December 2012 - 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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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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build error (latest and 1.0.1)
by Andreas S. 28 Mar '13

28 Mar '13

Hi the latest git fails to compile for me, so I tried getting something older (and stable?) but same breakage, included below. I'm on Ubuntu 10.10 (Maverick). Updated everything I was alerted about (compiled and installed x264-snapshot-20121215-2245-stable, with " --enable-shared --enable-pic "... What am I missing. ---------------------- build error quote: ---------------------- ..... LD ffmpeg_g libavcodec/libavcodec.a(libx264.o): In function `X264_frame': /opt/builds/snapshots/ffmpeg-1.0.1/libavcodec/libx264.c:159: undefined reference to `x264_picture_init' /opt/builds/snapshots/ffmpeg-1.0.1/libavcodec/libx264.c:161: undefined reference to `x264_bit_depth' libavcodec/libavcodec.a(libx264.o): In function `X264_init_static': /opt/builds/snapshots/ffmpeg-1.0.1/libavcodec/libx264.c:593: undefined reference to `x264_bit_depth' libavcodec/libavcodec.a(libx264.o): In function `X264_init': /opt/builds/snapshots/ffmpeg-1.0.1/libavcodec/libx264.c:292: undefined reference to `x264_param_default_preset' /opt/builds/snapshots/ffmpeg-1.0.1/libavcodec/libx264.c:459: undefined reference to `x264_param_apply_fastfirstpass' /opt/builds/snapshots/ffmpeg-1.0.1/libavcodec/libx264.c:490: undefined reference to `x264_param_apply_profile' /opt/builds/snapshots/ffmpeg-1.0.1/libavcodec/libx264.c:533: undefined reference to `x264_encoder_open_128' collect2: ld returned 1 exit status make: *** [ffmpeg_g] Error 1

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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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When to determine frames are progressive or interlaced ?
by tank pranav 15 Jan '13

15 Jan '13

Hello, I would like to know when to determine frames are progressive or interlaced ? Can it be known only through the parameter "progressive sequence" which exist in "sequence extension" ?or some other fields are also need to be validated (such as top field first, progressive frames, repeat first field , etc ..... ) or [ is this correct that in progressive frames , there always be parameter "top field first = 0 and progressive frames = 0 and repeate first field = 0) while in interlaced frames , all above 3 parameters will contains some values ) ??? if progressive sequence is 0 then its progressive frames. if progressive sequence is 1 then its its interlaced frames. -------------------------------------- I have another question also regarding interlaced frames. IF there are two frames A and B. I agree if frames are interlaced then either (odd/even) part of frame A can be part of frame B. But can it be case in which some color portion of frame B can remain present in frame A ? (where progressive sequence is 0 and "top field first = 1). -------------------------------------- It would be great if someone can answer for above questions. Regards, Pranav.

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No accelerated colorspace conversion found from yuv422p to rgb24
by Ronny Bismark 13 Jan '13

13 Jan '13

Hi all, I would like to use the hardware enabled mjpeg compression of my webcam (Logitech c920). I'm using a Pandaboard (ARM) and ubuntu 11.10 Server. I use as camera driver the ROS usb_cam package which works with YUYV but when I change the pixelformat to MJPEG I'll get following warning. *[swscaler @ 0xe8d640] No accelerated colorspace conversion found from yuv422p to rgb24.* The camera works but has a big time difference between the output and the actual image (about 2-3 seconds). According to some googling this warning leads to some lag and is caused by some old dependencies. Therefore I installed ffmpeg-0.10.6, ffmpeg-0.6.2 & ffmpeg-0.6.6 all with the same result. I always rebuild OpenCV afterward. my configuration options: *H.264* (git clone --depth 1 git://git.videolan.org/x264) ./configure --enable-static --enable-pic --enable-shared make sudo checkinstall --pkgname=libx264-dev --pkgversion="3:$(./version.sh | awk -F'[" ]' '/POINT/{print $4"+git"$5}')" --backup=no --deldoc=yes --fstrans=no --default *AAC audio encoder* (git clone --depth 1 git:// github.com/mstorsjo/fdk-aac.git) autoreconf -fiv ./configure --enable-pic --enable-shared make sudo checkinstall --pkgname=fdk-aac --pkgversion="$(date +%Y%m%d%H%M)-git" --backup=no --deldoc=yes --fstrans=no --default *libvpx* (git clone --depth 1 http://git.chromium.org/webm/libvpx.git) ./configure --enable-pic make sudo checkinstall --pkgname=libvpx --pkgversion="1:$(date +%Y%m%d%H%M)-git" --backup=no --deldoc=yes --fstrans=no --default *FFmpeg* (git clone --depth 1 git://source.ffmpeg.org/ffmpeg) ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-libvpx --enable-postproc --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-shared --enable-swscale --enable-pic make sudo checkinstall --pkgname=ffmpeg --pkgversion="7:$(date +%Y%m%d%H%M)-git" --backup=no --deldoc=yes --fstrans=no --default hash x264 ffmpeg ffplay ffprobe A weird thing is that when I use ffmpeg -h I get following message: *ffmpeg: error while loading shared libraries: libavfilter.so.2: cannot open shared object file: No such file or directory*

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