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audio_resampleLoss: Add support for multiple channels

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Brick 2023-08-24 10:47:37 +01:00 • committed by Ryan C. Gordon
parent 4f894e748e
commit f6a4080ff5
1 changed files with 26 additions and 12 deletions
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30
test/testautomation_audio.c
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@ -802,18 +802,22 @@ static int audio_resampleLoss(void *arg)
}; };
int spec_idx = 0; int spec_idx = 0;
int min_channels = 1;
int max_channels = 1 /*8*/;
int num_channels = min_channels;
for (spec_idx = 0; test_specs[spec_idx].time > 0; ++spec_idx) { for (spec_idx = 0; test_specs[spec_idx].time > 0; ++num_channels) {
const struct test_spec_t *spec = &test_specs[spec_idx]; const struct test_spec_t *spec = &test_specs[spec_idx];
const int frames_in = spec->time * spec->rate_in; const int frames_in = spec->time * spec->rate_in;
const int frames_target = spec->time * spec->rate_out; const int frames_target = spec->time * spec->rate_out;
const int len_in = frames_in * (int)sizeof(float); const int len_in = (frames_in * num_channels) * (int)sizeof(float);
const int len_target = frames_target * (int)sizeof(float); const int len_target = (frames_target * num_channels) * (int)sizeof(float);
SDL_AudioSpec tmpspec1, tmpspec2; SDL_AudioSpec tmpspec1, tmpspec2;
Uint64 tick_beg = 0; Uint64 tick_beg = 0;
Uint64 tick_end = 0; Uint64 tick_end = 0;
int i = 0; int i = 0;
int j = 0;
int ret = 0; int ret = 0;
SDL_AudioStream *stream = NULL; SDL_AudioStream *stream = NULL;
float *buf_in = NULL; float *buf_in = NULL;
@ -824,14 +828,19 @@ static int audio_resampleLoss(void *arg)
double sum_squared_value = 0; double sum_squared_value = 0;
double signal_to_noise = 0; double signal_to_noise = 0;
if (num_channels > max_channels) {
num_channels = 1;
++spec_idx;
}
SDLTest_AssertPass("Test resampling of %i s %i Hz %f phase sine wave from sampling rate of %i Hz to %i Hz", SDLTest_AssertPass("Test resampling of %i s %i Hz %f phase sine wave from sampling rate of %i Hz to %i Hz",
spec->time, spec->freq, spec->phase, spec->rate_in, spec->rate_out); spec->time, spec->freq, spec->phase, spec->rate_in, spec->rate_out);
tmpspec1.format = SDL_AUDIO_F32; tmpspec1.format = SDL_AUDIO_F32;
tmpspec1.channels = 1; tmpspec1.channels = num_channels;
tmpspec1.freq = spec->rate_in; tmpspec1.freq = spec->rate_in;
tmpspec2.format = SDL_AUDIO_F32; tmpspec2.format = SDL_AUDIO_F32;
tmpspec2.channels = 1; tmpspec2.channels = num_channels;
tmpspec2.freq = spec->rate_out; tmpspec2.freq = spec->rate_out;
stream = SDL_CreateAudioStream(&tmpspec1, &tmpspec2); stream = SDL_CreateAudioStream(&tmpspec1, &tmpspec2);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(SDL_AUDIO_F32, 1, %i, SDL_AUDIO_F32, 1, %i)", spec->rate_in, spec->rate_out); SDLTest_AssertPass("Call to SDL_CreateAudioStream(SDL_AUDIO_F32, 1, %i, SDL_AUDIO_F32, 1, %i)", spec->rate_in, spec->rate_out);
@ -848,7 +857,10 @@ static int audio_resampleLoss(void *arg)
} }
for (i = 0; i < frames_in; ++i) { for (i = 0; i < frames_in; ++i) {
*(buf_in + i) = (float)sine_wave_sample(i, spec->rate_in, spec->freq, spec->phase); float f = (float)sine_wave_sample(i, spec->rate_in, spec->freq, spec->phase);
for (j = 0; j < num_channels; ++j) {
*(buf_in + (i * num_channels) + j) = f;
}
} }
tick_beg = SDL_GetPerformanceCounter(); tick_beg = SDL_GetPerformanceCounter();
@ -890,14 +902,16 @@ static int audio_resampleLoss(void *arg)
tick_end = SDL_GetPerformanceCounter(); tick_end = SDL_GetPerformanceCounter();
SDLTest_Log("Resampling used %f seconds.", ((double)(tick_end - tick_beg)) / SDL_GetPerformanceFrequency()); SDLTest_Log("Resampling used %f seconds.", ((double)(tick_end - tick_beg)) / SDL_GetPerformanceFrequency());
for (i = 0; i < len_out / (int)sizeof(float); ++i) { for (i = 0; i < frames_target; ++i) {
const float output = *(buf_out + i);
const double target = sine_wave_sample(i, spec->rate_out, spec->freq, spec->phase); const double target = sine_wave_sample(i, spec->rate_out, spec->freq, spec->phase);
for (j = 0; j < num_channels; ++j) {
const float output = *(buf_out + (i * num_channels) + j);
const double error = SDL_fabs(target - output); const double error = SDL_fabs(target - output);
max_error = SDL_max(max_error, error); max_error = SDL_max(max_error, error);
sum_squared_error += error * error; sum_squared_error += error * error;
sum_squared_value += target * target; sum_squared_value += target * target;
} }
}
SDL_free(buf_out); SDL_free(buf_out);
signal_to_noise = 10 * SDL_log10(sum_squared_value / sum_squared_error); /* decibel */ signal_to_noise = 10 * SDL_log10(sum_squared_value / sum_squared_error); /* decibel */
SDLTest_AssertCheck(isfinite(sum_squared_value), "Sum of squared target should be finite."); SDLTest_AssertCheck(isfinite(sum_squared_value), "Sum of squared target should be finite.");