From 28b28bd8f4213d6ae9752dfa2fab84737008a702 Mon Sep 17 00:00:00 2001
From: Brick <[EMAIL REDACTED]>
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 aa719045cbee..1ed4b7476d89 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) */