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• libswresample : swr_convert() not producing enough samples

  20 septembre 2016, par Tsherr

  I’m trying to use ffmpeg/libswresample to resample streaming audio in my c++ application. Changing the sample width works well and the result sounds as one would expect ; however, when changing the sample rate the result is somewhat crackly. I am unsure if it is due to incorrect usage of the libswresample library, or if I’m misunderstanding the resampling theory.

  Here is my resampling process, simplified for demonstration’s sake :

  //Externally supplied data
  
  const uint8_t* in_samples //contains the audio data to be resampled
  
  int in_num_samples = 256
  
  
  
  //Set up resampling context
  
  SwrContext *swr = swr_alloc();
  
  av_opt_set_channel_layout(swr, "in_channel_layout", AV_CH_LAYOUT_STEREO, 0);
  
  av_opt_set_channel_layout(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
  
  av_opt_set_int(swr, "in_sample_rate", 44100, 0);
  
  av_opt_set_int(swr, "out_sample_rate", 22050, 0);
  
  av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLT, 0);
  
  av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_FLT, 0);
  
  swr_init(swr);
  
  
  
  //Perform the resampe
  
  uint8_t* out_samples;
  
  int out_num_samples = av_rescale_rnd(swr_get_delay(swr, in_samplerate) + in_num_samples, out_samplerate, in_samplerate, AV_ROUND_UP);
  
  av_samples_alloc(&out_samples, NULL, out_num_channels, out_num_samples, AV_SAMPLE_FMT_FLT, 0);
  
  out_num_samples = swr_convert(swr, &out_samples, out_num_samples, &in_samples, in_num_samples);
  
  av_freep(&out_samples);
  
  swr_free(&swr);

  I suspect that the reason the resampled audio does not sound right is because swr_convert() returns 112, where I expect it to return 128 (the number of samples of the resampled audio) :
  Downsampling 256 samples from a samplerate of 44100 to a samplerate of 22050 should yield 128 samples, yet swr_convert() is producing 112 samples. When expressed in terms of audio duration this is also puzzling. 256 samples at 44100 = 5.8 ms, but 112 samples at 22050 = 5.07 ms. Shouldn’t the downsampling process not alter the duration of the resampled audio ?

  I have also stepped through an example provided with ffmpeg, in which swr_convert() also returns a smaller number than I would expect. So, I suspect that the problem is not due to a bug in libswresample but rather my own lack of understanding.

• Prevent reporting 200% progress for chunked uploads on Firefox - Closes #2269.

  13 mai 2013, par blueimp

  Prevent reporting 200% progress for chunked uploads on Firefox - Closes #2269.

• vulkan_shaderc : fix error reporting for certain errors

  23 juillet 2024, par Lynne

  vulkan_shaderc : fix error reporting for certain errors
  The issue is that shaderc_result_get_num_errors may sometime
  
  return 0 even when shaderc_result_get_compilation_status returns
  
  a non-zero error code.
  
  Since we use the result from the former, override the status
  
  if it returned 0.
  

  • [DH] libavfilter/vulkan_shaderc.c