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• ffmpeg error creating thumbnail different frame rate

  18 décembre 2012, par KJS

  When using this at the command line I get very bad images with only grey or stripes in them.
  It seems "the frame rate differs from container frame rate : 59.94 (60000/1001) -> 29.97 (30000/1001)".

  Is there any way I can fix this in the ffmpeg statement ?

  ffmpeg -ss 00:00:10 -i FILENAME.mp4 -vframes 1 FILENAME.jpg

  This is the output I get :

  FFmpeg version 0.5.2, Copyright (c) 2000-2009 Fabrice Bellard, et al.
  
    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-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
  
    libavutil     49.15. 0 / 49.15. 0
  
    libavcodec    52.20. 1 / 52.20. 1
  
    libavformat   52.31. 0 / 52.31. 0
  
    libavdevice   52. 1. 0 / 52. 1. 0
  
    libavfilter    0. 4. 0 /  0. 4. 0
  
    libswscale     0. 7. 1 /  0. 7. 1
  
    libpostproc   51. 2. 0 / 51. 2. 0
  
    built on Jun 13 2010 23:44:18, gcc: 4.1.2 20080704 (Red Hat 4.1.2-48)

  Seems stream 0 codec frame rate differs from container frame rate : 59.94 (60000/1001) -> 29.97 (30000/1001)

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'FILENAME.mp4':
  
    Duration: 00:03:36.36, start: 0.000000, bitrate: 1305 kb/s
  
      Stream #0.0(eng): Video: h264, yuv420p, 640x428, 29.97 tbr, 29.97 tbn, 59.94 tbc
  
      Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16
  
  Output #0, image2, to 'FILENAME.jpg':
  
      Stream #0.0(eng): Video: mjpeg, yuvj420p, 640x428, q=2-31, 200 kb/s, 90k tbn, 29.97 tbc
  
  Stream mapping:
  
    Stream #0.0 -> #0.0
  
  Press [q] to stop encoding
  
  [h264 @ 0x307f6b0]brainfart cropping not supported, this could look slightly wrong ...
  
  [h264 @ 0x307f6b0]AVC: Consumed only 14978 bytes instead of 14984
  
  [h264 @ 0x307f6b0]AVC: Consumed only 1147 bytes instead of 1153
  
  [h264 @ 0x307f6b0]Missing reference picture
  
  [h264 @ 0x307f6b0]AVC: Consumed only 1947 bytes instead of 1953
  
  [h264 @ 0x307f6b0]Missing reference picture
  
  [h264 @ 0x307f6b0]AVC: Consumed only 1870 bytes instead of 1876
  
  [h264 @ 0x307f6b0]Missing reference picture
  
      Last message repeated 1 times
  
  [h264 @ 0x307f6b0]AVC: Consumed only 810 bytes instead of 816
  
  [h264 @ 0x307f6b0]Missing reference picture
  
      Last message repeated 1 times
  
  [h264 @ 0x307f6b0]AVC: Consumed only 955 bytes instead of 961
  
  [h264 @ 0x307f6b0]Missing reference picture
  
      Last message repeated 1 times
  
  [h264 @ 0x307f6b0]AVC: Consumed only 1036 bytes instead of 1042
  
  [h264 @ 0x307f6b0]Missing reference picture
  
      Last message repeated 1 times
  
  [h264 @ 0x307f6b0]AVC: Consumed only 998 bytes instead of 1004
  
  [h264 @ 0x307f6b0]Missing reference picture
  
  frame=    1 fps=  0 q=3.3 Lsize=      -0kB time=0.03 bitrate=  -5.3kbits/s
  
  video:14kB audio:0kB global headers:0kB muxing overhead -100.149568%

• Transcoding audio using xuggler

  23 juin 2014, par amd

  I am trying to convert an audio file with the header

  Opening audio decoder: [pcm] Uncompressed PCM audio decoder
  
  AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400)
  
  Selected audio codec: [pcm] afm: pcm (Uncompressed PCM)

  I want to transcode this file to mp3 format. I have following code snippet but its not working well. I have written it using XUGGLER code snippet for transcoding audio and video.

  Audio decoder is

      audioDecoder = IStreamCoder.make(IStreamCoder.Direction.DECODING, ICodec.findDecodingCodec(ICodec.ID.CODEC_ID_PCM_S16LE));
  
      audioDecoder.setSampleRate(44100);
  
      audioDecoder.setBitRate(176400);
  
      audioDecoder.setChannels(2);
  
      audioDecoder.setTimeBase(IRational.make(1,1000));
  
      if (audioDecoder.open(IMetaData.make(), IMetaData.make()) < 0)
  
          return false;
  
      return true; 

  Audio encoder is

      outContainer = IContainer.make();
  
      outContainerFormat = IContainerFormat.make();
  
      outContainerFormat.setOutputFormat("mp3", urlOut, null);
  
      int retVal = outContainer.open(urlOut, IContainer.Type.WRITE, outContainerFormat);
  
      if (retVal < 0) {
  
          System.out.println("Could not open output container");
  
          return false;
  
      }
  
      outAudioCoder = IStreamCoder.make(IStreamCoder.Direction.ENCODING, ICodec.findEncodingCodec(ICodec.ID.CODEC_ID_MP3));
  
      outAudioStream = outContainer.addNewStream(outAudioCoder);
  
      outAudioCoder.setSampleRate(new Integer(44100));
  
      outAudioCoder.setChannels(2);
  
      retVal = outAudioCoder.open(IMetaData.make(), IMetaData.make());
  
      if (retVal < 0) {
  
          System.out.println("Could not open audio coder");
  
          return false;
  
      }
  
      retVal = outContainer.writeHeader();
  
      if (retVal < 0) {
  
          System.out.println("Could not write output FLV header: ");
  
          return false;
  
      }
  
      return true;

  And here is encode method where i send packets of 32 byte to transcode

  public void encode(byte[] audioFrame){
  
      //duration of 1 video frame
  
      long lastVideoPts = 0;
  
  
  
      IPacket packet_out = IPacket.make();
  
      int lastPos = 0;
  
      int lastPos_out = 0;
  
  
  
      IAudioSamples audioSamples = IAudioSamples.make(48000, audioDecoder.getChannels());
  
      IAudioSamples audioSamples_resampled = IAudioSamples.make(48000, audioDecoder.getChannels());
  
  
  
      //we always have 32 bytes/sample
  
      int pos = 0;
  
      int audioFrameLength = audioFrame.length;
  
      int audioFrameCnt = 1;
  
      iBuffer = IBuffer.make(null, audioFrame, 0, audioFrameLength);
  
      IPacket packet = IPacket.make(iBuffer);
  
      //packet.setKeyPacket(true);
  
      packet.setTimeBase(IRational.make(1,1000));
  
      packet.setDuration(20);
  
      packet.setDts(audioFrameCnt*20);
  
      packet.setPts(audioFrameCnt*20);
  
      packet.setStreamIndex(1);
  
      packet.setPosition(lastPos);
  
      lastPos+=audioFrameLength;
  
      int pksz = packet.getSize();
  
      packet.setComplete(true, pksz);
  
      /*
  
      * A packet can actually contain multiple samples
  
      */
  
      int offset = 0;
  
      int retVal;
  
      while(offset < packet.getSize())
  
      {
  
          int bytesDecoded = audioDecoder.decodeAudio(audioSamples, packet, offset);
  
          if (bytesDecoded < 0)
  
              throw new RuntimeException("got error decoding audio ");
  
          offset += bytesDecoded;
  
          if (audioSamples.isComplete())
  
          {
  
              int samplesConsumed = 0;
  
              while (samplesConsumed < audioSamples.getNumSamples()) {
  
                  retVal = outAudioCoder.encodeAudio(packet_out, audioSamples, samplesConsumed);
  
                  if (retVal <= 0)
  
                      throw new RuntimeException("Could not encode audio");
  
                  samplesConsumed += retVal;
  
                  if (packet_out.isComplete()) {
  
                      packet_out.setPosition(lastPos_out);
  
                      packet_out.setStreamIndex(1);
  
                      lastPos_out+=packet_out.getSize();
  
                      retVal = outContainer.writePacket(packet_out);
  
                      if(retVal < 0){
  
                          throw new RuntimeException("Could not write data packet");
  
                      }
  
                  }
  
              }
  
          }
  
  
  
      }
  
  
  
  }

  I get an output file but it doesnt get played. I have very little experience of audio encoding and sampling. Thanks in advance.

• 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)