May 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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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,

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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

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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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LXF decoder ! Need help !
by Tuuls 03 May '12

03 May '12

Hello. Someone can help with unpacking LXF container? The fact that the LXF files from the new servers can not be preprocessed in ffmpeg. The case is probably not in the files themselves, as they are exported from the old systems are handled beautifully, but the problem is probably in the file header. And can we somehow add the module metadata extraction? http://refile.net/f/?6j Here is a small file that I exported from Harris server. It contains the metadata and the video format DVСPRO25 . when I try to open it in ffmpeg, I get an error [lxf @ 000000000035A920] checksum error [lxf @ 000000000035A920] expected 120 B size header, got 0 Thanks in advance ! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/LXF-decoder-Need-help-tp3507549p35… Sent from the FFmpeg-users mailing list archive at Nabble.com.

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