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);
}
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;
}
}
}
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);
}