Recherche avancée

Sur d’autres sites (9407)

• ffmpeg audio conversion distorted - half rate

  6 novembre 2013, par user1688971

  I'm trying to convert an asf audio to mp3 using ffmpeg.
  But I have one specific audio that gets distorted in the middle and starts like if the person was talking in slow motion (at half rate).

  The command I'm using is :

  ffmpeg - i input.asf -ac 2 output.mp3

  I've tried a lot of options, but about the middle of the audio is when it fails.
  The raw file sounds good, so it's not the recording. It is af in the middle of the transmission the frame rate went down for some reason.

  Thanks all !

  [EDIT]

  I'm adding the console response after running the command as a suggestion from LordNeckbeard :

  [root@mynasserver home]# ffmpeg -i recording-8532-1.asf -ac 2 -ab 64k -ar 44100 recording-8532-ac2-ar44100.mp3
  
  FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
  
  built on Jan 29 2012 23:56:18 with gcc 4.1.2 20080704 (Red Hat 4.1.2-51)
  
  configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib --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 -m32 -march=i386 -mtune=generic -fasynchronous-unwind-tables' --enable-avfilter --enable-avfilter-lavf --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --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
  
  [flv @ 0x86a4850]max_analyze_duration reached
  
  [flv @ 0x86a4850]Estimating duration from bitrate, this may be inaccurate
  
  Input #0, flv, from 'recording-8532-1.asf':
  
  Metadata:
  
  source          : STW MediaProxy v3.3.7.19894
  
  Duration: 04:00:08.49, start: 0.000000, bitrate: N/A
  
  Stream #0.0: Audio: aac, 44100 Hz, 2 channels (FC), s16
  
  Output #0, mp3, to 'recording-8532-ac2-ar44100.mp3':
  
  Metadata:
  
  TSSE            : Lavf52.64.2
  
  Stream #0.0: Audio: libmp3lame, 44100 Hz, 2 channels, s16, 64 kb/s
  
  Stream mapping:
  
  Stream #0.0 -> #0.0
  
  Press [q] to stop encoding
  
  size=  150906kB time=19315.93 bitrate=  64.0kbits/s    
  
  video:0kB audio:150906kB global headers:0kB muxing overhead 0.000021%

  So from the data above, you can see the input file is about 4hrs. The output ends up being around 5 hrs 20 mins.

• 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.