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• Using ffmpeg to combine small mp4 chunks ?

  20 mai 2015, par shiny formica

  I’m trying to convert batches of png images into a single mp4 x264 video using ffmpeg. The conversion, for reasons I won’t go into, converts groups of frames into short mp4 chunks and then I want to take those chunks and merge them into the final video at a specific fps (in this case 30fps).

  My understanding of ffmpeg and the x264 options is too limited, and while I can produce the individual mp4 chunks from the source png frames without trouble, the final merge always ends up duplicating and/or dropping frames especially with very short chunks (< 4 frames).

  The conversion from png to mp4 uses this command :

  ffmpeg -start_number 1001 -framerate 30 -f image2 -i 'intermediate.%d.png' -c:v libx264 -crf 1 -pix_fmt yuv420p -movflags +faststart -frames:v 4 -r 30 chunk.1.mp4 -y

  which appears to work as expected, I get a playable mp4 chunk of, in this case, 4 frames of the sequence of png images at 30fps. The length of each chunk can be anywhere from 1 frame to around 100 frames.

  When all the chunks are generated, I’ve been trying to use the concat demuxer to combine without re-encoding, placing all the source chunk paths in a file :

  concat.txt :

  file 'chunk.1.mp4'
  
  file 'chunk.2.mp4'
  
  file 'chunk.3.mp4'
  
  ...

  and then running this ffmpeg command :

  ffmpeg -f concat -i concat.txt -c:v copy merged.mp4 -y

  but it says this during the concatenation :

  [concat @ 0x315ff80] Estimating duration from bitrate, this may be inaccurate

  and the resulting mp4 has dropped/duplicated frames. So I tried adding duration info to the concat.txt file :

  file 'chunk.1.mp4'
  
  duration 0.133333
  
  file 'chunk.2.mp4'
  
  duration 0.133333
  
  file 'chunk.3.mp4'
  
  duration 0.066666

  in this case, two 4-frame/30fps chunks and one 2-frame/30fps chunk. Which gets rid of that estimation warning, but the result is still duplicating/dropping frames.

  I’m not sure where I’m going wrong here...what do I need to do either in the production of the short mp4 segments, or in the combination stage, to get a single mp4 at the right framerate with no duplicated or dropped frames ?

  As suggested, here’s the console output for the conversion from png->mp4 chunks :

  ffmpeg -loglevel verbose -start_number 1001 -framerate 30 -f image2 -i 'intermediate.%d.png' -c:v libx264 -crf 1 -pix_fmt yuv420p -movflags +faststart -frames:v 4 -r 30 chunk.1.mp4 -y
  
  ffmpeg version 2.5.4 Copyright (c) 2000-2015 the FFmpeg developers
  
    built on Feb 26 2015 10:23:42 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-3)
  
    configuration: --prefix=/dept/srd/vendor/ffmpeg/bundle.rhel6/ffmpeg2.5.4 --enable-static --enable-pthreads --enable-gpl --enable-version3 --disable-ffserver --disable-ffplay --disable-ffprobe --enable-x11grab --enable-nonfree --extra-cflags=-I/dept/srd/vendor/ffmpeg/extern/rhel6/include --extra-ldflags=-L/dept/srd/vendor/ffmpeg/extern/rhel6/lib --enable-libx264 --enable-fontconfig --enable-libfreetype --enable-swscale --enable-libmp3lame --enable-libfaac --disable-yasm
  
    libavutil      54. 15.100 / 54. 15.100
  
    libavcodec     56. 13.100 / 56. 13.100
  
    libavformat    56. 15.102 / 56. 15.102
  
    libavdevice    56.  3.100 / 56.  3.100
  
    libavfilter     5.  2.103 /  5.  2.103
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, image2, from 'intermediate.%d.png':
  
    Duration: 00:00:00.27, start: 0.000000, bitrate: N/A
  
      Stream #0:0: Video: png, rgba, 1024x1024 (0x0), 30 fps, 30 tbr, 30 tbn, 30 tbc
  
  [graph 0 input from stream 0:0 @ 0x273e9c0] w:1024 h:1024 pixfmt:rgba tb:1/30 fr:30/1 sar:0/1 sws_param:flags=2
  
  [auto-inserted scaler 0 @ 0x2737ea0] w:iw h:ih flags:'0x4' interl:0
  
  [format @ 0x273ece0] auto-inserting filter 'auto-inserted scaler 0' between the filter 'Parsed_null_0' and the filter 'format'
  
  [auto-inserted scaler 0 @ 0x2737ea0] w:1024 h:1024 fmt:rgba sar:0/1 -> w:1024 h:1024 fmt:yuv420p sar:0/1 flags:0x4
  
  [libx264 @ 0x273c540] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
  
  [libx264 @ 0x273c540] profile High, level 3.2
  
  [libx264 @ 0x273c540] 264 - core 142 - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - 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=36 lookahead_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=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
  
  Output #0, mp4, to 'chunk.1.mp4':
  
    Metadata:
  
      encoder         : Lavf56.15.102
  
      Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 1024x1024, q=-1--1, 30 fps, 15360 tbn, 30 tbc
  
      Metadata:
  
        encoder         : Lavc56.13.100 libx264
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (png (native) -> h264 (libx264))
  
  Press [q] to stop, [?] for help
  
  No more output streams to write to, finishing.
  
  [mp4 @ 0x273baa0] Starting second pass: moving the moov atom to the beginning of the file
  
  frame=    4 fps=0.0 q=-1.0 Lsize=     197kB time=00:00:00.06 bitrate=24228.7kbits/s    
  
  video:196kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.439751%
  
  Input file #0 (intermediate.%d.png):
  
    Input stream #0:0 (video): 8 packets read (2341016 bytes); 5 frames decoded; 
  
    Total: 8 packets (2341016 bytes) demuxed
  
  Output file #0 (chunk.3.mp4):
  
    Output stream #0:0 (video): 4 frames encoded; 4 packets muxed (201023 bytes); 
  
    Total: 4 packets (201023 bytes) muxed
  
  [libx264 @ 0x273c540] frame I:1     Avg QP: 0.47  size:116049
  
  [libx264 @ 0x273c540] frame P:1     Avg QP: 2.29  size: 37932
  
  [libx264 @ 0x273c540] frame B:2     Avg QP: 2.37  size: 23184
  
  [libx264 @ 0x273c540] consecutive B-frames: 25.0%  0.0% 75.0%  0.0%
  
  [libx264 @ 0x273c540] mb I  I16..4: 80.0%  4.5% 15.5%
  
  [libx264 @ 0x273c540] mb P  I16..4:  0.2%  0.1%  0.4%  P16..4:  8.1%  3.6%  3.7%  0.0%  0.0%    skip:83.9%
  
  [libx264 @ 0x273c540] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8:  4.8%  1.2%  1.6%  direct: 4.3%  skip:88.1%  L0:38.6% L1:39.3% BI:22.1%
  
  [libx264 @ 0x273c540] 8x8 transform intra:4.6% inter:14.8%
  
  [libx264 @ 0x273c540] coded y,uvDC,uvAC intra: 20.7% 22.9% 22.8% inter: 8.7% 10.1% 10.0%
  
  [libx264 @ 0x273c540] i16 v,h,dc,p: 95%  1%  3%  1%
  
  [libx264 @ 0x273c540] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 21% 22%  6%  6%  6%  7%  5%  6%
  
  [libx264 @ 0x273c540] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 20% 17% 18%  7%  8%  7%  8%  6%  8%
  
  [libx264 @ 0x273c540] i8c dc,h,v,p: 89%  4%  4%  3%
  
  [libx264 @ 0x273c540] Weighted P-Frames: Y:0.0% UV:0.0%
  
  [libx264 @ 0x273c540] ref B L1: 89.5% 10.5%
  
  [libx264 @ 0x273c540] kb/s:12020.88

  as I said, this appears to produce a valid mp4 at 30fps with no duplicated or dropped frames from the input images.

  Here’s the output of the combine phase :

  ffmpeg -loglevel verbose -f concat -i concat.txt -c:v copy merged.mp4 -y
  
  ffmpeg version 2.5.4 Copyright (c) 2000-2015 the FFmpeg developers
  
    built on Feb 26 2015 10:23:42 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-3)
  
    configuration: --prefix=/dept/srd/vendor/ffmpeg/bundle.rhel6/ffmpeg2.5.4 --enable-static --enable-pthreads --enable-gpl --enable-version3 --disable-ffserver --disable-ffplay --disable-ffprobe --enable-x11grab --enable-nonfree --extra-cflags=-I/dept/srd/vendor/ffmpeg/extern/rhel6/include --extra-ldflags=-L/dept/srd/vendor/ffmpeg/extern/rhel6/lib --enable-libx264 --enable-fontconfig --enable-libfreetype --enable-swscale --enable-libmp3lame --enable-libfaac --disable-yasm
  
    libavutil      54. 15.100 / 54. 15.100
  
    libavcodec     56. 13.100 / 56. 13.100
  
    libavformat    56. 15.102 / 56. 15.102
  
    libavdevice    56.  3.100 / 56.  3.100
  
    libavfilter     5.  2.103 /  5.  2.103
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, concat, from 'concat.txt':
  
    Duration: 00:00:00.67, start: 0.000000, bitrate: 2 kb/s
  
      Stream #0:0: Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1024x1024, 7791 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc
  
  Output #0, mp4, to 'merged.mp4':
  
    Metadata:
  
      encoder         : Lavf56.15.102
  
      Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 1024x1024 (0x0), q=2-31, 7791 kb/s, 30 fps, 15360 tbn, 15360 tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  No more output streams to write to, finishing.
  
  frame=   20 fps=0.0 q=-1.0 Lsize=     748kB time=00:00:00.56 bitrate=10805.0kbits/s    
  
  video:746kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.141687%
  
  Input file #0 (concat.txt):
  
    Input stream #0:0 (video): 20 packets read (764361 bytes); 
  
    Total: 20 packets (764361 bytes) demuxed
  
  Output file #0 (merged.mp4):
  
    Output stream #0:0 (video): 20 packets muxed (764361 bytes); 
  
    Total: 20 packets (764361 bytes) muxed

• Using ffmpeg to convert sound files for use in an android app

  10 janvier 2012, par stefs

  short : i'm trying to simply play a sound file converted with ffmpeg in my android app, but happen to have problems getting it to work.

  long : we have an iphone app and an android app doing the same thing, and i have to port the feature playing a sound on an user interaction. i have the source file in the aiff format, and tried to convert it to mp3 for android. but the app keeps crashing when it tries to load the file

  AssetFileDescriptor fileDescriptor = context.getResources().openRawResourceFd(resid);
  
  final MediaPlayer mp = new MediaPlayer();
  
  mp.setDataSource(fileDescriptor.getFileDescriptor(), fileDescriptor.getStartOffset(), fileDescriptor.getLength());
  
  fileDescriptor.close();
  
  mp.prepare();

  more specifically, mp.setDataSource crashes. some digging around led me to believe that something's wrong with the encoding. the sound file itself resides in res/raw.

  11-29 17:11:48.012: ERROR/SoundManager(15580): java.io.IOException: setDataSourceFD failed.: status=0x80000000
  
  11-29 17:11:48.012: ERROR/SoundManager(15580):     at android.media.MediaPlayer.setDataSource(Native Method)
  
  ...

  what i tried :

  • using a different mp3 that's already used with the same code in a different place. this works.
  • converted it to wav file. this didn't cause the app to crash, but it neither played a sound. that might be a different problem.
  • converted it to ogg ; crashed

  so, the the ffmpeg conversion parameters are as follows :

  $ ffmpeg -i click_24db.aif -f mp3 ~/foobar/wheel_click.mp3
  
  ffmpeg version 0.7.8, Copyright (c) 2000-2011 the FFmpeg developers
  
    built on Nov 24 2011 14:31:00 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-libopenjpeg --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/gcc-4.2 --arch=x86_64 --enable-yasm
  
    libavutil    50. 43. 0 / 50. 43. 0
  
    libavcodec   52.123. 0 / 52.123. 0
  
    libavformat  52.111. 0 / 52.111. 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
  
  Input #0, aiff, from 'click_24db.aif':
  
    Duration: 00:00:00.01, start: 0.000000, bitrate: 1570 kb/s
  
      Stream #0.0: Audio: pcm_s16be, 44100 Hz, 2 channels, s16, 1411 kb/s
  
  Output #0, mp3, to '/Users/xyz/foobar/wheel_click.mp3':
  
    Metadata:
  
      TSSE            : Lavf52.111.0
  
      Stream #0.0: Audio: libmp3lame, 44100 Hz, 2 channels, s16, 64 kb/s
  
  Stream mapping:
  
    Stream #0.0 -> #0.0
  
  Press [q] to stop, [?] for help
  
  size=       1kB time=00:00:00.05 bitrate=  92.9kbits/s    
  
  video:0kB audio:0kB global headers:0kB muxing overhead 45.563549%

  the resulting file plays nice in itunes, does not play in vlc and crashes when loaded with the android.media.MediaPlayer (note : i first tried it with the SoundPool lib, with both mp3 and ogg, but that didn't work either).

  i also tried the following paramters, which didn't work :

  ffmpeg -i inputfile.aif -f mp3 -acodec libmp3lame -ab 192000 -ar 44100 outputfile.mp3

  i'm working on osx, built ffmpeg with macports today, android api level is 7 (google api, 2.1-update1). looking at the "supported formats" table on dev.android didn't indicate my file to be out of the spec, but i may be mistaken in that.

  i don't have the slightest clue regarding bitrates and so on, so could anybody please point me to the right combination of ffmpeg parameters to get a working mp3 for android ? i don't care if the resulting file would be mp3, ogg or 3gp or whatever.

• ffmpeg merge wav-audio and rtmp-video stream with itsoffset parameter ?

  2 juillet 2015, par whatis01

  I tried to merge WAV audio and RTMP video stream with -itsoffset parameter :

  ffmpeg -i rtmp://server/video/stream_folder/stream_id -itsoffset 00:00:15.0000 -i /audio/1227121.wav \
  
  -codec:v copy -ac 2 -codec:a aac -strict experimental -f flv rtmp://server/sfolder/test_1

  But it works inappropriately (RTMP output is not steady, only voice or only video), and always prompts : mux overhead ratio 2000%, then conversion failed.