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• Streaming protocol relay without involving codec

  4 décembre 2015, par kiran_g

  I am trying to use libav to relay an RTSP stream. It involves PULLing the stream from an IP camera and then PUSHing to wowza.

  The video encoding in the IP camera stream is h264. To enable h264 in my libav application I need to enable x264. But as x264 is GPL, it will not work with my business plan.

  My questions is whether libav (ffmpeg) can be made to work like a dumb relay which is encoding-agnostic ? so that I dont need to integrate x264 with ffmpeg.

  This SO post says that I can use the "copy" argument, but does that allow me to exclude x264 ?

  BTW, is x264 actually needed by ffmpeg for decoding h264 ? Is x264 only used in encoding ?

  See here for my current code.

• avformat : add a concat protocol that takes a line break delimited list of resources

  29 juin 2021, par James Almer

  avformat : add a concat protocol that takes a line break delimited list of resources
  Suggested-by : ffmpeg@fb.com
  
  Reviewed-by : Nicolas George <george@nsup.org>
  
  Signed-off-by : James Almer <jamrial@gmail.com>
  

  • [DH] Changelog
  • [DH] doc/protocols.texi
  • [DH] libavformat/Makefile
  • [DH] libavformat/concat.c
  • [DH] libavformat/protocols.c
  • [DH] libavformat/version.h

• libswresample : swr_convert() returns empty buffer

  11 septembre 2019, par Герман Лиманський

  I try to convert audio in format AV_SAMPLE_FMT_S32. I use swr_convert(), but out buffer still empty.

  // frame is decoded frame, rframe - is empty frame(out buffer)

      if (!main_context->resampler) {
  
                      main_context->resampler = 
  
                      swr_alloc_set_opts(main_context->resampler,
  
                      AV_CH_LAYOUT_STEREO,                    // output
  
                      AV_SAMPLE_FMT_S32,                      // output
  
                      44100,                                  // output
  
                      audio_codec_context->channel_layout,    // input
  
                      audio_codec_context->sample_fmt,        // input
  
                      audio_codec_context->sample_rate,       // input
  
                      0,
  
                      nullptr);
  
                  swr_init(main_context->resampler);
  
  
  
              }
  
  
  
              //int in_samples = frame->nb_samples;
  
              int out_samples = av_rescale_rnd(swr_get_delay(
  
                  main_context->resampler, 44100) + 44100,
  
                  44100,
  
                  44100,
  
                  AV_ROUND_UP);
  
  
  
              size_t buffSize = av_samples_alloc(rframe->data, NULL,audio_codec_context->channels, out_samples, AV_SAMPLE_FMT_S32, 0);
  
              int len = swr_convert(main_context->resampler, rframe->data, frame->nb_samples, (const uint8_t * *)frame->data, frame->nb_samples);
  
  //here.. rframe->data should have some data, but its empty
  
              while (len > 0)
  
              {
  
                  size_t size_ = rframe->nb_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S32);
  
                  main_context->audio_buf.write(rframe->data[0], size_, 1);
  
  
  
                  len = swr_convert(main_context->resampler, rframe->data, frame->nb_samples, NULL, NULL);
  
              }