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• Anomalie #3332 : Message d’erreur erroné : le nombre d’erreur n’est pas le bon

  4 novembre 2014, par Valéry -

  Pour info il s’est avéré qu’il y avait bien deux erreurs : le login / pass ne correspondaient pas à ce qui étaient en base.

• 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 transcoded sound (AAC) stops after half video time

  17 août 2015, par TheSHEEEP

  I have a strange problem in my C/C++ FFmpeg transcoder, which takes an input MP4 (varying input codecs) and produces and output MP4 (x264, baseline & AAC LC @44100 sample rate with libfdk_aac) :

  The resulting mp4 video has fine images (x264) and the audio (AAC LC) works fine as well, but is only played until exactly the half of the video.

  The audio is not slowed down, not stretched and doesn’t stutter. It just stops right in the middle of the video.

  One hint may be that the input file has a sample rate of 22050 and 22050/44100 is 0.5, but I really don’t get why this would make the sound just stop after half the time. I’d expect such an error leading to sound being at the wrong speed. Everything works just fine if I don’t try to enforce 44100 and instead just use the incoming sample_rate.

  Another guess would be that the pts calculation doesn’t work. But the audio sounds just fine (until it stops) and I do exactly the same for the video part, where it works flawlessly. "Exactly", as in the same code, but "audio"-variables replaced with "video"-variables.

  FFmpeg reports no errors during the whole process. I also flush the decoders/encoders/interleaved_writing after all the package reading from the input is done. It works well for the video so I doubt there is much wrong with my general approach.

  Here are the functions of my code (stripped off the error handling & other class stuff) :

  AudioCodecContext Setup

  outContext->_audioCodec = avcodec_find_encoder(outContext->_audioTargetCodecID);
  
  outContext->_audioStream = 
  
          avformat_new_stream(outContext->_formatContext, outContext->_audioCodec);
  
  outContext->_audioCodecContext = outContext->_audioStream->codec;
  
  outContext->_audioCodecContext->channels = 2;
  
  outContext->_audioCodecContext->channel_layout = av_get_default_channel_layout(2);
  
  outContext->_audioCodecContext->sample_rate = 44100;
  
  outContext->_audioCodecContext->sample_fmt = outContext->_audioCodec->sample_fmts[0];
  
  outContext->_audioCodecContext->bit_rate = 128000;
  
  outContext->_audioCodecContext->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
  
  outContext->_audioCodecContext->time_base = 
  
          (AVRational){1, outContext->_audioCodecContext->sample_rate};
  
  outContext->_audioStream->time_base = (AVRational){1, outContext->_audioCodecContext->sample_rate};
  
  int retVal = avcodec_open2(outContext->_audioCodecContext, outContext->_audioCodec, NULL);

  Resampler Setup

  outContext->_audioResamplerContext = 
  
          swr_alloc_set_opts( NULL, outContext->_audioCodecContext->channel_layout,
  
                              outContext->_audioCodecContext->sample_fmt,
  
                              outContext->_audioCodecContext->sample_rate,
  
                              _inputContext._audioCodecContext->channel_layout,
  
                              _inputContext._audioCodecContext->sample_fmt,
  
                              _inputContext._audioCodecContext->sample_rate,
  
                              0, NULL);
  
  int retVal = swr_init(outContext->_audioResamplerContext);

  Decoding

  decodedBytes = avcodec_decode_audio4(   _inputContext._audioCodecContext, 
  
                                          _inputContext._audioTempFrame, 
  
                                          &p_gotAudioFrame, &_inputContext._currentPacket);

  Converting (only if decoding produced a frame, of course)

  int retVal = swr_convert(   outContext->_audioResamplerContext, 
  
                              outContext->_audioConvertedFrame->data, 
  
                              outContext->_audioConvertedFrame->nb_samples, 
  
                              (const uint8_t**)_inputContext._audioTempFrame->data, 
  
                              _inputContext._audioTempFrame->nb_samples);

  Encoding (only if decoding produced a frame, of course)

  outContext->_audioConvertedFrame->pts = 
  
          av_frame_get_best_effort_timestamp(_inputContext._audioTempFrame);
  
  
  
  // Init the new packet
  
  av_init_packet(&outContext->_audioPacket);
  
  outContext->_audioPacket.data = NULL;
  
  outContext->_audioPacket.size = 0;
  
  
  
  // Encode
  
  int retVal = avcodec_encode_audio2( outContext->_audioCodecContext, 
  
                                      &outContext->_audioPacket, 
  
                                      outContext->_audioConvertedFrame,
  
                                      &p_gotPacket);
  
  
  
  
  
  // Set pts/dts time stamps for writing interleaved
  
  av_packet_rescale_ts(   &outContext->_audioPacket, 
  
                          outContext->_audioCodecContext->time_base,
  
                          outContext->_audioStream->time_base);
  
  outContext->_audioPacket.stream_index = outContext->_audioStream->index;

  Writing (only if encoding produced a packet, of course)

  int retVal = av_interleaved_write_frame(outContext->_formatContext, &outContext->_audioPacket);

  I am quite out of ideas about what would cause such a behaviour.