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• Decode 16 kHz PCM aLaw with ffmpeg

  4 mai 2017, par Jim Rhodes

  I have an application that receives an RTP stream from an IP camera and decodes it with ffmpeg and I am having a problem with a particular audio format.

  If I configure the camera for G.711A at 8 kHz audio decodes correctly.
  If I configure the camera for G.711A at 16 kHz audio plays at the wrong pitch. It plays at the correct speed but voices are very low pitch.
  The camera sends a=rtpmap:8 PCMA/16000 for the audio track and I set up ffmeg like this :

  AVPacket avpkt;
  
  AVCodec* pCodec;
  
  AVCodecContext* pContext;
  
  AVFrame* pFrame;
  
  
  
  . . .
  
  
  
  av_init_packet(&avpkt);
  
  pCodec = avcodec_find_decoder(AV_CODEC_ID_PCM_ALAW);
  
  if ( pCodec )
  
  {
  
      pContext = avcodec_alloc_context3(pCodec);
  
      pContext->bits_per_coded_sample = 16;
  
      pContext->sample_rate = 16000;
  
      pContext->channels = 1;
  
      if ( avcodec_open2(pContext, pCodec, NULL) >= 0)
  
      {
  
          pFrame = av_frame_alloc();
  
      }
  
  }

  As audio data is received I use avcodec_decode_audio4 to decode the stream and pass the output to an audio device that accepts PCM16.

  Should ffmpeg be able to decode this format and if so is there anything else I need to set up ?

• Merge commit ’746c56b7730ce09397d3a8354acc131285e9d829’

  20 mars 2017, par Clément Bœsch

  Merge commit ’746c56b7730ce09397d3a8354acc131285e9d829’
  * commit ’746c56b7730ce09397d3a8354acc131285e9d829’ :
  
    indeo : Change type of array pitch parameters to ptrdiff_t
  Merged-by : Clément Bœsch <u@pkh.me>
  

  • [DH] libavcodec/indeo3.c
  • [DH] libavcodec/ivi.c
  • [DH] libavcodec/ivi.h
  • [DH] libavcodec/ivi_dsp.c
  • [DH] libavcodec/ivi_dsp.h

• opus : add a native Opus encoder

  11 février 2017, par Rostislav Pehlivanov

  opus : add a native Opus encoder
  This marks the first time anyone has written an Opus encoder without
  
  using any libopus code. The aim of the encoder is to prove how far
  
  the format can go by writing the craziest encoder for it.
  Right now the encoder’s basic, it only supports CBR encoding, however
  
  internally every single feature the CELT layer has is implemented
  
  (except the pitch pre-filter which needs to work well with the rest of
  
  whatever gets implemented). Psychoacoustic and rate control systems are
  
  under development.
  The encoder takes in frames of 120 samples and depending on the value of
  
  opus_delay the plan is to use the extra buffered frames as lookahead.
  
  Right now the encoder will pick the nearest largest legal frame size and
  
  won’t use the lookahead, but that’ll change once there’s a
  
  psychoacoustic system.
  Even though its a pretty basic encoder its already outperforming
  
  any other native encoder FFmpeg has by a huge amount.
  The PVQ search algorithm is faster and more accurate than libopus’s
  
  algorithm so the encoder’s performance is close to that of libopus
  
  at zero complexity (libopus has more SIMD).
  
  The algorithm might be ported to libopus or other codecs using PVQ in
  
  the future.
  The encoder still has a few minor bugs, like desyncs at ultra low
  
  bitrates (below 9kbps with 20ms frames).
  Signed-off-by : Rostislav Pehlivanov <atomnuker@gmail.com>
  

  • [DH] configure
  • [DH] libavcodec/Makefile
  • [DH] libavcodec/allcodecs.c
  • [DH] libavcodec/opus_celt.h
  • [DH] libavcodec/opus_pvq.c
  • [DH] libavcodec/opus_pvq.h
  • [DH] libavcodec/opusenc.c