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• checkasm : af_afir : Use a dynamic tolerance depending on values

  11 décembre 2019, par Martin Storsjö

  checkasm : af_afir : Use a dynamic tolerance depending on values
  As the values generated by av_bmg_get can be arbitrarily large
  
  (only the stddev is specified), we can't use a fixed tolerance.
  
  Calculate a dynamic tolerance (like in float_dsp from 38f966b2222db),
  
  based on the individual steps of the calculation.
  This fixes running this test with certain seeds, when built with
  
  clang for mingw/x86_32.
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DH] tests/checkasm/af_afir.c

• How to fill audio AVFrame (ffmpeg) with the data obtained from CMSampleBufferRef (AVFoundation) ?

  24 mai 2013, par Aleksei2414904

  I am writing program for streaming live audio and video from webcamera to rtmp-server. I work in MacOS X 10.8, so I use AVFoundation framework for obtaining audio and video frames from input devices. This frames come into delegate :

  -(void) captureOutput:(AVCaptureOutput*)captureOutput didOutputSampleBuffer:    (CMSampleBufferRef)sampleBuffer fromConnection:(AVCaptureConnection*)connection ,

  where sampleBuffer contains audio or video data.

  When I recieve audio data in the sampleBuffer, I'm trying to convert this data into AVFrame and encode AVFramewith libavcodec :

      aframe = avcodec_alloc_frame();  //AVFrame *aframe;
  
      int got_packet, ret;
  
      CMItemCount numSamples = CMSampleBufferGetNumSamples(sampleBuffer); //CMSampleBufferRef
  
  
  
      NSUInteger channelIndex = 0;
  
  
  
      CMBlockBufferRef audioBlockBuffer = CMSampleBufferGetDataBuffer(sampleBuffer);
  
  
  
      size_t audioBlockBufferOffset = (channelIndex * numSamples * sizeof(SInt16));
  
  
  
      size_t lengthAtOffset = 0;
  
  
  
      size_t totalLength = 0;
  
  
  
      SInt16 *samples = NULL;
  
  
  
      CMBlockBufferGetDataPointer(audioBlockBuffer, audioBlockBufferOffset, &lengthAtOffset, &totalLength, (char **)(&samples));
  
  
  
              const AudioStreamBasicDescription *audioDescription = CMAudioFormatDescriptionGetStreamBasicDescription(CMSampleBufferGetFormatDescription(sampleBuffer));
  
  
  
      aframe->nb_samples =(int) numSamples;
  
  
  
      aframe->channels=audioDescription->mChannelsPerFrame;
  
  
  
      aframe->sample_rate=(int)audioDescription->mSampleRate;
  
  
  
       //my webCamera configured to produce 16bit 16kHz LPCM mono, so sample format hardcoded here, and seems to be correct
  
      avcodec_fill_audio_frame(aframe, aframe->channels, AV_SAMPLE_FMT_S16,
  
  
  
                               (uint8_t *)samples,
  
  
  
                                aframe->nb_samples *
  
  
  
                               av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) *
  
  
  
                               aframe->channels, 0);  
  
      //encoding audio
  
      ret = avcodec_encode_audio2(c, &pkt, aframe, &got_packet);
  
      if (ret < 0) {
  
          fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
  
          exit(1);
  
      }

  The problem is that when I get so formed frames, I can hear the wanted sound, but it is slowing down and discontinuous (as if after each data frame comes the same frame of silence). It seems that something is wrong in the transformation from CMSampleBuffer to AVFrame , because the preview from the microphone created with AVFoundation from the same sample buffers played normally.

  I would be grateful for your help.

  UPD : Creating and initializing the AVCodceContext structure
   

      audio_codec= avcodec_find_encoder(AV_CODEC_ID_AAC);
  
      if (!(audio_codec)) {
  
          fprintf(stderr, "Could not find encoder for '%s'\n",
  
                  avcodec_get_name(AV_CODEC_ID_AAC));
  
          exit(1);
  
      }
  
      audio_st = avformat_new_stream(oc, audio_codec);  //AVFormatContext *oc;
  
      if (!audio_st) {
  
          fprintf(stderr, "Could not allocate stream\n");
  
          exit(1);
  
      }
  
  
  
        audio_st->id=1;
  
        audio_st->codec->sample_fmt= AV_SAMPLE_FMT_S16;
  
        audio_st->codec->bit_rate = 64000;
  
        audio_st->codec->sample_rate= 16000;
  
        audio_st->codec->channels=1;
  
        audio_st->codec->codec_type= AVMEDIA_TYPE_AUDIO;

• avformat/movenc : use enum values directly for colr atom

  31 mars 2020, par Michael Bradshaw

  avformat/movenc : use enum values directly for colr atom
  The switch cases were missing :
    - Primaries : bt470m, film, smpte428, and ebu3213.
  
    - TRCs : gamma22, gamma28, linear, log, log_sqrt, iec61966_2_4, bt1361,
  
      iec61966_2_1, bt2020_10bit, and bt2020_12bit.
  
    - Space : rgb, fcc, ycgco, bt2020_cl, smpte2085, chroma-derived-nc,
  
      chroma-derived-c, and ictcp.
  They also annoyingly remapped the following (which are functionally
  
  equivalent but can be treated differently by clients) :
    - smpte240m primaries to smpte170m.
  
    - smpte170m TRC to bt709.
  
    - bt470bg color space to smpte170m.
  The enum values in FFmpeg are the same values as ITU-T H.273 and
  
  ISO/IEC 23001-8 so we can just use them directly, which is both simpler
  
  and preserves the user intent.
  Signed-off-by : Michael Bradshaw <mjbshaw@google.com>
  

  • [DH] libavformat/movenc.c