June 2011 - ffmpeg-user - lists.ffmpeg.org

"error, non monotone timestamps"
by kshawkeye＠gmail.com 28 May '13

28 May '13

Hi, when trying to encode from a .mov with MPEG-4 video and PCM audio into .mp4 with MPEG-4 video and AAC audio I get: [avi @ 0x956cda0]st:0 error, non monotone timestamps 1028 >= 102889.9kbits/s av_interleaved_write_frame(): Operation not permitted I tried to demux and then mux them: Demux: ffmpeg -i 'Infile.mov' -vn -acodec libfaac -ab 320k -ar 48000 -ac 2 'Outfile.aac' and ffmpeg -i 'Infile.mov' -vcodec copy -an 'Outfilet.m2v' Mux: ffmpeg -i 'Outfile.m2v' -i 'Outfile.aac' -vcodec copy -acodec copy 'Final.mp4' While I finally got the audio to convert this way, the resulting Final.mp4 was out of sync. I tried to do a normal encode: ffmpeg -i 'Infile.mov' -vcodec copy -acodec libfaac -ab 320k -ar 48000 -ac 2 'Outfile.mp4' Which lead to my "error, non monotone timestamps" problem.

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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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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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convert a sequence of images to a video file
by joshjdevl＠gmail.com 04 Oct '12

04 Oct '12

Hi, I would like to convert a series of images (png files) to a video. Would something like this work? ffmpeg -r 29.97 -i /images/picture%.png out.avi When I try the above, I get picture%d.jpg: I/O error occurred Usually that means that input file is truncated and/or corrupted. Thanks, Josh

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compil ffmpeg with alsa support
by khenyh＠gmail.com 26 Aug '12

26 Aug '12

Hi, I need your help to configure ffmpeg so as to use alsa. I have downloaded version 0.5 of ffmpeg. But I can't find how to enable alsa. I have tried "configure --enable-muxer=alsa --enable-demuxer=alsa". Then I looked at config.h and I can see "#define ENABLE_ALSA_MUXER 0", "#define ENABLE_ALSA_MUXER 0". Is there something I have missed ? Nicolas Krieger

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Exclude ALL Metadata?
by Bob Zimmerman 08 Aug '12

08 Aug '12

I'm attempting to strip ALL TAGS (metadata) off of an MP3. While I have found other tools to do this, I naturally thought of FFMPEG first. I was sure FFMPEG could easily extract ONLY the AUDIO stream from an MP3 file and ignore everything else (ex. tags). The closest FFMPEG command I could come up with to attempt this was "ffmpeg -i original.mp3 -acodec copy -map_metadata 0:-1 notags.mp3". But with this command, FFMPEG still writes an ID3v2.4 header with a "TSSE" tag set to "Lavf53.2.0" to the "notags.mp3" file. Are there any other FFMPEG options I could specify to have it NOT create the ID3 header?

5 4

Vorbis codec
by acandido＠hi-iberia.es 12 Jul '12

12 Jul '12

Hello, I have two versions of FFmpeg, 0.5.2 and 0.6. I've noticed that FFmeg 0.5.2 is able to encode vorbis audio ("ffmpeg -formats" shows "DEA vorbis") but FFmpeg 0.6 is not ("D A vorbis"). Could anybody please explain me why does this happen? Maybe it is due to a wrong compilation? Regards, Andr?s

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how to extract h264 nal bitstream into a file from h264 quicktime file?
by Kaijun Tang 29 Jun '12

29 Jun '12

I have tried -vbsf h264_mp4toannexb or -f rawvideo. These options don't work. Thanks,

5 8

FFMpeg native aac decoder
by vubrhar＠gmail.com 24 Jun '12

24 Jun '12

Hi I am having some difficulty decoding audio from an IP camera, the RTSP negotiation for audio is such: v=0 o=- 1266870982069588 1266870982069588 IN IP4 10.11.11.189 s=Media Presentation e=NONE c=IN IP4 0.0.0.0 b=AS:50016 t=0 0 a=control:* a=range:npt=0.000000- m=video 0 RTP/AVP 96 b=AS:50000 a=framerate:10.0 a=control:trackID=1 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1; profile-level-id=420029; sprop-parameter-sets=Z0IAKeKQFAe2AtwEBAaQeJEV,aM48gA== m=audio 0 RTP/AVP 97 b=AS:16 a=control:trackID=2 a=rtpmap:97 mpeg4-generic/16000/1 a=fmtp:97 profile-level-id=15; mode=AAC-hbr;config=1408; SizeLength=13; IndexLength=3;IndexDeltaLength=3; Profile=1; bitrate=16000; in a previous discussion, it has been pointed out that config=1408 corresponds to Low Complexity Sampling Frequency, 16000 Hz, 1 channel So this should be supported by the native aac decoder. I also was unable to transcode the following aac file to vorbis ogg http://www.nch.com.au/acm/sample.aac I am using SVN-r19905

5 9

named pipes - > x264
by joshjdevl＠gmail.com 29 May '12

29 May '12

Hi, I was following these instructions to do a named pipe to x264 http://ffmpeg.x264.googlepages.com/piping mkfifo fifo.y4m ffmpeg -i test.mp4 -an -f yuv4mpegpipe - > fifo.y4m & x264 --quiet --crf 18 -o video.264 fifo.y4m For some reason, ffmpeg just sits here Stream mapping: Stream #0.1 -> #0.0 Press [q] to stop encoding It doesn't do any processing of test.mp4 and basically sits as a blank console. If I do something like ffmpeg -i test.mp4 -an -f yuv4mpegpipe - > test.y4m & Where test.y4m is not a named pipe, then test.y4m will end up being the yuv4 data... So I guess my question is, is there some extra option I need to compile ffmpeg or x264 with? Or is there another option I need to pass in? I tested this on Ubuntu 8.04 and centos 5.2 Thanks, Josh

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