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• IframeExtractor don't output sound with rtsp

  9 janvier 2013, par Kamax

  I use IframeExtractor from the git mooncatventure, it play nice the .mov file.
  But when i try to read a rtsp stream, i hear no sound.

  This is the FFMEG dump from the rtsp stream :

  Metadata:
  
  title           : unknown
  
  comment         : unknown
  
  Duration: N/A, start: 49435.000589, bitrate: 258 kb/s
  
  Program 3223 
  
  No Program
  
  Stream #0:0: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc
  
  Stream #0:1(fra): Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 142 kb/s
  
  Stream #0:2(fra): Subtitle: dvb_teletext ([6][0][0][0] / 0x0006)
  
  Stream #0:3(qad): Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, mono, fltp, 47 kb/s
  
  Stream #0:4(qaa): Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 68 kb/s

  And this is the dump from the local .mov file that work :

  Metadata:
  
  major_brand     : qt  
  
  minor_version   : 0
  
  compatible_brands: qt  
  
  creation_time   : 2010-01-17 21:52:33
  
  model           : iPhone 3GS
  
  model-eng       : iPhone 3GS
  
  date            : 2010-01-17T16:52:33-0500
  
  date-eng        : 2010-01-17T16:52:33-0500
  
  encoder         : 3.1.2
  
  encoder-eng     : 3.1.2
  
  make            : Apple
  
  make-eng        : Apple
  
  Duration: 00:00:03.25, start: 0.000000, bitrate: 3836 kb/s
  
  Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p, 640x480, 3695 kb/s, 30.02 fps, 30 tbr, 600 tbn, 1200 tbc
  
  Metadata:
  
    rotate          : 90
  
    creation_time   : 2010-01-17 21:52:33
  
    handler_name    : Core Media Data Handler
  
  Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 63 kb/s
  
  Metadata:
  
    creation_time   : 2010-01-17 21:52:33
  
    handler_name    : Core Media Data Handler

  The audio class that manage sounds contain a codec detector which say that the codec CODEC_ID_AAC is found for the two input :

  audioStreamBasicDesc_.mFormatFlags = 0;
  
  switch (_audioCodecContext->codec_id) {
  
      case CODEC_ID_MP3:
  
           audioStreamBasicDesc_.mFormatID = kAudioFormatMPEGLayer3;
  
          break;
  
      case CODEC_ID_AAC:
  
           audioStreamBasicDesc_.mFormatID = kAudioFormatMPEG4AAC;
  
           audioStreamBasicDesc_.mFormatFlags = kMPEG4Object_AAC_Main;
  
          NSLog(@"audio format aac %s (%d) is  supported",  _audioCodecContext->codec_name, _audioCodecContext->codec_id);
  
          break;
  
  }

  I see data going into the buffer but i hear nothing. It's maybe audioStreamBasicDesc_ which has wrong settings but i can't find what.

  Is it possible that it's not the same AAC codec ?

  Has someone experienced the same issue ?

  Any help are welcome, i'm on this problem since some days now.

  Edit :
  I have found a error that i had not before, i don't know how to resolve it. If i change audioStreamBasicDesc.mFramesPerPacket to 0 or divided by 2, the error message dissapear.

  AudioConverterNew returned 'fmt?'
  
  Prime failed ('fmt?'); will stop (72000/0 frames)

• Converting avi to mp4 (use ffmpeg library) is very slow [closed]

  7 décembre 2012, par yAnTar

  I have task - convert any video from user to mp4 format.
  I use command

  ffmpeg -i $file -vcodec libx264 -vpre ultrafast -bt 500k -acodec libfaac -ab 56k -ac 2 -s 480x320 $outputFileMp4

  I try to change options bitrate, dimensions, but spped of converting is slow - 3 mbytes convert arround 90 seconds.

  If i convert avi to flv - no problem.

  FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
  
    built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6)
  
    configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-avfilter-lavf --enable-libdc1394 --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
  
    libavutil     50.15. 1 / 50.15. 1
  
    libavcodec    52.72. 2 / 52.72. 2
  
    libavformat   52.64. 2 / 52.64. 2
  
    libavdevice   52. 2. 0 / 52. 2. 0
  
    libavfilter    1.19. 0 /  1.19. 0
  
    libswscale     0.11. 0 /  0.11. 0
  
    libpostproc   51. 2. 0 / 51. 2. 0
  
  Input #0, avi, from 'intro.avi':
  
    Duration: 00:07:48.53, start: 0.000000, bitrate: 244 kb/s
  
      Stream #0.0: Video: camtasia, bgr24, 800x600, 15 tbr, 15 tbn, 15 tbc
  
      Metadata:
  
        strn            : Camtasia Producer_render549ab1f.avi Video #1
  
      Stream #0.1: Audio: pcm_s16le, 12000 Hz, 1 channels, s16, 192 kb/s
  
      Metadata:
  
        strn            : Microsoft Waveform: Camtasia Producer_render549ab1f.wav
  
  [libx264 @ 0x1202e40]using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.1 Cache64
  
  [libx264 @ 0x1202e40]profile Constrained Baseline, level 2.2
  
  [libx264 @ 0x1202e40]264 - core 107 - H.264/MPEG-4 AVC codec - Copyleft 2003-2010 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0x1:0 me=dia subme=0 psy=1 psy_rd=0.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=0 intra_refresh=0 rc=abr mbtree=0 bitrate=200 ratetol=2.5 qcomp=0.60 qpmin=10 qpmax=51 qpstep=4 ip_ratio=1.41 aq=0
  
  Output #0, mp4, to 'test2.mp4':
  
    Metadata:
  
      encoder         : Lavf52.64.2
  
      Stream #0.0: Video: libx264, yuv420p, 640x480, q=10-51, 200 kb/s, 15 tbn, 15 tbc
  
      Metadata:
  
        strn            : Camtasia Producer_render549ab1f.avi Video #1
  
      Stream #0.1: Audio: libfaac, 12000 Hz, 2 channels, s16, 5 kb/s
  
      Metadata:
  
        strn            : Microsoft Waveform: Camtasia Producer_render549ab1f.wav
  
  Stream mapping:
  
    Stream #0.0 -> #0.0
  
    Stream #0.1 -> #0.1

  and result of executing

  frame= 7028 fps= 52 q=10.0 Lsize=    2654kB time=468.31 bitrate=  46.4kbits/s
  
  video:1883kB audio:672kB global headers:0kB muxing overhead 3.876805%
  
  [libx264 @ 0xe1ee40]frame I:29    Avg QP:10.83  size: 25781
  
  [libx264 @ 0xe1ee40]frame P:6999  Avg QP:10.08  size:   169
  
  [libx264 @ 0xe1ee40]mb I  I16..4: 80.1%  0.0% 19.9%
  
  [libx264 @ 0xe1ee40]mb P  I16..4:  0.2%  0.0%  0.0%  P16..4:  0.2%  0.0%  0.0%  0.0%  0.0%    skip:99.6%
  
  [libx264 @ 0xe1ee40]final ratefactor: 5.71
  
  [libx264 @ 0xe1ee40]coded y,uvDC,uvAC intra: 37.1% 3.3% 3.1% inter: 0.1% 0.0% 0.0%
  
  [libx264 @ 0xe1ee40]i16 v,h,dc,p: 87% 12%  1%  0%
  
  [libx264 @ 0xe1ee40]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 20% 23% 27%  8%  6%  3%  3%  2%  8%
  
  [libx264 @ 0xe1ee40]i8c dc,h,v,p: 94%  6%  0%  0%
  
  [libx264 @ 0xe1ee40]kb/s:32.91

• Call ffmpeg in c++ with system() function fails

  2 septembre 2014, par zhen lee

  I write a c++ program which needs to convert some(say:10) mp4 videos to flv videos.
  I use ffmpeg in my program for each video like this :

  system("ffmpeg -i video -filter:v yadif -ar 44100 -sameq -y -f flv temp.flv")

  however,it turns out :only first video will be converted successfully,the others will fail.
  it means :
  when i change the input order of which video to convert and re-execution the program,it behave the same:only the first video(will be different each time as i changed the input video order) will be converted successfully.

  The error message like :

  [h264 @ 0xaee0740] concealing 45 DC, 45 AC, 45 MV errors
  [h264 @ 0xaee0ce0] AVC : nal size 305665
  Last message repeated 1 times
  [h264 @ 0xaee0ce0] no frame !
  [h264 @ 0xaee1280] AVC : nal size 572993
  Last message repeated 1 times
  [h264 @ 0xaee1280] no frame !
  [aac @ 0xad9ccc0] channel element 0.13 is not allocated
  Error while decoding stream #0:1
  [aac @ 0xad9ccc0] channel element 0.13 is not allocated
  Error while decoding stream #0:1
  [aac @ 0xad9ccc0] channel element 0.13 is not allocated
  Error while decoding stream #0:1
  ......
  The most strange thing is :when i run ffmpeg command in bash shell,all video will be converted successfully .

  After google it,I have try these(certainly failed) :

  1. remove -sameq option,the result is same ;

  2. write ffmpeg commod in a shell script ConvertToFlv.sh like :

     /usr/local/bin/ffmpeg -i "$dir/$1" -filter:v yadif -ar 44100 -sameq -y -f flv "$dir/temp.fiv"

     then call this script in program like

     system("ConvertToFlv.sh"+video)

     or

     system("sh ConvertToFlv.sh"+video)

     The result is same.

  The ffmpeg configure is :

    ffmpeg version 0.9.1.git Copyright (c) 2000-2012 the FFmpeg developers
  
    built on Dec 17 2012 16:17:30 with gcc 4.1.2 20080704 (Red Hat 4.1.2-48)
  
    configuration: --enable-gpl --enable-postproc --enable-nonfree --enable-postproc --enable-swscale --enable-avfilter --enable-pthreads --enable-libxvid --enable-libx264 --enable-libmp3lame --enable-libfaac --disable-ffserver --disable-ffplay
  
    libavutil      51. 41.100 / 51. 41.100
  
    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

  and my machine envirment is :

  Linux master 2.6.18-194.el5 #1 SMP Fri Apr 2 14:58:14 EDT 2010 x86_64 x86_64 x86_64 GNU/Linux

  I’m irritable now,I hope someone can give me some advice,really appreciate it.