From x2305andy2305x at yahoo.com Thu Sep 1 06:55:07 2011 From: x2305andy2305x at yahoo.com (Andy Andy) Date: Wed, 31 Aug 2011 21:55:07 -0700 (PDT) Subject: [FFmpeg-user] (no subject) Message-ID: <1314852907.21025.androidMobile@web32508.mail.mud.yahoo.com> Thomas, You have been most helpful, i'd like to thank you very much for your patience and insight. Best regards, DAV Sent from Yahoo! Mail on Android From roundfile at tekbits.net Thu Sep 1 11:15:21 2011 From: roundfile at tekbits.net (Round File) Date: Thu, 1 Sep 2011 10:15:21 +0100 Subject: [FFmpeg-user] Emulate IP web camera In-Reply-To: <7D76607B-7DBC-4F8C-A0A5-7F0F5AEBB82B@pixelpartner.de> References: <1314779906.39111.YahooMailNeo@web32505.mail.mud.yahoo.com> <21840DFC-A92A-4E0A-9480-C3B7EC2CAD97@pixelpartner.de> <1314785759.95527.YahooMailNeo@web32505.mail.mud.yahoo.com> <7D76607B-7DBC-4F8C-A0A5-7F0F5AEBB82B@pixelpartner.de> Message-ID: <00b001cc6887$ad617220$08245660$@net> Hi all, It sounds basic for ffmpeg, but has anyone successfully emulated an IP web camera using ffmpeg? Example: ffmpeg -i file.mov -vcodec mpeg4 -an -s 176x144 -vcodec rawvideo -f rtp rtp://127.0.0.1:5000/ I'm trying to use ffmpeg as an IP camera for Wirecast, and can't seem to get it to work. Any help would be greatly appreciated. Kind regards, -N From pekon.gupta at gmail.com Thu Sep 1 15:10:40 2011 From: pekon.gupta at gmail.com (Pekon Gupta) Date: Thu, 1 Sep 2011 18:40:40 +0530 Subject: [FFmpeg-user] ffmpeg: encoding sub-titles Message-ID: Hi, Apologies for revisiting this question.. As i have googled a lot on this topic, and still could not find suitable answer.. (1) Is there a way to hard-code the ".srt" (text based subtitles) into the output video while encoding it ? If yes, then please illustrate it with command-line example.. (2) how is "-newsubtitle" switch is used ? Im using following command-line.. "ffmpeg -i in.avi -acodec libmp3lame -vcodec libxvid -bf 0 out.avi " Please tell me how to include ".srt" subtitles to this, so that stream could be added in output video.. thanks.. with regards, pekon From rogerdpack2 at gmail.com Thu Sep 1 16:33:42 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Thu, 1 Sep 2011 08:33:42 -0600 Subject: [FFmpeg-user] ffmpeg: encoding sub-titles In-Reply-To: References: Message-ID: > (1) Is there a way to hard-code the ".srt" (text based subtitles) into the > output video while encoding it ? > ? ? If yes, then please illustrate it with command-line example.. http://forums.creativecow.net/thread/291/155 may be helpful for you. From newton-san at rambler.ru Thu Sep 1 19:31:34 2011 From: newton-san at rambler.ru (Newton-san) Date: Thu, 1 Sep 2011 21:31:34 +0400 Subject: [FFmpeg-user] ffmpeg: encoding sub-titles In-Reply-To: References: Message-ID: <20110901213134.efa7d55d.newton-san@rambler.ru> On Thu, 1 Sep 2011 18:40:40 +0530 Pekon Gupta wrote: > Hi, > > Apologies for revisiting this question.. As i have googled a lot on this > topic, and still could not find suitable answer.. > > (1) Is there a way to hard-code the ".srt" (text based subtitles) into the > output video while encoding it ? > If yes, then please illustrate it with command-line example.. > > (2) how is "-newsubtitle" switch is used ? > > Im using following command-line.. > "ffmpeg -i in.avi -acodec libmp3lame -vcodec libxvid -bf 0 out.avi " > > Please tell me how to include ".srt" subtitles to this, so that stream could > be added in output video.. > > thanks.. > > with regards, pekon > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user You may use mplayer output & named pipe. Example: -========================================== mkfifo temp.yuv FNAME=in # input video-file, without extension # run mplayer in background mplayer -noass -fontconfig -font "DejaVu Sans" -subfont-autoscale 1 -subfont-text-scale 4.2 -subcp UTF-8 -subfont-outline 4 -subpos 100 -sub $FNAME.ass $FNAME.avi -vo yuv4mpeg:file=temp.yuv -noframedrop -ao null -really-quiet From narffy at gmail.com Thu Sep 1 21:00:44 2011 From: narffy at gmail.com (Narf!) Date: Thu, 1 Sep 2011 12:00:44 -0700 Subject: [FFmpeg-user] Large negative pts / dts from IP cameras Message-ID: I'm trying to capture the h.264 rtsp streams from some Vivotek FD8161 and FD8134 cameras. Usually ffmpeg can read the streams correctly. However, sometimes I get an error like: "Application provided invalid, non monotonically increasing dts to muxer in stream 0: 36000 >= -102295557451260" I'm using probesize so I don't have to wait for the probe buffer to fill up. When using ffprobe, you can see the pts and dts suddenly change into large negative values on the packets. Wireshark shows the time values correctly on the packets. Thanks! FFmpeg: ffmpeg -probesize 50 -loglevel debug -i rtsp://10.0.0.229/live.sdp -vcodec copy -an -y -f avi /tmp/test.avi ffmpeg version N-32141-g4becc86, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 29 2011 07:35:44 with gcc 4.1.2 20080704 (Red Hat 4.1.2-50) configuration: --datadir=/usr/share/ffmpeg --bindir=/usr/local/bin --libdir=/usr/local/lib --incdir=/usr/local/include --shlibdir=/usr/lib --mandir=/usr/share/man --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m32 -march=i386 -mtune=generic -fasynchronous-unwind-tables' --enable-avfilter --enable-libx264 --enable-gpl --enable-version3 --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-x11grab --enable-librtmp --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libmp3lame --disable-static --enable-libgsm --enable-libxvid --enable-libfaac --enable-nonfree libavutil 51. 14. 0 / 51. 14. 0 libavcodec 53. 12. 0 / 53. 12. 0 libavformat 53. 10. 0 / 53. 10. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 35. 0 / 2. 35. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [rtsp @ 0x9aaa9c0] SDP: v=0 o=RTSP 949361278 526 IN IP4 0.0.0.0 s=RTSP server c=IN IP4 0.0.0.0 t=0 0 a=charset:Shift_JIS a=range:npt=0- a=control:* a=etag:1234567890 m=video 0 RTP/AVP 98 b=AS:0 a=rtpmap:98 H264/90000 a=control:trackID=1 a=fmtp:98 packetization-mode=1; profile-level-id=4d001f; sprop-parameter-sets=J00AH9oCgPZA,KO48gA== [rtsp @ 0x9aaa9c0] video codec set to: h264 [NULL @ 0x9aae680] RTP Packetization Mode: 1 [NULL @ 0x9aae680] RTP Profile IDC: 4d Profile IOP: 0 Level: 1f [NULL @ 0x9aae680] Extradata set to 0x9aaeb80 (size: 19)!hello state=0 [h264 @ 0x9aae680] Unsupported bit depth: 0 [rtsp @ 0x9aaa9c0] Probe buffer size limit 50 reached [rtsp @ 0x9aaa9c0] Estimating duration from bitrate, this may be inaccurate Seems stream 0 codec frame rate differs from container frame rate: 180000.00 (180000/1) -> 90000.00 (180000/2) Input #0, rtsp, from 'rtsp://10.0.0.229/live.sdp': Metadata: title : RTSP server Duration: N/A, start: 0.000000, bitrate: N/A Stream #0.0, 1, 1/90000: Video: h264 (Main), yuv420p, 640x480, 1/180000, 90k tbr, 90k tbn, 180k tbc Output #0, avi, to '/tmp/test.avi': Metadata: INAM : RTSP server ISFT : Lavf53.10.0 Stream #0.0, 0, 1/90000: Video: h264 (H264 / 0x34363248), yuv420p, 640x480, 1/90000, q=2-31, 90k tbn, 90k tbc Stream mapping: Stream #0.0 -> #0.0: copy Press [q] to stop, [?] for help [avi @ 0x9aadca0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 36000 >= -102295549522620 av_interleaved_write_frame(): Invalid argument --------------------------------------------------------------------- ffprobe: ffprobe -show_packets -show_streams -pretty -loglevel debug -probesize 50 rtsp://10.0.0.229:554/live.sdp ffprobe version N-32153-gdb5effe, Copyright (c) 2007-2011 the FFmpeg developers built on Aug 30 2011 13:52:50 with gcc 4.1.2 20080704 (Red Hat 4.1.2-50) configuration: --datadir=/usr/share/ffmpeg --bindir=/usr/local/bin --libdir=/usr/local/lib --incdir=/usr/local/include --shlibdir=/usr/lib --mandir=/usr/share/man --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m32 -march=i386 -mtune=generic -fasynchronous-unwind-tables' --enable-avfilter --enable-libx264 --enable-gpl --enable-version3 --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-x11grab --enable-librtmp --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libmp3lame --disable-static --enable-libgsm --enable-libxvid --enable-libfaac --enable-nonfree libavutil 51. 14. 0 / 51. 14. 0 libavcodec 53. 12. 0 / 53. 12. 0 libavformat 53. 10. 0 / 53. 10. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 37. 0 / 2. 37. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [rtsp @ 0x84f29c0] SDP: v=0 o=RTSP 949358831 873 IN IP4 0.0.0.0 s=RTSP server c=IN IP4 0.0.0.0 t=0 0 a=charset:Shift_JIS a=range:npt=0- a=control:* a=etag:1234567890 m=video 0 RTP/AVP 98 b=AS:0 a=rtpmap:98 H264/90000 a=control:trackID=1 a=fmtp:98 packetization-mode=1; profile-level-id=4d001f; sprop-parameter-sets=J00AH9oCgPZA,KO48gA== [rtsp @ 0x84f29c0] video codec set to: h264 [NULL @ 0x84f6680] RTP Packetization Mode: 1 [NULL @ 0x84f6680] RTP Profile IDC: 4d Profile IOP: 0 Level: 1f [NULL @ 0x84f6680] Extradata set to 0x84f6b80 (size: 19)!hello state=0 [h264 @ 0x84f6680] Unsupported bit depth: 0 [rtsp @ 0x84f29c0] Probe buffer size limit 50 reached [rtsp @ 0x84f29c0] Estimating duration from bitrate, this may be inaccurate Input #0, rtsp, from 'rtsp://10.0.0.229:554/live.sdp': Metadata: title : RTSP server Duration: N/A, start: 0.000000, bitrate: N/A Stream #0.0, 1, 1/90000: Video: h264 (Main), yuv420p, 640x480, 1/180000, 90k tbr, 90k tbn, 180k tbc [PACKET] codec_type=video stream_index=0 pts=N/A pts_time=N/A dts=N/A dts_time=N/A duration=0 duration_time=0:00:00.000000 size=13.749 Kibyte pos=-1 flags=K [/PACKET] [PACKET] codec_type=video stream_index=0 pts=9000 pts_time=0:00:00.100000 dts=9000 dts_time=0:00:00.100000 duration=0 duration_time=0:00:00.000000 size=1.005 Kibyte pos=-1 flags=_ [/PACKET] [PACKET] codec_type=video stream_index=0 pts=18000 pts_time=0:00:00.200000 dts=18000 dts_time=0:00:00.200000 duration=0 duration_time=0:00:00.000000 size=562.000 byte pos=-1 flags=_ [/PACKET] [PACKET] codec_type=video stream_index=0 pts=27000 pts_time=0:00:00.300000 dts=27000 dts_time=0:00:00.300000 duration=0 duration_time=0:00:00.000000 size=483.000 byte pos=-1 flags=_ [/PACKET] [PACKET] codec_type=video stream_index=0 pts=36000 pts_time=0:00:00.400000 dts=36000 dts_time=0:00:00.400000 duration=0 duration_time=0:00:00.000000 size=355.000 byte pos=-1 flags=_ [/PACKET] [PACKET] codec_type=video stream_index=0 pts=-102295769715270 pts_time=-315727:-41:-3.503000 dts=-102295769715270 dts_time=-315727:-41:-3.503000 duration=0 duration_time=0:00:00.000000 size=13.724 Kibyte pos=-1 flags=K [/PACKET] [PACKET] codec_type=video stream_index=0 pts=-102295769706270 pts_time=-315727:-41:-3.403000 dts=-102295769706270 dts_time=-315727:-41:-3.403000 duration=0 duration_time=0:00:00.000000 size=1.078 Kibyte pos=-1 flags=_ [/PACKET] [PACKET] codec_type=video stream_index=0 pts=-102295769697270 pts_time=-315727:-41:-3.303000 dts=-102295769697270 dts_time=-315727:-41:-3.303000 duration=0 duration_time=0:00:00.000000 size=631.000 byte pos=-1 flags=_ [/PACKET] [PACKET] codec_type=video stream_index=0 pts=-102295769688270 pts_time=-315727:-41:-3.203000 dts=-102295769688270 dts_time=-315727:-41:-3.203000 duration=0 duration_time=0:00:00.000000 size=470.000 byte pos=-1 flags=_ [/PACKET] [PACKET] codec_type=video stream_index=0 pts=-102295769679270 pts_time=-315727:-41:-3.103000 dts=-102295769679270 dts_time=-315727:-41:-3.103000 duration=0 duration_time=0:00:00.000000 size=401.000 byte pos=-1 flags=_ [/PACKET] [PACKET] codec_type=video stream_index=0 pts=-102295769670270 pts_time=-315727:-41:-3.003000 dts=-102295769670270 dts_time=-315727:-41:-3.003000 duration=0 duration_time=0:00:00.000000 size=13.799 Kibyte pos=-1 flags=K [/PACKET] [PACKET] codec_type=video stream_index=0 pts=-102295769661270 pts_time=-315727:-41:-2.903000 dts=-102295769661270 dts_time=-315727:-41:-2.903000 duration=0 duration_time=0:00:00.000000 size=964.000 byte pos=-1 flags=_ [/PACKET] [PACKET] codec_type=video stream_index=0 pts=-102295769652270 pts_time=-315727:-41:-2.803000 dts=-102295769652270 dts_time=-315727:-41:-2.803000 duration=0 duration_time=0:00:00.000000 size=623.000 byte pos=-1 flags=_ [/PACKET] From rogerdpack2 at gmail.com Thu Sep 1 22:24:29 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Thu, 1 Sep 2011 14:24:29 -0600 Subject: [FFmpeg-user] Large negative pts / dts from IP cameras In-Reply-To: References: Message-ID: > "Application provided invalid, non monotonically increasing dts to > muxer in stream 0: 36000 >= -102295557451260" integer overflow? From pekon.gupta at gmail.com Thu Sep 1 22:43:05 2011 From: pekon.gupta at gmail.com (Pekon Gupta) Date: Fri, 2 Sep 2011 02:13:05 +0530 Subject: [FFmpeg-user] ffmpeg: encoding sub-titles In-Reply-To: <20110901213134.efa7d55d.newton-san@rambler.ru> References: <20110901213134.efa7d55d.newton-san@rambler.ru> Message-ID: hi, Thanks for reply..But the "-newsubtitle" switch of FFMPEG doesn't seem to work.. So, i used mencoder to first embed the sub-titles in video.. then used ffmpeg to transcode the video to different format.. 1) mencoder INFILE.avi -oac copy -ovc lavc -sub -font /usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf -subfont-autoscale 2 -o OUTFILE_WITH_SUBTITLES.avi 2) ffmpeg -i INFILE_WITH_SUBTITLES -s qvga -acodec libmp3lame -vcodec libxvid -bf 0 OUTFILE_WITH_SUBTITLES_XVID.avi However, i wished if ffmpeg could do this in single go.. anyways my work done for time being.. thanks again with regards, pekon From narffy at gmail.com Thu Sep 1 23:10:26 2011 From: narffy at gmail.com (Narf!) Date: Thu, 1 Sep 2011 14:10:26 -0700 Subject: [FFmpeg-user] Large negative pts / dts from IP cameras In-Reply-To: References: Message-ID: That's what I keep thinking too. However, the next timestamp value in Wireshark is small (like 45000) and I believe the pts is a 64 bit integer. On Thu, Sep 1, 2011 at 1:24 PM, Roger Pack wrote: >> "Application provided invalid, non monotonically increasing dts to >> muxer in stream 0: 36000 >= -102295557451260" > > integer overflow? > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From g at 8vue.com Thu Sep 1 23:15:05 2011 From: g at 8vue.com (G) Date: Thu, 01 Sep 2011 22:15:05 +0100 Subject: [FFmpeg-user] scaling and overlaying a png watermark to video Message-ID: <4E5FF5D9.3060100@8vue.com> Hi, so I'm trying to resize a png watermark image, and then overlay it to an video like so: ffmpeg -i input.mpg -vf "movie=wm.png [watermark];[in][watermark] scale=120:120,overlay=10:10 [out]" output/test.mpg However, I'm getting the error: Too many inputs specified for the scale filter. If I remove the scale option, it watermarks fine (as long as the wm.png is smaller than the video) ffmpeg version 0.8.2.git-8af9366 How do I scale it down and overlay the watermark? Thanks From stefano.sabatini-lala at poste.it Fri Sep 2 00:21:46 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Fri, 2 Sep 2011 00:21:46 +0200 Subject: [FFmpeg-user] scaling and overlaying a png watermark to video In-Reply-To: <4E5FF5D9.3060100@8vue.com> References: <4E5FF5D9.3060100@8vue.com> Message-ID: <20110901222146.GA22964@geppetto> On date Thursday 2011-09-01 22:15:05 +0100, G encoded: > Hi, so I'm trying to resize a png watermark image, and then overlay > it to an video like so: > > ffmpeg -i input.mpg -vf "movie=wm.png [watermark];[in][watermark] > scale=120:120,overlay=10:10 [out]" output/test.mpg [in][watermark]scale=120:120 you're specifying two inputs for the scale filter, which only accepts a single input. I suppose you mean: movie=wm.png, scale=120:120 [watermark]; [in][watermark] overlay=10:10 [out] -- ffmpeg-user random tip #4 ffmpeg -i FILE prints information about the container format and the streams in a file. From rhodri at kynesim.co.uk Fri Sep 2 12:58:49 2011 From: rhodri at kynesim.co.uk (Rhodri James) Date: Fri, 02 Sep 2011 11:58:49 +0100 Subject: [FFmpeg-user] Large negative pts / dts from IP cameras In-Reply-To: References: Message-ID: On Thu, 01 Sep 2011 22:10:26 +0100, Narf! wrote: > On Thu, Sep 1, 2011 at 1:24 PM, Roger Pack wrote: >>> "Application provided invalid, non monotonically increasing dts to >>> muxer in stream 0: 36000 >= -102295557451260" >> >> integer overflow? > > That's what I keep thinking too. However, the next timestamp value in > Wireshark is small (like 45000) and I believe the pts is a 64 bit > integer. PTS is a 33-bit integer in H.222. (Yes, I mean 33. It's weird.) Possibly something sign-extended that shouldn't have? -- Rhodri James Kynesim Ltd From mace.francois at gmail.com Fri Sep 2 13:40:02 2011 From: mace.francois at gmail.com (FrancoisMace) Date: Fri, 2 Sep 2011 04:40:02 -0700 (PDT) Subject: [FFmpeg-user] Generating v210 uncompressed AVIs from v210 individual files in 720p and 1080i formats Message-ID: <1314963602407-3785848.post@n4.nabble.com> Hello everybody, I am currently trying to generate uncompressed AVIs from individual files containing fields (for 1080i) or frames (for 720p) formatted in v210. I have already found some references on how to use FFMPEG to achieve generating an uncompressed AVI with v210 formats, but I have ran into two problems and I am hoping somebody might help me. 1. First of, for the generation on the uncompressed AVI in 1080i25, the issue I have is to know how I can actually generate the AVI with an aspect ratio of 1920x1080, knowing that my individual v210 files represent fields and how I can specify the actual field number of the first inserted field. I have used the following command: ffmpeg -s 1920x540 -vcodec v210 -f image2 -pix_fmt yuv422ple -i ./Decoded_5/NewMobCal_YUV10_1080i25//50000000/NM_1080i_%05d.v210 -vcodec v210 -r 25 -s 1920x540 -vframes 20 -y ./NewMobCal_YUV10_1080i25_50000.avi However, this generates a (working) AVI but with aspect ratio 1920x540 and where the image bounces a little bit up and downard (one frame over two) because fields are used as frames. 2. Secondly, when I try to generate a 720p AVI, I ran into an error. My original files are actually v210 with aspect ratio 1284x720 and I would like to be able to use the -cropright 4 option to come back to the actual 720p aspect ratio but I figured I should first manage to have the AVI generation working. My command line is the following: ffmpeg -s 1284x720 -vcodec v210 -f image2 -pix_fmt yuv422ple -i ./Decoded_5/NewMobCal_YUV10_720p50/50000000/NM_720p_%05d.v210 -vcodec v210 -r 50 -s 1284x720 -vframes 20 -y ./NewMobCal_YUV10_720p50_50000.avi And the error messages I got are: FFmpeg version SVN-r19283, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --disable-static --enable-shared libavutil 50. 3. 0 / 50. 3. 0 libavcodec 52.32. 0 / 52.32. 0 libavformat 52.36. 0 / 52.36. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0. 7. 1 / 0. 7. 1 built on Mar 23 2011 11:04:29, gcc: 4.4.3 Input #0, image2, from './Decoded_5/NewMobCal_YUV10_720p50/50000000/NM_720p_%05d.v210': Duration: 00:00:20.16, start: 0.000000, bitrate: N/A Stream #0.0: Video: v210, yuv422ple, 1284x720, 25 tbr, 25 tbn, 25 tbc [v210 @ 0x21d1030]bits per raw sample: 0 != 10-bit Output #0, avi, to './NewMobCal_YUV10_720p50_50000.avi': Stream #0.0: Video: v210, yuv422ple, 1284x720, q=2-31, 200 kb/s, 50 tbn, 50 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding [v210 @ 0x21cf9e0]packet too small Error while decoding stream #0.0 Error while decoding stream #0.0 Error while decoding stream #0.0 Error while decoding stream #0.0 Error while decoding stream #0.0 Error while decoding stream #0.0 Error while decoding stream #0.0 Error while decoding stream #0.0 Error while decoding stream #0.0 Error while decoding stream #0.0 Error while decoding stream #0.0 Error while decoding stream #0.0 ... Error while decoding stream #0.0 Error while decoding stream #0.0 Last message repeated 503 times [image2 @ 0x21ce390]Could not open file : ./Decoded_5/NewMobCal_YUV10_720p50/50000000/NM_720p_00504.v210 frame= 0 fps= 0 q=0.0 Lsize= 6kB time=10000000000.00 bitrate= 0.0kbits/s video:0kB audio:0kB global headers:0kB muxing overhead inf% My sequence has files starting with NM_720p_00000.v210 and ending at NM_720p_00503.v210. Each of this file is generatied with the same methodology as the one used for my 1080i fields (so I am pretty confident the files are ok) and the file size is 2465280 bytes (corresponding to the expected size of a v210 1284x720 image 1284x720/6*16). Thank you in advance for your help and suggestions. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Generating-v210-uncompressed-AVIs-from-v210-individual-files-in-720p-and-1080i-formats-tp3785848p3785848.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From tim.nicholson at bbc.co.uk Fri Sep 2 15:39:12 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Fri, 02 Sep 2011 14:39:12 +0100 Subject: [FFmpeg-user] Generating v210 uncompressed AVIs from v210 individual files in 720p and 1080i formats In-Reply-To: <1314963602407-3785848.post@n4.nabble.com> References: <1314963602407-3785848.post@n4.nabble.com> Message-ID: <4E60DC80.3050508@bbc.co.uk> On 02/09/11 12:40, FrancoisMace wrote: > Hello everybody, > > I am currently trying to generate uncompressed AVIs from individual files > containing fields (for 1080i) or frames (for 720p) formatted in v210. > I have already found some references on how to use FFMPEG to achieve > generating an uncompressed AVI with v210 formats, but I have ran into two > problems and I am hoping somebody might help me. > > 1. First of, for the generation on the uncompressed AVI in 1080i25, the > issue I have is to know how I can actually generate the AVI with an aspect > ratio of 1920x1080, knowing that my individual v210 files represent fields > and how I can specify the actual field number of the first inserted field. I > have used the following command: > > ffmpeg -s 1920x540 -vcodec v210 -f image2 -pix_fmt yuv422ple -i > ./Decoded_5/NewMobCal_YUV10_1080i25//50000000/NM_1080i_%05d.v210 -vcodec > v210 -r 25 -s 1920x540 -vframes 20 -y ./NewMobCal_YUV10_1080i25_50000.avi > > However, this generates a (working) AVI but with aspect ratio 1920x540 and > where the image bounces a little bit up and downard (one frame over two) > because fields are used as frames. I presume I'm missing something here because you specify an output size of 1920x540 and then complain when that's what you get..... As for turning frames into interlaced fields, thats one of those bits of magic that keeps slipping out of my personal FIFO so hopefully someone else can jog our memories.. -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From mace.francois at gmail.com Fri Sep 2 15:50:44 2011 From: mace.francois at gmail.com (FrancoisMace) Date: Fri, 2 Sep 2011 06:50:44 -0700 (PDT) Subject: [FFmpeg-user] Generating v210 uncompressed AVIs from v210 individual files in 720p and 1080i formats In-Reply-To: <4E60DC80.3050508@bbc.co.uk> References: <1314963602407-3785848.post@n4.nabble.com> <4E60DC80.3050508@bbc.co.uk> Message-ID: <1314971444894-3786119.post@n4.nabble.com> Hello Tim, Thanks for your quick answer. Indeed, my output image was specified as being 1920x540 since, when I specified 1920x1080, one hald of the ouptut AVI displayed completely green. So the command line was there in order to illustrate what I manage to get working, although what is slipping out of your mind correspond to what I have tried to get. The only notions of "interlacing" I found in the ffmpeg documentation seem to be related to the use of the AVC encoder/decoders which, since I am wanting to do an uncompressed AVI, do not interest me. PS: Any thought about the issue I am getting with the 720p ? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Generating-v210-uncompressed-AVIs-from-v210-individual-files-in-720p-and-1080i-formats-tp3785848p3786119.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From dev at rarevision.com Fri Sep 2 16:07:52 2011 From: dev at rarevision.com (Thomas Worth) Date: Fri, 2 Sep 2011 07:07:52 -0700 Subject: [FFmpeg-user] Generating v210 uncompressed AVIs from v210 individual files in 720p and 1080i formats In-Reply-To: <1314971444894-3786119.post@n4.nabble.com> References: <1314963602407-3785848.post@n4.nabble.com> <4E60DC80.3050508@bbc.co.uk> <1314971444894-3786119.post@n4.nabble.com> Message-ID: On Fri, Sep 2, 2011 at 6:50 AM, FrancoisMace wrote: > Hello Tim, > > Thanks for your quick answer. > > Indeed, my output image was specified as being 1920x540 since, when I > specified 1920x1080, one hald of the ouptut AVI displayed completely green. > > So the command line was there in order to illustrate what I manage to get > working, although what is slipping out of your mind correspond to what I > have tried to get. > > The only notions of "interlacing" I found in the ffmpeg documentation seem > to be related to the use of the AVC encoder/decoders which, since I am > wanting to do an uncompressed AVI, do not interest me. It sounds like you are trying to extract fields from an interlaced video and convert them to progressive (50i --> 50p). Am I correct? If so, I don't think it's as simple as scaling each field vertically to 1080 lines because you will get that jittery effect due to the fields being out of "registration." If I remember correctly, there is a bob filter in VirtualDub that will perform a shift to better align the fields and get rid of most of the jitter. That's a necessary step, and unless you do it prior to encoding with FFmpeg I don't think it'll look right. From tim.nicholson at bbc.co.uk Fri Sep 2 16:20:15 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Fri, 02 Sep 2011 15:20:15 +0100 Subject: [FFmpeg-user] Generating v210 uncompressed AVIs from v210 individual files in 720p and 1080i formats In-Reply-To: References: <1314963602407-3785848.post@n4.nabble.com> <4E60DC80.3050508@bbc.co.uk> <1314971444894-3786119.post@n4.nabble.com> Message-ID: <4E60E61F.3020809@bbc.co.uk> On 02/09/11 15:07, Thomas Worth wrote: > On Fri, Sep 2, 2011 at 6:50 AM, FrancoisMace wrote: >> Hello Tim, >> >> Thanks for your quick answer. >> >> Indeed, my output image was specified as being 1920x540 since, when I >> specified 1920x1080, one hald of the ouptut AVI displayed completely green. Because you only supplied 540 lines of input... >> >> So the command line was there in order to illustrate what I manage to get >> working, although what is slipping out of your mind correspond to what I >> have tried to get. >> >> The only notions of "interlacing" I found in the ffmpeg documentation seem >> to be related to the use of the AVC encoder/decoders which, since I am >> wanting to do an uncompressed AVI, do not interest me. > > It sounds like you are trying to extract fields from an interlaced > video and convert them to progressive (50i --> 50p). Am I correct? If Thomas I think he has already extracted the individual fields and they now exist as discrete image files. Now he wants to turn them back into an interlaced video. ISTR that mencoder managed that somewhat better than ffmpeg, but now ffmpeg supports mencoder filters it may be possible directly. I think having the input rate twice the output rate was involved somewhere. Someone must remember. However Francois is using a very old version of ffmpeg. > so, I don't think it's as simple as scaling each field vertically to > 1080 lines because you will get that jittery effect due to the fields > being out of "registration." If I remember correctly, there is a bob > filter in VirtualDub that will perform a shift to better align the > fields and get rid of most of the jitter. That's a necessary step, and > unless you do it prior to encoding with FFmpeg I don't think it'll > look right. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From mace.francois at gmail.com Fri Sep 2 17:48:42 2011 From: mace.francois at gmail.com (FrancoisMace) Date: Fri, 2 Sep 2011 08:48:42 -0700 (PDT) Subject: [FFmpeg-user] Generating v210 uncompressed AVIs from v210 individual files in 720p and 1080i formats In-Reply-To: <1314963602407-3785848.post@n4.nabble.com> References: <1314963602407-3785848.post@n4.nabble.com> Message-ID: <1314978522782-3786445.post@n4.nabble.com> Tim is indeed right, each of my v210 files for the 1080i represents a separate field. I will try to take a look at a more recent version of ffmpeg and to see if I can get the mencoder filter to work. Thanks. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Generating-v210-uncompressed-AVIs-from-v210-individual-files-in-720p-and-1080i-formats-tp3785848p3786445.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From mark at richardsemail.net Fri Sep 2 17:54:20 2011 From: mark at richardsemail.net (Mark Richards) Date: Fri, 2 Sep 2011 09:54:20 -0600 Subject: [FFmpeg-user] ffmpeg and HE-AAC audio Message-ID: <008301cc6988$94ad35f0$be07a1d0$@RichardsEMail.Net> I need to encode a mp4, h.264 with HE-AAC audio. I am having trouble finding what the parameters are to specify it. Can someone help me out ? Thanks, Mark Richards From batguano999 at hotmail.com Fri Sep 2 18:23:26 2011 From: batguano999 at hotmail.com (bat guano) Date: Fri, 2 Sep 2011 16:23:26 +0000 Subject: [FFmpeg-user] ffmpeg and HE-AAC audio In-Reply-To: <008301cc6988$94ad35f0$be07a1d0$@RichardsEMail.Net> References: <008301cc6988$94ad35f0$be07a1d0$@RichardsEMail.Net> Message-ID: > Subject: [FFmpeg-user] ffmpeg and HE-AAC audio > > I need to encode a mp4, h.264 with HE-AAC audio. I am having trouble > finding what the parameters are to specify it. Can someone help me out ? > > Thanks, > Mark Richards Hi (AFAIK) FFmpeg hasn't got an HE-AAC encoder. It uses libfaac which is based on FAAC encoder. http://en.wikipedia.org/wiki/FAAC From mayank.kargeti at evontech.com Fri Sep 2 09:19:56 2011 From: mayank.kargeti at evontech.com (mayank kargeti) Date: Fri, 2 Sep 2011 12:49:56 +0530 Subject: [FFmpeg-user] Unable to convert 3gp files to flv Message-ID: Hello we are unable to convert 3gp files to flv using ffmpeg . It is giving error of unsupported codec format i.e(Unsupported codec (id=73728) for input stream #0.1).I am working it on window server.Please reply asap -- Best Regards, *Mayank Kargeti* Software Developer *Evon Technologies* | Phone O: +91 135 65 41801 | Website: http://www.evontech.com This message and any attachments are solely for the intended recipient and may contain Evon Technologies confidential or privileged information. If you are not the intended recipient,any disclosure,copying, use, or distribution of the information included in this message and any attachments is prohibited. If you have received this communication in error, please notify us by reply e-mail at (administrator at evontech.com) and permanently delete this message and any attachments. Thank you. From bmurphy1976 at gmail.com Fri Sep 2 21:03:11 2011 From: bmurphy1976 at gmail.com (Bryan Murphy) Date: Fri, 2 Sep 2011 14:03:11 -0500 Subject: [FFmpeg-user] High Quality Single Pass Encodes Message-ID: Our encodes are targeted for streaming to various devices, so we've traditionally relied very heavily on fixed bitrates and two pass. For one customer, I'm seriously considering switching to one-pass encodes and using a variable bitrate. My problem is that when I've tried one-pass encodes in the past, scene changes in particular come out poorly. What settings should I be focusing on to smooth out scene changes, high motion sequences, and make single pass encodes in general look as good as we can get them? Our primary target is a 1280x720 1.5mbps h264 baseline encoded video. Anything we can do to speed up this transcode without significantly effecting video quality would be very helpful. Thanks! Bryan From mark at richardsemail.net Fri Sep 2 21:03:50 2011 From: mark at richardsemail.net (Mark Richards) Date: Fri, 2 Sep 2011 13:03:50 -0600 Subject: [FFmpeg-user] Unable to convert 3gp files to flv In-Reply-To: References: Message-ID: <009301cc69a3$0e1fc2d0$2a5f4870$@RichardsEMail.Net> These are the parameters I use to convert to FLV 8 in general: -vcodec flv -f flv -r 24 -s 640x360 -b 500k -cmp dct -subcmp dct -mbd 2 -flags +aic+cbp+mv0+mv4 -trellis 1 -ac 1 -ar 44100 -ab 96k -threads 0 Hope that is helpful. Mark -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of mayank kargeti Sent: Friday, September 02, 2011 1:20 AM To: ffmpeg-user at ffmpeg.org Subject: [FFmpeg-user] Unable to convert 3gp files to flv Hello we are unable to convert 3gp files to flv using ffmpeg . It is giving error of unsupported codec format i.e(Unsupported codec (id=73728) for input stream #0.1).I am working it on window server.Please reply asap -- Best Regards, *Mayank Kargeti* Software Developer *Evon Technologies* | Phone O: +91 135 65 41801 | Website: http://www.evontech.com This message and any attachments are solely for the intended recipient and may contain Evon Technologies confidential or privileged information. If you are not the intended recipient,any disclosure,copying, use, or distribution of the information included in this message and any attachments is prohibited. If you have received this communication in error, please notify us by reply e-mail at (administrator at evontech.com) and permanently delete this message and any attachments. Thank you. _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From mark at richardsemail.net Fri Sep 2 21:17:27 2011 From: mark at richardsemail.net (Mark Richards) Date: Fri, 2 Sep 2011 13:17:27 -0600 Subject: [FFmpeg-user] Padding in ffmpeg Message-ID: <009701cc69a4$f4ef53a0$decdfae0$@RichardsEMail.Net> Hello all, I am trying to understand the new padding syntax, but I am struggling to get it to work. My source video is 720x576 (5:4) My output size is 640x360 (16:9) The source scaled down to 360 pixels high is 450x360. I want to pillarbox the video, so it ends up with a 94 pixel pillar on each side of the video. Since I want the output to be 640x360, these are the params I tried (roughly) -vcodec libx264 -fpre libx264-fast.ffpreset -fpre libx264-main.ffpreset -aspect 16:9 -b 600k ........ -aspect 16:9 -w 640 -h 360 -vf pad=640:360:94:0:red But, ffmpeg gives me back this error. Input area 94:0:734:360 not within the padded area 0:0:440:360 or zero-sized I guess I clearly don't understand the documentation, cause I've tried every combination of parameters I can think of, and can't get around that message. I tried the formula from the documentation, and while it runs, it gives me a 640x640 video that is completely squashed. vf pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2" You help is greatly appreciated! Mark From jonathanbusby at gmail.com Fri Sep 2 21:47:20 2011 From: jonathanbusby at gmail.com (Jonathan Busby) Date: Fri, 2 Sep 2011 14:47:20 -0500 Subject: [FFmpeg-user] Need help with streaming raw h264 video from N900 to Linux PC using Gstreamer and ffdec_h264 Message-ID: Note that I posted this in gstreamer-devel but I have received no responses. I'm not sure if this is the appropriate forum but it's the closest to the subject at hand. ---------------------------------------------------------------- I'm trying to find a less bandwidth intensive way of streaming video from my Nokia N900's main back camera to my desktop other than sending it as a stream of jpeg frames. I thought the following would work : N900 : gst-launch v4l2camsrc device=/dev/video0 ! ffmpegcolorspace ! dsph264enc ! tcpserversink host=aleph1.clarkconnect.lan port=5000 PC : gst-launch tcpclientsrc host=192.168.1.179 port=5000 ! ffdec_h264 ! ffmpegcolorspace ! autovideosink but on the PC side I get this error : Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /GstPipeline:pipeline0/ffdec_ h264:ffdec_h2640: Internal GStreamer error: negotiation problem. Please file a bug at http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer. Additional debug info: gstffmpegdec.c(2684): gst_ffmpegdec_chain (): /GstPipeline:pipeline0/ffdec_h264:ffdec_h2640: ffdec_h264: input format was not set before data start ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ... Might anyone know what is going on above? Thanks, Jonathan From ted at tedpavlic.com Fri Sep 2 22:29:17 2011 From: ted at tedpavlic.com (Ted Pavlic) Date: Fri, 2 Sep 2011 16:29:17 -0400 Subject: [FFmpeg-user] ffplay disables DPMS timeouts; How to enable? (e.g., for electricsheep) Message-ID: I've noticed that my DPMS timeouts are never triggered while electricsheep runs despite "xset q" showing that DPMS is enabled. Drilling down further, I see that running mplayer actually causes the problem. Moreover, ffplay causes the same problem. If I'm playing a video with ffplay, how can I tell ffplay to *not* disable DPMS so that my screen will be blanked/etc. on schedule? Thanks -- Ted -- Ted Pavlic From mahu_berehat at yahoo.co.jp Sat Sep 3 17:00:57 2011 From: mahu_berehat at yahoo.co.jp (orang Aumori Jepun) Date: Sun, 4 Sep 2011 00:00:57 +0900 (JST) Subject: [FFmpeg-user] How to achieve old -croptop Message-ID: <20110903150057.19731.qmail@web3513.mail.bbt.yahoo.co.jp> Hi all. I recently installed the latest ffmpeg. And I noticed the parameters are little bit changed from the old debian ones. Now , I want to know how to achieve the old style ffmpeg -i INFILE.mpg -croptop 50 test.mpg with the latest ffmpeg ( as of 27.Aug.2011)? I'm not a native English speaker and I can not succeed in understanging the latest ffmpeg man page. So , someone please help me understand set up the parameter for the latest ffmpeg. T.I.A From coniophora at gmail.com Sat Sep 3 18:25:06 2011 From: coniophora at gmail.com (Jim Worrall) Date: Sat, 3 Sep 2011 10:25:06 -0600 Subject: [FFmpeg-user] Padding in ffmpeg In-Reply-To: <009701cc69a4$f4ef53a0$decdfae0$@RichardsEMail.Net> References: <009701cc69a4$f4ef53a0$decdfae0$@RichardsEMail.Net> Message-ID: <35169C72-4CF6-497C-9F56-6A8D7074654B@gmail.com> I'm far from an expert, but suspect you may need to scale the video down first, then add the padding. You might try -s WxH, or -vf scale=W:H I don't know what the difference is. I would try to add one of these before your -vf pad= On Sep 2, 2011, at 1:17 PM, Mark Richards wrote: Hello all, I am trying to understand the new padding syntax, but I am struggling to get it to work. My source video is 720x576 (5:4) My output size is 640x360 (16:9) The source scaled down to 360 pixels high is 450x360. I want to pillarbox the video, so it ends up with a 94 pixel pillar on each side of the video. Since I want the output to be 640x360, these are the params I tried (roughly) -vcodec libx264 -fpre libx264-fast.ffpreset -fpre libx264-main.ffpreset -aspect 16:9 -b 600k ........ -aspect 16:9 -w 640 -h 360 -vf pad=640:360:94:0:red But, ffmpeg gives me back this error. Input area 94:0:734:360 not within the padded area 0:0:440:360 or zero-sized I guess I clearly don't understand the documentation, cause I've tried every combination of parameters I can think of, and can't get around that message. I tried the formula from the documentation, and while it runs, it gives me a 640x640 video that is completely squashed. vf pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2" You help is greatly appreciated! Mark _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From coniophora at gmail.com Sat Sep 3 18:44:52 2011 From: coniophora at gmail.com (Jim Worrall) Date: Sat, 3 Sep 2011 10:44:52 -0600 Subject: [FFmpeg-user] How to achieve old -croptop In-Reply-To: <20110903150057.19731.qmail@web3513.mail.bbt.yahoo.co.jp> References: <20110903150057.19731.qmail@web3513.mail.bbt.yahoo.co.jp> Message-ID: On Sep 3, 2011, at 9:00 AM, orang Aumori Jepun wrote: > > Hi all. > > I recently installed the latest ffmpeg. > And I noticed the parameters are little bit changed from > the old debian ones. > > Now , I want to know how to achieve the old style > ffmpeg -i INFILE.mpg -croptop 50 test.mpg > with the latest ffmpeg ( as of 27.Aug.2011)? > > I'm not a native English speaker and I can not succeed in > understanging the latest ffmpeg man page. > > So , someone please help me understand set up the > parameter for the latest ffmpeg. > > T.I.A > As I understand, -vf crop=out_w:out_h:x:y out_w is the width in pixels of the final output you want out_h is the height in pixels of the final output you want and these are based on the ORIGINAL frame: x is the pixel position, counting from the left side of the original frame, of the left side of the frame you want to keep y is the pixel position, counting from the top of the original frame, of the top side of the frame you want to keep So, for example, -vf crop=640:480:60:40 will find the point that is 60 pixels from the left side of the original and 40 pixels from the top of the original, and then save a frame that is 640x480 from there, with that point as the upper left corner. If x and y are not given, it defaults to centering the final frame in the original frame. From belcampo at zonnet.nl Sat Sep 3 18:50:02 2011 From: belcampo at zonnet.nl (belcampo) Date: Sat, 03 Sep 2011 18:50:02 +0200 Subject: [FFmpeg-user] High Quality Single Pass Encodes In-Reply-To: References: Message-ID: <4E625ABA.2030702@zonnet.nl> On 09/02/2011 09:03 PM, Bryan Murphy wrote: > Our encodes are targeted for streaming to various devices, so we've > traditionally relied very heavily on fixed bitrates and two pass. For > one customer, I'm seriously considering switching to one-pass encodes > and using a variable bitrate. > > My problem is that when I've tried one-pass encodes in the past, scene > changes in particular come out poorly. > > What settings should I be focusing on to smooth out scene changes, > high motion sequences, and make single pass encodes in general look as > good as we can get them? > > Our primary target is a 1280x720 1.5mbps h264 baseline encoded video. > Anything we can do to speed up this transcode without significantly > effecting video quality would be very helpful. "Our primary target is a 1280x720 1.5mbps h264 baseline" This is enough bitrate for 'talking heads', but definitely not for content with a lot of activity. The crf parameter is for quality, and it will take the bits it needs for that quality, talking heads probably =< 1.5Mb/s and > 3.5Mb/s for lots of activity and/or grainy source material. The target 'you set' is an more or less impossible one. Baseline needs more bits then HighLevel, but the playing device needs more 'horse-power' to decode. > > Thanks! > Bryan > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From rogerdpack2 at gmail.com Sat Sep 3 18:59:29 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Sat, 3 Sep 2011 10:59:29 -0600 Subject: [FFmpeg-user] Emulate IP web camera In-Reply-To: <00b001cc6887$ad617220$08245660$@net> References: <1314779906.39111.YahooMailNeo@web32505.mail.mud.yahoo.com> <21840DFC-A92A-4E0A-9480-C3B7EC2CAD97@pixelpartner.de> <1314785759.95527.YahooMailNeo@web32505.mail.mud.yahoo.com> <7D76607B-7DBC-4F8C-A0A5-7F0F5AEBB82B@pixelpartner.de> <00b001cc6887$ad617220$08245660$@net> Message-ID: > Example: ? ffmpeg -i file.mov ? -vcodec mpeg4 -an -s 176x144 -vcodec > rawvideo -f rtp rtp://127.0.0.1:5000/ two vcodecs? From orenelrad at gmail.com Sat Sep 3 21:15:05 2011 From: orenelrad at gmail.com (Oren Elrad) Date: Sat, 3 Sep 2011 15:15:05 -0400 Subject: [FFmpeg-user] Compiled latest-git ffmpeg with --enable-decoder=libfaad but then is an unknown decoder Message-ID: Hi all, Backstory: I'm trying to decode a Quicktime movie and I get a repeated stream error message until I ctrl-C (see http://lzone.de/aac+decoding+with+ffmpeg+doesn't+work ). That site suggest that I use acodec libfaad instead of the built in aac decoder, so I downloaded and compiled libfaad from latest source and then recompiled ffmpeg from the latest git like so: ./configure --enable-gpl --enable-nonfree --enable-version3 --disable-dxva2 --enable-libfaac --enable-libxvid --enable-pthreads --enable-decoder=libfaad --enable-libmp3lame --enable-encoder=libfaad && make -j12 install That completed with no error messages. Now when I try to follow the instructions there for using libfaad instead of acc, it cannot find it as a decoder, nor is faad listed anywhere on the -codecs XXXXX at YYYYYYY:~$ ffmpeg -acodec libfaad -i Most.mov -ss 00:01:00 -vframes 1 test.png ffmpeg version N-32299-g23a287e, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 3 2011 21:06:21 with gcc 4.3.2 configuration: --enable-gpl --enable-nonfree --enable-version3 --disable-dxva2 --enable-libfaac --enable-libxvid --enable-pthreads --enable-decoder=libfaad --enable-libmp3lame --enable-encoder=libfaad libavutil 51. 14. 0 / 51. 14. 0 libavcodec 53. 12. 0 / 53. 12. 0 libavformat 53. 10. 0 / 53. 10. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 37. 0 / 2. 37. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Unknown decoder 'libfaad' Trying without it gives the same symptoms in the lzone.de thread: XXXXX at YYYYYYY:~$ ffmpeg-i Most.mov -ss 00:01:00 -vframes 1test.png ffmpeg version N-32299-g23a287e, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 3 2011 21:06:21 with gcc 4.3.2 configuration: --enable-gpl --enable-nonfree --enable-version3 --disable-dxva2 --enable-libfaac --enable-libxvid --enable-pthreads --enable-decoder=libfaad --enable-libmp3lame --enable-encoder=libfaad libavutil 51. 14. 0 / 51. 14. 0 libavcodec 53. 12. 0 / 53. 12. 0 libavformat 53. 10. 0 / 53. 10. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 37. 0 / 2. 37. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 1 codec frame rate differs from container frame rate: 2997.00 (2997/1) -> 29.97 (30000/1001) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Most.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2009-11-07 23:30:44 Duration: 00:43:41.83, start: 0.000000, bitrate: 2302 kb/s Stream #0.0(eng): Audio: wmav2 (WMA2 / 0x32414D57), 48000 Hz, 2 channels, s16, 346 kb/s Metadata: creation_time : 2009-11-07 23:30:44 Stream #0.1(eng): Video: wmv3 (Main) (WMV3 / 0x33564D57), yuv420p, 480x360, 1952 kb/s, 27.19 fps, 29.97 tbr, 2997 tbn, 2997 tbc Metadata: creation_time : 2009-11-07 23:30:44 Incompatible pixel format 'yuv420p' for codec 'png', auto-selecting format 'rgb24' [buffer @ 0x13ddfc0] w:480 h:360 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param: [buffersink @ 0x13e0a20] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' [scale @ 0x13d39a0] w:480 h:360 fmt:yuv420p -> w:480 h:360 fmt:rgb24 flags:0x4 Output #0, image2, to 'test.png': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2009-11-07 23:30:44 encoder : Lavf53.10.0 Stream #0.0(eng): Video: png, rgb24, 480x360, q=2-31, 200 kb/s, 90k tbn, 29.97 tbc Metadata: creation_time : 2009-11-07 23:30:44 Stream mapping: Stream #0.1 -> #0.0 (wmv3 -> png) Press [q] to stop, [?] for help Error while decoding stream #0.1 Error while decoding stream #0.1 Error while decoding stream #0.1 Error while decoding stream #0.1 [..... GOES ON UNTIL YOU CONTROL+C .... ] video:0kB audio:0kB global headers:0kB muxing overhead -inf% Received signal 2: terminating. Thanks for reading! From batguano999 at hotmail.com Sun Sep 4 00:28:59 2011 From: batguano999 at hotmail.com (bat guano) Date: Sat, 3 Sep 2011 22:28:59 +0000 Subject: [FFmpeg-user] Compiled latest-git ffmpeg with --enable-decoder=libfaad but then is an unknown decoder In-Reply-To: References: Message-ID: ---------------------------------------- > > Backstory: I'm trying to decode a Quicktime movie and I get a repeated > stream error message until I ctrl-C (see > http://lzone.de/aac+decoding+with+ffmpeg+doesn't+work ). That site > suggest that I use acodec libfaad instead of the built in aac decoder, Hi The information on the lzone.de webpage doesn't make any sense to me. And this means nothing:- "Backstory: I'm trying to decode a Quicktime movie". From lou at lrcd.com Sun Sep 4 03:03:34 2011 From: lou at lrcd.com (Lou) Date: Sat, 3 Sep 2011 17:03:34 -0800 Subject: [FFmpeg-user] Compiled latest-git ffmpeg with --enable-decoder=libfaad but then is an unknown decoder In-Reply-To: References: Message-ID: <20110903170334.62c30613@lrcd.com> On Sat, 3 Sep 2011 15:15:05 -0400 Oren Elrad wrote: > Now when I try to follow the instructions there for using libfaad > instead of acc, it cannot find it as a decoder, nor is faad listed > anywhere on the -codecs > > XXXXX at YYYYYYY:~$ ffmpeg -acodec libfaad -i Most.mov -ss 00:01:00 > -vframes 1 test.png ... > Unknown decoder 'libfaad' Support for libfaad in FFmpeg was dropped in 2010-06-20 in favor of the native FFmpeg AAC decoder. > Trying without it gives the same symptoms in the lzone.de thread: > > XXXXX at YYYYYYY:~$ ffmpeg-i Most.mov -ss 00:01:00 -vframes 1test.png > ffmpeg version N-32299-g23a287e, Copyright (c) 2000-2011 the FFmpeg > developers built on Sep 3 2011 21:06:21 with gcc 4.3.2 > configuration: --enable-gpl --enable-nonfree --enable-version3 > --disable-dxva2 --enable-libfaac --enable-libxvid --enable-pthreads > --enable-decoder=libfaad --enable-libmp3lame --enable-encoder=libfaad > libavutil 51. 14. 0 / 51. 14. 0 > libavcodec 53. 12. 0 / 53. 12. 0 > libavformat 53. 10. 0 / 53. 10. 0 > libavdevice 53. 3. 0 / 53. 3. 0 > libavfilter 2. 37. 0 / 2. 37. 0 > libswscale 2. 0. 0 / 2. 0. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > > Seems stream 1 codec frame rate differs from container frame rate: > 2997.00 (2997/1) -> 29.97 (30000/1001) > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Most.mov': > Metadata: > major_brand : qt > minor_version : 537199360 > compatible_brands: qt > creation_time : 2009-11-07 23:30:44 > Duration: 00:43:41.83, start: 0.000000, bitrate: 2302 kb/s > Stream #0.0(eng): Audio: wmav2 (WMA2 / 0x32414D57), 48000 Hz, 2 > channels, s16, 346 kb/s > Metadata: > creation_time : 2009-11-07 23:30:44 > Stream #0.1(eng): Video: wmv3 (Main) (WMV3 / 0x33564D57), yuv420p, > 480x360, 1952 kb/s, 27.19 fps, 29.97 tbr, 2997 tbn, 2997 tbc > Metadata: > creation_time : 2009-11-07 23:30:44 > Incompatible pixel format 'yuv420p' for codec 'png', auto-selecting > format 'rgb24' > [buffer @ 0x13ddfc0] w:480 h:360 pixfmt:yuv420p tb:1/1000000 sar:0/1 > sws_param: [buffersink @ 0x13e0a20] auto-inserting filter > 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' > [scale @ 0x13d39a0] w:480 h:360 fmt:yuv420p -> w:480 h:360 fmt:rgb24 > flags:0x4 Output #0, image2, to 'test.png': > Metadata: > major_brand : qt > minor_version : 537199360 > compatible_brands: qt > creation_time : 2009-11-07 23:30:44 > encoder : Lavf53.10.0 > Stream #0.0(eng): Video: png, rgb24, 480x360, q=2-31, 200 kb/s, > 90k tbn, 29.97 tbc > Metadata: > creation_time : 2009-11-07 23:30:44 > Stream mapping: > Stream #0.1 -> #0.0 (wmv3 -> png) > Press [q] to stop, [?] for help > Error while decoding stream #0.1 Unless FFmpeg is wrong then your input appears not to contain AAC. Your issue has nothing to do with AAC. FFmpeg is having trouble decoding stream 0.1 which is your video stream. Can you provide a sample of your input? From mahu_berehat at yahoo.co.jp Sun Sep 4 03:06:50 2011 From: mahu_berehat at yahoo.co.jp (orang Aumori Jepun) Date: Sun, 4 Sep 2011 10:06:50 +0900 (JST) Subject: [FFmpeg-user] How to achieve old -croptop In-Reply-To: Message-ID: <20110904010650.18539.qmail@web3503.mail.bbt.yahoo.co.jp> Hi, Jim. With your explanation , I could crop movies top and bottom with the latest ffmpeg. Thank yoiu very much!!! From akshar_tank at yahoo.com Sun Sep 4 06:59:17 2011 From: akshar_tank at yahoo.com (tank pranav) Date: Sat, 3 Sep 2011 21:59:17 -0700 (PDT) Subject: [FFmpeg-user] A doubt in generation of frames out of a ts file from ffmpeg. In-Reply-To: <20110903170334.62c30613@lrcd.com> References: <20110903170334.62c30613@lrcd.com> Message-ID: <1315112357.14953.YahooMailNeo@web122503.mail.ne1.yahoo.com> ?I am using ffmpeg for video ts file to image conversion by giving following command. ffmpeg -i video_file.ts destination_directory/img_%05d.jpeg it generates images in to ?destination_directory. But it generates some gray color frames in very begining part. and those number of gray color images is generated in every video ts files. number does not remain fixed , in some ts files those gray color images are 70 to 80, in some other video file it remains 2000-5000.? When I open that video ts file in to dvbcut then dvbcut only shows actual frames. Then I could not correlate what ffmpeg does and what dvbcut shows. In documentation of dvbcut says they are using ffmpeg as internal library Then why such thing should have happened ??? My conclusion :-? I think so, ffmpeg does not find where from 1st I frame starts ....... If it gives initial gray color frame then it definately means that it did not find I-frame thati s why ffmpeg keeps on inserting dummy frames. We need to change something in ffmpeg's?coding. Where should we change that I am still figuring out. If you have any idea how we should keep on drop the frame till we get I frame then it may solve my purpose. I dont know I am right or wrong, if u can help me out with this issue then it would be great. Thanks, Pranav. From mark at mdsh.com Sun Sep 4 11:12:37 2011 From: mark at mdsh.com (Mark Himsley) Date: Sun, 04 Sep 2011 10:12:37 +0100 Subject: [FFmpeg-user] Generating v210 uncompressed AVIs from v210 individual files in 720p and 1080i formats In-Reply-To: <4E60E61F.3020809@bbc.co.uk> References: <1314963602407-3785848.post@n4.nabble.com> <4E60DC80.3050508@bbc.co.uk> <1314971444894-3786119.post@n4.nabble.com> <4E60E61F.3020809@bbc.co.uk> Message-ID: <4E634105.2070903@mdsh.com> On 02/09/2011 15:20, Tim Nicholson wrote: > On 02/09/11 15:07, Thomas Worth wrote: >> On Fri, Sep 2, 2011 at 6:50 AM, FrancoisMace wrote: >>> Hello Tim, >>> >>> Thanks for your quick answer. >>> >>> Indeed, my output image was specified as being 1920x540 since, when I >>> specified 1920x1080, one hald of the ouptut AVI displayed completely green. > > Because you only supplied 540 lines of input... > >>> >>> So the command line was there in order to illustrate what I manage to get >>> working, although what is slipping out of your mind correspond to what I >>> have tried to get. >>> >>> The only notions of "interlacing" I found in the ffmpeg documentation seem >>> to be related to the use of the AVC encoder/decoders which, since I am >>> wanting to do an uncompressed AVI, do not interest me. >> >> It sounds like you are trying to extract fields from an interlaced >> video and convert them to progressive (50i --> 50p). Am I correct? If > > Thomas I think he has already extracted the individual fields and they > now exist as discrete image files. Now he wants to turn them back into > an interlaced video. > > ISTR that mencoder managed that somewhat better than ffmpeg, but now > ffmpeg supports mencoder filters it may be possible directly. I think > having the input rate twice the output rate was involved somewhere. > > Someone must remember. I haven't completely understood Francois' requirements, but if Francois is wanting to convert frames into fields then that's what the tinterlace filter does. There's the version in FFmbc, there's the version in the mp filters of FFmpeg (and therefore also in mencoder), and there's the patch that Stefano has posted to ffmpeg-devel - and which he said "I'm going to apply the patch in a few days if I see no comments..." a few days ago. If I've understood right, Francois has half-height frames that he wants to interlace, then he'll want tinterlace mode 0: "Move odd frames into the upper field of the new frame, even into the lower field, generating a double height frame at half framerate" > > However Francois is using a very old version of ffmpeg. > True enough. -- Mark From orenelrad at gmail.com Sun Sep 4 17:26:20 2011 From: orenelrad at gmail.com (Oren Elrad) Date: Sun, 4 Sep 2011 11:26:20 -0400 Subject: [FFmpeg-user] Compiled latest-git ffmpeg with --enable-decoder=libfaad but then is an unknown decoder Message-ID: OLD PARTS OF THE THREAD: On Sat, 3 Sep 2011 15:15:05 -0400 Oren Elrad wrote: > Now when I try to follow the instructions there for using libfaad > instead of acc, it cannot find it as a decoder, nor is faad listed > anywhere on the -codecs > > XXXXX at YYYYYYY:~$ ffmpeg -acodec libfaad -i Most.mov -ss 00:01:00 > -vframes 1 test.png ... > Unknown decoder 'libfaad' Support for libfaad in FFmpeg was dropped in 2010-06-20 in favor of the native FFmpeg AAC decoder. > Trying without it gives the same symptoms in the lzone.de thread: > > XXXXX at YYYYYYY:~$ ffmpeg-i Most.mov -ss 00:01:00 -vframes 1test.png > ffmpeg version N-32299-g23a287e, Copyright (c) 2000-2011 the FFmpeg > developers built on Sep ?3 2011 21:06:21 with gcc 4.3.2 > ? configuration: --enable-gpl --enable-nonfree --enable-version3 > --disable-dxva2 --enable-libfaac --enable-libxvid --enable-pthreads > --enable-decoder=libfaad --enable-libmp3lame --enable-encoder=libfaad > ? libavutil ? ?51. 14. 0 / 51. 14. 0 > ? libavcodec ? 53. 12. 0 / 53. 12. 0 > ? libavformat ?53. 10. 0 / 53. 10. 0 > ? libavdevice ?53. ?3. 0 / 53. ?3. 0 > ? libavfilter ? 2. 37. 0 / ?2. 37. 0 > ? libswscale ? ?2. ?0. 0 / ?2. ?0. 0 > ? libpostproc ?51. ?2. 0 / 51. ?2. 0 > > Seems stream 1 codec frame rate differs from container frame rate: > 2997.00 (2997/1) -> 29.97 (30000/1001) > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Most.mov': > ? Metadata: > ? ? major_brand ? ? : qt > ? ? minor_version ? : 537199360 > ? ? compatible_brands: qt > ? ? creation_time ? : 2009-11-07 23:30:44 > ? Duration: 00:43:41.83, start: 0.000000, bitrate: 2302 kb/s > ? ? Stream #0.0(eng): Audio: wmav2 (WMA2 / 0x32414D57), 48000 Hz, 2 > channels, s16, 346 kb/s > ? ? Metadata: > ? ? ? creation_time ? : 2009-11-07 23:30:44 > ? ? Stream #0.1(eng): Video: wmv3 (Main) (WMV3 / 0x33564D57), yuv420p, > 480x360, 1952 kb/s, 27.19 fps, 29.97 tbr, 2997 tbn, 2997 tbc > ? ? Metadata: > ? ? ? creation_time ? : 2009-11-07 23:30:44 > Incompatible pixel format 'yuv420p' for codec 'png', auto-selecting > format 'rgb24' > [buffer @ 0x13ddfc0] w:480 h:360 pixfmt:yuv420p tb:1/1000000 sar:0/1 > sws_param: [buffersink @ 0x13e0a20] auto-inserting filter > 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' > [scale @ 0x13d39a0] w:480 h:360 fmt:yuv420p -> w:480 h:360 fmt:rgb24 > flags:0x4 Output #0, image2, to 'test.png': > ? Metadata: > ? ? major_brand ? ? : qt > ? ? minor_version ? : 537199360 > ? ? compatible_brands: qt > ? ? creation_time ? : 2009-11-07 23:30:44 > ? ? encoder ? ? ? ? : Lavf53.10.0 > ? ? Stream #0.0(eng): Video: png, rgb24, 480x360, q=2-31, 200 kb/s, > 90k tbn, 29.97 tbc > ? ? Metadata: > ? ? ? creation_time ? : 2009-11-07 23:30:44 > Stream mapping: > ? Stream #0.1 -> #0.0 (wmv3 -> png) > Press [q] to stop, [?] for help > Error while decoding stream #0.1 Unless FFmpeg is wrong then your input appears not to contain AAC. Your issue has nothing to do with AAC. FFmpeg is having trouble decoding stream 0.1 which is your video stream. Can you provide a sample of your input? ------------------------------ NEW STUFF ----------------------------- Ah, I didn't think hard enough about that huh. Sorry to have provided a red herring for a solution with the same symptoms and different causes. By the way, perhaps ./configure --with-decoder=libfaad should fail with "this decoder is not supported" or some other message of that ilk? That would have definitely saved me from the wrong track. What do you mean by a 'sample'? I didn't encode this and all I have is a rather large file. I can dd the first N bytes if that will help ... Oren From maxence.rioblanc at ixmotion.com Sun Sep 4 12:13:29 2011 From: maxence.rioblanc at ixmotion.com (Maxence Rioblanc) Date: Sun, 04 Sep 2011 12:13:29 +0200 (CEST) Subject: [FFmpeg-user] Live streaming Message-ID: <8654D442-8AA5-43A5-9F13-5BAF947C2FB4@ixmotion.com> Dear All, I am currently working on a project where it is needed to stream a video from a camera. I am working with OpenCV, manage to capture image (opencv Mat format) , do some image treatment ( creating a new Mat object). I read a lot of topics but all were about converting existing movie or pictures (in bmp, jpeg format) to something else. Yet I did not find how to use ffmep on the fly with my raw buffer ( I can get image width, heigth, pixel depth, the Mat creation is working at 25Hz) and output a flv file which can be accessed by another computer (in my local network). Any idea or suggestion will be highly appreciated as I am a little bit stuck by this nowadays, Regards, Maxence From mace.francois at gmail.com Sun Sep 4 20:01:12 2011 From: mace.francois at gmail.com (FrancoisMace) Date: Sun, 4 Sep 2011 11:01:12 -0700 (PDT) Subject: [FFmpeg-user] Generating v210 uncompressed AVIs from v210 individual files in 720p and 1080i formats In-Reply-To: <4E634105.2070903@mdsh.com> References: <1314963602407-3785848.post@n4.nabble.com> <4E60DC80.3050508@bbc.co.uk> <1314971444894-3786119.post@n4.nabble.com> <4E60E61F.3020809@bbc.co.uk> <4E634105.2070903@mdsh.com> Message-ID: <1315159272754-3789686.post@n4.nabble.com> Hello Mark, Thank you for your answer. Your completely right about what I am trying to do: I have a bunch of files formated in v210. Each file refers to one of the two fields of the image, the even numbered one corresponds to the bottom field and the odd one correspond to the top field. I have taken a look at your suggestion to use the mp filter and to go for a more recent version of the ffmpeg software. To do so, I have downloaded an available windows build (the latest 32 bit static build from http://ffmpeg.zeranoe.com/builds/). However I am still experiencing some odd behavior: Here is the command line launched and the obtained result: ./ffmpeg.exe -s 1920x540 -vcodec v210 -f image2 -pix_fmt yuv422p16le -i "PATH_TO_FILES\ParkRun_YUV10_1080i25\PR_1080i_%05d.v210" -pix_fmt yuv422p10le -vcodec v210 -vf mp=tinterlace:0 -vcodec v210 -pix_fmt yuv422p16le -r 25 -s 1920x540 "test.avi" ffmpeg version N-32290-g6fc1299, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 2 2011 00:40:00 with gcc 4.6.1 configuration: --enable-gpl --enable-version3 --enable-memalign-hack --enable- runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libo pencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm -- enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enabl e-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 14. 0 / 51. 14. 0 libavcodec 53. 12. 0 / 53. 12. 0 libavformat 53. 10. 0 / 53. 10. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 37. 0 / 2. 37. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, image2, from 'PATH_TO_FILE\ParkRun_YUV10_1080i25\PR_1080i_%05d.v210': Duration: 00:00:20.16, start: 0.000000, bitrate: N/A Stream #0.0: Video: v210, yuv422p10le, 1920x540, 25 tbr, 25 tbn, 25 tbc File 'test.avi' already exists. Overwrite ? [y/N] y Incompatible pixel format 'yuv422p16le' for codec 'v210', auto-selecting format 'yuv422p10le' [buffer @ 01ED2500] w:1920 h:540 pixfmt:yuv422p10le tb:1/1000000 sar:0/1 sws_param: [mp @ 01ED2AC0] This is a unholy filter, it will be purified by the ffmpeg exorcist team which will change its syntax from dark -vf mp to light -vf. Thou shalst not make spells or scripts that depend on it [buffersink @ 01ED2780] auto-inserting filter 'auto-inserted scaler 0' between the filter 'Parsed_mp_0' and the filter 'out' [mp @ 01ED2AC0] auto-inserting filter 'auto-inserted scaler 1' between the filte r 'src' and the filter 'Parsed_mp_0' [scale @ 01ED3960] w:1920 h:540 fmt:yuv422p10le -> w:1920 h:540 fmt:yuv420p flags:0x4 [scale @ 01ED29A0] w:1920 h:1080 fmt:yuv420p -> w:1920 h:1080 fmt:yuv422p10le flags:0x4 Output #0, avi, to 'test.avi': Metadata: ISFT : Lavf53.10.0 Stream #0.0: Video: v210 (v210 / 0x30313276), yuv422p10le, 1920x1080, q=2-31 , 200 kb/s, 25 tbn, 25 tbc Stream mapping: Stream #0.0 -> #0.0 (v210 -> v210) Press [q] to stop, [?] for help [buffer @ 01ED2500] request_frame() called with no available frame! frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 bitrate=1107056.0kbit frame= 1 fps= 1 q=0.0 size= 5406kB time=00:00:00.04 bitrate=1107056.0kbit frame= 1 fps= 1 q=0.0 size= 5406kB time=00:00:00.04 bitrate=1107056.0kbit frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 bitrate=1107056.0kbit frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 bitrate=1107056.0kbit frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 bitrate=1107056.0kbit frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 bitrate=1107056.0kbit frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 bitrate=1107056.0kbit frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 bitrate=1107056.0kbit frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 bitrate=1107056.0kbit frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 bitrate=1107056.0kbit frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 bitrate=1107056.0kbit frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 bitrate=1107056.0kbit frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 bitrate=1107056.0kbit Last message repeated 46 times request_frame() called with no available frame! frame= 1 fps= 0 q=0.0 Lsize= 5406kB time=00:00:00.04 bitrate=1107060.8kbi ts/s dup=0 drop=46 Two behaviors I cannot explain: - I do not undersand why a scaler filter is automatically insterted two times (goint from 422 to 420 and back) - I do not understand why the output file size does not increase during the encoding process. Thanks for your help! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Generating-v210-uncompressed-AVIs-from-v210-individual-files-in-720p-and-1080i-formats-tp3785848p3789686.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From mark at mdsh.com Mon Sep 5 00:14:32 2011 From: mark at mdsh.com (Mark Himsley) Date: Sun, 04 Sep 2011 23:14:32 +0100 Subject: [FFmpeg-user] Generating v210 uncompressed AVIs from v210 individual files in 720p and 1080i formats In-Reply-To: <1315159272754-3789686.post@n4.nabble.com> References: <1314963602407-3785848.post@n4.nabble.com> <4E60DC80.3050508@bbc.co.uk> <1314971444894-3786119.post@n4.nabble.com> <4E60E61F.3020809@bbc.co.uk> <4E634105.2070903@mdsh.com> <1315159272754-3789686.post@n4.nabble.com> Message-ID: <4E63F848.902@mdsh.com> On 04/09/2011 19:01, FrancoisMace wrote: > Hello Mark, > > Thank you for your answer. Your completely right about what I am trying to > do: > > I have a bunch of files formated in v210. Each file refers to one of the two > fields of the image, the even numbered one corresponds to the bottom field > and the odd one correspond to the top field. > > I have taken a look at your suggestion to use the mp filter and to go for a > more recent version of the ffmpeg software. To do so, I have downloaded an > available windows build (the latest 32 bit static build from > http://ffmpeg.zeranoe.com/builds/). > > However I am still experiencing some odd behavior: > > Here is the command line launched and the obtained result: > > ./ffmpeg.exe -s 1920x540 -vcodec v210 -f image2 -pix_fmt yuv422p16le -i > "PATH_TO_FILES\ParkRun_YUV10_1080i25\PR_1080i_%05d.v210" -pix_fmt > yuv422p10le -vcodec v210 -vf mp=tinterlace:0 -vcodec v210 -pix_fmt > yuv422p16le -r 25 -s 1920x540 "test.avi" > > ffmpeg version N-32290-g6fc1299, Copyright (c) 2000-2011 the FFmpeg > developers > built on Sep 2 2011 00:40:00 with gcc 4.6.1 > configuration: --enable-gpl --enable-version3 --enable-memalign-hack > --enable- > runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r > --enable-libo > pencore-amrnb --enable-libopencore-amrwb --enable-libfreetype > --enable-libgsm -- > enable-libmp3lame --enable-libopenjpeg --enable-librtmp > --enable-libschroedinger > --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx > --enabl > e-libx264 --enable-libxavs --enable-libxvid --enable-zlib > libavutil 51. 14. 0 / 51. 14. 0 > libavcodec 53. 12. 0 / 53. 12. 0 > libavformat 53. 10. 0 / 53. 10. 0 > libavdevice 53. 3. 0 / 53. 3. 0 > libavfilter 2. 37. 0 / 2. 37. 0 > libswscale 2. 0. 0 / 2. 0. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > Input #0, image2, from > 'PATH_TO_FILE\ParkRun_YUV10_1080i25\PR_1080i_%05d.v210': > Duration: 00:00:20.16, start: 0.000000, bitrate: N/A > Stream #0.0: Video: v210, yuv422p10le, 1920x540, 25 tbr, 25 tbn, 25 tbc > File 'test.avi' already exists. Overwrite ? [y/N] y > Incompatible pixel format 'yuv422p16le' for codec 'v210', auto-selecting > format > 'yuv422p10le' > [buffer @ 01ED2500] w:1920 h:540 pixfmt:yuv422p10le tb:1/1000000 sar:0/1 > sws_param: > [mp @ 01ED2AC0] This is a unholy filter, it will be purified by the ffmpeg > exorcist team which will change its syntax from dark -vf mp to light -vf. > Thou shalst not make spells or scripts that depend on it > [buffersink @ 01ED2780] auto-inserting filter 'auto-inserted scaler 0' > between the filter 'Parsed_mp_0' and the filter 'out' > [mp @ 01ED2AC0] auto-inserting filter 'auto-inserted scaler 1' between the > filte r 'src' and the filter 'Parsed_mp_0' > [scale @ 01ED3960] w:1920 h:540 fmt:yuv422p10le -> w:1920 h:540 fmt:yuv420p > flags:0x4 > [scale @ 01ED29A0] w:1920 h:1080 fmt:yuv420p -> w:1920 h:1080 > fmt:yuv422p10le flags:0x4 > Output #0, avi, to 'test.avi': > Metadata: > ISFT : Lavf53.10.0 > Stream #0.0: Video: v210 (v210 / 0x30313276), yuv422p10le, 1920x1080, > q=2-31 > , 200 kb/s, 25 tbn, 25 tbc > Stream mapping: > Stream #0.0 -> #0.0 (v210 -> v210) > Press [q] to stop, [?] for help > [buffer @ 01ED2500] request_frame() called with no available frame! > frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 > bitrate=1107056.0kbit > frame= 1 fps= 1 q=0.0 size= 5406kB time=00:00:00.04 > bitrate=1107056.0kbit > frame= 1 fps= 1 q=0.0 size= 5406kB time=00:00:00.04 > bitrate=1107056.0kbit > frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 > bitrate=1107056.0kbit > frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 > bitrate=1107056.0kbit > frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 > bitrate=1107056.0kbit > frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 > bitrate=1107056.0kbit > frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 > bitrate=1107056.0kbit > frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 > bitrate=1107056.0kbit > frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 > bitrate=1107056.0kbit > frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 > bitrate=1107056.0kbit > frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 > bitrate=1107056.0kbit > frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 > bitrate=1107056.0kbit > frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 > bitrate=1107056.0kbit > Last message repeated 46 times > request_frame() called with no available frame! > frame= 1 fps= 0 q=0.0 Lsize= 5406kB time=00:00:00.04 > bitrate=1107060.8kbi > ts/s dup=0 drop=46 > > > > Two behaviors I cannot explain: > > - I do not undersand why a scaler filter is automatically insterted two > times (goint from 422 to 420 and back) Looking at the source, the mp tinterlace filter accepts these mplayer formats only: IMGFMT_YV12, IMGFMT_IYUV, IMGFMT_I420. And in the shim between FFmpeg and mplayer filter all three equate to FFmpeg's YUV420P. Therefore the FFmpeg filter chain has done the conversion to the only available format in the tinterlace filter. Stefano's new tinterlace filter will do all these - so, not the 10 bit you're after, but better than 420. PIX_FMT_YUV420P, PIX_FMT_YUV422P, PIX_FMT_YUV444P, PIX_FMT_YUV444P, PIX_FMT_YUV410P, PIX_FMT_YUVA420P, PIX_FMT_YUVJ420P, PIX_FMT_YUVJ422P, PIX_FMT_YUVJ444P, PIX_FMT_YUVJ440P, PIX_FMT_GRAY8 Now I'm awake and can see you're using the EBU test material. Good man :-) > - I do not understand why the output file size does not increase during the > encoding process. I don't quite understand your question. But, at the input part you are not specifying the rate of your image stream and therefore FFmpeg is defaulting to 25. You have 50 frames per second at the input and 25 frames per second at the output, so add -r 50 before the -i. Then FFmpeg won't drop half your frames (the "drop=46" at the end of your output. -- Mark From dashing.meng at gmail.com Mon Sep 5 02:38:14 2011 From: dashing.meng at gmail.com (littlebat) Date: Mon, 5 Sep 2011 08:38:14 +0800 Subject: [FFmpeg-user] Padding in ffmpeg In-Reply-To: <35169C72-4CF6-497C-9F56-6A8D7074654B@gmail.com> References: <009701cc69a4$f4ef53a0$decdfae0$@RichardsEMail.Net> <35169C72-4CF6-497C-9F56-6A8D7074654B@gmail.com> Message-ID: <20110905083814.cd8c4b69.dashing.meng@gmail.com> On Sat, 3 Sep 2011 10:25:06 -0600 Jim Worrall wrote: > I'm far from an expert, but suspect you may need to scale the video > down first, then add the padding. You might try -s WxH, or > -vf scale=W:H > > I don't know what the difference is. I would try to add one of these > before your -vf pad= I always expand or crop the source video to the proper aspect then scale to the result video, with mencoder. --- littlebat From joe_pocari at yahoo.com Mon Sep 5 06:20:37 2011 From: joe_pocari at yahoo.com (joe pocari) Date: Sun, 4 Sep 2011 21:20:37 -0700 (PDT) Subject: [FFmpeg-user] Capturing one frame of MacBook iSight output with a delay (...no longer works) Message-ID: <1315196437.59668.YahooMailClassic@web36202.mail.mud.yahoo.com> For about five years I've used ffmpeg in a shell script to grab one frame from my linux'd-up Macbook's iSight: ffmpeg -f video4linux2 -s 640x480 -r 15 -i /dev/video0 -an -vframes 1 -vcodec mjpeg -y -sameq -ss 1.5 snapshot.jpg I upgraded my Ubuntu distro from Lucid to Natty (now using ffmpeg ver. 0.6.2-4:0.6.2-1ubuntu1) and that syntax no longer works. The cam turns on and ffmpeg spits out an infinitely-repeated [er, unverified] stream of the following line at the rate of ~2/sec: frame= 0 fps= 0 q=0.0 size= -0kB time=10000000000.00 bitrate= -0.0kbit Without the '-ss 1.5' flag it will grab the first frame and exit immediately. That's no good, however, because the cam needs that 1.5 seconds to settle at a reasonable exposure. Any permutation of '-ss', '-itsoffset', and '-t' I've haphazardly tried result one of those two behaviors. (Also unexpected, albeit not important atm, is that when vframes > 3, three files are written as expected within ~0.6s and then it merely seems to iterate the 'drop=' total indefinitely.) Suggestions? JP From googol at wolke7.net Mon Sep 5 06:58:12 2011 From: googol at wolke7.net (lovelove) Date: Sun, 4 Sep 2011 21:58:12 -0700 (PDT) Subject: [FFmpeg-user] extracting images from video: please help me understand output Message-ID: <1315198692074-3790405.post@n4.nabble.com> Hi. I am trying to extract 1 image from a video every 10 seconds. My command is: ffmpeg.exe -i inputfile.flv -r 0.1 -f image2 output-%03d.jpeg The jpeg output files are as expected. Nevertheless I am wondering if everything works correctly, because extraction is slow and in the ffmpeg output each timecode (e.g. 00:00:20) is repeated ~5 times before going to the next time code (e.g. 00:00:30). Thank you for any remarks/comments. ffmpeg output: frame= 2 fps= 0 q=2.7 size= -0kB time=00:00:20.00 bitrate= -0.0kbits/s frame= 2 fps= 0 q=2.7 size= -0kB time=00:00:20.00 bitrate= -0.0kbits/s frame= 2 fps= 1 q=2.7 size= -0kB time=00:00:20.00 bitrate= -0.0kbits/s frame= 2 fps= 1 q=2.7 size= -0kB time=00:00:20.00 bitrate= -0.0kbits/s frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= -0.0kbits/s frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= -0.0kbits/s frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= -0.0kbits/s frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= -0.0kbits/s frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= -0.0kbits/s frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= -0.0kbits/s frame= 4 fps= 1 q=1.6 size= -0kB time=00:00:40.00 bitrate= -0.0kbits/s frame= 4 fps= 1 q=1.6 size= -0kB time=00:00:40.00 bitrate= -0.0kbits/s frame= 4 fps= 1 q=1.6 size= -0kB time=00:00:40.00 bitrate= -0.0kbits/s frame= 4 fps= 1 q=1.6 size= -0kB time=00:00:40.00 bitrate= -0.0kbits/s frame= 4 fps= 1 q=1.6 size= -0kB time=00:00:40.00 bitrate= -0.0kbits/s frame= 4 fps= 0 q=1.6 size= -0kB time=00:00:40.00 bitrate= -0.0kbits/s frame= 5 fps= 1 q=1.6 size= -0kB time=00:00:50.00 bitrate= -0.0kbits/s frame= 5 fps= 1 q=1.6 size= -0kB time=00:00:50.00 bitrate= -0.0kbits/s frame= 5 fps= 1 q=1.6 size= -0kB time=00:00:50.00 bitrate= -0.0kbits/s frame= 5 fps= 0 q=1.6 size= -0kB time=00:00:50.00 bitrate= -0.0kbits/s frame= 5 fps= 0 q=1.6 size= -0kB time=00:00:50.00 bitrate= -0.0kbits/s frame= 5 fps= 0 q=1.6 size= -0kB time=00:00:50.00 bitrate= -0.0kbits/s frame= 6 fps= 1 q=1.6 size= -0kB time=00:01:00.00 bitrate= -0.0kbits/s frame= 6 fps= 0 q=1.6 size= -0kB time=00:01:00.00 bitrate= -0.0kbits/s frame= 6 fps= 0 q=1.6 size= -0kB time=00:01:00.00 bitrate= -0.0kbits/s frame= 6 fps= 0 q=1.6 size= -0kB time=00:01:00.00 bitrate= -0.0kbits/s frame= 6 fps= 0 q=1.6 size= -0kB time=00:01:00.00 bitrate= -0.0kbits/s frame= 6 fps= 0 q=1.6 size= -0kB time=00:01:00.00 bitrate= -0.0kbits/s frame= 7 fps= 0 q=1.6 size= -0kB time=00:01:10.00 bitrate= -0.0kbits/s frame= 7 fps= 0 q=1.6 size= -0kB time=00:01:10.00 bitrate= -0.0kbits/s frame= 7 fps= 0 q=1.6 size= -0kB time=00:01:10.00 bitrate= -0.0kbits/s frame= 7 fps= 0 q=1.6 size= -0kB time=00:01:10.00 bitrate= -0.0kbits/s frame= 7 fps= 0 q=1.6 size= -0kB time=00:01:10.00 bitrate= -0.0kbits/s frame= 7 fps= 0 q=1.6 size= -0kB time=00:01:10.00 bitrate= -0.0kbits/s frame= 8 fps= 0 q=1.6 size= -0kB time=00:01:20.00 bitrate= -0.0kbits/s frame= 8 fps= 0 q=1.6 size= -0kB time=00:01:20.00 bitrate= -0.0kbits/s frame= 8 fps= 0 q=1.6 size= -0kB time=00:01:20.00 bitrate= -0.0kbits/s frame= 8 fps= 0 q=1.6 size= -0kB time=00:01:20.00 bitrate= -0.0kbits/s frame= 8 fps= 0 q=1.6 size= -0kB time=00:01:20.00 bitrate= -0.0kbits/s frame= 8 fps= 0 q=1.6 size= -0kB time=00:01:20.00 bitrate= -0.0kbits/s frame= 9 fps= 0 q=1.6 size= -0kB time=00:01:30.00 bitrate= -0.0kbits/s frame= 9 fps= 0 q=1.6 size= -0kB time=00:01:30.00 bitrate= -0.0kbits/s frame= 9 fps= 0 q=1.6 size= -0kB time=00:01:30.00 bitrate= -0.0kbits/s frame= 9 fps= 0 q=1.6 size= -0kB time=00:01:30.00 bitrate= -0.0kbits/s frame= 9 fps= 0 q=1.6 size= -0kB time=00:01:30.00 bitrate= -0.0kbits/s frame= 9 fps= 0 q=1.6 size= -0kB time=00:01:30.00 bitrate= -0.0kbits/s frame= 10 fps= 0 q=1.6 size= -0kB time=00:01:40.00 bitrate= -0.0kbits/s frame= 10 fps= 0 q=1.6 size= -0kB time=00:01:40.00 bitrate= -0.0kbits/s frame= 10 fps= 0 q=1.6 size= -0kB time=00:01:40.00 bitrate= -0.0kbits/s frame= 10 fps= 0 q=1.6 size= -0kB time=00:01:40.00 bitrate= -0.0kbits/s frame= 10 fps= 0 q=1.6 size= -0kB time=00:01:40.00 bitrate= -0.0kbits/s frame= 10 fps= 0 q=1.6 Lsize= -0kB time=00:01:40.00 bitrate= -0.0kbits/s -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/extracting-images-from-video-please-help-me-understand-output-tp3790405p3790405.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From mace.francois at gmail.com Mon Sep 5 08:33:54 2011 From: mace.francois at gmail.com (FrancoisMace) Date: Sun, 4 Sep 2011 23:33:54 -0700 (PDT) Subject: [FFmpeg-user] Generating v210 uncompressed AVIs from v210 individual files in 720p and 1080i formats In-Reply-To: <4E63F848.902@mdsh.com> References: <1314963602407-3785848.post@n4.nabble.com> <4E60DC80.3050508@bbc.co.uk> <1314971444894-3786119.post@n4.nabble.com> <4E60E61F.3020809@bbc.co.uk> <4E634105.2070903@mdsh.com> <1315159272754-3789686.post@n4.nabble.com> <4E63F848.902@mdsh.com> Message-ID: <1315204434989-3790523.post@n4.nabble.com> Hello Mark, Thanks for your reply. I am indeed using the EBU test content. The purpose of my application is to have a way of vizualizing the ouptut of a hardware compression-decompression chain I have applied on the EBU test content for one of our customer, our tools handling only v210 yuv 10 bit files. My last question is related to the fact that my output .avi file size does not increase with frames being added. I applied the changes you indicated and it seems that all the frames, except from the last one, are dropped, leaving the output.avi file with only one frame inside it, hence the line > frame= 1 fps= 0 q=0.0 size= 5406kB time=00:00:00.04 bitrate=1107056.0kbit with the size parameter remaining always the same (size keeps being = 5460 kB). -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Generating-v210-uncompressed-AVIs-from-v210-individual-files-in-720p-and-1080i-formats-tp3785848p3790523.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From dev at rarevision.com Mon Sep 5 11:38:30 2011 From: dev at rarevision.com (Thomas Worth) Date: Mon, 5 Sep 2011 02:38:30 -0700 Subject: [FFmpeg-user] Generating v210 uncompressed AVIs from v210 individual files in 720p and 1080i formats In-Reply-To: <4E63F848.902@mdsh.com> References: <1314963602407-3785848.post@n4.nabble.com> <4E60DC80.3050508@bbc.co.uk> <1314971444894-3786119.post@n4.nabble.com> <4E60E61F.3020809@bbc.co.uk> <4E634105.2070903@mdsh.com> <1315159272754-3789686.post@n4.nabble.com> <4E63F848.902@mdsh.com> Message-ID: > Looking at the source, the mp tinterlace filter accepts these mplayer > formats only: IMGFMT_YV12, IMGFMT_IYUV, IMGFMT_I420. > > And in the shim between FFmpeg and mplayer filter all three equate to > FFmpeg's YUV420P. > > Therefore the FFmpeg filter chain has done the conversion to the only > available format in the tinterlace filter. > > Stefano's new tinterlace filter will do all these - so, not the 10 bit > you're after, but better than 420. > > ?PIX_FMT_YUV420P, ?PIX_FMT_YUV422P, ?PIX_FMT_YUV444P, > ?PIX_FMT_YUV444P, ?PIX_FMT_YUV410P, ?PIX_FMT_YUVA420P, > ?PIX_FMT_YUVJ420P, PIX_FMT_YUVJ422P, PIX_FMT_YUVJ444P, > ?PIX_FMT_YUVJ440P, PIX_FMT_GRAY8 Since v210 is still 4:2:2, is it possible that the filter will just reduce precision to 8 bit? Or does it fail completely since it can't match the pixel format exactly? From mark at mdsh.com Mon Sep 5 12:29:11 2011 From: mark at mdsh.com (Mark Himsley) Date: Mon, 05 Sep 2011 11:29:11 +0100 Subject: [FFmpeg-user] Generating v210 uncompressed AVIs from v210 individual files in 720p and 1080i formats In-Reply-To: References: <1314963602407-3785848.post@n4.nabble.com> <4E60DC80.3050508@bbc.co.uk> <1314971444894-3786119.post@n4.nabble.com> <4E60E61F.3020809@bbc.co.uk> <4E634105.2070903@mdsh.com> <1315159272754-3789686.post@n4.nabble.com> <4E63F848.902@mdsh.com> Message-ID: <4E64A477.606@mdsh.com> On 05/09/11 10:38, Thomas Worth wrote: >> Looking at the source, the mp tinterlace filter accepts these mplayer >> formats only: IMGFMT_YV12, IMGFMT_IYUV, IMGFMT_I420. >> >> And in the shim between FFmpeg and mplayer filter all three equate to >> FFmpeg's YUV420P. >> >> Therefore the FFmpeg filter chain has done the conversion to the only >> available format in the tinterlace filter. >> >> Stefano's new tinterlace filter will do all these - so, not the 10 bit >> you're after, but better than 420. >> >> PIX_FMT_YUV420P, PIX_FMT_YUV422P, PIX_FMT_YUV444P, >> PIX_FMT_YUV444P, PIX_FMT_YUV410P, PIX_FMT_YUVA420P, >> PIX_FMT_YUVJ420P, PIX_FMT_YUVJ422P, PIX_FMT_YUVJ444P, >> PIX_FMT_YUVJ440P, PIX_FMT_GRAY8 > > Since v210 is still 4:2:2, is it possible that the filter will just > reduce precision to 8 bit? Or does it fail completely since it can't > match the pixel format exactly? Yes. FFmpeg's filter chain logic adds format conversion when it's needed - although I haven't looked into how it chooses the best quality format to convert to. Maybe Stefano could describe that? Personally I would format the filter line to force the conversion to the format I want, using a scale filter followed by a format filter (format conversion is done in the scale filter). I add the scale filter myself, instead of letting the filter chain add a scale filter for me because, by default, non-interlaced aware scale filters are added, which *can* wreck the chroma of format conversions. -- Mark From mightydreams at gmail.com Mon Sep 5 17:51:30 2011 From: mightydreams at gmail.com (Tapas Mishra) Date: Mon, 5 Sep 2011 21:21:30 +0530 Subject: [FFmpeg-user] Live streaming In-Reply-To: <8654D442-8AA5-43A5-9F13-5BAF947C2FB4@ixmotion.com> References: <8654D442-8AA5-43A5-9F13-5BAF947C2FB4@ixmotion.com> Message-ID: There is streaming server an open source one known as Red you can try that. On Sun, Sep 4, 2011 at 3:43 PM, Maxence Rioblanc wrote: > Dear All, > > I am currently working on a project where it is needed to stream a video from a camera. > I am working with OpenCV, manage to capture image (opencv Mat format) , do some image treatment ( creating a new Mat object). > I read a lot of topics but all were about converting existing movie or pictures (in bmp, jpeg format) to something else. Yet I did not find how to use ffmep on the fly with my raw buffer ( I can get image width, heigth, pixel depth, the Mat creation is working at 25Hz) and output a flv file which can be accessed by another computer (in my local network). > Any idea or suggestion will be highly appreciated as I am a little bit stuck by this nowadays, > > Regards, > > Maxence > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- ------------------------------ http://mightydreams.blogspot.com Tapas From etienne.buira.lists at free.fr Mon Sep 5 20:23:33 2011 From: etienne.buira.lists at free.fr (Etienne Buira) Date: Mon, 5 Sep 2011 20:23:33 +0200 Subject: [FFmpeg-user] -map weirdness Message-ID: <20110905182333.GO11600@epicure.lazyet.homelinux.net> Hi all. I use 6f8b1fcd4b34 (from today), and I'm having trouble using -map option. The input file looks like: 0.0 -> Video 0.1 -> Audio (fra) 0.2 -> Subtitle (fra) 0.3 -> Audio (deu) (that is actually english track) When I $ ffmpeg -i input.ts -vcodec libx264 -preset ultrafast -crf 25 -acodec libmp3lame -aq 9 -scodec ass -vf crop=... -map 0:0 -map 0:1 -map 0:2 -map 0:3 -acodec libfaac -aq 9 -alang eng output.mp4 I get: Output: 0.0 -> Video 0.1 -> Audio aac (eng) 0.2 -> Subtitle (fra) 0.3 -> Audio aac (deu) Obviously, output 0.1 is plain wrong (wrong encoder, and wrong alang), and 0.3 is weird also. Subtitles are not working (showing as ? -> ass in stream mapping), it have been noticed and reported by Julian earlier today, but it's not my primary concern. My ideal goal is to have video, french sound track, french subtitles, and english sound track (correctly flagged as english if possible). Looking at the source, I didn't find a way to correctly use map and set options for each stream. So, is there a workable solution I didn't find? Regards From tim.nicholson at bbc.co.uk Tue Sep 6 09:18:26 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Tue, 06 Sep 2011 08:18:26 +0100 Subject: [FFmpeg-user] Generating v210 uncompressed AVIs from v210 individual files in 720p and 1080i formats In-Reply-To: <4E634105.2070903@mdsh.com> References: <1314963602407-3785848.post@n4.nabble.com> <4E60DC80.3050508@bbc.co.uk> <1314971444894-3786119.post@n4.nabble.com> <4E60E61F.3020809@bbc.co.uk> <4E634105.2070903@mdsh.com> Message-ID: <4E65C942.6060104@bbc.co.uk> On 04/09/11 10:12, Mark Himsley wrote: > On 02/09/2011 15:20, Tim Nicholson wrote: >> On 02/09/11 15:07, Thomas Worth wrote: >>> On Fri, Sep 2, 2011 at 6:50 AM, FrancoisMace wrote: >>>> Hello Tim, >>>> >>>> Thanks for your quick answer. >>>> >>>> Indeed, my output image was specified as being 1920x540 since, when I >>>> specified 1920x1080, one hald of the ouptut AVI displayed completely green. >> >> Because you only supplied 540 lines of input... >> >>>> >>>> So the command line was there in order to illustrate what I manage to get >>>> working, although what is slipping out of your mind correspond to what I >>>> have tried to get. >>>> >>>> The only notions of "interlacing" I found in the ffmpeg documentation seem >>>> to be related to the use of the AVC encoder/decoders which, since I am >>>> wanting to do an uncompressed AVI, do not interest me. >>> >>> It sounds like you are trying to extract fields from an interlaced >>> video and convert them to progressive (50i --> 50p). Am I correct? If >> >> Thomas I think he has already extracted the individual fields and they >> now exist as discrete image files. Now he wants to turn them back into >> an interlaced video. >> >> ISTR that mencoder managed that somewhat better than ffmpeg, but now >> ffmpeg supports mencoder filters it may be possible directly. I think >> having the input rate twice the output rate was involved somewhere. >> >> Someone must remember. > > I haven't completely understood Francois' requirements, but if Francois > is wanting to convert frames into fields then that's what the tinterlace > filter does. There's the version in FFmbc, there's the version in the mp > filters of FFmpeg (and therefore also in mencoder), and there's the > patch that Stefano has posted to ffmpeg-devel - and which he said "I'm > going to apply the patch in a few days if I see no comments..." a few > days ago. > Tinterlace...that's the one. I knew my memory wasn't completely fubar....(just could not for the life of me remember its name, just knew it was a mencoder one.) > If I've understood right, Francois has half-height frames that he wants > to interlace, then he'll want tinterlace mode 0: "Move odd frames into > the upper field of the new frame, even into the lower field, generating > a double height frame at half framerate" > > >> >> However Francois is using a very old version of ffmpeg. >> > > True enough. > -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From tim.nicholson at bbc.co.uk Tue Sep 6 09:19:54 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Tue, 06 Sep 2011 08:19:54 +0100 Subject: [FFmpeg-user] Generating v210 uncompressed AVIs from v210 individual files in 720p and 1080i formats In-Reply-To: <4E64A477.606@mdsh.com> References: <1314963602407-3785848.post@n4.nabble.com> <4E60DC80.3050508@bbc.co.uk> <1314971444894-3786119.post@n4.nabble.com> <4E60E61F.3020809@bbc.co.uk> <4E634105.2070903@mdsh.com> <1315159272754-3789686.post@n4.nabble.com> <4E63F848.902@mdsh.com> <4E64A477.606@mdsh.com> Message-ID: <4E65C99A.5000204@bbc.co.uk> On 05/09/11 11:29, Mark Himsley wrote: > [...] > > Personally I would format the filter line to force the conversion to the > format I want, using a scale filter followed by a format filter (format > conversion is done in the scale filter). I add the scale filter myself, > instead of letting the filter chain add a scale filter for me because, > by default, non-interlaced aware scale filters are added, which *can* > wreck the chroma of format conversions. > Now I didn't know that.... Which scaler do you use btw? -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From katmai at keptprivate.com Tue Sep 6 10:35:34 2011 From: katmai at keptprivate.com (Stefanita Rares Dumitrescu) Date: Tue, 06 Sep 2011 10:35:34 +0200 Subject: [FFmpeg-user] ffmpeg svn compilation fails on freebsd 8.2 64bit Message-ID: <4E65DB56.7020800@keptprivate.com> hi guys, i try to compile the latest ffmpeg from svn with the following options: ./configure --prefix=/usr/local/ffmpeg \ --cc=gcc45 \ --enable-cross-compile \ --target-os=freebsd \ --arch=amd64 \ --extra-cflags='-msse -I/usr/local/include/vorbis -I/usr/local/include -D__BSD_VISIBLE' \ --extra-ldflags='-L/usr/local/lib ' \ --enable-shared \ --enable-nonfree \ --enable-gpl \ --enable-pthreads \ --enable-libopencore-amrwb \ --enable-libopencore-amrnb \ --enable-libtheora \ --enable-libvorbis \ --enable-libx264 \ --enable-libxvid \ --enable-libopenjpeg \ --enable-memalign-hack \ --enable-runtime-cpudetect \ --enable-version3 \ --enable-postproc however it just stops when i run gmake: CC libavformat/udp.o CC libavformat/utils.o libavformat/utils.c: In function 'av_iformat_next': libavformat/utils.c:126:5: warning: 'first_iformat' is deprecated (declared at libavformat/utils.c:116) libavformat/utils.c: In function 'av_oformat_next': libavformat/utils.c:132:5: warning: 'first_oformat' is deprecated (declared at libavformat/utils.c:121) libavformat/utils.c: In function 'av_register_input_format': libavformat/utils.c:138:5: warning: 'first_iformat' is deprecated (declared at libavformat/utils.c:116) libavformat/utils.c: In function 'av_register_output_format': libavformat/utils.c:147:5: warning: 'first_oformat' is deprecated (declared at libavformat/utils.c:121) libavformat/utils.c: In function 'ff_av_destruct_packet_nofree': libavformat/utils.c:296:5: warning: 'av_destruct_packet_nofree' is deprecated (declared at ./libavcodec/avcodec.h:3055) libavformat/utils.c: In function 'av_seek_frame_generic': libavformat/utils.c:1709:5: warning: 'read_seek' is deprecated (declared at libavformat/avformat.h:448) libavformat/utils.c:1710:9: warning: 'read_seek' is deprecated (declared at libavformat/avformat.h:448) libavformat/utils.c: In function 'av_seek_frame': libavformat/utils.c:1742:5: warning: 'read_seek' is deprecated (declared at libavformat/avformat.h:448) libavformat/utils.c:1743:9: warning: 'read_seek' is deprecated (declared at libavformat/avformat.h:448) libavformat/utils.c: In function 'avformat_seek_file': libavformat/utils.c:1772:5: warning: 'read_seek' is deprecated (declared at libavformat/avformat.h:448) libavformat/utils.c: In function 'av_close_input_stream': libavformat/utils.c:2563:9: warning: 'filename' is deprecated (declared at libavformat/avformat.h:625) libavformat/utils.c:2571:9: warning: 'provider_name' is deprecated (declared at libavformat/avformat.h:706) libavformat/utils.c:2572:9: warning: 'name' is deprecated (declared at libavformat/avformat.h:707) libavformat/utils.c:2582:9: warning: 'title' is deprecated (declared at libavformat/avformat.h:724) libavformat/utils.c: In function 'ff_new_chapter': libavformat/utils.c:2701:5: warning: 'title' is deprecated (declared at libavformat/avformat.h:724) libavformat/utils.c: In function 'av_gettime': libavformat/utils.c:3316:5: error: implicit declaration of function 'gettimeofday' libavformat/utils.c: In function 'parse_date': libavformat/utils.c:3403:13: warning: cast discards qualifiers from pointer target type gmake: *** [libavformat/utils.o] Error 1 ??(root at beast)(03:32:56) any help appreciated. From isambhav at gmail.com Tue Sep 6 11:39:47 2011 From: isambhav at gmail.com (Sambhav) Date: Tue, 6 Sep 2011 15:09:47 +0530 Subject: [FFmpeg-user] FFMPEG initial analysing delay Message-ID: Hi, I am trying to combine h264 elementray video bitstream and AMR audio bitstream into a 3gp container. Both audio and video is read from pipes. My application writes data into the pipe in realtime. ffmpeg -analyzeduration 0 -i vpipe -analyzeduration 0 -i apipe -vcodec copy -acodec copy output.3gp It is taking around 2-3 seconds for the audio input(apipe) to be analysed after which it prints the input properties and then the actual packing into the container starts. When "-analyzeduration 0 " is given, it almost takes 10sec to analyse the input. Is this the expected behaviour, or am I missing any parameter for it to start analyzing input without the initial huge delay ? Regards, Sambhav From batguano999 at hotmail.com Tue Sep 6 15:13:09 2011 From: batguano999 at hotmail.com (bat guano) Date: Tue, 6 Sep 2011 13:13:09 +0000 Subject: [FFmpeg-user] ffmpeg svn compilation fails on freebsd 8.2 64bit In-Reply-To: <4E65DB56.7020800@keptprivate.com> References: <4E65DB56.7020800@keptprivate.com> Message-ID: > Date: Tue, 6 Sep 2011 10:35:34 +0200 > From: katmai at keptprivate.com > To: ffmpeg-user at ffmpeg.org > Subject: [FFmpeg-user] ffmpeg svn compilation fails on freebsd 8.2 64bit > > hi guys, > > i try to compile the latest ffmpeg from svn with the following options: > Hi FFmpeg from svn is obsolete now. Try with FFmpeg Git instead. Here:- http://ffmpeg.org/download.html From david at davidfavor.com Tue Sep 6 16:22:22 2011 From: david at davidfavor.com (David Favor) Date: Tue, 06 Sep 2011 09:22:22 -0500 Subject: [FFmpeg-user] Problem concatenating .mp4 files - dimensions not set Message-ID: <4E662C9E.6000209@davidfavor.com> Trying to concatenate a set of .mp4 files together with no intermediate disk space and having no luck. Looks like something simple is missing. Suggestions welcome. Thanks. Version of ffmpeg is... David-Favor-iMac> ffmpeg -version 2>&1 | grep version ffmpeg version 0.7.3, Copyright (c) 2000-2011 the FFmpeg developers Files are... David-Favor-iMac> for file in 00*.mp4 ; do ffmpeg -i $file 2>&1 | grep Video ; done Stream #0.0(und): Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 22698 kb/s, 59.94 fps, 59.94 tbr, 30k tbn, 59.94 tbc Stream #0.0(und): Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 22699 kb/s, 59.94 fps, 59.94 tbr, 30k tbn, 59.94 tbc Stream #0.0(und): Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 22699 kb/s, 59.94 fps, 59.94 tbr, 30k tbn, 59.94 tbc Stream #0.0(und): Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 22700 kb/s, 59.94 fps, 59.94 tbr, 30k tbn, 59.94 tbc Stream #0.0(und): Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 22695 kb/s, 59.94 fps, 59.94 tbr, 30k tbn, 59.94 tbc Command (one of many attempted) is... cat 00*.mp4 | ffmpeg -benchmark -y -f mp4 -s 1920x1080 -i - -vcodec copy -acodec copy -f mp4 -s 1920x1080 test.mp4 Errors are... David-Favor-iMac> cat 00*.mp4 | ffmpeg -benchmark -y -f mp4 -s 1920x1080 -i - -vcodec copy -acodec copy -f mp4 -s 1920x1080 test.mp4 ffmpeg version 0.7.3, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 23 2011 08:57:36 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --mandir=/opt/local/share/man --enable-shared --enable-pthreads --disable-indevs --cc=/usr/bin/gcc-4.2 --arch=x86_64 --enable-yasm libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 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 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x102054400] multiple edit list entries, a/v desync might occur, patch welcome Last message repeated 1 times [mov,mp4,m4a,3gp,3g2,mj2 @ 0x102054400] stream 0, offset 0x30: partial file [mov,mp4,m4a,3gp,3g2,mj2 @ 0x102054400] Could not find codec parameters (Video: h264, 22698 kb/s) Seems stream 0 codec frame rate differs from container frame rate: 60000.00 (60000/1) -> 30000.00 (60000/2) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'pipe:': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.110.0 Duration: 00:11:20.70, bitrate: N/A Stream #0.0(und): Video: h264, 22698 kb/s, 59.94 fps, 30k tbr, 30k tbn, 60k tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(und): Audio: ac3, 48000 Hz, 2 channels, s16, 256 kb/s Metadata: creation_time : 1970-01-01 00:00:00 [mp4 @ 0x102047600] dimensions not set Output #0, mp4, to 'test.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.110.0 Stream #0.0(und): Video: libx264, q=2-31, 22698 kb/s, 90k tbn, 30k tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(und): Audio: ac3, 48000 Hz, 2 channels, 256 kb/s Metadata: creation_time : 1970-01-01 00:00:00 Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Could not write header for output file #0 (incorrect codec parameters ?) -- Love feeling your best ever, all day, every day? Click http://RadicalHealth.com for the easy way! From etienne.buira.lists at free.fr Tue Sep 6 16:29:29 2011 From: etienne.buira.lists at free.fr (Etienne Buira) Date: Tue, 6 Sep 2011 16:29:29 +0200 Subject: [FFmpeg-user] Problem concatenating .mp4 files - dimensions not set In-Reply-To: <4E662C9E.6000209@davidfavor.com> References: <4E662C9E.6000209@davidfavor.com> Message-ID: <20110906142929.GQ11600@epicure.lazyet.homelinux.net> On Tue, Sep 06, 2011 at 09:22:22AM -0500, David Favor wrote: > Trying to concatenate a set of .mp4 files together > with no intermediate disk space and having no luck. Hi. This should not be very hard to do with MP4Box. AFAIK, ffmpeg can only concatenate streams when fed using a concatenable format (mpegts for instance). Regards. From psychesnet at gmail.com Tue Sep 6 16:40:41 2011 From: psychesnet at gmail.com (Psychesnet) Date: Tue, 6 Sep 2011 07:40:41 -0700 (PDT) Subject: [FFmpeg-user] How to calculate pts for mpegts, contain raw data H264 + AAC Message-ID: <1315320041776-3793605.post@n4.nabble.com> hi guys, I have a big issue, please help me, thanks a lot. I am working at camera, I have raw data of H264+AAC and I packet it into mpegts by FFMPEG, but I do not calculate the pts and the relationship with pcr. any one has ideal, please reply me, thanks a lot. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-to-calculate-pts-for-mpegts-contain-raw-data-H264-AAC-tp3793605p3793605.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From david at davidfavor.com Tue Sep 6 17:58:45 2011 From: david at davidfavor.com (David Favor) Date: Tue, 06 Sep 2011 10:58:45 -0500 Subject: [FFmpeg-user] Problem concatenating .mp4 files - dimensions notset In-Reply-To: <20110906142929.GQ11600@epicure.lazyet.homelinux.net> References: <4E662C9E.6000209@davidfavor.com> <20110906142929.GQ11600@epicure.lazyet.homelinux.net> Message-ID: <4E664335.2090506@davidfavor.com> Etienne Buira wrote: > On Tue, Sep 06, 2011 at 09:22:22AM -0500, David Favor wrote: >> Trying to concatenate a set of .mp4 files together >> with no intermediate disk space and having no luck. > > Hi. > > This should not be very hard to do with MP4Box. Unfortunately MP4Box audio drifts out of sync for every added file. Concatenate 5-6 files and the final file is way out of sync. > AFAIK, ffmpeg can only concatenate streams when fed using a concatenable > format (mpegts for instance). > > Regards. Thanks. -- Love feeling your best ever, all day, every day? Click http://RadicalHealth.com for the easy way! From rogerdpack2 at gmail.com Tue Sep 6 18:26:01 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Tue, 6 Sep 2011 10:26:01 -0600 Subject: [FFmpeg-user] How to calculate pts for mpegts, contain raw data H264 + AAC In-Reply-To: <1315320041776-3793605.post@n4.nabble.com> References: <1315320041776-3793605.post@n4.nabble.com> Message-ID: > but I do not calculate the pts and the relationship with pcr. > any one has ideal, please reply me, thanks a lot. mencoder -idx ? From mark at mdsh.com Tue Sep 6 18:29:08 2011 From: mark at mdsh.com (Mark Himsley) Date: Tue, 06 Sep 2011 17:29:08 +0100 Subject: [FFmpeg-user] Generating v210 uncompressed AVIs from v210 individual files in 720p and 1080i formats In-Reply-To: <4E65C99A.5000204@bbc.co.uk> References: <1314963602407-3785848.post@n4.nabble.com> <4E60DC80.3050508@bbc.co.uk> <1314971444894-3786119.post@n4.nabble.com> <4E60E61F.3020809@bbc.co.uk> <4E634105.2070903@mdsh.com> <1315159272754-3789686.post@n4.nabble.com> <4E63F848.902@mdsh.com> <4E64A477.606@mdsh.com> <4E65C99A.5000204@bbc.co.uk> Message-ID: <4E664A54.2090806@mdsh.com> On 06/09/11 08:19, Tim Nicholson wrote: > > Which scaler do you use btw? Hi Tim, It depends on what I'm doing. Sorry, I think it deserves a much longer answer than that. -- Mark From david at davidfavor.com Tue Sep 6 19:20:39 2011 From: david at davidfavor.com (David Favor) Date: Tue, 06 Sep 2011 12:20:39 -0500 Subject: [FFmpeg-user] Need help rewrapping vixia hfs100 .mts files and concatenating to .mp4 In-Reply-To: <4E664335.2090506@davidfavor.com> References: <4E662C9E.6000209@davidfavor.com> <20110906142929.GQ11600@epicure.lazyet.homelinux.net> <4E664335.2090506@davidfavor.com> Message-ID: <4E665667.9050406@davidfavor.com> Flow seems to be... 1) rewrap 2) concatenate files To rewrap I've tried these commands: ffmpeg -y -i $name.mts -vcodec copy -acodec copy $name.mp4 ffmpeg -y -i $name.mts -f mp4 -vcodec copy -acodec libfaac -ab 256000 -ac 2 $name.mp4 To concatenate: mp4box -cat 00178.mp4 -cat 00179.mp4 -cat 00180.mp4 -cat 00181.mp4 -cat 00182.mp4 -new final.mp4 Problem is audio drifts out of sync with each new file concatenated. So... It appears I have to transcode these files. This being the case, if anyone has a command set that works for either general .mts clips or specifically vixia clips, pass it along to me. Thanks. -- Love feeling your best ever, all day, every day? Click http://RadicalHealth.com for the easy way! From pgoldweic at northwestern.edu Tue Sep 6 19:38:49 2011 From: pgoldweic at northwestern.edu (Patricia N Goldweic) Date: Tue, 6 Sep 2011 17:38:49 +0000 Subject: [FFmpeg-user] Need help rewrapping vixia hfs100 .mts files and concatenating to .mp4 In-Reply-To: <4E665667.9050406@davidfavor.com> References: <4E662C9E.6000209@davidfavor.com> <20110906142929.GQ11600@epicure.lazyet.homelinux.net> <4E664335.2090506@davidfavor.com> <4E665667.9050406@davidfavor.com> Message-ID: <316ABB7EE42BC448BDBDE73B4323B09B1E00BF57@evcspmbx4.ads.northwestern.edu> I use myself a somewhat different command for concatenating mp4s with MP4Box. You might want to try it and see if this works better for you perhaps: MP4Box -force-cat -add .mp4 -cat .mp4 -new .mp4 -Patricia -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of David Favor Sent: Tuesday, September 06, 2011 12:21 PM To: FFmpeg user questions and RTFMs Subject: [FFmpeg-user] Need help rewrapping vixia hfs100 .mts files and concatenating to .mp4 Flow seems to be... 1) rewrap 2) concatenate files To rewrap I've tried these commands: ffmpeg -y -i $name.mts -vcodec copy -acodec copy $name.mp4 ffmpeg -y -i $name.mts -f mp4 -vcodec copy -acodec libfaac -ab 256000 -ac 2 $name.mp4 To concatenate: mp4box -cat 00178.mp4 -cat 00179.mp4 -cat 00180.mp4 -cat 00181.mp4 -cat 00182.mp4 -new final.mp4 Problem is audio drifts out of sync with each new file concatenated. So... It appears I have to transcode these files. This being the case, if anyone has a command set that works for either general .mts clips or specifically vixia clips, pass it along to me. Thanks. -- Love feeling your best ever, all day, every day? Click http://RadicalHealth.com for the easy way! _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From psychesnet at gmail.com Tue Sep 6 09:33:46 2011 From: psychesnet at gmail.com (Psychesnet) Date: Tue, 6 Sep 2011 00:33:46 -0700 (PDT) Subject: [FFmpeg-user] milliseconds to mpeg-ts pts/dts In-Reply-To: <4A158E6A.4090707@unixsystems.de> References: <4A158E6A.4090707@unixsystems.de> Message-ID: <1315294426486-3792778.post@n4.nabble.com> Hello, I have this problem, too, Anyone has ideal about raw data h264+AAC to be contained into mpegts, please help me, thanks a lot. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/milliseconds-to-mpeg-ts-pts-dts-tp939814p3792778.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From ted at tedpavlic.com Tue Sep 6 23:02:02 2011 From: ted at tedpavlic.com (Ted Pavlic) Date: Tue, 06 Sep 2011 17:02:02 -0400 Subject: [FFmpeg-user] ffplay disables DPMS timeouts; How to enable? (e.g., for electricsheep) In-Reply-To: References: Message-ID: <4E668A4A.20802@tedpavlic.com> RESOLVED on ffmpeg-devel list. Solution: set SDL_VIDEO_ALLOW_SCREENSAVER environment variable to 1 [ FYI, a related problem in mplayer was due to a bug that was causing it to ignore -nostop-xscreensaver. That problem should be fixed soon. ] On 09/02/2011 04:29 PM, Ted Pavlic wrote: > I've noticed that my DPMS timeouts are never triggered while > electricsheep runs despite "xset q" showing that DPMS is enabled. > Drilling down further, I see that running mplayer actually causes the > problem. Moreover, ffplay causes the same problem. > > If I'm playing a video with ffplay, how can I tell ffplay to *not* > disable DPMS so that my screen will be blanked/etc. on schedule? > > Thanks -- > Ted > -- Ted Pavlic From rogerdpack2 at gmail.com Wed Sep 7 00:45:53 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Tue, 6 Sep 2011 16:45:53 -0600 Subject: [FFmpeg-user] ffplay should stop Message-ID: Hello all. It appears that this: ffplay -f dshow -i audio=virtual-audio-capturer -t 6 "hangs" after 6 seconds, instead of quitting as I would have expected it to. Bug report worthy? Thanks! -r From rogerdpack2 at gmail.com Wed Sep 7 01:10:00 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Tue, 6 Sep 2011 17:10:00 -0600 Subject: [FFmpeg-user] ffplay vol control possible? Message-ID: It seems from the scattered documentation on the web that ffplay -vol 1 g:\yo.mp3 should work. However, it doesn't: Unrecognized option 'vol' c:\dev\ruby\sensible-cinema\vendor\cache\ffmpeg\ffplay.exe: failed to set value '1' for option 'vol' Bug report worthy? Or is there another way out there? Thanks! -r From psychesnet at gmail.com Wed Sep 7 02:28:08 2011 From: psychesnet at gmail.com (Psychesnet) Date: Tue, 6 Sep 2011 17:28:08 -0700 (PDT) Subject: [FFmpeg-user] How to calculate pts for mpegts, contain raw data H264 + AAC In-Reply-To: References: <1315320041776-3793605.post@n4.nabble.com> Message-ID: <1315355288332-3794925.post@n4.nabble.com> Hi Roger, what's your mean? I have no ideal. ---> mencoder -idx ? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-to-calculate-pts-for-mpegts-contain-raw-data-H264-AAC-tp3793605p3794925.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From dwkang at camars.kaist.ac.kr Wed Sep 7 07:27:30 2011 From: dwkang at camars.kaist.ac.kr (Dongwon Kang) Date: Wed, 7 Sep 2011 14:27:30 +0900 Subject: [FFmpeg-user] Question on implementing a custom IO callback in ByteIOContext. Message-ID: Hi all, I am trying to implement custom IO call backs by using ByteIOContext in avio.h in order to support http sources. The implementation of the most callbacks were pretty much straight forward, but I faced a problem while handling seek() operation with SEEK_END flag. The problem is that there are some cases that we don't know the actual size of the source. (e.g. http source which chunked transfer encoding ) Regarding the callback for seek operation, I found some article which says: http://cdry.wordpress.com/2009/09/09/using-custom-io-callbacks-with-ffmpeg/ "int64_t SeekFunc(void *opaque, int64_t offset, int whence) ; you must return the current position of your stream in bytes (that is, after the seeking is performed). If the seek has failed you must return <0." So, my question is: 1. Will it be okay to return -1 if we don't know the actual size of the source? 2. If so, will the ffmpeg demux code just fail to demux the source or treat it as streaming source and continue parsing? Any comments are welcome! Thanks, -- Dongwon Kang From googol at wolke7.net Wed Sep 7 10:00:55 2011 From: googol at wolke7.net (lovelove) Date: Wed, 7 Sep 2011 01:00:55 -0700 (PDT) Subject: [FFmpeg-user] Scene detection In-Reply-To: <20110808222557.GC9924@geppetto> References: <354784.24180.qm@web110607.mail.gq1.yahoo.com> <1312726584077-3724931.post@n4.nabble.com> <39659E17-5BE5-4C7C-8E1E-0E79DB34CCE5@avpreserve.com> <20110808161254.88776wg7sdrgb946@webmail.tuwien.ac.at> <20110808222557.GC9924@geppetto> Message-ID: <1315382455055-3795513.post@n4.nabble.com> Mark Himsley, Dave Rice, Peter B. and Stefano Sabatini: Thank you all for your replies and useful suggestions. Sorry for the delay, I got a bit side-tracked, but here is my follow-up post in the scene change detection thread. Stefano Sabatini wrote: > > Scene-detection looks like a much requested feature, please file a > feature request on the issue tracker. I have written ticket#442: http://ffmpeg.org/trac/ffmpeg/ticket/442 and also included your ideas and links. Sorry I can't do more, my coding skills aren't even enough for "hello world" greetings :-( Stefano Sabatini wrote: > > On the technical side: this has been already discussed on > ffmpeg-devel, an underkill solution may be as simple as tweaking the > select filter and adopt a simple pixel-per-pixel average difference as > metric. I think this would work only with motionless scenes, but could fail on e.g. a continuous shot of a crowded street. Or spinning elements ( http://www.wheeloffortuneinfo.com/WheelAnimated-White.gif ). But I think you did address this issue later on: Stefano Sabatini wrote: > For images with moving objects that's not a proper solution, and you need > motion estimation code, which is already integrated in libavcodec, so the > ideal solution would be to refactor the ME code in libavcodec and move it > to libavutil. By doing this you may also improve the code itself, and thus > improve the codecs using it at the same time. > What about the -sc_threshold parameter in FFmpeg? Isn't this actually the implementation of what we are talking here? Peter B. wrote: > > For ideas/details about how to implement a scene-cut detection, I > would suggest looking at the code of shotdetect (Johan Mathe is really > nice, and the license is LGPL by the way). > His approach is actually really simple and straightforward, and we've > now been using it in our long-term archive solution for quite a while > and could provide information based on hands-on experience about what > to expect from that straight-forwards algorithm. good call! In ticket#442 I also listed a couple of other open source examples which can possibly be re-used. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Scene-detection-tp941845p3795513.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From googol at wolke7.net Wed Sep 7 10:45:12 2011 From: googol at wolke7.net (lovelove) Date: Wed, 7 Sep 2011 01:45:12 -0700 (PDT) Subject: [FFmpeg-user] extracting images from video: please help me understand output In-Reply-To: <1315198692074-3790405.post@n4.nabble.com> References: <1315198692074-3790405.post@n4.nabble.com> Message-ID: <1315385112511-3795588.post@n4.nabble.com> lovelove wrote: > > Hi. I am trying to extract 1 image from a video every 10 seconds. > > My command is: ffmpeg.exe -i inputfile.flv -r 0.1 -f image2 > output-%03d.jpeg > > The jpeg output files are as expected. Nevertheless I am wondering if > everything works correctly, because extraction is slow and in the ffmpeg > output each timecode (e.g. 00:00:20) is repeated ~5 times before going to > the next time code (e.g. 00:00:30). Thank you for any remarks/comments. > > ffmpeg output: > > frame= 2 fps= 0 q=2.7 size= -0kB time=00:00:20.00 bitrate= > -0.0kbits/s > frame= 2 fps= 0 q=2.7 size= -0kB time=00:00:20.00 bitrate= > -0.0kbits/s > frame= 2 fps= 1 q=2.7 size= -0kB time=00:00:20.00 bitrate= > -0.0kbits/s > frame= 2 fps= 1 q=2.7 size= -0kB time=00:00:20.00 bitrate= > -0.0kbits/s > frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= > -0.0kbits/s > frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= > -0.0kbits/s > frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= > -0.0kbits/s > frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= > -0.0kbits/s > frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= > -0.0kbits/s > frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= > -0.0kbits/s > frame= 4 fps= 1 q=1.6 size= -0kB time=00:00:40.00 bitrate= > -0.0kbits/s > frame= 4 fps= 1 q=1.6 size= -0kB time=00:00:40.00 bitrate= > -0.0kbits/s > frame= 4 fps= 1 q=1.6 size= -0kB time=00:00:40.00 bitrate= > -0.0kbits/s > frame= 4 fps= 1 q=1.6 size= -0kB time=00:00:40.00 bitrate= > -0.0kbits/s > frame= 4 fps= 1 q=1.6 size= -0kB time=00:00:40.00 bitrate= > -0.0kbits/s > frame= 4 fps= 0 q=1.6 size= -0kB time=00:00:40.00 bitrate= > -0.0kbits/s > [...] > Any idea why the lines are repeated (same timecode, same frame)? Does that mean the output image is overwritten 5 times and thus the process could be 5 times faster if done right? Where can I find a explanation/documentation on what FFmpeg output (like this one) means? thank you. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/extracting-images-from-video-please-help-me-understand-output-tp3790405p3795588.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From googol at wolke7.net Wed Sep 7 11:01:51 2011 From: googol at wolke7.net (lovelove) Date: Wed, 7 Sep 2011 02:01:51 -0700 (PDT) Subject: [FFmpeg-user] extract all key frames from a video Message-ID: <1315386111552-3795626.post@n4.nabble.com> Hi. I tried to use the new key-frame filter ( http://ffmpeg.org/trac/ffmpeg/ticket/92 Ticket #92 ) , but it doesn't work for me. What am I doing wrong? My command is: *ffmpeg.exe -i inputfile.avi -vf select="eq(pict_type\,PICT_TYPE_I)" -s 400x300 -f image2 outputfile.AVI-%03d.tif* According to #92 the -vf parameter should select only key frames. And yes, the output images are key frames. But way too much. After 3 seconds, I already had 378 output images, of which only 3 were unique (the rest were duplicates). What do I have to do to make FFmpeg output only the key frames (without duplicating them) ? Thank you. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/extract-all-key-frames-from-a-video-tp3795626p3795626.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From googol at wolke7.net Wed Sep 7 11:12:36 2011 From: googol at wolke7.net (lovelove) Date: Wed, 7 Sep 2011 02:12:36 -0700 (PDT) Subject: [FFmpeg-user] Scene detection In-Reply-To: <1315382455055-3795513.post@n4.nabble.com> References: <354784.24180.qm@web110607.mail.gq1.yahoo.com> <1312726584077-3724931.post@n4.nabble.com> <39659E17-5BE5-4C7C-8E1E-0E79DB34CCE5@avpreserve.com> <20110808161254.88776wg7sdrgb946@webmail.tuwien.ac.at> <20110808222557.GC9924@geppetto> <1315382455055-3795513.post@n4.nabble.com> Message-ID: <1315386756035-3795646.post@n4.nabble.com> lovelove wrote: > > > Stefano Sabatini wrote: >> >> an underkill solution may be as simple as tweaking the >> select filter and adopt a simple pixel-per-pixel average difference as >> metric. > I think this would work only with motionless scenes, but could fail on > e.g. a continuous shot of a crowded street. Or spinning elements ( > http://www.wheeloffortuneinfo.com/WheelAnimated-White.gif ). > An alternative to the pixel-per-pixel average difference would be to detect sudden histogram changes (if that's any easier to implement). This shouldn't be fooled by the 2 examples I gave above, I think. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Scene-detection-tp941845p3795646.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From steve.henshaw at gmail.com Wed Sep 7 11:17:23 2011 From: steve.henshaw at gmail.com (Steve Henshaw) Date: Wed, 07 Sep 2011 10:17:23 +0100 Subject: [FFmpeg-user] building ffmpeg - configure script error Message-ID: <4E6736A3.6050100@gmail.com> Hello, Im trying to build ffmpeg on Ubuntu 10.04 but the configure script wont complete. I've cloned the git repo for ffmpeg, when I run the configure script it doesnt seem to end. Ive included a small part of the output here which is where I think the problem is (Ive added line numbers in at the start): #1 eval: 1: seek_114:yuv_avi_test=yes: not found #2 eval: 1: 1:ac3_fixed_test_checking=yes: not found #3 eval: 1: bsfs:ac3_fixed_test_deps_checking=yes: not found #4 eval: 1: yes:ac3_fixed_test_deps_deps_checking=yes: not found #5 eval: 1: :ac3_fixed_test_deps_deps_deps_checking=yes: not found #6 eval: 1: $:ac3_fixed_test_deps_deps_deps_deps_checking=yes: not found Things seem to be ok until line #3 the string ac3_fixed_test_deps_checking seems to get bigger, adding "_deps_" between "test" and "checking". The length of that string increases through the rest of the output which doesnt seem to end. Could anyone suggest how I can solve this problem? Thanks, Steve From stefano.sabatini-lala at poste.it Wed Sep 7 11:28:19 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Wed, 7 Sep 2011 11:28:19 +0200 Subject: [FFmpeg-user] extract all key frames from a video In-Reply-To: <1315386111552-3795626.post@n4.nabble.com> References: <1315386111552-3795626.post@n4.nabble.com> Message-ID: <20110907092819.GB7009@geppetto> On date Wednesday 2011-09-07 02:01:51 -0700, lovelove encoded: > Hi. I tried to use the new key-frame filter ( > http://ffmpeg.org/trac/ffmpeg/ticket/92 Ticket #92 ) , but it doesn't work > for me. What am I doing wrong? > > My command is: > > *ffmpeg.exe -i inputfile.avi -vf select="eq(pict_type\,PICT_TYPE_I)" -s > 400x300 -f image2 outputfile.AVI-%03d.tif* > > According to #92 the -vf parameter should select only key frames. > > And yes, the output images are key frames. But way too much. After 3 > seconds, I already had 378 output images, of which only 3 were unique (the > rest were duplicates). > What do I have to do to make FFmpeg output only the key frames (without > duplicating them) ? -vsync 0 (-vsync -1 is the default) > Thank you. -- ffmpeg-user random tip #17 A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing in e-mail? From stefano.sabatini-lala at poste.it Wed Sep 7 11:31:11 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Wed, 7 Sep 2011 11:31:11 +0200 Subject: [FFmpeg-user] ffplay vol control possible? In-Reply-To: References: Message-ID: <20110907093111.GC7009@geppetto> On date Tuesday 2011-09-06 17:10:00 -0600, Roger Pack encoded: > It seems from the scattered documentation on the web that > ffplay -vol 1 g:\yo.mp3 > should work. > However, it doesn't: > > Unrecognized option 'vol' > c:\dev\ruby\sensible-cinema\vendor\cache\ffmpeg\ffplay.exe: failed to > set value '1' for option 'vol' That's correct, ffplay *never* supported a -vol option, maybe you're confusing with the ffmpeg option. > Bug report worthy? Or is there another way out there? Wait for audio filtering, but yes you're welcome to file a feature request. -- ffmpeg-user random tip #3 Some few multimedia formats (mpeg-1, mpeg-2:PS, DV) support physical merge consistency, which means you can merge two files with cat. E.g.: cat input1.mpeg input2.mpeg > output.mpeg See: http://www.ffmpeg.org/faq.html#SEC25 From gkinsey at ad-holdings.co.uk Wed Sep 7 14:40:42 2011 From: gkinsey at ad-holdings.co.uk (Gavin Kinsey) Date: Wed, 7 Sep 2011 13:40:42 +0100 Subject: [FFmpeg-user] FFMPEG initial analysing delay In-Reply-To: References: Message-ID: <201109071340.42124.gkinsey@ad-holdings.co.uk> On Tuesday 06 September 2011 10:39:47 Sambhav wrote: > > ffmpeg -analyzeduration 0 -i vpipe -analyzeduration 0 -i apipe -vcodec > copy -acodec copy output.3gp > > It is taking around 2-3 seconds for the audio input(apipe) to be analysed > after which it prints the input properties and then the actual packing > into the container starts. > When "-analyzeduration 0 " is given, it almost takes 10sec to analyse > the input. > > Is this the expected behaviour, or am I missing any parameter for it to > start analyzing input without the initial huge delay ? -probesize -- Gavin Kinsey AD Holdings Plc Closed IPTV, the new safe and secure deterministic IP Video solution from Dedicated Micros, is now shipping through selected distributors. - Come and see this award winning plug and play, IP Video innovation at one of our Roadshows around the UK visit: https://www.dedicatedmicros.com/europe/ClosedIPTVRoadshow2011 - Contact our Customer Services Team for more information regarding how to enter the world of safe and secure hybrid IP and analogue video surveillance systems. From batguano999 at hotmail.com Wed Sep 7 15:13:40 2011 From: batguano999 at hotmail.com (bat guano) Date: Wed, 7 Sep 2011 13:13:40 +0000 Subject: [FFmpeg-user] building ffmpeg - configure script error In-Reply-To: <4E6736A3.6050100@gmail.com> References: <4E6736A3.6050100@gmail.com> Message-ID: ---------------------------------------- > Date: Wed, 7 Sep 2011 10:17:23 +0100 > From: steve.henshaw at gmail.com > To: ffmpeg-user at ffmpeg.org > Subject: [FFmpeg-user] building ffmpeg - configure script error > > Hello, > > Im trying to build ffmpeg on Ubuntu 10.04 but the configure script wont > complete. > Which configure script is that? From steve.henshaw at gmail.com Wed Sep 7 15:34:59 2011 From: steve.henshaw at gmail.com (Steve Henshaw) Date: Wed, 07 Sep 2011 14:34:59 +0100 Subject: [FFmpeg-user] building ffmpeg - configure script error In-Reply-To: References: <4E6736A3.6050100@gmail.com> Message-ID: <4E677303.1030806@gmail.com> On 07/09/11 14:13, bat guano wrote: > > Which configure script is that? > Its the main configure script at the root of the repo. The steps I took: git clone git://git.videolan.org/ffmpeg cd ffmpeg ./configure I also tried it with the flags as suggested here (http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289) ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-postproc \ --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb \ --enable-libtheora --enable-libvorbis --enable-libx264 \ --enable-libxvid --enable-x11grab Both give the same results of endless output. From googol at wolke7.net Wed Sep 7 17:34:49 2011 From: googol at wolke7.net (lovelove) Date: Wed, 7 Sep 2011 08:34:49 -0700 (PDT) Subject: [FFmpeg-user] extract all key frames from a video In-Reply-To: <20110907092819.GB7009@geppetto> References: <1315386111552-3795626.post@n4.nabble.com> <20110907092819.GB7009@geppetto> Message-ID: <1315409689810-3796429.post@n4.nabble.com> Stefano Sabatini wrote: > > On date Wednesday 2011-09-07 02:01:51 -0700, lovelove encoded: >> Hi. I tried to use the new key-frame filter ( >> http://ffmpeg.org/trac/ffmpeg/ticket/92 Ticket #92 ) [...] >> >> My command is: >> >> *ffmpeg.exe -i inputfile.avi -vf select="eq(pict_type\,PICT_TYPE_I)" -s >> 400x300 -f image2 outputfile.AVI-%03d.tif* >> >> [...] >> yes, the output images are key frames. But way too much. After 3 >> seconds, I already had 378 output images, of which only 3 were unique >> (the >> rest were duplicates). > > -vsync 0 (-vsync -1 is the default) > That works, thank you so much! I read up on the -vsync param. It says: 0 = Each frame is passed with its timestamp from the demuxer to the muxer 2 = Frames are passed through with their timestamp or dropped so as to prevent 2 frames from having the same timestamp. I tried both 0 and 2 in the quoted ffmpeg above and both seem to generate the same output in this case. Do they really do the same in my case, or does one of them have any advantage/disadvantage for me which I'm missing? Should 0 or 2 in theory be faster? In my tests there was no speed (performance) difference. Other issue: the output images are fine, however the console text output mysteriously repeats again so many lines (just as I posted earlier on today). Any idea why? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/extract-all-key-frames-from-a-video-tp3795626p3796429.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From kshawkeye at gmail.com Wed Sep 7 20:50:07 2011 From: kshawkeye at gmail.com (Kyle) Date: Wed, 07 Sep 2011 14:50:07 -0400 Subject: [FFmpeg-user] ffmpeg.c:4492: undefined reference to `_avdevice_register_all' Message-ID: <4E67BCDF.8080008@gmail.com> I'm really trying to get this bug: https://ffmpeg.org/trac/ffmpeg/ticket/282 resolved. It seems to not even be linking avdevice. I tried running: i686-w64-mingw32-gcc -Llibavdevice -Wl,--as-needed -Wl,--warn-common -Wl,-rpath-link=libavdevice -o ffmpeg_g.exe ffmpeg.o cmdutils.o -lavdevice 2>&1 | grep "_avdevice_register_all" just for a basic link, but that even fails with: "ffmpeg.c:4492: undefined reference to `_avdevice_register_all'" Also: i686-w64-mingw32-nm "./libavdevice/avdevice.dll" | grep "_avdevice_register_all" Returns: 6d7814c0 T _avdevice_register_all Any ideas? I'm using the latest FFmpeg and trying to compile with MinGW-w64, the problem seems to only happen with "--disable-static --enable-shared" and when SDL is found. (Though the bug doesn't seem related to SDL so it's stumping me). Any help would be great. From eric_hsieh at alphanetworks.com Wed Sep 7 11:10:03 2011 From: eric_hsieh at alphanetworks.com (Eric Hsieh) Date: Wed, 07 Sep 2011 17:10:03 +0800 Subject: [FFmpeg-user] milliseconds to mpeg-ts pts/dts In-Reply-To: <4A158E6A.4090707@unixsystems.de> References: <4A158E6A.4090707@unixsystems.de> Message-ID: <4E6734EB.8040601@alphanetworks.com> Dear *Heribert Steuer, I am working on Camera, I have the same problem with your issue *https://lists.libav.org/pipermail/ffmpeg-user/2009-May/020738.html, What would be the correct way to calculate the pts and dts of the mpegts stream? By the way, my source is raw data H264+AAC and I will packet raw data into mpegts container by ffmpeg. But the pts is always error. do you know now? Please help me. Thanks a lot. Regards, Eric Hsieh, 09/07 From googol at wolke7.net Wed Sep 7 22:23:30 2011 From: googol at wolke7.net (lovelove) Date: Wed, 7 Sep 2011 13:23:30 -0700 (PDT) Subject: [FFmpeg-user] set image width only? Message-ID: <1315427010151-3797158.post@n4.nabble.com> Hi. Sorry for being a newbie, but I have looked everywhere in http://ffmpeg.org/ffmpeg-doc.html and only found *-s WxH*, but how can I /set the width/ e.g. to 400 /*without specifying the height*/ (which FFmpeg should determine itself based on the aspect ratio)? (I'm outputting still images). Many thanks. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/set-image-width-only-tp3797158p3797158.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From mark at mdsh.com Wed Sep 7 22:29:02 2011 From: mark at mdsh.com (Mark Himsley) Date: Wed, 07 Sep 2011 21:29:02 +0100 Subject: [FFmpeg-user] set image width only? In-Reply-To: <1315427010151-3797158.post@n4.nabble.com> References: <1315427010151-3797158.post@n4.nabble.com> Message-ID: <4E67D40E.2050505@mdsh.com> On 07/09/2011 21:23, lovelove wrote: > Hi. Sorry for being a newbie, but I have looked everywhere in > http://ffmpeg.org/ffmpeg-doc.html and only found *-s WxH*, but how can I > /set the width/ e.g. to 400 /*without specifying the height*/ (which FFmpeg > should determine itself based on the aspect ratio)? (I'm outputting still > images). Many thanks. You missed this part of ffmpeg-doc.html http://ffmpeg.org/ffmpeg-doc.html#SEC89 -- Mark From stefano.sabatini-lala at poste.it Thu Sep 8 01:23:01 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Thu, 8 Sep 2011 01:23:01 +0200 Subject: [FFmpeg-user] set image width only? In-Reply-To: <4E67D40E.2050505@mdsh.com> References: <1315427010151-3797158.post@n4.nabble.com> <4E67D40E.2050505@mdsh.com> Message-ID: <20110907232301.GA15115@geppetto> On date Wednesday 2011-09-07 21:29:02 +0100, Mark Himsley encoded: > On 07/09/2011 21:23, lovelove wrote: > >Hi. Sorry for being a newbie, but I have looked everywhere in > >http://ffmpeg.org/ffmpeg-doc.html and only found *-s WxH*, but how can I > >/set the width/ e.g. to 400 /*without specifying the height*/ (which FFmpeg > >should determine itself based on the aspect ratio)? (I'm outputting still > >images). Many thanks. > > You missed this part of ffmpeg-doc.html > > http://ffmpeg.org/ffmpeg-doc.html#SEC89 Better to read instead: http://ffmpeg.org/ffmpeg.html ffmpeg-doc.html is not updated since ages (and maybe should be removed from the web-server). -- ffmpeg-user random tip #0 Please try to use a reasonably recent version of ffmpeg before asking questions to ffmpeg-user: older versions aren't supported here! From lclemens at gmail.com Thu Sep 8 03:13:17 2011 From: lclemens at gmail.com (Luke Clemens) Date: Wed, 7 Sep 2011 19:13:17 -0600 Subject: [FFmpeg-user] How to play asf file using ffmpeg ? In-Reply-To: References: Message-ID: to "play" you'll want to use ffplay. i've never tried it myself, but what happens when you try something like this? : C:\ffmpeg\bin>ffplay.exe http://serveraddress/streamvideo.asf On Fri, Aug 26, 2011 at 5:55 AM, Amit Mangal wrote: > Hi everyone, > i want to play asf stream using ffmpeg is it possible ? > > anbody is having any clue. > > > thanks > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- - - - - Luke Clemens http://clemens.bytehammer.com From batguano999 at hotmail.com Thu Sep 8 03:15:48 2011 From: batguano999 at hotmail.com (bat guano) Date: Thu, 8 Sep 2011 01:15:48 +0000 Subject: [FFmpeg-user] building ffmpeg - configure script error In-Reply-To: <4E677303.1030806@gmail.com> References: <4E6736A3.6050100@gmail.com>, , <4E677303.1030806@gmail.com> Message-ID: ---------------------------------------- > > > I also tried it with the flags as suggested here > (http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289) > Hi That's a good tutorial. I've used it too. Start again and work your way through it from Step 1. From spyfeng at gmail.com Thu Sep 8 03:47:28 2011 From: spyfeng at gmail.com (zhentan feng) Date: Thu, 8 Sep 2011 09:47:28 +0800 Subject: [FFmpeg-user] How to play asf file using ffmpeg ? In-Reply-To: References: Message-ID: Hi On Fri, Aug 26, 2011 at 7:55 PM, Amit Mangal wrote: > Hi everyone, > i want to play asf stream using ffmpeg is it possible ? > > anbody is having any clue. > > > thanks > you said that "asf stream". Is it the fie in the remote server? Maybe you can try mmsh://x.x.x.x/file.asf ? -- Best wishes~ From lou at lrcd.com Thu Sep 8 04:05:40 2011 From: lou at lrcd.com (Lou) Date: Wed, 7 Sep 2011 18:05:40 -0800 Subject: [FFmpeg-user] set image width only? In-Reply-To: <20110907232301.GA15115@geppetto> References: <1315427010151-3797158.post@n4.nabble.com> <4E67D40E.2050505@mdsh.com> <20110907232301.GA15115@geppetto> Message-ID: <20110907180540.0ba5adab@lrcd.com> On Thu, 8 Sep 2011 01:23:01 +0200 Stefano Sabatini wrote: > On date Wednesday 2011-09-07 21:29:02 +0100, Mark Himsley encoded: > > On 07/09/2011 21:23, lovelove wrote: > > >Hi. Sorry for being a newbie, but I have looked everywhere in > > >http://ffmpeg.org/ffmpeg-doc.html and only found *-s WxH*, but how > > >can I /set the width/ e.g. to 400 /*without specifying the > > >height*/ (which FFmpeg should determine itself based on the aspect > > >ratio)? (I'm outputting still images). Many thanks. > > > > You missed this part of ffmpeg-doc.html > > > > http://ffmpeg.org/ffmpeg-doc.html#SEC89 > > Better to read instead: > http://ffmpeg.org/ffmpeg.html More specifically see the scale filter information: http://ffmpeg.org/ffmpeg.html#SEC155 Which currently states: "If the value for width or height is -1, the scale filter will use, for the respective output size, a value that maintains the aspect ratio of the input image." From dwkang at camars.kaist.ac.kr Thu Sep 8 06:56:18 2011 From: dwkang at camars.kaist.ac.kr (Dongwon Kang) Date: Thu, 8 Sep 2011 13:56:18 +0900 Subject: [FFmpeg-user] Question on implementing a custom IO callback in ByteIOContext. In-Reply-To: References: Message-ID: I realized I am using very old version of ffmpeg and found out there is "seekable" flag in AVIOContext struct in the current code. This looks like what I wanted. =) Setting it to 0 seems to work for our usage. /** * A combination of AVIO_SEEKABLE_ flags or 0 when the stream is not seekable. */ int seekable; On Wed, Sep 7, 2011 at 2:27 PM, Dongwon Kang wrote: > Hi all, > > I am trying to implement custom IO call backs by using ByteIOContext in > avio.h in order to support http sources. > The implementation of the most callbacks were pretty much straight forward, > but I faced a problem while handling seek() operation with SEEK_END flag. > The problem is that there are some cases that we don't know the actual size > of the source. (e.g. http source which chunked transfer encoding > ) > > Regarding the callback for seek operation, I found some article which says: > http://cdry.wordpress.com/2009/09/09/using-custom-io-callbacks-with-ffmpeg/ > "int64_t SeekFunc(void *opaque, int64_t offset, > int whence) ; > you must return the current position of your stream in bytes (that is, > after the seeking is performed). If the seek has failed you must return <0." > > So, my question is: > 1. Will it be okay to return -1 if we don't know the actual size of the > source? > 2. If so, will the ffmpeg demux code just fail to demux the source or treat > it as streaming source and continue parsing? > > Any comments are welcome! > Thanks, > -- > Dongwon Kang > > -- Dongwon Kang CA Lab, Div of CS, Dept of EECS, KAIST dwkang at camars.kaist.ac.kr (official) dongwon.kang at gmail.com (non-official) +82) 42-869-5578 +82) 16-705-2950 From akshar_tank at yahoo.com Thu Sep 8 09:30:17 2011 From: akshar_tank at yahoo.com (tank pranav) Date: Thu, 8 Sep 2011 00:30:17 -0700 (PDT) Subject: [FFmpeg-user] extract all key frames from a video In-Reply-To: <1315409689810-3796429.post@n4.nabble.com> References: <1315386111552-3795626.post@n4.nabble.com> <20110907092819.GB7009@geppetto> <1315409689810-3796429.post@n4.nabble.com> Message-ID: <1315467017.443.YahooMailNeo@web122503.mail.ne1.yahoo.com> This is pranav, I am using ffmpeg for video ts file to image conversion by giving following command. ffmpeg -i video_file.ts destination_directory/img_%05d.jpeg it generates images in to ?destination_directory. But it generates some gray color frames in very begining part. and those number of gray color images is generated in every video ts files. number does not remain fixed , in some ts files those gray color images are 70 to 80, in some other video file it remains 2000-5000.? When I open that video ts file in to dvbcut then dvbcut only shows actual frames. Then I could not correlate what ffmpeg does and what dvbcut shows. In documentation of dvbcut says they are using ffmpeg as internal library then why such thing should have happened ??? My conclusion :-? I think so, ffmpeg does not find where from 1st I frame starts ....... If it gives initial gray color frame then it definately means that it did not find I-frame thati s why ffmpeg keeps on inserting dummy frames. We need to change something in ffmpeg's?coding. Where should we change that I am still figuring out. If you have any idea how we should keep on drop the frame till we get I frame then it may solve my purpose. I dont know I am right or wrong, if u can help me out with this issue then it would be great. ________________________________ From: lovelove To: ffmpeg-user at ffmpeg.org Sent: Wednesday, September 7, 2011 9:04 PM Subject: Re: [FFmpeg-user] extract all key frames from a video Stefano Sabatini wrote: > > On date Wednesday 2011-09-07 02:01:51 -0700, lovelove encoded: >> Hi. I tried to use the new key-frame filter ( >> http://ffmpeg.org/trac/ffmpeg/ticket/92 Ticket #92 ) [...] >> >> My command is: >> >> *ffmpeg.exe -i inputfile.avi -vf select="eq(pict_type\,PICT_TYPE_I)" -s >> 400x300 -f image2 outputfile.AVI-%03d.tif* >> >> [...] >> yes, the output images are key frames. But way too much. After 3 >> seconds, I already had 378 output images, of which only 3 were unique >> (the >> rest were duplicates).? > > -vsync 0 (-vsync -1 is the default) > That works, thank you so much! I read up on the -vsync param. It says: 0 = Each frame is passed with its timestamp from the demuxer to the muxer 2 = Frames are passed through with their timestamp or dropped so as to prevent 2 frames from having the same timestamp. I tried both 0 and 2 in the quoted ffmpeg above and both seem to generate the same output in this case. Do they really do the same in my case, or does one of them have any advantage/disadvantage for me which I'm missing? Should 0 or 2 in theory be faster? In my tests there was no speed (performance) difference. Other issue: the output images are fine, however the console text output mysteriously repeats again so many lines (just as I posted earlier on today). Any idea why? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/extract-all-key-frames-from-a-video-tp3795626p3796429.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 googol at wolke7.net Thu Sep 8 09:50:37 2011 From: googol at wolke7.net (lovelove) Date: Thu, 8 Sep 2011 00:50:37 -0700 (PDT) Subject: [FFmpeg-user] extract all key frames from a video In-Reply-To: <1315467017.443.YahooMailNeo@web122503.mail.ne1.yahoo.com> References: <1315386111552-3795626.post@n4.nabble.com> <20110907092819.GB7009@geppetto> <1315409689810-3796429.post@n4.nabble.com> <1315467017.443.YahooMailNeo@web122503.mail.ne1.yahoo.com> Message-ID: <1315468237178-3798104.post@n4.nabble.com> tank pranav wrote: > > This is pranav, I am using ffmpeg for video ts file to image conversion by > giving following command. > > ffmpeg -i video_file.ts destination_directory/img_%05d.jpeg > > it generates images in to ?destination_directory. But it generates some > gray color frames in very begining part. and those number of gray color > images is generated in every video ts files. number does not remain fixed > , in some ts files those gray color images are 70 to 80, in some other > video file it remains 2000-5000.? Normally this occurs when the first frame of your video is not a key frame, so subsequent dependent frames cannot be properly decoded (hence the gray images). The application should recover from this condition upon reaching the next key frame. (Please start your own thread instead of thread hijacking with an unrelated problem.) -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/extract-all-key-frames-from-a-video-tp3795626p3798104.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From googol at wolke7.net Thu Sep 8 10:22:19 2011 From: googol at wolke7.net (lovelove) Date: Thu, 8 Sep 2011 01:22:19 -0700 (PDT) Subject: [FFmpeg-user] set image width only? In-Reply-To: <20110907232301.GA15115@geppetto> References: <1315427010151-3797158.post@n4.nabble.com> <4E67D40E.2050505@mdsh.com> <20110907232301.GA15115@geppetto> Message-ID: <1315470139469-3798162.post@n4.nabble.com> Stefano Sabatini wrote: > > Better to read instead: > http://ffmpeg.org/ffmpeg.html > > ffmpeg-doc.html is not updated since ages (and maybe should be removed > from the web-server). > Thank you all, it works now. Although I don't really understand the difference between *-s 400x300* and *-vf "scale=400:300"* *IF* there is no functional difference, then one parameter should be discontinued, because it is IMO bad practice to have redundant (and thus confusing) parameters. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/set-image-width-only-tp3797158p3798162.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From isambhav at gmail.com Thu Sep 8 13:54:50 2011 From: isambhav at gmail.com (Sambhav) Date: Thu, 8 Sep 2011 17:24:50 +0530 Subject: [FFmpeg-user] FFMPEG initial analysing delay In-Reply-To: <201109071340.42124.gkinsey@ad-holdings.co.uk> References: <201109071340.42124.gkinsey@ad-holdings.co.uk> Message-ID: Thanks. -probesize did reduce the initial analysing period. But still it has a delay of ~2 seconds. What is the unit of the probesize ? The minimum value it is taking is 32. Is there any way to avoid the initial probing ? I know all the parameters of the input file. On Wed, Sep 7, 2011 at 6:10 PM, Gavin Kinsey wrote: > On Tuesday 06 September 2011 10:39:47 Sambhav wrote: > > > > ffmpeg -analyzeduration 0 -i vpipe -analyzeduration 0 -i apipe -vcodec > > copy -acodec copy output.3gp > > > > It is taking around 2-3 seconds for the audio input(apipe) to be analysed > > after which it prints the input properties and then the actual packing > > into the container starts. > > When "-analyzeduration 0 " is given, it almost takes 10sec to analyse > > the input. > > > > Is this the expected behaviour, or am I missing any parameter for it to > > start analyzing input without the initial huge delay ? > > -probesize > > -- > Gavin Kinsey > AD Holdings Plc > > > Closed IPTV, the new safe and secure deterministic IP Video solution from > Dedicated Micros, is now shipping through selected distributors. > - Come and see this award winning plug and play, IP Video innovation at one > of our Roadshows around the UK visit: > https://www.dedicatedmicros.com/europe/ClosedIPTVRoadshow2011 > - Contact our Customer Services Team for more information regarding how to > enter the world of safe and secure hybrid IP and analogue video surveillance > systems. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From gkinsey at ad-holdings.co.uk Thu Sep 8 14:28:52 2011 From: gkinsey at ad-holdings.co.uk (Gavin Kinsey) Date: Thu, 8 Sep 2011 13:28:52 +0100 Subject: [FFmpeg-user] FFMPEG initial analysing delay In-Reply-To: References: <201109071340.42124.gkinsey@ad-holdings.co.uk> Message-ID: <201109081328.52074.gkinsey@ad-holdings.co.uk> On Thursday 08 September 2011 12:54:50 Sambhav wrote: > Thanks. -probesize did reduce the initial analysing period. > But still it has a delay of ~2 seconds. > > What is the unit of the probesize ? The minimum value it is taking is 32. It's in bytes. > Is there any way to avoid the initial probing ? I know all the parameters > of the input file. Not that I know of, short of writing your own app that uses the ffmpeg libraries. It's the av_find_stream_info call that causes the delay, but ffmpeg always calls it. How long it takes depends on the nature of the stream. -- Gavin Kinsey AD Holdings Plc Closed IPTV, the new safe and secure deterministic IP Video solution from Dedicated Micros, is now shipping through selected distributors. - Come and see this award winning plug and play, IP Video innovation at one of our Roadshows around the UK visit: https://www.dedicatedmicros.com/europe/ClosedIPTVRoadshow2011 - Contact our Customer Services Team for more information regarding how to enter the world of safe and secure hybrid IP and analogue video surveillance systems. From mark at mdsh.com Thu Sep 8 19:14:39 2011 From: mark at mdsh.com (Mark Himsley) Date: Thu, 08 Sep 2011 18:14:39 +0100 Subject: [FFmpeg-user] set image width only? In-Reply-To: <20110907232301.GA15115@geppetto> References: <1315427010151-3797158.post@n4.nabble.com> <4E67D40E.2050505@mdsh.com> <20110907232301.GA15115@geppetto> Message-ID: <4E68F7FF.3040606@mdsh.com> On 08/09/11 00:23, Stefano Sabatini wrote: > ffmpeg-doc.html is not updated since ages (and maybe should be removed > from the web-server). Or perhaps a redirect... -- Mark From rogerdpack2 at gmail.com Thu Sep 8 19:33:39 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Thu, 8 Sep 2011 11:33:39 -0600 Subject: [FFmpeg-user] default mp4 encoding changed to x264 Message-ID: Hello. I noticed with this version of (windows) ffmpeg: N-31706-g335bbe4 And this command line: $ ffmpeg -f dshow -i video=screen-capture-recorder -r 5 -t 1 "yo3.mp4" The output video encoding appears to default to Output #0, mp4, to 'yo3.mp4': Metadata: encoder : Lavf53.6.0 Stream #0.0: Video: mpeg4, yuv420p, 500x500, q=2-31, 200 kb/s, 5 tbn, 5 tbc However with this newer version: N-32138-g312645e The output appears to default to: Output #0, mp4, to 'yo3.mp4': Metadata: encoder : Lavf53.10.0 Stream #0.0: Video: h264 (![0][0][0] / 0x0021), yuv420p, 500x500, q=2-31, 5 tbn, 5 tbc (so newer one uses h264 instead of mpeg4 for default). Is this change expected? https://gist.github.com/1204022 has more complete outputs Cheers! -roger- From lou at lrcd.com Thu Sep 8 19:38:37 2011 From: lou at lrcd.com (Lou) Date: Thu, 8 Sep 2011 09:38:37 -0800 Subject: [FFmpeg-user] default mp4 encoding changed to x264 In-Reply-To: References: Message-ID: <20110908093837.01840446@lrcd.com> On Thu, 8 Sep 2011 11:33:39 -0600 Roger Pack wrote: > Hello. > I noticed with this version of (windows) ffmpeg: N-31706-g335bbe4 > > And this command line: > > $ ffmpeg -f dshow -i video=screen-capture-recorder -r 5 -t 1 > "yo3.mp4" > > The output video encoding appears to default to > > Output #0, mp4, to 'yo3.mp4': > Metadata: > encoder : Lavf53.6.0 > Stream #0.0: Video: mpeg4, yuv420p, 500x500, q=2-31, 200 kb/s, 5 > tbn, 5 tbc > > However with this newer version: N-32138-g312645e > > The output appears to default to: > > Output #0, mp4, to 'yo3.mp4': > Metadata: > encoder : Lavf53.10.0 > Stream #0.0: Video: h264 (![0][0][0] / 0x0021), yuv420p, 500x500, > q=2-31, 5 tbn, 5 tbc > > (so newer one uses h264 instead of mpeg4 for default). > Is this change expected? > https://gist.github.com/1204022 has more complete outputs > Cheers! > -roger- Yes, it's expected behavior due to a recent commit: movenc: use libx264 by default when possible for mov, mp4 and psp You can add "-vcodec mpeg4" (did the syntax change for this recently? I recall another recent commit...) if you want the previous behavior. From rogerdpack2 at gmail.com Thu Sep 8 19:47:17 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Thu, 8 Sep 2011 11:47:17 -0600 Subject: [FFmpeg-user] filter to "combine" audio tracks (audio mixer) Message-ID: Hello all. My basic question is "is there a way to combine audio inputs" (like an audio mixer) using ffmpeg. My target would be something like input1 = -f dshow -audio="something" input2= -f dshow -audio="something else" input3= -v dshow -video="something else2" and then overlay input2 on input2 as the "final" (single) audio track for the output. I noticed on http://wiki.multimedia.cx/index.php?title=Ffmpeg_audio_api that some previous work apparently has been discussed about an audio API. Would this be as "easy" as a new libavfilter audio filter that merged (mixed) the two? I guess such a one doesn't yet exist, judging by the list: http://ffmpeg.org/libavfilter.html#SEC15 Sorry if it's a repeat question just wanted to know if a filter would be "the best way" or not. Cheers! -roger- From rogerdpack2 at gmail.com Thu Sep 8 20:32:01 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Thu, 8 Sep 2011 12:32:01 -0600 Subject: [FFmpeg-user] default mp4 encoding changed to x264 In-Reply-To: <20110908093837.01840446@lrcd.com> References: <20110908093837.01840446@lrcd.com> Message-ID: > Yes, it's expected behavior due to a recent commit: > > movenc: use libx264 by default when possible for mov, mp4 and psp > > > You can add "-vcodec mpeg4" (did the syntax change for this recently? I > recall another recent commit...) if you want the previous behavior. Cool thanks! -r From stefano.sabatini-lala at poste.it Thu Sep 8 22:24:57 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Thu, 8 Sep 2011 22:24:57 +0200 Subject: [FFmpeg-user] filter to "combine" audio tracks (audio mixer) In-Reply-To: References: Message-ID: <20110908202457.GA6118@geppetto> On date Thursday 2011-09-08 11:47:17 -0600, Roger Pack encoded: > Hello all. > My basic question is "is there a way to combine audio inputs" (like an > audio mixer) using ffmpeg. > My target would be something like > > > input1 = -f dshow -audio="something" > input2= -f dshow -audio="something else" > input3= -v dshow -video="something else2" > > and then overlay input2 on input2 as the "final" (single) audio track > for the output. > > I noticed on > http://wiki.multimedia.cx/index.php?title=Ffmpeg_audio_api > that some previous work apparently has been discussed about an audio API. > > Would this be as "easy" as a new libavfilter audio filter that merged > (mixed) the two? I guess such a one doesn't yet exist, judging by the > list: > > http://ffmpeg.org/libavfilter.html#SEC15 > > Sorry if it's a repeat question just wanted to know if a filter would > be "the best way" or not. Yes a mixing audio filter (2 in, 1 out) is the right solution. Audio filtering is a work in progress, many bits have been already committed and we're close to integration (where "close" is hardly definable, as it depends on many variable factors including my available free time). -- ffmpeg-user random tip #2 The -y option will force the overwrite of the output, without to prompt you for your confirmation. Example: ffmpeg -i movie.avi -y movie.mp4 From kshawkeye at gmail.com Thu Sep 8 23:07:13 2011 From: kshawkeye at gmail.com (Kyle) Date: Thu, 08 Sep 2011 17:07:13 -0400 Subject: [FFmpeg-user] ffmpeg.c:4492: undefined reference to `_avdevice_register_all' In-Reply-To: <4E67BCDF.8080008@gmail.com> References: <4E67BCDF.8080008@gmail.com> Message-ID: <4E692E81.1030407@gmail.com> On 9/7/2011 2:50 PM, Kyle wrote: > I'm really trying to get this bug: > https://ffmpeg.org/trac/ffmpeg/ticket/282 resolved. > > It seems to not even be linking avdevice. > > I tried running: i686-w64-mingw32-gcc -Llibavdevice -Wl,--as-needed > -Wl,--warn-common -Wl,-rpath-link=libavdevice -o ffmpeg_g.exe ffmpeg.o > cmdutils.o -lavdevice 2>&1 | grep "_avdevice_register_all" > > just for a basic link, but that even fails with: > > "ffmpeg.c:4492: undefined reference to `_avdevice_register_all'" > > Also: i686-w64-mingw32-nm "./libavdevice/avdevice.dll" | grep > "_avdevice_register_all" > > Returns: 6d7814c0 T _avdevice_register_all > > Any ideas? > > I'm using the latest FFmpeg and trying to compile with MinGW-w64, the > problem seems to only happen with "--disable-static --enable-shared" > and when SDL is found. (Though the bug doesn't seem related to SDL so > it's stumping me). > > Any help would be great. I contacted MinGW-w64 and they told me that __declspec(dllexport), needs to be used to export the symbols. Meaning: __declspec(dllexport) void avdevice_register_all(void); and so on. From rogerdpack2 at gmail.com Thu Sep 8 23:14:19 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Thu, 8 Sep 2011 15:14:19 -0600 Subject: [FFmpeg-user] ffmpeg.c:4492: undefined reference to `_avdevice_register_all' In-Reply-To: <4E692E81.1030407@gmail.com> References: <4E67BCDF.8080008@gmail.com> <4E692E81.1030407@gmail.com> Message-ID: >> Any help would be great. > > I contacted MinGW-w64 and they told me that __declspec(dllexport), needs to > be used to export the symbols. > > Meaning: > __declspec(dllexport) void avdevice_register_all(void); > > and so on. Or you can have it in an extern ?C? {} block or a .def file can't you? http://betterlogic.com/roger/2010/07/visual-studio-unable-to-register-dll -r From ian at agrajag.eclipse.co.uk Fri Sep 9 01:05:42 2011 From: ian at agrajag.eclipse.co.uk (Ian Harper) Date: Fri, 09 Sep 2011 00:05:42 +0100 Subject: [FFmpeg-user] ffmpeg Newbie: VDPAU/ION, H.264, Non-Standard y Resolutions (and XBMC) Message-ID: <853bd06607d4b9d22524b4ca5b3aff2b8189092e@webmail.eclipse.net.uk> Apologies, long post, but please bear with it if the title makes any sense to you.? Okay, how to phrase this... I'm happy to wade into ffmpeg and try to track this down, but it's all a new world so I thought I'd at least ask first! Linux (Ubuntu 10.04 and 10.10), Nvidia ION platform (Acer Aspire Revo 3600), several different Nvidia driver versions tried. Easiest test is simply to go with the latest git XBMC code or a pre-compile such as the PVR version that uses the mainline codebase. I picked up an issue on XBMC when the internal version of ffmpeg was updated early this year (from ffmpeg r24229 to c3beafa0f1 - here's the commit [1] that introduced the problem. After much bludgeoning, this seems to be the essence of the problem as seen in XBMC, at least: 1. Corruption on certain H.264 video streams - the very bottom of the image dissolves into green soup. Attractive, but somewhat distracting. 2. The files played okay on the previous ffmpeg code (the version in 10.1 Dharma, versus the version that's now in the pre-11.0 Eden release). 3. It only shows on XBMC when VDPAU is enabled, but that could be due to how XBMC handles VDPAU/non-VDPAU *or* due to how ffmpeg uses VDPAU. So, disable VDPAU, the files play; enable it, they fall apart. 4. It appears on files from the on-demand UK terrestrial broadcasters, such as the BBC (especially HD) and Channel 5 - so, for example, iPlayer for BBC HD programmes streamed within XBMC. 5. It seems to be related to the "odd" file dimensions they use: 1080x720 plays, 1080x724 doesn't - 640x360 does, 640x372 doesn't. From reading around, it may be that these additional lines should be cropped out, but not all players understand that. 6. libvdpau works, because mplayer with VDPAU can play the files on the same system (-vo vdpau -vc ffh264vdpau). 7. ffplay testing is inconclusive, since it doesn't seem to support VDPAU no matter how much you shout at it. It thus struggles to play an HD file on this system - but the few frames it does play show corruption at the bottom of the image (albeit a different corruption from XBMC's). Adding a -vf 1280:720:0:0 to ffplay does away with that corruption (although the system is still too slow to play the files). More information to anaesthetise you here [2]if you're feeling masochistic. SOOOO, WHERE NEXT.. OH, YES: MY POINT! I'm about to embark on a round of endless re-compilation - XBMC using external ffmpeg, compiled against multiple SVN and git versions to see if I can find where the issue arrived.? However, before I do that - because it'll drive me and the family insane, most likely - does this make sense to anyone? Has anyone else seen problems with non-standard y resolutions, or issues with such files and VDPAU, or anything else that rings a bell with this ramble? Any thoughts from the collected wisdom very much appreciated. Links: ------ [1] https://github.com/xbmc/xbmc/commit/1a6a927ec5a0c305f58fa44bc0d023e007820b64 [2] http://forum.xbmc.org/showthread.php?t=99448 From jeffreyskirby at gmail.com Thu Sep 8 20:32:41 2011 From: jeffreyskirby at gmail.com (Jeff Kirby) Date: Thu, 8 Sep 2011 14:32:41 -0400 Subject: [FFmpeg-user] Encoding and Concatenating end-to-end MPEG-TS Files to MPEG-TS Message-ID: I?m having a problem with concatenating similarly x264/libfaac encoded MPEG TS files end-to-end. The problem is a visible via a 1 to 2 second visual/audible interrupt/break during playback of the output file at the point where one segment ends and the subsequent segments begins. If I concatenate the raw mpegts files, via ?cat? or the concat option in FFMpeg, the playback is smooth. I?m wondering if there is something that I can do to help alleviate this interrupt/break in the playback of my compressed concatenated file. Any information or help is much appreciated. If necessary I can provide links to the actual files being used. ffprobe Output of a raw mpegts segment ffprobe capture-2.ts ffprobe version N-32226-ga9c6936, Copyright (c) 2007-2011 the FFmpeg developers built on Aug 31 2011 15:56:19 with gcc 4.4.3 configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab --enable-libvpx --enable-libmp3lame libavutil 51. 14. 0 / 51. 14. 0 libavcodec 53. 12. 0 / 53. 12. 0 libavformat 53. 10. 0 / 53. 10. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 37. 0 / 2. 37. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpegts @ 0x2538420] Continuity Check Failed [mpegts @ 0x2538420] max_analyze_duration 5000000 reached at 5013333 [mpegts @ 0x2538420] Continuity Check Failed [mpegts @ 0x2538420] Continuity Check Failed Input #0, mpegts, from 'capture-2.ts': Duration: 00:00:30.58, start: 30.083044, bitrate: 7187 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0.0[0x100]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 59.96 fps, 59.94 tbr, 90k tbn, 119.88 tbc Stream #0.1[0x101]: Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, s16, 127 kb/s ffprobe output of the encoded file ffprobe encoded-segment-2.ts ffprobe version N-32226-ga9c6936, Copyright (c) 2007-2011 the FFmpeg developers built on Aug 31 2011 15:56:19 with gcc 4.4.3 configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab --enable-libvpx --enable-libmp3lame libavutil 51. 14. 0 / 51. 14. 0 libavcodec 53. 12. 0 / 53. 12. 0 libavformat 53. 10. 0 / 53. 10. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 37. 0 / 2. 37. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpegts @ 0x235a420] Continuity Check Failed [mpegts @ 0x235a420] Continuity Check Failed [mpegts @ 0x235a420] max_analyze_duration 5000000 reached at 5003333 [mpegts @ 0x235a420] Continuity Check Failed [mpegts @ 0x235a420] Continuity Check Failed Input #0, mpegts, from 'encoded-segment-2.ts': Duration: 00:00:30.56, start: 1.400000, bitrate: 1945 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, 856x480 [SAR 320:321 DAR 16:9], 59.96 fps, 59.94 tbr, 90k tbn, 119.88 tbc Stream #0.1[0x101]: Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, s16, 61 kb/s Currently I?m compressing the segments with the command ffmpeg -y -i capture-1.ts -copyts -s 856x480 -vcodec libx264 -bt 2M -vprofile baseline -acodec libfaac -ab 64K -ac 2 -async 1 -threads 0 -f mpegts encoded-segment-1.ts From googol at wolke7.net Fri Sep 9 04:35:34 2011 From: googol at wolke7.net (lovelove) Date: Thu, 8 Sep 2011 19:35:34 -0700 (PDT) Subject: [FFmpeg-user] set image width only? In-Reply-To: <1315470139469-3798162.post@n4.nabble.com> References: <1315427010151-3797158.post@n4.nabble.com> <4E67D40E.2050505@mdsh.com> <20110907232301.GA15115@geppetto> <1315470139469-3798162.post@n4.nabble.com> Message-ID: <1315535734691-3800652.post@n4.nabble.com> lovelove wrote: > > I don't really understand the difference between > *-s 400x300* > and > *-vf "scale=400:300"* > Aside from the difference from the -s parameter being unclear, the -vf scale parameter you suggested to me somehow interferes with key frame detection. This extracts all key frames of a video as single images: *ffmpeg.exe -i inputfile.avi -vf select="eq(pict_type\,PICT_TYPE_I)" -vsync 0 -f image2 outputfile.AVI-%03d.tif** But when adding -vf scale, *ALL* images are extracted, not just keyframes: *ffmpeg.exe -i inputfile.avi -vf select="eq(pict_type\,PICT_TYPE_I)" -vsync 0 -vf "scale=400:-1" -f image2 outputfile.AVI-%03d.tif** What's going wrong? thank you. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/set-image-width-only-tp3797158p3800652.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From bernicesurprise at gmail.com Fri Sep 9 05:25:34 2011 From: bernicesurprise at gmail.com (k b) Date: Fri, 9 Sep 2011 15:25:34 +1200 Subject: [FFmpeg-user] Dies without message when streaming Message-ID: Hi 8) I'm trying to set up live webcam streaming with ffmpeg and ffserver, but not having much luck. I can save to a file no problem, but if I try to send the output to ffserver, ffmpeg just stops, with no indication of why. It gets as far as: > Press [q] to stop encoding But then immediately drops back to the command prompt. It's noticed by ffserver, which outputs: > [POST] "/feed1.ffm HTTP/1.1" 200 1280 So this works: ffmpeg -f video4linux2 -i /dev/video0 -t 00:00:30 -r 3 -s 160x128 -vcodec mpeg4 -f avi -an /home/user/temp/test.avi But this doesn't: ffmpeg -f video4linux2 -i /dev/video0 -t 00:00:30 -r 3 -s 160x128 -vcodec mpeg4 -f avi -an http://localhost:8090/feed1.ffm The relevant section of ffserver.conf is: Feed feed1.ffm Format avi AudioBitRate 32 AudioChannels 1 AudioSampleRate 44100 VideoBitRate 64 VideoBufferSize 40 VideoFrameRate 3 VideoSize 160x128 VideoGopSize 12 VideoCodec mpeg4 NoAudio Preroll 10 Any ideas? At this stage I'm not fussy about format; I'd be happy to get _any_ kind of video output into ffserver, from any source (I've tried an existing video file instead of the webcam, no difference). Running on Ubuntu 11.04. bs From tim.nicholson at bbc.co.uk Fri Sep 9 08:20:26 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Fri, 09 Sep 2011 07:20:26 +0100 Subject: [FFmpeg-user] set image width only? In-Reply-To: <1315535734691-3800652.post@n4.nabble.com> References: <1315427010151-3797158.post@n4.nabble.com> <4E67D40E.2050505@mdsh.com> <20110907232301.GA15115@geppetto> <1315470139469-3798162.post@n4.nabble.com> <1315535734691-3800652.post@n4.nabble.com> Message-ID: <4E69B02A.6060001@bbc.co.uk> On 09/09/11 03:35, lovelove wrote: > > lovelove wrote: >> >> I don't really understand the difference between >> *-s 400x300* >> and >> *-vf "scale=400:300"* >> > > Aside from the difference from the -s parameter being unclear, > the -vf scale parameter you suggested to me somehow interferes with key > frame detection. > > This extracts all key frames of a video as single images: > *ffmpeg.exe -i inputfile.avi -vf select="eq(pict_type\,PICT_TYPE_I)" -vsync > 0 -f image2 outputfile.AVI-%03d.tif** > > But when adding -vf scale, *ALL* images are extracted, not just keyframes: > *ffmpeg.exe -i inputfile.avi -vf select="eq(pict_type\,PICT_TYPE_I)" -vsync > 0 -vf "scale=400:-1" -f image2 outputfile.AVI-%03d.tif** > > What's going wrong? thank you. > At a guess, because you haven't ordered the filters the resize is happening first and therefore all frames become keyframes... You need to take the ouptut of the select and feed it into the scale... Try:- ffmpeg.exe -i inputfile.avi -vsync 0 -vf "select=eq(pict_type\,PICT_TYPE_I) [select]; [select] scale=400:-1 [out]" -f image2 outputfile.AVI-%03d.tif -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From tim.nicholson at bbc.co.uk Fri Sep 9 08:39:41 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Fri, 09 Sep 2011 07:39:41 +0100 Subject: [FFmpeg-user] set image width only? In-Reply-To: <4E69B02A.6060001@bbc.co.uk> References: <1315427010151-3797158.post@n4.nabble.com> <4E67D40E.2050505@mdsh.com> <20110907232301.GA15115@geppetto> <1315470139469-3798162.post@n4.nabble.com> <1315535734691-3800652.post@n4.nabble.com> <4E69B02A.6060001@bbc.co.uk> Message-ID: <4E69B4AD.7070204@bbc.co.uk> On 09/09/11 07:20, Tim Nicholson wrote: > On 09/09/11 03:35, lovelove wrote: >> >> lovelove wrote: >>> >>> I don't really understand the difference between >>> *-s 400x300* >>> and >>> *-vf "scale=400:300"* >>> >> >> Aside from the difference from the -s parameter being unclear, >> the -vf scale parameter you suggested to me somehow interferes with key >> frame detection. >> >> This extracts all key frames of a video as single images: >> *ffmpeg.exe -i inputfile.avi -vf select="eq(pict_type\,PICT_TYPE_I)" >> -vsync >> 0 -f image2 outputfile.AVI-%03d.tif** >> >> But when adding -vf scale, *ALL* images are extracted, not just >> keyframes: >> *ffmpeg.exe -i inputfile.avi -vf select="eq(pict_type\,PICT_TYPE_I)" >> -vsync >> 0 -vf "scale=400:-1" -f image2 outputfile.AVI-%03d.tif** >> >> What's going wrong? thank you. >> > > At a guess, because you haven't ordered the filters the resize is > happening first and therefore all frames become keyframes... > > You need to take the ouptut of the select and feed it into the scale... > > Try:- > > ffmpeg.exe -i inputfile.avi -vsync 0 -vf > "select=eq(pict_type\,PICT_TYPE_I) [select]; [select] scale=400:-1 > [out]" -f image2 outputfile.AVI-%03d.tif > > And before someone else points it out, this can be simplified to:- ffmpeg.exe -i inputfile.avi -vsync 0 -vf "select=eq(pict_type\,PICT_TYPE_I), scale=400:-1" -f image2 outputfile.AVI-%03d.tif Read up on http://ffmpeg.org/ffmpeg.html#SEC113 about the filtergraph syntax. -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From rodney.baker at iinet.net.au Fri Sep 9 08:56:57 2011 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Fri, 9 Sep 2011 16:26:57 +0930 Subject: [FFmpeg-user] Video capture from V4L2 devices with multiple inputs Message-ID: <201109091626.57384.rodney.baker@iinet.net.au> Hi all. I have a DVB-T tuner card (Leadtek DTV1800H) that also has an analog tuner and both composite video and S-video capture inputs. This card is supported in the mainline kernel from 3.1rc1 onwards. I have been trying to do video capture via the composite video input (which appears as input 1 on /dev/video1) but ffmpeg does not seem to allow passing V4L2 options to switch inputs. With mencoder I can pass V4L2 options through from the command line (e.g. tv:// -tv device=/dev/video1:input=1...etc.) but I cannot find an equivalent in the ffmepg man page or online docs. The docs suggest that the appropriate input must first be activated using an external program (e.g. xawtv) and if I do so I can view the composite video input, but as soon as I invoke ffplay or ffmpeg with [ffmpeg|ffplay] -f video4linux2 -i /dev/video1 it resets the device and selectes input0 (the analog TV tuner) instead of input1 (the composite video input). How hard would it be to add the ability to parse V4L2 options and pass them through via the V4L2 API? I am not a programmer so I don't really have any idea where to start, although I'd be happy to help with testing. BTW, I've confirmed that I can capture both video and audio via this card using VLC - video via V4L2 and audio via ALSA - although the options for successful encoding via VLC seem limited and I would prefer to use ffmpeg if possible. Regards, Rodney. -- =================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au =================================================== From psychesnet at gmail.com Fri Sep 9 09:01:56 2011 From: psychesnet at gmail.com (Psychesnet) Date: Fri, 9 Sep 2011 00:01:56 -0700 (PDT) Subject: [FFmpeg-user] How to produce mp4 and quicktime can read it. Message-ID: <1315551716973-3800959.post@n4.nabble.com> Hello guys, I meet a problem, I get raw data of H264+AAC from camera's driver and I produce mp4 file by ffmpeg. The problem is, vlc can play it normal, but the quicktime is always show error -2401. Does anyone have ideal about this, please reply me, thanks a lot. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-to-produce-mp4-and-quicktime-can-read-it-tp3800959p3800959.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From googol at wolke7.net Fri Sep 9 09:55:34 2011 From: googol at wolke7.net (lovelove) Date: Fri, 9 Sep 2011 00:55:34 -0700 (PDT) Subject: [FFmpeg-user] Scene detection In-Reply-To: <20110808161254.88776wg7sdrgb946@webmail.tuwien.ac.at> References: <354784.24180.qm@web110607.mail.gq1.yahoo.com> <1312726584077-3724931.post@n4.nabble.com> <39659E17-5BE5-4C7C-8E1E-0E79DB34CCE5@avpreserve.com> <20110808161254.88776wg7sdrgb946@webmail.tuwien.ac.at> Message-ID: <1315554934172-3801047.post@n4.nabble.com> Peter B. wrote: > > For getting a list of detected scene cut offsets, I can also suggest > using "shotdetect" by Johan Mathe: > http://shotdetect.nonutc.fr/ > > (NOTE: The current release is quite old and contains some issues, but > I've fixed some of them and Johan agreed to merge them back upstream - > there's no official release containing those changes, but if you need > them, let me know) > Thank you Peter. I tested shotdetect today and it really has some nice concepts. The http://shotdetect.nonutc.fr/ graphs are really cool. Problem is, almost all input videos failed. They were either outright refused or failed to produce any meaningful output. For example, just try any video from youtube, they all fail. I got only a single video to work. It's really a pity because I that's the only command line tool with scene change detection I found and I really quite badly wanted this to work ... If you want to send me your improved release, I would of course be happy to test it and give feedback. I'd have some comments/suggestions, but I think they are secondary to the input issue. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Scene-detection-tp941845p3801047.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From psychesnet at gmail.com Fri Sep 9 10:47:23 2011 From: psychesnet at gmail.com (Psychesnet) Date: Fri, 9 Sep 2011 01:47:23 -0700 (PDT) Subject: [FFmpeg-user] How to produce mp4 and quicktime can read it. In-Reply-To: <1315551716973-3800959.post@n4.nabble.com> References: <1315551716973-3800959.post@n4.nabble.com> Message-ID: <1315558043215-3801126.post@n4.nabble.com> Hello guys, I found some clues about error 2401. We need to fill extradata to ffmpeg, and quicktime would not appear the error 2401. But I would not calculate the 2 bytes, does anyone have ideal about this. uint8_t extradata[] = { ??, ??}; AVCodecContextAudio->extradata = extradata; AVCodecContextAudio->extradata_size = 2; My audio is aac, 16kb/s, 1 channel, please help me, thanks a lot. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-to-produce-mp4-and-quicktime-can-read-it-tp3800959p3801126.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From googol at wolke7.net Fri Sep 9 11:19:27 2011 From: googol at wolke7.net (lovelove) Date: Fri, 9 Sep 2011 02:19:27 -0700 (PDT) Subject: [FFmpeg-user] set image width only? In-Reply-To: <4E69B4AD.7070204@bbc.co.uk> References: <1315427010151-3797158.post@n4.nabble.com> <4E67D40E.2050505@mdsh.com> <20110907232301.GA15115@geppetto> <1315470139469-3798162.post@n4.nabble.com> <1315535734691-3800652.post@n4.nabble.com> <4E69B02A.6060001@bbc.co.uk> <4E69B4AD.7070204@bbc.co.uk> Message-ID: <1315559967160-3801166.post@n4.nabble.com> Tim Nicholson wrote: > > ffmpeg.exe -i inputfile.avi -vsync 0 -vf > "select=eq(pict_type\,PICT_TYPE_I), scale=400:-1" -f image2 > outputfile.AVI-%03d.tif > just to confirm that this works and to thank you. What I find extremely curious, is that this command doesn't generate a console output with repeating lines. The output is as follows: frame= 1 fps= 0 q=0.0 size= -0kB time=00:00:00.04 bitrate= -4.4kbits/s frame= 2 fps= 0 q=0.0 size= -0kB time=00:00:03.04 bitrate= -0.1kbits/s frame= 3 fps= 2 q=0.0 size= -0kB time=00:00:06.04 bitrate= -0.0kbits/s frame= 4 fps= 2 q=0.0 size= -0kB time=00:00:09.04 bitrate= -0.0kbits/s frame= 5 fps= 2 q=0.0 size= -0kB time=00:00:12.04 bitrate= -0.0kbits/s frame= 6 fps= 2 q=0.0 size= -0kB time=00:00:15.04 bitrate= -0.0kbits/s frame= 7 fps= 2 q=0.0 size= -0kB time=00:00:18.04 bitrate= -0.0kbits/s frame= 8 fps= 2 q=0.0 size= -0kB time=00:00:21.04 bitrate= -0.0kbits/s frame= 9 fps= 2 q=0.0 size= -0kB time=00:00:24.04 bitrate= -0.0kbits/s frame= 10 fps= 2 q=0.0 size= -0kB time=00:00:27.04 bitrate= -0.0kbits/s frame= 11 fps= 2 q=0.0 size= -0kB time=00:00:30.04 bitrate= -0.0kbits/s [...] whereas *ffmpeg.exe -i inputfile.flv -r 0.1 -f image2 output-%03d.jpeg* generates repeating lines in console and is much slower (hence my fear that each output image is overwritten several times, thus slowing down the process). Is there something wrong with the command? frame= 2 fps= 0 q=2.7 size= -0kB time=00:00:20.00 bitrate= -0.0kbits/s frame= 2 fps= 0 q=2.7 size= -0kB time=00:00:20.00 bitrate= -0.0kbits/s frame= 2 fps= 1 q=2.7 size= -0kB time=00:00:20.00 bitrate= -0.0kbits/s frame= 2 fps= 1 q=2.7 size= -0kB time=00:00:20.00 bitrate= -0.0kbits/s frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= -0.0kbits/s frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= -0.0kbits/s frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= -0.0kbits/s frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= -0.0kbits/s frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= -0.0kbits/s frame= 3 fps= 1 q=1.6 size= -0kB time=00:00:30.00 bitrate= -0.0kbits/s frame= 4 fps= 1 q=1.6 size= -0kB time=00:00:40.00 bitrate= -0.0kbits/s frame= 4 fps= 1 q=1.6 size= -0kB time=00:00:40.00 bitrate= -0.0kbits/s frame= 4 fps= 1 q=1.6 size= -0kB time=00:00:40.00 bitrate= -0.0kbits/s frame= 4 fps= 1 q=1.6 size= -0kB time=00:00:40.00 bitrate= -0.0kbits/s frame= 4 fps= 1 q=1.6 size= -0kB time=00:00:40.00 bitrate= -0.0kbits/s frame= 4 fps= 0 q=1.6 size= -0kB time=00:00:40.00 bitrate= -0.0kbits/s frame= 5 fps= 1 q=1.6 size= -0kB time=00:00:50.00 bitrate= -0.0kbits/s frame= 5 fps= 1 q=1.6 size= -0kB time=00:00:50.00 bitrate= -0.0kbits/s frame= 5 fps= 1 q=1.6 size= -0kB time=00:00:50.00 bitrate= -0.0kbits/s frame= 5 fps= 0 q=1.6 size= -0kB time=00:00:50.00 bitrate= -0.0kbits/s frame= 5 fps= 0 q=1.6 size= -0kB time=00:00:50.00 bitrate= -0.0kbits/s frame= 5 fps= 0 q=1.6 size= -0kB time=00:00:50.00 bitrate= -0.0kbits/s -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/set-image-width-only-tp3797158p3801166.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From rogerdpack2 at gmail.com Fri Sep 9 16:00:12 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Fri, 9 Sep 2011 08:00:12 -0600 Subject: [FFmpeg-user] Scene detection In-Reply-To: <20110808222557.GC9924@geppetto> References: <354784.24180.qm@web110607.mail.gq1.yahoo.com> <1312726584077-3724931.post@n4.nabble.com> <39659E17-5BE5-4C7C-8E1E-0E79DB34CCE5@avpreserve.com> <20110808161254.88776wg7sdrgb946@webmail.tuwien.ac.at> <20110808222557.GC9924@geppetto> Message-ID: >> Or is there any way to feed some sort of "edit list" to ffmpeg? you can specify start and stop times. EDL support would be a nice feature at some point :) > Scene-detection looks like a much requested feature, please file a > feature request on the issue tracker. I'll hope we'll be able to find > some mechanisms for allowing users to (micro?)fund and vote on the > issues/features they want to be fixed. I've been interested in creating something like this recently. Contact me offline to discuss it and requirements :) re: scene detection You could split your video to jpeg's then use imagemagick's "diff" tool metric, and detect frames that have large diff's (sorry if this was already mentioned previously). Cheers! -roger- From rogerdpack2 at gmail.com Fri Sep 9 16:02:13 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Fri, 9 Sep 2011 08:02:13 -0600 Subject: [FFmpeg-user] Dies without message when streaming In-Reply-To: References: Message-ID: > Any ideas? At this stage I'm not fussy about format; I'd be happy to get > _any_ kind of video output into ffserver, from any source (I've tried an > existing video file instead of the webcam, no difference). Running on Ubuntu what exit code? (my first idea is "maybe use VLC for streaming in the meantime" but that doesn't help you much I guess...) -r From rogerdpack2 at gmail.com Fri Sep 9 16:18:19 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Fri, 9 Sep 2011 08:18:19 -0600 Subject: [FFmpeg-user] Scene detection In-Reply-To: <20110808222557.GC9924@geppetto> References: <354784.24180.qm@web110607.mail.gq1.yahoo.com> <1312726584077-3724931.post@n4.nabble.com> <39659E17-5BE5-4C7C-8E1E-0E79DB34CCE5@avpreserve.com> <20110808161254.88776wg7sdrgb946@webmail.tuwien.ac.at> <20110808222557.GC9924@geppetto> Message-ID: > Scene-detection looks like a much requested feature, please file a > feature request on the issue tracker. I'll hope we'll be able to find > some mechanisms for allowing users to (micro?)fund and vote on the > issues/features they want to be fixed. A bit unrelated, but I'd be willing to donate say $100 toward a "motion detection" filter (you specify rectangles and intensity thresholds, if any is triggered it allows the frame through, otherwise it drops it). Thanks! -roger- From ancelotpinto at gmail.com Fri Sep 9 12:22:10 2011 From: ancelotpinto at gmail.com (Ancelot Mark Pinto) Date: Fri, 9 Sep 2011 15:52:10 +0530 Subject: [FFmpeg-user] RTSP Server RECORD Message-ID: Mischa Spiegelmock's RTSP server, https://github.com/revmischa/rtsp-server#readme describes ffmpeg as a client supporting the "RECORD" request. I can't get the necessary flags to send the record request on Darwin Streaming Server. As mentioned in the documentation at, http://ffmpeg.org/ffmpeg-doc.html ffmpeg -re -i input -f rtsp -muxdelay 0.1 rtsp://server/live.sdp works just fine for streaming for others to watch. Please help with the flag for recording. -- Ancelot Mark Pinto Mumbai, India From anupyadav1234 at gmail.com Fri Sep 9 14:25:29 2011 From: anupyadav1234 at gmail.com (anup yadav) Date: Fri, 9 Sep 2011 17:55:29 +0530 Subject: [FFmpeg-user] Please help can't configure ffmpeg with libfreetype Message-ID: Hi there, I was trying to enable libfree type with ffmpeg so i installed freetype also but when i configure ffmpeg with '--enable-libfreetype' It gives me an following error: Package freetyp2 was not found in the pkg-config search path. Parhaps you should add the directory containing 'freetype2.pc' to the PKG_CONFIG_PATH environment variable No package 'freetype2' found ERROR: libfreetype not found If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solving the problem. Please help! Thanks, Anup Yadav. From cr-ffmpeg at ponzo.net Fri Sep 9 22:30:42 2011 From: cr-ffmpeg at ponzo.net (Scott Doty) Date: Fri, 09 Sep 2011 13:30:42 -0700 Subject: [FFmpeg-user] Reduce audio noise In-Reply-To: <87liun92ig.fsf@Compaq.site> References: <87pqk07zcw.fsf@Compaq.site> <20110820200425.GB13947@geppetto> <87liun92ig.fsf@Compaq.site> Message-ID: <4E6A7772.7040109@ponzo.net> On 08/20/2011 02:49 PM, Cecil Westerhof wrote: > Op zaterdag 20 aug 2011 22:04 CEST schreef Stefano Sabatini: > >> On date Saturday 2011-08-20 19:43:27 +0200, Cecil Westerhof encoded: >>> My camera is not the best there is and I can not connect a microphone >>> to it. So the audio is recorded with the internal microphone. It is >>> not bad, but it gives some audio noise. How do I filter this noise >>> away? >> Audio filtering is a work in progress, and will hopefully land soon, >> right now not yet possible. > Okay, thanks. > Incidentally, Audacity has a noise removal function. -Scott From cr-ffmpeg at ponzo.net Fri Sep 9 22:39:15 2011 From: cr-ffmpeg at ponzo.net (Scott Doty) Date: Fri, 09 Sep 2011 13:39:15 -0700 Subject: [FFmpeg-user] Reduce audio noise In-Reply-To: <4E6A7772.7040109@ponzo.net> References: <87pqk07zcw.fsf@Compaq.site> <20110820200425.GB13947@geppetto> <87liun92ig.fsf@Compaq.site> <4E6A7772.7040109@ponzo.net> Message-ID: <4E6A7973.30201@ponzo.net> Just read the thread -- sorry, ignore me. ;P -Scott From help at archivomas.com Sat Sep 10 01:20:05 2011 From: help at archivomas.com (help at archivomas.com) Date: Fri, 9 Sep 2011 18:20:05 -0500 Subject: [FFmpeg-user] Help with REDCODE Message-ID: <20110909182005.10466saeso6jgm4g@webmail.archivomas.com> Hi, i need help, i want decode r3d (redcode) videos on a h264 (mp4) file, i try: ffmpeg -i A001_C004_1124MU_001.R3D -vcodec libx264 new.mp4 but, i get Input #0, r3d, from 'A001_C004_1124MU_001.R3D': Duration: 00:00:04.96, start: 0.000000, bitrate: 314771 kb/s Stream #0.0(und): Video: 0x0000, 4096x2304p, 23.98 fps Metadata: filename: A001_C004_1124MU_001.R3D Stream #0.1(und): Audio: pcm_s32be, 48000 Hz, 2 channels, s32, 3072 kb/s File 'uno.mp4' already exists. Overwrite ? [y/N] y Video pixel format is unknown, stream cannot be encoded my ffmpeg compillation is ffmpeg version 0.8.3, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 9 2011 10:54:26 with gcc 4.1.2 20080704 (Red Hat 4.1.2-50) configuration: --disable-shared --enable-static --disable-ffplay --disable-ffserver --enable-gpl --enable-pthreads --enable-postproc --enable-gray --enable-libfaac --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-bzlib --enable-zlib --enable-nonfree --enable-version3 --enable-runtime-cpudetect --enable-libopenjpeg any idea ? From stefano.sabatini-lala at poste.it Sat Sep 10 01:05:49 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Sat, 10 Sep 2011 01:05:49 +0200 Subject: [FFmpeg-user] Please help can't configure ffmpeg with libfreetype In-Reply-To: References: Message-ID: <20110909230549.GB24963@geppetto> On date Friday 2011-09-09 17:55:29 +0530, anup yadav encoded: > Hi there, > I was trying to enable libfree type with ffmpeg so i installed > freetype also but when i configure ffmpeg with '--enable-libfreetype' It > gives me an following error: > > > Package freetyp2 was not found in the pkg-config search path. > Parhaps you should add the directory containing 'freetype2.pc' > to the PKG_CONFIG_PATH environment variable > No package 'freetype2' found > ERROR: libfreetype not found > > If you think configure made a mistake, make sure you are using the latest > version from Git. If the latest version fails, report the problem to the > ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. > Include the log file "config.log" produced by configure as this will help > solving the problem. > > Please help! If you're on a distro, make sure that the freetype-dev/-devel package is installed on your system. On my debian the package is called libfreetype6-dev. In the devel package you should also find a file: /usr/lib/pkgconfig/freetype2.pc which contains the metadata required for letting other libraries to compile against the library, and in particular is required by the FFmpeg build system. -- ffmpeg-user random tip #1 The -b (bitrate) and -ab (audio bitrate) ffmpeg options take as argument a bitrate expressed in bits per second, and not in Kbits per second as in previous versions. Change your command lines accordingly. From kshawkeye at gmail.com Sat Sep 10 08:42:57 2011 From: kshawkeye at gmail.com (Kyle) Date: Sat, 10 Sep 2011 02:42:57 -0400 Subject: [FFmpeg-user] ffmpeg.c:4492: undefined reference to `_avdevice_register_all' In-Reply-To: References: <4E67BCDF.8080008@gmail.com> <4E692E81.1030407@gmail.com> Message-ID: <4E6B06F1.30003@gmail.com> On 9/8/2011 5:14 PM, Roger Pack wrote: >>> Any help would be great. >> I contacted MinGW-w64 and they told me that __declspec(dllexport), needs to >> be used to export the symbols. >> >> Meaning: >> __declspec(dllexport) void avdevice_register_all(void); >> >> and so on. > > Or you can have it in an extern ?C? {} block or a .def file can't you? > > http://betterlogic.com/roger/2010/07/visual-studio-unable-to-register-dll > > -r > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Perhaps, that is what was mentioned to me. Sorry, I don't know C. Check out: http://sourceforge.net/mailarchive/forum.php?thread_name=CAEwic4ZvRDjZ_qn%3DJCwNQcaa6zAa4ZOpno_LReRs4xVc0E0M2Q%40mail.gmail.com&forum_name=mingw-w64-public if you would like more info, the mingw-w64 dev's are very friendly and can help with whatever is needed, I would just like to see this bug resolved. From bernicesurprise at gmail.com Sat Sep 10 11:08:27 2011 From: bernicesurprise at gmail.com (k b) Date: Sat, 10 Sep 2011 21:08:27 +1200 Subject: [FFmpeg-user] Dies without message when streaming In-Reply-To: References: Message-ID: On Sat, Sep 10, 2011 at 2:02 AM, Roger Pack wrote: > what exit code? > How do I find out? > (my first idea is "maybe use VLC for streaming in the meantime" but > that doesn't help you much I guess...) > I could live with VLC - not sure what streaming method to select, though? Or the best way to actually embed the video on a website? I've got flowplayer, but that seems to only cope with RTMP for live video, which isn't one of VLC's options. kelly From batguano999 at hotmail.com Sat Sep 10 13:58:36 2011 From: batguano999 at hotmail.com (bat guano) Date: Sat, 10 Sep 2011 11:58:36 +0000 Subject: [FFmpeg-user] "aac_adtstoasc" filter seems to be broken. Message-ID: Hi With FFmpeg? version N-32404-gf6e99bf? the "aac_adtstoasc" filter seems to be broken. Using a command like this:- ffmpeg -i foo.flv -vn -acodec copy -absf aac_adtstoasc foo.aac Gives an error message:- Unrecognized option 'absf' Failed to set value 'aac_adtstoasc' for option 'absf' INFO: Command exit code 1 (raw code = 256) WARNING: aac conversion failed - retaining aac file When I use the command with FFmpeg version 0.8.2 it does the job OK, no errors. (The command is used with get_iplayer program to make the downloaded files acceptable for iTunes) Is this a bug with version N-32404-gf6e99bf, or has the filter been deliberately removed, or has the syntax changed? Console output is here:- http://pastebin.com/TjieDTYQ From batguano999 at hotmail.com Sat Sep 10 15:22:48 2011 From: batguano999 at hotmail.com (bat guano) Date: Sat, 10 Sep 2011 13:22:48 +0000 Subject: [FFmpeg-user] "aac_adtstoasc" filter seems to be broken. In-Reply-To: References: Message-ID: ---------------------------------------- > From: batguano999 at hotmail.com > To: ffmpeg-user at ffmpeg.org > Date: Sat, 10 Sep 2011 11:58:36 +0000 > Subject: [FFmpeg-user] "aac_adtstoasc" filter seems to be broken. > > > Hi > With FFmpeg version N-32404-gf6e99bf the "aac_adtstoasc" filter seems to be broken. > Using a command like this:- > ffmpeg -i foo.flv -vn -acodec copy -absf aac_adtstoasc foo.aac > Gives an error message:- > Unrecognized option 'absf' > Failed to set value 'aac_adtstoasc' for option 'absf' > INFO: Command exit code 1 (raw code = 256) > WARNING: aac conversion failed - retaining aac file > > > When I use the command with FFmpeg version 0.8.2 it does the job OK, no errors. > > (The command is used with get_iplayer program to make the downloaded files acceptable for iTunes) > > Is this a bug with version N-32404-gf6e99bf, or has the filter been deliberately removed, or has the syntax changed? > > Console output is here:- http://pastebin.com/TjieDTYQ > > > Hi I've just re-compiled FFmpeg version N-32432-gb10ba11. The error's still there:- http://pastebin.com/F05rfeb5 I think that this problem has crept into FFmpeg git sometime within the past 1 or 2 weeks. From batguano999 at hotmail.com Sat Sep 10 15:31:54 2011 From: batguano999 at hotmail.com (bat guano) Date: Sat, 10 Sep 2011 13:31:54 +0000 Subject: [FFmpeg-user] "aac_adtstoasc" filter seems to be broken. In-Reply-To: References: , Message-ID: ---------------------------------------- > From: batguano999 at hotmail.com > To: ffmpeg-user at ffmpeg.org > Date: Sat, 10 Sep 2011 13:22:48 +0000 > Subject: Re: [FFmpeg-user] "aac_adtstoasc" filter seems to be broken. > > > > > ---------------------------------------- > > From: batguano999 at hotmail.com > > To: ffmpeg-user at ffmpeg.org > > Date: Sat, 10 Sep 2011 11:58:36 +0000 > > Subject: [FFmpeg-user] "aac_adtstoasc" filter seems to be broken. > > > > > > Hi > > With FFmpeg version N-32404-gf6e99bf the "aac_adtstoasc" filter seems to be broken. > > Using a command like this:- > > ffmpeg -i foo.flv -vn -acodec copy -absf aac_adtstoasc foo.aac > > Gives an error message:- > > Unrecognized option 'absf' > > Failed to set value 'aac_adtstoasc' for option 'absf' > > INFO: Command exit code 1 (raw code = 256) > > WARNING: aac conversion failed - retaining aac file > > > > > > When I use the command with FFmpeg version 0.8.2 it does the job OK, no errors. > > > > (The command is used with get_iplayer program to make the downloaded files acceptable for iTunes) > > > > Is this a bug with version N-32404-gf6e99bf, or has the filter been deliberately removed, or has the syntax changed? > > > > Console output is here:- http://pastebin.com/TjieDTYQ > > > > > > > > Hi > I've just re-compiled FFmpeg version N-32432-gb10ba11. > The error's still there:- http://pastebin.com/F05rfeb5 > I think that this problem has crept into FFmpeg git sometime within the past 1 or 2 weeks. > > > _______________________________________________ Hi This is the good result when using FFmpeg version 0.8.2 :- http://pastebin.com/Tcbh85Rb From phil_rhodes at rocketmail.com Sat Sep 10 23:13:19 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Sat, 10 Sep 2011 22:13:19 +0100 Subject: [FFmpeg-user] Help with REDCODE In-Reply-To: <20110909182005.10466saeso6jgm4g@webmail.archivomas.com> References: <20110909182005.10466saeso6jgm4g@webmail.archivomas.com> Message-ID: > ffmpeg -i A001_C004_1124MU_001.R3D -vcodec libx264 new.mp4 Red files have been encrypted for some time. Unless your file is at least a year or two old, it can't be read by anything other than Red's own SDK. I'm not sure if there's any appetite for attacking the encryption. P From kshawkeye at gmail.com Sun Sep 11 07:00:41 2011 From: kshawkeye at gmail.com (Kyle) Date: Sun, 11 Sep 2011 01:00:41 -0400 Subject: [FFmpeg-user] multiple definition of `_SDL_GetError' Message-ID: <4E6C4079.6080700@gmail.com> After working through this bug: https://ffmpeg.org/trac/ffmpeg/ticket/282 I ran into this: /home/kyle/software/ffmpeg/packages/sdl/sdl-1.2.14-win32/lib/libSDL.a(SDL_error.o):SDL_error.c:(.text+0x477): multiple definition of `_SDL_GetError' libavdevice/libavdevice.dll.a(d000060.o):(.text+0x0): first defined here /home/kyle/software/ffmpeg/packages/sdl/sdl-1.2.14-win32/lib/libSDL.a(SDL_thread.o):SDL_thread.c:(.text+0x2c1): multiple definition of `_SDL_CreateThread' libavdevice/libavdevice.dll.a(d000030.o):(.text+0x0): first defined here /home/kyle/software/ffmpeg/packages/sdl/sdl-1.2.14-win32/lib/libSDL.a(SDL_thread.o):SDL_thread.c:(.text+0x411): multiple definition of `_SDL_WaitThread' libavdevice/libavdevice.dll.a(d000168.o):(.text+0x0): first defined here I'm not sure if this is related to the other bug or not. The command: grep -ir "SDL_GetError" * returns: Binary file ffplay.o matches Binary file libavdevice/avdevice.dll matches libavdevice/avdevice-53.def: SDL_GetError @57 Binary file libavdevice/libavdevice.dll.a matches Binary file libavdevice/sdl.o matches Binary file libavdevice/avdevice-53.dll matches The command: make V=1 returns: i686-w64-mingw32-gcc -Llibavcodec -Llibavdevice -Llibavfilter -Llibavformat -Llibavutil -Llibpostproc -Llibswscale -L/home/kyle/software/ffmpeg/packages/sdl/sdl-1.2.14-win32/lib -Wl,--as-needed -Wl,--warn-common -Wl,-rpath-link=libpostproc:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil -o ffplay_g.exe ffplay.o cmdutils.o -lavdevice -lavfilter -lavformat -lavcodec -lpostproc -lswscale -lavutil -lavicap32 -lpsapi -lole32 -lstrmiids -luuid -lws2_32 -L/home/kyle/software/ffmpeg/packages/sdl/sdl-1.2.14-win32/lib -lmingw32 -lSDLmain -lSDL -lm -luser32 -lgdi32 -lwinmm -ldxguid -lm -lpthread -lpsapi -L/home/kyle/software/ffmpeg/packages/sdl/sdl-1.2.14-win32/lib -lmingw32 -lSDLmain -lSDL -lm -luser32 -lgdi32 -lwinmm -ldxguid SDL_GetError can be found in libSDL.a and libSDLmain.a Any ideas? From jon_y at users.sourceforge.net Sun Sep 11 08:28:26 2011 From: jon_y at users.sourceforge.net (JonY) Date: Sun, 11 Sep 2011 14:28:26 +0800 Subject: [FFmpeg-user] multiple definition of `_SDL_GetError' In-Reply-To: <4E6C4079.6080700@gmail.com> References: <4E6C4079.6080700@gmail.com> Message-ID: <4E6C550A.9060209@users.sourceforge.net> On 9/11/2011 13:00, Kyle wrote: > After working through this bug: > https://ffmpeg.org/trac/ffmpeg/ticket/282 I ran into this: > /home/kyle/software/ffmpeg/packages/sdl/sdl-1.2.14-win32/lib/libSDL.a(SDL_error.o):SDL_error.c:(.text+0x477): > multiple definition of `_SDL_GetError' > libavdevice/libavdevice.dll.a(d000060.o):(.text+0x0): first defined here > /home/kyle/software/ffmpeg/packages/sdl/sdl-1.2.14-win32/lib/libSDL.a(SDL_thread.o):SDL_thread.c:(.text+0x2c1): > multiple definition of `_SDL_CreateThread' > libavdevice/libavdevice.dll.a(d000030.o):(.text+0x0): first defined here > /home/kyle/software/ffmpeg/packages/sdl/sdl-1.2.14-win32/lib/libSDL.a(SDL_thread.o):SDL_thread.c:(.text+0x411): > multiple definition of `_SDL_WaitThread' > libavdevice/libavdevice.dll.a(d000168.o):(.text+0x0): first defined here Hi, This would be a symptom of not using dllexport properly (eg using -export-all-symbols). It is also possible that dllexport directives leaked into the SDL static lib, and finally into libavdevice, caused by sloppy use of the dllexport attribute and the use of static libs to build DLLs. Anyway, libavdevice shouldn't be exporting SDL symbols. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 196 bytes Desc: OpenPGP digital signature URL: From snthdiueoa at gmail.com Sun Sep 11 11:36:06 2011 From: snthdiueoa at gmail.com (Tom) Date: Sun, 11 Sep 2011 11:36:06 +0200 Subject: [FFmpeg-user] Option video_size Message-ID: Hi, I'm running the latest ffmpeg (built this weekend from git), and got this error when trying to transform an flv to a set of jpegs: $ ffmpeg -s 720x576 -i media.flv -s 640x512 -b 8000K -an -r 25 -vframes 30 media/640x480x25/build%d.jpg ffmpeg version git-Sat Sep 10 05:24:39 2011 +0200-b10ba11, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 10 2011 08:00:13 with gcc 4.2.4 (Ubuntu 4.2.4-1ubuntu4) configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libxvid --enable-x11grab --enable-pthreads libavutil 51. 16. 0 / 51. 16. 0 libavcodec 53. 13. 0 / 53. 13. 0 libavformat 53. 12. 0 / 53. 12. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 39. 0 / 2. 39. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 Option video_size not found. I've googled, but option video_size is apparently quite unknown on the documentation page.. Can someone help? Cheers, Tom From etienne.buira.lists at free.fr Sun Sep 11 11:57:19 2011 From: etienne.buira.lists at free.fr (Etienne Buira) Date: Sun, 11 Sep 2011 11:57:19 +0200 Subject: [FFmpeg-user] Option video_size In-Reply-To: References: Message-ID: <20110911095719.GS11600@epicure.lazyet.homelinux.net> On Sun, Sep 11, 2011 at 11:36:06AM +0200, Tom wrote: > Hi, > > I'm running the latest ffmpeg (built this weekend from git), and got > this error when trying to transform an flv to a set of jpegs: > > $ ffmpeg -s 720x576 -i media.flv -s 640x512 -b 8000K -an -r 25 > -vframes 30 media/640x480x25/build%d.jpg > > ffmpeg version git-Sat Sep 10 05:24:39 2011 +0200-b10ba11, Copyright > (c) 2000-2011 the FFmpeg developers > built on Sep 10 2011 08:00:13 with gcc 4.2.4 (Ubuntu 4.2.4-1ubuntu4) > configuration: --enable-gpl --enable-version3 --enable-nonfree > --enable-postproc --enable-libfaac --enable-libopencore-amrnb > --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora > --enable-libvorbis --enable-libxvid --enable-x11grab --enable-pthreads > libavutil 51. 16. 0 / 51. 16. 0 > libavcodec 53. 13. 0 / 53. 13. 0 > libavformat 53. 12. 0 / 53. 12. 0 > libavdevice 53. 3. 0 / 53. 3. 0 > libavfilter 2. 39. 0 / 2. 39. 0 > libswscale 2. 1. 0 / 2. 1. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > Option video_size not found. > > > I've googled, but option video_size is apparently quite unknown on the > documentation page.. > > Can someone help? Hi. It's a string describing frame size (instead of -s before -i). And true, not in doc, but present in ffmpeg -h. Regards. From batguano999 at hotmail.com Sun Sep 11 13:50:20 2011 From: batguano999 at hotmail.com (bat guano) Date: Sun, 11 Sep 2011 11:50:20 +0000 Subject: [FFmpeg-user] "aac_adtstoasc" filter seems to be broken. In-Reply-To: References: , , , Message-ID: ---------------------------------------- > > I think that this problem has crept into FFmpeg git sometime within the past 1 or 2 weeks. > > > > > > _______________________________________________ > Hi > This is the good result when using FFmpeg version 0.8.2 :- http://pastebin.com/Tcbh85Rb > > _______________________________________________ Hi The problem was caused by a syntax change at commit d821cbe2ffb5c06073b6c841e6d19bc7ca9ae835 Backward compatibility has been added at commit cec92fe49a004a2e8b8b8bbba95d756e43cdd93e The problem is solved. :-) Here:- http://pastebin.com/D8zrFbPi From snthdiueoa at gmail.com Sun Sep 11 20:14:32 2011 From: snthdiueoa at gmail.com (Tom) Date: Sun, 11 Sep 2011 20:14:32 +0200 Subject: [FFmpeg-user] Option video_size In-Reply-To: <20110911095719.GS11600@epicure.lazyet.homelinux.net> References: <20110911095719.GS11600@epicure.lazyet.homelinux.net> Message-ID: Hi, So I changed "-s" to "--video_size" and got this error now: $ ffmpeg --video_size 720x576 -i media.flv --video_size 640x512 -b 8000K -an -r 25 -vframes 30 media/640x480x25/build%d.jpg ffmpeg version git-Sat Sep 10 05:24:39 2011 +0200-b10ba11, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 10 2011 08:00:13 with gcc 4.2.4 (Ubuntu 4.2.4-1ubuntu4) configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libxvid --enable-x11grab --enable-pthreads libavutil 51. 16. 0 / 51. 16. 0 libavcodec 53. 13. 0 / 53. 13. 0 libavformat 53. 12. 0 / 53. 12. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 39. 0 / 2. 39. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 Unrecognized option '-video_size' Failed to set value '720x576' for option '-video_size' So first a complaint about not knowing "-video_size", and then a complaint about it being a wrong value. What am i doing wrong? Cheers, Tom On Sun, Sep 11, 2011 at 11:57 AM, Etienne Buira wrote: > On Sun, Sep 11, 2011 at 11:36:06AM +0200, Tom wrote: >> Hi, >> >> I'm running the latest ffmpeg (built this weekend from git), and got >> this error when trying to transform an flv to a set of jpegs: >> >> $ ffmpeg -s 720x576 -i media.flv -s 640x512 -b 8000K -an -r 25 >> -vframes 30 media/640x480x25/build%d.jpg >> >> ffmpeg version git-Sat Sep 10 05:24:39 2011 +0200-b10ba11, Copyright >> (c) 2000-2011 the FFmpeg developers >> ? built on Sep 10 2011 08:00:13 with gcc 4.2.4 (Ubuntu 4.2.4-1ubuntu4) >> ? configuration: --enable-gpl --enable-version3 --enable-nonfree >> --enable-postproc --enable-libfaac --enable-libopencore-amrnb >> --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora >> --enable-libvorbis --enable-libxvid --enable-x11grab --enable-pthreads >> ? libavutil ? ?51. 16. 0 / 51. 16. 0 >> ? libavcodec ? 53. 13. 0 / 53. 13. 0 >> ? libavformat ?53. 12. 0 / 53. 12. 0 >> ? libavdevice ?53. ?3. 0 / 53. ?3. 0 >> ? libavfilter ? 2. 39. 0 / ?2. 39. 0 >> ? libswscale ? ?2. ?1. 0 / ?2. ?1. 0 >> ? libpostproc ?51. ?2. 0 / 51. ?2. 0 >> Option video_size not found. >> >> >> I've googled, but option video_size is apparently quite unknown on the >> documentation page.. >> >> Can someone help? > > Hi. > > It's a string describing frame size (instead of -s before -i). And true, > not in doc, but present in ffmpeg -h. > > Regards. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From etienne.buira.lists at free.fr Sun Sep 11 20:42:26 2011 From: etienne.buira.lists at free.fr (Etienne Buira) Date: Sun, 11 Sep 2011 20:42:26 +0200 Subject: [FFmpeg-user] Option video_size In-Reply-To: References: <20110911095719.GS11600@epicure.lazyet.homelinux.net> Message-ID: <20110911184226.GT11600@epicure.lazyet.homelinux.net> On Sun, Sep 11, 2011 at 08:14:32PM +0200, Tom wrote: > Hi, > > So I changed "-s" to "--video_size" and got this error now: > > $ ffmpeg --video_size 720x576 -i media.flv --video_size 640x512 -b > 8000K -an -r 25 > -vframes 30 media/640x480x25/build%d.jpg Didn't notice at first that you were using flv, so you should actually not use -video_size (which is to set _input_ frame size on formats that doesn't hold this information). From snthdiueoa at gmail.com Sun Sep 11 21:28:48 2011 From: snthdiueoa at gmail.com (Tom) Date: Sun, 11 Sep 2011 21:28:48 +0200 Subject: [FFmpeg-user] Option video_size In-Reply-To: <20110911184226.GT11600@epicure.lazyet.homelinux.net> References: <20110911095719.GS11600@epicure.lazyet.homelinux.net> <20110911184226.GT11600@epicure.lazyet.homelinux.net> Message-ID: On Sun, Sep 11, 2011 at 8:42 PM, Etienne Buira wrote: > On Sun, Sep 11, 2011 at 08:14:32PM +0200, Tom wrote: >> Hi, >> >> So I changed "-s" to "--video_size" and got this error now: >> >> $ ffmpeg --video_size 720x576 -i media.flv --video_size 640x512 -b >> 8000K -an -r 25 >> ?-vframes 30 media/640x480x25/build%d.jpg > > Didn't notice at first that you were using flv, so you should actually > not use -video_size (which is to set _input_ frame size on formats that > doesn't hold this information). So i removed the first --video_size now, and ffmpeg got a step further: ffmpeg version git-Sat Sep 10 05:24:39 2011 +0200-b10ba11, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 10 2011 08:00:13 with gcc 4.2.4 (Ubuntu 4.2.4-1ubuntu4) configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libxvid --enable-x11grab --enable-pthreads libavutil 51. 16. 0 / 51. 16. 0 libavcodec 53. 13. 0 / 53. 13. 0 libavformat 53. 12. 0 / 53. 12. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 39. 0 / 2. 39. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 [flv @ 0x8e02aa0] Estimating duration from bitrate, this may be inaccurate Seems stream 0 codec frame rate differs from container frame rate: 1000.00 (1000/1) -> 25.00 (25/1) Input #0, flv, from 'media.flv': Metadata: duration : 23 width : 720 height : 576 videodatarate : 195 framerate : 50 videocodecid : 2 audiodatarate : 125 audiosamplerate : 44100 audiosamplesize : 16 stereo : true audiocodecid : 2 encoder : Lavf52.83.0 filesize : 19656777 Duration: 00:00:23.06, start: 0.000000, bitrate: 328 kb/s Stream #0.0: Video: flv1, yuv420p, 720x576, 200 kb/s, 25 tbr, 1k tbn, 1k tbc Stream #0.1: Audio: mp3, 44100 Hz, stereo, s16, 128 kb/s Unrecognized option '-video_size' Failed to set value '640x512' for option '-video_size' but still the same error.. > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From snthdiueoa at gmail.com Mon Sep 12 07:18:36 2011 From: snthdiueoa at gmail.com (Tom) Date: Mon, 12 Sep 2011 07:18:36 +0200 Subject: [FFmpeg-user] Option video_size In-Reply-To: References: <20110911095719.GS11600@epicure.lazyet.homelinux.net> <20110911184226.GT11600@epicure.lazyet.homelinux.net> Message-ID: Sigh, silly me, i used "--video_size" instead of "-video_size". Don't even know why.. Works now! Thanks Etienne! On Sun, Sep 11, 2011 at 9:28 PM, Tom wrote: > On Sun, Sep 11, 2011 at 8:42 PM, Etienne Buira > wrote: >> On Sun, Sep 11, 2011 at 08:14:32PM +0200, Tom wrote: >>> Hi, >>> >>> So I changed "-s" to "--video_size" and got this error now: >>> >>> $ ffmpeg --video_size 720x576 -i media.flv --video_size 640x512 -b >>> 8000K -an -r 25 >>> ?-vframes 30 media/640x480x25/build%d.jpg >> >> Didn't notice at first that you were using flv, so you should actually >> not use -video_size (which is to set _input_ frame size on formats that >> doesn't hold this information). > > So i removed the first --video_size now, and ffmpeg got a step further: > > ffmpeg version git-Sat Sep 10 05:24:39 2011 +0200-b10ba11, Copyright > (c) 2000-2011 the FFmpeg developers > ?built on Sep 10 2011 08:00:13 with gcc 4.2.4 (Ubuntu 4.2.4-1ubuntu4) > ?configuration: --enable-gpl --enable-version3 --enable-nonfree > --enable-postproc --enable-libfaac --enable-libopencore-amrnb > --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora > --enable-libvorbis --enable-libxvid --enable-x11grab --enable-pthreads > ?libavutil ? ?51. 16. 0 / 51. 16. 0 > ?libavcodec ? 53. 13. 0 / 53. 13. 0 > ?libavformat ?53. 12. 0 / 53. 12. 0 > ?libavdevice ?53. ?3. 0 / 53. ?3. 0 > ?libavfilter ? 2. 39. 0 / ?2. 39. 0 > ?libswscale ? ?2. ?1. 0 / ?2. ?1. 0 > ?libpostproc ?51. ?2. 0 / 51. ?2. 0 > [flv @ 0x8e02aa0] Estimating duration from bitrate, this may be inaccurate > > Seems stream 0 codec frame rate differs from container frame rate: > 1000.00 (1000/1) -> 25.00 (25/1) > Input #0, flv, from 'media.flv': > ?Metadata: > ? ?duration ? ? ? ?: 23 > ? ?width ? ? ? ? ? : 720 > ? ?height ? ? ? ? ?: 576 > ? ?videodatarate ? : 195 > ? ?framerate ? ? ? : 50 > ? ?videocodecid ? ?: 2 > ? ?audiodatarate ? : 125 > ? ?audiosamplerate : 44100 > ? ?audiosamplesize : 16 > ? ?stereo ? ? ? ? ?: true > ? ?audiocodecid ? ?: 2 > ? ?encoder ? ? ? ? : Lavf52.83.0 > ? ?filesize ? ? ? ?: 19656777 > ?Duration: 00:00:23.06, start: 0.000000, bitrate: 328 kb/s > ? ?Stream #0.0: Video: flv1, yuv420p, 720x576, 200 kb/s, 25 tbr, 1k tbn, 1k tbc > ? ?Stream #0.1: Audio: mp3, 44100 Hz, stereo, s16, 128 kb/s > Unrecognized option '-video_size' > Failed to set value '640x512' for option '-video_size' > > but still the same error.. > > >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > From atish.kathpal at gmail.com Mon Sep 12 07:26:43 2011 From: atish.kathpal at gmail.com (Atish Kathpal) Date: Mon, 12 Sep 2011 10:56:43 +0530 Subject: [FFmpeg-user] What is the transcoding unit for FFMPEG? Message-ID: Hi friends What is the transcoding unit for FFMEG? Is it a file or can we transcode a single file as user defined chunks of equal/variable size? Eg: Given a video- Austral.mpeg (10mins). Can I divide it into 10 chunks of 1 min each while transcoding, such that each 1min chunk is accessible once transcoding for it is over? Pardon me if this doubt has already been addressed, please point me to the link/mail-thread where I could get information on the same. Best regards Atish From bernicesurprise at gmail.com Mon Sep 12 12:50:27 2011 From: bernicesurprise at gmail.com (k b) Date: Mon, 12 Sep 2011 22:50:27 +1200 Subject: [FFmpeg-user] Dies without message when streaming In-Reply-To: References: Message-ID: Progress; no luck with ffserver, but I do have streaming via crtmpserver ( http://www.rtmpd.com/) working! This looks like the perfect option for people who want to stream video to embedded flash players. So far, I've only got it working for a prerecorded file, though. So this works fine: ffmpeg -i "testfile.avi" -re -vcodec libx264 -vpre default -vpre baseline -b 500000 -s 320x180 -strict experimental -g 25 -me_method zero -acodec aac -ab 96000 -ar 48000 -ac 2 -vbsf h264_mp4toannexb -f mpegts udp:// 127.0.0.1:10000?pkt_size=1316 But this doesn't: ffmpeg -f video4linux2 -i /dev/video0 -re -vcodec libx264 -vpre default -vpre baseline -b 500000 -s 320x180 -strict experimental -g 25 -me_method zero -acodec aac -ab 96000 -ar 48000 -ac 2 -vbsf h264_mp4toannexb -f mpegts udp://127.0.0.1:10000?pkt_size=1316 ffmpeg gives this error: [mpegts @ 0x71d2c0]st:0 error, non monotone timestamps 1 >= 1 av_interleaved_write_frame(): Operation not permitted ioctl(VIDIOC_QBUF) I've tried experimenting, but haven't been able to come up with anything that doesn't fail for either ffmpeg or crtmpserver. Any idea how to fix the av_interleaved_write_frame error? bs From bernicesurprise at gmail.com Mon Sep 12 12:52:07 2011 From: bernicesurprise at gmail.com (k b) Date: Mon, 12 Sep 2011 22:52:07 +1200 Subject: [FFmpeg-user] Dies without message when streaming In-Reply-To: References: Message-ID: Progress; no luck with ffserver, but I do have streaming via crtmpserver ( http://www.rtmpd.com/) working! This looks like the perfect option for people who want to stream video to embedded flash players. So far, I've only got it working for a prerecorded file, though. So this works fine: ffmpeg -i "testfile.avi" -re -vcodec libx264 -vpre default -vpre baseline -b 500000 -s 320x180 -strict experimental -g 25 -me_method zero -acodec aac -ab 96000 -ar 48000 -ac 2 -vbsf h264_mp4toannexb -f mpegts udp:// 127.0.0.1:10000?pkt_size=1316 But this doesn't: ffmpeg -f video4linux2 -i /dev/video0 -re -vcodec libx264 -vpre default -vpre baseline -b 500000 -s 320x180 -strict experimental -g 25 -me_method zero -acodec aac -ab 96000 -ar 48000 -ac 2 -vbsf h264_mp4toannexb -f mpegts udp://127.0.0.1:10000?pkt_size=1316 ffmpeg gives this error: [mpegts @ 0x71d2c0]st:0 error, non monotone timestamps 1 >= 1 av_interleaved_write_frame(): Operation not permitted ioctl(VIDIOC_QBUF) I've tried experimenting, but haven't been able to come up with anything that doesn't fail for either ffmpeg or crtmpserver. Any idea how to fix the av_interleaved_write_frame error? bs From scary10 at mail.ru Sun Sep 11 21:41:59 2011 From: scary10 at mail.ru (=?UTF-8?B?U2Nhcnk=?=) Date: Sun, 11 Sep 2011 23:41:59 +0400 Subject: [FFmpeg-user] =?utf-8?q?ffmpeg_in_deamond_mode?= Message-ID: help me run ffmpeg in ?deamond mode with cron on linux system ...? -- ??????????? ????????? ????? - ??? ?????????? ? ?????????. ????? ????????? m.mail.ru ? ?????? ???????? From atish.kathpal at gmail.com Tue Sep 13 06:40:17 2011 From: atish.kathpal at gmail.com (Atish Kathpal) Date: Tue, 13 Sep 2011 10:10:17 +0530 Subject: [FFmpeg-user] What is the transcoding unit for FFMPEG? In-Reply-To: References: Message-ID: Hello guys Please help me resolve this doubt. Waiting eagerly for your replies. Regards Atish On Mon, Sep 12, 2011 at 10:56 AM, Atish Kathpal wrote: > Hi friends > > What is the transcoding unit for FFMEG? Is it a file or can we transcode a > single file as user defined chunks of equal/variable size? > > Eg: Given a video- Austral.mpeg (10mins). Can I divide it into 10 chunks of > 1 min each while transcoding, such that each 1min chunk is accessible once > transcoding for it is over? > > Pardon me if this doubt has already been addressed, please point me to the > link/mail-thread where I could get information on the same. > > Best regards > Atish > From etienne.buira.lists at free.fr Tue Sep 13 06:51:20 2011 From: etienne.buira.lists at free.fr (Etienne Buira) Date: Tue, 13 Sep 2011 06:51:20 +0200 Subject: [FFmpeg-user] What is the transcoding unit for FFMPEG? In-Reply-To: References: Message-ID: <20110913045120.GH2506@epicure.lazyet.homelinux.net> On Mon, Sep 12, 2011 at 10:56:43AM +0530, Atish Kathpal wrote: > Hi friends > > What is the transcoding unit for FFMEG? Is it a file or can we transcode a > single file as user defined chunks of equal/variable size? > > Eg: Given a video- Austral.mpeg (10mins). Can I divide it into 10 chunks of > 1 min each while transcoding, such that each 1min chunk is accessible once > transcoding for it is over? > > Pardon me if this doubt has already been addressed, please point me to the > link/mail-thread where I could get information on the same. > > Best regards > Atish Hi. Look at -ss and -t documentation, I think that's the only way to achieve what you want. Regards. From gaurav.g at sasken.com Tue Sep 13 09:20:11 2011 From: gaurav.g at sasken.com (Gaurav G) Date: Tue, 13 Sep 2011 12:50:11 +0530 Subject: [FFmpeg-user] trick play maximum support speed Message-ID: <6F91E0FFDA542149961F7BDED2D2B94B1EBE8230D8@EXGMBX01.sasken.com> Hi Michael, This is my very first mail to the community. In my project i am using FFMPEG parser to get the raw/elementary stream and decode it using my own code. In one of the issue i am observing that if i try to do the trick play (Fast Forward) for more than 64x then i am not getting the proper data. Instead i am getting the following error [asf @ 0x13bac0]freeing incomplete packet size 26152, new 25636 the error continued with different value of pkt.size and packet_obj_size; in the last i receive the EOF. I want to cross check is their any constraint in the amount of speed that we can use in the trickplay use-case. If not can you please guide me to resolve the issue. Thanks, Gaurav SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html From bernicesurprise at gmail.com Tue Sep 13 10:19:46 2011 From: bernicesurprise at gmail.com (k b) Date: Tue, 13 Sep 2011 20:19:46 +1200 Subject: [FFmpeg-user] Dies without message when streaming In-Reply-To: References: Message-ID: On Mon, Sep 12, 2011 at 10:50 PM, k b wrote: > I've tried experimenting, but haven't been able to come up with anything > that doesn't fail for either ffmpeg or crtmpserver. Any idea how to fix the > av_interleaved_write_frame error? > Ok, specifying a frame rate seems to do it. That was simple... Live video streaming from webcam to flowplayer via ffmpeg and crtmpserver is go 8) bs From galacap at gmail.com Tue Sep 13 13:02:38 2011 From: galacap at gmail.com (Roee Kashi) Date: Tue, 13 Sep 2011 14:02:38 +0300 Subject: [FFmpeg-user] converting yuv420p rawdata to avi - colors issue Message-ID: hi, i'm trying to convert my yuv raw data to .. actually anything. my yuv type is YV12 which i understand is yuv420p, so in order to convert it i tried the following command line: 'ffmpeg.exe -s 352x288 -vcodec rawvideo -f rawvideo -pix_fmt yuv420p -i 1.yuv -sameq test.avi' i'm getting a beautiful avi file, but not exactly with my original colors. is there a place in command line where i can somehow try to fix it? if not, what do you think should i change in my yuv rawdata in order to get a better output? you can see in this picture the different between the original video color (in the right) to the converted one: http://oi55.tinypic.com/2cp5lc9.jpg thanks. From rhodri at kynesim.co.uk Tue Sep 13 13:27:00 2011 From: rhodri at kynesim.co.uk (Rhodri James) Date: Tue, 13 Sep 2011 12:27:00 +0100 Subject: [FFmpeg-user] converting yuv420p rawdata to avi - colors issue In-Reply-To: References: Message-ID: On Tue, 13 Sep 2011 12:02:38 +0100, Roee Kashi wrote: > hi, > i'm trying to convert my yuv raw data to .. actually anything. > my yuv type is YV12 which i understand is yuv420p, Sadly, no. YV12 has the U and V planes in the opposite order to yuv420p, giving the wrong colours you see. What you need is "yvu420p", except that it doesn't exist... -- Rhodri James Kynesim Ltd From galacap at gmail.com Tue Sep 13 13:36:10 2011 From: galacap at gmail.com (Roee Kashi) Date: Tue, 13 Sep 2011 14:36:10 +0300 Subject: [FFmpeg-user] converting yuv420p rawdata to avi - colors issue In-Reply-To: References: Message-ID: VLC is based on ffmpeg libs isn't it? because in VLC the yuv file displayed ok. if i want to add a "yvu240p" option, where should i start (in what c file should i start)? or is there a manual way to define strict yv placement? :\ Thanks a lot. 2011/9/13 Rhodri James > On Tue, 13 Sep 2011 12:02:38 +0100, Roee Kashi wrote: > > hi, >> i'm trying to convert my yuv raw data to .. actually anything. >> my yuv type is YV12 which i understand is yuv420p, >> > > Sadly, no. YV12 has the U and V planes in the opposite order > to yuv420p, giving the wrong colours you see. What you need > is "yvu420p", except that it doesn't exist... > > -- > Rhodri James > Kynesim Ltd > ______________________________**_________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/**listinfo/ffmpeg-user > From forum.amit.mangal at gmail.com Tue Sep 13 17:16:23 2011 From: forum.amit.mangal at gmail.com (Amit Mangal) Date: Tue, 13 Sep 2011 08:16:23 -0700 Subject: [FFmpeg-user] playing asf file command with ffplay Message-ID: Hi everybody i want to play asf stream with ffplay any body knows command for playing asf stream with ffplay. thanks From isaac.wang at visorcorp.com Tue Sep 13 18:34:41 2011 From: isaac.wang at visorcorp.com (isyiwang) Date: Tue, 13 Sep 2011 09:34:41 -0700 (PDT) Subject: [FFmpeg-user] FFmpeg lib + UDP + Qt issues Message-ID: <1315931681363-3810551.post@n4.nabble.com> Hello, First of all, thank you to this amazing community of developers! I have received a tremendous amount of help both directly and indirectly thanks to you all. I am in need of some help regarding reading encoded video udp packets and decoding them into a Qt display. I have managed to read the video packets and display them in Qt but video playback 'stutters'. Checking the FPS of frames transmitted via the FFMpeg library to Qt, I get ~30 frames per second, so I'm led to believe that FFMpeg itself is sending me the same frames multiple times a second. Here are the steps I took: 1. Broadcast an H.264 encoded video stream via VLC to udp:/238.38.38.38:1234 2. Use FFMpeg lib to decode the stream. Here is my function (apologies in advance, if there is a better way to display code, please let me know!): bool QVideoDecoder::decodeSeekFrame(int after) { bool done=false; while(!done) { // Read a frame timeoutTimer.start(); if(av_read_frame(pFormatCtx, &packet)<0) return false; // Frame read failed (e.g. end of stream) timeoutTimer.stop(); if(packet.stream_index==videoStream) { int frameFinished; avcodec_decode_video2(pCodecCtx,pFrame,&frameFinished,&packet); // Did we get a video frame? if(frameFinished) { ffmpeg::AVRational millisecondbase = {1, 1000}; int f = packet.dts; int t = ffmpeg::av_rescale_q(packet.dts,pFormatCtx->streams[videoStream]->time_base,millisecondbase); if(LastFrameOk==false) { LastFrameOk=true; LastLastFrameTime=LastFrameTime=t; LastLastFrameNumber=LastFrameNumber=f; } else { // If we decoded 2 frames in a row, the last times are okay LastLastFrameTime = LastFrameTime; LastLastFrameNumber = LastFrameNumber; LastFrameTime=t; LastFrameNumber=f; } // Is this frame the desired frame? if(after==-1 || LastFrameNumber>=after) { // Convert the image format (init the context the first time) int w = pCodecCtx->width; int h = pCodecCtx->height; img_convert_ctx = ffmpeg::sws_getCachedContext(img_convert_ctx,w, h, pCodecCtx->pix_fmt, w, h, ffmpeg::PIX_FMT_RGB24, SWS_BICUBIC, NULL, NULL, NULL); if(img_convert_ctx == NULL) { printf("Cannot initialize the conversion context!\n"); return false; } ffmpeg::sws_scale(img_convert_ctx, pFrame->data, pFrame->linesize, 0, pCodecCtx->height, pFrameRGB->data, pFrameRGB->linesize); // Convert the frame to QImage LastFrame=QImage(w,h,QImage::Format_RGB888); for(int y=0;y<h;y++) memcpy(LastFrame.scanLine(y),pFrameRGB->data[0]+y*pFrameRGB->linesize[0],w*3); // Set the time DesiredFrameTime = ffmpeg::av_rescale_q(after,pFormatCtx->streams[videoStream]->time_base,millisecondbase); LastFrameOk=true; done = true; } } } av_free_packet(&packet); } return done; } I have a periodic timer that requests the latest frame from FFMpeg. If I set the timer to a low rate (20 FPS), I get the video played back but at a slow rate. If I set the timer to a high rate (60 FPS), I get the frames as soon as they are ready, but the video stutters. I figured that I should request the frames back as fast as possible.. Any insight on how to avoid the stuttering is appreciated. 3. Send the QImage back to Qt to be displayed -- I have tested the frames received by my Qt callback per second and can confirm that I am receiving ~ 30 FPS which leads me to believe the stuttering is caused by FFMpeg lib. 4. I also wrote an interrupting callback to be called by FFMpeg if any of the blocking functions (i.e. av_read_frame) took too long: Timer timeoutTimer; static int decode_interrupt_cb(void) { if (timeoutTimer.getElapsedTimeInMilliSec() > 1000) { qDebug() << "Video Timed Out!"; return 1; } return 0; } Although I sometimes do receive the "Video Timed Out!" call, the av_read_frame function still blocks... Any help would be greatly appreciated! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/FFmpeg-lib-UDP-Qt-issues-tp3810551p3810551.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From rogerdpack2 at gmail.com Tue Sep 13 19:22:53 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Tue, 13 Sep 2011 11:22:53 -0600 Subject: [FFmpeg-user] filter to "combine" audio tracks (audio mixer) In-Reply-To: <20110908202457.GA6118@geppetto> References: <20110908202457.GA6118@geppetto> Message-ID: > Yes a mixing audio filter (2 in, 1 out) is the right solution. > > Audio filtering is a work in progress, many bits have been already > committed and we're close to integration (where "close" is hardly > definable, as it depends on many variable factors including my > available free time). Good to hear. I'll put a $150 bounty on a working system that can two audio inputs and combines them into one (ex: 1 mono + 1 stereo -> 1 stereo). I have no idea how so that's my contribution :) -roger- From t.step at comcast.net Tue Sep 13 23:21:29 2011 From: t.step at comcast.net (Tom Step) Date: Tue, 13 Sep 2011 15:21:29 -0600 Subject: [FFmpeg-user] ffmpeg installed properly? Noob Message-ID: <813AE82C-97C3-4BF4-98EA-D0134C80A675@comcast.net> Tyring to install ffmpeg on my Mac (OS 10.6.8) to learn CLI. Used MacPorts to do install, it came back with a ton of errors, then did port clean ffmpeg, then: svn checkout svn://svn.ffmpeg.org/ffmpeg/trunk ffmpeg Checked out revision 26402. ./configure --enable-shared --disable-mmx --arch=x86_64install prefix /usr/local source path /Users/shoei/ffmpeg/ffmpeg C compiler gcc ARCH x86 (generic) sudo make install make FFmpeg version SVN-r26402, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 8 2011 16:56:02 with gcc 4.2.1 (Apple Inc. build 5646) (dot 1) configuration: --enable-shared --disable-mmx --arch=x86_64 libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 Hyper fast Audio and Video encoder , which seems to have installed ffmeg. Problem is that I get errors when running basic flips (.m4a to mp3 for example). See below for CLI transcript. Checked lib*, -format, -codecs to make sure they're available, don't know what else to do. Curiously I have a ~/ffmpeg/ffmpeg/ stucture, could that be the issue where the PATH is not being seen by ffmpeg? ===================================================================================== shoeis-computer:Volumes shoei$ ffmpeg -i ~/Desktop/Hunger.m4a mp3 -ab 192k ~/Desktop/Hunger.mp3 FFmpeg version SVN-r26402, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 8 2011 16:56:02 with gcc 4.2.1 (Apple Inc. build 5646) (dot 1) configuration: --enable-shared --disable-mmx --arch=x86_64 libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Users/shoei/Desktop/Hunger.m4a': Metadata: major_brand : M4A minor_version : 0 compatible_brands: M4A mp42isom creation_time : 2007-10-04 07:53:54 title : Hunger artist : Hans Zimmer composer : Hans Zimmer album : Black Hawk Down track : 1 date : 2002 encoder : iTunes v7.3.2.6, QuickTime 7.2 Duration: 00:06:35.83, start: 0.000000, bitrate: 129 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16, 128 kb/s Metadata: creation_time : 2007-10-04 07:53:54 Unable to find a suitable output format for 'mp3' ============================================================== shoeis-computer:ffmpeg shoei$ ffmpeg -i ~/Desktop/NoSuchThing.wav aac -ab 192k ~/Desktop/Hunger.aac FFmpeg version SVN-r26402, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 8 2011 16:56:02 with gcc 4.2.1 (Apple Inc. build 5646) (dot 1) configuration: --enable-shared --disable-mmx --arch=x86_64 libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 [wav @ 0x1009600] max_analyze_duration reached Input #0, wav, from '/Users/shoei/Desktop/NoSuchThing.wav': Duration: 00:03:51.28, bitrate: 1411 kb/s Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s Unable to find a suitable output format for 'aac' ============================================== shoeis-computer:ffmpeg shoei$ ffmpeg -i ~/Desktop/NoSuchThing.wav aiff -ab 192k ~/Desktop/Hunger.aiff FFmpeg version SVN-r26402, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 8 2011 16:56:02 with gcc 4.2.1 (Apple Inc. build 5646) (dot 1) configuration: --enable-shared --disable-mmx --arch=x86_64 libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 [wav @ 0x1009600] max_analyze_duration reached Input #0, wav, from '/Users/shoei/Desktop/NoSuchThing.wav': Duration: 00:03:51.28, bitrate: 1411 kb/s Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s Unable to find a suitable output format for 'aiff' shoeis-computer:ffmpeg shoei$ ffmpeg -i ~/Desktop/NoSuchThing.wav wav ~/Desktop/testflip.wav FFmpeg version SVN-r26402, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 8 2011 16:56:02 with gcc 4.2.1 (Apple Inc. build 5646) (dot 1) configuration: --enable-shared --disable-mmx --arch=x86_64 libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 [wav @ 0x1009600] max_analyze_duration reached Input #0, wav, from '/Users/shoei/Desktop/NoSuchThing.wav': Duration: 00:03:51.28, bitrate: 1411 kb/s Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s Unable to find a suitable output format for 'wav' ======================================================================= shoeis-computer:usr shoei$ ffmpeg -i ~/Desktop/Hunger.m4a -acodec libmp3lame -ab 320k ~/Desktop/Hunger.mp3 FFmpeg version SVN-r26402, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 8 2011 16:56:02 with gcc 4.2.1 (Apple Inc. build 5646) (dot 1) configuration: --enable-shared --disable-mmx --arch=x86_64 libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Users/shoei/Desktop/Hunger.m4a': Metadata: major_brand : M4A minor_version : 0 compatible_brands: M4A mp42isom creation_time : 2007-10-04 07:53:54 title : Hunger artist : Hans Zimmer composer : Hans Zimmer album : Black Hawk Down track : 1 date : 2002 encoder : iTunes v7.3.2.6, QuickTime 7.2 Duration: 00:06:35.83, start: 0.000000, bitrate: 129 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16, 128 kb/s Metadata: creation_time : 2007-10-04 07:53:54 Unknown encoder 'libmp3lame' =============================================================== etc... It did do these however: +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ shoeis-computer:usr shoei$ ffmpeg -i ~/Desktop/Hunger.m4a -acodec real_144 -ac 1 ~/Desktop/testflip4.rm FFmpeg version SVN-r26402, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 8 2011 16:56:02 with gcc 4.2.1 (Apple Inc. build 5646) (dot 1) configuration: --enable-shared --disable-mmx --arch=x86_64 libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Users/shoei/Desktop/Hunger.m4a': Metadata: major_brand : M4A minor_version : 0 compatible_brands: M4A mp42isom creation_time : 2007-10-04 07:53:54 title : Hunger artist : Hans Zimmer composer : Hans Zimmer album : Black Hawk Down track : 1 date : 2002 encoder : iTunes v7.3.2.6, QuickTime 7.2 Duration: 00:06:35.83, start: 0.000000, bitrate: 129 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16, 128 kb/s Metadata: creation_time : 2007-10-04 07:53:54 Output #0, rm, to '/Users/shoei/Desktop/testflip4.rm': Metadata: major_brand : M4A minor_version : 0 compatible_brands: M4A mp42isom creation_time : 2007-10-04 07:53:54 title : Hunger artist : Hans Zimmer composer : Hans Zimmer album : Black Hawk Down track : 1 date : 2002 encoder : Lavf52.93.0 Stream #0.0(und): Audio: real_144, 44100 Hz, 1 channels, s16, 8 kb/s Metadata: creation_time : 2007-10-04 07:53:54 Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding [real_144 @ 0x100b600] Overflow. Broken sample? Last message repeated 25 timese= 70.6kbits/s size= 3410kB time=395.83 bitrate= 70.6kbits/s video:0kB audio:2131kB global headers:0kB muxing overhead 60.012511% +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ shoeis-computer:usr shoei$ ffmpeg -i ~/Desktop/Hunger.m4a -acodec mp2 ~/Desktop/testflip.mp2 FFmpeg version SVN-r26402, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 8 2011 16:56:02 with gcc 4.2.1 (Apple Inc. build 5646) (dot 1) configuration: --enable-shared --disable-mmx --arch=x86_64 libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Users/shoei/Desktop/Hunger.m4a': Metadata: major_brand : M4A minor_version : 0 compatible_brands: M4A mp42isom creation_time : 2007-10-04 07:53:54 title : Hunger artist : Hans Zimmer composer : Hans Zimmer album : Black Hawk Down track : 1 date : 2002 encoder : iTunes v7.3.2.6, QuickTime 7.2 Duration: 00:06:35.83, start: 0.000000, bitrate: 129 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16, 128 kb/s Metadata: creation_time : 2007-10-04 07:53:54 Output #0, mp2, to '/Users/shoei/Desktop/testflip.mp2': Metadata: major_brand : M4A minor_version : 0 compatible_brands: M4A mp42isom creation_time : 2007-10-04 07:53:54 title : Hunger artist : Hans Zimmer composer : Hans Zimmer album : Black Hawk Down track : 1 date : 2002 encoder : Lavf52.93.0 Stream #0.0(und): Audio: mp2, 44100 Hz, stereo, s16, 64 kb/s Metadata: creation_time : 2007-10-04 07:53:54 Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding size= 3092kB time=395.83 bitrate= 64.0kbits/s video:0kB audio:3092kB global headers:0kB muxing overhead 0.000000% ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ END From isaac.wang at visorcorp.com Tue Sep 13 23:43:54 2011 From: isaac.wang at visorcorp.com (isyiwang) Date: Tue, 13 Sep 2011 14:43:54 -0700 (PDT) Subject: [FFmpeg-user] filter to "combine" audio tracks (audio mixer) In-Reply-To: References: <20110908202457.GA6118@geppetto> Message-ID: <1315950234245-3811284.post@n4.nabble.com> do you have to use ffmpeg to combine the audio? i'm familiar with lightweight c++ audio libraries that you can use to accomplish the same thing -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/filter-to-combine-audio-tracks-audio-mixer-tp3799520p3811284.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From rogerdpack2 at gmail.com Wed Sep 14 01:31:11 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Tue, 13 Sep 2011 17:31:11 -0600 Subject: [FFmpeg-user] filter to "combine" audio tracks (audio mixer) In-Reply-To: <1315950234245-3811284.post@n4.nabble.com> References: <20110908202457.GA6118@geppetto> <1315950234245-3811284.post@n4.nabble.com> Message-ID: > do you have to use ffmpeg to combine the audio? ?i'm familiar with > lightweight c++ audio libraries that you can use to accomplish the same > thing My first attempt will be to record to two audio streams, then pull the out and combine them with sox then add them back in. It should work I hope. -r From rogerdpack2 at gmail.com Wed Sep 14 01:32:22 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Tue, 13 Sep 2011 17:32:22 -0600 Subject: [FFmpeg-user] ffmpeg installed properly? Noob In-Reply-To: <813AE82C-97C3-4BF4-98EA-D0134C80A675@comcast.net> References: <813AE82C-97C3-4BF4-98EA-D0134C80A675@comcast.net> Message-ID: > Unable to find a suitable output format for 'mp3' are there any port variants that might help? From lou at lrcd.com Wed Sep 14 01:50:15 2011 From: lou at lrcd.com (Lou) Date: Tue, 13 Sep 2011 15:50:15 -0800 Subject: [FFmpeg-user] ffmpeg installed properly? Noob In-Reply-To: <813AE82C-97C3-4BF4-98EA-D0134C80A675@comcast.net> References: <813AE82C-97C3-4BF4-98EA-D0134C80A675@comcast.net> Message-ID: <20110913155015.6d08f4e0@lrcd.com> On Tue, 13 Sep 2011 15:21:29 -0600 Tom Step wrote: > Tyring to install ffmpeg on my Mac (OS 10.6.8) to learn CLI. Used > MacPorts to do install, it came back with a ton of errors, then did > port clean ffmpeg, then: > > svn checkout svn://svn.ffmpeg.org/ffmpeg/trunk ffmpeg > Checked out revision 26402. FFmpeg uses Git now. The SVN repository is outdated. You should switch to the Git repository as the last SVN commit was in January 2011 (or simply download a Git snapshot): http://git.videolan.org/?p=ffmpeg.git;a=snapshot;h=HEAD;sf=tgz > ./configure --enable-shared --disable-mmx --arch=x86_64 --disable-mmx was used for PowerPC I think. I don't think you need it for the recent architecture. But IANAMacUser. More importantly, if you want mp3 encoding support, you need to install lame and build ffmpeg with --enable-libmp3lame. > shoeis-computer:Volumes shoei$ ffmpeg -i ~/Desktop/Hunger.m4a mp3 -ab > 192k ~/Desktop/Hunger.mp3 Your command is telling ffmpeg that your output is a file named "mp3". Once you recompile ffmpeg with --enable-libmp3lame the remove the superfluous "mp3" from your command and it might work, and if not then add "-acodec libmp3lame". From maheshraja8 at gmail.com Wed Sep 14 11:24:20 2011 From: maheshraja8 at gmail.com (mahesh raja) Date: Wed, 14 Sep 2011 14:54:20 +0530 Subject: [FFmpeg-user] Invocation of ffplay API/functions from Aplication code Message-ID: Hi We have downloaded FFMpeg and built FFplay. The player is executing fine. We have seen that the all the functions like stream_open, event_loop are defined as static in ffplay.c and are only part of the ffplay executable. They are not available as part of any library. If we want to invoke these APIs from our application so that the application can control when to play and when to pause, is there a way to do ? We have seen that this has the license LGPL 2.1, so please suggest if there is a way to invoke the APIs from our application. Thanks, mahesh From rogerdpack2 at gmail.com Wed Sep 14 16:11:03 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 14 Sep 2011 08:11:03 -0600 Subject: [FFmpeg-user] Invocation of ffplay API/functions from Aplication code In-Reply-To: References: Message-ID: > We have downloaded FFMpeg and built FFplay. The player is executing fine. > We have seen that the all the functions like stream_open, event_loop are > defined as static in ffplay.c and are only part of the ffplay executable. > They are not available as part of any library. If we want to invoke these > APIs from our application so that the application can control > when to play and when to pause, ?is there a way to do ? We have seen that > this has the license LGPL 2.1, so please suggest if there is a way to > invoke the APIs from our application. cop out answer, but mplayer has a slave mode... From Peter.Wright at Renishaw.com Wed Sep 14 17:45:45 2011 From: Peter.Wright at Renishaw.com (Peter Wright) Date: Wed, 14 Sep 2011 16:45:45 +0100 Subject: [FFmpeg-user] mp42 installation Message-ID: Is it possible to add the Microsoft MP42 audio codec to a windows build of FFMPEG without rebuilding FFMPEG? Regards Peter Wright BEng (Hons) Website Developer -------------------------------------------------------------------------------------------------- This email and any attachments are confidential and are for the use of the addressee only. If you are not the addressee, you must not use or disclose the contents to any other person. Please immediately notify the sender and delete the email. Statements and opinions expressed here may not represent those of the company. Email correspondence is monitored by the company. This information may be subject to export control regulation. You are obliged to comply with such regulations. Renishaw plc (company number 1106260), Wotton Travel Limited (company number 01973158), PulseTeq Limited (company number 4392865) and Renishaw Advanced Materials Limited (company number 04632041), are companies registered in England and Wales with a registered office at New Mills, Wotton-under-Edge, Gloucestershire, GL12 8JR, United Kingdom, Telephone +44 1453 524524. -------------------------------------------------------------------------------------------------- From t.step at comcast.net Wed Sep 14 19:45:45 2011 From: t.step at comcast.net (Tom Step) Date: Wed, 14 Sep 2011 11:45:45 -0600 Subject: [FFmpeg-user] ffmpeg installed properly? Noob In-Reply-To: <20110913155015.6d08f4e0@lrcd.com> References: <813AE82C-97C3-4BF4-98EA-D0134C80A675@comcast.net> <20110913155015.6d08f4e0@lrcd.com> Message-ID: <51911F64-9A2C-464E-B78A-50BDCE0BF909@comcast.net> Downloaded Git version to Downloads folder. it created a folder called "ffmpeg-HEAD-d2a847c" which has another "ffmpeg-HEAD-d2a847c" folder within it, which has the configure executable. Now in Terminal do I: cd ~/Downloads/ffmpeg-HEAD-d2a847c/ffmpeg-HEAD-d2a847c ./configure --enable-libmp3lame --enable-shared make sudo make install Wil this make a "clean" install with necessary parts? and What about other errors with different codecs? FWIW I did run the original configure with--libmp3lame in it.And I do have libfaac but can't make a .aac versions either... Thanks for helping a noob out Tom On Sep 13, 2011, at 5:50 PM, Lou wrote: > On Tue, 13 Sep 2011 15:21:29 -0600 > Tom Step wrote: > >> Tyring to install ffmpeg on my Mac (OS 10.6.8) to learn CLI. Used >> MacPorts to do install, it came back with a ton of errors, then did >> port clean ffmpeg, then: >> >> svn checkout svn://svn.ffmpeg.org/ffmpeg/trunk ffmpeg >> Checked out revision 26402. > > FFmpeg uses Git now. The SVN repository is outdated. You should switch > to the Git repository as the last SVN commit was in January 2011 (or > simply download a Git snapshot): > > http://git.videolan.org/?p=ffmpeg.git;a=snapshot;h=HEAD;sf=tgz > >> ./configure --enable-shared --disable-mmx --arch=x86_64 > > --disable-mmx was used for PowerPC I think. I don't think you need it > for the recent architecture. But IANAMacUser. > > More importantly, if you want mp3 encoding support, you need to install > lame and build ffmpeg with --enable-libmp3lame. > >> shoeis-computer:Volumes shoei$ ffmpeg -i ~/Desktop/Hunger.m4a mp3 -ab >> 192k ~/Desktop/Hunger.mp3 > > Your command is telling ffmpeg that your output is a file named "mp3". > Once you recompile ffmpeg with --enable-libmp3lame the remove the > superfluous "mp3" from your command and it might work, and if not then > add "-acodec libmp3lame". > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From t.step at comcast.net Wed Sep 14 23:03:41 2011 From: t.step at comcast.net (Tom Step) Date: Wed, 14 Sep 2011 15:03:41 -0600 Subject: [FFmpeg-user] ffmpeg installed properly? Noob In-Reply-To: <51911F64-9A2C-464E-B78A-50BDCE0BF909@comcast.net> References: <813AE82C-97C3-4BF4-98EA-D0134C80A675@comcast.net> <20110913155015.6d08f4e0@lrcd.com> <51911F64-9A2C-464E-B78A-50BDCE0BF909@comcast.net> Message-ID: installed LAME, x264, and faac/faad libs, then tried to install ffmpeg again with all libs, then without, still get yasm error (isn't that a linux thing?) shoeis-computer:~ shoei$ cd ~/ffmpeg/ffmpeg-HEAD-d2a847c/ shoeis-computer:ffmpeg-HEAD-d2a847c shoei$ ./configure yasm not found, use --disable-yasm for a crippled build 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. On Sep 14, 2011, at 11:45 AM, Tom Step wrote: > Downloaded Git version to Downloads folder. it created a folder called "ffmpeg-HEAD-d2a847c" which has another "ffmpeg-HEAD-d2a847c" folder within it, which has the configure executable. > > Now in Terminal do I: > cd ~/Downloads/ffmpeg-HEAD-d2a847c/ffmpeg-HEAD-d2a847c > ./configure --enable-libmp3lame --enable-shared > make > sudo make install > > Wil this make a "clean" install with necessary parts? > and > What about other errors with different codecs? FWIW I did run the original configure with--libmp3lame in it.And I do have libfaac but can't make a .aac versions either... > > Thanks for helping a noob out > Tom > > On Sep 13, 2011, at 5:50 PM, Lou wrote: > >> On Tue, 13 Sep 2011 15:21:29 -0600 >> Tom Step wrote: >> >>> Tyring to install ffmpeg on my Mac (OS 10.6.8) to learn CLI. Used >>> MacPorts to do install, it came back with a ton of errors, then did >>> port clean ffmpeg, then: >>> >>> svn checkout svn://svn.ffmpeg.org/ffmpeg/trunk ffmpeg >>> Checked out revision 26402. >> >> FFmpeg uses Git now. The SVN repository is outdated. You should switch >> to the Git repository as the last SVN commit was in January 2011 (or >> simply download a Git snapshot): >> >> http://git.videolan.org/?p=ffmpeg.git;a=snapshot;h=HEAD;sf=tgz >> >>> ./configure --enable-shared --disable-mmx --arch=x86_64 >> >> --disable-mmx was used for PowerPC I think. I don't think you need it >> for the recent architecture. But IANAMacUser. >> >> More importantly, if you want mp3 encoding support, you need to install >> lame and build ffmpeg with --enable-libmp3lame. >> >>> shoeis-computer:Volumes shoei$ ffmpeg -i ~/Desktop/Hunger.m4a mp3 -ab >>> 192k ~/Desktop/Hunger.mp3 >> >> Your command is telling ffmpeg that your output is a file named "mp3". >> Once you recompile ffmpeg with --enable-libmp3lame the remove the >> superfluous "mp3" from your command and it might work, and if not then >> add "-acodec libmp3lame". >> _______________________________________________ >> 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 coniophora at gmail.com Thu Sep 15 01:16:05 2011 From: coniophora at gmail.com (Jim Worrall) Date: Wed, 14 Sep 2011 17:16:05 -0600 Subject: [FFmpeg-user] ffmpeg installed properly? Noob In-Reply-To: References: <813AE82C-97C3-4BF4-98EA-D0134C80A675@comcast.net> <20110913155015.6d08f4e0@lrcd.com> <51911F64-9A2C-464E-B78A-50BDCE0BF909@comcast.net> Message-ID: <409585E4-17B4-4714-9F3D-4BE5E92DE9A0@gmail.com> I know little to nothing about installing unix programs, but using the detailed instructions here: http://ffmpegmac.net/ with a little help, I was able to do it, and have since updated it. You can download two different mac binaries or click the "How To" link and follow the instructions to download all the files and compile them yourself. Best is to use a script. Sebastian is doing a great service by putting this information up and keeping it updated for the less knowledgeable among us. Jim On Sep 14, 2011, at 3:03 PM, Tom Step wrote: installed LAME, x264, and faac/faad libs, then tried to install ffmpeg again with all libs, then without, still get yasm error (isn't that a linux thing?) shoeis-computer:~ shoei$ cd ~/ffmpeg/ffmpeg-HEAD-d2a847c/ shoeis-computer:ffmpeg-HEAD-d2a847c shoei$ ./configure yasm not found, use --disable-yasm for a crippled build 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. On Sep 14, 2011, at 11:45 AM, Tom Step wrote: > Downloaded Git version to Downloads folder. it created a folder called "ffmpeg-HEAD-d2a847c" which has another "ffmpeg-HEAD-d2a847c" folder within it, which has the configure executable. > > Now in Terminal do I: > cd ~/Downloads/ffmpeg-HEAD-d2a847c/ffmpeg-HEAD-d2a847c > ./configure --enable-libmp3lame --enable-shared > make > sudo make install > > Wil this make a "clean" install with necessary parts? > and > What about other errors with different codecs? FWIW I did run the original configure with--libmp3lame in it.And I do have libfaac but can't make a .aac versions either... > > Thanks for helping a noob out > Tom > > On Sep 13, 2011, at 5:50 PM, Lou wrote: > >> On Tue, 13 Sep 2011 15:21:29 -0600 >> Tom Step wrote: >> >>> Tyring to install ffmpeg on my Mac (OS 10.6.8) to learn CLI. Used >>> MacPorts to do install, it came back with a ton of errors, then did >>> port clean ffmpeg, then: >>> >>> svn checkout svn://svn.ffmpeg.org/ffmpeg/trunk ffmpeg >>> Checked out revision 26402. >> >> FFmpeg uses Git now. The SVN repository is outdated. You should switch >> to the Git repository as the last SVN commit was in January 2011 (or >> simply download a Git snapshot): >> >> http://git.videolan.org/?p=ffmpeg.git;a=snapshot;h=HEAD;sf=tgz >> >>> ./configure --enable-shared --disable-mmx --arch=x86_64 >> >> --disable-mmx was used for PowerPC I think. I don't think you need it >> for the recent architecture. But IANAMacUser. >> >> More importantly, if you want mp3 encoding support, you need to install >> lame and build ffmpeg with --enable-libmp3lame. >> >>> shoeis-computer:Volumes shoei$ ffmpeg -i ~/Desktop/Hunger.m4a mp3 -ab >>> 192k ~/Desktop/Hunger.mp3 >> >> Your command is telling ffmpeg that your output is a file named "mp3". >> Once you recompile ffmpeg with --enable-libmp3lame the remove the >> superfluous "mp3" from your command and it might work, and if not then >> add "-acodec libmp3lame". >> _______________________________________________ >> 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 lautarus at gmail.com Thu Sep 15 05:23:24 2011 From: lautarus at gmail.com (Lautaro) Date: Thu, 15 Sep 2011 00:23:24 -0300 Subject: [FFmpeg-user] vdpau render ? Message-ID: <4E716FAC.5090803@gmail.com> Hi all, I'm trying render a file.ts (h264) to pal-dvd compliant mpeg, the command is: $ffmpeg -i input.ts -target pal-dvd -ab 192k -threads 4 output.mpg It's work fine, but the fps when Im work with ffmpeg is '29' always, the same numbers what it get with a dual-core @ 2.6ghz. Im using a quad-core @ 2.4ghz -actually- , with a nvidia gt210 -with vdpau support, of course-, but the fps=29 , I say, the same speed. Is this ok? Or I m 'Lost in translation' ??? I wrote in console: 'ffmpeg -formats|grep vdpau' but nothing... Thanks in advance & Cheers From hardik.sharma22 at yahoo.com Thu Sep 15 05:55:13 2011 From: hardik.sharma22 at yahoo.com (Hardik Sharma) Date: Wed, 14 Sep 2011 20:55:13 -0700 (PDT) Subject: [FFmpeg-user] Any error inserting help available? Message-ID: <1316058913.13662.YahooMailNeo@web46210.mail.sp1.yahoo.com> Hi guys, I am looking to insert error in h264 files in the form of skipping slices from decoding one or multi at a time. I have one error insert executable for JM, do we have any such kind of help available for ffmpeg/x264. I can delete slice manually with help of hexeditor for h264 as I want to automate it so can't use hex editor for that. Thanks. Hardik Sharma ? ? ? From vas at mpeks.tomsk.su Thu Sep 15 07:48:39 2011 From: vas at mpeks.tomsk.su (Victor Sudakov) Date: Thu, 15 Sep 2011 12:48:39 +0700 Subject: [FFmpeg-user] Using ffmpeg to send multicast Message-ID: <20110915054839.GB68193@admin.sibptus.tomsk.ru> Colleagues, I am trying to stream multicast RTP sound into the network. When I use the following command line on FreeBSD: ffmpeg -i conference.mp3 -acodec copy -f rtp rtp://239.8.8.8:5000 it does work but in a weird way. It spews the whole content of conference.mp3 into the network instantly and exits. You can even actually hear _part_ of the podcast in the multicast receiver, perhaps as much as the receiver's buffer can save (I tested with VLC). What is the correct command line to multicast the file smoothly for everybody to listen to the podcast? I would like to avoid using ffserver if possible since ffmpeg can output RTP natively. TIA for any input. -- Victor Sudakov, VAS4-RIPE, VAS47-RIPN sip:sudakov at sibptus.tomsk.ru From maheshraja8 at gmail.com Thu Sep 15 08:32:41 2011 From: maheshraja8 at gmail.com (mahesh raja) Date: Thu, 15 Sep 2011 12:02:41 +0530 Subject: [FFmpeg-user] Invocation of ffplay API/functions from Aplication code In-Reply-To: References: Message-ID: Hi Roger, Thanks for the info about mplayer. Does it mean that we cannot control ffplay from our application source code by invoking functions defined in ffplay.c with respect to LGPL 2.1 version license.? Any help will be really appreciated. Thanks, Mahesh On Wed, Sep 14, 2011 at 7:41 PM, Roger Pack wrote: > > We have downloaded FFMpeg and built FFplay. The player is executing fine. > > We have seen that the all the functions like stream_open, event_loop are > > defined as static in ffplay.c and are only part of the ffplay executable. > > They are not available as part of any library. If we want to invoke these > > APIs from our application so that the application can control > > when to play and when to pause, is there a way to do ? We have seen that > > this has the license LGPL 2.1, so please suggest if there is a way to > > invoke the APIs from our application. > > cop out answer, but mplayer has a slave mode... > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From wernam at hotmail.com Thu Sep 15 10:25:12 2011 From: wernam at hotmail.com (Wernam Wer) Date: Thu, 15 Sep 2011 10:25:12 +0200 Subject: [FFmpeg-user] Using ffmpeg to send multicast In-Reply-To: <20110915054839.GB68193@admin.sibptus.tomsk.ru> References: <20110915054839.GB68193@admin.sibptus.tomsk.ru> Message-ID: Hi Victor If what you want is send the file little by little, so someone can connect to the stream after you start the stream, you can try this command ffmpeg -i conference.mp3 -acodec copy -f rtp rtp://239.8.8.8:5000 -re With -re you read the input at native frame rate, so you will be sending it as long as the file duration. I hope it helps. Wernam > Date: Thu, 15 Sep 2011 12:48:39 +0700 > From: vas at mpeks.tomsk.su > To: ffmpeg-user at ffmpeg.org > Subject: [FFmpeg-user] Using ffmpeg to send multicast > > Colleagues, > > I am trying to stream multicast RTP sound into the network. When I use > the following command line on FreeBSD: > > ffmpeg -i conference.mp3 -acodec copy -f rtp rtp://239.8.8.8:5000 > > it does work but in a weird way. It spews the whole content of > conference.mp3 into the network instantly and exits. You can even > actually hear _part_ of the podcast in the multicast receiver, perhaps > as much as the receiver's buffer can save (I tested with VLC). > > What is the correct command line to multicast the file smoothly for > everybody to listen to the podcast? I would like to avoid using > ffserver if possible since ffmpeg can output RTP natively. > > TIA for any input. > > -- > Victor Sudakov, VAS4-RIPE, VAS47-RIPN > sip:sudakov at sibptus.tomsk.ru > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From ankuj2004 at gmail.com Thu Sep 15 10:57:36 2011 From: ankuj2004 at gmail.com (Ankuj Gupta) Date: Thu, 15 Sep 2011 14:27:36 +0530 Subject: [FFmpeg-user] What is the use of sws_scale Message-ID: Hi What is sw_scale used for. What I can get is that it is used for format conversion. Are there any more use of it ? Regards Ankuj From vas at mpeks.tomsk.su Thu Sep 15 11:59:01 2011 From: vas at mpeks.tomsk.su (Victor Sudakov) Date: Thu, 15 Sep 2011 16:59:01 +0700 Subject: [FFmpeg-user] Using ffmpeg to send multicast In-Reply-To: References: <20110915054839.GB68193@admin.sibptus.tomsk.ru> Message-ID: <20110915095901.GA76525@admin.sibptus.tomsk.ru> Wernam, thank you very much indeed. With "-re" it works like charm. Wernam Wer wrote: > > Hi Victor > > If what you want is send the file little by little, so someone can connect to the stream after you start the stream, you can try this command > ffmpeg -i conference.mp3 -acodec copy -f rtp rtp://239.8.8.8:5000 -re > With -re you read the input at native frame rate, so you will be sending it as long as the file duration. > I hope it helps. > > Wernam > > > Date: Thu, 15 Sep 2011 12:48:39 +0700 > > From: vas at mpeks.tomsk.su > > To: ffmpeg-user at ffmpeg.org > > Subject: [FFmpeg-user] Using ffmpeg to send multicast > > > > Colleagues, > > > > I am trying to stream multicast RTP sound into the network. When I use > > the following command line on FreeBSD: > > > > ffmpeg -i conference.mp3 -acodec copy -f rtp rtp://239.8.8.8:5000 > > > > it does work but in a weird way. It spews the whole content of > > conference.mp3 into the network instantly and exits. You can even > > actually hear _part_ of the podcast in the multicast receiver, perhaps > > as much as the receiver's buffer can save (I tested with VLC). > > > > What is the correct command line to multicast the file smoothly for > > everybody to listen to the podcast? I would like to avoid using > > ffserver if possible since ffmpeg can output RTP natively. > > > > TIA for any input. > > > > -- > > Victor Sudakov, VAS4-RIPE, VAS47-RIPN > > sip:sudakov at sibptus.tomsk.ru > > _______________________________________________ > > 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 -- Victor Sudakov, VAS4-RIPE, VAS47-RIPN sip:sudakov at sibptus.tomsk.ru From daisyren.jing at gmail.com Thu Sep 15 15:12:47 2011 From: daisyren.jing at gmail.com (daisy.ren) Date: Thu, 15 Sep 2011 21:12:47 +0800 Subject: [FFmpeg-user] consult about ffmpeg can not deal with the downloaded YouTube video Message-ID: Hi, I have downloaded some video from YouTube, and used ffmpeg to transcode them. Some of them can be transcoded , others can not. but they can all be played by video player. when trascode the video, it gives the header info of the video as following: Input #0, flv, from 'G:\test.flv': duration: 00:05:18.1, strat: 0.000000, bitrate: N/A Stream #0.0: Video: 0x0007, 25.00 tb Stream #0.1: Audio: 0x000a, 44100Hz, stereo Output #0, au, to 'G:\test.au' Stream #0.0: Audio: pcm_s16be, 44100Hz, stereo, 1411 kb/s Stream mapping? Stream #0.1 -> #0.0 Unsupported codec for input stream #0.1 the info of the video test.flv, the video and audio format are all ASCII, does this effect the reading of the video? many video player can play the video, means they can read the header info. So i want to know if ffmpeg can fix this or does the ASCII related to the format, such as h.264, .acc and so on? thanks in advance! -- Best Regards Jing From prince_goyal_1987 at yahoo.com Thu Sep 15 15:33:14 2011 From: prince_goyal_1987 at yahoo.com (johnnydepp) Date: Thu, 15 Sep 2011 06:33:14 -0700 (PDT) Subject: [FFmpeg-user] ffmpeg: Setting GOP (i-frame interval) Message-ID: <1316093594614-3815575.post@n4.nabble.com> I am converting one video format (.vob format) to another format (.ts segments having h264 video). Using this command: ffmpeg -ss 0:0:0 -i "D:\sample videos\input9.vob" -ac 2 -f mpegts -acodec libmp3lame -ar 48000 -ab 64000 -s 720x408 -vcodec libx264 -b 1200k -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 5 -trellis 0 -refs 1 -coder 0 -me_range 16 -keyint_min 1 -sc_threshold 40 -i_qfactor 0.71 -bt 1300k -bufsize 524288 -qmin 2 -qmax 48 -aspect 1.7647059 -g 10 -r 30 -async 2 out.ts I want out.ts to have "i-frames at intervals of 10 frames' Problem: Output file contains i-frames at the interval of 126 frames. Am I missing some command line parameter? (For the record, with previous versions of ffmpeg its perfectly putting i-frames at required interval. But with new version only its creating problem) . -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-Setting-GOP-i-frame-interval-tp3815575p3815575.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From prince_goyal_1987 at yahoo.com Thu Sep 15 15:45:04 2011 From: prince_goyal_1987 at yahoo.com (johnnydepp) Date: Thu, 15 Sep 2011 06:45:04 -0700 (PDT) Subject: [FFmpeg-user] changing GOP size In-Reply-To: <4C87CAED.3000505@gmail.com> References: <4C87CAED.3000505@gmail.com> Message-ID: <1316094304613-3815609.post@n4.nabble.com> Hey I am using this command but still its not putting i-frames at required intervals ffmpeg -ss 0:0:0 -i "D:\sample videos\input9.vob" -ac 2 -f mpegts -acodec libmp3lame -ar 48000 -ab 64000 -s 720x408 -vcodec libx264 -b 1200k -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 5 -trellis 0 -refs 1 -coder 0 -me_range 16 -keyint_min 1 -sc_threshold 40 -i_qfactor 0.71 -bt 1300k -bufsize 524288 -qmin 2 -qmax 48 -aspect 1.7647059 -g 5 -r 30 -async 2 out.ts In out.ts i-frame interval is 96. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/changing-GOP-size-tp2531576p3815609.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From andycivil at gmail.com Thu Sep 15 15:58:19 2011 From: andycivil at gmail.com (Andy Civil) Date: Thu, 15 Sep 2011 09:58:19 -0400 Subject: [FFmpeg-user] ffmpeg: Setting GOP (i-frame interval) In-Reply-To: <1316093594614-3815575.post@n4.nabble.com> References: <1316093594614-3815575.post@n4.nabble.com> Message-ID: <4E72047B.2040607@gmail.com> It could be this? http://ffmpeg.org/trac/ffmpeg/ticket/157 On 2011-09-15 9:33 AM, johnnydepp wrote: > I am converting one video format (.vob format) to another format (.ts > segments having h264 video). > > Using this command: > > ffmpeg -ss 0:0:0 -i "D:\sample videos\input9.vob" -ac 2 -f mpegts > -acodec libmp3lame -ar 48000 -ab 64000 -s 720x408 -vcodec libx264 -b 1200k > -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 5 > -trellis 0 -refs 1 -coder 0 -me_range 16 -keyint_min 1 -sc_threshold 40 > -i_qfactor 0.71 -bt 1300k -bufsize 524288 -qmin 2 -qmax 48 -aspect 1.7647059 > -g 10 -r 30 -async 2 out.ts > > > I want out.ts to have "i-frames at intervals of 10 frames' > > Problem: Output file contains i-frames at the interval of 126 frames. > > Am I missing some command line parameter? > > (For the record, with previous versions of ffmpeg its perfectly putting > i-frames at required interval. But with new version only its creating > problem) . > > > > > -- > View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-Setting-GOP-i-frame-interval-tp3815575p3815575.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 rogerdpack2 at gmail.com Thu Sep 15 16:10:29 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Thu, 15 Sep 2011 08:10:29 -0600 Subject: [FFmpeg-user] What is the use of sws_scale In-Reply-To: References: Message-ID: > What is sw_scale used for. What I can get is that it is used for > format conversion. Are there any more use of it ? You can select a more sophisticated resizing algorithm if you want higher quality (ex: lanczos takes more cpu, higher quality result). From rogerdpack2 at gmail.com Thu Sep 15 16:11:38 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Thu, 15 Sep 2011 08:11:38 -0600 Subject: [FFmpeg-user] Invocation of ffplay API/functions from Aplication code In-Reply-To: References: Message-ID: > Thanks for the info about mplayer. > Does it mean that we cannot control ffplay from our application source code > by invoking functions defined in ffplay.c with respect to LGPL 2.1 version > license.? > Any help will be really appreciated. I'm sure you can. Dunno if anybody else has done that before.. From pango99 at tom.com Thu Sep 15 05:43:40 2011 From: pango99 at tom.com (pango) Date: Wed, 14 Sep 2011 20:43:40 -0700 (PDT) Subject: [FFmpeg-user] VLC can not play the mpeg-ts file muxed by ffmpeg Message-ID: <1316058220270-3814625.post@n4.nabble.com> I use below command line to mux mpv & ac3 es file to a mpeg-ts file: ffmpeg -genpts 1 -i h:\Test.mpv -vcodec copy - i h:\Test.ac3 -acodec copy -f mpegts -y h:\Output.ts but the generated file can not be played by vlc media player,vlc report below errors: main error: ES_OUT_SET_(GROUP_)PCR is called too late (jitter of 5203 ms ignored) main error: ES_OUT_SET_(GROUP_)PCR is called too late (jitter of 5320 ms ignored) main error: ES_OUT_SET_(GROUP_)PCR is called too late (jitter of 5441 ms ignored) main error: ES_OUT_SET_(GROUP_)PCR is called too late (jitter of 5563 ms ignored) main error: ES_OUT_SET_(GROUP_)PCR is called too late (jitter of 5683 ms ignored) main error: ES_OUT_SET_(GROUP_)PCR is called too late (jitter of 5801 ms ignored) main error: ES_OUT_SET_(GROUP_)PCR is called too late (jitter of 5922 ms ignored) main error: ES_OUT_SET_(GROUP_)PCR is called too late (jitter of 6042 ms ignored) so,what's wrong about the command line I type?how to solve? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/VLC-can-not-play-the-mpeg-ts-file-muxed-by-ffmpeg-tp3814625p3814625.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From michael.rampe at gmail.com Fri Sep 16 01:06:21 2011 From: michael.rampe at gmail.com (FFmichael) Date: Thu, 15 Sep 2011 16:06:21 -0700 (PDT) Subject: [FFmpeg-user] Centre justify text with -drawtext filter Message-ID: <1316127981905-3816998.post@n4.nabble.com> Hi All, I have been experimenting with the drawtext filter and am looking at a way to centre justify text. For example: I have a 640x480 size video I want the set the centre of the text to 320x400 so it is centred on the bottom of the screen.... Is this currently possible or something that is being looked at? Have I missed something in the docs? Regards, FFmichael -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Centre-justify-text-with-drawtext-filter-tp3816998p3816998.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From stefasab at gmail.com Fri Sep 16 01:39:03 2011 From: stefasab at gmail.com (Stefano Sabatini) Date: Fri, 16 Sep 2011 01:39:03 +0200 Subject: [FFmpeg-user] Centre justify text with -drawtext filter In-Reply-To: <1316127981905-3816998.post@n4.nabble.com> References: <1316127981905-3816998.post@n4.nabble.com> Message-ID: <20110915233903.GD22539@geppetto> On date Thursday 2011-09-15 16:06:21 -0700, FFmichael encoded: > Hi All, > > I have been experimenting with the drawtext filter and am looking at a way > to centre justify text. > > For example: > I have a 640x480 size video > I want the set the centre of the text to 320x400 so it is centred on the > bottom of the screen.... > > Is this currently possible or something that is being looked at? Have I > missed something in the docs? Not yet supported: https://ffmpeg.org/trac/ffmpeg/ticket/378 -- ffmpeg-user random tip #1 The -b (bitrate) and -ab (audio bitrate) ffmpeg options take as argument a bitrate expressed in bits per second, and not in Kbits per second as in previous versions. Change your command lines accordingly. From michael.rampe at gmail.com Fri Sep 16 01:50:16 2011 From: michael.rampe at gmail.com (FFmichael) Date: Thu, 15 Sep 2011 16:50:16 -0700 (PDT) Subject: [FFmpeg-user] Centre justify text with -drawtext filter In-Reply-To: <20110915233903.GD22539@geppetto> References: <1316127981905-3816998.post@n4.nabble.com> <20110915233903.GD22539@geppetto> Message-ID: <1316130616942-3817056.post@n4.nabble.com> On date Thursday 2011-09-15 16:06:21 -0700, FFmichael encoded: > Hi All, > > I have been experimenting with the drawtext filter and am looking at a way > to centre justify text. > > For example: > I have a 640x480 size video > I want the set the centre of the text to 320x400 so it is centred on the > bottom of the screen.... > > Is this currently possible or something that is being looked at? Have I > missed something in the docs? Not yet supported: https://ffmpeg.org/trac/ffmpeg/ticket/378 -- Thanks Stefano. Looking forward to this feature.... FFmicahel -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Centre-justify-text-with-drawtext-filter-tp3816998p3817056.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From dashing.meng at gmail.com Fri Sep 16 02:56:26 2011 From: dashing.meng at gmail.com (littlebat) Date: Fri, 16 Sep 2011 08:56:26 +0800 Subject: [FFmpeg-user] consult about ffmpeg can not deal with the downloaded YouTube video In-Reply-To: References: Message-ID: <20110916085626.4b5b0489.dashing.meng@gmail.com> On Thu, 15 Sep 2011 21:12:47 +0800 "daisy.ren" wrote: > Unsupported codec for input stream #0.1 It seems can't recognise your input audio, try to identify your audio codec using your media player or other tool(mediainfo), and re-compile ffmpeg with the support of this audio codec. From vas at mpeks.tomsk.su Fri Sep 16 04:48:25 2011 From: vas at mpeks.tomsk.su (Victor Sudakov) Date: Fri, 16 Sep 2011 09:48:25 +0700 Subject: [FFmpeg-user] Recording sound on FreeBSD 8 Message-ID: <20110916024825.GA92446@admin.sibptus.tomsk.ru> Colleagues, My next goal is to multicast live sound. All my attemps to record sound from a USB audio interface have resulted in a severely distorted growl instead of normal voice. I have tried various parameters around ffmpeg -f oss -i /dev/dsp1 -acodec libmp3lame -re -f rtp rtp://239.1.1.1:5000 or even simply ffmpeg -f oss -i /dev/dsp1 out.wav I suppose I need to adjust sampling rate or something like this. What is the way to determine the correct parameters? Is there a way to analyze the audio interface input? "cat /dev/sndstat" does not give anything really useful. The audio interface is not to blame because I use it all the time with linphone for SIP calls. TIA for any ideas. -- Victor Sudakov, VAS4-RIPE, VAS47-RIPN sip:sudakov at sibptus.tomsk.ru From vas at mpeks.tomsk.su Fri Sep 16 05:09:00 2011 From: vas at mpeks.tomsk.su (Victor Sudakov) Date: Fri, 16 Sep 2011 10:09:00 +0700 Subject: [FFmpeg-user] Using ffmpeg to send multicast In-Reply-To: References: <20110915054839.GB68193@admin.sibptus.tomsk.ru> Message-ID: <20110916030900.GB92792@admin.sibptus.tomsk.ru> Wernam, This ffmpeg-generated stream is played very well by VLC (both Windows and Linux), but mplayer/FreeBSD somehow cannot play rtp://239.8.8.8:5000 with the "Stream not seekable!" error. Is it a limitation of mplayer or is there something wrong with the rtp stream I am sending? In Wireshark, it looks like normal, good RTP. Wernam Wer wrote: > > Hi Victor > > If what you want is send the file little by little, so someone can connect to the stream after you start the stream, you can try this command > ffmpeg -i conference.mp3 -acodec copy -f rtp rtp://239.8.8.8:5000 -re > With -re you read the input at native frame rate, so you will be sending it as long as the file duration. > I hope it helps. > > Wernam > > > Date: Thu, 15 Sep 2011 12:48:39 +0700 > > From: vas at mpeks.tomsk.su > > To: ffmpeg-user at ffmpeg.org > > Subject: [FFmpeg-user] Using ffmpeg to send multicast > > > > Colleagues, > > > > I am trying to stream multicast RTP sound into the network. When I use > > the following command line on FreeBSD: > > > > ffmpeg -i conference.mp3 -acodec copy -f rtp rtp://239.8.8.8:5000 > > > > it does work but in a weird way. It spews the whole content of > > conference.mp3 into the network instantly and exits. You can even > > actually hear _part_ of the podcast in the multicast receiver, perhaps > > as much as the receiver's buffer can save (I tested with VLC). > > > > What is the correct command line to multicast the file smoothly for > > everybody to listen to the podcast? I would like to avoid using > > ffserver if possible since ffmpeg can output RTP natively. > > > > TIA for any input. > > > > -- > > Victor Sudakov, VAS4-RIPE, VAS47-RIPN > > sip:sudakov at sibptus.tomsk.ru > > _______________________________________________ > > 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 -- Victor Sudakov, VAS4-RIPE, VAS47-RIPN sip:sudakov at sibptus.tomsk.ru From prince_goyal_1987 at yahoo.com Fri Sep 16 12:23:14 2011 From: prince_goyal_1987 at yahoo.com (johnnydepp) Date: Fri, 16 Sep 2011 03:23:14 -0700 (PDT) Subject: [FFmpeg-user] (SOLUTION ) ffmpeg: Setting GOP (i-frame interval) In-Reply-To: <4E72047B.2040607@gmail.com> References: <1316093594614-3815575.post@n4.nabble.com> <4E72047B.2040607@gmail.com> Message-ID: <1316168594449-3817755.post@n4.nabble.com> I Found the solution. ffmpeg have changed cmd input format for x264. For x264 you have to do like this: ffmpeg -i input.vob -f mpegts -vcodec libx264 -x264opts keyint=5:min-keyint=1 output.ts Note: all x264 opts go through "x264opts". So all the x264 options you have to be given from the "x264opts" methods. Otherwise they can be overridden by default values. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-Setting-GOP-i-frame-interval-tp3815575p3817755.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From maheshraja8 at gmail.com Fri Sep 16 12:23:17 2011 From: maheshraja8 at gmail.com (mahesh raja) Date: Fri, 16 Sep 2011 15:53:17 +0530 Subject: [FFmpeg-user] Invocation of ffplay API/functions from Aplication code In-Reply-To: References: Message-ID: Hi Roger, Thanks for your valuable info and reply. Thanks Mahesh On Thu, Sep 15, 2011 at 7:41 PM, Roger Pack wrote: > > Thanks for the info about mplayer. > > Does it mean that we cannot control ffplay from our application source > code > > by invoking functions defined in ffplay.c with respect to LGPL 2.1 > version > > license.? > > Any help will be really appreciated. > > I'm sure you can. Dunno if anybody else has done that before.. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From wernam at hotmail.com Fri Sep 16 12:55:06 2011 From: wernam at hotmail.com (Wernam Wer) Date: Fri, 16 Sep 2011 12:55:06 +0200 Subject: [FFmpeg-user] Using ffmpeg to send multicast In-Reply-To: <20110916030900.GB92792@admin.sibptus.tomsk.ru> References: <20110915054839.GB68193@admin.sibptus.tomsk.ru>, , <20110916030900.GB92792@admin.sibptus.tomsk.ru> Message-ID: Hi Victor How are you using mplayer?What command? I'm currently using it to read a stream and it work fine, this is what I do: *Execute ffmpeg, the you get something like this: .... v=0 o=- 0 0 IN IP4 127.0.0.1 s=No Name c=IN IP4 224.52.52.22 t=0 0 a=tool:libavformat 53.6.0 m=video 7004 RTP/AVP 96 a=rtpmap:96 MP4V-ES/90000 a=fmtp:96 profile-level-id=1; config=000001B003000001B50900000100000001200086C400668582120A31 .... *Save only those lines into a file and name it name.sdp *Then I execute: mplayer sdp://name.sdp If you executed ffmpeg again with the same file and parameters you always will obtain the same sdp file, so you only have to generate once. This should work perfect, I hope it helps. Wernam > Date: Fri, 16 Sep 2011 10:09:00 +0700 > From: vas at mpeks.tomsk.su > To: ffmpeg-user at ffmpeg.org > Subject: Re: [FFmpeg-user] Using ffmpeg to send multicast > > Wernam, > > This ffmpeg-generated stream is played very well by VLC (both Windows > and Linux), but mplayer/FreeBSD somehow cannot play > rtp://239.8.8.8:5000 with the "Stream not seekable!" error. Is it a > limitation of mplayer or is there something wrong with the rtp stream > I am sending? > > In Wireshark, it looks like normal, good RTP. > > Wernam Wer wrote: > > > > Hi Victor > > > > If what you want is send the file little by little, so someone can connect to the stream after you start the stream, you can try this command > > ffmpeg -i conference.mp3 -acodec copy -f rtp rtp://239.8.8.8:5000 -re > > With -re you read the input at native frame rate, so you will be sending it as long as the file duration. > > I hope it helps. > > > > Wernam > > > > > Date: Thu, 15 Sep 2011 12:48:39 +0700 > > > From: vas at mpeks.tomsk.su > > > To: ffmpeg-user at ffmpeg.org > > > Subject: [FFmpeg-user] Using ffmpeg to send multicast > > > > > > Colleagues, > > > > > > I am trying to stream multicast RTP sound into the network. When I use > > > the following command line on FreeBSD: > > > > > > ffmpeg -i conference.mp3 -acodec copy -f rtp rtp://239.8.8.8:5000 > > > > > > it does work but in a weird way. It spews the whole content of > > > conference.mp3 into the network instantly and exits. You can even > > > actually hear _part_ of the podcast in the multicast receiver, perhaps > > > as much as the receiver's buffer can save (I tested with VLC). > > > > > > What is the correct command line to multicast the file smoothly for > > > everybody to listen to the podcast? I would like to avoid using > > > ffserver if possible since ffmpeg can output RTP natively. > > > > > > TIA for any input. > > > > > > -- > > > Victor Sudakov, VAS4-RIPE, VAS47-RIPN > > > sip:sudakov at sibptus.tomsk.ru > > > _______________________________________________ > > > 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 > > -- > Victor Sudakov, VAS4-RIPE, VAS47-RIPN > sip:sudakov at sibptus.tomsk.ru > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From maheshraja8 at gmail.com Fri Sep 16 13:36:44 2011 From: maheshraja8 at gmail.com (mahesh raja) Date: Fri, 16 Sep 2011 17:06:44 +0530 Subject: [FFmpeg-user] Streaming using ffserver problems Message-ID: hi, ffserver -f doc/ffserver.conf & is working. but ffmpeg -i /root/Desktop/niceday.asf http://10.130.100.150:8090/feed1.ffm giving these ERRORS Stream #0.1(eng): Video: wmv3 (Main), yuv420p, 320x240, 250 kb/s, 30 tbr, 1k tbn, 1k tbc Fri Sep 16 16:56:53 2011 File '/feed1.ffm' not found Fri Sep 16 16:56:53 2011 10.130.100.150 - - [GET] "/feed1.ffm HTTP/1.1" 404 149 [http @ 0x948a0e0] HTTP error 404 Not Found http://10.130.100.150:8090/feed1.ffm: Input/output error help me to be out of these bugs Thanks Mahesh From hardik.sharma22 at yahoo.com Fri Sep 16 21:59:43 2011 From: hardik.sharma22 at yahoo.com (Hardik Sharma) Date: Fri, 16 Sep 2011 12:59:43 -0700 (PDT) Subject: [FFmpeg-user] Any error inserting help available? In-Reply-To: <1316131685.9987.YahooMailNeo@web46212.mail.sp1.yahoo.com> References: <1316058913.13662.YahooMailNeo@web46210.mail.sp1.yahoo.com> <1316131685.9987.YahooMailNeo@web46212.mail.sp1.yahoo.com> Message-ID: <1316203183.81495.YahooMailNeo@web46215.mail.sp1.yahoo.com> So now I am trying to use my parser code which is going through the bytes of h264 file by using a buffer to find out NAL start code. I want to ask that is it possible from here to skip one particular NAL unit/slice every time and rest I can put in new error 264 which is going to have every other data leaving that one particular slice. In this way I am generating error file by dropping one slice at a time. Let me know if I am missing any crucial point. Thanks. ? ? ? ________________________________ From: Hardik Sharma To: ffmpeg dev Sent: Thursday, 15 September 2011 5:08 PM Subject: Re: Any error inserting help available? I already have boundaries for slices from h264 file. Can I directly delete the?particular slices according to my need somehow from h264 file?? ________________________________ From: Hardik Sharma To: "x264-devel at videolan.org" ; ffmpeg dev ; ffmpeg Sent: Wednesday, 14 September 2011 8:55 PM Subject: Any error inserting help available? Hi guys, I am looking to insert error in h264 files in the form of skipping slices from decoding one or multi at a time. I have one error insert executable for JM, do we have any such kind of help available for ffmpeg/x264. I can delete slice manually with help of hexeditor for h264 as I want to automate it so can't use hex editor for that. Thanks. Hardik Sharma ? ? ? From micahsutton at gmail.com Sun Sep 18 01:50:44 2011 From: micahsutton at gmail.com (Micah Sutton) Date: Sat, 17 Sep 2011 18:50:44 -0500 Subject: [FFmpeg-user] Unable to compile ffmpeg with libx264 support under OS X Lion Message-ID: Trying to get ffmpeg up and running on a brand new clean installation of Lion. I've installed developer tools 4.1.1, downloaded latest libx264 and ffmpeg. Attempting to compile ffmpeg with the following: ./configure --enable-gpl --enable-libx264 results in: LD ffmpeg_g Undefined symbols for architecture x86_64: "_MM_FIX_0_707106781", referenced from: _filter in libavfilter.a(vf_fspp.o) "_MM_FIX_0_541196100", referenced from: _filter in libavfilter.a(vf_fspp.o) ld: symbol(s) not found for architecture x86_64 collect2: ld returned 1 exit status make: *** [ffmpeg_g] Error 1 From rodney.baker at iinet.net.au Sun Sep 18 02:28:15 2011 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Sun, 18 Sep 2011 09:58:15 +0930 Subject: [FFmpeg-user] Unable to compile ffmpeg with libx264 support under OS X Lion In-Reply-To: References: Message-ID: <201109180958.15869.rodney.baker@iinet.net.au> On Sun, 18 Sep 2011 09:20:44 Micah Sutton wrote: > Trying to get ffmpeg up and running on a brand new clean installation of > Lion. I've installed developer tools 4.1.1, downloaded latest libx264 and > ffmpeg. > > Attempting to compile ffmpeg with the following: ./configure --enable-gpl > --enable-libx264 > > results in: > > LD ffmpeg_g > Undefined symbols for architecture x86_64: > "_MM_FIX_0_707106781", referenced from: > _filter in libavfilter.a(vf_fspp.o) > "_MM_FIX_0_541196100", referenced from: > _filter in libavfilter.a(vf_fspp.o) > ld: symbol(s) not found for architecture x86_64 > collect2: ld returned 1 exit status > make: *** [ffmpeg_g] Error 1 > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user How did you install libx264? From source? If not, you will need to install the relevant devel package as well to make sure that you have the headers for ffmpeg to compile against. -- =================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au =================================================== From micahsutton at gmail.com Sun Sep 18 02:42:46 2011 From: micahsutton at gmail.com (Micah Sutton) Date: Sat, 17 Sep 2011 19:42:46 -0500 Subject: [FFmpeg-user] Unable to compile ffmpeg with libx264 support under OS X Lion In-Reply-To: <201109180958.15869.rodney.baker@iinet.net.au> References: <201109180958.15869.rodney.baker@iinet.net.au> Message-ID: > How did you install libx264? From source? If not, you will need to install > the relevant devel package as well to make sure that you have the headers > for > ffmpeg to compile against. > > I did install libx264 from source. It was suggested on the irc channel to use the -cc=clang flag, this seems to work. Why doesn't gcc?! From hno2005 at gmail.com Sun Sep 18 13:58:44 2011 From: hno2005 at gmail.com (hno2005) Date: Sun, 18 Sep 2011 15:28:44 +0330 Subject: [FFmpeg-user] Problem with converting 3gp to flv Message-ID: <000301cc75fa$5300b1d0$f9021570$@com> HI, I have wrote a code which convert video files to flv but the problem is this that it converts almost all of the files type to flv but it can?t convert 3gp to flv. This is the command that I use: exec( "$ffmpegpath -i uploads/$video_name -ar 22050 -ab 32 -f flv -s 320?240 uploads/a.flv", $ret ); Actually I have surf most of the website to find the reason but I can?t find out the reason. Please help me . Thanks every one From dashing.meng at gmail.com Sun Sep 18 16:52:46 2011 From: dashing.meng at gmail.com (littlebat) Date: Sun, 18 Sep 2011 22:52:46 +0800 Subject: [FFmpeg-user] Problem with converting 3gp to flv In-Reply-To: <000301cc75fa$5300b1d0$f9021570$@com> References: <000301cc75fa$5300b1d0$f9021570$@com> Message-ID: <20110918225246.c6741050.dashing.meng@gmail.com> On Sun, 18 Sep 2011 15:28:44 +0330 "hno2005" wrote: > HI, > > I have wrote a code which convert video files to flv but the problem > is this that it converts almost all of the files type to flv but it > can?t convert 3gp to flv. > > This is the command that I use: > > exec( "$ffmpegpath -i uploads/$video_name -ar 22050 -ab 32 -f flv -s > 320?240 uploads/a.flv", $ret ); You should post the error message of converting. Try "-ab 32k" From lautarus at gmail.com Sun Sep 18 18:38:07 2011 From: lautarus at gmail.com (Lautaro) Date: Sun, 18 Sep 2011 13:38:07 -0300 Subject: [FFmpeg-user] Problem with a lot a images to .avi In-Reply-To: <20110918225246.c6741050.dashing.meng@gmail.com> References: <000301cc75fa$5300b1d0$f9021570$@com> <20110918225246.c6741050.dashing.meng@gmail.com> Message-ID: <4E761E6F.5050708@gmail.com> Hi list, I have this command to create a movie.avi from a lot (40) images.jpg $ffmpeg -r 0.25 -f image2 -i "imagen%d.jpg" -target pal-dvd -threads 4 -y alerces-01.mpg The video resultate in an error, but I cannot say how... This is a link from vid: http://www.mediafire.com/?kklf48863bkr3lz (38mb) I hope any help, what's wrong with my command? The video have 'vibrations' for talk any way. Cheers. Lautaro From hno2005 at gmail.com Sun Sep 18 19:53:22 2011 From: hno2005 at gmail.com (hno2005) Date: Sun, 18 Sep 2011 21:23:22 +0330 Subject: [FFmpeg-user] Problem with converting 3gp to flv In-Reply-To: <20110918225246.c6741050.dashing.meng@gmail.com> References: <000301cc75fa$5300b1d0$f9021570$@com> <20110918225246.c6741050.dashing.meng@gmail.com> Message-ID: <000301cc762b$de2f12c0$9a8d3840$@com> Thanks for your reply I tried that but It didn't work again. Actually , I'm new to ffmpeg .How can I see the errors ? Thanks -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of littlebat Sent: Sunday, September 18, 2011 6:23 PM To: FFmpeg user questions and RTFMs Subject: Re: [FFmpeg-user] Problem with converting 3gp to flv On Sun, 18 Sep 2011 15:28:44 +0330 "hno2005" wrote: > HI, > > I have wrote a code which convert video files to flv but the problem > is this that it converts almost all of the files type to flv but it > can?t convert 3gp to flv. > > This is the command that I use: > > exec( "$ffmpegpath -i uploads/$video_name -ar 22050 -ab 32 -f flv -s > 320?240 uploads/a.flv", $ret ); You should post the error message of converting. Try "-ab 32k" _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From marcosfrm at gmail.com Sun Sep 18 18:54:36 2011 From: marcosfrm at gmail.com (Marcos Felipe Rasia de Mello) Date: Sun, 18 Sep 2011 13:54:36 -0300 Subject: [FFmpeg-user] DirectShow - Unrecognized option 'list_options' Message-ID: Hi, I'm playing with the new DirectShow input device support and when I query the supported options (following the man page example), ffmpeg prints this ffmpeg -list_options true -f dshow -i video="Lenovo EasyCamera" ffmpeg version N-32611-gd55b06b, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 15 2011 00:23:27 with gcc 4.6.1 configuration: --enable-gpl --enable-version3 --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 16. 0 / 51. 16. 0 libavcodec 53. 14. 0 / 53. 14. 0 libavformat 53. 12. 0 / 53. 12. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 40. 0 / 2. 40. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 Unrecognized option 'list_options' Failed to set value 'true' for option 'list_options' From butting.mp at gmail.com Mon Sep 19 02:11:31 2011 From: butting.mp at gmail.com (Bryce Utting) Date: Mon, 19 Sep 2011 12:11:31 +1200 Subject: [FFmpeg-user] Problem with converting 3gp to flv In-Reply-To: <000301cc75fa$5300b1d0$f9021570$@com> References: <000301cc75fa$5300b1d0$f9021570$@com> Message-ID: <4E7688B3.5030608@gmail.com> hno2005 wrote: > HI, > > I have wrote a code which convert video files to flv but the problem is this > that it converts almost all of the files type to flv but it can?t convert > 3gp to flv. > > This is the command that I use: > > exec( "$ffmpegpath -i uploads/$video_name -ar 22050 -ab 32 -f flv -s 320?240 > uploads/a.flv", $ret ); What errors does ffmpeg report when you run that command from a shell with the appropriate $video_name value? Does $video_name have any characters in which might break from not being quoted? (spaces, quote marks, etc) butting From vas at mpeks.tomsk.su Mon Sep 19 07:40:31 2011 From: vas at mpeks.tomsk.su (Victor Sudakov) Date: Mon, 19 Sep 2011 12:40:31 +0700 Subject: [FFmpeg-user] Using ffmpeg to send multicast In-Reply-To: References: <20110916030900.GB92792@admin.sibptus.tomsk.ru> Message-ID: <20110919054031.GA50274@admin.sibptus.tomsk.ru> Wernam Wer wrote: > > How are you using mplayer?What command? I was trying "mplayer rtp://239.8.8.8:5000" just like in VLC, but now I see I was wrong. > I'm currently using it to read a stream and it work fine, this is what I do: > *Execute ffmpeg, the you get something like this: [dd] > .... > *Save only those lines into a file and name it name.sdp > *Then I execute: mplayer sdp://name.sdp > If you executed ffmpeg again with the same file and parameters you > always will obtain the same sdp file, so you only have to generate > once. Thank you, this works. I would like to know if it is possible to pass all the parameters in the mplayer command line, without creating an .sdp file. But our world is not perfect :) -- Victor Sudakov, VAS4-RIPE, VAS47-RIPN sip:sudakov at sibptus.tomsk.ru From rogerdpack2 at gmail.com Mon Sep 19 07:57:37 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Sun, 18 Sep 2011 23:57:37 -0600 Subject: [FFmpeg-user] DirectShow - Unrecognized option 'list_options' In-Reply-To: References: Message-ID: > I'm playing with the new DirectShow input device support and when I > query the supported options (following the man page example), ffmpeg > prints this > > ffmpeg -list_options true -f dshow -i video="Lenovo EasyCamera" > > ffmpeg version N-32611-gd55b06b, Copyright (c) 2000-2011 the FFmpeg developers > ?built on Sep 15 2011 00:23:27 with gcc 4.6.1 > ... > Unrecognized option 'list_options' > Failed to set value 'true' for option 'list_options' The commits to support that were only added recently (sep. 16 or so) so you'll either have to wait for a new build or build it yourself locally from git. Or I "think" I published a build that included those here: http://groups.google.com/group/roger-projects/browse_thread/thread/f4bf40f1e1d875dd http://groups.google.com/group/roger-projects/attach/52e1201b1b65017d/ffmpeg-dshow-params-4.7z?part=2 that Ramiro had sent me, so it is a "one off" build for prototyping the dshow stuff but might give you a head start until another build is ready. Cheers! -r From forum at numerizik.com Mon Sep 19 09:52:54 2011 From: forum at numerizik.com (numerizik) Date: Mon, 19 Sep 2011 00:52:54 -0700 (PDT) Subject: [FFmpeg-user] =?utf-8?q?Add_little_video_IN_other_vid=C3=A9o?= Message-ID: <1316418773989-3823216.post@n4.nabble.com> Hello For one TV local project, i need to do this : I have big Interview vid?o (2 minutes) and some little videos (5 secondes). I need to add little video on the big vid?o using audio of big vid?o. I start to do this : - Extract audio from big video - Extract point a,b,c,d,... - Cut each part - merge each part with audio file http://ffmpeg-users.933282.n4.nabble.com/file/n3823216/timeline.png It's complex and i want to know if you have better solution for this ??? thanks for your help Guillaume -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Add-little-video-IN-other-video-tp3823216p3823216.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From hno2005 at gmail.com Mon Sep 19 10:59:33 2011 From: hno2005 at gmail.com (hno2005) Date: Mon, 19 Sep 2011 12:29:33 +0330 Subject: [FFmpeg-user] Problem with converting 3gp to flv In-Reply-To: <4E7688B3.5030608@gmail.com> References: <000301cc75fa$5300b1d0$f9021570$@com> <4E7688B3.5030608@gmail.com> Message-ID: <002701cc76aa$77a6cdc0$66f46940$@com> Ffmpeg convert the file and make flv file but with 0 filezie . I execute the command by shell_exec but It still return nothing. If I convert any file except 3gp it's work fine, What can be the problem? Thanks a lot -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Bryce Utting Sent: Monday, September 19, 2011 3:42 AM To: ffmpeg-user at ffmpeg.org Subject: Re: [FFmpeg-user] Problem with converting 3gp to flv hno2005 wrote: > HI, > > I have wrote a code which convert video files to flv but the problem is this > that it converts almost all of the files type to flv but it can?t convert > 3gp to flv. > > This is the command that I use: > > exec( "$ffmpegpath -i uploads/$video_name -ar 22050 -ab 32 -f flv -s 320?240 > uploads/a.flv", $ret ); What errors does ffmpeg report when you run that command from a shell with the appropriate $video_name value? Does $video_name have any characters in which might break from not being quoted? (spaces, quote marks, etc) butting _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From stefasab at gmail.com Mon Sep 19 10:51:31 2011 From: stefasab at gmail.com (Stefano Sabatini) Date: Mon, 19 Sep 2011 10:51:31 +0200 Subject: [FFmpeg-user] DirectShow - Unrecognized option 'list_options' In-Reply-To: References: Message-ID: <20110919085131.GA13022@geppetto> On date Sunday 2011-09-18 13:54:36 -0300, Marcos Felipe Rasia de Mello encoded: > Hi, > > I'm playing with the new DirectShow input device support and when I > query the supported options (following the man page example), ffmpeg > prints this > > ffmpeg -list_options true -f dshow -i video="Lenovo EasyCamera" > > ffmpeg version N-32611-gd55b06b, Copyright (c) 2000-2011 the FFmpeg developers > built on Sep 15 2011 00:23:27 with gcc 4.6.1 > configuration: --enable-gpl --enable-version3 > --enable-runtime-cpudetect --enable-avisynth --enable-bzlib > --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-libfreetype --enable-libgsm --enable-libmp3lame > --enable-libopenjpeg --enable-librtmp --enable-libschroedinger > --enable-libspeex --enable-libtheora --enable-libvo-aacenc > --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx > --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib > libavutil 51. 16. 0 / 51. 16. 0 > libavcodec 53. 14. 0 / 53. 14. 0 > libavformat 53. 12. 0 / 53. 12. 0 > libavdevice 53. 3. 0 / 53. 3. 0 You need libavdevice >= 53.3.2, update your FFmpeg. From mimeini at gmail.com Mon Sep 19 16:42:50 2011 From: mimeini at gmail.com (mikkel meinike) Date: Mon, 19 Sep 2011 16:42:50 +0200 Subject: [FFmpeg-user] =?iso-8859-1?q?Add_little_video_IN_other_vid=E9o?= In-Reply-To: <1316418773989-3823216.post@n4.nabble.com> References: <1316418773989-3823216.post@n4.nabble.com> Message-ID: That sounds like the way but if audio an video is going to stay in sync you would have to. Split audio and video in to separate files. cut out of the audio exactly the same measure at as the video you want to insert. Than ad the pieces together. That's how I would do this. Mikkel On Mon, Sep 19, 2011 at 9:52 AM, numerizik wrote: > Hello > For one TV local project, i need to do this : > > I have big Interview vid?o (2 minutes) and some little videos (5 secondes). > I need to add little video on the big vid?o using audio of big vid?o. > I start to do this : > - Extract audio from big video > - Extract point a,b,c,d,... > - Cut each part > - merge each part with audio file > http://ffmpeg-users.933282.n4.nabble.com/file/n3823216/timeline.png > > It's complex and i want to know if you have better solution for this ??? > > thanks for your help > Guillaume > > -- > View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Add-little-video-IN-other-video-tp3823216p3823216.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 mimeini at gmail.com Mon Sep 19 16:45:22 2011 From: mimeini at gmail.com (mikkel meinike) Date: Mon, 19 Sep 2011 16:45:22 +0200 Subject: [FFmpeg-user] =?iso-8859-1?q?Add_little_video_IN_other_vid=E9o?= In-Reply-To: References: <1316418773989-3823216.post@n4.nabble.com> Message-ID: Sorry mistake one more time. That sounds like the way but if audio an video is going to stay in sync you would have to. Split audio and video in to separate files. cut out of the long video exactly the same time measure at as the duration of the video you want to insert. Than ad the pieces together. And then add the audio to the video. That's how I would do this. On Mon, Sep 19, 2011 at 4:42 PM, mikkel meinike wrote: > That sounds like the way but if audio an video is going to stay in > sync you would have to. Split audio and video in to separate files. > cut out of the audio exactly the same measure at as the video you want > to insert. Than ad the pieces together. That's how I would do this. > > Mikkel > > On Mon, Sep 19, 2011 at 9:52 AM, numerizik wrote: >> Hello >> For one TV local project, i need to do this : >> >> I have big Interview vid?o (2 minutes) and some little videos (5 secondes). >> I need to add little video on the big vid?o using audio of big vid?o. >> I start to do this : >> - Extract audio from big video >> - Extract point a,b,c,d,... >> - Cut each part >> - merge each part with audio file >> http://ffmpeg-users.933282.n4.nabble.com/file/n3823216/timeline.png >> >> It's complex and i want to know if you have better solution for this ??? >> >> thanks for your help >> Guillaume >> >> -- >> View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Add-little-video-IN-other-video-tp3823216p3823216.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 hanschke-michael at t-online.de Mon Sep 19 20:54:47 2011 From: hanschke-michael at t-online.de (hanschke-michael at t-online.de) Date: Mon, 19 Sep 2011 20:54:47 +0200 Subject: [FFmpeg-user] Bug configure ffmpeg ERROR: libx264 not found Message-ID: <1R5izP-0swkue0@fwd07.aul.t-online.de> Hello he is a Bug ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab ERROR: libx264 not found MFG Attachment: config.log -------------- next part -------------- A non-text attachment was scrubbed... Name: config.log Type: application/octet-stream Size: 124511 bytes Desc: URL: From 111gopher at seznam.cz Mon Sep 19 22:04:31 2011 From: 111gopher at seznam.cz (111gopher) Date: Mon, 19 Sep 2011 13:04:31 -0700 (PDT) Subject: [FFmpeg-user] Encoding video to stream format Message-ID: <1316462671700-3825026.post@n4.nabble.com> Hi, I have question to you, is possible to encode video to this specific format: / Stream #0.0(eng): Audio: mp4a / 0x6134706D, 8000 Hz, mono, s16 Stream #0.1(eng): Data: rtp / 0x20707472 Stream #0.2(eng): Video: h263, yuv420p, 352x288 [PAR 12:11 DAR 4:3], 29.97 tbr, 100 tbn, 29.97 tbc Stream #0.3(eng): Data: rtp / 0x20707472/ I am using Asterisk to save and play video files but I need convert some video files into this specific format that Asterisk is able to handle it. Many thanks for you answer. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Encoding-video-to-stream-format-tp3825026p3825026.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From thomas at pixelpartner.de Tue Sep 20 00:03:48 2011 From: thomas at pixelpartner.de (Thomas Kumlehn) Date: Tue, 20 Sep 2011 00:03:48 +0200 Subject: [FFmpeg-user] =?utf-8?q?Add_little_video_IN_other_vid=C3=A9o?= In-Reply-To: References: <1316418773989-3823216.post@n4.nabble.com> Message-ID: <5CAD57BD-5936-499E-B954-F51BA67EDD6C@pixelpartner.de> I would suggest to use AviSynth if you are on the Windows32 side. It's very powerful when it comes to this complexity. Thomas Kumlehn PIXEL PARTNER (R) Send from my iPad 3-D http://www.pixelpartner.de Am 19.09.2011 um 16:45 schrieb mikkel meinike : > Sorry mistake one more time. That sounds like the way but if audio an > video is going to stay in > sync you would have to. Split audio and video in to separate files. > cut out of the long video exactly the same time measure at as the > duration of the video you want > to insert. Than ad the pieces together. And then add the audio to the > video. That's how I would do this. > > On Mon, Sep 19, 2011 at 4:42 PM, mikkel meinike wrote: >> That sounds like the way but if audio an video is going to stay in >> sync you would have to. Split audio and video in to separate files. >> cut out of the audio exactly the same measure at as the video you want >> to insert. Than ad the pieces together. That's how I would do this. >> >> Mikkel >> >> On Mon, Sep 19, 2011 at 9:52 AM, numerizik wrote: >>> Hello >>> For one TV local project, i need to do this : >>> >>> I have big Interview vid?o (2 minutes) and some little videos (5 secondes). >>> I need to add little video on the big vid?o using audio of big vid?o. >>> I start to do this : >>> - Extract audio from big video >>> - Extract point a,b,c,d,... >>> - Cut each part >>> - merge each part with audio file >>> http://ffmpeg-users.933282.n4.nabble.com/file/n3823216/timeline.png >>> >>> It's complex and i want to know if you have better solution for this ??? >>> >>> thanks for your help >>> Guillaume >>> >>> -- >>> View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Add-little-video-IN-other-video-tp3823216p3823216.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 >>> >> > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From h.reindl at thelounge.net Tue Sep 20 00:01:35 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Tue, 20 Sep 2011 00:01:35 +0200 Subject: [FFmpeg-user] Bug configure ffmpeg ERROR: libx264 not found In-Reply-To: <1R5izP-0swkue0@fwd07.aul.t-online.de> References: <1R5izP-0swkue0@fwd07.aul.t-online.de> Message-ID: <4E77BBBF.80007@thelounge.net> Am 19.09.2011 20:54, schrieb hanschke-michael at t-online.de: > Hello > > he is a Bug > > ./configure --enable-gpl --enable-version3 --enable-nonfree > --enable-postproc --enable-libfaac --enable-libmp3lame > --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 > --enable-libxvid --enable-x11grab > ERROR: libx264 not found the main question is have you installed x264-devel? if yes - where is it installed? -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From marcosfrm at gmail.com Tue Sep 20 01:07:08 2011 From: marcosfrm at gmail.com (Marcos Felipe Rasia de Mello) Date: Mon, 19 Sep 2011 20:07:08 -0300 Subject: [FFmpeg-user] DirectShow - Unrecognized option 'list_options' In-Reply-To: <20110919085131.GA13022@geppetto> References: <20110919085131.GA13022@geppetto> Message-ID: 2011/9/19 Stefano Sabatini : [...] > > You need libavdevice >= 53.3.2, update your FFmpeg. Thank you guys and sorry for the noise. I was confident that Zeranoe builds were up to date without checking the version and compile date... Marcos From spligak at gmail.com Tue Sep 20 09:13:53 2011 From: spligak at gmail.com (Jason Johnson) Date: Tue, 20 Sep 2011 02:13:53 -0500 Subject: [FFmpeg-user] Issues streaming to Twitch.tv Message-ID: I am currently using the following command to stream my desktop to Twitch.tv: ffmpeg -f x11grab -s "1024x768" -r "15" -i :0.0 -vcodec libx264 -s > "1024x768" -f flv "rtmp://live.justin.tv/app/my_chan_key" > The command does result in a "stream" which is picked up by the servers fine, except the stream is entirely black. The output from the above command is as follows (please excuse the length, trying to be complete): ffmpeg version N-32697-g4bc447a, Copyright (c) 2000-2011 the FFmpeg > developers > built on Sep 18 2011 15:42:01 with gcc 4.5.2 > configuration: --enable-gpl --enable-libfaac --enable-libmp3lame > --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora > --enable-libvorbis --enable-libx264 --enable-libxvid --enable-nonfree > --enable-postproc --enable-version3 --enable-x11grab > libavutil 51. 16. 0 / 51. 16. 0 > libavcodec 53. 16. 0 / 53. 16. 0 > libavformat 53. 12. 0 / 53. 12. 0 > libavdevice 53. 4. 0 / 53. 4. 0 > libavfilter 2. 43. 2 / 2. 43. 2 > libswscale 2. 1. 0 / 2. 1. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > [x11grab @ 0x2feca00] device: :0.0 -> display: :0.0 x: 0 y: 0 width: 1024 > height: 768 > [x11grab @ 0x2feca00] shared memory extension found > [x11grab @ 0x2feca00] Estimating duration from bitrate, this may be > inaccurate > Input #0, x11grab, from ':0.0': > Duration: N/A, start: 1316500191.510707, bitrate: 377487 kb/s > Stream #0.0: Video: rawvideo (BGRA / 0x41524742), bgra, 1024x768, > 377487 kb/s, 15 tbr, 1000k tbn, 15 tbc > Incompatible pixel format 'bgra' for codec 'libx264', auto-selecting format > 'yuv420p' > [buffer @ 0x2ffd2e0] w:1024 h:768 pixfmt:bgra tb:1/1000000 sar:0/1 > sws_param: > [buffersink @ 0x2ffd6e0] auto-inserting filter 'auto-inserted scale 0' > between the filter 'src' and the filter 'out' > [scale @ 0x2ffdee0] w:1024 h:768 fmt:bgra -> w:1024 h:768 fmt:yuv420p > flags:0x4 > [libx264 @ 0x2ffa3e0] using cpu capabilities: MMX2 SSE2Fast FastShuffle > SSEMisalign LZCNT > [libx264 @ 0x2ffa3e0] profile High, level 3.1 > [libx264 @ 0x2ffa3e0] 264 - core 116 r2074 2641b9e - H.264/MPEG-4 AVC codec > - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: > cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 > psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 > cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 > 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=15 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, flv, to 'rtmp://live.justin.tv/app/my_chan_key': > Metadata: > encoder : Lavf53.12.0 > Stream #0.0: Video: h264 ([7][0][0][0] / 0x0007), yuv420p, 1024x768, > q=-1--1, 1k tbn, 15 tbc > Stream mapping: > Stream #0.0 -> #0.0 (rawvideo -> libx264) > Press [q] to stop, [?] for help > frame= 109 fps= 13 q=32766.0 Lsize= 461kB time=00:00:08.73 bitrate= > 432.0kbits/s dup=0 drop=10 > video:458kB audio:0kB global headers:0kB muxing overhead 0.522560% > frame I:1 Avg QP:11.86 size:121831 > [libx264 @ 0x2ffa3e0] frame P:36 Avg QP:17.86 size: 9491 > [libx264 @ 0x2ffa3e0] frame B:72 Avg QP:27.49 size: 70 > [libx264 @ 0x2ffa3e0] consecutive B-frames: 11.0% 1.8% 2.8% 84.4% > [libx264 @ 0x2ffa3e0] mb I I16..4: 14.5% 49.7% 35.8% > [libx264 @ 0x2ffa3e0] mb P I16..4: 0.3% 0.4% 2.8% P16..4: 2.6% 0.6% > 0.5% 0.0% 0.0% skip:92.8% > [libx264 @ 0x2ffa3e0] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 0.6% 0.0% > 0.0% direct: 0.0% skip:99.4% L0:34.5% L1:65.5% BI: 0.1% > [libx264 @ 0x2ffa3e0] 8x8 transform intra:28.0% inter:35.3% > [libx264 @ 0x2ffa3e0] coded y,uvDC,uvAC intra: 77.7% 45.2% 32.5% inter: > 0.4% 0.3% 0.1% > [libx264 @ 0x2ffa3e0] i16 v,h,dc,p: 30% 51% 4% 14% > [libx264 @ 0x2ffa3e0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 13% 18% 34% 4% 7% > 5% 7% 5% 6% > [libx264 @ 0x2ffa3e0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 26% 27% 17% 4% 5% > 5% 5% 4% 6% > [libx264 @ 0x2ffa3e0] i8c dc,h,v,p: 62% 23% 13% 2% > [libx264 @ 0x2ffa3e0] Weighted P-Frames: Y:0.0% UV:0.0% > [libx264 @ 0x2ffa3e0] ref P L0: 80.8% 4.3% 12.3% 2.5% > [libx264 @ 0x2ffa3e0] ref B L0: 55.2% 27.5% 17.3% > [libx264 @ 0x2ffa3e0] ref B L1: 90.7% 9.3% > [libx264 @ 0x2ffa3e0] kb/s:422.72 > When I output the file to my desktop like so: ffmpeg -f x11grab -s "1024x768" -r "15" -i :0.0 -vcodec libx264 -s > "1024x768" -f flv ~/Desktop/test.flv > It plays fine. Also of note, while ffmpeg reports to be streaming the data - however, the Ubuntu System Monitor shows abrupt momentary network activity, then zero network activity. This I believe would indicate some issue in configuration of the RTMP stuff. Any help in understanding what's going wrong here would be much appreciated. From balnaborju at gmail.com Tue Sep 20 10:50:40 2011 From: balnaborju at gmail.com (Balna) Date: Tue, 20 Sep 2011 08:50:40 +0000 (UTC) Subject: [FFmpeg-user] mp4 interleaving Message-ID: Hello, I?d like to use mp4box for swapping the audio in some mp4 files, generated earlier with ffmpeg (h.264+aac). I am however not sure, whether the resulting container will be interleaved in a suitable way for, say, offline playback on a computer or a set-top box. The same interleaving structure would be fine, as ffmpeg did originally, but I don?t have a clue, how ffmpeg does it by default. What I'm most afraid is seeking problems and audio sync problems. So I wonder how ffmpeg interleaves an mp4 output ? (chunk size, whatever..) Thanks in advance, Peter From rapsey at gmail.com Tue Sep 20 14:57:14 2011 From: rapsey at gmail.com (Rapsey) Date: Tue, 20 Sep 2011 14:57:14 +0200 Subject: [FFmpeg-user] libvpx encoding CBR does not work Message-ID: Hello, I'm trying to encode a file to .webm, with constant bitrate, but it pretty much ignores whatever I set for bitrate. What am I doing wrong? ffmpeg -i aerobika.avi -vcodec libvpx -maxrate 500k -minrate 500k -b 500k -bt 500k -quality realtime -acodec libvorbis -ab 128k -threads 0 -f matroska -y aerobika.webm Sergej From mahakcay at gmail.com Tue Sep 20 16:52:30 2011 From: mahakcay at gmail.com (Mahmut Akcay) Date: Tue, 20 Sep 2011 15:52:30 +0100 Subject: [FFmpeg-user] Problem converting video file with mp3 to mp4 Message-ID: I'm having problem when converting avi/flv files with mp3 audio to mp4. It plays well with ffplay and VLC players. However, I can't hear audio when using quicktime player. It looks Quicktime Player doesn't like the way ffmpeg copies mp3 into mp4 file. Is there any solution for it? Thanks in advance mahmut Below is the ffmpeg version I'm using: FFmpeg version SVN-r26402, Copyright (c) 2000-2011 the FFmpeg developers built on Feb 8 2011 10:59:10 with gcc 4.2.1 (Apple Inc. build 5664) configuration: --disable-yasm --enable-gpl libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 From unga888 at yahoo.com Tue Sep 20 17:26:39 2011 From: unga888 at yahoo.com (Unga) Date: Tue, 20 Sep 2011 08:26:39 -0700 (PDT) Subject: [FFmpeg-user] sws_scale down scale produces incorrect video Message-ID: <1316532399.51957.YahooMailClassic@web160120.mail.bf1.yahoo.com> Hi all I use ffmpeg version 0.7-rc1. The sws_scale down scale (full HD -> HD) produces incorrect video with vertical different colour lines. But the upscale (HD->Full HD) produces video without any issue. Was this a bug in ffmpeg which is already been fixed or am I making a mistake in my code? Many thanks in advance. Best regards Unga From alan_fasone at yahoo.com Mon Sep 19 23:17:18 2011 From: alan_fasone at yahoo.com (phase72259) Date: Mon, 19 Sep 2011 14:17:18 -0700 (PDT) Subject: [FFmpeg-user] Converting .MXF to .wmv (or even .flv) Message-ID: <1316467038496-3825254.post@n4.nabble.com> I'm trying to use ffmpeg to convert .MXF (Material eXchange Format) files (video and audio) to preferably .wmv, but I'd settle for .flv at this point. Has anyone does this and could you post the ffmpeg parameters? Thanks. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Converting-MXF-to-wmv-or-even-flv-tp3825254p3825254.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From luj125 at gmail.com Tue Sep 20 19:39:16 2011 From: luj125 at gmail.com (James Lu) Date: Tue, 20 Sep 2011 10:39:16 -0700 Subject: [FFmpeg-user] libvpx encoding CBR does not work In-Reply-To: References: Message-ID: On Tue, Sep 20, 2011 at 5:57 AM, Rapsey wrote: > Hello, > > I'm trying to encode a file to .webm, with constant bitrate, but it pretty > much ignores whatever I set for bitrate. What am I doing wrong? > > ffmpeg -i aerobika.avi -vcodec libvpx -maxrate 500k -minrate 500k -b 500k > -bt 500k -quality realtime -acodec libvorbis -ab 128k -threads 0 -f > matroska -y aerobika.webm > > > > Sergej > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Hey Sergej, In my experience with encoding into webm video , I've had to use the webmproject.org tool vpxenc. For some reason, the mappings of the -b ffmpeg parameter doesn't match to libvpx's bitrate control options. Using vpxenc, the command line you would want to use seems like: ffmpeg -i aerobika.avi -an aerobika.y4m vpxenc --end-usage=cbr --target-bitrate=500 --rt --threads=0 --output=video.webm aerobika.y4m ffmpeg -i aerobika.avi -vn -acodec libvorbis -aq 4 -threads 0 audio.ogg mkvmerge aerobika.avi audio.ogg video.webm Notes on the commands: 1) Vorbis uses aq commands, the mapping of aq to ab can be found at http://en.wikipedia.org/wiki/Vorbis#Technical_details 2) After generating the audio and video seperatly, you need to combine without reencoding; this is done using mkvmerge tool found here: http://www.bunkus.org/videotools/mkvtoolnix/ I haven't checked the bug lists for this particular oddity, but it would help if you submitted one or confirmed an existing report. Hope this helps! ~James From rogerdpack2 at gmail.com Tue Sep 20 20:14:48 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Tue, 20 Sep 2011 12:14:48 -0600 Subject: [FFmpeg-user] sws_scale down scale produces incorrect video In-Reply-To: <1316532399.51957.YahooMailClassic@web160120.mail.bf1.yahoo.com> References: <1316532399.51957.YahooMailClassic@web160120.mail.bf1.yahoo.com> Message-ID: > Was this a bug in ffmpeg which is already been fixed or am I making a mistake in my code? Maybe you can test with a newer version of ffmpeg and check? -r From marcosfrm at gmail.com Tue Sep 20 22:16:15 2011 From: marcosfrm at gmail.com (Marcos Felipe Rasia de Mello) Date: Tue, 20 Sep 2011 17:16:15 -0300 Subject: [FFmpeg-user] libvpx encoding CBR does not work In-Reply-To: References: Message-ID: 2011/9/20 James Lu : <...> > > In my experience with encoding into webm video , I've had to use the > webmproject.org tool vpxenc. For some reason, the mappings of the -b ffmpeg > parameter doesn't match to libvpx's bitrate control options. Using vpxenc, > the command line you would want to use seems like: It should, because setting "minrate = maxrate = vb" enable libvpx VPX_CBR encoding mode. http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/libvpxenc.c;h=01749a29#l289 http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/encoders.texi;h=f50ee10d#l485 > > ffmpeg -i aerobika.avi -an aerobika.y4m > vpxenc --end-usage=cbr --target-bitrate=500 --rt --threads=0 > --output=video.webm aerobika.y4m > ffmpeg -i aerobika.avi -vn -acodec libvorbis -aq 4 -threads 0 audio.ogg > mkvmerge aerobika.avi audio.ogg video.webm > You can also use pipe: ffmpeg -i input.xxx -pix_fmt yuv420p -f yuv4mpegpipe - 2>/dev/null | vpxenc - -o video.webm For audio too: ffmpeg -i input.xxx -f wav - 2>/dev/null | oggenc -b 96 --ignorelength - -o audio.ogg > Notes on the commands: > 1) Vorbis uses aq commands, the mapping of aq to ab can be found at > http://en.wikipedia.org/wiki/Vorbis#Technical_details > 2) After generating the audio and video seperatly, you need to combine > without reencoding; this is done using mkvmerge tool found here: > http://www.bunkus.org/videotools/mkvtoolnix/ > > I haven't checked the bug lists for this particular oddity, but it would > help if you submitted one or confirmed an existing report. > > Hope this helps! > > ~James Regards, Marcos From rickcorteza at gmail.com Wed Sep 21 13:24:58 2011 From: rickcorteza at gmail.com (Rick C.) Date: Wed, 21 Sep 2011 19:24:58 +0800 Subject: [FFmpeg-user] invalid pixel error Message-ID: Hello, Could someone explain what this error means: [scale @ 0x101403960] w:608 h:256 fmt:yuv420p -> w:720 h:304 fmt:yuv420p flags:0x4 [mpeg4 @ 0x101819e00] Invalid pixel aspect ratio 361/360, limit is 255/255 That's just a small clip of the output and I get this from time to time so any help in understanding what this means would be great. Thanks! rc From xflyway at gmail.com Wed Sep 21 14:05:38 2011 From: xflyway at gmail.com (xfly) Date: Wed, 21 Sep 2011 14:05:38 +0200 Subject: [FFmpeg-user] rtp streaming - ok with ffmpeg 0.8.3 - error with ffmpeg git version N-32748-g358d837 Message-ID: Hi, I want to stream the video captured from my camera via rtp. I'm using this simple command: $ffmpeg -f video4linux2 -s 640x480 -r 25 -i /dev/video0 -f rtp -muxdelay 0 -vcodec mpeg2video -sameq rtp://$DEFAULT_HOST:$VIDEO_PORT It works with ffmpeg version 0.8.3. Details: ffmpeg version 0.8.3, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 21 2011 11:47:11 with gcc 4.4.3 configuration: --prefix=/usr --enable-shared --enable-pthreads --enable-version3 --enable-libtheora --enable-libdc1394 --enable-libvorbis --enable-libschroedinger --enable-zlib --enable-bzlib --enable-vdpau --enable-libdirac libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 7. 0 / 53. 7. 0 libavformat 53. 4. 0 / 53. 4. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 It does NOT work with ffmpeg version N-32748-g358d837 pulled down TODAY from git repo. Details: ffmpeg version N-32748-g358d837, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 21 2011 11:33:34 with gcc 4.4.3 configuration: --prefix=/usr --enable-shared --enable-pthreads --enable-version3 --enable-libtheora --enable-libdc1394 --enable-libvorbis --enable-libschroedinger --enable-zlib --enable-bzlib --enable-vdpau --enable-libdirac libavutil 51. 16. 1 / 51. 16. 1 libavcodec 53. 16. 0 / 53. 16. 0 libavformat 53. 12. 0 / 53. 12. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 43. 2 / 2. 43. 2 libswscale 2. 1. 0 / 2. 1. 0 This what I get from ffmpeg version N-32748-g358d837 when I run the command: $ffmpeg -f video4linux2 -s 640x480 -r 25 -i /dev/video0 -f rtp -muxdelay 0 -vcodec mpeg2video -sameq rtp://$DEFAULT_HOST:$VIDEO_PORT Output: ffmpeg version N-32748-g358d837, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 21 2011 11:33:34 with gcc 4.4.3 configuration: --prefix=/usr --enable-shared --enable-pthreads --enable-version3 --enable-libtheora --enable-libdc1394 --enable-libvorbis --enable-libschroedinger --enable-zlib --enable-bzlib --enable-vdpau --enable-libdirac libavutil 51. 16. 1 / 51. 16. 1 libavcodec 53. 16. 0 / 53. 16. 0 libavformat 53. 12. 0 / 53. 12. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 43. 2 / 2. 43. 2 libswscale 2. 1. 0 / 2. 1. 0 [video4linux2 @ 0x97533a0] The driver changed the time per frame from 1/25 to 1/30 [video4linux2 @ 0x97533a0] Estimating duration from bitrate, this may be inaccurate Input #0, video4linux2, from '/dev/video0': Duration: N/A, start: 1316603488.369544, bitrate: 147456 kb/s Stream #0.0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 640x480, 147456 kb/s, 30 tbr, 1000k tbn, 30 tbc Output #0, rtp, to 'rtp://127.0.0.1:5000': Output file #0 does not contain any stream I also tried the following commands obtaining the same result "Output file #0 does not contain any stream" $ ffmpeg -f video4linux2 -s 640x480 -r 25 -i /dev/video0 -f rtp -vcodec mpegts -vframes 25 rtp://127.0.0.1:5000 $ ffmpeg -f video4linux2 -s 640x480 -r 25 -i /dev/video0 -f mpegts -f rtp rtp://127.0.0.1:5000 Is there any problem with the syntax of the command in the new ffmpeg version? Did the syntax changed for rtp streaming? Thank you in advance for your help. Xfly From teomer1984 at gmail.com Wed Sep 21 15:33:31 2011 From: teomer1984 at gmail.com (Matteo) Date: Wed, 21 Sep 2011 06:33:31 -0700 (PDT) Subject: [FFmpeg-user] RTSP Video Problem Message-ID: <1316612011506-3829983.post@n4.nabble.com> Hi, I've a problem with recording from rtsp network camera. Sometimes, the file output remains blocked on same time, as it is visible in screenshot: http://ffmpeg-users.933282.n4.nabble.com/file/n3829983/screenshot_ffmpeg.png The command line that I use is: ffmpeg -i rtsp://....... -vcodec mpeg4 -r 25 out.mp4 The codec of the camera's stream is mpeg4. Somebody can help me? thank you Matteo -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/RTSP-Video-Problem-tp3829983p3829983.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From unga888 at yahoo.com Wed Sep 21 18:48:52 2011 From: unga888 at yahoo.com (Unga) Date: Wed, 21 Sep 2011 09:48:52 -0700 (PDT) Subject: [FFmpeg-user] Is av_malloc() and av_free() thread safe? Message-ID: <1316623732.1599.YahooMailClassic@web160106.mail.bf1.yahoo.com> Hi all Is av_malloc() and av_free() thread safe? Many thanks in advance. Best regards Unga From swathiep2009 at gmail.com Wed Sep 21 09:02:30 2011 From: swathiep2009 at gmail.com (swathi e.p) Date: Wed, 21 Sep 2011 12:32:30 +0530 Subject: [FFmpeg-user] Issue with cross compiling ffmpeg for linux in windows Message-ID: Hi I am in the process of configuring ffmpeg. The file attached with this mail is the script i used to configure ffmpeg. I have tried in all possible ways i know, but always getting "No such file or directory" error as below. WARNING: /cygdrive/e/cygwin/home/mbt/Android-NDK/android-ndk/toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows/bin/arm-linux-androideabi-pkg-config not found, library detection may fail. mktemp -u XXXXXX 6X6NG4 WARNING: Unknown C compiler /cygdrive/e/cygwin/home/mbt/Android-NDK/android-ndk/toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows/bin/arm-linux-androideabi-gcc, unable to select optimal CFLAGS check_ld check_cc BEGIN /tmp/ffconf.6dDK4GNb.c 1 int main(void){ return 0; } END /tmp/ffconf.6dDK4GNb.c /cygdrive/e/cygwin/home/mbt/Android-NDK/android-ndk/toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows/bin/arm-linux-androideabi-gcc -I//usr/include -c -o /tmp/ffconf.IvgodWsv.o /tmp/ffconf.6dDK4GNb.c ./configure: line 595: /cygdrive/e/cygwin/home/mbt/Android-NDK/android-ndk/toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows/bin/arm-linux-androideabi-gcc: No such file or directory C compiler test failed. I tried to define path in different ways as below: >> By setting environment variable called NDK and defined PREBUILT and PLATFORM paths as: PLATFORM=$NDK/platforms/android-8/arch-arm/ PREBUILT=$NDK/toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows >> Then by defining locally as: 1. NDK=/cygdrive/e/cygwin/home/mbt/Android-NDK/android-ndk PLATFORM=$NDK/platforms/android-8/arch-arm/ PREBUILT=$NDK/toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows 2. NDK=e/cygwin/home/mbt/Android-NDK/android-ndk PLATFORM=$NDK/platforms/android-8/arch-arm/ PREBUILT=$NDK/toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows 3. NDK=e:/cygwin/home/mbt/Android-NDK/android-ndk PLATFORM=$NDK/platforms/android-8/arch-arm/ PREBUILT=$NDK/toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows 4. NDK=/home/mbt/Android-NDK/android-ndk PLATFORM=$NDK/platforms/android-8/arch-arm/ PREBUILT=$NDK/toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows 5. NDK=~/Android-NDK/android-ndk PLATFORM=$NDK/platforms/android-8/arch-arm/ PREBUILT=$NDK/toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows 6. NDK=E:/cygwin/home/mbt/Android-NDK/android-ndk PLATFORM=$NDK/platforms/android-8/arch-arm/ PREBUILT=$NDK/toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows All ended with same error. Cygwin version - 1.7.9 ffmpeg version - 0.8.2 Android NDK version - r5 Please let me know, if i have done any mistake in the steps defined above. Thanks & Reagrds, Swathi -------------- next part -------------- A non-text attachment was scrubbed... Name: new 2.sh Type: application/x-sh Size: 584 bytes Desc: not available URL: From mb at powerflasher.de Wed Sep 21 17:11:02 2011 From: mb at powerflasher.de (Malte Beyer) Date: Wed, 21 Sep 2011 17:11:02 +0200 Subject: [FFmpeg-user] FFmpeg can't get access to webcam when excuted via Adobe AIR? Message-ID: <24E4D111F793154782E3D8FC9E0125C5606F65455D@mail1.powerflasher.de> Hello, I am trying to use FFmpeg via the NativeProcess-API in Adobe AIR 2.0. My goal is to encode the video of the webcam with H.264 and stream it to a Flash Media Server. The application will be x-platform (Win and Mac). Additionally I need to switch between different webcams during the live stream. I already made a small prototype and I nearly managed to stream the webcam to my streaming server. I can send the stream of the webcam to the streaming server when I am using the windows console. I also can communicate with FFmpeg via AIR and I can e.g. send a local video to the streaming server when using AIR. But I cannot manage it to send the video of the webcam to the streaming server when using AIR. I am using the same command on the command line and in AIR. It works from the command line, but it does not work from AIR. I am using this command: ffmpeg -f dshow -i video="Logitech QuickCam E2500":audio="Mikrofon (IDT High Definition A" -f flv rtmp://xyz As I said it works, when executed via the command line. But when I execute the same command from AIR I get the following error message in my AIR app: [dshow @ 01998640] Could not find video device. video="Logitech QuickCam E2500": Input/output error The question is: might this be a rights problem? Is it not allowed for FFmpeg to access my webcam when FFmpeg is executed via AIR? And if so, how can I solve it? Or is it some other problem? Two other questions I was wondering about (probably much easier to answer): - I cannot see a solution how to send the video of the webcam to a streaming server on Mac (dshow seems to work just for Windows). Is there no chance to have input via the webcam on Mac? Just Linux and Windows? - What do you think about switching the cameras during runtime? Is that a problem? Will I stop the dshow-command by sending a "q" and start the next dshow-command with the different cam right after? How will this look like in the frontend? Black screen during the switch? Thanks for your help. Malte Beyer Technical Director ------------------------------------------------------ Powerflasher GmbH Drehturm Belvedereallee 5 52070 Aachen, Germany E mb at powerflasher.de W http://www.powerflasher.de B http://blog.powerflasher.de S Facebook | Twitter | Xing T +49 241 91880 247 Powerflasher Showreel 2011 FDT 4 - Coding Comfort Gesch?ftsf?hrung: Carlo Blatz, Amtsgericht Aachen - HRB 8683, Ust-IdNr.: DE220959918 From rogerdpack2 at gmail.com Wed Sep 21 20:00:45 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 21 Sep 2011 12:00:45 -0600 Subject: [FFmpeg-user] Issue with cross compiling ffmpeg for linux in windows In-Reply-To: References: Message-ID: > WARNING: > /cygdrive/e/cygwin/home/mbt/Android-NDK/android-ndk/toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows/bin/arm-linux-androideabi-pkg-config > not found, library detection may fail. > mktemp -u XXXXXX So you're trying to build for ARM from within cygwin? From rogerdpack2 at gmail.com Wed Sep 21 20:02:14 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 21 Sep 2011 12:02:14 -0600 Subject: [FFmpeg-user] RTSP Video Problem In-Reply-To: <1316612011506-3829983.post@n4.nabble.com> References: <1316612011506-3829983.post@n4.nabble.com> Message-ID: > I've a problem with recording from rtsp network camera. Sometimes, the file > output remains blocked on same time, as it is visible in screenshot: > > http://ffmpeg-users.933282.n4.nabble.com/file/n3829983/screenshot_ffmpeg.png > > The command line that I use is: > ffmpeg -i rtsp://....... -vcodec mpeg4 -r 25 out.mp4 Does it block forever? does it repeat that message forever? if you hit 'q' to quit does it quit eventually? -r From afcrowe at gmail.com Wed Sep 21 21:34:08 2011 From: afcrowe at gmail.com (system159) Date: Wed, 21 Sep 2011 12:34:08 -0700 (PDT) Subject: [FFmpeg-user] Convert Video stream, while keeping all other streams Message-ID: <1316633648333-3831170.post@n4.nabble.com> Hi all, I'm taking in an mpeg2 transport stream that has video, audio, and KLV metadata on separate streams. I'm using ffmpeg to transcode the video to the format successfully, however the KLV metadata stream gets dropped completely. Is there a way to tell ffmpeg to pass through any elementary streams when it doesn't know what they are? Thanks! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Convert-Video-stream-while-keeping-all-other-streams-tp3831170p3831170.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From rogerdpack2 at gmail.com Wed Sep 21 23:58:29 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 21 Sep 2011 15:58:29 -0600 Subject: [FFmpeg-user] slanted line when playing a video with an odd width Message-ID: Hello. I noticed that ffmpeg can "capture" windows that have an odd width, ex: 777x777 however, when played in ffplay, there is a curious diagonal line displayed [1] which doesn't appear in mplayer. Bug reportable? Cheers! -r [1] http://twitpic.com/6ofrpe From rogerdpack2 at gmail.com Thu Sep 22 00:10:12 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 21 Sep 2011 16:10:12 -0600 Subject: [FFmpeg-user] FFmpeg can't get access to webcam when excuted via Adobe AIR? In-Reply-To: <24E4D111F793154782E3D8FC9E0125C5606F65455D@mail1.powerflasher.de> References: <24E4D111F793154782E3D8FC9E0125C5606F65455D@mail1.powerflasher.de> Message-ID: > ffmpeg -f dshow -i video="Logitech QuickCam E2500":audio="Mikrofon (IDT High Definition A" -f flv rtmp://xyz > > As I said it works, when executed via the command line. But when I execute the same command from AIR I get the following error message in my AIR app: > > [dshow @ 01998640] Could not find video device. > video="Logitech QuickCam E2500": Input/output error > > The question is: might this be a rights problem? Is it not allowed for FFmpeg to access my webcam when FFmpeg is executed via AIR? And if so, how can I solve it? Or is it some other problem? Sounds like a rights problem. Maybe you can specify in AIR preferences which camera AIR has access to, or maybe...you can access the raw bytes from the camera through AIR somehow? > - ? ? ? ? ?I cannot see a solution how to send the video of the webcam to a streaming server on Mac (dshow seems to work just for Windows). Is there no chance to have input via the webcam on Mac? Just Linux and Windows? I have thought about that recently myself. I don't think it can yet. VLC has a qtcapture module, and once it gets working audio capture [1] I hope to port it over to ffmpeg. > - ? ? ? ? ?What do you think about switching the cameras during runtime? Is that a problem? Will I stop the dshow-command by sending a "q" and start the next dshow-command with the different cam right after? How will this look like in the frontend? Black screen during the switch? I believe so. Or rather one window disappears, the next appears. Related: VLC has an --sout-keep option. That...might maybe work. Also I could hack you up a directshow video source device that presents AviSynth (which has runtime scripts so it can change its source...I think so anyway), but that would only be for windows so not sure... GL. -r [1] http://forum.videolan.org/viewtopic.php?f=12&t=89140 From gavr.mail at gmail.com Thu Sep 22 06:20:06 2011 From: gavr.mail at gmail.com (Kirill Gavrilov) Date: Thu, 22 Sep 2011 08:20:06 +0400 Subject: [FFmpeg-user] Is av_malloc() and av_free() thread safe? In-Reply-To: <1316623732.1599.YahooMailClassic@web160106.mail.bf1.yahoo.com> References: <1316623732.1599.YahooMailClassic@web160106.mail.bf1.yahoo.com> Message-ID: On Wed, Sep 21, 2011 at 8:48 PM, Unga wrote: > > Hi all > > Is av_malloc() and av_free() thread safe? > av_malloc() and av_free() just redirects to standard malloc() / posix_memalign() / free() functions. The main difference is that them ensure alignment for some instructions like SSE / MMX and so on available on your CPU and used in FFmpeg decoders/encoders. No global storage and no any advanced memory management are implemented in FFmpeg. Thus yes - these FFmpeg functions should be thread-safe if memory allocator of your CRT library is thread-safe ;) (generally it is). ----------------------------------------------- Kirill Gavrilov, Software designer. From teomer1984 at gmail.com Thu Sep 22 08:41:29 2011 From: teomer1984 at gmail.com (Matteo) Date: Wed, 21 Sep 2011 23:41:29 -0700 (PDT) Subject: [FFmpeg-user] RTSP Video Problem In-Reply-To: References: <1316612011506-3829983.post@n4.nabble.com> Message-ID: <1316673689943-3832556.post@n4.nabble.com> yes, the message is repeated and the time remains the same. if from another shell at the same time I watch the video stream with ffplay, this can be seen. if i stop and I redo the same command, the ffmpeg maybe work but not always. Thank you Matteo -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/RTSP-Video-Problem-tp3829983p3832556.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From unga888 at yahoo.com Thu Sep 22 11:39:07 2011 From: unga888 at yahoo.com (Unga) Date: Thu, 22 Sep 2011 02:39:07 -0700 (PDT) Subject: [FFmpeg-user] Is av_malloc() and av_free() thread safe? In-Reply-To: Message-ID: <1316684347.32357.YahooMailClassic@web160113.mail.bf1.yahoo.com> --- On Thu, 9/22/11, Kirill Gavrilov wrote: > From: Kirill Gavrilov > Subject: Re: [FFmpeg-user] Is av_malloc() and av_free() thread safe? > To: "FFmpeg user questions and RTFMs" > Date: Thursday, September 22, 2011, 12:20 PM > On Wed, Sep 21, 2011 at 8:48 PM, Unga > > wrote: > > > > Hi all > > > > Is av_malloc() and av_free() thread safe? > > > av_malloc() and av_free() just redirects to standard > malloc() / > posix_memalign() / free() functions. > The main difference is that them ensure alignment for some > instructions like > SSE / MMX and so on available on your CPU and used in > FFmpeg > decoders/encoders. > > No global storage and no any advanced memory management are > implemented in > FFmpeg. > Thus yes - these FFmpeg functions should be thread-safe if > memory allocator > of your CRT library is thread-safe ;) (generally it is). > ----------------------------------------------- > Kirill Gavrilov, > Software designer. > Hi Kirill Thank you for the info. I'll take it as thread safe. Best regards Unga From jdmls at yahoo.com Thu Sep 22 17:46:14 2011 From: jdmls at yahoo.com (John Doe) Date: Thu, 22 Sep 2011 08:46:14 -0700 (PDT) Subject: [FFmpeg-user] ffmpeg and external libs... Message-ID: <1316706374.87629.YahooMailNeo@web114709.mail.gq1.yahoo.com> Hi, I am back to compiling ffmpeg and I was wondering what external libraries I still need to include... I saw that amr stuff, h264, etc seems to be supported internally by ffmpeg. Are there any external libs that are still must have...? By example is the x264 external lib no more better? Another question: I did not see a rotate filter... is it an external option? Thx, JD From naveen.kumar at halosys.com Thu Sep 22 09:25:54 2011 From: naveen.kumar at halosys.com (Kumar, Naveen) Date: Thu, 22 Sep 2011 12:55:54 +0530 Subject: [FFmpeg-user] ffmpeg issue Message-ID: Hello Sir, Good morning, i want to play video file in android using your ffmpeg lib , here i am not getting success with git hub all ffmpeg samples. Please need help for implementing ffmpeg in my own video play app Thanks, Naveen From c.guo at nbsystems.tv Thu Sep 22 09:36:49 2011 From: c.guo at nbsystems.tv (Chao Guo) Date: Thu, 22 Sep 2011 15:36:49 +0800 Subject: [FFmpeg-user] Can FFMPEG encode DVCPRO HD? Message-ID: <4e7ae53a.44cae30a.7aae.ffffe0af@mx.google.com> Dear Guys I found ffmpeg can only encoder image to DVCPRO SD. Can anybody tell me how to make it be able to encode the DVCPRO HD frame like 1080i and 720p ? Thanks a lot Edward From rogerdpack2 at gmail.com Thu Sep 22 20:12:33 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Thu, 22 Sep 2011 12:12:33 -0600 Subject: [FFmpeg-user] RTSP Video Problem In-Reply-To: <1316673689943-3832556.post@n4.nabble.com> References: <1316612011506-3829983.post@n4.nabble.com> <1316673689943-3832556.post@n4.nabble.com> Message-ID: > yes, the message is repeated and the time remains the same. if from another > shell at the same time I watch the video stream with ffplay, this can be > seen. if i stop and I redo the same command, the ffmpeg maybe work but not > always. Maybe a bug in ffmpeg's rtsp? maybe it doesn't handle not being able to receive enough packets (cpu not fast enough) right? does ffplay never stop? From rogerdpack2 at gmail.com Thu Sep 22 20:56:08 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Thu, 22 Sep 2011 12:56:08 -0600 Subject: [FFmpeg-user] FFmpeg can't get access to webcam when excuted via Adobe AIR? In-Reply-To: References: <24E4D111F793154782E3D8FC9E0125C5606F65455D@mail1.powerflasher.de> Message-ID: >> - ? ? ? ? ?I cannot see a solution how to send the video of the webcam to a streaming server on Mac (dshow seems to work just for Windows). Is there no chance to have input via the webcam on Mac? Just Linux and Windows? > > I have thought about that recently myself. ?I don't think it can yet. > VLC has a qtcapture module, and once it gets working audio capture [1] > I hope to port it over to ffmpeg. hot off the presses, looks like there was a patch made: http://mailman.videolan.org/pipermail/vlc-devel/2011-September/082063.html though apparently it doesn't work great "quite yet" or something. From hardik.sharma22 at yahoo.com Thu Sep 22 21:20:02 2011 From: hardik.sharma22 at yahoo.com (Hardik Sharma) Date: Thu, 22 Sep 2011 12:20:02 -0700 (PDT) Subject: [FFmpeg-user] Read h264 file into byte stream format Message-ID: <1316719202.20186.YahooMailNeo@web46203.mail.sp1.yahoo.com> Hi, Can I access h264 encoded file as a byte stream format?? From michael at aram.at Thu Sep 22 21:54:55 2011 From: michael at aram.at (Michael Aram) Date: Thu, 22 Sep 2011 21:54:55 +0200 Subject: [FFmpeg-user] max_analyze_duration reached Message-ID: Hi ffmpegers, when transforming a video from MOV to FLV i get the warning "[mov,mp4,m4a,3gp,3g2,mj2 @ 0x12180c800]max_analyze_duration reached". This is my command/output: ################# ffmpeg -i test.mov -ar 22050 -croptop 76 -cropbottom 76 -cropright 12 -cropleft 12 -s 655x375 -sameq test.flv FFmpeg version 0.6.1, Copyright (c) 2000-2010 the FFmpeg developers built on Jan 6 2011 22:05:42 with gcc 4.2.1 (Apple Inc. build 5664) configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-avfilter-lavf --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libfaac --enable-libfaad --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --enable-nonfree --mandir=/opt/local/share/man --enable-shared --enable-pthreads --disable-indevs --cc=/usr/bin/gcc-4.2 --arch=x86_64 libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.19. 0 / 1.19. 0 libswscale 1.11. 0 / 1.11. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x12180c800]max_analyze_duration reached Seems stream 0 codec frame rate differs from container frame rate: 5000.00 (5000/1) -> 25.00 (25/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt Duration: 00:01:46.69, start: 0.000000, bitrate: 6706 kb/s Stream #0.0(eng): Video: h264, yuv420p, 720x576, 6577 kb/s, PAR 10000:9375 DAR 4:3, 25 fps, 25 tbr, 2500 tbn, 5k tbc Stream #0.1(eng): Audio: aac, 44100 Hz, stereo, s16, 125 kb/s Stream #0.2(eng): Data: tmcd / 0x64636D74 Output #0, flv, to 'test.flv': Metadata: encoder : Lavf52.64.2 Stream #0.0(eng): Video: flv, yuv420p, 655x375 [PAR 100:131 DAR 4:3], q=2-31, 200 kb/s, 1k tbn, 25 tbc Stream #0.1(eng): Audio: libmp3lame, 22050 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding frame= 2665 fps=164 q=0.0 Lsize= 23156kB time=106.64 bitrate=1778.8kbits/s dup=0 drop=2 video:22216kB audio:834kB global headers:0kB muxing overhead 0.458958% ################# Two questions: 1) Is this a warning I have to worry about? 2) Are there any obvious important options missing if I want to have good video quality? Cheers, Michael! From rehman_saeed at live.com Thu Sep 22 21:56:36 2011 From: rehman_saeed at live.com (Rehman Saeed) Date: Thu, 22 Sep 2011 20:56:36 +0100 Subject: [FFmpeg-user] delay/lag in playing stream from network camera in ffplay In-Reply-To: References: Message-ID: Hi All, I am having problem (delay or lag of 3-4 seconds) in playing stream from network camera. Those 3-4 seconds are with resolution of 320x240. If I change my resolution to 640x480, ffplay almost get dead. Please find below my environment OS: "Window 7 Home Premium" Environment: "MINGW32_NT-6.1 SAEED-PC 1.0.17(0.48/3/2) 2011-04-24 23:39 i686 Msys" Gcc version: "4.6.1" ffplay/ffmpeg version: "0.8.3" Commandline in MingW: {ffplay.exe "http://192.168.0.111:8080/videostream.asf?user=admin&pwd=admin"} Network camera stream: "MJPEG" I have validated the build of ffplay/ffmpeg with bigbuckbunny file and everything worked perfectly even with full HD mode using below command: ./ffplay.exe "C:\work\Demo_Clips\big_buck_bunny_1080p_surround.avi" The above command line shows the video perfectly. I have verified that network camera is also working and streaming images as firefox can show stream if i give the url "http://192.168.0.111:8080/videostream.cgi?user=admin&pwd=admin". It seems like I am missing something very basic in order to access/decode video stream from network camera. Can someone please enlighten me what I am missing or point me to right direction? Please let me know if I need to provide any more information. Many thanks. Kind regards, Rehman From dave.bevan at bbc.co.uk Thu Sep 22 23:01:26 2011 From: dave.bevan at bbc.co.uk (Dave Bevan) Date: Thu, 22 Sep 2011 22:01:26 +0100 Subject: [FFmpeg-user] Can FFMPEG encode DVCPRO HD? References: <4e7ae53a.44cae30a.7aae.ffffe0af@mx.google.com> Message-ID: >Dear Guys >I found ffmpeg can only encoder image to DVCPRO SD. Can anybody tell me how to make it be able to encode the DVCPRO HD frame like 1080i and 720p ? http://code.google.com/p/ffmbc/ >Thanks a lot >Edward http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 2927 bytes Desc: not available URL: From stefasab at gmail.com Fri Sep 23 00:35:41 2011 From: stefasab at gmail.com (Stefano Sabatini) Date: Fri, 23 Sep 2011 00:35:41 +0200 Subject: [FFmpeg-user] slanted line when playing a video with an odd width In-Reply-To: References: Message-ID: <20110922223541.GA2911@geppetto> On date Wednesday 2011-09-21 15:58:29 -0600, Roger Pack encoded: > Hello. I noticed that ffmpeg can "capture" windows that have an odd width, ex: > 777x777 > > however, when played in ffplay, there is a curious diagonal line > displayed [1] which doesn't appear in mplayer. > Bug reportable? > Cheers! I remember the bug and I'm not yet sure it was a bug of FFmpeg or of SDL, and I can't apparently reproduce it in my Debian. And yes reporting may be a good idea anyway (providing sample and command, or you may use the lavfi device + testsrc). From rogerdpack2 at gmail.com Fri Sep 23 01:44:15 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Thu, 22 Sep 2011 17:44:15 -0600 Subject: [FFmpeg-user] delay/lag in playing stream from network camera in ffplay In-Reply-To: References: Message-ID: > I am having problem (delay or lag of 3-4 seconds) in > playing stream from network camera. Those 3-4 ?seconds are with > resolution of 320x240. If I change my resolution to 640x480, ffplay > almost get dead. Please find below my environment is fps high? I know I ran into the same problem with a 1 fps source the other day...it hung for like 5s before starting to play back, possible because of message " Estimating duration from bitrate, this may be inaccurate" maybe there's a way to disable that, or some way to turn off cacheing? From dashing.meng at gmail.com Fri Sep 23 02:34:59 2011 From: dashing.meng at gmail.com (littlebat) Date: Fri, 23 Sep 2011 08:34:59 +0800 Subject: [FFmpeg-user] ffmpeg and external libs... In-Reply-To: <1316706374.87629.YahooMailNeo@web114709.mail.gq1.yahoo.com> References: <1316706374.87629.YahooMailNeo@web114709.mail.gq1.yahoo.com> Message-ID: <20110923083459.c5f43cb8.dashing.meng@gmail.com> On Thu, 22 Sep 2011 08:46:14 -0700 (PDT) John Doe wrote: > Hi, > > I am back to compiling ffmpeg and I was wondering what external > libraries I still need to include... I saw that amr stuff, h264, etc > seems to be supported internally by ffmpeg. Are there any external > libs that are still must have...? By example is the x264 external lib > no more better? You can test them to see if it is ok for you. e.g., I test internal "aac" encoder to convert video for my mobilephone, it is ok for me. Another question: I did not see a rotate filter... is > it an external option? Try -vf ?transpose=N? From edward.guo at hotmail.com Fri Sep 23 04:33:45 2011 From: edward.guo at hotmail.com (=?gb2312?B?zPogxL3I3Q==?=) Date: Fri, 23 Sep 2011 10:33:45 +0800 Subject: [FFmpeg-user] Can FFMPEG encode DVCPRO HD? Message-ID: Dear Guys I found ffmpeg can only encoder image to DVCPRO SD. Can anybody tell me how to make it be able to encode the DVCPRO HD frame like 1080i and 720p ? Thanks a lot! Kind Wishes Yours Chao.Guo From teomer1984 at gmail.com Fri Sep 23 08:51:50 2011 From: teomer1984 at gmail.com (Matteo) Date: Thu, 22 Sep 2011 23:51:50 -0700 (PDT) Subject: [FFmpeg-user] RTSP Video Problem In-Reply-To: References: <1316612011506-3829983.post@n4.nabble.com> <1316673689943-3832556.post@n4.nabble.com> Message-ID: <1316760710350-3835954.post@n4.nabble.com> I have tried other release of ffmpeg but the problem persist. FFplay works normally and never does stop. I should add some video options? Thank you Matteo -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/RTSP-Video-Problem-tp3829983p3835954.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From rehman_saeed at live.com Fri Sep 23 09:44:43 2011 From: rehman_saeed at live.com (Rehman Saeed) Date: Fri, 23 Sep 2011 08:44:43 +0100 Subject: [FFmpeg-user] delay/lag in playing stream from network camera in ffplay In-Reply-To: References: , , Message-ID: Thanks Roger for your reply. Indeed the fps is 30fps for 320x240 and 15fps for 640x480. I have not reached to the point where I can see those messages. Maybe providing more information like bitrates etc. should solve the problem but I haven't tried yet Do you see delay/lag in case of 1 fps after video starts? > From: rogerdpack2 at gmail.com > Date: Thu, 22 Sep 2011 17:44:15 -0600 > To: ffmpeg-user at ffmpeg.org > Subject: Re: [FFmpeg-user] delay/lag in playing stream from network camera in ffplay > > > I am having problem (delay or lag of 3-4 seconds) in > > playing stream from network camera. Those 3-4 seconds are with > > resolution of 320x240. If I change my resolution to 640x480, ffplay > > almost get dead. Please find below my environment > > is fps high? I know I ran into the same problem with a 1 fps source > the other day...it hung for like 5s before starting to play back, > possible because of message > " Estimating duration from bitrate, this may be inaccurate" > maybe there's a way to disable that, or some way to turn off cacheing? > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From drabner at zoobe.com Fri Sep 23 09:54:08 2011 From: drabner at zoobe.com (Drabner) Date: Fri, 23 Sep 2011 00:54:08 -0700 (PDT) Subject: [FFmpeg-user] av_write_frame crashes without error message Message-ID: <1316764448768-3836086.post@n4.nabble.com> Hey there, Before I start, I should point out that I have almost no expertise in audio and video codecs, and the same goes for ffmpeg/libav. I'm trying to get all frames of an audio file (*.wav) and then save those frames to a new audio file with the same format (so a more or less copied file should be the result). This works well. But now I'm trying to get all video frames of a file (*.mp4) that has both audio and video content. The getting seems to work fine without any errors. But when trying to write the frames I got into a new video file with the same format - but this time, using video frames only, no audio - av_write_frame just crashes without giving any clue why. Here is the function that reads the frames in: //--------------------------------------------------------------------- //--------------------------------------------------------------------- //--------------------------------------------------------------------- void FramesCollection::DecodeVideo(const char* p_filename) { av_log(NULL, AV_LOG_ERROR, "%s","Starting video decoding\n"); // Get the format context... int ret = avformat_open_input(&_videoFormatContext, p_filename, NULL, NULL); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "%s","Error opening file '%s'\n", p_filename); exit(-1); } // ... and fill it with stream info (contains number of frames, etc.) av_find_stream_info(_videoFormatContext); // Find the video stream AVStream* stream; for(int i = 0; i < (int)_videoFormatContext->nb_streams; i++) { if(_videoFormatContext->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO) { stream = _videoFormatContext->streams[i]; break; } } // Use stream to set the context _videoCodecContext = stream->codec; // Find the codec _videoCodec = avcodec_find_decoder(_videoCodecContext->codec_id); if (!_videoCodec) { av_log(_videoCodecContext, AV_LOG_ERROR, "%s","codec not found\n"); exit(-1); } // Open the codec if (avcodec_open(_videoCodecContext, _videoCodec) < 0) { av_log(_videoCodecContext, AV_LOG_ERROR, "%s","could not open codec\n"); exit(-1); } // Read each frame into a packet AVPacket packet; av_init_packet(&packet); int frameCount = 0; while (av_read_frame(_videoFormatContext, &packet) == 0) { // Save the frame AVPacket* tempPacket = new AVPacket(packet); _videoPackets.push_back(tempPacket); } // END reading each frame av_free_packet(&packet); } And here is the function that should write the new video: //--------------------------------------------------------------------- //--------------------------------------------------------------------- //--------------------------------------------------------------------- void FramesCollection::TestVideoEncode() { // Try writing AVFormatContext* outputContext; outputContext = avformat_alloc_context(); outputContext->oformat = av_guess_format(NULL, _videoFormatContext->filename, NULL); char buffer[512]; sprintf(buffer, "C:\\test\\outTest.%s", outputContext->oformat->name); _snprintf_s(outputContext->filename, sizeof(outputContext->filename), strlen(buffer), "%s", buffer); // Add stream if (!av_new_stream(outputContext, 0)) { av_log(outputContext, AV_LOG_ERROR, "%s","Error while creating stream.\n"); exit(-1); } // Open Codec if (avcodec_open(outputContext->streams[0]->codec, _videoCodec) < 0) { av_log(outputContext->streams[0]->codec, AV_LOG_ERROR, "%s","Error while opening codec for context.\n"); exit(-1); } // Set time base, dimensions, etc. outputContext->streams[0]->codec->time_base = _videoCodecContext->time_base; outputContext->streams[0]->codec->width = _videoCodecContext->width; outputContext->streams[0]->codec->height = _videoCodecContext->height; // Some container formats want stream headers to be global if(outputContext->oformat->flags & AVFMT_GLOBALHEADER) { outputContext->streams[0]->codec->flags |= CODEC_FLAG_GLOBAL_HEADER; } // Open output file if (url_fopen(&outputContext->pb, outputContext->filename, URL_WRONLY) < 0) { av_log(outputContext, AV_LOG_ERROR, "%s","Could not open '%s'\n", outputContext->filename); exit(-1); } // Write Header if (avformat_write_header(outputContext, NULL) < 0) { av_log(outputContext, AV_LOG_ERROR, "%s","Error while writing header.\n"); exit(-1); } // Write each packet for (int i = 0; i < (int)_videoPackets.size(); i++) { if (av_write_frame(outputContext, _videoPackets[i]) < 0) { av_log(outputContext, AV_LOG_ERROR, "%s","Error while writing frame.\n"); exit(-1); } } // Write trailer if (av_write_trailer(outputContext) < 0) { av_log(outputContext, AV_LOG_ERROR, "%s","Error while writing trailer.\n"); exit(-1); } // Close stuff avcodec_close(outputContext->streams[0]->codec); url_fclose(outputContext->pb); avformat_free_context(outputContext); } As said, it always crashes while trying to write the packages. Maybe I should also point out that the reading and writing of the audio frames is done in more or less identical functions, and works very well. I have no clue... please help! :( -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/av-write-frame-crashes-without-error-message-tp3836086p3836086.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From hno2005 at gmail.com Fri Sep 23 10:40:25 2011 From: hno2005 at gmail.com (hno2005) Date: Fri, 23 Sep 2011 12:10:25 +0330 Subject: [FFmpeg-user] (no subject) Message-ID: <000001cc79cc$73696cc0$5a3c4640$@com> HI, I got this error when I'm trying to convert 3gp to flv the result file has 0 size .I'm using this command for converting: "ffmpeg.exe -i uploads/v_1.3gp -ar 22050 -ab 32 -f flv -s 320?240 uploads/a.flv" and I got this: FFmpeg version SVN-r12665, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --enable-gpl --enable-postproc --enable-swscale --enable-avfilt er-lavf --enable-pthreads --enable-liba52 --enable-avisynth --enable-libfaac --e nable-libfaad --enable-libgsm --enable-memalign-hack --enable-libmp3lame --enabl e-libnut --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --cpu=i686 --extra-ldflags=-static libavutil version: 49.6.0 libavcodec version: 51.54.0 libavformat version: 52.13.0 libavdevice version: 52.0.0 built on Apr 2 2008 22:35:11, gcc: 4.2.3 Seems stream 0 codec frame rate differs from container frame rate: 29.97 (30000/ 1001) -> 15.00 (15/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'uploads/v_1.3gp': Duration: 00:01:03.6, start: 0.000000, bitrate: 189 kb/s Stream #0.0(und): Video: h263, yuv420p, 176x144 [PAR 12:11 DAR 4:3], 15.00 t b(r) Stream #0.1(und): Audio: samr / 0x726D6173, 8000 Hz, mono WARNING: The bitrate parameter is set too low. It takes bits/s as argument, not kbits/s Output #0, flv, to 'uploads/a.flv': Stream #0.0(und): Video: flv, yuv420p, 320x240 [PAR 1:1 DAR 4:3], q=2-31, 20 0 kb/s, 15.00 tb(c) Stream #0.1(und): Audio: libmp3lame, 22050 Hz, mono, 0 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Unsupported codec (id=73728) for input stream #0.1 Can anybody helps me please? Thanks From etienne.buira.lists at free.fr Fri Sep 23 10:54:36 2011 From: etienne.buira.lists at free.fr (Etienne Buira) Date: Fri, 23 Sep 2011 10:54:36 +0200 Subject: [FFmpeg-user] Unsupported codec (was (no subject)) In-Reply-To: <000001cc79cc$73696cc0$5a3c4640$@com> References: <000001cc79cc$73696cc0$5a3c4640$@com> Message-ID: <20110923085436.GM2506@epicure.lazyet.homelinux.net> Hi. On Fri, Sep 23, 2011 at 12:10:25PM +0330, hno2005 wrote: > "ffmpeg.exe -i uploads/v_1.3gp -ar 22050 -ab 32 -f flv -s 320?240 > uploads/a.flv" > > FFmpeg version SVN-r12665, Copyright (c) 2000-2008 Fabrice Bellard, et al. Your version is years old. Try with git version (alternatively release). > Stream #0.1(und): Audio: samr / 0x726D6173, 8000 Hz, mono Don't know about this particular codec, but I know amr have been a often discussed topic on development list over last months, again, try with a new version. > WARNING: The bitrate parameter is set too low. It takes bits/s as argument, > not > kbits/s This warning makes sense. > Unsupported codec (id=73728) for input stream #0.1 From hno2005 at gmail.com Fri Sep 23 11:07:18 2011 From: hno2005 at gmail.com (hno2005) Date: Fri, 23 Sep 2011 12:37:18 +0330 Subject: [FFmpeg-user] Unsupported codec (was (no subject)) In-Reply-To: <20110923085436.GM2506@epicure.lazyet.homelinux.net> References: <000001cc79cc$73696cc0$5a3c4640$@com> <20110923085436.GM2506@epicure.lazyet.homelinux.net> Message-ID: <000801cc79d0$33d04f30$9b70ed90$@com> Actually I use 32k for amr and It was just a test. Is there any way that I can find information about this codec? I try that In my local xampp and It's not working in my host.Is there any way that I can get this return in my host using php? Thanks -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Etienne Buira Sent: Friday, September 23, 2011 12:25 PM To: ffmpeg-user at ffmpeg.org Subject: Re: [FFmpeg-user] Unsupported codec (was (no subject)) Hi. On Fri, Sep 23, 2011 at 12:10:25PM +0330, hno2005 wrote: > "ffmpeg.exe -i uploads/v_1.3gp -ar 22050 -ab 32 -f flv -s 320?240 > uploads/a.flv" > > FFmpeg version SVN-r12665, Copyright (c) 2000-2008 Fabrice Bellard, et al. Your version is years old. Try with git version (alternatively release). > Stream #0.1(und): Audio: samr / 0x726D6173, 8000 Hz, mono Don't know about this particular codec, but I know amr have been a often discussed topic on development list over last months, again, try with a new version. > WARNING: The bitrate parameter is set too low. It takes bits/s as argument, > not > kbits/s This warning makes sense. > Unsupported codec (id=73728) for input stream #0.1 _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From drabner at zoobe.com Fri Sep 23 13:19:18 2011 From: drabner at zoobe.com (Drabner) Date: Fri, 23 Sep 2011 04:19:18 -0700 (PDT) Subject: [FFmpeg-user] Access Violation in avcodec_encode_video Message-ID: <1316776758671-3836503.post@n4.nabble.com> I am trying to write frames into an output file (mp4 container, codec is h264). This is the code that always gets an Access Violation: /avcodec_encode_video(outputFormatContext->streams[0]->codec, outbuf, outbufSize, frame);/ outbuf is correctly allocated with av_malloc(outbufSize) with outbufSize being FF_MIN_BUFFER_SIZE. But all sizes bigger than that also cause the same error. frame is an AVFrame* that was filled by sws_scale, converting to PIX_FMT_YUV420P. The format context (and the codec context) is also correctly initialized and opened with the correct codec. Up to the point with the access violation crash, ffmpeg reports absolutely no errors or warnings, so I guess I did everything correct. Any ideas what could go wrong here? Here is the complete code (_videoFrames is a vector filled with previously decoded AVFrames, the alloc_picture function is at the bottom): ------------------------------------------------------- ------------------------------------------------------- void FramesCollection::TestVideoEncode() { av_log(NULL, AV_LOG_ERROR, "%s","Start video encoding.\n"); // Try writing AVFormatContext* outputContext; outputContext = avformat_alloc_context(); outputContext->oformat = av_guess_format("mp4", NULL, NULL);//_videoFormatContext->filename, NULL); char buffer[512]; sprintf(buffer, "C:\\zoobe\\outTest_video.%s", outputContext->oformat->name); _snprintf_s(outputContext->filename, sizeof(outputContext->filename), strlen(buffer), "%s", buffer); // Add stream if (!av_new_stream(outputContext, 0)) { av_log(outputContext, AV_LOG_ERROR, "%s","Error while creating stream.\n"); exit(-1); } // Find Codec AVCodec* outCodec = avcodec_find_decoder(CODEC_ID_H264); if (!outCodec) { av_log(outputContext, AV_LOG_ERROR, "%s","codec not found\n"); exit(-1); } // Open Codec if (avcodec_open(outputContext->streams[0]->codec, outCodec) < 0) { av_log(outputContext->streams[0]->codec, AV_LOG_ERROR, "%s","Error while opening codec for context.\n"); exit(-1); } // Open Codec /*if (avcodec_open(outputContext->streams[0]->codec, _videoCodec) < 0) { av_log(outputContext->streams[0]->codec, AV_LOG_ERROR, "%s","Error while opening codec for context.\n"); exit(-1); }*/ // Set time base, dimensions, etc. outputContext->streams[0]->codec->time_base = _videoCodecContext->time_base; outputContext->streams[0]->codec->width = _videoCodecContext->width; outputContext->streams[0]->codec->height = _videoCodecContext->height; // Some container formats want stream headers to be global if(outputContext->oformat->flags & AVFMT_GLOBALHEADER) { outputContext->streams[0]->codec->flags |= CODEC_FLAG_GLOBAL_HEADER; } // Open output file if (url_fopen(&outputContext->pb, outputContext->filename, URL_WRONLY) < 0) { av_log(outputContext, AV_LOG_ERROR, "%s","Could not open '%s'\n", outputContext->filename); exit(-1); } // Write Header if (avformat_write_header(outputContext, NULL) < 0) { av_log(outputContext, AV_LOG_ERROR, "%s","Error while writing header.\n"); exit(-1); } // Write each frame int outbufSize = FF_MIN_BUFFER_SIZE; uint8_t* outbuf = new uint8_t[outbufSize]; for (int i = 0; i < (int)_videoFrames.size(); i++) { // Convert the frame to correct format SwsContext* convertContext = sws_getContext( _videoFrames[i]->width, _videoFrames[i]->height, _videoCodecContext->pix_fmt, _videoFrames[i]->width, _videoFrames[i]->height, PIX_FMT_YUV420P, SWS_BICUBIC, NULL, NULL, NULL ); if (!convertContext) { av_log(outputContext->streams[0]->codec, AV_LOG_ERROR, "%s","Cannot initialize the conversion context\n"); exit(1); } AVFrame *frame; frame = alloc_picture(PIX_FMT_YUV420P, _videoFrames[i]->width, _videoFrames[i]->height); int outSize = sws_scale(convertContext, _videoFrames[i]->data, _videoFrames[i]->linesize, 0, _videoFrames[i]->height, frame->data, frame->linesize); if (outSize <= 0) { av_log(outputContext, AV_LOG_ERROR, "%s","Error while scaling frame.\n"); exit(-1); } // Encode the frame av_log(outputContext, AV_LOG_ERROR, "%s %i \n","Writing frame:",i); outSize = avcodec_encode_video( outputContext->streams[0]->codec, outbuf, outbufSize, frame); if (outSize < 0) { av_log(outputContext, AV_LOG_ERROR, "%s","Error while encoding frame.\n"); exit(-1); } else if (outSize > 0) { // Write the frame AVPacket pkt; av_init_packet(&pkt); if (outputContext->streams[0]->codec->coded_frame->pts != AV_NOPTS_VALUE) { pkt.pts = av_rescale_q(outputContext->streams[0]->codec->coded_frame->pts, outputContext->streams[0]->codec->time_base, outputContext->streams[0]->time_base); } if(outputContext->streams[0]->codec->coded_frame->key_frame) pkt.flags |= AV_PKT_FLAG_KEY; pkt.stream_index = outputContext->streams[0]->index; pkt.data = outbuf; pkt.size = outSize; int ret = av_write_frame(outputContext, &pkt); if (ret < 0) { av_log(outputContext, AV_LOG_ERROR, "%s","Error while writing frame.\n"); exit(-1); } } } // Write trailer if (av_write_trailer(outputContext) < 0) { av_log(outputContext, AV_LOG_ERROR, "%s","Error while writing trailer.\n"); exit(-1); } // Close stuff avcodec_close(outputContext->streams[0]->codec); url_fclose(outputContext->pb); avformat_free_context(outputContext); } ------------------------------------------------------- ------------------------------------------------------- AVFrame *alloc_picture(enum PixelFormat pix_fmt, int width, int height ) { AVFrame *picture; uint8_t *picture_buf; int size; picture = avcodec_alloc_frame(); if (!picture) return NULL; size = avpicture_get_size(pix_fmt, width, height); picture_buf = (uint8_t *) av_malloc(size); if (!picture_buf) { av_free(picture); return NULL; } avpicture_fill((AVPicture *)picture, picture_buf, pix_fmt, width, height); return picture; } -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Access-Violation-in-avcodec-encode-video-tp3836503p3836503.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From gagandeep.bawa at ironroad.com Fri Sep 23 15:45:26 2011 From: gagandeep.bawa at ironroad.com (Gagandeep Bawa) Date: Fri, 23 Sep 2011 19:15:26 +0530 Subject: [FFmpeg-user] non monotonically increasing dts to muxer in stream Message-ID: Hi Guys, I am getting following error when i try to convert .mov file to .mp4 in IPhone using 0.8.4 build libraries. Please help. It seems problem in setting dts and pts. Find below my code which i am using to write audio frames. The output video is having less duration (8-10 sec) and in start there are some green frames. Console Output: ** *Output #0, mp4, to '/var/mobile/Applications/20A7BB04-C8E0-4357-ACF6-28BE57C72FED/Documents/FFMPeg.mp4': * * Stream #0.0: Video: mpeg4, yuv420p, 426x320, q=2-31, 384 kb/s, 90k tbn, 25 tbc Stream #0.1: Audio: aac, 44100 Hz, 1 channels, s16, 63 kb/s 2011-09-23 18:34:02.223 iFrameExtractor[3036:707] Audio-pts ...............0.000000 2011-09-23 18:34:02.227 iFrameExtractor[3036:707] Video-pts ...............0.000000 2011-09-23 18:34:02.268 iFrameExtractor[3036:707] Audio-pts ...............0.000000 2011-09-23 18:34:02.273 iFrameExtractor[3036:707] Video-pts ...............0.040000 2011-09-23 18:34:02.279 iFrameExtractor[3036:707] Audio-pts ...............0.023220 2011-09-23 18:34:02.283 iFrameExtractor[3036:707] Video-pts ...............0.040000 2011-09-23 18:34:02.286 iFrameExtractor[3036:707] Audio pts ...............0.046440 2011-09-23 18:34:02.289 iFrameExtractor[3036:707] Video pts ...............0.040000 2011-09-23 18:34:02.307 iFrameExtractor[3036:707] Audio-pts ...............0.046440 2011-09-23 18:34:02.310 iFrameExtractor[3036:707] Video-pts ...............0.080000 [mp4 @ 0x11fc400] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 2048 >= 2048 Error while writing audio frame 2011-09-23 18:34:02.327 iFrameExtractor[3036:707] Audio-pts ...............0.069660 2011-09-23 18:34:02.330 iFrameExtractor[3036:707] Video-pts ...............0.080000 2011-09-23 18:34:02.339 iFrameExtractor[3036:707] Audio pts ...............0.092880 2011-09-23 18:34:02.342 iFrameExtractor[3036:707] Video pts ...............0.080000 2011-09-23 18:34:02.356 iFrameExtractor[3036:707] Audio-pts ...............0.092880 2011-09-23 18:34:02.359 iFrameExtractor[3036:707] Video-pts ...............0.120000 [mp4 @ 0x11fc400] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 4096 >= 4096 Error while writing audio frame 2011-09-23 18:34:02.374 iFrameExtractor[3036:707] Audio-pts ...............0.116100 2011-09-23 18:34:02.377 iFrameExtractor[3036:707] Video-pts ...............0.120000 2011-09-23 18:34:02.385 iFrameExtractor[3036:707] Audio pts ...............0.139320 2011-09-23 18:34:02.388 iFrameExtractor[3036:707] Video pts ...............0.120000 2011-09-23 18:34:02.397 iFrameExtractor[3036:707] Audio-pts ...............0.139320 2011-09-23 18:34:02.401 iFrameExtractor[3036:707] Video-pts ...............0.160000 [mp4 @ 0x11fc400] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 6144 >= 6144 Error while writing audio frame 2011-09-23 18:34:02.415 iFrameExtractor[3036:707] Audio pts ...............0.162540 * Code To Write Audio Frames : -(bool) write_audio_frame:(AVFormatContext *)oc: (AVStream *)st{ AVPacket pkt; av_init_packet(&pkt); int ss,ret; c = st->codec; while( av_fifo_size(fifo) >= frame_bytes ) { av_fifo_generic_read( fifo, samples, frame_bytes, NULL ); ss = avcodec_encode_audio(c, audio_outbuf, frame_bytes, samples); if (ss > 0){ pkt.size = ss; if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE){ pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base); //pkt.pts = av_rescale_q(packetNew.dts, pFormatCtx->streams[audioStream]->time_base, st->time_base); } if(c->coded_frame && c->coded_frame->key_frame) pkt.flags |= AV_PKT_FLAG_KEY; pkt.stream_index= st->index; pkt.data= audio_outbuf; ret = av_interleaved_write_frame(oc, &pkt); } else { ret = 0; } } av_free_packet(&pkt); if (ret != 0) { fprintf(stderr, "Error while writing audio frame\n"); //exit(1); } return true; } -- Thanks & Regards, Gagandeep From dashing.meng at gmail.com Fri Sep 23 16:44:00 2011 From: dashing.meng at gmail.com (littlebat) Date: Fri, 23 Sep 2011 22:44:00 +0800 Subject: [FFmpeg-user] (no subject) In-Reply-To: <000001cc79cc$73696cc0$5a3c4640$@com> References: <000001cc79cc$73696cc0$5a3c4640$@com> Message-ID: <20110923224400.d27c937a.dashing.meng@gmail.com> On Fri, 23 Sep 2011 12:10:25 +0330 "hno2005" wrote: > HI, > > I got this error when I'm trying to convert 3gp to flv the result > file has 0 size .I'm using this command for converting: > > > > "ffmpeg.exe -i uploads/v_1.3gp -ar 22050 -ab 32 -f flv -s 320?240 > uploads/a.flv" > > > > and I got this: > > > > FFmpeg version SVN-r12665, Copyright (c) 2000-2008 Fabrice Bellard, > et al. > > configuration: --enable-gpl --enable-postproc --enable-swscale > --enable-avfilt > > er-lavf --enable-pthreads --enable-liba52 --enable-avisynth > --enable-libfaac > --e > > nable-libfaad --enable-libgsm --enable-memalign-hack > --enable-libmp3lame > --enabl > > e-libnut --enable-libtheora --enable-libvorbis --enable-libx264 > --enable-libxvid > > --cpu=i686 --extra-ldflags=-static > > libavutil version: 49.6.0 > > libavcodec version: 51.54.0 > > libavformat version: 52.13.0 > > libavdevice version: 52.0.0 > > built on Apr 2 2008 22:35:11, gcc: 4.2.3 > > > > Seems stream 0 codec frame rate differs from container frame rate: > 29.97 (30000/ > > 1001) -> 15.00 (15/1) > > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'uploads/v_1.3gp': > > Duration: 00:01:03.6, start: 0.000000, bitrate: 189 kb/s > > Stream #0.0(und): Video: h263, yuv420p, 176x144 [PAR 12:11 DAR > 4:3], 15.00 t > > b(r) > > Stream #0.1(und): Audio: samr / 0x726D6173, 8000 Hz, mono > > WARNING: The bitrate parameter is set too low. It takes bits/s as > argument, not > > kbits/s > Output #0, flv, to 'uploads/a.flv': > > Stream #0.0(und): Video: flv, yuv420p, 320x240 [PAR 1:1 DAR 4:3], > q=2-31, 20 > > 0 kb/s, 15.00 tb(c) > > Stream #0.1(und): Audio: libmp3lame, 22050 Hz, mono, 0 kb/s > > Stream mapping: > > Stream #0.0 -> #0.0 > > Stream #0.1 -> #0.1 > > Unsupported codec (id=73728) for input stream #0.1 > Try replace "-ab 32" with "-ab 32k", and, it seems missing amr decoder in your system. From mboorshtein at gmail.com Fri Sep 23 17:23:46 2011 From: mboorshtein at gmail.com (Marc Boorshtein) Date: Fri, 23 Sep 2011 11:23:46 -0400 Subject: [FFmpeg-user] Floating point exception when converting ogg to mp4 Message-ID: All, I'm trying to convert an ogv of a desktop recording to mp4. If I do something really basic: $ ffmpeg -i out.ogv test.mp4 it encodes, but the video is really bad. I tried using one of the lossless presets: $ ffmpeg -i ~/out.ogv -vpre libx264-lossless_fast test.mp4 ffmpeg version git-snapshot-oldabi-20110612-RPMFusion, Copyright (c) 2000-2011 the FFmpeg developers built on Jul 1 2011 17:40:35 with gcc 4.6.0 20110509 (Red Hat 4.6.0-7) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --enable-bzlib --enable-libdc1394 --enable-libdirac --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect libavutil 50. 41. 0 / 50. 41. 0 libavcodec 52.121. 0 / 52.121. 0 libavformat 52.108. 0 / 52.108. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 78. 0 / 1. 78. 0 libswscale 0. 14. 0 / 0. 14. 0 libpostproc 51. 2. 0 / 51. 2. 0 [ogg @ 0xf69920] max_analyze_duration reached Input #0, ogg, from '/home/mlb/out.ogv': Duration: 00:00:25.46, start: 0.000000, bitrate: 1286 kb/s Stream #0.0: Data: skeleton Stream #0.1: Video: theora, yuv420p, 1354x768 [PAR 1:1 DAR 677:384], 15 fps, 15 tbr, 15 tbn, 15 tbc Stream #0.2: Audio: vorbis, 22050 Hz, mono, s16, 89 kb/s File 'test.mp4' already exists. Overwrite ? [y/N] y [buffer @ 0xf72820] w:1354 h:768 pixfmt:yuv420p tb:1/1000000 sar:1/1 Floating point exception (core dumped) But its giving me the floating point exception. Running on fedora 15 on an intel i7 quadcore with 8gb of ram. Any help would be greatly appreciated. Thanks Marc From gagandeep.bawa at ironroad.com Fri Sep 23 12:52:29 2011 From: gagandeep.bawa at ironroad.com (Gagandeep Bawa) Date: Fri, 23 Sep 2011 16:22:29 +0530 Subject: [FFmpeg-user] Error in compiling FFMPEG 0.8.4 for IPhone OS 4.3 SDK Message-ID: Hi, I am trying to compile new FFMPEG "Love" 0.8.4 version for ios 4.3 sdk but it fails. Please have a look in to this issue and let me know if i am doing anything wrong. Please find attached "Build Run.rtf" file which shows the error which i am getting after trying to compile library. Also find attached my two build script files i tried with both batch file but it doesn't work (don't confuse with name ....working). I am using gas-preprocessor.pl file which i get from following link to build for IPhone but it doesn't help. I am trying this from so many hours with no luck, so disappointing. Can you please let me know that whether current FFMPEG 0.8.4 libraries are compatible and stable with IPhone 4 (ios 4.3 sdk) or not, because i want to compress .mov file to .mp4 in IPhone 4G device itself through my IPhone application. https://github.com/yuvi/gas-preprocessor -- Thanks & Regards, Gagandeep -------------- next part -------------- A non-text attachment was scrubbed... Name: Build Run.rtf Type: application/rtf Size: 3654 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: build_armv7 Type: application/octet-stream Size: 558 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: build_armv7_working Type: application/octet-stream Size: 877 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: config.log Type: application/octet-stream Size: 73791 bytes Desc: not available URL: From phil.eki at gmail.com Thu Sep 22 20:20:44 2011 From: phil.eki at gmail.com (Philip Iyobose Eki) Date: Thu, 22 Sep 2011 12:20:44 -0600 Subject: [FFmpeg-user] Converting to VP6 FLV Message-ID: Hey guys, does anyone know how to convert to VP6 (FLV) using ffmpeg? It can't be sorenson. I checked the codec list and VP6 doesn't seem to be encodable. What would one need to to do make it encodable? Thanks! Philip Eki From kistlerjm9 at yahoo.com Fri Sep 23 00:46:28 2011 From: kistlerjm9 at yahoo.com (kistlerjm9) Date: Thu, 22 Sep 2011 15:46:28 -0700 (PDT) Subject: [FFmpeg-user] ffmpeg converting Message-ID: <1316731588057-3835161.post@n4.nabble.com> is there a way to have ffmpeg convert 3 different flv files at once. I am setting up a flash player with the option of 320, 480, and hq bitrates aka 800, 1200, 2000k. I have been told you have to have 3 seperate flv files of the same video each converted with the seperate bitrates. Any help would be appreciated. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-converting-tp3835161p3835161.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From l.jozsa at gmail.com Fri Sep 23 20:47:52 2011 From: l.jozsa at gmail.com (=?UTF-8?Q?Ladislav_J=C3=B3zsa?=) Date: Fri, 23 Sep 2011 20:47:52 +0200 Subject: [FFmpeg-user] Delaying audio and video stream Message-ID: Hi guys, I have a video file with one audio track already attached to it and its perfectly in sync with video. Then I have another audio track which I need to add to the video as second audio track. But this audio track is ahead of the audio/video stream so I need to delay it. When I use the following command ffmpeg -threads 4 -y -i ./DriveAng_CZE_2.pal.flac -itsoffset 1.2 -i ./DriveAngry_3D.mpg -map 1:0 -map 0:0 -vcodec mpeg2video -b 8192k -minrate 8192k -maxrate 8192k -bufsize 1835k -acodec mp2 -ab 192k -metadata:s:0 language=ces -acodec ac3 -ab 192k -metadata:s:0 language=eng ./DriveAngry_3D.mixed.ts I get second audio track perfectly synced with the video but the first audio track which originally came with video is out of sync. It is ahead of video. Apparently, the command above has delayed only the video stream. Then I tried ffmpeg -threads 4 -y -i ./DriveAng_CZE_2.pal.flac -itsoffset 1.2 -i ./DriveAngry_3D.mpg -itsoffset 1.2 -i ./DriveAngry_3D.mpg -map 1:0 -map 0:0 -map 2:1 -vcodec mpeg2video -b 8192k -minrate 8192k -maxrate 8192k -bufsize 1835k -acodec mp2 -ab 192k -metadata:s:0 language=ces -acodec ac3 -ab 192k -metadata:s:0 language=eng ./DriveAngry_3D.mixed.ts but it didn't help. My ffmpeg version is ffmpeg version N-32816-g3f7dc48, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 23 2011 19:42:22 with gcc 4.6.1 configuration: --enable-filter=movie libavutil 51. 17. 0 / 51. 17. 0 libavcodec 53. 17. 0 / 53. 17. 0 libavformat 53. 13. 0 / 53. 13. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 43. 3 / 2. 43. 3 libswscale 2. 1. 0 / 2. 1. 0 Is there a way how to delay audio and video file at once? Or do I need to separate the video first and go for two pass audio sync (first sync Czech audio with the delayed video only and then add original audio stream)? Thanks Ladislav From listes at defendingthemusic.fr Fri Sep 23 21:20:52 2011 From: listes at defendingthemusic.fr (Paul-Antoine Arras) Date: Fri, 23 Sep 2011 21:20:52 +0200 Subject: [FFmpeg-user] Aspect ratio preservation in WMV Message-ID: <4E7CDC14.3070809@defendingthemusic.fr> Hello, I'm trying to convert the sound of a video from WMA to MP3. To do so, I use the following command line: ffmpeg -i input.wmv -acodec libmp3lame -aq 4 -vcodec copy output.wmv The sound is converted correctly but the aspect ratio (AR) becomes 4:3 instead of 16:9. Then I've tried to force the AR with: -aspect 16:9 without success. Why does ffmpeg modify the AR while all it has to do in this case is a simple copy of the video part? How to make it preserve the original AR (16:9)? Thanks in advance for your help. -- Paul-Antoine ARRAS From dmc at sunrise.ch Fri Sep 23 22:34:46 2011 From: dmc at sunrise.ch (dmc at sunrise.ch) Date: Fri, 23 Sep 2011 22:34:46 +0200 Subject: [FFmpeg-user] how to concatenate multiple .ts files as input? Message-ID: Hi all I have several MPEG2 transport stream files "file_n.ts" (1<=n<=N) from a PVR and would like to avoid merging them before passing them through ffmpeg. How can I submit them as a joint input argument? Thanks From rodney.baker at iinet.net.au Sat Sep 24 04:17:11 2011 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Sat, 24 Sep 2011 11:47:11 +0930 Subject: [FFmpeg-user] Aspect ratio preservation in WMV In-Reply-To: <4E7CDC14.3070809@defendingthemusic.fr> References: <4E7CDC14.3070809@defendingthemusic.fr> Message-ID: <201109241147.11523.rodney.baker@iinet.net.au> On Sat, 24 Sep 2011 04:50:52 Paul-Antoine Arras wrote: > Hello, > > I'm trying to convert the sound of a video from WMA to MP3. To do so, I > use the following command line: > > ffmpeg -i input.wmv -acodec libmp3lame -aq 4 -vcodec copy output.wmv > > The sound is converted correctly but the aspect ratio (AR) becomes 4:3 > instead of 16:9. Then I've tried to force the AR with: > > -aspect 16:9 > > without success. > > Why does ffmpeg modify the AR while all it has to do in this case is a > simple copy of the video part? How to make it preserve the original AR > (16:9)? > > Thanks in advance for your help. You need to provide the full output from ffmpeg. -- =================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au =================================================== From hno2005 at gmail.com Sat Sep 24 07:49:43 2011 From: hno2005 at gmail.com (hno2005) Date: Sat, 24 Sep 2011 09:19:43 +0330 Subject: [FFmpeg-user] (no subject) In-Reply-To: <20110923224400.d27c937a.dashing.meng@gmail.com> References: <000001cc79cc$73696cc0$5a3c4640$@com> <20110923224400.d27c937a.dashing.meng@gmail.com> Message-ID: <000001cc7a7d$d0b2c820$72185860$@com> Yes I have tried that but the problem didn't solved.What can be the problem? -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of littlebat Sent: Friday, September 23, 2011 6:14 PM To: FFmpeg user questions and RTFMs Subject: Re: [FFmpeg-user] (no subject) On Fri, 23 Sep 2011 12:10:25 +0330 "hno2005" wrote: > HI, > > I got this error when I'm trying to convert 3gp to flv the result > file has 0 size .I'm using this command for converting: > > > > "ffmpeg.exe -i uploads/v_1.3gp -ar 22050 -ab 32 -f flv -s 320?240 > uploads/a.flv" > > > > and I got this: > > > > FFmpeg version SVN-r12665, Copyright (c) 2000-2008 Fabrice Bellard, > et al. > > configuration: --enable-gpl --enable-postproc --enable-swscale > --enable-avfilt > > er-lavf --enable-pthreads --enable-liba52 --enable-avisynth > --enable-libfaac > --e > > nable-libfaad --enable-libgsm --enable-memalign-hack > --enable-libmp3lame > --enabl > > e-libnut --enable-libtheora --enable-libvorbis --enable-libx264 > --enable-libxvid > > --cpu=i686 --extra-ldflags=-static > > libavutil version: 49.6.0 > > libavcodec version: 51.54.0 > > libavformat version: 52.13.0 > > libavdevice version: 52.0.0 > > built on Apr 2 2008 22:35:11, gcc: 4.2.3 > > > > Seems stream 0 codec frame rate differs from container frame rate: > 29.97 (30000/ > > 1001) -> 15.00 (15/1) > > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'uploads/v_1.3gp': > > Duration: 00:01:03.6, start: 0.000000, bitrate: 189 kb/s > > Stream #0.0(und): Video: h263, yuv420p, 176x144 [PAR 12:11 DAR > 4:3], 15.00 t > > b(r) > > Stream #0.1(und): Audio: samr / 0x726D6173, 8000 Hz, mono > > WARNING: The bitrate parameter is set too low. It takes bits/s as > argument, not > > kbits/s > Output #0, flv, to 'uploads/a.flv': > > Stream #0.0(und): Video: flv, yuv420p, 320x240 [PAR 1:1 DAR 4:3], > q=2-31, 20 > > 0 kb/s, 15.00 tb(c) > > Stream #0.1(und): Audio: libmp3lame, 22050 Hz, mono, 0 kb/s > > Stream mapping: > > Stream #0.0 -> #0.0 > > Stream #0.1 -> #0.1 > > Unsupported codec (id=73728) for input stream #0.1 > Try replace "-ab 32" with "-ab 32k", and, it seems missing amr decoder in your system. _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From dashing.meng at gmail.com Sat Sep 24 09:14:28 2011 From: dashing.meng at gmail.com (littlebat) Date: Sat, 24 Sep 2011 15:14:28 +0800 Subject: [FFmpeg-user] (no subject) In-Reply-To: <000001cc7a7d$d0b2c820$72185860$@com> References: <000001cc79cc$73696cc0$5a3c4640$@com> <20110923224400.d27c937a.dashing.meng@gmail.com> <000001cc7a7d$d0b2c820$72185860$@com> Message-ID: <20110924151428.16f24957.dashing.meng@gmail.com> On Sat, 24 Sep 2011 09:19:43 +0330 "hno2005" wrote: > > > > Unsupported codec (id=73728) for input stream #0.1 > > Missing amr decoder in your system. But I have no a Windows FFmpeg to tell you how to add it. From br at bitberry.com Sat Sep 24 22:36:03 2011 From: br at bitberry.com (Brian Rasmusson) Date: Sat, 24 Sep 2011 22:36:03 +0200 Subject: [FFmpeg-user] libfaac Message-ID: <4E7E3F33.20506@bitberry.com> Hi, I read a year old thread that libfaac is not compliant with the GPL license, and that the FFmpeg team is working on their own AAC encoder. What is the status on this? Has it reached a stage where the quality can match libfaac? Sorry if this is a repeat question, I didn't find the answer in the forum archive. Brian From listes at defendingthemusic.fr Sun Sep 25 18:48:34 2011 From: listes at defendingthemusic.fr (Paul-Antoine Arras) Date: Sun, 25 Sep 2011 18:48:34 +0200 Subject: [FFmpeg-user] Aspect ratio preservation in WMV In-Reply-To: <201109241147.11523.rodney.baker@iinet.net.au> References: <4E7CDC14.3070809@defendingthemusic.fr> <201109241147.11523.rodney.baker@iinet.net.au> Message-ID: <4E7F5B62.3030400@defendingthemusic.fr> Le 24/09/2011 04:17, Rodney Baker a ?crit : > On Sat, 24 Sep 2011 04:50:52 Paul-Antoine Arras wrote: >> Hello, >> >> I'm trying to convert the sound of a video from WMA to MP3. To do so, I >> use the following command line: >> >> ffmpeg -i input.wmv -acodec libmp3lame -aq 4 -vcodec copy output.wmv >> >> The sound is converted correctly but the aspect ratio (AR) becomes 4:3 >> instead of 16:9. Then I've tried to force the AR with: >> >> -aspect 16:9 >> >> without success. >> >> Why does ffmpeg modify the AR while all it has to do in this case is a >> simple copy of the video part? How to make it preserve the original AR >> (16:9)? >> >> Thanks in advance for your help. > > You need to provide the full output from ffmpeg. > Here is the full output from ffmpeg: FFmpeg version 0.6.2-4:0.6.2-1ubuntu1.1, Copyright (c) 2000-2010 the Libav developers built on Sep 16 2011 16:57:46 with gcc 4.5.2 configuration: --extra-version=4:0.6.2-1ubuntu1.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --disable-stripping --enable-runtime-cpudetect --enabl-libdirac --enable-libfaad --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libavcodec configuration: --extra-version=4:0.6.2-1ubuntu2 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libdirac --enable-libgsm --enable-libopenjpeg --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --disable-stripping --enable-runtime-cpudetect --enable-vaapi --enable-libopenjpeg --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libfaad --enable-libdirac --enable-libfaad --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libavformat configuration: --extra-version=4:0.6.2-1ubuntu1.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --disable-stripping --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libavdevice configuration: --extra-version=4:0.6.2-1ubuntu1.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --disable-stripping --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libavfilter configuration: --extra-version=4:0.6.2-1ubuntu1.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --disable-stripping --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libswscale configuration: --extra-version=4:0.6.2-1ubuntu1.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --disable-stripping --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libpostproc configuration: --extra-version=4:0.6.2-1ubuntu1.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --disable-stripping --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.19. 0 / 1.19. 0 libswscale 0.11. 0 / 0.11. 0 libpostproc 51. 2. 0 / 51. 2. 0 [wmv3 @ 0x80f41f0]Extra data: 8 bits left, value: 0 Seems stream 1 codec frame rate differs from container frame rate: 1000.00 (1000/1) -> 25.00 (25/1) Input #0, asf, from '[NoDRM]-Les Guignols de l'info Episode 14 - Les Guignols de l'info (prod7600121mpp6Id).wmv': Metadata: WMFSDKVersion : 12.0.7600.16385 WMFSDKNeeded : 0.0.0.0000 IsVBR : 0 title : author : copyright : comment : Duration: 00:07:46.51, start: 3.000000, bitrate: 1494 kb/s Stream #0.0: Audio: wmav2, 48000 Hz, 2 channels, s16, 128 kb/s Stream #0.1: Video: wmv3, yuv420p, 720x576, 25 tbr, 1k tbn, 1k tbc Output #0, asf, to 'output.wmv': Metadata: WM/EncodingSettings: Lavf52.64.2 Stream #0.0: Video: WMV3 / 0x33564D57, yuv420p, 720x576 [PAR 64:45 DAR 16:9], q=2-31, 1k tbn, 1k tbc Stream #0.1: Audio: libmp3lame, 48000 Hz, 2 channels, s16, 64 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Press [q] to stop encoding frame=11663 fps=808 q=-1.0 Lsize= 87185kB time=466.44 bitrate=1531.2kbits/s video:76705kB audio:8558kB global headers:0kB muxing overhead 2.253658% *** Thanks -- Paul-Antoine ARRAS From rodney.baker at iinet.net.au Mon Sep 26 01:23:22 2011 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Mon, 26 Sep 2011 08:53:22 +0930 Subject: [FFmpeg-user] Aspect ratio preservation in WMV In-Reply-To: <4E7F5B62.3030400@defendingthemusic.fr> References: <4E7CDC14.3070809@defendingthemusic.fr> <201109241147.11523.rodney.baker@iinet.net.au> <4E7F5B62.3030400@defendingthemusic.fr> Message-ID: <201109260853.22804.rodney.baker@iinet.net.au> On Mon, 26 Sep 2011 02:18:34 Paul-Antoine Arras wrote: > Le 24/09/2011 04:17, Rodney Baker a ?crit : > > On Sat, 24 Sep 2011 04:50:52 Paul-Antoine Arras wrote: > >> Hello, > >> > >> I'm trying to convert the sound of a video from WMA to MP3. To do so, I > >> > >> use the following command line: > >> ffmpeg -i input.wmv -acodec libmp3lame -aq 4 -vcodec copy > >> output.wmv > >> > >> The sound is converted correctly but the aspect ratio (AR) becomes 4:3 > >> > >> instead of 16:9. Then I've tried to force the AR with: > >> -aspect 16:9 > >> > >> without success. > >> > >> Why does ffmpeg modify the AR while all it has to do in this case is a > >> simple copy of the video part? How to make it preserve the original AR > >> (16:9)? > >> > >> Thanks in advance for your help. > > > > You need to provide the full output from ffmpeg. > > Here is the full output from ffmpeg: > > FFmpeg version 0.6.2-4:0.6.2-1ubuntu1.1, Copyright (c) 2000-2010 the > Libav developers [...] > Output #0, asf, to 'output.wmv': > Metadata: > WM/EncodingSettings: Lavf52.64.2 > Stream #0.0: Video: WMV3 / 0x33564D57, yuv420p, 720x576 [PAR 64:45 > DAR 16:9], q=2-31, 1k tbn, 1k tbc ^^^^^^^^^^^^ ffmpeg is writing the stream out with an aspect ratio of 16:9 (Display Aspect Ratio) and pixel aspect ratio of 64:45 which, with a standard definition 720x576 picture will be correct for a 16:9 display. What are you using to view the resultant output file? Is it showing the correct aspect ratio? Can you force it? -- =================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au =================================================== From dashing.meng at gmail.com Mon Sep 26 02:41:20 2011 From: dashing.meng at gmail.com (littlebat) Date: Mon, 26 Sep 2011 08:41:20 +0800 Subject: [FFmpeg-user] libfaac In-Reply-To: <4E7E3F33.20506@bitberry.com> References: <4E7E3F33.20506@bitberry.com> Message-ID: <20110926084120.7ee0c543.dashing.meng@gmail.com> On Sat, 24 Sep 2011 22:36:03 +0200 Brian Rasmusson
wrote: > Hi, > > I read a year old thread that libfaac is not compliant with the GPL > license, and that the FFmpeg team is working on their own AAC encoder. > > What is the status on this? Has it reached a stage where the quality > can match libfaac? > > Sorry if this is a repeat question, I didn't find the answer in the > forum archive. I have tested on converting some videos to mp4(mpeg4+ native ffmpeg aac) for my mobilephone. It works ok. I sugguest you do some test on you work then report here. From lou at lrcd.com Mon Sep 26 02:51:22 2011 From: lou at lrcd.com (Lou) Date: Sun, 25 Sep 2011 16:51:22 -0800 Subject: [FFmpeg-user] libfaac In-Reply-To: <4E7E3F33.20506@bitberry.com> References: <4E7E3F33.20506@bitberry.com> Message-ID: <20110925165122.49f19717@lrcd.com> On Sat, 24 Sep 2011 22:36:03 +0200 Brian Rasmusson
wrote: > Hi, > > I read a year old thread that libfaac is not compliant with the GPL > license, and that the FFmpeg team is working on their own AAC encoder. > > What is the status on this? Has it reached a stage where the quality > can match libfaac? > > Sorry if this is a repeat question, I didn't find the answer in the > forum archive. > > Brian FFmpeg also supports vo-aacenc which is under the Apache License 2.0. It requires FFmpeg to be configured with: "--enable-libvo-aacenc --enable-version3". Just another AAC encoder to add to your list. From swathiep2009 at gmail.com Mon Sep 26 09:01:51 2011 From: swathiep2009 at gmail.com (swathi) Date: Mon, 26 Sep 2011 00:01:51 -0700 (PDT) Subject: [FFmpeg-user] Issue with cross compiling ffmpeg for linux in windows In-Reply-To: References: Message-ID: <1317020511089-3842906.post@n4.nabble.com> i don't have any idea in resolving the error, please tell me how to resolve? Thanks, Swathi -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Issue-with-cross-compiling-ffmpeg-for-linux-in-windows-tp3830825p3842906.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From swathiep2009 at gmail.com Mon Sep 26 09:29:40 2011 From: swathiep2009 at gmail.com (swathi) Date: Mon, 26 Sep 2011 00:29:40 -0700 (PDT) Subject: [FFmpeg-user] using ffmpeg in android In-Reply-To: References: Message-ID: <1317022180281-3842948.post@n4.nabble.com> can you please tell me, how did you succeed in generating .so files? Even i am working on it, but facing lot of issues in compiling. I am using windows, with cygwin-1.7.9 and i downloaded ffmpeg-android from the link: http://bambuser.com/opensource Please find the attached build script and config.log, containing error description. http://ffmpeg-users.933282.n4.nabble.com/file/n3842948/config.log config.log http://ffmpeg-users.933282.n4.nabble.com/file/n3842948/build.sh build.sh -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/using-ffmpeg-in-android-tp3638397p3842948.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From gkinsey at ad-holdings.co.uk Mon Sep 26 11:47:20 2011 From: gkinsey at ad-holdings.co.uk (Gavin Kinsey) Date: Mon, 26 Sep 2011 10:47:20 +0100 Subject: [FFmpeg-user] Error in compiling FFMPEG 0.8.4 for IPhone OS 4.3 SDK In-Reply-To: References: Message-ID: <201109261047.20256.gkinsey@ad-holdings.co.uk> On Friday 23 September 2011 11:52:29 Gagandeep Bawa wrote: > > I am trying to compile new FFMPEG "Love" 0.8.4 version for ios 4.3 sdk > but it fails. Please have a look in to this issue and let me know if i > am doing anything wrong. configure line for ARMv7 iOS. export IPH_ROOT="/Developer/Platforms/iPhoneOS.platform/Developer" export IPH_SDK="$IPH_ROOT/SDKs/iPhoneOS4.3.sdk" ../configure --cc="$IPH_ROOT/usr/bin/gcc -arch armv7" --sysroot="$IPH_SDK" --extra-ldflags="-isysroot $IPH_SDK" --enable-cross-compile --arch=arm --cpu=cortex-a8 --target-os=darwin --enable-static --disable-shared --enable-pic Haven't done a build in a few weeks, so it's possible ffmpeg has become broken since then, but this did work and probably still does. -- Gavin Kinsey AD Holdings Plc Closed IPTV, the new safe and secure deterministic IP Video solution from Dedicated Micros, is now shipping through selected distributors. - Come and see this award winning plug and play, IP Video innovation at one of our Roadshows around the UK visit: https://www.dedicatedmicros.com/europe/ClosedIPTVRoadshow2011 - Contact our Customer Services Team for more information regarding how to enter the world of safe and secure hybrid IP and analogue video surveillance systems. From gagandeep.bawa at ironroad.com Mon Sep 26 12:19:11 2011 From: gagandeep.bawa at ironroad.com (Gagandeep Bawa) Date: Mon, 26 Sep 2011 15:49:11 +0530 Subject: [FFmpeg-user] Error in compiling FFMPEG 0.8.4 for IPhone OS 4.3 SDK In-Reply-To: <201109261047.20256.gkinsey@ad-holdings.co.uk> References: <201109261047.20256.gkinsey@ad-holdings.co.uk> Message-ID: Hi Gavin, Thanks for reply ! I was able to make build for IPhone. I am also able to convert my .mov file in to .mp4 file in IPhone but the only problem which i am facing is that my output file have less duration, Means if i compress 1 min .mov file then i got 53 sec duration .mp4 file and the audio is playing little bit faster in this file. While compressing file i am getting following error in my console output many times, It seems i am loosing some audio frames. Any idea how can i resolve this. I did not get how to resolve this if this is dts pts problem. *[mp4 @ 0x10e6800] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 2048 >= 2048* *Error while writing audio frame* * * On Mon, Sep 26, 2011 at 3:17 PM, Gavin Kinsey wrote: > On Friday 23 September 2011 11:52:29 Gagandeep Bawa wrote: > > > > I am trying to compile new FFMPEG "Love" 0.8.4 version for ios 4.3 sdk > > but it fails. Please have a look in to this issue and let me know if i > > am doing anything wrong. > > configure line for ARMv7 iOS. > export IPH_ROOT="/Developer/Platforms/iPhoneOS.platform/Developer" > export IPH_SDK="$IPH_ROOT/SDKs/iPhoneOS4.3.sdk" > ../configure --cc="$IPH_ROOT/usr/bin/gcc -arch armv7" > --sysroot="$IPH_SDK" > --extra-ldflags="-isysroot $IPH_SDK" > --enable-cross-compile > --arch=arm > --cpu=cortex-a8 > --target-os=darwin > --enable-static > --disable-shared > --enable-pic > > Haven't done a build in a few weeks, so it's possible ffmpeg has become > broken since then, but this did work and probably still does. > > -- > Gavin Kinsey > AD Holdings Plc > > > Closed IPTV, the new safe and secure deterministic IP Video solution from > Dedicated Micros, is now shipping through selected distributors. > - Come and see this award winning plug and play, IP Video innovation at one > of our Roadshows around the UK visit: > https://www.dedicatedmicros.com/europe/ClosedIPTVRoadshow2011 > - Contact our Customer Services Team for more information regarding how to > enter the world of safe and secure hybrid IP and analogue video surveillance > systems. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Thanks & Regards, Gagandeep, Tech Lead Mobility +91- 7307470044 ------------------------------------------------------------------------------- Ironroad IT India Pvt. Ltd. IT-6, C-6,SEBIZ SQUARE, Sector-67, Mohali, Punjab, India From rogerdpack2 at gmail.com Mon Sep 26 17:37:35 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 26 Sep 2011 09:37:35 -0600 Subject: [FFmpeg-user] Delaying audio and video stream In-Reply-To: References: Message-ID: > I have a video file with one audio track already attached to it and its > perfectly in sync with video. Then I have another audio track which I need > to add to the video as second audio track. But this audio track is ahead of > the audio/video stream so I need to delay it. When I use the following > command Maybe use sox to add some extra whitespace at the beginning? From rogerdpack2 at gmail.com Mon Sep 26 17:38:45 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 26 Sep 2011 09:38:45 -0600 Subject: [FFmpeg-user] av_write_frame crashes without error message In-Reply-To: <1316764448768-3836086.post@n4.nabble.com> References: <1316764448768-3836086.post@n4.nabble.com> Message-ID: > But when trying to write the frames I got into a new video file with the > same format - but this time, using video frames only, no audio - > av_write_frame just crashes without giving any clue why. does running it in gdb reveal a backtrace? From rogerdpack2 at gmail.com Mon Sep 26 17:39:59 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 26 Sep 2011 09:39:59 -0600 Subject: [FFmpeg-user] Read h264 file into byte stream format In-Reply-To: <1316719202.20186.YahooMailNeo@web46203.mail.sp1.yahoo.com> References: <1316719202.20186.YahooMailNeo@web46203.mail.sp1.yahoo.com> Message-ID: > Can I access h264 encoded file as a byte stream format? byte stream of what? raw bits? From rogerdpack2 at gmail.com Mon Sep 26 17:41:15 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 26 Sep 2011 09:41:15 -0600 Subject: [FFmpeg-user] delay/lag in playing stream from network camera in ffplay In-Reply-To: References: Message-ID: > Indeed the fps is 30fps for 320x240 and 15fps for 640x480. I have not reached to the point where I can see those messages. > > Maybe providing more information like bitrates etc. should solve the problem but I haven't tried yet > > Do you see delay/lag in case of 1 fps after video starts? What type output does your camera provide? mjpeg? -r From rogerdpack2 at gmail.com Mon Sep 26 17:42:54 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 26 Sep 2011 09:42:54 -0600 Subject: [FFmpeg-user] ffmpeg converting In-Reply-To: <1316731588057-3835161.post@n4.nabble.com> References: <1316731588057-3835161.post@n4.nabble.com> Message-ID: On Thu, Sep 22, 2011 at 4:46 PM, kistlerjm9 wrote: > is there a way to have ffmpeg convert 3 different flv files at once. I am > setting up a flash player with the option of 320, 480, and hq bitrates aka > 800, 1200, 2000k. I have been told you have to have 3 seperate flv files of > the same video each converted with the seperate bitrates. Any help would be > appreciated. Run 3 processes at once? Actually with today's libavfilter style you might be able to split the input into 3 outputs. From help at archivomas.com Mon Sep 26 18:59:24 2011 From: help at archivomas.com (help at archivomas.com) Date: Mon, 26 Sep 2011 11:59:24 -0500 Subject: [FFmpeg-user] Help with REDCODE In-Reply-To: References: <20110909182005.10466saeso6jgm4g@webmail.archivomas.com> Message-ID: <20110926115924.21464lw3iueyw2o0@webmail.archivomas.com> I finish the job giving back to mac os x + quicktime + qt_tool... apparently no way in ffmpeg Thanks Phil Phil Rhodes escribi?: > >> ffmpeg -i A001_C004_1124MU_001.R3D -vcodec libx264 new.mp4 > > Red files have been encrypted for some time. Unless your file is at > least a year or two old, it can't be read by anything other than > Red's own SDK. > > I'm not sure if there's any appetite for attacking the encryption. > > P > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From victorpaesa at googlemail.com Mon Sep 26 20:02:14 2011 From: victorpaesa at googlemail.com (=?UTF-8?Q?V=C3=ADctor_Paesa?=) Date: Mon, 26 Sep 2011 20:02:14 +0200 Subject: [FFmpeg-user] using ffmpeg in android In-Reply-To: <1317022180281-3842948.post@n4.nabble.com> References: <1317022180281-3842948.post@n4.nabble.com> Message-ID: Hi, On Mon, Sep 26, 2011 at 09:29, swathi wrote: > can you please tell me, how did you succeed in generating .so files? Even i > am working on it, but facing lot of issues in compiling. I am using windows, > with cygwin-1.7.9 and i downloaded ffmpeg-android from the link: > > http://bambuser.com/opensource > > Please find the attached build script and config.log, containing error > description. > http://ffmpeg-users.933282.n4.nabble.com/file/n3842948/config.log config.log The last line in config.log offers some hint: ./configure: line 572: arm-linux-androideabi-gcc: command not found So your gcc is not where you specified in your configure line: ./configure --target-os=linux --cross-prefix=arm-linux-androideabi- \ --arch=arm --sysroot='e:/Android-NDK/android-ndk/platforms/android-8/arch-arm' \ --soname-prefix=/data/data/com.bambuser.broadcaster/lib/ --enable-shared \ --disable-symver --enable-small --optimization-flags=-O2 --disable-everything \ --enable-encoder=mpeg2video --enable-encoder=nellymoser --prefix=../build/ffmpeg/armeabi --extra-cflags= --extra-ldflags= Please avoid DOS style paths in Cygwin, i.e. try /cygdrive/e/ instead of e:/ Regards, V?ctor From listes at defendingthemusic.fr Mon Sep 26 21:13:46 2011 From: listes at defendingthemusic.fr (Paul-Antoine Arras) Date: Mon, 26 Sep 2011 21:13:46 +0200 Subject: [FFmpeg-user] Aspect ratio preservation in WMV In-Reply-To: <201109260853.22804.rodney.baker@iinet.net.au> References: <4E7CDC14.3070809@defendingthemusic.fr> <201109241147.11523.rodney.baker@iinet.net.au> <4E7F5B62.3030400@defendingthemusic.fr> <201109260853.22804.rodney.baker@iinet.net.au> Message-ID: <4E80CEEA.3070908@defendingthemusic.fr> Le 26/09/2011 01:23, Rodney Baker a ?crit : > On Mon, 26 Sep 2011 02:18:34 Paul-Antoine Arras wrote: >> Le 24/09/2011 04:17, Rodney Baker a ?crit : >>> On Sat, 24 Sep 2011 04:50:52 Paul-Antoine Arras wrote: >>>> Hello, >>>> >>>> I'm trying to convert the sound of a video from WMA to MP3. To do so, I >>>> >>>> use the following command line: >>>> ffmpeg -i input.wmv -acodec libmp3lame -aq 4 -vcodec copy >>>> output.wmv >>>> >>>> The sound is converted correctly but the aspect ratio (AR) becomes 4:3 >>>> >>>> instead of 16:9. Then I've tried to force the AR with: >>>> -aspect 16:9 >>>> >>>> without success. >>>> >>>> Why does ffmpeg modify the AR while all it has to do in this case is a >>>> simple copy of the video part? How to make it preserve the original AR >>>> (16:9)? >>>> >>>> Thanks in advance for your help. >>> >>> You need to provide the full output from ffmpeg. >> >> Here is the full output from ffmpeg: >> >> FFmpeg version 0.6.2-4:0.6.2-1ubuntu1.1, Copyright (c) 2000-2010 the >> Libav developers > [...] >> Output #0, asf, to 'output.wmv': >> Metadata: >> WM/EncodingSettings: Lavf52.64.2 >> Stream #0.0: Video: WMV3 / 0x33564D57, yuv420p, 720x576 [PAR 64:45 >> DAR 16:9], q=2-31, 1k tbn, 1k tbc > ^^^^^^^^^^^^ > > ffmpeg is writing the stream out with an aspect ratio of 16:9 (Display Aspect > Ratio) and pixel aspect ratio of 64:45 which, with a standard definition > 720x576 picture will be correct for a 16:9 display. What are you using to view > the resultant output file? Is it showing the correct aspect ratio? Can you > force it? --- When I open output.wmv with VLC, the video is displayed as 4:3, then I can force it to 16:9 by software. But the purpose of what I'm trying to do is making the video readable on my set-top box, which renders it as 4:3 (just like VLC) and can't be forced to another ratio. -- Paul-Antoine ARRAS From yadnesh at gmail.com Mon Sep 26 21:13:49 2011 From: yadnesh at gmail.com (Yadnesh Phadke) Date: Mon, 26 Sep 2011 12:13:49 -0700 Subject: [FFmpeg-user] using ffmpeg in android In-Reply-To: References: <1317022180281-3842948.post@n4.nabble.com> Message-ID: I read in android-ndk google group that Google no longer supports NDK compilation with Cygwin. I would advice to use Linux, to avoid getting into issues because of Cygwin. -- Yadnesh 2011/9/26 V?ctor Paesa > Hi, > > On Mon, Sep 26, 2011 at 09:29, swathi wrote: > > can you please tell me, how did you succeed in generating .so files? Even > i > > am working on it, but facing lot of issues in compiling. I am using > windows, > > with cygwin-1.7.9 and i downloaded ffmpeg-android from the link: > > > > http://bambuser.com/opensource > > > > Please find the attached build script and config.log, containing error > > description. > > http://ffmpeg-users.933282.n4.nabble.com/file/n3842948/config.logconfig.log > > > The last line in config.log offers some hint: > ./configure: line 572: arm-linux-androideabi-gcc: command not found > > So your gcc is not where you specified in your configure line: > ./configure --target-os=linux --cross-prefix=arm-linux-androideabi- \ > --arch=arm > --sysroot='e:/Android-NDK/android-ndk/platforms/android-8/arch-arm' > \ > --soname-prefix=/data/data/com.bambuser.broadcaster/lib/ --enable-shared \ > --disable-symver --enable-small --optimization-flags=-O2 > --disable-everything \ > --enable-encoder=mpeg2video --enable-encoder=nellymoser > --prefix=../build/ffmpeg/armeabi --extra-cflags= --extra-ldflags= > > Please avoid DOS style paths in Cygwin, i.e. try /cygdrive/e/ > instead of e:/ > > Regards, > V?ctor > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- ---------------------------------------------------------------------- Christopher Morley - There is only one success - to be able to spend your life in your own way. From mark at richardsemail.net Mon Sep 26 21:19:45 2011 From: mark at richardsemail.net (Mark Richards) Date: Mon, 26 Sep 2011 13:19:45 -0600 Subject: [FFmpeg-user] Aspect ratio preservation in WMV In-Reply-To: <4E80CEEA.3070908@defendingthemusic.fr> References: <4E7CDC14.3070809@defendingthemusic.fr> <201109241147.11523.rodney.baker@iinet.net.au> <4E7F5B62.3030400@defendingthemusic.fr> <201109260853.22804.rodney.baker@iinet.net.au> <4E80CEEA.3070908@defendingthemusic.fr> Message-ID: <038801cc7c81$41276cf0$c37646d0$@RichardsEmail.Net> The problem is that ffmpeg has defaults for ratios, dimensions, bitrates, etc..., and I am guessing ratio is one of them. You probably need to specify those values yourself when you transcode, to ensure you preserve the dimensions, bitrate, etc... Some of the values of "copy" options. Unfortunately, you'll have to scour the documentation. I'd gladly tell you them if I knew, but I haven't needed to "copy" values myself. Mark -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Paul-Antoine Arras Sent: Monday, September 26, 2011 1:14 PM To: FFmpeg user questions and RTFMs Subject: Re: [FFmpeg-user] Aspect ratio preservation in WMV Le 26/09/2011 01:23, Rodney Baker a ?crit : > On Mon, 26 Sep 2011 02:18:34 Paul-Antoine Arras wrote: >> Le 24/09/2011 04:17, Rodney Baker a ?crit : >>> On Sat, 24 Sep 2011 04:50:52 Paul-Antoine Arras wrote: >>>> Hello, >>>> >>>> I'm trying to convert the sound of a video from WMA to MP3. To do >>>> so, I >>>> >>>> use the following command line: >>>> ffmpeg -i input.wmv -acodec libmp3lame -aq 4 -vcodec copy >>>> output.wmv >>>> >>>> The sound is converted correctly but the aspect ratio (AR) becomes >>>> 4:3 >>>> >>>> instead of 16:9. Then I've tried to force the AR with: >>>> -aspect 16:9 >>>> >>>> without success. >>>> >>>> Why does ffmpeg modify the AR while all it has to do in this case >>>> is a simple copy of the video part? How to make it preserve the >>>> original AR (16:9)? >>>> >>>> Thanks in advance for your help. >>> >>> You need to provide the full output from ffmpeg. >> >> Here is the full output from ffmpeg: >> >> FFmpeg version 0.6.2-4:0.6.2-1ubuntu1.1, Copyright (c) 2000-2010 the >> Libav developers > [...] >> Output #0, asf, to 'output.wmv': >> Metadata: >> WM/EncodingSettings: Lavf52.64.2 >> Stream #0.0: Video: WMV3 / 0x33564D57, yuv420p, 720x576 [PAR >> 64:45 DAR 16:9], q=2-31, 1k tbn, 1k tbc > ^^^^^^^^^^^^ > > ffmpeg is writing the stream out with an aspect ratio of 16:9 (Display > Aspect > Ratio) and pixel aspect ratio of 64:45 which, with a standard > definition > 720x576 picture will be correct for a 16:9 display. What are you using > to view the resultant output file? Is it showing the correct aspect > ratio? Can you force it? --- When I open output.wmv with VLC, the video is displayed as 4:3, then I can force it to 16:9 by software. But the purpose of what I'm trying to do is making the video readable on my set-top box, which renders it as 4:3 (just like VLC) and can't be forced to another ratio. -- Paul-Antoine ARRAS _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From l.jozsa at gmail.com Mon Sep 26 21:22:40 2011 From: l.jozsa at gmail.com (=?UTF-8?Q?Ladislav_J=C3=B3zsa?=) Date: Mon, 26 Sep 2011 21:22:40 +0200 Subject: [FFmpeg-user] Delaying audio and video stream In-Reply-To: References: Message-ID: On Mon, Sep 26, 2011 at 5:37 PM, Roger Pack wrote: > > I have a video file with one audio track already attached to it and its > > perfectly in sync with video. Then I have another audio track which I > need > > to add to the video as second audio track. But this audio track is ahead > of > > the audio/video stream so I need to delay it. When I use the following > > command > > Maybe use sox to add some extra whitespace at the beginning? > Thanks for the tip, Roger. It seems that method you've described is the fastest one. Until I find out how to do it by ffmpeg itself I will be using the method you have posted. Ladislav > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From rogerdpack2 at gmail.com Mon Sep 26 21:40:27 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 26 Sep 2011 13:40:27 -0600 Subject: [FFmpeg-user] -t 5 doesn't seem to work with certain live streams Message-ID: Hello all. When receiving an mp3 stream VLC (which may or may not be a high quality streamer), if I specify that it record for 1s, I get the following (it records for 16 seconds): $ ffmpeg.exe -t 1 -y -i http://127.0.0.1:8081/go.mp3 "C:/Users/packrd/AppData/Local/Temp/wav.2220.32829.wav" ffmpeg version N-32726-ga254452, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 19 2011 19:49:13 with gcc 4.6.1 configuration: --enable-gpl --enable-version3 --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 16. 1 / 51. 16. 1 libavcodec 53. 16. 0 / 53. 16. 0 libavformat 53. 12. 0 / 53. 12. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 43. 2 / 2. 43. 2 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 Ignoring attempt to set invalid timebase for st:0 # THIS MAY BE A CLUE. [ogg @ 00038D00] Estimating duration from bitrate, this may be inaccurate Input #0, ogg, from 'http://127.0.0.1:8081/go.mp3': Duration: N/A, start: 0.000000, bitrate: 268435 kb/s Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 268435 kb/s Output #0, wav, to 'C:/Users/packrd/AppData/Local/Temp/wav.2220.32829.wav': Metadata: encoder : Lavf53.12.0 Stream #0.0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s Stream mapping: Stream #0.0 -> #0.0 (mp3 -> pcm_s16le) Press [q] to stop, [?] for help size= 2781kB time=00:00:16.14 bitrate=1411.2kbits/s video:0kB audio:2781kB global headers:0kB muxing overhead 0.001545% Any insights here? Thanks! -roger- From listes at defendingthemusic.fr Mon Sep 26 22:09:41 2011 From: listes at defendingthemusic.fr (Paul-Antoine Arras) Date: Mon, 26 Sep 2011 22:09:41 +0200 Subject: [FFmpeg-user] Aspect ratio preservation in WMV In-Reply-To: <038801cc7c81$41276cf0$c37646d0$@RichardsEmail.Net> References: <4E7CDC14.3070809@defendingthemusic.fr> <201109241147.11523.rodney.baker@iinet.net.au> <4E7F5B62.3030400@defendingthemusic.fr> <201109260853.22804.rodney.baker@iinet.net.au> <4E80CEEA.3070908@defendingthemusic.fr> <038801cc7c81$41276cf0$c37646d0$@RichardsEmail.Net> Message-ID: <4E80DC05.6020103@defendingthemusic.fr> Le 26/09/2011 21:19, Mark Richards a ?crit : > The problem is that ffmpeg has defaults for ratios, dimensions, bitrates, > etc..., and I am guessing ratio is one of them. You probably need to > specify those values yourself when you transcode, to ensure you preserve the > dimensions, bitrate, etc... Some of the values of "copy" options. > Unfortunately, you'll have to scour the documentation. I'd gladly tell you > them if I knew, but I haven't needed to "copy" values myself. > > Mark --- Thanks for the advice. I'll try to inquire into this... From batguano999 at hotmail.com Mon Sep 26 22:28:58 2011 From: batguano999 at hotmail.com (bat guano) Date: Mon, 26 Sep 2011 20:28:58 +0000 Subject: [FFmpeg-user] -t 5 doesn't seem to work with certain live streams In-Reply-To: References: Message-ID: ---------------------------------------- > From: rogerdpack2 at gmail.com > Date: Mon, 26 Sep 2011 13:40:27 -0600 > To: ffmpeg-user at ffmpeg.org > Subject: [FFmpeg-user] -t 5 doesn't seem to work with certain live streams > > Hello all. > When receiving an mp3 stream VLC (which may or may not be a high > quality streamer), > if I specify that it record for 1s, I get the following (it records > for 16 seconds): > > > > $ ffmpeg.exe -t 1 -y -i http://127.0.0.1:8081/go.mp3 > "C:/Users/packrd/AppData/Local/Temp/wav.2220.32829.wav" > ffmpeg version N-32726-ga254452, Copyright (c) 2000-2011 the FFmpeg developers > built on Sep 19 2011 19:49:13 with gcc 4.6.1 > configuration: --enable-gpl --enable-version3 > --enable-runtime-cpudetect --enable-avisynth --enable-bzlib > --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-libfreetype --enable-libgsm --enable-libmp3lame > --enable-libopenjpeg --enable-librtmp --enable-libschroedinger > --enable-libspeex --enable-libtheora --enable-libvo-aacenc > --enable-libvo-amrwbenc --enable-libvorbis > --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib > libavutil 51. 16. 1 / 51. 16. 1 > libavcodec 53. 16. 0 / 53. 16. 0 > libavformat 53. 12. 0 / 53. 12. 0 > libavdevice 53. 4. 0 / 53. 4. 0 > libavfilter 2. 43. 2 / 2. 43. 2 > libswscale 2. 1. 0 / 2. 1. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > Ignoring attempt to set invalid timebase for st:0 # THIS MAY BE A CLUE. > [ogg @ 00038D00] Estimating duration from bitrate, this may be inaccurate > Input #0, ogg, from 'http://127.0.0.1:8081/go.mp3': > Duration: N/A, start: 0.000000, bitrate: 268435 kb/s > Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 268435 kb/s > Output #0, wav, to 'C:/Users/packrd/AppData/Local/Temp/wav.2220.32829.wav': > Metadata: > encoder : Lavf53.12.0 > Stream #0.0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, > stereo, s16, 1411 kb/s > Stream mapping: > Stream #0.0 -> #0.0 (mp3 -> pcm_s16le) > Press [q] to stop, [?] for help > size= 2781kB time=00:00:16.14 bitrate=1411.2kbits/s > video:0kB audio:2781kB global headers:0kB muxing overhead 0.001545% > > > Any insights here? > Thanks! > -roger- Hi Try putting -t 1 in a different place. Like this:- $ ffmpeg.exe -y -i http://127.0.0.1:8081/go.mp3" -t 1 C:/Users/packrd/AppData/Local/Temp/wav.2220.32829.wav" From rehman_saeed at live.com Tue Sep 27 00:33:37 2011 From: rehman_saeed at live.com (Rehman Saeed) Date: Mon, 26 Sep 2011 23:33:37 +0100 Subject: [FFmpeg-user] delay/lag in playing stream from network camera in ffplay In-Reply-To: References: , , , , Message-ID: Hi Roger, Yes it is mjpeg Regards, Rehman > From: rogerdpack2 at gmail.com > Date: Mon, 26 Sep 2011 09:41:15 -0600 > To: ffmpeg-user at ffmpeg.org > Subject: Re: [FFmpeg-user] delay/lag in playing stream from network camera in ffplay > > > Indeed the fps is 30fps for 320x240 and 15fps for 640x480. I have not reached to the point where I can see those messages. > > > > Maybe providing more information like bitrates etc. should solve the problem but I haven't tried yet > > > > Do you see delay/lag in case of 1 fps after video starts? > > What type output does your camera provide? mjpeg? > -r > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From rogerdpack2 at gmail.com Tue Sep 27 00:42:47 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 26 Sep 2011 16:42:47 -0600 Subject: [FFmpeg-user] -t 5 doesn't seem to work with certain live streams In-Reply-To: References: Message-ID: > Hi > Try putting -t 1 in a different place. > Like this:- > $ ffmpeg.exe -y -i http://127.0.0.1:8081/go.mp3" -t 1 C:/Users/packrd/AppData/Local/Temp/wav.2220.32829.wav" That's a great idea. In this instance it didn't help though. However, if I instructed VLC to stream with a "raw" mux instead of the "ogg" mux then it suddenly works just peachy, recording exactly 2 seconds worth. Other symptoms of this odd problem: The first time I attempt to record from the stream, I have to use -t 1, which captures about 10 seconds worth. The next time I attempt to record from the stream, I have to use *at least* -t 10, to capture anything at all. Which does capture say 11 seconds worth. Weird. Current suspect is VLC's ogg mux (I know it's not sending any PTS timestamps through), but I don't know for certain what the culprit is, and now have a work around :) Thanks! -roger- From rectalogic at rectalogic.com Tue Sep 27 04:51:03 2011 From: rectalogic at rectalogic.com (Andrew Wason) Date: Mon, 26 Sep 2011 22:51:03 -0400 Subject: [FFmpeg-user] Aspect ratio preservation in WMV In-Reply-To: <201109260853.22804.rodney.baker@iinet.net.au> References: <4E7CDC14.3070809@defendingthemusic.fr> <201109241147.11523.rodney.baker@iinet.net.au> <4E7F5B62.3030400@defendingthemusic.fr> <201109260853.22804.rodney.baker@iinet.net.au> Message-ID: On Sun, Sep 25, 2011 at 7:23 PM, Rodney Baker wrote: > On Mon, 26 Sep 2011 02:18:34 Paul-Antoine Arras wrote: >> Output #0, asf, to 'output.wmv': >> ? ?Metadata: >> ? ? ?WM/EncodingSettings: Lavf52.64.2 >> ? ? ?Stream #0.0: Video: WMV3 / 0x33564D57, yuv420p, 720x576 [PAR 64:45 >> DAR 16:9], q=2-31, 1k tbn, 1k tbc > ^^^^^^^^^^^^ > > ffmpeg is writing the stream out with an aspect ratio of 16:9 >(Display Aspect Ratio) ASF supports a couple of different mechanisms for storing aspect ratio information, looking in ffmpeg libavformat/asfenc.c it doesn't seem to use any of them. So I think this is a missing feature. I think in asfenc.c:asf_write_header1() you would need to write AspectRatioX and AspectRatioY into the ff_asf_extended_content_header section. See the spec: http://msdn.microsoft.com/en-us/library/bb643323.aspx Andrew From rogerdpack2 at gmail.com Tue Sep 27 05:57:53 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 26 Sep 2011 21:57:53 -0600 Subject: [FFmpeg-user] How can we enable dshow format in ffmpeg when compiling on windows? In-Reply-To: <1313736975522-3754481.post@n4.nabble.com> References: <1313147406257-3738949.post@n4.nabble.com> <1313736975522-3754481.post@n4.nabble.com> Message-ID: appears you need to use mingw-w64 for it to work [?] http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=7&t=119 From swathiep2009 at gmail.com Tue Sep 27 06:52:44 2011 From: swathiep2009 at gmail.com (swathi) Date: Mon, 26 Sep 2011 21:52:44 -0700 (PDT) Subject: [FFmpeg-user] using ffmpeg in android In-Reply-To: References: <1317022180281-3842948.post@n4.nabble.com> Message-ID: <1317099164052-3846177.post@n4.nabble.com> I am able to configure now. But make command is failing due to some issue. Please find configuration result and make result attached. http://ffmpeg-users.933282.n4.nabble.com/file/n3846177/configuration.txt configuration.txt http://ffmpeg-users.933282.n4.nabble.com/file/n3846177/makeResult.txt makeResult.txt The configure command is as follows: http://ffmpeg-users.933282.n4.nabble.com/file/n3846177/info.txt info.txt -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/using-ffmpeg-in-android-tp3638397p3846177.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From drabner at zoobe.com Tue Sep 27 09:13:19 2011 From: drabner at zoobe.com (Jan drabner) Date: Tue, 27 Sep 2011 09:13:19 +0200 Subject: [FFmpeg-user] av_write_frame crashes without error message In-Reply-To: References: <1316764448768-3836086.post@n4.nabble.com> Message-ID: Sorry, I should probably have pointed out that I'm running on Windows 7 using VS2010. I'm not even sure if GDB would run on this setup, andf of course, I have never worked with that ;) 2011/9/26 Roger Pack > > But when trying to write the frames I got into a new video file with the > > same format - but this time, using video frames only, no audio - > > av_write_frame just crashes without giving any clue why. > > does running it in gdb reveal a backtrace? > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Jan Drabner ? TD Programming Engine & Animation zoobe message entertainment gmbh kurf?rstendamm 226 l 10719 berlin email: drabner at zoobe.com l mob: 0172 7017640 gesch?ftsf?hrer: lenard f. krawinkel tel: +49 30. 288 838 88 l site: *www.zoobe.com* ? email: *info at zoobe.com* amtsgericht charlottenburg, berlin ? hrb-nr. 11 42 27 9b From krueger at signal7.de Tue Sep 27 10:54:45 2011 From: krueger at signal7.de (=?iso-8859-1?Q?Robert_Kr=FCger?=) Date: Tue, 27 Sep 2011 10:54:45 +0200 Subject: [FFmpeg-user] Aspect ratio preservation in WMV In-Reply-To: References: <4E7CDC14.3070809@defendingthemusic.fr> <201109241147.11523.rodney.baker@iinet.net.au> <4E7F5B62.3030400@defendingthemusic.fr> <201109260853.22804.rodney.baker@iinet.net.au> Message-ID: <2B5F50D0-AA81-4E9D-89C1-B32267E22369@signal7.de> On Sep 27, 2011, at 04:51 , Andrew Wason wrote: > On Sun, Sep 25, 2011 at 7:23 PM, Rodney Baker wrote: >> On Mon, 26 Sep 2011 02:18:34 Paul-Antoine Arras wrote: >>> Output #0, asf, to 'output.wmv': >>> Metadata: >>> WM/EncodingSettings: Lavf52.64.2 >>> Stream #0.0: Video: WMV3 / 0x33564D57, yuv420p, 720x576 [PAR 64:45 >>> DAR 16:9], q=2-31, 1k tbn, 1k tbc >> ^^^^^^^^^^^^ >> >> ffmpeg is writing the stream out with an aspect ratio of 16:9 >> (Display Aspect Ratio) > > > ASF supports a couple of different mechanisms for storing aspect ratio > information, looking in ffmpeg libavformat/asfenc.c it doesn't seem to > use any of them. So I think this is a missing feature. > > I think in asfenc.c:asf_write_header1() you would need to write > AspectRatioX and AspectRatioY into the ff_asf_extended_content_header > section. See the spec: > http://msdn.microsoft.com/en-us/library/bb643323.aspx > Paul-Antoine, could you take the time and file this as an issue in the tracker, if possible with a reproducible test case and include the information Andrew provided, so it does not get lost if you don't intend to fix it yourself or it isn't fixed immediately by anyone else? This normally increases your chances of getting your problem solved, you get notified when something happens and it helps the project. From gagandeep.bawa at ironroad.com Tue Sep 27 13:41:09 2011 From: gagandeep.bawa at ironroad.com (Gagandeep Bawa) Date: Tue, 27 Sep 2011 17:11:09 +0530 Subject: [FFmpeg-user] Rotate video with ffmpeg library in iPhone application Message-ID: Is it possible to adjust the orientation of a video programmatically with the ffmeg library using objective-c? I have a recorded video (mov) file that is playing in the wrong orientation and would like to perform the adjustment. If ffmpeg is not the right approach, what is the simplest way to take a video and rotate it by 90 degrees? I tried with following code but it only tells how to detect recorded video orientation http://www.humbug.in/stackoverflow/es/como-para-detectar-iphone-sdk-si-un-archivo-de-video-consto-en-orientacion-de-retrato-o-paisaje-4627940.html I also try following url steps but it describe how to get translation matrix of ffmpeg video stream and adjust IPhone UIView for display purpose. But i don't want to display compress video file in IPhone instead i want to send it to server. So it's orientation should be same as it was at the time it was recorded, So end user can see it on server properly. Currently if i record movie in "landscape right" mode then ffmpeg result video is totally reverse 180 degree. If i record in "Landscape left" then ffmpeg result video is ok but in "portrait" it moves 90 left . Need help ............ http://www.seqoy.com/correct-orientation-for-iphone-recorded-movies-with-ffmpeg/ -- Thanks & Regards, Gagandeep From rickcorteza at gmail.com Tue Sep 27 14:34:55 2011 From: rickcorteza at gmail.com (Rick C.) Date: Tue, 27 Sep 2011 20:34:55 +0800 Subject: [FFmpeg-user] -target ntsc-dvd Message-ID: Hello, I have been doing some conversions like this: ffmpeg -i original.avi -target ntsc-dvd -sameq output.vob And I found that my converted files were often double in size. So I removed the -sameq and it's still the same. Since I'm using -target and everything is being set automatically what can you recommend so that I can keep the converted file size about the same as the original? Thanks! rc From mimeini at gmail.com Tue Sep 27 15:21:35 2011 From: mimeini at gmail.com (mikkel meinike) Date: Tue, 27 Sep 2011 15:21:35 +0200 Subject: [FFmpeg-user] Rotate video with ffmpeg library in iPhone application In-Reply-To: References: Message-ID: I do it with mplayer. But then you will have to rotate only the video and than add the sound afterwords but that is not to difficult. If know one else have better ideas I can explain you how I do that but then you will have to ask me for it because I have to consult some old notes before i can explain you and I don't want to use time on that if you don't need it :-) Mikkel From gagandeep.bawa at ironroad.com Tue Sep 27 15:22:38 2011 From: gagandeep.bawa at ironroad.com (Gagan) Date: Tue, 27 Sep 2011 13:22:38 +0000 (UTC) Subject: [FFmpeg-user] iPhone Matrix Rotation Info References: <8ef05635a4775fe0875c801aa51b0911@mail.gmail.com> Message-ID: Hi Norbert, Can you provide some sample code for this fix. As i tried the patch given by dave (see below) but doesn't work. I am using 0.8.4 version libraries of FFMPEG. In IPhone portrait video rotate to 90 degree after FFMPEG encoding. Need help. //av_dict_set(&st->metadata, "rotate", "180", 0); av_metadata_set2(&st->metadata, "rotate", "180", 0); From rectalogic at rectalogic.com Tue Sep 27 18:03:33 2011 From: rectalogic at rectalogic.com (Andrew Wason) Date: Tue, 27 Sep 2011 12:03:33 -0400 Subject: [FFmpeg-user] Rotate video with ffmpeg library in iPhone application In-Reply-To: References: Message-ID: On Tue, Sep 27, 2011 at 7:41 AM, Gagandeep Bawa wrote: > Is it possible to adjust the orientation of a video > programmatically with the ffmeg library using objective-c? You could use libavfilter. e.g. the corresponding ffmpeg command lines would be: -vf transpose=1 (rotate 90) -vf transpose=2 (rotate 270) -vf vflip,hflip (rotate 180) Andrew From gagandeep.bawa at ironroad.com Tue Sep 27 19:03:43 2011 From: gagandeep.bawa at ironroad.com (Gagandeep) Date: Tue, 27 Sep 2011 22:33:43 +0530 Subject: [FFmpeg-user] Rotate video with ffmpeg library in iPhone application In-Reply-To: References: Message-ID: <6E533182-9ACA-4506-BA9F-99B9C655D2BD@ironroad.com> Sent from my iPhone4 On 27-Sep-2011, at 6:51 PM, mikkel meinike wrote: > I do it with mplayer. But then you will have to rotate only the video > and than add the sound afterwords but that is not to difficult. If > know one else have better ideas I can explain you how I do that but > then you will have to ask me for it because I have to consult some old > notes before i can explain you and I don't want to use time on that if > you don't need it :-) > > Mikkel > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From gagandeep.bawa at ironroad.com Tue Sep 27 19:12:26 2011 From: gagandeep.bawa at ironroad.com (Gagandeep) Date: Tue, 27 Sep 2011 22:42:26 +0530 Subject: [FFmpeg-user] Rotate video with ffmpeg library in iPhone application In-Reply-To: References: Message-ID: Hi Mikkel, Yes I need this please provide code if you have any. I am trying to compress video in iPhone and then sending compress video to server. I need correct orientation of compress video so that it can be seen on server side and on other user's devices easily. While compression I am doing audio encoding which already making audio little bit faster then video and I am getting non monotone timestamp error while writing audio frames and getting few second shorter video So my query is if I will again write audio separately for rotation purpose will it be sync properly with video? Sent from my iPhone4 On 27-Sep-2011, at 6:51 PM, mikkel meinike wrote: > I do it with mplayer. But then you will have to rotate only the video > and than add the sound afterwords but that is not to difficult. If > know one else have better ideas I can explain you how I do that but > then you will have to ask me for it because I have to consult some old > notes before i can explain you and I don't want to use time on that if > you don't need it :-) > > Mikkel > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From rogerdpack2 at gmail.com Tue Sep 27 20:37:48 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Tue, 27 Sep 2011 12:37:48 -0600 Subject: [FFmpeg-user] Can't combine 2 pulse input streams in screencast In-Reply-To: References: Message-ID: > I should also say that whenever I use the input hw:0,0 or something similar, > when I play back the resulting video, the audio seems to get gradually out > of sync. Can anyone help? Maybe it's the same reason that ffserver gets out of sync eventually (if it still does...) ? From youngshingjun at 163.com Tue Sep 27 11:01:49 2011 From: youngshingjun at 163.com (=?GBK?B?0e7Qy778?=) Date: Tue, 27 Sep 2011 17:01:49 +0800 (CST) Subject: [FFmpeg-user] ffmpeg for iPhone fails Message-ID: <155359f.795e.132aa1e5e5a.Coremail.youngshingjun@163.com> Hello! I want to use ffmpeg for iPhone app. But I fail. Xcode version: 3.2.6. iOS SDK version:4.3. iOS device: iPod touch 4, 4.3.3(8J2). The Xcode console messages: "warning: Unable to read symbols for /Developer/Platforms/iPhoneOS.platform/DeviceSupport/4.3.3 (8J2)/Symbols/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.3.sdk/usr/lib/libavutil.dylib (file not found). warning: Unable to read symbols from "libavutil.dylib" (not yet mapped into memory)." "dyld: Library not loaded: /Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.3.sdk/usr/lib/libavutil.dylib Referenced from: /var/mobile/Applications/462CB733-4699-4BB4-B2BF-F2C157DA5C7B/Book+.app/Book+ Reason: image not found" Get the latest FFmpeg from SVN . Configure FFmpeg with one of the following commands: (1) ./configure --enable-cross-compile --cross-prefix=/Developer/Platforms/iPhoneOS.platform/Developer/usr/bin/ --cc=/Developer/Platforms/iPhoneOS.platform/Developer/usr/bin/arm-apple-darwin10-gcc-4.2.1 --prefix=/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.3.sdk/usr --extra-cflags="-isysroot /Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.3.sdk" --extra-ldflags="-isysroot /Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.3.sdk -Wl,-syslibroot /Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.3.sdk" --arch=armv6 --enable-armv6 --enable-gpl --enable-shared --enable-swscale --enable-zlib --enable-bzlib --disable-ffmpeg --disable-ffplay --disable-ffserver (2) make (3) sudo make install I wish you help me solve the problem. Thanks a lot!!!! From listes at defendingthemusic.fr Tue Sep 27 21:28:49 2011 From: listes at defendingthemusic.fr (Paul-Antoine Arras) Date: Tue, 27 Sep 2011 21:28:49 +0200 Subject: [FFmpeg-user] Aspect ratio preservation in WMV In-Reply-To: <2B5F50D0-AA81-4E9D-89C1-B32267E22369@signal7.de> References: <4E7CDC14.3070809@defendingthemusic.fr> <201109241147.11523.rodney.baker@iinet.net.au> <4E7F5B62.3030400@defendingthemusic.fr> <201109260853.22804.rodney.baker@iinet.net.au> <2B5F50D0-AA81-4E9D-89C1-B32267E22369@signal7.de> Message-ID: <4E8223F1.5080601@defendingthemusic.fr> Le 27/09/2011 10:54, Robert Kr?ger a ?crit : > > On Sep 27, 2011, at 04:51 , Andrew Wason wrote: > >> On Sun, Sep 25, 2011 at 7:23 PM, Rodney Baker wrote: >>> On Mon, 26 Sep 2011 02:18:34 Paul-Antoine Arras wrote: >>>> Output #0, asf, to 'output.wmv': >>>> Metadata: >>>> WM/EncodingSettings: Lavf52.64.2 >>>> Stream #0.0: Video: WMV3 / 0x33564D57, yuv420p, 720x576 [PAR 64:45 >>>> DAR 16:9], q=2-31, 1k tbn, 1k tbc >>> ^^^^^^^^^^^^ >>> >>> ffmpeg is writing the stream out with an aspect ratio of 16:9 >>> (Display Aspect Ratio) >> >> >> ASF supports a couple of different mechanisms for storing aspect ratio >> information, looking in ffmpeg libavformat/asfenc.c it doesn't seem to >> use any of them. So I think this is a missing feature. >> >> I think in asfenc.c:asf_write_header1() you would need to write >> AspectRatioX and AspectRatioY into the ff_asf_extended_content_header >> section. See the spec: >> http://msdn.microsoft.com/en-us/library/bb643323.aspx >> > > Paul-Antoine, could you take the time and file this as an issue in the tracker, if possible with a reproducible test case and include the information Andrew provided, so it does not get lost if you don't intend to fix it yourself or it isn't fixed immediately by anyone else? This normally increases your chances of getting your problem solved, you get notified when something happens and it helps the project. > --- OK I'm gonna file an issue for this. Also I've already retrieved a copy of the ASF spec, so I'm gonna see if can manage to fix it myself. -- Paul-Antoine ARRAS From a_titov82 at mail.ru Tue Sep 27 21:29:31 2011 From: a_titov82 at mail.ru (Alexey Titov) Date: Tue, 27 Sep 2011 23:29:31 +0400 Subject: [FFmpeg-user] player skips frames in video after encoding Message-ID: <1317151771.2028.11.camel@alex-desktop> Hello. I encoded video from my camera with the following commands ffmpeg -i file.mp4 -sameq -an -threads 2 -deinterlace -r 25 converted/file.mp4.temp.MP4 ffmpeg -i converted/file.mp4.temp.MP4.temp.MP4 -vcodec libx264 -vpre normal -threads 3 converted/file.mp4 I applied this command to many video files. Some files are played well after encoding but several files are not. The first command produces big file (converted/file.mp4.temp.MP4) which can be played by any player. The second command reduces file size (converted/file.mp4) but most of this video is played very slowly and lots frames are skipped. Video looks like a slideshow. I don't want to show the video itself, but I can provide any information about this video and any debug information. Please help me to investigate what do I do wrong. ffmpeg version 0.8.3, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 18 2011 13:18:53 with gcc 4.5.2 configuration: --enable-gpl --enable-libx264 libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 7. 0 / 53. 7. 0 libavformat 53. 4. 0 / 53. 4. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 From briane at elementaltechnologies.com Wed Sep 28 01:00:22 2011 From: briane at elementaltechnologies.com (Brian Enigma) Date: Tue, 27 Sep 2011 16:00:22 -0700 Subject: [FFmpeg-user] Can extract DVB-Sub, cannot extract DVB-T Message-ID: So I have two sets of transport stream files, one with a DVB-Sub subtitle track and one with a DVB-T Teletext track. In both instances, there's an audio track, a video track, and a data track and ffmpeg recognizes all of them correctly. What I'd like to do is extract out just the data track (i.e. strip out the audio and video tracks) for analysis and post-processing. In the DVB-Sub case, I can do something like this (with both ffmpeg 0.8.4 and the latest git) and everything works as expected: ffmpeg -i file1.ts -vn -an -scodec copy -f rawvideo dvbsub.dat When I try to do the very same thing for the DVB-Teletext track, I always end up with a zero-length file. I know there should be data in there, as I've looked at the data PID in an analyzer and can see the content of the teletext payloads. Ffmpeg sees and recognizes the stream as dvb_teletext, but it doesn't seem to want to export it. Am I doing something wrong? Am I missing something? Or is this something that ffmpeg does not support? ?Brian From humer4489 at gmail.com Wed Sep 28 02:21:54 2011 From: humer4489 at gmail.com (Patrick) Date: Tue, 27 Sep 2011 20:21:54 -0400 Subject: [FFmpeg-user] ffmpeg for iPhone fails In-Reply-To: <155359f.795e.132aa1e5e5a.Coremail.youngshingjun@163.com> References: <155359f.795e.132aa1e5e5a.Coremail.youngshingjun@163.com> Message-ID: On Sep 27, 2011, at 5:01 AM, ??? wrote: > Hello! > I want to use ffmpeg for iPhone app. But I fail. > Xcode version: 3.2.6. > iOS SDK version:4.3. > iOS device: iPod touch 4, 4.3.3(8J2). > > > The Xcode console messages: > "warning: Unable to read symbols for /Developer/Platforms/iPhoneOS.platform/DeviceSupport/4.3.3 (8J2)/Symbols/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.3.sdk/usr/lib/libavutil.dylib (file not found). > warning: Unable to read symbols from "libavutil.dylib" (not yet mapped into memory)." > "dyld: Library not loaded: /Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.3.sdk/usr/lib/libavutil.dylib > > Referenced from: /var/mobile/Applications/462CB733-4699-4BB4-B2BF-F2C157DA5C7B/Book+.app/Book+ > > Reason: image not found" > > > Get the latest FFmpeg from SVN . > Configure FFmpeg with one of the following commands: > (1) ./configure --enable-cross-compile --cross-prefix=/Developer/Platforms/iPhoneOS.platform/Developer/usr/bin/ --cc=/Developer/Platforms/iPhoneOS.platform/Developer/usr/bin/arm-apple-darwin10-gcc-4.2.1 --prefix=/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.3.sdk/usr --extra-cflags="-isysroot /Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.3.sdk" --extra-ldflags="-isysroot /Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.3.sdk -Wl,-syslibroot /Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.3.sdk" --arch=armv6 --enable-armv6 --enable-gpl --enable-shared --enable-swscale --enable-zlib --enable-bzlib --disable-ffmpeg --disable-ffplay --disable-ffserver > (2) make > (3) sudo make install > > > I wish you help me solve the problem. Thanks a lot!!!! > > > "Unable to read symbols"..... means architecture mismatch. Remember compile i386 for simulator, and armv6/7 for device. Use this as a starting point: http://www.codza.com/extracting-frames-from-movies-on-iphone Patrick From gagandeep.bawa at ironroad.com Wed Sep 28 04:13:01 2011 From: gagandeep.bawa at ironroad.com (Gagandeep) Date: Wed, 28 Sep 2011 07:43:01 +0530 Subject: [FFmpeg-user] Rotate video with ffmpeg library in iPhone application In-Reply-To: References: Message-ID: <1FDFE4BC-7526-4327-814E-80093E4AE7AC@ironroad.com> Hi Andrew, I know this from command line. I want to do it through code using library and need sample code for this. Sent from my iPhone4 On 27-Sep-2011, at 9:33 PM, Andrew Wason wrote: > On Tue, Sep 27, 2011 at 7:41 AM, Gagandeep Bawa > wrote: >> Is it possible to adjust the orientation of a video >> programmatically with the ffmeg library using objective-c? > > You could use libavfilter. e.g. the corresponding ffmpeg command lines would be: > > -vf transpose=1 (rotate 90) > -vf transpose=2 (rotate 270) > -vf vflip,hflip (rotate 180) > > Andrew > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From vandung.tran at vn.panasonic.com Wed Sep 28 06:55:40 2011 From: vandung.tran at vn.panasonic.com (vandung.tran at vn.panasonic.com) Date: Wed, 28 Sep 2011 11:55:40 +0700 Subject: [FFmpeg-user] can't build ffplay for ARM Message-ID: Building ffmpeg for ARM, I can get ffmpeg, ffprobe, ffserver but I can't get ffplay. I have searched and someone said the reason is lack of SDL lib. I build SDL-1.2.14 for ARM, using arm-none-linux-gnueabi-gcc. I informed the SDL directory to compiler by $export CFLAGS="-I//include/SDL -L//lib" However, I still can't get ffplay. I checked make.log and I didn't find "CC ffplay". I don't know why ffplay wasn't compiled. If you have any idea to fix this, please let me know. Thanks and Best Regards, ===================== Tran Van Dung "The information in this e-mail (including attachments) is confidential and is only intended for use by the addressee. If you are not the intended recipient or addressee, please notify us immediately. Any unauthorized disclosure, use or dissemination either in whole or in part is prohibited. Opinions, conclusions and other information contained in this message are personal opinions of the sender and do not necessarily represent the views of the Panasonic Group of companies." From dusa.adrian at gmail.com Wed Sep 28 07:22:45 2011 From: dusa.adrian at gmail.com (Adrian Dusa) Date: Wed, 28 Sep 2011 08:22:45 +0300 Subject: [FFmpeg-user] CLI ffmpeg on MacOS Message-ID: Dear list, I am recently working on a Mac (64bit, Lion), and wanted to use the CLI version of the ffmpeg just like in my former Linux box. The previous command, to grab a portion of the screen, was: ffmpeg -f alsa -i pulse -f x11grab -r 15 -s 500x500 -i :0.0+304,198 -vcodec libx264 -vpre lossless_ultrafast -threads 0 blahvideo.mkv Obviously, the options cannot be the same in the Mac box (there's no alsa, there's no x11grab), and after much researching on the web I finally got to my last resource, that is asking this list. Is is possible to mimic the above command on a Mac? Thanks in advance, Adrian From fab_tafelmak at hotmail.com Wed Sep 28 11:22:52 2011 From: fab_tafelmak at hotmail.com (Fab Tafel) Date: Wed, 28 Sep 2011 09:22:52 +0000 Subject: [FFmpeg-user] Encoding from MOV AppleProRes to AVI Message-ID: Hi all! I'm trying since yesterday to encode an AppleProRes video into AVI. I'm not sure if AppleProRes is supported this way. I'm using a Mac (Mac OSX 10.5.8) and I compiled ffmpeg with MacPorts. -version command returns: ffmpeg version 0.7.5, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 26 2011 19:02:27 with gcc 4.0.1 (Apple Inc. build 5493) configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --mandir=/opt/local/share/man --enable-shared --enable-pthreads --disable-indevs --cc=/usr/bin/gcc-4.0 --arch=i386 --enable-yasm libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 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 ffmpeg 0.7.5 libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 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 And my command to encode the stuff looks like this: ffmpeg -i video.mov -sameq -f mov -acodec pcm_s16be -y video.avi and returns: Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2011-03-21 09:38:07 Duration: 00:10:28.28, start: -15.880000, bitrate: 107517 kb/s Stream #0.0(eng): Video: Apple ProRes 422, 1920x1080, 105958 kb/s, PAR 1:1 DAR 16:9, 25 fps, 25 tbr, 25 tbn, 25 tbc Metadata: creation_time : 2011-03-21 09:38:07 Stream #0.1(eng): Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s Metadata: creation_time : 2011-03-21 09:38:07 Stream #0.2(eng): Data: tmcd / 0x64636D74 Metadata: creation_time : 2011-03-21 12:25:44 [buffer @ 0x1b29040] Invalid pixel format string '-1' Error opening filters! Does someone know what's wrong - if something is wrong - and how to correct it? a very hudge thankyou! rekam From mimeini at gmail.com Wed Sep 28 18:39:02 2011 From: mimeini at gmail.com (mikkel meinike) Date: Wed, 28 Sep 2011 18:39:02 +0200 Subject: [FFmpeg-user] Rotate video with ffmpeg library in iPhone application In-Reply-To: <1FDFE4BC-7526-4327-814E-80093E4AE7AC@ironroad.com> References: <1FDFE4BC-7526-4327-814E-80093E4AE7AC@ironroad.com> Message-ID: OK first a little about stripping and adding audio First, strip the video into a silent file: ffmpeg -i in.avi -an -vcodec copy out.avi. This will copy just the video stream with no audio. Then, add your audio track: ffmpeg -i out.avi -vcodec copy -i audio.file new_out.avi -newaudio -------------------------------- The mplayer command for rotating video and saving it to a file. mplayer -vo xv -vf rotate=2 yuv4mpeg:file=ret0tmp.y4m Video0011.3gp That file than have to be convert to something usable. For that you'll need mjpegtool. I talk a littel about it in this blogpost. http://nerdcore-enthusiasm.blogspot.com/2010/12/add-subtitles-to-video-clip-with-pure.html Mikkel From benoit at friry.net Wed Sep 28 19:25:45 2011 From: benoit at friry.net (Benoit) Date: Wed, 28 Sep 2011 19:25:45 +0200 Subject: [FFmpeg-user] Bug for bug file? Message-ID: <20110928172545.GA17932@friry.net> Hello, I'm trying to convert mjpeg into h264. But ffmpeg is "killed" before finishing the job. Details and outputs below. I don't know how to investigate. Should I open a bug report? Can you please help me? Thanks, benoit ======================================== My script and the file $ cat convert_h264.sh #!/bin/ksh time nice ffmpeg \ -i $1 \ -acodec libfaac -aq 120 \ -vcodec libx264 -r 50 -vf "yadif=1:0,crop=in_w-24:in_h-8:8:0" \ -preset veryfast -profile $3 -crf 23 \ -f mp4 $2 myvideo.avi is a 7.2 GB file, grabbed from a VHS via a Dazzle device. ======================================== First test: profile main $ ./convert_h264.sh myvideo.avi myvideo.veryfast_main_23.mp4 main ffmpeg version 0.8.3, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 10 2011 13:42:49 with gcc 4.6.1 configuration: --prefix=/usr --extra-cflags='-Wall -g ' --cc='ccache cc' --enable-shared --enable-libmp3lame --enable-gpl --enable-nonfree --enable-libdirac --disable-decoder=libdirac --enable-libvorbis --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab --enable-libgsm --enable-libtheora --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libspeex --enable-nonfree --disable-stripping --enable-libschroedinger --disable-encoder=libschroedinger --enable-version3 --enable-libopenjpeg --enable-libvpx --enable-librtmp --enable-avfilter --enable-frei0r --enable-libopencv --enable-libfreetype --enable-libvo-aacenc --enable-libvo-amrwbenc --libdir=/usr/lib/x86_64-linux-gnu --enable-libdc1394 --disable-altivec --disable-armv5te --disable-armv6 --disable-vis --shlibdir=/usr/lib/x86_64-linux-gnu libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 7. 0 / 53. 7. 0 libavformat 53. 4. 0 / 53. 4. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, avi, from 'myvideo.avi': Duration: 00:47:29.56, start: 0.000000, bitrate: 21670 kb/s Stream #0.0: Video: mjpeg, yuvj420p, 720x576, 25 tbr, 25 tbn, 25 tbc Stream #0.1: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s [buffer @ 0x1204a00] w:720 h:576 pixfmt:yuvj420p tb:1/1000000 sar:0/1 sws_param: [yadif @ 0x120bb20] mode:1 parity:0 [crop @ 0x1207e20] w:720 h:576 -> w:696 h:568 [libx264 @ 0x1203ae0] using cpu capabilities: MMX2 SSE2Slow SlowCTZ [libx264 @ 0x1203ae0] profile Main, level 3.1 [libx264 @ 0x1203ae0] 264 - core 116 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=1 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=2 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 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=1 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=10 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 'myvideo.veryfast_main_23.mp4': Metadata: encoder : Lavf53.4.0 Stream #0.0: Video: libx264, yuvj420p, 696x568, q=2-31, 200 kb/s, 50 tbn, 50 tbc Stream #0.1: Audio: libfaac, 44100 Hz, 2 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop, [?] for help Killed 9261 fps= 22 q=31.0 size= 408590kB time=00:26:24.96 bitrate=2111.8kbits/s dup=6289 drop=0 6138.46s real 5820.28s user 30.55s system ======================================== Second test: profile baseline $ ./convert_h264.sh myvideo.avi myvideo.veryfast_baseline_23.mp4 baseline ffmpeg version 0.8.3, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 10 2011 13:42:49 with gcc 4.6.1 configuration: --prefix=/usr --extra-cflags='-Wall -g ' --cc='ccache cc' --enable-shared --enable-libmp3lame --enable-gpl --enable-nonfree --enable-libdirac --disable-decoder=libdirac --enable-libvorbis --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab --enable-libgsm --enable-libtheora --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libspeex --enable-nonfree --disable-stripping --enable-libschroedinger --disable-encoder=libschroedinger --enable-version3 --enable-libopenjpeg --enable-libvpx --enable-librtmp --enable-avfilter --enable-frei0r --enable-libopencv --enable-libfreetype --enable-libvo-aacenc --enable-libvo-amrwbenc --libdir=/usr/lib/x86_64-linux-gnu --enable-libdc1394 --disable-altivec --disable-armv5te --disable-armv6 --disable-vis --shlibdir=/usr/lib/x86_64-linux-gnu libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 7. 0 / 53. 7. 0 libavformat 53. 4. 0 / 53. 4. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, avi, from 'myvideo.avi': Duration: 00:47:29.56, start: 0.000000, bitrate: 21670 kb/s Stream #0.0: Video: mjpeg, yuvj420p, 720x576, 25 tbr, 25 tbn, 25 tbc Stream #0.1: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s [buffer @ 0x1d67a00] w:720 h:576 pixfmt:yuvj420p tb:1/1000000 sar:0/1 sws_param: [yadif @ 0x1d6eb20] mode:1 parity:0 [crop @ 0x1d6ae20] w:720 h:576 -> w:696 h:568 [libx264 @ 0x1d66ae0] using cpu capabilities: MMX2 SSE2Slow SlowCTZ [libx264 @ 0x1d66ae0] profile Constrained Baseline, level 3.1 [libx264 @ 0x1d66ae0] 264 - core 116 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=2 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=10 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 'myvideo.veryfast_baseline_23.mp4': Metadata: encoder : Lavf53.4.0 Stream #0.0: Video: libx264, yuvj420p, 696x568, q=2-31, 200 kb/s, 50 tbn, 50 tbc Stream #0.1: Audio: libfaac, 44100 Hz, 2 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop, [?] for help Killed 9557 fps= 15 q=31.0 size= 434826kB time=00:19:50.92 bitrate=2991.0kbits/s dup=4937 drop=0 6632.78s real 5736.91s user 31.41s system ======================================== Third test: profile baseline and without filter I remove [-r 50 -vf "yadif=1:0,crop=in_w-24:in_h-8:8:0"] in the script $ ./convert_h264.sh myvideo.avi myvideo.veryfast_baseline_23_nocropyadif.mp4 baseline ffmpeg version 0.8.3, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 10 2011 13:42:49 with gcc 4.6.1 configuration: --prefix=/usr --extra-cflags='-Wall -g ' --cc='ccache cc' --enable-shared --enable-libmp3lame --enable-gpl --enable-nonfree --enable-libdirac --disable-decoder=libdirac --enable-libvorbis --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab --enable-libgsm --enable-libtheora --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libspeex --enable-nonfree --disable-stripping --enable-libschroedinger --disable-encoder=libschroedinger --enable-version3 --enable-libopenjpeg --enable-libvpx --enable-librtmp --enable-avfilter --enable-frei0r --enable-libopencv --enable-libfreetype --enable-libvo-aacenc --enable-libvo-amrwbenc --libdir=/usr/lib/x86_64-linux-gnu --enable-libdc1394 --disable-altivec --disable-armv5te --disable-armv6 --disable-vis --shlibdir=/usr/lib/x86_64-linux-gnu libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 7. 0 / 53. 7. 0 libavformat 53. 4. 0 / 53. 4. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, avi, from 'myvideo.avi': Duration: 00:47:29.56, start: 0.000000, bitrate: 21670 kb/s Stream #0.0: Video: mjpeg, yuvj420p, 720x576, 25 tbr, 25 tbn, 25 tbc Stream #0.1: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s [buffer @ 0x19fdfa0] w:720 h:576 pixfmt:yuvj420p tb:1/1000000 sar:0/1 sws_param: [libx264 @ 0x19fdae0] using cpu capabilities: MMX2 SSE2Slow SlowCTZ [libx264 @ 0x19fdae0] profile Constrained Baseline, level 3.1 [libx264 @ 0x19fdae0] 264 - core 116 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=2 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=10 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 'myvideo.veryfast_baseline_23_nocropyadif.mp4': Metadata: encoder : Lavf53.4.0 Stream #0.0: Video: libx264, yuvj420p, 720x576, q=2-31, 200 kb/s, 50 tbn, 50 tbc Stream #0.1: Audio: libfaac, 44100 Hz, 2 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop, [?] for help Memory fault ps= 19 q=31.0 size= 185766kB time=00:06:23.14 bitrate=3971.9kbits/s dup=10522 drop=0 1658.33s real 1590.50s user 10.18s system From doseofvitamink at gmail.com Wed Sep 28 19:33:34 2011 From: doseofvitamink at gmail.com (Kurt Nordstrom) Date: Wed, 28 Sep 2011 12:33:34 -0500 Subject: [FFmpeg-user] Stripping chapter metadata from video Message-ID: Can anybody give me a recipe for how to strip the chapter metadata from a video? Previously, I was able to do something like: -map_chapters -1:0 However, recently the syntax for the -map_chapters option has changed, as per the changelog: -map_chapters now takes only an input file index and applies to the next output file. This is consistent with how all the other options work. It doesn't explain how to say "don't copy anything at all." And, since chapters are copied by default, I'm a little stumped. Any ideas? -- http://www.blar.net/kurt/blog/ From p.nejedly at gmail.com Wed Sep 28 16:40:11 2011 From: p.nejedly at gmail.com (Petr Nejedly) Date: Wed, 28 Sep 2011 15:40:11 +0100 Subject: [FFmpeg-user] HTTP headers Message-ID: Hi, is it possible to pass custom HTTP headers (or at least change user-agent) to a command like the bellow? ffmpeg -i http://www.someserver.com/file.mp4 /tmp/out/mp4 I have seen some code in http.c, but I'm not sure if it can be invoked via the ffmpeg command line. Cheers, Petr From doseofvitamink at gmail.com Wed Sep 28 20:05:42 2011 From: doseofvitamink at gmail.com (Kurt Nordstrom) Date: Wed, 28 Sep 2011 13:05:42 -0500 Subject: [FFmpeg-user] HTTP headers In-Reply-To: References: Message-ID: Could you possibly try piping the output from wget or curl into ffmpeg? On Wed, Sep 28, 2011 at 9:40 AM, Petr Nejedly wrote: > Hi, > > is it possible to pass custom HTTP headers (or at least change user-agent) > to a command like the bellow? > > ffmpeg -i http://www.someserver.com/file.mp4 /tmp/out/mp4 > > I have seen some code in http.c, but I'm not sure if it can be invoked via > the ffmpeg command line. > > Cheers, > > Petr > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- http://www.blar.net/kurt/blog/ From hansooloo at gmail.com Thu Sep 29 01:21:17 2011 From: hansooloo at gmail.com (Han Sooloo) Date: Wed, 28 Sep 2011 19:21:17 -0400 Subject: [FFmpeg-user] How to enable all options for compilation Message-ID: <945899D8-D5F2-4C21-B6D0-A085B9555C35@gmail.com> Downloaded the source code and trying to configure the source tree. How can I enable "all" encoders, decoders, etc etc? From gagandeep.bawa at ironroad.com Thu Sep 29 05:15:37 2011 From: gagandeep.bawa at ironroad.com (Gagandeep) Date: Thu, 29 Sep 2011 08:45:37 +0530 Subject: [FFmpeg-user] Rotate video with ffmpeg library in iPhone application In-Reply-To: References: <1FDFE4BC-7526-4327-814E-80093E4AE7AC@ironroad.com> Message-ID: <31DAE84D-E2A3-4F75-80A9-5C455D0E6C07@ironroad.com> Thanks for reply I need this rotation thing to be done from ffmpeg library code inside iphone app itself not from command line tool. Otherwise I can use ffmpeg transpose command on server which rotate video 90. Is this transpose command will leave audio from file after rotation? Sent from my iPhone4 On 28-Sep-2011, at 10:09 PM, mikkel meinike wrote: > OK first a little about stripping and adding audio > > > First, strip the video into a silent file: > > ffmpeg -i in.avi -an -vcodec copy out.avi. > > This will copy just the video stream with no audio. > > Then, add your audio track: > > ffmpeg -i out.avi -vcodec copy -i audio.file new_out.avi -newaudio > > -------------------------------- > > The mplayer command for rotating video and saving it to a file. > > mplayer -vo xv -vf rotate=2 yuv4mpeg:file=ret0tmp.y4m Video0011.3gp > > That file than have to be convert to something usable. For that you'll > need mjpegtool. > > I talk a littel about it in this blogpost. > > http://nerdcore-enthusiasm.blogspot.com/2010/12/add-subtitles-to-video-clip-with-pure.html > > > Mikkel > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From jd1008 at gmail.com Thu Sep 29 05:25:11 2011 From: jd1008 at gmail.com (JD) Date: Wed, 28 Sep 2011 20:25:11 -0700 Subject: [FFmpeg-user] Compilation failed Message-ID: <4E83E517.9090602@gmail.com> A non-text attachment was scrubbed... Name: ffmpeg.build.log.bz2 Type: application/x-bzip Size: 13380 bytes Desc: not available URL: From mimeini at gmail.com Thu Sep 29 07:24:17 2011 From: mimeini at gmail.com (mikkel meinike) Date: Thu, 29 Sep 2011 07:24:17 +0200 Subject: [FFmpeg-user] Rotate video with ffmpeg library in iPhone application In-Reply-To: <31DAE84D-E2A3-4F75-80A9-5C455D0E6C07@ironroad.com> References: <1FDFE4BC-7526-4327-814E-80093E4AE7AC@ironroad.com> <31DAE84D-E2A3-4F75-80A9-5C455D0E6C07@ironroad.com> Message-ID: > I need this rotation thing to be done from ffmpeg library code inside iphone app itself not from command ine tool. Unfortunately I can't help you with that. I don't know anything about it iPhone and apps and how this function. Is this transpose command will leave audio from file after rotation? This command do the rotation and save it to a file. This is not done with ffmpeg but with mplayer. mplayer -vo xv -vf rotate=2 yuv4mpeg:file=ret0tmp.y4m Video0011.3gp And it will leave audio out. Mikkel From tim.nicholson at bbc.co.uk Thu Sep 29 09:19:44 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Thu, 29 Sep 2011 08:19:44 +0100 Subject: [FFmpeg-user] Encoding from MOV AppleProRes to AVI In-Reply-To: References: Message-ID: <4E841C10.7030908@bbc.co.uk> On 28/09/11 10:22, Fab Tafel wrote: > > Hi all! > > I'm trying since yesterday to encode an AppleProRes video into AVI. I'm not sure if AppleProRes is supported this way. > > I'm using a Mac (Mac OSX 10.5.8) and I compiled ffmpeg with MacPorts. -version command returns: > > [....] > > And my command to encode the stuff looks like this: > ffmpeg -i video.mov -sameq -f mov -acodec pcm_s16be -y video.avi > >[...] > Error opening filters! > > > Does someone know what's wrong - if something is wrong - and how to correct it? So you are specifying creation of a mov format file but with an avi extension, and without specifying a codec want the same quantiser as a decode only format, and you are surprised it doesn't work? lose the -f mov and try setting a sensible codec for the avi wrapper. -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From belcampo at zonnet.nl Thu Sep 29 09:46:58 2011 From: belcampo at zonnet.nl (belcampo) Date: Thu, 29 Sep 2011 09:46:58 +0200 Subject: [FFmpeg-user] How to enable all options for compilation In-Reply-To: <945899D8-D5F2-4C21-B6D0-A085B9555C35@gmail.com> References: <945899D8-D5F2-4C21-B6D0-A085B9555C35@gmail.com> Message-ID: <4E842272.9080306@zonnet.nl> On 09/29/2011 01:21 AM, Han Sooloo wrote: > Downloaded the source code and trying to configure the source tree. How can I enable "all" encoders, decoders, etc etc? AFAIK by having all development packages installed and have them all --enable-lib..... on your configure command line. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From mistrybhavinm at yahoo.co.in Thu Sep 29 11:34:29 2011 From: mistrybhavinm at yahoo.co.in (Bhavin Mistry) Date: Thu, 29 Sep 2011 15:04:29 +0530 (IST) Subject: [FFmpeg-user] Convert m4a to mp3 Message-ID: <1317288869.54060.YahooMailNeo@web95515.mail.in.yahoo.com> Hello, I want to conver m4a file to mp3. I use following command, from my php file $cmd="ffmpeg -i from.m4a to.mp3"; exec ($cmd); But converted sound is somethink junk. I can not describe it properly, but it feels like it plays very fast. Can anyone help me with this? ? Regards, Bhavin M Mistry. From c.guo at nbsystems.tv Thu Sep 29 11:34:46 2011 From: c.guo at nbsystems.tv (Edward.guo) Date: Thu, 29 Sep 2011 02:34:46 -0700 (PDT) Subject: [FFmpeg-user] Can FFMPEG encode DVCPRO HD? In-Reply-To: References: <4e7ae53a.44cae30a.7aae.ffffe0af@mx.google.com> Message-ID: <1317288886665-3854976.post@n4.nabble.com> Thanks Dave for your kind help. My problem has been solved. Dave Bevan wrote: > > > > >>Dear Guys >>I found ffmpeg can only encoder image to DVCPRO SD. Can anybody tell me how to make it be able to encode the DVCPRO HD frame like 1080i and 720p ? > > > http://code.google.com/p/ffmbc/ > > >>Thanks a lot >>Edward > > http://www.bbc.co.uk/ > This e-mail (and any attachments) is confidential and may contain personal > views which are not the views of the BBC unless specifically stated. > If you have received it in error, please delete it from your system. > Do not use, copy or disclose the information in any way nor act in > reliance on it and notify the sender immediately. > Please note that the BBC monitors e-mails sent or received. > Further communication will signify your consent to this. > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user@ > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Can-FFMPEG-encode-DVCPRO-HD-tp3834258p3854976.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From krueger at signal7.de Thu Sep 29 11:43:56 2011 From: krueger at signal7.de (=?iso-8859-1?Q?Robert_Kr=FCger?=) Date: Thu, 29 Sep 2011 11:43:56 +0200 Subject: [FFmpeg-user] image stabilization avfilter options? Message-ID: Hi, does anyone know of an avfilter that allows to reduce camera shake of raw footage by applying some image stabilization algorithm? AFAIR there was someone who had experimented with that quite a while ago but I have never heard anything of that since. Any pointers would be appreciated. Cheers, Robert From blekenbleu at gmail.com Thu Sep 29 16:31:57 2011 From: blekenbleu at gmail.com (Steven Weed) Date: Thu, 29 Sep 2011 10:31:57 -0400 Subject: [FFmpeg-user] anomalous JPEG resolution Message-ID: Resolutions (and file sizes) for JPEG files from .MOV frames vary based on extraction sequence. Specifically, the third and fourth files are always larger (more than 2x average size) basically independent of the corresponding frame. Third images typically resolve 80 lines horizontally, while others resolve 70. For the following example, changing "-ss 0.6" to "-ss 0.3" yields different files but with similar relative sizes and resolutions. PNG output shows similar resolution behavior, although file sizes vary less. Can other command-line options more consistently extract still images of highest resolution? Example: * captured short iPad 1280x72 videos of ITU Fax chart, which includes targets of converging black/white lines. * transferred to OS X 10.6 iMac, converted to JPEG, e.g. $ ffmpeg -b 2000K -ss 0.6 -vframes 10 -i IMG_0031.MOV IMG31ss.6_v%d.jpg ffmpeg version 0.7.4, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 15 2011 10:14:37 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --mandir=/opt/local/share/man --enable-shared --enable-pthreads --disable-indevs --cc=/usr/bin/gcc-4.2 --arch=x86_64 --enable-yasm --enable-nonfree --enable-libfaac libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 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 Seems stream 0 codec frame rate differs from container frame rate: 1200.00 (1200/1) -> 29.97 (30000/1001) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'IMG_0031.MOV': Metadata: major_brand : qt minor_version : 0 compatible_brands: qt creation_time : 2011-09-26 18:41:29 Duration: 00:00:01.09, start: 0.000000, bitrate: 10384 kb/s Stream #0.0(und): Video: h264 (Baseline), yuv420p, 1280x720, 10516 kb/s, 29.95 fps, 29.97 tbr, 600 tbn, 1200 tbc Metadata: creation_time : 2011-09-26 18:41:29 Stream #0.1(und): Audio: aac, 44100 Hz, mono, s16, 59 kb/s Metadata: creation_time : 2011-09-26 18:41:29 Incompatible pixel format 'yuv420p' for codec 'mjpeg', auto-selecting format 'yuvj420p' [buffer @ 0x101732780] w:1280 h:720 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param: [buffersink @ 0x101732980] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' [scale @ 0x101732c80] w:1280 h:720 fmt:yuv420p -> w:1280 h:720 fmt:yuvj420p flags:0x4 Output #0, image2, to 'IMG31ss.6_v%d.jpg': Metadata: major_brand : qt minor_version : 0 compatible_brands: qt creation_time : 2011-09-26 18:41:29 encoder : Lavf52.110.0 Stream #0.0(und): Video: mjpeg, yuvj420p, 1280x720, q=2-31, 200 kb/s, 90k tbn, 29.97 tbc Metadata: creation_time : 2011-09-26 18:41:29 Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop, [?] for help frame= 10 fps= 0 q=20.0 Lsize= -0kB time=00:00:00.33 bitrate= -0.5kbits/s dup=0 drop=17 video:722kB audio:0kB global headers:0kB muxing overhead -100.002975% $ ls -lt | head -11 total 72456 -rw-r--r-- 1 stevenweed staff 72198 Sep 29 09:31 IMG31ss.6_v1.jpg -rw-r--r-- 1 stevenweed staff 45372 Sep 29 09:31 IMG31ss.6_v10.jpg -rw-r--r-- 1 stevenweed staff 85183 Sep 29 09:31 IMG31ss.6_v2.jpg -rw-r--r-- 1 stevenweed staff 106895 Sep 29 09:31 IMG31ss.6_v3.jpg -rw-r--r-- 1 stevenweed staff 107535 Sep 29 09:31 IMG31ss.6_v4.jpg -rw-r--r-- 1 stevenweed staff 85583 Sep 29 09:31 IMG31ss.6_v5.jpg -rw-r--r-- 1 stevenweed staff 69671 Sep 29 09:31 IMG31ss.6_v6.jpg -rw-r--r-- 1 stevenweed staff 62969 Sep 29 09:31 IMG31ss.6_v7.jpg -rw-r--r-- 1 stevenweed staff 54815 Sep 29 09:31 IMG31ss.6_v8.jpg -rw-r--r-- 1 stevenweed staff 49305 Sep 29 09:31 IMG31ss.6_v9.jpg From rectalogic at rectalogic.com Thu Sep 29 17:11:31 2011 From: rectalogic at rectalogic.com (Andrew Wason) Date: Thu, 29 Sep 2011 11:11:31 -0400 Subject: [FFmpeg-user] Rotate video with ffmpeg library in iPhone application In-Reply-To: <1FDFE4BC-7526-4327-814E-80093E4AE7AC@ironroad.com> References: <1FDFE4BC-7526-4327-814E-80093E4AE7AC@ironroad.com> Message-ID: On Tue, Sep 27, 2011 at 10:13 PM, Gagandeep wrote: > Hi Andrew, > > I know this from command line. I want to do it through code > using library and need sample code for this. The example code for how to use libavfilter is here: http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/examples/filtering.c Andrew From stefasab at gmail.com Thu Sep 29 19:17:22 2011 From: stefasab at gmail.com (Stefano Sabatini) Date: Thu, 29 Sep 2011 19:17:22 +0200 Subject: [FFmpeg-user] image stabilization avfilter options? In-Reply-To: References: Message-ID: <20110929171722.GA31374@geppetto> On date Thursday 2011-09-29 11:43:56 +0200, Robert Kr?ger encoded: > Hi, > > does anyone know of an avfilter that allows to reduce camera shake > of raw footage by applying some image stabilization algorithm? AFAIR > there was someone who had experimented with that quite a while ago > but I have never heard anything of that since. > > Any pointers would be appreciated. deshake libavfilter filter branch bg G. Taylor: https://gitorious.org/ffmpeg/ffmpeg-deshake transcode stabilize filter: http://www.transcoding.org/transcode?Filter_Plugins/Filter_Stabilize Feel free to file a feature request on trac, if there is enough interest we may try to port/integrate it. -- ffmpeg-user random tip #6 Please follow netiquette rules while posting to ffmpeg-user: http://linux.sgms-centre.com/misc/netiquette.php From surferrob at aol.com Thu Sep 29 18:14:09 2011 From: surferrob at aol.com (Bob Russo) Date: Thu, 29 Sep 2011 12:14:09 -0400 Subject: [FFmpeg-user] Avid QuickTime Codec Examples Message-ID: <003a01cc7ec2$d1e14b60$75a3e220$@aol.com> I saw FFMPEG was looking for examples of Avid QT files so I uploaded sample files here: http://www.filefactory.com/file/ce0b206/n/Avid11RGB.mov http://www.filefactory.com/file/ce0b005/n/AVCI100.mov http://www.filefactory.com/file/ce0b0e0/n/AVCI50.mov From jd1008 at gmail.com Thu Sep 29 19:43:44 2011 From: jd1008 at gmail.com (JD) Date: Thu, 29 Sep 2011 10:43:44 -0700 Subject: [FFmpeg-user] Compilation failed In-Reply-To: <4E83E517.9090602@gmail.com> References: <4E83E517.9090602@gmail.com> Message-ID: <4E84AE50.8040609@gmail.com> A non-text attachment was scrubbed... Name: ffmpeg.build.log.bz2 Type: application/x-bzip Size: 13380 bytes Desc: not available URL: From gerardojuarez at buyteknet.info Thu Sep 29 20:05:18 2011 From: gerardojuarez at buyteknet.info (gerardo Juarez) Date: Thu, 29 Sep 2011 13:05:18 -0500 Subject: [FFmpeg-user] Compilation failed In-Reply-To: <4E84AE50.8040609@gmail.com> References: <4E83E517.9090602@gmail.com> <4E84AE50.8040609@gmail.com> Message-ID: <4E84B35E.4090304@buyteknet.info> Could you isolate the relevant section where the compilation fails and include it in your message please? JD wrote: > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From jai.kumar at memeo-inc.com Thu Sep 29 20:26:19 2011 From: jai.kumar at memeo-inc.com (Jai Kumar) Date: Thu, 29 Sep 2011 11:26:19 -0700 Subject: [FFmpeg-user] How to identify a video is 3D Message-ID: <4E84B84B.6080107@memeo-inc.com> I am working on a program which transcodes video files. if file is 3D, then convert to 2D if file is 2D, then don't do anything So how would I know the input video is 3D? and also which type of 3D is it..framepacked or split? I am unable to find any info about these two things in the metadata of the file. I would appreciate any help. Thanks. Jai From jd1008 at gmail.com Thu Sep 29 20:45:52 2011 From: jd1008 at gmail.com (JD) Date: Thu, 29 Sep 2011 11:45:52 -0700 Subject: [FFmpeg-user] Compilation failed In-Reply-To: <4E84B35E.4090304@buyteknet.info> References: <4E83E517.9090602@gmail.com> <4E84AE50.8040609@gmail.com> <4E84B35E.4090304@buyteknet.info> Message-ID: <4E84BCE0.30302@gmail.com> On 09/29/2011 11:05 AM, gerardo Juarez wrote: > Could you isolate the relevant section where the compilation fails and > include it in your message please? > > JD wrote: I included the full build log as a bzipp'ed attachment. Since perhaps the server truncates it, I am appending the relevant output below: ------------------------------- . . . CC libavcodec/crystalhd.o In file included from libavcodec/crystalhd.c:82:0: /usr/include/libcrystalhd/bc_dts_types.h:64:14: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'bc_bool_t' In file included from libavcodec/crystalhd.c:84:0: /usr/include/libcrystalhd/libcrystalhd_if.h:149:14: error: expected ')' before '*' token /usr/include/libcrystalhd/libcrystalhd_if.h:177:12: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:215:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:255:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:300:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:339:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:371:12: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:402:12: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:452:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:486:16: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:522:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:569:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:604:12: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:635:12: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:666:13: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:695:15: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:726:15: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:757:12: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:786:12: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:829:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:878:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:908:12: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:954:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:990:15: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1031:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1068:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1104:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1138:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1170:12: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1198:12: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1228:13: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1265:13: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1298:13: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1329:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1363:14: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1393:21: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1423:10: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1451:10: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1478:13: error: expected ')' before 'hDevice' /usr/include/libcrystalhd/libcrystalhd_if.h:1507:13: error: expected ')' before 'hDevice' libavcodec/crystalhd.c:125:5: error: expected specifier-qualifier-list before 'HANDLE' libavcodec/crystalhd.c:152:7: error: 'CHDContext' has no member named 'sWidth' libavcodec/crystalhd.c:153:7: warning: missing braces around initializer libavcodec/crystalhd.c:153:7: warning: (near initialization for 'options[0].default_val') libavcodec/crystalhd.c: In function 'id2subtype': libavcodec/crystalhd.c:177:20: error: 'CHDContext' has no member named 'is_nal' libavcodec/crystalhd.c: In function 'opaque_list_push': libavcodec/crystalhd.c:238:14: error: 'CHDContext' has no member named 'head' libavcodec/crystalhd.c:240:13: error: 'CHDContext' has no member named 'head' libavcodec/crystalhd.c:240:9: warning: statement with no effect libavcodec/crystalhd.c:242:39: error: 'CHDContext' has no member named 'tail' libavcodec/crystalhd.c:242:45: error: 'AVOption' has no member named 'fake_timestamp' libavcodec/crystalhd.c:242:33: warning: assignment makes integer from pointer without a cast libavcodec/crystalhd.c:243:13: error: 'CHDContext' has no member named 'tail' libavcodec/crystalhd.c:243:19: error: 'AVOption' has no member named 'next' libavcodec/crystalhd.c:243:9: warning: statement with no effect libavcodec/crystalhd.c:245:9: error: 'CHDContext' has no member named 'tail' libavcodec/crystalhd.c:245:5: warning: statement with no effect libavcodec/crystalhd.c: In function 'opaque_list_pop': libavcodec/crystalhd.c:261:28: error: 'CHDContext' has no member named 'head' libavcodec/crystalhd.c:261:24: warning: initialization from incompatible pointer type libavcodec/crystalhd.c:263:14: error: 'CHDContext' has no member named 'head' libavcodec/crystalhd.c:273:13: error: 'CHDContext' has no member named 'head' libavcodec/crystalhd.c:273:19: error: 'AVOption' has no member named 'fake_timestamp' libavcodec/crystalhd.c:273:36: warning: comparison between pointer and integer libavcodec/crystalhd.c:274:13: error: 'CHDContext' has no member named 'head' libavcodec/crystalhd.c:274:9: warning: statement with no effect libavcodec/crystalhd.c:276:18: error: 'CHDContext' has no member named 'head' libavcodec/crystalhd.c:276:24: error: 'AVOption' has no member named 'next' libavcodec/crystalhd.c:277:17: error: 'CHDContext' has no member named 'tail' libavcodec/crystalhd.c:277:30: error: 'CHDContext' has no member named 'head' libavcodec/crystalhd.c:277:13: warning: statement with no effect libavcodec/crystalhd.c:293:20: error: 'CHDContext' has no member named 'tail' libavcodec/crystalhd.c:293:16: warning: statement with no effect libavcodec/crystalhd.c: In function 'flush': libavcodec/crystalhd.c:317:9: error: 'CHDContext' has no member named 'last_picture' libavcodec/crystalhd.c:317:5: warning: statement with no effect libavcodec/crystalhd.c:318:9: error: 'CHDContext' has no member named 'output_ready' libavcodec/crystalhd.c:318:5: warning: statement with no effect libavcodec/crystalhd.c:319:9: error: 'CHDContext' has no member named 'need_second_field' libavcodec/crystalhd.c:319:5: warning: statement with no effect libavcodec/crystalhd.c:320:9: error: 'CHDContext' has no member named 'skip_next_output' libavcodec/crystalhd.c:320:5: warning: statement with no effect libavcodec/crystalhd.c:321:9: error: 'CHDContext' has no member named 'decode_wait' libavcodec/crystalhd.c:321:5: warning: statement with no effect libavcodec/crystalhd.c:327:5: error: implicit declaration of function 'DtsFlushInput' libavcodec/crystalhd.c:327:23: error: 'CHDContext' has no member named 'dev' libavcodec/crystalhd.c: In function 'uninit': libavcodec/crystalhd.c:334:5: error: 'HANDLE' undeclared (first use in this function) libavcodec/crystalhd.c:334:5: note: each undeclared identifier is reported only once for each function it appears in libavcodec/crystalhd.c:334:5: warning: statement with no effect libavcodec/crystalhd.c:334:12: error: expected ';' before 'device' libavcodec/crystalhd.c:336:5: error: 'device' undeclared (first use in this function) libavcodec/crystalhd.c:336:18: error: 'CHDContext' has no member named 'dev' libavcodec/crystalhd.c:336:5: warning: statement with no effect libavcodec/crystalhd.c:337:5: error: implicit declaration of function 'DtsStopDecoder' libavcodec/crystalhd.c:338:5: error: implicit declaration of function 'DtsCloseDecoder' libavcodec/crystalhd.c:339:5: error: implicit declaration of function 'DtsDeviceClose' libavcodec/crystalhd.c:341:25: error: 'CHDContext' has no member named 'parser' libavcodec/crystalhd.c:341:5: warning: passing argument 1 of 'av_parser_close' from incompatible pointer type libavcodec/avcodec.h:4110:6: note: expected 'struct AVCodecParserContext *' but argument is of type 'const struct AVOption *' libavcodec/crystalhd.c:342:13: error: 'CHDContext' has no member named 'bsfc' libavcodec/crystalhd.c:343:39: error: 'CHDContext' has no member named 'bsfc' libavcodec/crystalhd.c:343:9: warning: passing argument 1 of 'av_bitstream_filter_close' from incompatible pointer type libavcodec/avcodec.h:4138:6: note: expected 'struct AVBitStreamFilterContext *' but argument is of type 'const struct AVOption *' libavcodec/crystalhd.c:346:17: error: 'CHDContext' has no member named 'sps_pps_buf' libavcodec/crystalhd.c:346:5: warning: passing argument 1 of 'av_free' discards qualifiers from pointer target type ./libavutil/mem.h:97:6: note: expected 'void *' but argument is of type 'const struct AVOption *' libavcodec/crystalhd.c:351:13: error: 'CHDContext' has no member named 'head' libavcodec/crystalhd.c:352:31: error: 'CHDContext' has no member named 'head' libavcodec/crystalhd.c:352:27: warning: initialization from incompatible pointer type libavcodec/crystalhd.c: In function 'init': libavcodec/crystalhd.c:370:24: error: 'FALSE' undeclared (first use in this function) libavcodec/crystalhd.c:370:9: warning: initialization makes integer from pointer without a cast libavcodec/crystalhd.c:371:24: error: 'TRUE' undeclared (first use in this function) libavcodec/crystalhd.c:371:9: warning: initialization makes integer from pointer without a cast libavcodec/crystalhd.c:372:37: error: 'vdecFrameRate59_94' undeclared (first use in this function) libavcodec/crystalhd.c:372:35: error: invalid operands to binary | (have 'unsigned int' and 'const struct AVOption *') libavcodec/crystalhd.c:372:56: error: invalid operands to binary | (have 'const struct AVOption *' and 'int') libavcodec/crystalhd.c:372:9: warning: initialization makes integer from pointer without a cast libavcodec/crystalhd.c:384:21: error: 'vdecRESOLUTION_1080p23_976' undeclared (first use in this function) libavcodec/crystalhd.c:384:21: error: invalid operands to binary << (have 'const struct AVOption *' and 'int') libavcodec/crystalhd.c:383:46: error: invalid operands to binary | (have 'int' and 'const struct AVOption *') libavcodec/crystalhd.c:379:21: warning: initialization makes integer from pointer without a cast libavcodec/crystalhd.c:394:9: error: 'CHDContext' has no member named 'is_nal' libavcodec/crystalhd.c:394:5: warning: statement with no effect libavcodec/crystalhd.c:395:9: error: 'CHDContext' has no member named 'last_picture' libavcodec/crystalhd.c:395:5: warning: statement with no effect libavcodec/crystalhd.c:396:9: error: 'CHDContext' has no member named 'decode_wait' libavcodec/crystalhd.c:396:5: warning: statement with no effect libavcodec/crystalhd.c:415:17: error: 'CHDContext' has no member named 'bsfc' libavcodec/crystalhd.c:415:13: warning: statement with no effect libavcodec/crystalhd.c:416:22: error: 'CHDContext' has no member named 'bsfc' libavcodec/crystalhd.c:422:44: error: 'CHDContext' has no member named 'bsfc' libavcodec/crystalhd.c:423:40: warning: passing argument 1 of 'av_bitstream_filter_filter' from incompatible pointer type libavcodec/avcodec.h:4134:5: note: expected 'struct AVBitStreamFilterContext *' but argument is of type 'const struct AVOption *' libavcodec/crystalhd.c:425:17: error: 'CHDContext' has no member named 'sps_pps_buf' libavcodec/crystalhd.c:425:13: warning: statement with no effect libavcodec/crystalhd.c:426:17: error: 'CHDContext' has no member named 'sps_pps_size' libavcodec/crystalhd.c:426:13: warning: statement with no effect libavcodec/crystalhd.c:430:38: error: 'CHDContext' has no member named 'sps_pps_buf' libavcodec/crystalhd.c:430:32: warning: assignment from incompatible pointer type libavcodec/crystalhd.c:431:38: error: 'CHDContext' has no member named 'sps_pps_size' libavcodec/crystalhd.c:431:32: warning: assignment makes integer from pointer without a cast libavcodec/crystalhd.c:454:13: error: 'CHDContext' has no member named 'sWidth' libavcodec/crystalhd.c:456:43: error: 'CHDContext' has no member named 'sWidth' libavcodec/crystalhd.c:456:37: warning: assignment makes integer from pointer without a cast libavcodec/crystalhd.c:462:5: error: implicit declaration of function 'DtsDeviceOpen' libavcodec/crystalhd.c:462:30: error: 'CHDContext' has no member named 'dev' libavcodec/crystalhd.c:468:5: error: implicit declaration of function 'DtsCrystalHDVersion' libavcodec/crystalhd.c:468:35: error: 'CHDContext' has no member named 'dev' libavcodec/crystalhd.c:474:9: error: 'CHDContext' has no member named 'is_70012' libavcodec/crystalhd.c:474:5: warning: statement with no effect libavcodec/crystalhd.c:476:13: error: 'CHDContext' has no member named 'is_70012' libavcodec/crystalhd.c:483:5: error: implicit declaration of function 'DtsSetInputFormat' libavcodec/crystalhd.c:483:33: error: 'CHDContext' has no member named 'dev' libavcodec/crystalhd.c:489:5: error: implicit declaration of function 'DtsOpenDecoder' libavcodec/crystalhd.c:489:30: error: 'CHDContext' has no member named 'dev' libavcodec/crystalhd.c:495:5: error: implicit declaration of function 'DtsSetColorSpace' libavcodec/crystalhd.c:495:32: error: 'CHDContext' has no member named 'dev' libavcodec/crystalhd.c:500:5: error: implicit declaration of function 'DtsStartDecoder' libavcodec/crystalhd.c:500:31: error: 'CHDContext' has no member named 'dev' libavcodec/crystalhd.c:505:5: error: implicit declaration of function 'DtsStartCapture' libavcodec/crystalhd.c:505:31: error: 'CHDContext' has no member named 'dev' libavcodec/crystalhd.c:512:13: error: 'CHDContext' has no member named 'parser' libavcodec/crystalhd.c:512:9: warning: statement with no effect libavcodec/crystalhd.c:513:18: error: 'CHDContext' has no member named 'parser' libavcodec/crystalhd.c:517:13: error: 'CHDContext' has no member named 'parser' libavcodec/crystalhd.c:517:9: error: assignment of read-only location '*(const struct AVOption *)&' libavcodec/crystalhd.c:517:9: warning: statement with no effect libavcodec/crystalhd.c: In function 'copy_frame': libavcodec/crystalhd.c:542:53: error: 'VDEC_FLAG_BOTTOMFIELD' undeclared (first use in this function) libavcodec/crystalhd.c:542:51: error: invalid operands to binary & (have 'uint32_t' and 'const struct AVOption *') libavcodec/crystalhd.c:544:55: error: 'VDEC_FLAG_BOTTOM_FIRST' undeclared (first use in this function) libavcodec/crystalhd.c:544:53: error: invalid operands to binary & (have 'uint32_t' and 'const struct AVOption *') libavcodec/crystalhd.c:577:5: error: implicit declaration of function 'DtsGetDriverStatus' libavcodec/crystalhd.c:577:34: error: 'CHDContext' has no member named 'dev' libavcodec/crystalhd.c:601:50: error: 'VDEC_FLAG_UNKNOWN_SRC' undeclared (first use in this function) libavcodec/crystalhd.c:601:48: error: invalid operands to binary & (have 'uint32_t' and 'const struct AVOption *') libavcodec/crystalhd.c:602:28: error: 'CHDContext' has no member named 'need_second_field' libavcodec/crystalhd.c:612:47: error: 'CHDContext' has no member named 'last_picture' libavcodec/crystalhd.c:612:40: warning: comparison between pointer and integer libavcodec/crystalhd.c:612:70: error: 'CHDContext' has no member named 'need_second_field' libavcodec/crystalhd.c:619:43: error: 'VDEC_FLAG_INTERLACED_SRC' undeclared (first use in this function) libavcodec/crystalhd.c:619:41: error: invalid operands to binary & (have 'uint32_t' and 'const struct AVOption *') libavcodec/crystalhd.c:630:35: error: 'CHDContext' has no member named 'need_second_field' libavcodec/crystalhd.c:633:9: error: 'CHDContext' has no member named 'need_second_field' libavcodec/crystalhd.c:633:50: error: 'CHDContext' has no member named 'need_second_field' libavcodec/crystalhd.c:633:5: warning: statement with no effect libavcodec/crystalhd.c:645:13: error: 'CHDContext' has no member named 'is_70012' libavcodec/crystalhd.c:691:14: error: 'CHDContext' has no member named 'need_second_field' libavcodec/crystalhd.c:703:47: error: invalid operands to binary & (have 'uint32_t' and 'const struct AVOption *') libavcodec/crystalhd.c:713:16: error: 'CHDContext' has no member named 'need_second_field' libavcodec/crystalhd.c:714:36: error: invalid operands to binary & (have 'uint32_t' and 'const struct AVOption *') libavcodec/crystalhd.c: In function 'receive_frame': libavcodec/crystalhd.c:728:5: error: 'HANDLE' undeclared (first use in this function) libavcodec/crystalhd.c:728:5: warning: statement with no effect libavcodec/crystalhd.c:728:12: error: expected ';' before 'dev' libavcodec/crystalhd.c:733:5: error: implicit declaration of function 'DtsProcOutputNoCopy' libavcodec/crystalhd.c:733:31: error: 'dev' undeclared (first use in this function) libavcodec/crystalhd.c:742:21: error: 'CHDContext' has no member named 'last_picture' libavcodec/crystalhd.c:742:36: warning: comparison between pointer and integer libavcodec/crystalhd.c:747:21: error: 'CHDContext' has no member named 'last_picture' libavcodec/crystalhd.c:747:17: warning: statement with no effect libavcodec/crystalhd.c:751:54: error: 'CHDContext' has no member named 'bframe_bug' libavcodec/crystalhd.c:754:21: error: 'CHDContext' has no member named 'last_picture' libavcodec/crystalhd.c:754:17: error: lvalue required as increment operand libavcodec/crystalhd.c:754:17: warning: statement with no effect libavcodec/crystalhd.c:755:17: error: implicit declaration of function 'DtsReleaseOutputBuffs' libavcodec/crystalhd.c:755:50: error: 'FALSE' undeclared (first use in this function) libavcodec/crystalhd.c:761:21: error: 'CHDContext' has no member named 'last_picture' libavcodec/crystalhd.c:761:40: warning: comparison between pointer and integer libavcodec/crystalhd.c:774:20: error: 'CHDContext' has no member named 'last_picture' libavcodec/crystalhd.c:774:16: warning: statement with no effect libavcodec/crystalhd.c:780:21: error: 'CHDContext' has no member named 'last_picture' libavcodec/crystalhd.c:780:17: error: lvalue required as increment operand libavcodec/crystalhd.c:780:17: warning: statement with no effect libavcodec/crystalhd.c: In function 'decode': libavcodec/crystalhd.c:811:5: error: 'HANDLE' undeclared (first use in this function) libavcodec/crystalhd.c:811:5: warning: statement with no effect libavcodec/crystalhd.c:811:12: error: expected ';' before 'dev' libavcodec/crystalhd.c:812:5: warning: ISO C90 forbids mixed declarations and code libavcodec/crystalhd.c:817:34: error: 'CHDContext' has no member named 'bframe_bug' libavcodec/crystalhd.c:823:13: error: 'CHDContext' has no member named 'bframe_bug' libavcodec/crystalhd.c:823:9: warning: statement with no effect libavcodec/crystalhd.c:824:40: error: 'CHDContext' has no member named 'bframe_bug' libavcodec/crystalhd.c:830:13: error: 'CHDContext' has no member named 'bframe_bug' libavcodec/crystalhd.c:830:9: warning: statement with no effect libavcodec/crystalhd.c:834:9: error: implicit declaration of function 'DtsTxFreeSize' libavcodec/crystalhd.c:834:50: error: 'dev' undeclared (first use in this function) libavcodec/crystalhd.c:836:17: error: 'CHDContext' has no member named 'parser' libavcodec/crystalhd.c:841:21: error: 'CHDContext' has no member named 'bsfc' libavcodec/crystalhd.c:842:54: error: 'CHDContext' has no member named 'bsfc' libavcodec/crystalhd.c:844:50: warning: passing argument 1 of 'av_bitstream_filter_filter' from incompatible pointer type libavcodec/avcodec.h:4134:5: note: expected 'struct AVBitStreamFilterContext *' but argument is of type 'const struct AVOption *' libavcodec/crystalhd.c:851:38: error: 'CHDContext' has no member named 'parser' libavcodec/crystalhd.c:851:46: error: 'AVOption' has no member named 'priv_data' libavcodec/crystalhd.c:851:34: warning: initialization from incompatible pointer type libavcodec/crystalhd.c:853:46: error: 'CHDContext' has no member named 'parser' libavcodec/crystalhd.c:855:42: warning: passing argument 1 of 'av_parser_parse2' from incompatible pointer type libavcodec/avcodec.h:4099:5: note: expected 'struct AVCodecParserContext *' but argument is of type 'const struct AVOption *' libavcodec/crystalhd.c:898:13: error: implicit declaration of function 'DtsProcInput' libavcodec/crystalhd.c:918:13: error: 'CHDContext' has no member named 'skip_next_output' libavcodec/crystalhd.c:920:13: error: 'CHDContext' has no member named 'skip_next_output' libavcodec/crystalhd.c:920:9: warning: statement with no effect libavcodec/crystalhd.c:939:13: error: 'CHDContext' has no member named 'output_ready' libavcodec/crystalhd.c:939:28: warning: comparison between pointer and integer libavcodec/crystalhd.c:941:17: error: 'CHDContext' has no member named 'output_ready' libavcodec/crystalhd.c:941:13: error: lvalue required as increment operand libavcodec/crystalhd.c:941:13: warning: statement with no effect libavcodec/crystalhd.c:953:13: error: 'CHDContext' has no member named 'decode_wait' libavcodec/crystalhd.c:953:9: warning: statement with no effect libavcodec/crystalhd.c:981:28: error: 'CHDContext' has no member named 'decode_wait' libavcodec/crystalhd.c:981:17: warning: passing argument 1 of 'usleep' makes integer from pointer without a cast /usr/include/unistd.h:457:12: note: expected '__useconds_t' but argument is of type 'const struct AVOption *' libavcodec/crystalhd.c:998:17: error: 'CHDContext' has no member named 'skip_next_output' libavcodec/crystalhd.c:998:13: warning: statement with no effect libavcodec/crystalhd.c:1009:16: error: 'CHDContext' has no member named 'decode_wait' libavcodec/crystalhd.c:1009:5: warning: passing argument 1 of 'usleep' makes integer from pointer without a cast /usr/include/unistd.h:457:12: note: expected '__useconds_t' but argument is of type 'const struct AVOption *' make: *** [libavcodec/crystalhd.o] Error 1 error: Bad exit status from /var/tmp/rpm-tmp.270FFX (%build) RPM build errors: Recursion depth(17) greater than max(16) Recursion depth(17) greater than max(16) Recursion depth(17) greater than max(16) Recursion depth(17) greater than max(16) Recursion depth(17) greater than max(16) Recursion depth(17) greater than max(16) Bad exit status from /var/tmp/rpm-tmp.270FFX (%build) From lou at lrcd.com Thu Sep 29 20:49:28 2011 From: lou at lrcd.com (Lou) Date: Thu, 29 Sep 2011 10:49:28 -0800 Subject: [FFmpeg-user] Avid QuickTime Codec Examples In-Reply-To: <003a01cc7ec2$d1e14b60$75a3e220$@aol.com> References: <003a01cc7ec2$d1e14b60$75a3e220$@aol.com> Message-ID: <20110929104928.02a4264f@lrcd.com> On Thu, 29 Sep 2011 12:14:09 -0400 "Bob Russo" wrote: > I saw FFMPEG was looking for examples of Avid QT files so I uploaded > sample files here: > > > > http://www.filefactory.com/file/ce0b206/n/Avid11RGB.mov > > > > http://www.filefactory.com/file/ce0b005/n/AVCI100.mov > > > > http://www.filefactory.com/file/ce0b0e0/n/AVCI50.mov Thank you for the samples, but 934.12 MB (Avid11RGB.mov) is very large. Do you have a smaller sample? If you do have something smaller you could use a better service such as mediafire (200 MB max) or datafilehost (100 MB max). From h.reindl at thelounge.net Thu Sep 29 20:51:20 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Thu, 29 Sep 2011 20:51:20 +0200 Subject: [FFmpeg-user] Compilation failed In-Reply-To: <4E84BCE0.30302@gmail.com> References: <4E83E517.9090602@gmail.com> <4E84AE50.8040609@gmail.com> <4E84B35E.4090304@buyteknet.info> <4E84BCE0.30302@gmail.com> Message-ID: <4E84BE28.7000902@thelounge.net> what is your configure-line? what is your distribution? what is your compiler and what version? Am 29.09.2011 20:45, schrieb JD: > On 09/29/2011 11:05 AM, gerardo Juarez wrote: >> Could you isolate the relevant section where the compilation fails and include it in your message please? >> >> JD wrote: > I included the full build log as a bzipp'ed attachment. > > Since perhaps the server truncates it, I am appending the relevant output below: > > ------------------------------- > > . > . > . > CC libavcodec/crystalhd.o > In file included from libavcodec/crystalhd.c:82:0: > /usr/include/libcrystalhd/bc_dts_types.h:64:14: error: expected '=', ',', ';', 'asm' or '__attribute__' before > 'bc_bool_t' > In file included from libavcodec/crystalhd.c:84:0: > /usr/include/libcrystalhd/libcrystalhd_if.h:149:14: error: expected ')' before '*' token > /usr/include/libcrystalhd/libcrystalhd_if.h:177:12: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:215:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:255:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:300:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:339:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:371:12: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:402:12: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:452:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:486:16: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:522:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:569:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:604:12: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:635:12: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:666:13: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:695:15: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:726:15: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:757:12: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:786:12: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:829:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:878:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:908:12: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:954:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:990:15: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1031:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1068:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1104:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1138:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1170:12: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1198:12: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1228:13: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1265:13: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1298:13: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1329:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1363:14: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1393:21: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1423:10: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1451:10: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1478:13: error: expected ')' before 'hDevice' > /usr/include/libcrystalhd/libcrystalhd_if.h:1507:13: error: expected ')' before 'hDevice' > libavcodec/crystalhd.c:125:5: error: expected specifier-qualifier-list before 'HANDLE' > libavcodec/crystalhd.c:152:7: error: 'CHDContext' has no member named 'sWidth' > libavcodec/crystalhd.c:153:7: warning: missing braces around initializer > libavcodec/crystalhd.c:153:7: warning: (near initialization for 'options[0].default_val') > libavcodec/crystalhd.c: In function 'id2subtype': > libavcodec/crystalhd.c:177:20: error: 'CHDContext' has no member named 'is_nal' > libavcodec/crystalhd.c: In function 'opaque_list_push': > libavcodec/crystalhd.c:238:14: error: 'CHDContext' has no member named 'head' > libavcodec/crystalhd.c:240:13: error: 'CHDContext' has no member named 'head' > libavcodec/crystalhd.c:240:9: warning: statement with no effect > libavcodec/crystalhd.c:242:39: error: 'CHDContext' has no member named 'tail' > libavcodec/crystalhd.c:242:45: error: 'AVOption' has no member named 'fake_timestamp' > libavcodec/crystalhd.c:242:33: warning: assignment makes integer from pointer without a cast > libavcodec/crystalhd.c:243:13: error: 'CHDContext' has no member named 'tail' > libavcodec/crystalhd.c:243:19: error: 'AVOption' has no member named 'next' > libavcodec/crystalhd.c:243:9: warning: statement with no effect > libavcodec/crystalhd.c:245:9: error: 'CHDContext' has no member named 'tail' > libavcodec/crystalhd.c:245:5: warning: statement with no effect > libavcodec/crystalhd.c: In function 'opaque_list_pop': > libavcodec/crystalhd.c:261:28: error: 'CHDContext' has no member named 'head' > libavcodec/crystalhd.c:261:24: warning: initialization from incompatible pointer type > libavcodec/crystalhd.c:263:14: error: 'CHDContext' has no member named 'head' > libavcodec/crystalhd.c:273:13: error: 'CHDContext' has no member named 'head' > libavcodec/crystalhd.c:273:19: error: 'AVOption' has no member named 'fake_timestamp' > libavcodec/crystalhd.c:273:36: warning: comparison between pointer and integer > libavcodec/crystalhd.c:274:13: error: 'CHDContext' has no member named 'head' > libavcodec/crystalhd.c:274:9: warning: statement with no effect > libavcodec/crystalhd.c:276:18: error: 'CHDContext' has no member named 'head' > libavcodec/crystalhd.c:276:24: error: 'AVOption' has no member named 'next' > libavcodec/crystalhd.c:277:17: error: 'CHDContext' has no member named 'tail' > libavcodec/crystalhd.c:277:30: error: 'CHDContext' has no member named 'head' > libavcodec/crystalhd.c:277:13: warning: statement with no effect > libavcodec/crystalhd.c:293:20: error: 'CHDContext' has no member named 'tail' > libavcodec/crystalhd.c:293:16: warning: statement with no effect > libavcodec/crystalhd.c: In function 'flush': > libavcodec/crystalhd.c:317:9: error: 'CHDContext' has no member named 'last_picture' > libavcodec/crystalhd.c:317:5: warning: statement with no effect > libavcodec/crystalhd.c:318:9: error: 'CHDContext' has no member named 'output_ready' > libavcodec/crystalhd.c:318:5: warning: statement with no effect > libavcodec/crystalhd.c:319:9: error: 'CHDContext' has no member named 'need_second_field' > libavcodec/crystalhd.c:319:5: warning: statement with no effect > libavcodec/crystalhd.c:320:9: error: 'CHDContext' has no member named 'skip_next_output' > libavcodec/crystalhd.c:320:5: warning: statement with no effect > libavcodec/crystalhd.c:321:9: error: 'CHDContext' has no member named 'decode_wait' > libavcodec/crystalhd.c:321:5: warning: statement with no effect > libavcodec/crystalhd.c:327:5: error: implicit declaration of function 'DtsFlushInput' > libavcodec/crystalhd.c:327:23: error: 'CHDContext' has no member named 'dev' > libavcodec/crystalhd.c: In function 'uninit': > libavcodec/crystalhd.c:334:5: error: 'HANDLE' undeclared (first use in this function) > libavcodec/crystalhd.c:334:5: note: each undeclared identifier is reported only once for each function it appears in > libavcodec/crystalhd.c:334:5: warning: statement with no effect > libavcodec/crystalhd.c:334:12: error: expected ';' before 'device' > libavcodec/crystalhd.c:336:5: error: 'device' undeclared (first use in this function) > libavcodec/crystalhd.c:336:18: error: 'CHDContext' has no member named 'dev' > libavcodec/crystalhd.c:336:5: warning: statement with no effect > libavcodec/crystalhd.c:337:5: error: implicit declaration of function 'DtsStopDecoder' > libavcodec/crystalhd.c:338:5: error: implicit declaration of function 'DtsCloseDecoder' > libavcodec/crystalhd.c:339:5: error: implicit declaration of function 'DtsDeviceClose' > libavcodec/crystalhd.c:341:25: error: 'CHDContext' has no member named 'parser' > libavcodec/crystalhd.c:341:5: warning: passing argument 1 of 'av_parser_close' from incompatible pointer type > libavcodec/avcodec.h:4110:6: note: expected 'struct AVCodecParserContext *' but argument is of type 'const struct > AVOption *' > libavcodec/crystalhd.c:342:13: error: 'CHDContext' has no member named 'bsfc' > libavcodec/crystalhd.c:343:39: error: 'CHDContext' has no member named 'bsfc' > libavcodec/crystalhd.c:343:9: warning: passing argument 1 of 'av_bitstream_filter_close' from incompatible pointer > type > libavcodec/avcodec.h:4138:6: note: expected 'struct AVBitStreamFilterContext *' but argument is of type 'const > struct AVOption *' > libavcodec/crystalhd.c:346:17: error: 'CHDContext' has no member named 'sps_pps_buf' > libavcodec/crystalhd.c:346:5: warning: passing argument 1 of 'av_free' discards qualifiers from pointer target type > ./libavutil/mem.h:97:6: note: expected 'void *' but argument is of type 'const struct AVOption *' > libavcodec/crystalhd.c:351:13: error: 'CHDContext' has no member named 'head' > libavcodec/crystalhd.c:352:31: error: 'CHDContext' has no member named 'head' > libavcodec/crystalhd.c:352:27: warning: initialization from incompatible pointer type > libavcodec/crystalhd.c: In function 'init': > libavcodec/crystalhd.c:370:24: error: 'FALSE' undeclared (first use in this function) > libavcodec/crystalhd.c:370:9: warning: initialization makes integer from pointer without a cast > libavcodec/crystalhd.c:371:24: error: 'TRUE' undeclared (first use in this function) > libavcodec/crystalhd.c:371:9: warning: initialization makes integer from pointer without a cast > libavcodec/crystalhd.c:372:37: error: 'vdecFrameRate59_94' undeclared (first use in this function) > libavcodec/crystalhd.c:372:35: error: invalid operands to binary | (have 'unsigned int' and 'const struct AVOption *') > libavcodec/crystalhd.c:372:56: error: invalid operands to binary | (have 'const struct AVOption *' and 'int') > libavcodec/crystalhd.c:372:9: warning: initialization makes integer from pointer without a cast > libavcodec/crystalhd.c:384:21: error: 'vdecRESOLUTION_1080p23_976' undeclared (first use in this function) > libavcodec/crystalhd.c:384:21: error: invalid operands to binary << (have 'const struct AVOption *' and 'int') > libavcodec/crystalhd.c:383:46: error: invalid operands to binary | (have 'int' and 'const struct AVOption *') > libavcodec/crystalhd.c:379:21: warning: initialization makes integer from pointer without a cast > libavcodec/crystalhd.c:394:9: error: 'CHDContext' has no member named 'is_nal' > libavcodec/crystalhd.c:394:5: warning: statement with no effect > libavcodec/crystalhd.c:395:9: error: 'CHDContext' has no member named 'last_picture' > libavcodec/crystalhd.c:395:5: warning: statement with no effect > libavcodec/crystalhd.c:396:9: error: 'CHDContext' has no member named 'decode_wait' > libavcodec/crystalhd.c:396:5: warning: statement with no effect > libavcodec/crystalhd.c:415:17: error: 'CHDContext' has no member named 'bsfc' > libavcodec/crystalhd.c:415:13: warning: statement with no effect > libavcodec/crystalhd.c:416:22: error: 'CHDContext' has no member named 'bsfc' > libavcodec/crystalhd.c:422:44: error: 'CHDContext' has no member named 'bsfc' > libavcodec/crystalhd.c:423:40: warning: passing argument 1 of 'av_bitstream_filter_filter' from incompatible > pointer type > libavcodec/avcodec.h:4134:5: note: expected 'struct AVBitStreamFilterContext *' but argument is of type 'const > struct AVOption *' > libavcodec/crystalhd.c:425:17: error: 'CHDContext' has no member named 'sps_pps_buf' > libavcodec/crystalhd.c:425:13: warning: statement with no effect > libavcodec/crystalhd.c:426:17: error: 'CHDContext' has no member named 'sps_pps_size' > libavcodec/crystalhd.c:426:13: warning: statement with no effect > libavcodec/crystalhd.c:430:38: error: 'CHDContext' has no member named 'sps_pps_buf' > libavcodec/crystalhd.c:430:32: warning: assignment from incompatible pointer type > libavcodec/crystalhd.c:431:38: error: 'CHDContext' has no member named 'sps_pps_size' > libavcodec/crystalhd.c:431:32: warning: assignment makes integer from pointer without a cast > libavcodec/crystalhd.c:454:13: error: 'CHDContext' has no member named 'sWidth' > libavcodec/crystalhd.c:456:43: error: 'CHDContext' has no member named 'sWidth' > libavcodec/crystalhd.c:456:37: warning: assignment makes integer from pointer without a cast > libavcodec/crystalhd.c:462:5: error: implicit declaration of function 'DtsDeviceOpen' > libavcodec/crystalhd.c:462:30: error: 'CHDContext' has no member named 'dev' > libavcodec/crystalhd.c:468:5: error: implicit declaration of function 'DtsCrystalHDVersion' > libavcodec/crystalhd.c:468:35: error: 'CHDContext' has no member named 'dev' > libavcodec/crystalhd.c:474:9: error: 'CHDContext' has no member named 'is_70012' > libavcodec/crystalhd.c:474:5: warning: statement with no effect > libavcodec/crystalhd.c:476:13: error: 'CHDContext' has no member named 'is_70012' > libavcodec/crystalhd.c:483:5: error: implicit declaration of function 'DtsSetInputFormat' > libavcodec/crystalhd.c:483:33: error: 'CHDContext' has no member named 'dev' > libavcodec/crystalhd.c:489:5: error: implicit declaration of function 'DtsOpenDecoder' > libavcodec/crystalhd.c:489:30: error: 'CHDContext' has no member named 'dev' > libavcodec/crystalhd.c:495:5: error: implicit declaration of function 'DtsSetColorSpace' > libavcodec/crystalhd.c:495:32: error: 'CHDContext' has no member named 'dev' > libavcodec/crystalhd.c:500:5: error: implicit declaration of function 'DtsStartDecoder' > libavcodec/crystalhd.c:500:31: error: 'CHDContext' has no member named 'dev' > libavcodec/crystalhd.c:505:5: error: implicit declaration of function 'DtsStartCapture' > libavcodec/crystalhd.c:505:31: error: 'CHDContext' has no member named 'dev' > libavcodec/crystalhd.c:512:13: error: 'CHDContext' has no member named 'parser' > libavcodec/crystalhd.c:512:9: warning: statement with no effect > libavcodec/crystalhd.c:513:18: error: 'CHDContext' has no member named 'parser' > libavcodec/crystalhd.c:517:13: error: 'CHDContext' has no member named 'parser' > libavcodec/crystalhd.c:517:9: error: assignment of read-only location '*(const struct AVOption > *)&' > libavcodec/crystalhd.c:517:9: warning: statement with no effect > libavcodec/crystalhd.c: In function 'copy_frame': > libavcodec/crystalhd.c:542:53: error: 'VDEC_FLAG_BOTTOMFIELD' undeclared (first use in this function) > libavcodec/crystalhd.c:542:51: error: invalid operands to binary & (have 'uint32_t' and 'const struct AVOption *') > libavcodec/crystalhd.c:544:55: error: 'VDEC_FLAG_BOTTOM_FIRST' undeclared (first use in this function) > libavcodec/crystalhd.c:544:53: error: invalid operands to binary & (have 'uint32_t' and 'const struct AVOption *') > libavcodec/crystalhd.c:577:5: error: implicit declaration of function 'DtsGetDriverStatus' > libavcodec/crystalhd.c:577:34: error: 'CHDContext' has no member named 'dev' > libavcodec/crystalhd.c:601:50: error: 'VDEC_FLAG_UNKNOWN_SRC' undeclared (first use in this function) > libavcodec/crystalhd.c:601:48: error: invalid operands to binary & (have 'uint32_t' and 'const struct AVOption *') > libavcodec/crystalhd.c:602:28: error: 'CHDContext' has no member named 'need_second_field' > libavcodec/crystalhd.c:612:47: error: 'CHDContext' has no member named 'last_picture' > libavcodec/crystalhd.c:612:40: warning: comparison between pointer and integer > libavcodec/crystalhd.c:612:70: error: 'CHDContext' has no member named 'need_second_field' > libavcodec/crystalhd.c:619:43: error: 'VDEC_FLAG_INTERLACED_SRC' undeclared (first use in this function) > libavcodec/crystalhd.c:619:41: error: invalid operands to binary & (have 'uint32_t' and 'const struct AVOption *') > libavcodec/crystalhd.c:630:35: error: 'CHDContext' has no member named 'need_second_field' > libavcodec/crystalhd.c:633:9: error: 'CHDContext' has no member named 'need_second_field' > libavcodec/crystalhd.c:633:50: error: 'CHDContext' has no member named 'need_second_field' > libavcodec/crystalhd.c:633:5: warning: statement with no effect > libavcodec/crystalhd.c:645:13: error: 'CHDContext' has no member named 'is_70012' > libavcodec/crystalhd.c:691:14: error: 'CHDContext' has no member named 'need_second_field' > libavcodec/crystalhd.c:703:47: error: invalid operands to binary & (have 'uint32_t' and 'const struct AVOption *') > libavcodec/crystalhd.c:713:16: error: 'CHDContext' has no member named 'need_second_field' > libavcodec/crystalhd.c:714:36: error: invalid operands to binary & (have 'uint32_t' and 'const struct AVOption *') > libavcodec/crystalhd.c: In function 'receive_frame': > libavcodec/crystalhd.c:728:5: error: 'HANDLE' undeclared (first use in this function) > libavcodec/crystalhd.c:728:5: warning: statement with no effect > libavcodec/crystalhd.c:728:12: error: expected ';' before 'dev' > libavcodec/crystalhd.c:733:5: error: implicit declaration of function 'DtsProcOutputNoCopy' > libavcodec/crystalhd.c:733:31: error: 'dev' undeclared (first use in this function) > libavcodec/crystalhd.c:742:21: error: 'CHDContext' has no member named 'last_picture' > libavcodec/crystalhd.c:742:36: warning: comparison between pointer and integer > libavcodec/crystalhd.c:747:21: error: 'CHDContext' has no member named 'last_picture' > libavcodec/crystalhd.c:747:17: warning: statement with no effect > libavcodec/crystalhd.c:751:54: error: 'CHDContext' has no member named 'bframe_bug' > libavcodec/crystalhd.c:754:21: error: 'CHDContext' has no member named 'last_picture' > libavcodec/crystalhd.c:754:17: error: lvalue required as increment operand > libavcodec/crystalhd.c:754:17: warning: statement with no effect > libavcodec/crystalhd.c:755:17: error: implicit declaration of function 'DtsReleaseOutputBuffs' > libavcodec/crystalhd.c:755:50: error: 'FALSE' undeclared (first use in this function) > libavcodec/crystalhd.c:761:21: error: 'CHDContext' has no member named 'last_picture' > libavcodec/crystalhd.c:761:40: warning: comparison between pointer and integer > libavcodec/crystalhd.c:774:20: error: 'CHDContext' has no member named 'last_picture' > libavcodec/crystalhd.c:774:16: warning: statement with no effect > libavcodec/crystalhd.c:780:21: error: 'CHDContext' has no member named 'last_picture' > libavcodec/crystalhd.c:780:17: error: lvalue required as increment operand > libavcodec/crystalhd.c:780:17: warning: statement with no effect > libavcodec/crystalhd.c: In function 'decode': > libavcodec/crystalhd.c:811:5: error: 'HANDLE' undeclared (first use in this function) > libavcodec/crystalhd.c:811:5: warning: statement with no effect > libavcodec/crystalhd.c:811:12: error: expected ';' before 'dev' > libavcodec/crystalhd.c:812:5: warning: ISO C90 forbids mixed declarations and code > libavcodec/crystalhd.c:817:34: error: 'CHDContext' has no member named 'bframe_bug' > libavcodec/crystalhd.c:823:13: error: 'CHDContext' has no member named 'bframe_bug' > libavcodec/crystalhd.c:823:9: warning: statement with no effect > libavcodec/crystalhd.c:824:40: error: 'CHDContext' has no member named 'bframe_bug' > libavcodec/crystalhd.c:830:13: error: 'CHDContext' has no member named 'bframe_bug' > libavcodec/crystalhd.c:830:9: warning: statement with no effect > libavcodec/crystalhd.c:834:9: error: implicit declaration of function 'DtsTxFreeSize' > libavcodec/crystalhd.c:834:50: error: 'dev' undeclared (first use in this function) > libavcodec/crystalhd.c:836:17: error: 'CHDContext' has no member named 'parser' > libavcodec/crystalhd.c:841:21: error: 'CHDContext' has no member named 'bsfc' > libavcodec/crystalhd.c:842:54: error: 'CHDContext' has no member named 'bsfc' > libavcodec/crystalhd.c:844:50: warning: passing argument 1 of 'av_bitstream_filter_filter' from incompatible > pointer type > libavcodec/avcodec.h:4134:5: note: expected 'struct AVBitStreamFilterContext *' but argument is of type 'const > struct AVOption *' > libavcodec/crystalhd.c:851:38: error: 'CHDContext' has no member named 'parser' > libavcodec/crystalhd.c:851:46: error: 'AVOption' has no member named 'priv_data' > libavcodec/crystalhd.c:851:34: warning: initialization from incompatible pointer type > libavcodec/crystalhd.c:853:46: error: 'CHDContext' has no member named 'parser' > libavcodec/crystalhd.c:855:42: warning: passing argument 1 of 'av_parser_parse2' from incompatible pointer type > libavcodec/avcodec.h:4099:5: note: expected 'struct AVCodecParserContext *' but argument is of type 'const struct > AVOption *' > libavcodec/crystalhd.c:898:13: error: implicit declaration of function 'DtsProcInput' > libavcodec/crystalhd.c:918:13: error: 'CHDContext' has no member named 'skip_next_output' > libavcodec/crystalhd.c:920:13: error: 'CHDContext' has no member named 'skip_next_output' > libavcodec/crystalhd.c:920:9: warning: statement with no effect > libavcodec/crystalhd.c:939:13: error: 'CHDContext' has no member named 'output_ready' > libavcodec/crystalhd.c:939:28: warning: comparison between pointer and integer > libavcodec/crystalhd.c:941:17: error: 'CHDContext' has no member named 'output_ready' > libavcodec/crystalhd.c:941:13: error: lvalue required as increment operand > libavcodec/crystalhd.c:941:13: warning: statement with no effect > libavcodec/crystalhd.c:953:13: error: 'CHDContext' has no member named 'decode_wait' > libavcodec/crystalhd.c:953:9: warning: statement with no effect > libavcodec/crystalhd.c:981:28: error: 'CHDContext' has no member named 'decode_wait' > libavcodec/crystalhd.c:981:17: warning: passing argument 1 of 'usleep' makes integer from pointer without a cast > /usr/include/unistd.h:457:12: note: expected '__useconds_t' but argument is of type 'const struct AVOption *' > libavcodec/crystalhd.c:998:17: error: 'CHDContext' has no member named 'skip_next_output' > libavcodec/crystalhd.c:998:13: warning: statement with no effect > libavcodec/crystalhd.c:1009:16: error: 'CHDContext' has no member named 'decode_wait' > libavcodec/crystalhd.c:1009:5: warning: passing argument 1 of 'usleep' makes integer from pointer without a cast > /usr/include/unistd.h:457:12: note: expected '__useconds_t' but argument is of type 'const struct AVOption *' > make: *** [libavcodec/crystalhd.o] Error 1 > error: Bad exit status from /var/tmp/rpm-tmp.270FFX (%build) > > > RPM build errors: > Recursion depth(17) greater than max(16) > Recursion depth(17) greater than max(16) > Recursion depth(17) greater than max(16) > Recursion depth(17) greater than max(16) > Recursion depth(17) greater than max(16) > Recursion depth(17) greater than max(16) > Bad exit status from /var/tmp/rpm-tmp.270FFX (%build) > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- Mit besten Gr??en, Reindl Harald the lounge interactive design GmbH A-1060 Vienna, Hofm?hlgasse 17 CTO / software-development / cms-solutions p: +43 (1) 595 3999 33, m: +43 (676) 40 221 40 icq: 154546673, http://www.thelounge.net/ http://www.thelounge.net/signature.asc.what.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From lou at lrcd.com Thu Sep 29 21:08:27 2011 From: lou at lrcd.com (Lou) Date: Thu, 29 Sep 2011 11:08:27 -0800 Subject: [FFmpeg-user] anomalous JPEG resolution In-Reply-To: References: Message-ID: <20110929110827.6795c76c@lrcd.com> On Thu, 29 Sep 2011 10:31:57 -0400 Steven Weed wrote: > Resolutions (and file sizes) for JPEG files from > .MOV frames vary based on extraction sequence. > Specifically, the third and fourth files are always larger > (more than 2x average size) basically independent > of the corresponding frame. > > Third images typically resolve 80 lines horizontally, while others > resolve 70. > For the following example, > changing "-ss 0.6" to "-ss 0.3" yields different files > but with similar relative sizes and resolutions. > PNG output shows similar resolution behavior, although file sizes > vary less. > > Can other command-line options more consistently extract > still images of highest resolution? > > Example: > * captured short iPad 1280x72 videos of ITU Fax chart, > which includes targets of converging black/white lines. > * transferred to OS X 10.6 iMac, converted to JPEG, e.g. > > $ ffmpeg -b 2000K -ss 0.6 -vframes 10 -i IMG_0031.MOV > IMG31ss.6_v%d.jpg ffmpeg version 0.7.4, Copyright (c) 2000-2011 the > FFmpeg developers built on Sep 15 2011 10:14:37 with gcc 4.2.1 (Apple > Inc. build 5666) (dot 3) Can you duplicate this if you move -ss after "-i IMG_0031.MOV"? Moving -ss as an output option changes its behavior: -ss after -i: Decode until -ss value, and then start encoding. -ss before -i: Seek before decoding. Can be significantly faster than the other method, but has the potential of not being frame accurate (depending on your input format). ...at least that's how I interpret it. Consider moving your other options as well because placing options before -i is generally meant to apply to the input only. From krueger at signal7.de Thu Sep 29 23:00:36 2011 From: krueger at signal7.de (=?iso-8859-1?Q?Robert_Kr=FCger?=) Date: Thu, 29 Sep 2011 23:00:36 +0200 Subject: [FFmpeg-user] image stabilization avfilter options? In-Reply-To: <20110929171722.GA31374@geppetto> References: <20110929171722.GA31374@geppetto> Message-ID: <94ACAE19-20DF-4570-B2F3-DB5EC9D927D7@signal7.de> On Sep 29, 2011, at 19:17 , Stefano Sabatini wrote: > On date Thursday 2011-09-29 11:43:56 +0200, Robert Kr?ger encoded: >> Hi, >> >> does anyone know of an avfilter that allows to reduce camera shake >> of raw footage by applying some image stabilization algorithm? AFAIR >> there was someone who had experimented with that quite a while ago >> but I have never heard anything of that since. >> >> Any pointers would be appreciated. > > deshake libavfilter filter branch bg G. Taylor: > https://gitorious.org/ffmpeg/ffmpeg-deshake > > transcode stabilize filter: > http://www.transcoding.org/transcode?Filter_Plugins/Filter_Stabilize > > Feel free to file a feature request on trac, if there is enough > interest we may try to port/integrate it. Thanks a lot for the links. The ticket is now #519. Regards, Robert From hardik.sharma22 at yahoo.com Fri Sep 30 00:56:39 2011 From: hardik.sharma22 at yahoo.com (Hardik Sharma) Date: Thu, 29 Sep 2011 15:56:39 -0700 (PDT) Subject: [FFmpeg-user] how to enable intra-refresh? Message-ID: <1317336999.57838.YahooMailNeo@web46204.mail.sp1.yahoo.com> I am trying to enable intra-refresh or IDR frame to replace keyint/I frames for all the time when new gop starts. I am giving following command - ffmpeg -y -s 352x288 -r 30 -pix_fmt yuv420p -f rawvideo -i foreman_cif.yuv -vcodec libx264 -y -b 512k -r 30 -preset fast -s 352x288 -flags +loop -cmp +chroma -flags2 -fastpskip -refs 6 -x264opts slice-max-size=300:bframes=1:keyint=20:no-interlaced=1:trellis=2:intra-refresh=1:subme=10 -f h264 -mbd 1 -profile high -deblockalpha 0 -deblockbeta 0 -threads 0 out.264 and getting error as - [libx264 @ 0xa896320] ref > 1 + intra-refresh is not supported [libx264 @ 0xa896320] intra-refresh is not compatible with open-gop my gop is not open as you can see there is keyint=20, which proves that the size of gop is 20 frames. PLease help me for this. Thanks? From michaelni at gmx.at Fri Sep 30 02:30:07 2011 From: michaelni at gmx.at (Michael Niedermayer) Date: Fri, 30 Sep 2011 02:30:07 +0200 Subject: [FFmpeg-user] [FFmpeg-devel] how to enable intra-refresh? In-Reply-To: <1317336999.57838.YahooMailNeo@web46204.mail.sp1.yahoo.com> References: <1317336999.57838.YahooMailNeo@web46204.mail.sp1.yahoo.com> Message-ID: <20110930003006.GR3513@kiste2> On Thu, Sep 29, 2011 at 03:56:39PM -0700, Hardik Sharma wrote: > I am trying to enable intra-refresh or IDR frame to replace keyint/I frames for all the time when new gop starts. I am giving following command - > > ffmpeg -y -s 352x288 -r 30 -pix_fmt yuv420p -f rawvideo -i foreman_cif.yuv -vcodec libx264 -y -b 512k -r 30 -preset fast -s 352x288 -flags +loop -cmp +chroma -flags2 -fastpskip -refs 6 -x264opts slice-max-size=300:bframes=1:keyint=20:no-interlaced=1:trellis=2:intra-refresh=1:subme=10 -f h264 -mbd 1 -profile high -deblockalpha 0 -deblockbeta 0 -threads 0 out.264 > > and getting error as - > [libx264 @ 0xa896320] ref > 1 + intra-refresh is not supported > [libx264 @ 0xa896320] intra-refresh is not compatible with open-gop not reproduceable with git master [...] -- Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB The real ebay dictionary, page 3 "Rare item" - "Common item with rare defect or maybe just a lie" "Professional" - "'Toy' made in china, not functional except as doorstop" "Experts will know" - "The seller hopes you are not an expert" -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From hardik.sharma22 at yahoo.com Fri Sep 30 02:41:50 2011 From: hardik.sharma22 at yahoo.com (Hardik Sharma) Date: Thu, 29 Sep 2011 17:41:50 -0700 (PDT) Subject: [FFmpeg-user] [FFmpeg-devel] how to enable intra-refresh? In-Reply-To: <20110930003006.GR3513@kiste2> References: <1317336999.57838.YahooMailNeo@web46204.mail.sp1.yahoo.com> <20110930003006.GR3513@kiste2> Message-ID: <1317343310.98422.YahooMailNeo@web46214.mail.sp1.yahoo.com> Still getting the error. Any suggestion to try? ________________________________ From: Michael Niedermayer To: FFmpeg development discussions and patches ; ffmpeg Sent: Thursday, 29 September 2011 5:30 PM Subject: Re: [FFmpeg-user] [FFmpeg-devel] how to enable intra-refresh? On Thu, Sep 29, 2011 at 03:56:39PM -0700, Hardik Sharma wrote: > I am trying to enable intra-refresh or IDR frame to replace keyint/I frames for all the time when new gop starts. I am giving following command - > > ffmpeg -y -s 352x288 -r 30 -pix_fmt yuv420p -f rawvideo -i foreman_cif.yuv -vcodec libx264 -y -b 512k -r 30 -preset fast -s 352x288 -flags +loop -cmp +chroma -flags2 -fastpskip -refs 6 -x264opts slice-max-size=300:bframes=1:keyint=20:no-interlaced=1:trellis=2:intra-refresh=1:subme=10 -f h264 -mbd 1 -profile high -deblockalpha 0 -deblockbeta 0 -threads 0 out.264 > > and getting error as - > [libx264 @ 0xa896320] ref > 1 + intra-refresh is not supported > [libx264 @ 0xa896320] intra-refresh is not compatible with open-gop not reproduceable with git master [...] -- Michael? ? GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB The real ebay dictionary, page 3 "Rare item" - "Common item with rare defect or maybe just a lie" "Professional" - "'Toy' made in china, not functional except as doorstop" "Experts will know" - "The seller hopes you are not an expert" _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From hardik.sharma22 at yahoo.com Fri Sep 30 02:43:11 2011 From: hardik.sharma22 at yahoo.com (HardikSharma) Date: Thu, 29 Sep 2011 17:43:11 -0700 (PDT) Subject: [FFmpeg-user] [FFmpeg-devel] how to enable intra-refresh? In-Reply-To: <20110930003006.GR3513@kiste2> References: <1317336999.57838.YahooMailNeo@web46204.mail.sp1.yahoo.com> <20110930003006.GR3513@kiste2> Message-ID: <1317343391891-3858269.post@n4.nabble.com> Still getting error. Any suggestion to try? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/how-to-enable-intra-refresh-tp3858040p3858269.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From dashing.meng at gmail.com Fri Sep 30 03:06:44 2011 From: dashing.meng at gmail.com (littlebat) Date: Fri, 30 Sep 2011 09:06:44 +0800 Subject: [FFmpeg-user] Convert m4a to mp3 In-Reply-To: <1317288869.54060.YahooMailNeo@web95515.mail.in.yahoo.com> References: <1317288869.54060.YahooMailNeo@web95515.mail.in.yahoo.com> Message-ID: <20110930090644.11fe2b6c.dashing.meng@gmail.com> On Thu, 29 Sep 2011 15:04:29 +0530 (IST) Bhavin Mistry wrote: > $cmd="ffmpeg -i from.m4a to.mp3"; Try set more detail parameters, such as "-ac 2 -ar 44100 -ab 384k" From gavr.mail at gmail.com Fri Sep 30 09:58:44 2011 From: gavr.mail at gmail.com (Kirill Gavrilov) Date: Fri, 30 Sep 2011 11:58:44 +0400 Subject: [FFmpeg-user] How to identify a video is 3D In-Reply-To: <4E84B84B.6080107@memeo-inc.com> References: <4E84B84B.6080107@memeo-inc.com> Message-ID: Hi, On Thu, Sep 29, 2011 at 10:26 PM, Jai Kumar wrote: > if file is 3D, then convert to 2D > So you should know what kind of 3D files and formats you support? Currently I know only about stereo tags in MKV container . This parameter is output as metadata in FFmpeg, however I don't remember is it in LibAV fork or not. Also metadata tags syntax may be changed since I last used it... AVStream* theStream; AVFormatContext* theFormatCtx; AVMetadataTag* aTag = av_metadata_get(theStream->metadata, "STEREO_MODE", NULL, 0); if(aTag == NULL) { aTag = av_metadata_get(theFormatCtx->metadata, "STEREO_MODE", NULL, 0); } Flags could be mono, right_left, left_right, bottom_top, top_bottom, row_interleaved_rl, row_interleaved_lr, col_interleaved_rl, col_interleaved_lr, block_lr, block_rl, anaglyph_cyan_red, anaglyph_green_magenta. ----------------------------------------------- Kirill Gavrilov, Software designer. From satonio at gmail.com Fri Sep 30 15:39:24 2011 From: satonio at gmail.com (SaToNiO) Date: Fri, 30 Sep 2011 15:39:24 +0200 Subject: [FFmpeg-user] Audio lost Message-ID: Hi, I'm trying to convert a file to other format (mpg to concatenate after) and i've noticed the audio track is missing in the converted file. look: satonio at satonio-laptop:~$ ffmpeg -i salida2.mp4 -s 352x288 -y salida.3gp ffmpeg version N-33094-gecf31a6, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 29 2011 19:39:35 with gcc 4.4.3 configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab libavutil 51. 18. 0 / 51. 18. 0 libavcodec 53. 18. 0 / 53. 18. 0 libavformat 53. 13. 0 / 53. 13. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 43. 6 / 2. 43. 6 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1950780] multiple edit list entries, a/v desync might occur, patch welcome Last message repeated 2 times [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1950780] stream 1, missing mandatory atoms, broken header [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1950780] multiple edit list entries, a/v desync might occur, patch welcome [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1950780] stream 3, missing mandatory atoms, broken header [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1950780] max_analyze_duration 5000000 reached at 5020000 Seems stream 0 codec frame rate differs from container frame rate: 600.00 (600/1) -> 150.00 (150/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'salida2.mp4': Metadata: major_brand : 3gp5 minor_version : 256 compatible_brands: 3gp53gp4isom creation_time : 2011-09-29 17:43:06 Duration: 00:00:59.36, start: 0.000000, bitrate: 1000 kb/s Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 864x488, 920 kb/s, 20.01 fps, 150 tbr, 300 tbn, 600 tbc Metadata: creation_time : 1945-09-21 14:05:41 Stream #0:1(eng): Data: none (stts / 0x73747473) Metadata: creation_time : 1945-09-21 14:05:41 Stream #0:2(eng): Audio: pcm_mulaw (ulaw / 0x77616C75), 8000 Hz, 1 channels, s16, 64 kb/s Metadata: creation_time : 1945-09-21 14:05:41 Stream #0:3(eng): Data: none (stts / 0x73747473) Metadata: creation_time : 1945-09-21 14:05:41 [buffer @ 0x197af00] w:864 h:488 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param: [scale @ 0x194fea0] w:864 h:488 fmt:yuv420p -> w:352 h:288 fmt:yuv420p flags:0x4 [libopencore_amrnb @ 0x194edc0] bitrate not supported: use one of 4.75k, 5.15k, 5.90k, 6.70k, 7.40k, 7.95k, 10.20k, 12.20k, using 12.20k Output #0, 3gp, to 'salida.3gp': Metadata: major_brand : 3gp5 minor_version : 256 compatible_brands: 3gp53gp4isom creation_time : 2011-09-29 17:43:06 encoder : Lavf53.13.0 Stream #0:0(eng): Video: h263 (s263 / 0x33363273), yuv420p, 352x288, q=2-31, 200 kb/s, 150 tbn, 150 tbc Metadata: creation_time : 1945-09-21 14:05:41 Stream #0:1(eng): Audio: amr_nb (samr / 0x726D6173), 8000 Hz, 1 channels, s16, 128 kb/s Metadata: creation_time : 1945-09-21 14:05:41 Stream mapping: Stream #0.0 -> #0.0 (h264 -> h263) Stream #0.2 -> #0.1 (pcm_mulaw -> libopencore_amrnb) Press [q] to stop, [?] for help [h263 @ 0x194b040] warning, clipping 1 dct coefficients to -127..127 Last message repeated 7 times frame= 8904 fps=770 q=31.0 Lsize= 4016kB time=00:00:00.00 bitrate= 0.0kbits/s dup=7716 drop=0 video:3943kB audio:0kB global headers:0kB muxing overhead 1.853415% And after: satonio at satonio-laptop:~$ ffmpeg -i salida.3gp ffmpeg version N-33094-gecf31a6, Copyright (c) 2000-2011 the FFmpeg developers built on Sep 29 2011 19:39:35 with gcc 4.4.3 configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab libavutil 51. 18. 0 / 51. 18. 0 libavcodec 53. 18. 0 / 53. 18. 0 libavformat 53. 13. 0 / 53. 13. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 43. 6 / 2. 43. 6 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 0 codec frame rate differs from container frame rate: 29.97 (30000/1001) -> 150.00 (150/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'salida.3gp': Metadata: major_brand : 3gp4 minor_version : 512 compatible_brands: isomiso23gp4 creation_time : 2011-09-28 23:00:00 Duration: 00:00:59.36, start: 0.000000, bitrate: 554 kb/s Stream #0:0(eng): Video: h263 (s263 / 0x33363273), yuv420p, 352x288 [SAR 12:11 DAR 4:3], 544 kb/s, 150 fps, 150 tbr, 150 tbn, 29.97 tbc Metadata: creation_time : 2011-09-28 23:00:00 At least one output file must be specified It doesn't work neither converting it to mpg, avi, etc etc and all ends without audio. Also, i've noticed that those videos doesn't have sound played with mplayer (but with vlc work well). Thanks From christian at ch-sc.de Fri Sep 30 10:02:46 2011 From: christian at ch-sc.de (Christian Schneider) Date: Fri, 30 Sep 2011 10:02:46 +0200 Subject: [FFmpeg-user] need help for transcoding Message-ID: <4E8577A6.5070304@ch-sc.de> Hi, I'm trying to convert a video file, which contains one video two audio and one subtitle streams. See here: Duration: 00:55:46.57, start: 0.000000, bitrate: 832 kb/s Stream #0.0(eng): Video: h264 (High), yuv420p, 1280x720, PAR 1:1 DAR 16:9, 25 fps, 25 tbr, 1k tbn, 47.95 tbc (default) Stream #0.1(ger): Audio: ac3, 48000 Hz, stereo, s16, 192 kb/s (default) Stream #0.2(eng): Audio: ac3, 48000 Hz, 5.1, s16, 640 kb/s Stream #0.3(eng): Subtitle: [0][0][0][0] / 0x0000 but it seems to have a problem with the subtitles, ffmpeg seems it can't read them: Decoder (codec id 94210) not found for input stream #0.3 What kind of codec is this, and is there a way to make ffmpeg be able to read it? (scodec copy works, but i'd like to have it in ass, if possible) Another small point i just noticed: the stream bitrates are all in units of "kb/s", but for total it is k-bytes/s and for audio it's k-bit/s, is this true? Should the unit in some way denote that they are different? Thanks in advance Christian