From 28b28bd8f4213d6ae9752dfa2fab84737008a702 Mon Sep 17 00:00:00 2001 From: Brick <6098371+0x1F9F1@users.noreply.github.com> Date: Thu, 31 Aug 2023 22:35:59 +0100 Subject: [PATCH] Added audio_formatChange test --- test/testautomation_audio.c | 159 +++++++++++++++++++++++++++++++++++- 1 file changed, 158 insertions(+), 1 deletion(-) diff --git a/test/testautomation_audio.c b/test/testautomation_audio.c index aa719045c..1ed4b7476 100644 --- a/test/testautomation_audio.c +++ b/test/testautomation_audio.c @@ -1073,6 +1073,159 @@ static int audio_convertAccuracy(void *arg) return TEST_COMPLETED; } + +/** + * \brief Check accuracy when switching between formats + * + * \sa SDL_SetAudioStreamFormat + */ +static int audio_formatChange(void *arg) +{ + int i; + SDL_AudioSpec spec1, spec2, spec3; + int frames_1, frames_2, frames_3; + int length_1, length_2, length_3; + int retval = 0; + int status = TEST_ABORTED; + float* buffer_1 = NULL; + float* buffer_2 = NULL; + float* buffer_3 = NULL; + SDL_AudioStream* stream = NULL; + double max_error = 0; + double sum_squared_error = 0; + double sum_squared_value = 0; + double signal_to_noise = 0; + double target_max_error = 0.02; + double target_signal_to_noise = 75.0; + int sine_freq = 500; + + spec1.format = SDL_AUDIO_F32SYS; + spec1.channels = 1; + spec1.freq = 20000; + + spec2.format = SDL_AUDIO_F32SYS; + spec2.channels = 1; + spec2.freq = 40000; + + spec3.format = SDL_AUDIO_F32SYS; + spec3.channels = 1; + spec3.freq = 80000; + + frames_1 = spec1.freq; + frames_2 = spec2.freq; + frames_3 = spec3.freq * 2; + + length_1 = (int)(frames_1 * sizeof(*buffer_1)); + buffer_1 = (float*) SDL_malloc(length_1); + if (!SDLTest_AssertCheck(buffer_1 != NULL, "Expected buffer_1 to be created.")) { + goto cleanup; + } + + length_2 = (int)(frames_2 * sizeof(*buffer_2)); + buffer_2 = (float*) SDL_malloc(length_2); + if (!SDLTest_AssertCheck(buffer_2 != NULL, "Expected buffer_2 to be created.")) { + goto cleanup; + } + + length_3 = (int)(frames_3 * sizeof(*buffer_3)); + buffer_3 = (float*) SDL_malloc(length_3); + if (!SDLTest_AssertCheck(buffer_3 != NULL, "Expected buffer_3 to be created.")) { + goto cleanup; + } + + for (i = 0; i < frames_1; ++i) { + buffer_1[i] = (float) sine_wave_sample(i, spec1.freq, sine_freq, 0.0f); + } + + for (i = 0; i < frames_2; ++i) { + buffer_2[i] = (float) sine_wave_sample(i, spec2.freq, sine_freq, 0.0f); + } + + stream = SDL_CreateAudioStream(NULL, NULL); + if (!SDLTest_AssertCheck(stream != NULL, "Expected SDL_CreateAudioStream to succeed")) { + goto cleanup; + } + + retval = SDL_SetAudioStreamFormat(stream, &spec1, &spec3); + if (!SDLTest_AssertCheck(retval == 0, "Expected SDL_SetAudioStreamFormat(spec1, spec3) to succeed")) { + goto cleanup; + } + + retval = SDL_GetAudioStreamAvailable(stream); + if (!SDLTest_AssertCheck(retval == 0, "Expected SDL_GetAudioStreamAvailable return 0")) { + goto cleanup; + } + + retval = SDL_PutAudioStreamData(stream, buffer_1, length_1); + if (!SDLTest_AssertCheck(retval == 0, "Expected SDL_PutAudioStreamData(buffer_1) to succeed")) { + goto cleanup; + } + + retval = SDL_FlushAudioStream(stream); + if (!SDLTest_AssertCheck(retval == 0, "Expected SDL_FlushAudioStream to succeed")) { + goto cleanup; + } + + retval = SDL_SetAudioStreamFormat(stream, &spec2, &spec3); + if (!SDLTest_AssertCheck(retval == 0, "Expected SDL_SetAudioStreamFormat(spec2, spec3) to succeed")) { + goto cleanup; + } + + retval = SDL_PutAudioStreamData(stream, buffer_2, length_2); + if (!SDLTest_AssertCheck(retval == 0, "Expected SDL_PutAudioStreamData(buffer_1) to succeed")) { + goto cleanup; + } + + retval = SDL_FlushAudioStream(stream); + if (!SDLTest_AssertCheck(retval == 0, "Expected SDL_FlushAudioStream to succeed")) { + goto cleanup; + } + + retval = SDL_GetAudioStreamAvailable(stream); + if (!SDLTest_AssertCheck(retval == length_3, "Expected SDL_GetAudioStreamAvailable to return %i, got %i", length_3, retval)) { + goto cleanup; + } + + retval = SDL_GetAudioStreamData(stream, buffer_3, length_3); + if (!SDLTest_AssertCheck(retval == length_3, "Expected SDL_GetAudioStreamData to return %i, got %i", length_3, retval)) { + goto cleanup; + } + + retval = SDL_GetAudioStreamAvailable(stream); + if (!SDLTest_AssertCheck(retval == 0, "Expected SDL_GetAudioStreamAvailable to return 0")) { + goto cleanup; + } + + for (i = 0; i < frames_3; ++i) { + const float output = buffer_3[i]; + const float target = (float) sine_wave_sample(i, spec3.freq, sine_freq, 0.0f); + const double error = SDL_fabs(target - output); + max_error = SDL_max(max_error, error); + sum_squared_error += error * error; + sum_squared_value += target * target; + } + + 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_error), "Sum of squared error should be finite."); + /* Infinity is theoretically possible when there is very little to no noise */ + SDLTest_AssertCheck(!isnan(signal_to_noise), "Signal-to-noise ratio should not be NaN."); + SDLTest_AssertCheck(isfinite(max_error), "Maximum conversion error should be finite."); + SDLTest_AssertCheck(signal_to_noise >= target_signal_to_noise, "Conversion signal-to-noise ratio %f dB should be no less than %f dB.", + signal_to_noise, target_signal_to_noise); + SDLTest_AssertCheck(max_error <= target_max_error, "Maximum conversion error %f should be no more than %f.", + max_error, target_max_error); + + status = TEST_COMPLETED; + +cleanup: + SDL_free(buffer_1); + SDL_free(buffer_2); + SDL_free(buffer_3); + SDL_DestroyAudioStream(stream); + + return status; +} /* ================= Test Case References ================== */ /* Audio test cases */ @@ -1146,12 +1299,16 @@ static const SDLTest_TestCaseReference audioTest17 = { audio_convertAccuracy, "audio_convertAccuracy", "Check accuracy converting between audio formats.", TEST_ENABLED }; +static const SDLTest_TestCaseReference audioTest18 = { + audio_formatChange, "audio_formatChange", "Check handling of format changes.", TEST_ENABLED +}; + /* Sequence of Audio test cases */ static const SDLTest_TestCaseReference *audioTests[] = { &audioTest1, &audioTest2, &audioTest3, &audioTest4, &audioTest5, &audioTest6, &audioTest7, &audioTest8, &audioTest9, &audioTest10, &audioTest11, &audioTest12, &audioTest13, &audioTest14, &audioTest15, &audioTest16, - &audioTest17, NULL + &audioTest17, &audioTest18, NULL }; /* Audio test suite (global) */