예제 #1
0
파일: main.c 프로젝트: Jalle19/RetroArch
int main(int argc, char *argv[])
{
   int16_t input_i[1024];
   int16_t output_i[1024 * 8];

   float input_f[1024];
   float output_f[1024 * 8];

   if (argc != 3)
   {
      fprintf(stderr, "Usage: %s <in-rate> <out-rate> (max ratio: 8.0)\n", argv[0]);
      return 1;
   }

   double in_rate = strtod(argv[1], NULL);
   double out_rate = strtod(argv[2], NULL);

   double ratio = out_rate / in_rate;
   if (ratio >= 7.99)
   {
      fprintf(stderr, "Ratio is too high.\n");
      return 1;
   }

   const rarch_resampler_t *resampler = NULL;
   void *re = NULL;
   if (!rarch_resampler_realloc(&re, &resampler, NULL))
   {
      fprintf(stderr, "Failed to allocate resampler ...\n");
      return 1;
   }

   for (;;)
   {
      if (fread(input_i, sizeof(int16_t), 1024, stdin) != 1024)
         break;

      audio_convert_s16_to_float(input_f, input_i, 1024, 1.0f);

      struct resampler_data data = {
         .data_in = input_f,
         .data_out = output_f,
         .input_frames = sizeof(input_f) / (2 * sizeof(float)),
         .ratio = ratio,
      };

      rarch_resampler_process(resampler, re, &data);

      size_t output_samples = data.output_frames * 2;

      audio_convert_float_to_s16(output_i, output_f, output_samples);

      if (fwrite(output_i, sizeof(int16_t), output_samples, stdout) != output_samples)
         break;
   }

   rarch_resampler_freep(&resampler, &re);
}
예제 #2
0
static void ffmpeg_audio_resample(ffmpeg_t *handle, struct ffemu_audio_data *data)
{
   if (!handle->audio.use_float && !handle->audio.resampler)
      return;

   if (data->frames > handle->audio.float_conv_frames)
   {
      handle->audio.float_conv = (float*)av_realloc(handle->audio.float_conv,
            data->frames * handle->params.channels * sizeof(float));
      if (!handle->audio.float_conv)
         return;

      handle->audio.float_conv_frames = data->frames;

      // To make sure we don't accidentially overflow.
      handle->audio.resample_out_frames = data->frames * handle->audio.ratio + 16;

      handle->audio.resample_out = (float*)av_realloc(handle->audio.resample_out,
            handle->audio.resample_out_frames * handle->params.channels * sizeof(float));
      if (!handle->audio.resample_out)
         return;

      handle->audio.fixed_conv_frames = max(handle->audio.resample_out_frames, handle->audio.float_conv_frames);
      handle->audio.fixed_conv = (int16_t*)av_realloc(handle->audio.fixed_conv,
            handle->audio.fixed_conv_frames * handle->params.channels * sizeof(int16_t));
      if (!handle->audio.fixed_conv)
         return;
   }

   if (handle->audio.use_float || handle->audio.resampler)
   {
      audio_convert_s16_to_float(handle->audio.float_conv,
            (const int16_t*)data->data, data->frames * handle->params.channels, 1.0);
      data->data = handle->audio.float_conv;
   }

   if (handle->audio.resampler)
   {
      // It's always two channels ...
      struct resampler_data info = {0};
      info.data_in      = (const float*)data->data;
      info.data_out     = handle->audio.resample_out;
      info.input_frames = data->frames;
      info.ratio        = handle->audio.ratio;

      rarch_resampler_process(handle->audio.resampler, handle->audio.resampler_data, &info);
      data->data   = handle->audio.resample_out;
      data->frames = info.output_frames;

      if (!handle->audio.use_float)
      {
         audio_convert_float_to_s16(handle->audio.fixed_conv, handle->audio.resample_out,
               data->frames * handle->params.channels);
         data->data = handle->audio.fixed_conv;
      }
   }
}
예제 #3
0
int main(int argc, char *argv[])
{
   srand(time(NULL));
   int16_t input_i[1024];
   int16_t output_i[1024 * 8];

   float input_f[1024];
   float output_f[1024 * 8];

   double ratio_max_deviation = 0.0;

   if (argc < 3 || argc > 4)
   {
      fprintf(stderr, "Usage: %s <in-rate> <out-rate> [ratio deviation] (max ratio: 8.0)\n", argv[0]);
      return 1;
   }
   else if (argc == 4)
   {
      ratio_max_deviation = fabs(strtod(argv[3], NULL));
      fprintf(stderr, "Ratio deviation: %.4f.\n", ratio_max_deviation);
   }

   double in_rate = strtod(argv[1], NULL);
   double out_rate = strtod(argv[2], NULL);

   double ratio = out_rate / in_rate;
   if (ratio >= 7.99)
   {
      fprintf(stderr, "Ratio is too high.\n");
      return 1;
   }

   const rarch_resampler_t *resampler = NULL;
   void *re = NULL;
   if (!rarch_resampler_realloc(&re, &resampler, RESAMPLER_IDENT, out_rate / in_rate))
   {
      fprintf(stderr, "Failed to allocate resampler ...\n");
      return 1;
   }

   for (;;)
   {
      if (fread(input_i, sizeof(int16_t), 1024, stdin) != 1024)
         break;

      double uniform = (2.0 * rand()) / RAND_MAX - 1.0;
      double rate_mod = 1.0 + ratio_max_deviation * uniform;

      audio_convert_s16_to_float(input_f, input_i, 1024, 1.0f);

      struct resampler_data data = {
         .data_in = input_f,
         .data_out = output_f,
         .input_frames = sizeof(input_f) / (2 * sizeof(float)),
         .ratio = ratio * rate_mod,
      };

      rarch_resampler_process(resampler, re, &data);

      size_t output_samples = data.output_frames * 2;

      audio_convert_float_to_s16(output_i, output_f, output_samples);

      if (fwrite(output_i, sizeof(int16_t), output_samples, stdout) != output_samples)
         break;
   }

   rarch_resampler_freep(&resampler, &re);
}