to be there in the first place? If I also understood this correctly, then it means that huge amounts of RAM are allocated for reading the pass-logfile in the 2nd pass. > How large is the passlog-file that does not work? In the case of this video the logfile is around 7.2 GiB. Puzzling though, I just saw that on the second computer where I tested this, the logfile for the same video has around 12.8 GiB. Unfortunately, the runtime to produce this 1st pass logfile takes a few hours, so it's not something that I can easily reproduce ad-hoc :( I'll try disabling the check in libavutil/mem.c and report back as soon as I know more :) Thanks, Pb From cehoyos at ag.or.at Tue Oct 22 14:16:37 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 22 Oct 2013 12:16:37 +0000 (UTC) Subject: [FFmpeg-user] Remux to a "brand mp42" MP4 file References: <66567361A4EC46149E5014E23C83D4CE@vasonote> <1373183970642-4659859.post@n4.nabble.com> Message-ID: Tarun singhal gmail.com> writes: > I am responding in continuation this email chain Which is not a good idea imo. [...] > But I downloaded one mp4 video from dolby > http://download.dolby.com/Content_Download/channelcheck-cXp.mp4 > > This video contains eac3 codec, the only diff is its mp4v2. No, the "difference" is that FFmpeg's mov/isom muxer does not contain a function mov_write_eac3_tag() which is necessary to write compliant files. Carl Eugen From cehoyos at ag.or.at Tue Oct 22 14:18:43 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 22 Oct 2013 12:18:43 +0000 (UTC) Subject: [FFmpeg-user] FFV1 multi-pass: Could not allocate file buffer References: <52658E34.1060007@das-werkstatt.com> <20131022134136.86626n23jsi449dc@webmail.tuwien.ac.at> <20131022141335.11347dspn2ou9drj@webmail.tuwien.ac.at> Message-ID: Peter B. das-werkstatt.com> writes: > > You could remove that check but I start to realize > > that this is indeed something that has to be changed. > > I'll give that a try! It may be easier to just test "-max_alloc 100000000000" Carl Eugen From tsinghal18 at gmail.com Tue Oct 22 14:48:24 2013 From: tsinghal18 at gmail.com (Tarun singhal) Date: Tue, 22 Oct 2013 18:18:24 +0530 Subject: [FFmpeg-user] Remux to a "brand mp42" MP4 file In-Reply-To: References: <66567361A4EC46149E5014E23C83D4CE@vasonote> <1373183970642-4659859.post@n4.nabble.com> Message-ID: On Tue, Oct 22, 2013 at 5:46 PM, Carl Eugen Hoyos wrote: > Tarun singhal gmail.com> writes: > > > I am responding in continuation this email chain > > Which is not a good idea imo. > > [...] > > > But I downloaded one mp4 video from dolby > > http://download.dolby.com/Content_Download/channelcheck-cXp.mp4 > > > > This video contains eac3 codec, the only diff is its mp4v2. > > No, the "difference" is that FFmpeg's mov/isom muxer does > not contain a function mov_write_eac3_tag() which is > necessary to write compliant files. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > so ffmpeg muxer doesnt contains function mov_write_eac3_tag(), that means I cant get EAC3 codec in mp4 container? Is there any roadmap for including this in upcoming releases? Thanks, Tarun From onemda at gmail.com Tue Oct 22 14:58:48 2013 From: onemda at gmail.com (Paul B Mahol) Date: Tue, 22 Oct 2013 12:58:48 +0000 Subject: [FFmpeg-user] Remux to a "brand mp42" MP4 file In-Reply-To: References: <66567361A4EC46149E5014E23C83D4CE@vasonote> <1373183970642-4659859.post@n4.nabble.com> Message-ID: On 10/22/13, Tarun singhal wrote: > On Tue, Oct 22, 2013 at 5:46 PM, Carl Eugen Hoyos wrote: > >> Tarun singhal gmail.com> writes: >> >> > I am responding in continuation this email chain >> >> Which is not a good idea imo. >> >> [...] >> >> > But I downloaded one mp4 video from dolby >> > http://download.dolby.com/Content_Download/channelcheck-cXp.mp4 >> > >> > This video contains eac3 codec, the only diff is its mp4v2. >> >> No, the "difference" is that FFmpeg's mov/isom muxer does >> not contain a function mov_write_eac3_tag() which is >> necessary to write compliant files. >> >> Carl Eugen >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > so ffmpeg muxer doesnt contains function mov_write_eac3_tag(), that means I > cant get EAC3 codec in mp4 container? > Is there any roadmap for including this in upcoming releases? Usual way is to report feature requst on bug tracker. There are no roadmaps for anything. > > Thanks, > Tarun > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From pb at das-werkstatt.com Tue Oct 22 14:59:56 2013 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 22 Oct 2013 14:59:56 +0200 Subject: [FFmpeg-user] FFV1 multi-pass: Could not allocate file buffer In-Reply-To: References: <52658E34.1060007@das-werkstatt.com> <20131022134136.86626n23jsi449dc@webmail.tuwien.ac.at> <20131022141335.11347dspn2ou9drj@webmail.tuwien.ac.at> Message-ID: <20131022145956.39541yj6h5rklpzg@webmail.tuwien.ac.at> Quoting Carl Eugen Hoyos : > Peter B. das-werkstatt.com> writes: > >> > You could remove that check but I start to realize >> > that this is indeed something that has to be changed. >> >> I'll give that a try! > > It may be easier to just test "-max_alloc 100000000000" Tried it right now. Returns the same error :( Pb From ib at wupperonline.de Tue Oct 22 19:49:46 2013 From: ib at wupperonline.de (=?ISO-8859-1?Q?Ingo=20Br=FCckl?=) Date: Tue, 22 Oct 2013 19:49:46 +0200 Subject: [FFmpeg-user] Blurry artefacts during playback In-Reply-To: <52658bfb.492373de.bm000@wupperonline.de> Message-ID: <5266babb.0368b98b.bm002@wupperonline.de> Can anyone confirm now? Will anyone fix it? Ingo From ib at wupperonline.de Tue Oct 22 19:10:43 2013 From: ib at wupperonline.de (=?ISO-8859-1?Q?Ingo=20Br=FCckl?=) Date: Tue, 22 Oct 2013 19:10:43 +0200 Subject: [FFmpeg-user] strip doesn't work In-Reply-To: Message-ID: <5266babb.5b163da6.bm001@wupperonline.de> Paul B Mahol wrote on Tue, 22 Oct 2013 08:14:51 +0000: > On 10/21/13, Ingo Brueckl wrote: >> CC libswscale/utils.o >> YASM libswscale/x86/input.o >> STRIP libswscale/x86/input.o >> /bin/sh: 1: -wN: not found >> >> Apparently commit e52567c2954f627d420b30f75f71af2f2e4afe80. > Perhaps you have very old binutils. No. Actually, the error only occurs when compiling with MPlayer, because STRIP isn't defined there. Although one might think that it is a pure MPlayer problem, it isn't, because the condition if $(STRIP) ... is kinda lazy. $(STRIP) will always expand (to @printf STRIP\t%s\n ...), even if there is no strip util available. It should rather be checked if there is a "; SOMETHING" at the end of $(STRIP). (Well, and MPlayer maybe should defined STRIP, but that will be discussed over there.) Ingo From arminkappeler2011 at u.northwestern.edu Tue Oct 22 05:31:52 2013 From: arminkappeler2011 at u.northwestern.edu (Armin Kappeler) Date: Mon, 21 Oct 2013 22:31:52 -0500 Subject: [FFmpeg-user] ffmpeg noise command: How is noise level defined Message-ID: I am using ffmpeg to add noise to my video by using the "noise" command (Section 9.53 in user documentation). The noise level can be choosen between 0 and 100, but how is the noise level calculated/defined? (PSNR, variance, deviation etc.) Thank you for your help Armin From james at dmi1011.com Tue Oct 22 19:37:00 2013 From: james at dmi1011.com (James Cahall) Date: Tue, 22 Oct 2013 10:37:00 -0700 Subject: [FFmpeg-user] configure with libx265 Message-ID: Hi, I have the latest source from GIT and getting ERROR: libx265 not found. I'm running the following configure command: ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libaacplus --enable-libass --enable-libcelt --enable-libfaac --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-openssl --enable-libopus --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-libx265 --prefix=/usr/local Any help is appreciated. Thanks! Regards, James Cahall From lou at lrcd.com Tue Oct 22 20:21:47 2013 From: lou at lrcd.com (Lou) Date: Tue, 22 Oct 2013 10:21:47 -0800 Subject: [FFmpeg-user] ffmpeg noise command: How is noise level defined In-Reply-To: References: Message-ID: <20131022102147.163a5ae2@lrcd.com> On Mon, 21 Oct 2013 22:31:52 -0500 Armin Kappeler wrote: > I am using ffmpeg to add noise to my video by using the "noise" command > (Section 9.53 in user documentation). The noise level can be choosen > between 0 and 100, but how is the noise level calculated/defined? (PSNR, > variance, deviation etc.) Please include your ffmpeg command and the complete ffmpeg console output so I have a better idea of what you're referring to. From krueger at lesspain.de Tue Oct 22 20:48:01 2013 From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=) Date: Tue, 22 Oct 2013 20:48:01 +0200 Subject: [FFmpeg-user] strip doesn't work In-Reply-To: References: <52655121.5af7e7a4.bm000@wupperonline.de> Message-ID: On Tue, Oct 22, 2013 at 10:14 AM, Paul B Mahol wrote: > On 10/21/13, Ingo Brueckl wrote: >> CC libswscale/utils.o >> YASM libswscale/x86/input.o >> STRIP libswscale/x86/input.o >> /bin/sh: 1: -wN: not found >> >> Apparently commit e52567c2954f627d420b30f75f71af2f2e4afe80. > > Perhaps you have very old binutils. I am observing the same with recent revisions of ffmpeg on Mac OS X. CC libswscale/utils.o YASM libswscale/x86/input.o STRIP libswscale/x86/input.o strip: unrecognized option: -wN Usage: strip [-AnuSXx] [-] [-d filename] [-s filename] [-R filename] [-o output] file [...] make: [libswscale/x86/input.o] Error 1 (ignored) ... I have never noticed this before. From belcampo at zonnet.nl Tue Oct 22 23:28:27 2013 From: belcampo at zonnet.nl (Henk D. Schoneveld) Date: Tue, 22 Oct 2013 23:28:27 +0200 Subject: [FFmpeg-user] Multiple outputs worse performance than multiple instances In-Reply-To: <20131022153335.c72f338922c6fdce8eacd3a0@gmail.com> References: <20131022153335.c72f338922c6fdce8eacd3a0@gmail.com> Message-ID: <9AC788CF-7B55-466C-A48A-4E2E52359929@zonnet.nl> On Oct 22, 2013, at 9:33 AM, littlebat wrote: > On Mon, 21 Oct 2013 14:36:47 +0200 > Carl Lindqvist wrote: > >> Sorry for the previous top post. Gmail is giving me a headache.. >> >> I did a small typo/brain fart, what I meant to say was: >> >> Everything works perfectly, but cpu usage on the very fast server (16 >> cores) is quite bad. I can set -preset slow and get the same fps >> encoding speed, but with higher cpu usage. If you do a simple test on 1 file, just change threads 1, threads 2 ?. threads 0, and note what each added core adds, you'll discover that every additional core adds LESS. Your original multiple instances AFAIK will be the optimum you'll get out your CPU-cores. Breaking up the source file to get 1 single core to encode 1 piece and then catting them together will be maximum encoding speed you'll get. Henk >> > Don't know more about multiple threads decoding or encoding, But I > think you need understand some things: > > 1, CPU usage is low, if there are some other bottle neck? e.g., disk IO? > > 2, What kinds of codecs support multiple threads decoding or encoding? > > 3, If a codecs support multiple threads decoding or encoding, if these > threads can occupy all the CPUs' usage? > > Just a clue. If we can't issue enough commands at same time to occupy > all the CPU cores, and how to use the multiple decoding and encoding > ability of ffmpeg to achieve this goal, I am also interest in this > question :) > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From cehoyos at ag.or.at Wed Oct 23 00:04:50 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 22 Oct 2013 22:04:50 +0000 (UTC) Subject: [FFmpeg-user] configure with libx265 References: Message-ID: James Cahall dmi1011.com> writes: > I have the latest source from GIT and getting > ERROR: libx265 not found. There is no configure option --enable-libx265 (Is there really libx265?) Carl Eugen From retrogradesnowcone at gmail.com Wed Oct 23 03:03:34 2013 From: retrogradesnowcone at gmail.com (Sarah) Date: Tue, 22 Oct 2013 20:03:34 -0500 Subject: [FFmpeg-user] ideal settings for mp3 + jpeg to video output? Message-ID: I want to take mp3's and jpg's and output them to a video file, with the intention they'll be uploaded to youtube. I'm wondering what the ideal settings would be. One stipulation, however, is that I want the mp3 stream to be used without it being reencoded. Thanks for any help :) From dashing.meng at gmail.com Wed Oct 23 03:48:43 2013 From: dashing.meng at gmail.com (littlebat) Date: Wed, 23 Oct 2013 09:48:43 +0800 Subject: [FFmpeg-user] ideal settings for mp3 + jpeg to video output? In-Reply-To: References: Message-ID: <20131023094843.84cbc1d6008ee75a8632b77c@gmail.com> On Tue, 22 Oct 2013 20:03:34 -0500 Sarah wrote: > I want to take mp3's and jpg's and output them to a video file, with > the intention they'll be uploaded to youtube. I'm wondering what the > ideal settings would be. One stipulation, however, is that I want the > mp3 stream to be used without it being reencoded. You can try to encode jpg's into a video stream, then mux the video stream and your original mp3 stream together. You can keep the best quality of your video stream, once you upload your muxed video+audio to youtube, youtube should take care of it to encode to the proper formats. From ez at efs.net.br Wed Oct 23 04:06:32 2013 From: ez at efs.net.br (=?ISO-8859-1?Q?Ezequiel_Fran=E7a?=) Date: Wed, 23 Oct 2013 00:06:32 -0200 Subject: [FFmpeg-user] Help with Image render into Video Message-ID: <52672F28.5000103@efs.net.br> Hi guys, how are you ? This is my first post here and I am needing some help with render images into video. I did a research on the web and asked guys on IRC and no one could help me with it. I am creating a bash script for a customer that converts multiple images into a video-slideshow. I need the imagens to appear into video on a specific order. I can generate the video using the pattern * but i am struggling using individual files. I got the file list and the order via JSON and created the following syntax: ffmpeg -v verbose -r 1/2 -i /var/www/html/Symfony/web/uploads/timeline/preview/imgvideo/testeez/45b37c51e89f2bd68b84e0e3d2bea1d83c5c8d96.png -i /var/www/html/Symfony/web/uploads/timeline/preview/imgvideo/testeez/39fe3494eb2dba3e7b9a4cacfc98450204d0137f.png -i /var/www/html/Symfony/web/uploads/timeline/preview/imgvideo/testeez/8d74d1399caad2b8dfc23c9183ddbec5b44a086b.png -i /var/www/html/Symfony/web/uploads/timeline/preview/imgvideo/testeez/09c26fc89b23dd742aaad8421938e224c35c68fa.png -i /var/www/html/Symfony/web/uploads/timeline/preview/imgvideo/testeez/9cad7bfd980c6cebae3d1b78520f35842114c27a.png -i /var/www/html/Symfony/web/uploads/timeline/preview/imgvideo/testeez/42f395cd036957ea51bae2f7719efccad4cbbcfa.png -i /var/www/html/Symfony/web/uploads/timeline/preview/imgvideo/testeez/0d442a4c4aa0b6c69682fbbbcf9d654562222f8c.png -i /var/www/html/Symfony/web/uploads/timeline/preview/imgvideo/testeez/58e0c5e9ba77cfc6a4b5f2a20debd6d74dc78a25.png -i /var/www/html/Symfony/web/uploads/timeline/preview/imgvideo/testeez/0c3b05645e810235b9fc266e4d0cc60320b11ab0.png /var/www/html/Symfony/web/uploads/documents/video0.mp4 It is not working and it seems to get only the first image. Another question is: There is a way to have different times for each image, something like "1st image stays for 2 sec, 2nd for 5 sec, etc" Is this the correct way to do this ? A read the documentation and It WORKS using the pattern *, for all imagens on a specific directory. I do not want to double the size copying the images to a directory and using the wildcard syntax. Thanks in advance and sorry about my english. It is not my native language. From lingjiujianke at gmail.com Wed Oct 23 04:37:52 2013 From: lingjiujianke at gmail.com (=?GB2312?B?wfUg4ao=?=) Date: Wed, 23 Oct 2013 10:37:52 +0800 Subject: [FFmpeg-user] ffmpeg and ffplay can not play the TrueHD audio Message-ID: <6F1B6CD4-D12E-4494-8AA8-959640D345A6@gmail.com> Hi guy, This test case is come from Jacky, I found ffmpeg can not decode the TrueHD(AC3). test command ffplay ~/Download/truehd.m2ts the TrueHD in m2ts test sample is here : http://blog.fs-linux.org/truehd.m2ts http://blog.fs-linux.org/truehd2.m2ts Can not play the TrueHD audio stream, but VLC can play it. Perhaps this is an bug. Carl ,is it? Thanks Steven Liu From lou at lrcd.com Wed Oct 23 05:47:56 2013 From: lou at lrcd.com (Lou) Date: Tue, 22 Oct 2013 19:47:56 -0800 Subject: [FFmpeg-user] ideal settings for mp3 + jpeg to video output? In-Reply-To: References: Message-ID: <1382500076.500.37355109.7B20C8A5@webmail.messagingengine.com> On Tue, Oct 22, 2013, at 05:03 PM, Sarah wrote: > I want to take mp3's and jpg's and output them to a video file, with the > intention they'll be uploaded to youtube. I'm wondering what the ideal > settings would be. One stipulation, however, is that I want the mp3 > stream > to be used without it being reencoded. Good, since YouTube will re-encode whatever you feed it anyway. > Thanks for any help :) ffmpeg -loop 1 -r 2 -i image.jpg -i audio.mp3 -c:v libx264 -crf 18 \ -c:a copy output.mkv Also see: http://trac.ffmpeg.org/wiki/EncodeforYouTube (might be a little outdated but it will give a few more examples) From lou at lrcd.com Wed Oct 23 05:58:00 2013 From: lou at lrcd.com (Lou) Date: Tue, 22 Oct 2013 19:58:00 -0800 Subject: [FFmpeg-user] ideal settings for mp3 + jpeg to video output? In-Reply-To: <1382500076.500.37355109.7B20C8A5@webmail.messagingengine.com> References: <1382500076.500.37355109.7B20C8A5@webmail.messagingengine.com> Message-ID: <1382500680.2995.37357761.1AE8B72C@webmail.messagingengine.com> On Tue, Oct 22, 2013, at 07:47 PM, Lou wrote: > > ffmpeg -loop 1 -r 2 -i image.jpg -i audio.mp3 -c:v libx264 -crf 18 \ > -c:a copy output.mkv I forgot to add "-shortest" (as an output option) to the command. From cehoyos at ag.or.at Wed Oct 23 09:53:49 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 23 Oct 2013 07:53:49 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg and ffplay can not play the TrueHD audio References: <6F1B6CD4-D12E-4494-8AA8-959640D345A6@gmail.com> Message-ID: ? ? gmail.com> writes: > I found ffmpeg can not decode the TrueHD(AC3). ^^^^^^ > test command > ffplay ~/Download/truehd.m2ts ^^^^^^ ffmpeg != ffplay Complete, uncut console output missing. > the TrueHD in m2ts test sample is here : > http://blog.fs-linux.org/truehd.m2ts The sample contains eight audio tracks, all eight audio tracks play fine here with ffplay. > http://blog.fs-linux.org/truehd2.m2ts The sample contains five audio tracks, all five audio tracks play fine here with ffplay. Carl Eugen From cehoyos at ag.or.at Wed Oct 23 09:55:33 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 23 Oct 2013 07:55:33 +0000 (UTC) Subject: [FFmpeg-user] Help with Image render into Video References: <52672F28.5000103@efs.net.br> Message-ID: Ezequiel Fran?a efs.net.br> writes: > Another question is: There is a way to have different > times for each image, something like "1st image stays > for 2 sec, 2nd for 5 sec, etc" Assuming you are using "-r 1", "copy" (see below) the first image once, the second four times, etc. > Is this the correct way to do this ? > > A read the documentation and It WORKS using the > pattern *, for all imagens on a specific directory. > I do not want to double the size copying the images > to a directory and using the wildcard syntax. Use symbolic links instead. Carl Eugen From jan at janwijma.nl Wed Oct 23 10:13:26 2013 From: jan at janwijma.nl (Jan Wijma) Date: Wed, 23 Oct 2013 10:13:26 +0200 Subject: [FFmpeg-user] Intel Evansport support Message-ID: Dear All, Not sure if this is the right place, but is the best i could think of. I currently own a NAS with an Intel Evansport CE5335 SOC. This CPU supports 1080P hardware transcoding. See: http://newsroom.intel.com/community/intel_newsroom/blog/2013/03/04/intel-launches-system-on-chip-storage-solution-designed-for-simple-video-transcoding-and-streaming To be able to fully leverage this CPU, suppor of mediaserver apps like Plex is vital. Plex however, says the can't support this kind of transcoding until FFMPEG does. Is FFMPEG going to be able to do so? Kind regards From elliottbalsley at gmail.com Wed Oct 23 10:26:12 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Wed, 23 Oct 2013 01:26:12 -0700 Subject: [FFmpeg-user] How to configure x86_64 Message-ID: From the latest git head, I try to configure ffmpeg with ?arch=x86_64, but it ends up with x86 (generic). I?m running OS 10.9 on a Mac Pro 5,1. It?s a 64 bit machine and a 64 bit kernel, so why is this happening? Am I correct to assume that using 64 bit ffmpeg would mean faster encoding? Full output is here: http://pastebin.com/ce8sADAL From onemda at gmail.com Wed Oct 23 10:44:53 2013 From: onemda at gmail.com (Paul B Mahol) Date: Wed, 23 Oct 2013 08:44:53 +0000 Subject: [FFmpeg-user] Help with Image render into Video In-Reply-To: References: <52672F28.5000103@efs.net.br> Message-ID: On 10/23/13, Carl Eugen Hoyos wrote: > Ezequiel Franc,a efs.net.br> writes: > >> Another question is: There is a way to have different >> times for each image, something like "1st image stays >> for 2 sec, 2nd for 5 sec, etc" > > Assuming you are using "-r 1", "copy" (see below) the > first image once, the second four times, etc. > >> Is this the correct way to do this ? >> >> A read the documentation and It WORKS using the >> pattern *, for all imagens on a specific directory. >> I do not want to double the size copying the images >> to a directory and using the wildcard syntax. > > Use symbolic links instead. Why? That is extremly bad idea. > > 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 Oct 23 10:52:40 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 23 Oct 2013 08:52:40 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?How_to_configure_x86=5F64?= References: Message-ID: Elliott Balsley gmail.com> writes: > From the latest git head, I try to configure ffmpeg with > ?arch=x86_64, but it ends up with x86 (generic). > I?m running OS 10.9 on a Mac Pro 5,1. It?s a 64 bit > machine and a 64 bit kernel, so why is this happening? Please try the following: $ ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-static --enable-libass --enable-libbluray --enable-libfdk-aac --enable-libmp3lame --enable-libvpx --enable-libx264 --enable-opencl --enable-libfreetype --cc='gcc -m64' > Am I correct to assume that using 64 bit ffmpeg would > mean faster encoding? To some degree, yes. > Full output is here: http://pastebin.com/ce8sADAL Please never use external resources to post short console output, always post it here. Carl Eugen From lulebo at gmail.com Wed Oct 23 10:55:37 2013 From: lulebo at gmail.com (Carl Lindqvist) Date: Wed, 23 Oct 2013 10:55:37 +0200 Subject: [FFmpeg-user] Multiple outputs worse performance than multiple instances In-Reply-To: <9AC788CF-7B55-466C-A48A-4E2E52359929@zonnet.nl> References: <20131022153335.c72f338922c6fdce8eacd3a0@gmail.com> <9AC788CF-7B55-466C-A48A-4E2E52359929@zonnet.nl> Message-ID: 2013/10/22 Henk D. Schoneveld : > > On Oct 22, 2013, at 9:33 AM, littlebat wrote: > >> On Mon, 21 Oct 2013 14:36:47 +0200 >> Carl Lindqvist wrote: >> >>> Sorry for the previous top post. Gmail is giving me a headache.. >>> >>> I did a small typo/brain fart, what I meant to say was: >>> >>> Everything works perfectly, but cpu usage on the very fast server (16 >>> cores) is quite bad. I can set -preset slow and get the same fps >>> encoding speed, but with higher cpu usage. > If you do a simple test on 1 file, just change threads 1, threads 2 ?. threads 0, and note what each added core adds, you'll discover that every additional core adds LESS. > Your original multiple instances AFAIK will be the optimum you'll get out your CPU-cores. > Breaking up the source file to get 1 single core to encode 1 piece and then catting them together will be maximum encoding speed you'll get. > Henk >>> >> Don't know more about multiple threads decoding or encoding, But I >> think you need understand some things: >> >> 1, CPU usage is low, if there are some other bottle neck? e.g., disk IO? >> >> 2, What kinds of codecs support multiple threads decoding or encoding? >> >> 3, If a codecs support multiple threads decoding or encoding, if these >> threads can occupy all the CPUs' usage? >> >> Just a clue. If we can't issue enough commands at same time to occupy >> all the CPU cores, and how to use the multiple decoding and encoding >> ability of ffmpeg to achieve this goal, I am also interest in this >> question :) >> _______________________________________________ >> 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 I will do some more tests today. One of the reasons I don't want to do encode with multiple instances of ffmpeg is that I don't want to tax the network. The source files are high bitrate mxf broadcast files, and the file system is shared, so 5 ffmpeg instances each reading the same file might impact other parts of the network. I would then have to copy the file locally before transcoding, which would negate some of the speed advantages. (Network speed is not a bottleneck; especially in my tests, as I have the file locally on hard drive when testing) As I understand it from the console output, ffmpeg spawns 5 seperate x264 encoders each with their own threads. I don't see why it should be slower than starting five instances of ffmpeg. I should be doing less work this way (only decode/deinterlace once, instead of five times), so I expected the speed to go up, not down. Maybe I am doing something weird in my filter graph that is inefficient? I will report if I find anything interesting. /Carl From elliottbalsley at gmail.com Wed Oct 23 11:33:18 2013 From: elliottbalsley at gmail.com (Elliott Balsley) Date: Wed, 23 Oct 2013 02:33:18 -0700 Subject: [FFmpeg-user] How to configure x86_64 In-Reply-To: References: Message-ID: <1E4DBE0E-F600-4F98-ABCF-14D2A70F80D3@gmail.com> Elliott Balsley gmail.com> writes: > >> From the latest git head, I try to configure ffmpeg with >> ?arch=x86_64, but it ends up with x86 (generic). >> I?m running OS 10.9 on a Mac Pro 5,1. It?s a 64 bit >> machine and a 64 bit kernel, so why is this happening? > > Please try the following: > $ ./configure --enable-gpl --enable-version3 --enable-nonfree > --enable-static --enable-libass --enable-libbluray > --enable-libfdk-aac --enable-libmp3lame --enable-libvpx > --enable-libx264 --enable-opencl --enable-libfreetype --cc='gcc -m64? That still gives the result of ?x86 (generic)" $ make distclean $ ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-static --enable-libass --enable-libbluray --enable-libfdk-aac --enable-libmp3lame --enable-libvpx --enable-libx264 --enable-opencl --enable-libfreetype --cc='gcc -m64' Unknown option: n Unknown option: 1 Usage: head [-options] ... -m use method for the request (default is 'HEAD') -f make request even if head believes method is illegal -b Use the specified URL as base -t Set timeout value -i