November 2012 - ffmpeg-user - lists.ffmpeg.org

Changing aspect ratio flag without re-encoding
by mi.lists.ffmpeg＠alma.ch 16 May '13

16 May '13

I tried to use FFmpeg to change the aspect ratio flag in an MPEG file, but it didn't. Am I doing it wrong, or is it not possible without re-encoding? Here, the Output part correctly shows "DAR 16:9", but the resulting file is still "DAR 4:3" : $ ffmpeg -i INPUT.m2v -aspect 16:9 -vcodec copy OUTPUT.m2v FFmpeg version SVN-r18021, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --enable-gpl --enable-postproc --enable-pthreads --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libx264 --enable-libxvid libavutil 50. 0. 0 / 50. 0. 0 libavcodec 52.21. 0 / 52.21. 0 libavformat 52.31. 1 / 52.31. 1 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Mar 18 2009 14:40:53, gcc: 4.1.2 20061115 (prerelease) (Debian 4.1.1-21) Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 25.00 (25/1) Input #0, mpegvideo, from 'INPUT.m2v': Duration: 00:00:11.33, bitrate: 8000 kb/s Stream #0.0: Video: mpeg2video, yuv420p, 720x576 [PAR 16:15 DAR 4:3], 8000 kb/s, 25 tbr, 1200k tbn, 50 tbc Output #0, mpeg2video, to 'OUTPUT.m2v': Stream #0.0: Video: mpeg2video, yuv420p, 720x576 [PAR 64:45 DAR 16:9], q=2-31, 8000 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 282 fps= 0 q=-1.0 Lsize= 11072kB time=11.28 bitrate=8041.3kbits/s video:11072kB audio:0kB global headers:0kB muxing overhead 0.000000% Looking at the result: $ ffmpeg -i OUTPUT.m2v FFmpeg version SVN-r18021, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --enable-gpl --enable-postproc --enable-pthreads --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libx264 --enable-libxvid libavutil 50. 0. 0 / 50. 0. 0 libavcodec 52.21. 0 / 52.21. 0 libavformat 52.31. 1 / 52.31. 1 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Mar 18 2009 14:40:53, gcc: 4.1.2 20061115 (prerelease) (Debian 4.1.1-21) Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 25.00 (25/1) Input #0, mpegvideo, from 'OUTPUT.m2v': Duration: 00:00:11.33, bitrate: 8000 kb/s Stream #0.0: Video: mpeg2video, yuv420p, 720x576 [PAR 16:15 DAR 4:3], 8000 kb/s, 25 tbr, 1200k tbn, 50 tbc

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RGB-YUV color shift
by George L.P. Fitz 29 Apr '13

29 Apr '13

Hello, I am seeing a slight color shift when using ffmpeg to encode to yuv based codecs like prores and v210 from an rgb source. The resulting file has slightly more saturation and gamma than the original- it doesn't look way different, but there is enough of a difference to be noticed. I'm assuming this has to do with the rgb->yuv colorspace conversion, since encoding from a yuv source to a yuv codec shows no perceivable color difference. Has anyone run across this and if so, found a way to keep the color shift negligible? Here's a sample command: ffmpeg -f rawvideo -pix_fmt rgb24 -s 1920x1080 -r 23.976 -i c3yoz4vuuy.video.raw -vcodec prores -profile 3 test.mov Thanks for any help you can provide -George

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Problem transcoding from MP4 (H264 - AAC) to OGG (DIRAC - FLAC)
by Alexis Prus 12 Apr '13

12 Apr '13

Hello, I am trying to do a lossy-to-lossless transcoding from an MP4 file (with one H264 video stream and one AAC audio stream) to an OGG file (with a DIRAC video stream and a FLAC audio stream). The command line I am using is this: ffmpeg -i input.mp4 -vcodec libschroedinger -q:v 0 -acodec flac output.ogg and I am getting the following errors (complete output is at the end): [ogg @ 0000000001F29570] Unsupported codec id in stream 0 Could not write header for output file #0 (incorrect codec parameters ?) Does anybody know an appropriate way to do this kind of transcoding? Thanks in advance, Alexis ffmpeg version N-38148-gb6ff81d Copyright (c) 2000-2012 the FFmpeg developers built on Feb 23 2012 12:30:44 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 40.100 / 51. 40.100 libavcodec 54. 4.100 / 54. 4.100 libavformat 54. 1.100 / 54. 1.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 62.101 / 2. 62.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 7.100 / 0. 7.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'h264aac1.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 2010-02-16 19:06:52 encoder : Lavf52.36.0 Duration: 00:03:40.03, start: 0.000000, bitrate: 2571 kb/s Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 2537 kb/s, 30 fps, 30 tbr, 30 tbn, 60 tbc Metadata: creation_time : 2010-02-16 19:06:52 handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 11025 Hz, stereo, s16, 31 kb/s Metadata: creation_time : 2010-02-16 19:06:52 handler_name : [buffer @ 0000000001F29F30] w:640 h:480 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param: [ogg @ 0000000001F29570] Unsupported codec id in stream 0 Output #0, ogg, to 'h264aac.ogg': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 2010-02-16 19:06:52 encoder : Lavf54.1.100 Stream #0:0(und): Video: dirac, yuv420p, 640x480 [SAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 30 tbn, 30 tbc Metadata: creation_time : 2010-02-16 19:06:52 handler_name : VideoHandler Stream #0:1(und): Audio: flac, 11025 Hz, stereo, s16, 128 kb/s Metadata: creation_time : 2010-02-16 19:06:52 handler_name : Stream mapping: Stream #0:0 -> #0:0 (h264 -> libschroedinger) Stream #0:1 -> #0:1 (aac -> flac) Could not write header for output file #0 (incorrect codec parameters ?)

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codec copy fails with malformed bitstreams in mpegts
by Rares Pop 26 Mar '13

26 Mar '13

Hi, Every now and then ffmpeg fails when remuxing a mpegts file with the following error: [mpegts @ 0x1677a60] H.264 bitstream malformed, no startcode found, use the h264_mp4toannexb bitstream filter (-bsf h264_mp4toannexb) av_interleaved_write_frame(): Invalid argument The file I am remuxing can be downloaded here: https://www.dropbox.com/sh/b8v81l4l7lhjgj0/6Klxfejcki I am thinking this is a bug and even if there is an error in the stream the remuxing shouldn't break. I am able to play this file just fine. The comand and output: /usr/local/bin/ffmpeg -i /cutchup/record/channel7/_/2012-09-17.00.00.7-0.rec/00066.ts -vcodec copy -an -y /cutchup/record/channel7/_/2012-09-17.00.00.7-0.rec/00066.ts-video.ts ffmpeg version git-2012-08-27-60924df Copyright (c) 2000-2012 the FFmpeg developers built on Sep 1 2012 18:38:17 with gcc 4.5.3 (GCC) 20120403 (ALT Linux 4.5.3-alt1.M60P.1) configuration: --enable-gpl --enable-libfdk-aac --enable-libvo-aacenc --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libx264 --enable-nonfree --enable-version3 --enable-yasm libavutil 51. 70.100 / 51. 70.100 libavcodec 54. 54.100 / 54. 54.100 libavformat 54. 25.104 / 54. 25.104 libavdevice 54. 2.100 / 54. 2.100 libavfilter 3. 13.101 / 3. 13.101 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 [h264 @ 0x1674140] mmco: unref short failure Last message repeated 3 times [mp3 @ 0x1674a40] Header missing [h264 @ 0x1674140] mmco: unref short failure Last message repeated 1 times [mpegts @ 0x1670240] max_analyze_duration 5000000 reached at 5016000 [mpegts @ 0x1670240] PES packet size mismatch Input #0, mpegts, from '/cutchup/record/channel7/_/2012-09-17.00.00.7-0.rec/00066.ts': Duration: 00:12:23.66, start: 77203.089300, bitrate: 2256 kb/s Program 132 Stream #0:0[0x2be]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p, 720x576 [SAR 12:11 DAR 15:11], 50 fps, 50 tbr, 90k tbn, 50 tbc Stream #0:1[0x2bf](rus): Audio: mp2 ([4][0][0][0] / 0x0004), 48000 Hz, stereo, s16, 128 kb/s [mpegts @ 0x1677a60] muxrate VBR, pcr every 2 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to '/cutchup/record/channel7/_/2012-09-17.00.00.7-0.rec/00066.ts-video.ts': Metadata: encoder : Lavf54.25.104 Stream #0:0: Video: h264 ([27][0][0][0] / 0x001B), yuv420p, 720x576 [SAR 12:11 DAR 15:11], q=2-31, 50 fps, 90k tbn, 25 tbc Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help [mpegts @ 0x1677a60] H.264 bitstream malformed, no startcode found, use the h264_mp4toannexb bitstream filter (-bsf h264_mp4toannexb) av_interleaved_write_frame(): Invalid argument -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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Using -re with -ss to stream a ts file as live in rtmp
by Paulo Flores 18 Mar '13

18 Mar '13

Hi People. My Scenery: I'm using ffmpeg to streaming a local .ts file (in h264) to Wowza Server using RTMP protocol as realtime, using -re parameter. In some cases because the internet connection for example the ffmpeg lost connection with Wowza and stops to stream, in this case I need to continue the stream at same file time it is stopped. Now the problem: To start ffmpeg in same position I'm using the parameter -ss to set start time but I'm using too -re parameter to simulate the live broadcast and in this case the seek to start time delay the same time of "start time" to do. How I can do the seek in file instantly and to the live in real time? Thanks, Paulo

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What does "-vol" parameter mean in ffmpeg?
by littlebat 01 Feb '13

01 Feb '13

Hi, I can't understand how to use "-vol" parameter to increase volume in ffmpeg. I know "-vol 256" mean keep volume unchanged. But I don't know how to choose a proper number for a video file to increase volume. For example, does "-vol 512" mean volume intensity is twice as original file? ( volume intensity: http://en.wikipedia.org/wiki/Decibel#Power_quantities ) thanks. littlebat

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Encoding to mpeg2video results in buffer underflows
by Jonathan Isom 13 Jan '13

13 Jan '13

Attached is 2 logs. 1 is from a ATSC mpeg2 stream, the other a h264 mkv video. Fresh build from today from a live ebuild on gentoo that uses no patches. Problem affects mplayer and vlc as well(system ffmpeg). command line used: ffmpeg -i test.mkv -t 60 -s 720x480 -target ntsc-dvd -b:v 8000000 -b:a 224000 -c:v mpeg2video -c:a ac3 test.mpeg ffmpeg -i 3031_20120927210000.mpg -t 60 -s 720x480 -target ntsc-dvd -b:v 8000000 -b:a 224000 -c:v mpeg2video -c:a ac3 test.mpeg Thanks Jonathan

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X11grab playback is too fast
by . 06 Jan '13

06 Jan '13

I'm using X11grab (here's the code- http://pastebin.com/f7Tjw8b6 The problem is playback is fast (speed is like and old time movie) (as a side note....I can't get the audio to be captured either...but I'll just work on the speed first) How do I fix it?

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NAV PACK System Clock Refrence.
by kifayat ullah 04 Jan '13

04 Jan '13

Hello Carl, Accroding to DVD Standards Each Title set first VOB should always contain System clock refrence value = 0 in its first Navigation Pack. When I try to create a vob from ffmpeg command line using following command it doesn't seems valid. ffmpeg -i VTS_01_1.VOB -target ntsc-dvd out.vob I try to look the value of Nav Pack 0 @ 000000 in VobEdit tool and it seems other then 0. Any idea how to fix this. Many thanks kifayat

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Post problem: export long wav files (incorrect/limited length)
by Stas Fomin 01 Jan '13

01 Jan '13

I have problem with export long wav files (>8 hours) from a video. Exported files have length only about 2.5 hour (but "good" size). <pre> ffmpeg.exe -analyzeduration 999999999 -i aud.scenario.avs$lossless.avi -f wav d2.wav ffmpeg version N-46469-gc995644 Copyright (c) 2000-2012 the FFmpeg developers built on Nov 5 2012 17:54:28 with gcc 4.7.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-runt ime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass - -enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enab le-libgsm --enable-libmp3lame --enable-libnut --enable-libopenjpeg --enable-libo pus --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheo ra --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-li bvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --ena ble-zlib libavutil 52. 5.100 / 52. 5.100 libavcodec 54. 71.100 / 54. 71.100 libavformat 54. 36.100 / 54. 36.100 libavdevice 54. 3.100 / 54. 3.100 libavfilter 3. 21.106 / 3. 21.106 libswscale 2. 1.102 / 2. 1.102 libswresample 0. 16.100 / 0. 16.100 libpostproc 52. 1.100 / 52. 1.100 [avi @ 022ab5a0] non-interleaved AVI Guessed Channel Layout for Input Stream #0.1 : stereo Input #0, avi, from 'aud.scenario.avs$lossless.avi': Metadata: encoder : Lavf53.2.0 Duration: 08:26:37.00, start: 0.000000, bitrate: 2395 kb/s Stream #0:0: Video: h264 (High 4:4:4 Predictive) (H264 / 0x34363248), yuv420 p, 128x64, 25 fps, 25 tbr, 25 tbn, 50 tbc Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16 , 1536 kb/s File 'd2.wav' already exists. Overwrite ? [y/N] y Output #0, wav, to 'd2.wav': Metadata: ISFT : Lavf54.36.100 Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16 , 1536 kb/s Stream mapping: Stream #0:1 -> #0:0 (pcm_s16le -> pcm_s16le) Press [q] to stop, [?] for help size= 5699438kB time=08:26:37.00 bitrate=1536.0kbits/s video:0kB audio:5699438kB subtitle:0 global headers:0kB muxing overhead 0.000001 % </pre> <pre> D:\bug-export-wav>ffmpeg -analyzeduration 999999999 -i d2.wav ffmpeg version N-46469-gc995644 Copyright (c) 2000-2012 the FFmpeg developers built on Nov 5 2012 17:54:28 with gcc 4.7.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-runt ime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass - -enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enab le-libgsm --enable-libmp3lame --enable-libnut --enable-libopenjpeg --enable-libo pus --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheo ra --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-li bvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --ena ble-zlib libavutil 52. 5.100 / 52. 5.100 libavcodec 54. 71.100 / 54. 71.100 libavformat 54. 36.100 / 54. 36.100 libavdevice 54. 3.100 / 54. 3.100 libavfilter 3. 21.106 / 3. 21.106 libswscale 2. 1.102 / 2. 1.102 libswresample 0. 16.100 / 0. 16.100 libpostproc 52. 1.100 / 52. 1.100 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, wav, from 'd2.wav': Metadata: encoder : Lavf54.36.100 Duration: 02:13:47.38, bitrate: 5816 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16 , 1536 kb/s At least one output file must be specified </pre> That is "Duration: 02:13:47.38, bitrate: 5816 kb/s" instead of "size= 5699438kB time=08:26:37.00 bitrate=1536.0kbits/s" VirtualDub export correct wav ("aud.scenario.avs.avi.wav", approximately same size as "bad ffmpeg wav", but with correct length) without any problem. <pre> D:\bug-export-wav>ffmpeg -i aud.scenario.avs.avi.wav ffmpeg version N-46469-gc995644 Copyright (c) 2000-2012 the FFmpeg developers built on Nov 5 2012 17:54:28 with gcc 4.7.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-runt ime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass - -enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enab le-libgsm --enable-libmp3lame --enable-libnut --enable-libopenjpeg --enable-libo pus --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheo ra --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-li bvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --ena ble-zlib libavutil 52. 5.100 / 52. 5.100 libavcodec 54. 71.100 / 54. 71.100 libavformat 54. 36.100 / 54. 36.100 libavdevice 54. 3.100 / 54. 3.100 libavfilter 3. 21.106 / 3. 21.106 libswscale 2. 1.102 / 2. 1.102 libswresample 0. 16.100 / 0. 16.100 libpostproc 52. 1.100 / 52. 1.100 [w64 @ 0036b400] max_analyze_duration 5000000 reached at 5013333 [w64 @ 0036b400] Estimating duration from bitrate, this may be inaccurate Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, w64, from 'aud.scenario.avs.avi.wav': Duration: 08:26:37.00, bitrate: 1535 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16 , 1536 kb/s At least one output file must be specified </pre> ---- I have this problem with every tried version of ffmpeg since 2010 (arrozcru/zeranoe builds), tried a lot of options — no success. Is this bug or know feature? Are the any workaroud? Sincerely, Stas Fomin P.S. (Sample files are too great, 17GB, but I can share if this will be useful).

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