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Randomly split the data in audio_resampleLoss 

This helps ensure correct resampling across track boundaries
main
Brick 2024-04-15 14:46:23 +01:00 committed by Sam Lantinga
parent 8f6f9cadc4
commit ae57b0c9d8
1 changed files with 131 additions and 25 deletions
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156
test/testautomation_audio.c
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@ -821,6 +821,130 @@ static double sine_wave_sample(const Sint64 idx, const Sint64 rate, const Sint64
return SDL_sin(((double)(idx * freq % rate)) / ((double)rate) * (SDL_PI_D * 2) + phase);
}
static void free_audio_buffer(void* userdata, const void* buf, int len)
{
SDL_free((void*) buf);
}
/* Split the data into randomly sized chunks */
static int put_audio_data_split(SDL_AudioStream* stream, const void* buf, int len)
{
SDL_AudioSpec spec;
int frame_size;
int ret = SDL_GetAudioStreamFormat(stream, &spec, NULL);
if (ret != 0) {
return ret;
}
frame_size = SDL_AUDIO_FRAMESIZE(spec);
while (len > 0) {
int n = SDLTest_RandomIntegerInRange(1, 10000) * frame_size;
n = SDL_min(n, len);
ret = SDL_PutAudioStreamData(stream, buf, n);
if (ret != 0) {
return ret;
}
buf = ((const Uint8*) buf) + n;
len -= n;
}
return 0;
}
/* Read the data in randomly sized chunks */
static int get_audio_data_split(SDL_AudioStream* stream, void* buf, int len) {
SDL_AudioSpec spec;
int frame_size;
int ret = SDL_GetAudioStreamFormat(stream, NULL, &spec);
int total = 0;
if (ret != 0) {
return ret;
}
frame_size = SDL_AUDIO_FRAMESIZE(spec);
while (len > 0) {
int n = SDLTest_RandomIntegerInRange(1, 10000) * frame_size;
n = SDL_min(n, len);
ret = SDL_GetAudioStreamData(stream, buf, n);
if (ret <= 0) {
return total ? total : ret;
}
buf = ((Uint8*) buf) + ret;
total += ret;
len -= ret;
}
return total;
}
/* Convert the data in chunks, putting/getting randomly sized chunks until finished */
static int convert_audio_chunks(SDL_AudioStream* stream, const void* src, int srclen, void* dst, int dstlen)
{
SDL_AudioSpec src_spec, dst_spec;
int src_frame_size, dst_frame_size;
int total_in = 0, total_out = 0;
int ret = SDL_GetAudioStreamFormat(stream, &src_spec, &dst_spec);
if (ret) {
return ret;
}
src_frame_size = SDL_AUDIO_FRAMESIZE(src_spec);
dst_frame_size = SDL_AUDIO_FRAMESIZE(dst_spec);
while ((total_in < srclen) || (total_out < dstlen)) {
int to_put = SDLTest_RandomIntegerInRange(1, 40000) * src_frame_size;
int to_get = SDLTest_RandomIntegerInRange(1, (int)((40000.0f * dst_spec.freq) / src_spec.freq)) * dst_frame_size;
to_put = SDL_min(to_put, srclen - total_in);
to_get = SDL_min(to_get, dstlen - total_out);
if (to_put)
{
ret = put_audio_data_split(stream, (const Uint8*)(src) + total_in, to_put);
if (ret) {
return total_out ? total_out : ret;
}
total_in += to_put;
if (total_in == srclen) {
ret = SDL_FlushAudioStream(stream);
if (ret) {
return total_out ? total_out : ret;
}
}
}
if (to_get)
{
ret = get_audio_data_split(stream, (Uint8*)(dst) + total_out, to_get);
if ((ret == 0) && (total_in == srclen)) {
ret = -1;
}
if (ret < 0) {
return total_out ? total_out : ret;
}
total_out += ret;
}
}
return total_out;
}
/**
* Check signal-to-noise ratio and maximum error of audio resampling.
*
@ -868,7 +992,6 @@ static int audio_resampleLoss(void *arg)
Uint64 tick_end = 0;
int i = 0;
int j = 0;
int ret = 0;
SDL_AudioStream *stream = NULL;
float *buf_in = NULL;
float *buf_out = NULL;
@ -909,24 +1032,7 @@ static int audio_resampleLoss(void *arg)
}
tick_beg = SDL_GetPerformanceCounter();
ret = SDL_PutAudioStreamData(stream, buf_in, len_in);
SDLTest_AssertPass("Call to SDL_PutAudioStreamData(stream, buf_in, %i)", len_in);
SDLTest_AssertCheck(ret == 0, "Expected SDL_PutAudioStreamData to succeed.");
SDL_free(buf_in);
if (ret != 0) {
SDL_DestroyAudioStream(stream);
return TEST_ABORTED;
}
ret = SDL_FlushAudioStream(stream);
SDLTest_AssertPass("Call to SDL_FlushAudioStream(stream)");
SDLTest_AssertCheck(ret == 0, "Expected SDL_FlushAudioStream to succeed");
if (ret != 0) {
SDL_DestroyAudioStream(stream);
return TEST_ABORTED;
}
buf_out = (float *)SDL_malloc(len_target);
SDLTest_AssertCheck(buf_out != NULL, "Expected output buffer to be created.");
if (buf_out == NULL) {
@ -934,13 +1040,13 @@ static int audio_resampleLoss(void *arg)
return TEST_ABORTED;
}
len_out = SDL_GetAudioStreamData(stream, buf_out, len_target);
SDLTest_AssertPass("Call to SDL_GetAudioStreamData(stream, buf_out, %i)", len_target);
SDLTest_AssertCheck(len_out == len_target, "Expected output length to be no larger than %i, got %i.",
len_out = convert_audio_chunks(stream, buf_in, len_in, buf_out, len_target);
SDLTest_AssertPass("Call to convert_audio_chunks(stream, buf_in, %i, buf_out, %i)", len_in, len_target);
SDLTest_AssertCheck(len_out == len_target, "Expected output length to be %i, got %i.",
len_target, len_out);
SDL_DestroyAudioStream(stream);
if (len_out > len_target) {
SDL_free(buf_out);
SDL_free(buf_in);
if (len_out != len_target) {
SDL_DestroyAudioStream(stream);
return TEST_ABORTED;
}