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SDL/test/testautomation_audio.c
Brick edaab8ad9f Refactored audio conversion to reduce copying 
More of the logic has been moved into SDL_AudioQueue,
allowing data to be converted directly from the input buffer.
2024-04-15 11:47:18 -10:00

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/**
* Original code: automated SDL audio test written by Edgar Simo "bobbens"
* New/updated tests: aschiffler at ferzkopp dot net
*/
/* quiet windows compiler warnings */
#if defined(_MSC_VER) && !defined(_CRT_SECURE_NO_WARNINGS)
#define _CRT_SECURE_NO_WARNINGS
#endif
#include <math.h>
#include <stdio.h>
#include <SDL3/SDL.h>
#include <SDL3/SDL_test.h>
#include "testautomation_suites.h"
/* ================= Test Case Implementation ================== */
/* Fixture */
static void audioSetUp(void *arg)
{
/* Start SDL audio subsystem */
int ret = SDL_InitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_InitSubSystem(SDL_INIT_AUDIO)");
SDLTest_AssertCheck(ret == 0, "Check result from SDL_InitSubSystem(SDL_INIT_AUDIO)");
if (ret != 0) {
SDLTest_LogError("%s", SDL_GetError());
}
}
static void audioTearDown(void *arg)
{
/* Remove a possibly created file from SDL disk writer audio driver; ignore errors */
(void)remove("sdlaudio.raw");
SDLTest_AssertPass("Cleanup of test files completed");
}
#if 0 /* !!! FIXME: maybe update this? */
/* Global counter for callback invocation */
static int g_audio_testCallbackCounter;
/* Global accumulator for total callback length */
static int g_audio_testCallbackLength;
/* Test callback function */
static void SDLCALL audio_testCallback(void *userdata, Uint8 *stream, int len)
{
/* track that callback was called */
g_audio_testCallbackCounter++;
g_audio_testCallbackLength += len;
}
#endif
static SDL_AudioDeviceID g_audio_id = 0;
/* Test case functions */
/**
* Stop and restart audio subsystem
*
* \sa SDL_QuitSubSystem
* \sa SDL_InitSubSystem
*/
static int audio_quitInitAudioSubSystem(void *arg)
{
/* Stop SDL audio subsystem */
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO)");
/* Restart audio again */
audioSetUp(NULL);
return TEST_COMPLETED;
}
/**
* Start and stop audio directly
*
* \sa SDL_InitAudio
* \sa SDL_QuitAudio
*/
static int audio_initQuitAudio(void *arg)
{
int result;
int i, iMax;
const char *audioDriver;
const char *hint = SDL_GetHint(SDL_HINT_AUDIO_DRIVER);
/* Stop SDL audio subsystem */
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO)");
/* Loop over all available audio drivers */
iMax = SDL_GetNumAudioDrivers();
SDLTest_AssertPass("Call to SDL_GetNumAudioDrivers()");
SDLTest_AssertCheck(iMax > 0, "Validate number of audio drivers; expected: >0 got: %d", iMax);
for (i = 0; i < iMax; i++) {
audioDriver = SDL_GetAudioDriver(i);
SDLTest_AssertPass("Call to SDL_GetAudioDriver(%d)", i);
SDLTest_Assert(audioDriver != NULL, "Audio driver name is not NULL");
SDLTest_AssertCheck(audioDriver[0] != '\0', "Audio driver name is not empty; got: %s", audioDriver); /* NOLINT(clang-analyzer-core.NullDereference): Checked for NULL above */
if (hint && SDL_strcmp(audioDriver, hint) != 0) {
continue;
}
/* Call Init */
SDL_SetHint("SDL_AUDIO_DRIVER", audioDriver);
result = SDL_InitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_InitSubSystem(SDL_INIT_AUDIO) with driver='%s'", audioDriver);
SDLTest_AssertCheck(result == 0, "Validate result value; expected: 0 got: %d", result);
/* Call Quit */
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO)");
}
/* NULL driver specification */
audioDriver = NULL;
/* Call Init */
SDL_SetHint("SDL_AUDIO_DRIVER", audioDriver);
result = SDL_InitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_AudioInit(NULL)");
SDLTest_AssertCheck(result == 0, "Validate result value; expected: 0 got: %d", result);
/* Call Quit */
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO)");
/* Restart audio again */
audioSetUp(NULL);
return TEST_COMPLETED;
}
/**
* Start, open, close and stop audio
*
* \sa SDL_InitAudio
* \sa SDL_OpenAudioDevice
* \sa SDL_CloseAudioDevice
* \sa SDL_QuitAudio
*/
static int audio_initOpenCloseQuitAudio(void *arg)
{
int result;
int i, iMax, j, k;
const char *audioDriver;
SDL_AudioSpec desired;
const char *hint = SDL_GetHint(SDL_HINT_AUDIO_DRIVER);
/* Stop SDL audio subsystem */
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO)");
/* Loop over all available audio drivers */
iMax = SDL_GetNumAudioDrivers();
SDLTest_AssertPass("Call to SDL_GetNumAudioDrivers()");
SDLTest_AssertCheck(iMax > 0, "Validate number of audio drivers; expected: >0 got: %d", iMax);
for (i = 0; i < iMax; i++) {
audioDriver = SDL_GetAudioDriver(i);
SDLTest_AssertPass("Call to SDL_GetAudioDriver(%d)", i);
SDLTest_Assert(audioDriver != NULL, "Audio driver name is not NULL");
SDLTest_AssertCheck(audioDriver[0] != '\0', "Audio driver name is not empty; got: %s", audioDriver); /* NOLINT(clang-analyzer-core.NullDereference): Checked for NULL above */
if (hint && SDL_strcmp(audioDriver, hint) != 0) {
continue;
}
/* Change specs */
for (j = 0; j < 2; j++) {
/* Call Init */
SDL_SetHint("SDL_AUDIO_DRIVER", audioDriver);
result = SDL_InitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_InitSubSystem(SDL_INIT_AUDIO) with driver='%s'", audioDriver);
SDLTest_AssertCheck(result == 0, "Validate result value; expected: 0 got: %d", result);
/* Set spec */
SDL_memset(&desired, 0, sizeof(desired));
switch (j) {
case 0:
/* Set standard desired spec */
desired.freq = 22050;
desired.format = SDL_AUDIO_S16;
desired.channels = 2;
break;
case 1:
/* Set custom desired spec */
desired.freq = 48000;
desired.format = SDL_AUDIO_F32;
desired.channels = 2;
break;
}
/* Call Open (maybe multiple times) */
for (k = 0; k <= j; k++) {
result = SDL_OpenAudioDevice(SDL_AUDIO_DEVICE_DEFAULT_OUTPUT, &desired);
if (k == 0) {
g_audio_id = result;
}
SDLTest_AssertPass("Call to SDL_OpenAudioDevice(SDL_AUDIO_DEVICE_DEFAULT_OUTPUT, desired_spec_%d), call %d", j, k + 1);
SDLTest_AssertCheck(result > 0, "Verify return value; expected: > 0, got: %d", result);
}
/* Call Close (maybe multiple times) */
for (k = 0; k <= j; k++) {
SDL_CloseAudioDevice(g_audio_id);
SDLTest_AssertPass("Call to SDL_CloseAudioDevice(), call %d", k + 1);
}
/* Call Quit (maybe multiple times) */
for (k = 0; k <= j; k++) {
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO), call %d", k + 1);
}
} /* spec loop */
} /* driver loop */
/* Restart audio again */
audioSetUp(NULL);
return TEST_COMPLETED;
}
/**
* Pause and unpause audio
*
* \sa SDL_PauseAudioDevice
* \sa SDL_PlayAudioDevice
*/
static int audio_pauseUnpauseAudio(void *arg)
{
int iMax;
int i, j /*, k, l*/;
int result;
const char *audioDriver;
SDL_AudioSpec desired;
const char *hint = SDL_GetHint(SDL_HINT_AUDIO_DRIVER);
/* Stop SDL audio subsystem */
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO)");
/* Loop over all available audio drivers */
iMax = SDL_GetNumAudioDrivers();
SDLTest_AssertPass("Call to SDL_GetNumAudioDrivers()");
SDLTest_AssertCheck(iMax > 0, "Validate number of audio drivers; expected: >0 got: %d", iMax);
for (i = 0; i < iMax; i++) {
audioDriver = SDL_GetAudioDriver(i);
SDLTest_AssertPass("Call to SDL_GetAudioDriver(%d)", i);
SDLTest_Assert(audioDriver != NULL, "Audio driver name is not NULL");
SDLTest_AssertCheck(audioDriver[0] != '\0', "Audio driver name is not empty; got: %s", audioDriver); /* NOLINT(clang-analyzer-core.NullDereference): Checked for NULL above */
if (hint && SDL_strcmp(audioDriver, hint) != 0) {
continue;
}
/* Change specs */
for (j = 0; j < 2; j++) {
/* Call Init */
SDL_SetHint("SDL_AUDIO_DRIVER", audioDriver);
result = SDL_InitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_InitSubSystem(SDL_INIT_AUDIO) with driver='%s'", audioDriver);
SDLTest_AssertCheck(result == 0, "Validate result value; expected: 0 got: %d", result);
/* Set spec */
SDL_memset(&desired, 0, sizeof(desired));
switch (j) {
case 0:
/* Set standard desired spec */
desired.freq = 22050;
desired.format = SDL_AUDIO_S16;
desired.channels = 2;
break;
case 1:
/* Set custom desired spec */
desired.freq = 48000;
desired.format = SDL_AUDIO_F32;
desired.channels = 2;
break;
}
/* Call Open */
g_audio_id = SDL_OpenAudioDevice(SDL_AUDIO_DEVICE_DEFAULT_OUTPUT, &desired);
result = g_audio_id;
SDLTest_AssertPass("Call to SDL_OpenAudioDevice(SDL_AUDIO_DEVICE_DEFAULT_OUTPUT, desired_spec_%d)", j);
SDLTest_AssertCheck(result > 0, "Verify return value; expected > 0 got: %d", result);
#if 0 /* !!! FIXME: maybe update this? */
/* Start and stop audio multiple times */
for (l = 0; l < 3; l++) {
SDLTest_Log("Pause/Unpause iteration: %d", l + 1);
/* Reset callback counters */
g_audio_testCallbackCounter = 0;
g_audio_testCallbackLength = 0;
/* Un-pause audio to start playing (maybe multiple times) */
for (k = 0; k <= j; k++) {
SDL_PlayAudioDevice(g_audio_id);
SDLTest_AssertPass("Call to SDL_PlayAudioDevice(g_audio_id), call %d", k + 1);
}
/* Wait for callback */
int totalDelay = 0;
do {
SDL_Delay(10);
totalDelay += 10;
} while (g_audio_testCallbackCounter == 0 && totalDelay < 1000);
SDLTest_AssertCheck(g_audio_testCallbackCounter > 0, "Verify callback counter; expected: >0 got: %d", g_audio_testCallbackCounter);
SDLTest_AssertCheck(g_audio_testCallbackLength > 0, "Verify callback length; expected: >0 got: %d", g_audio_testCallbackLength);
/* Pause audio to stop playing (maybe multiple times) */
for (k = 0; k <= j; k++) {
const int pause_on = (k == 0) ? 1 : SDLTest_RandomIntegerInRange(99, 9999);
if (pause_on) {
SDL_PauseAudioDevice(g_audio_id);
SDLTest_AssertPass("Call to SDL_PauseAudioDevice(g_audio_id), call %d", k + 1);
} else {
SDL_PlayAudioDevice(g_audio_id);
SDLTest_AssertPass("Call to SDL_PlayAudioDevice(g_audio_id), call %d", k + 1);
}
}
/* Ensure callback is not called again */
const int originalCounter = g_audio_testCallbackCounter;
SDL_Delay(totalDelay + 10);
SDLTest_AssertCheck(originalCounter == g_audio_testCallbackCounter, "Verify callback counter; expected: %d, got: %d", originalCounter, g_audio_testCallbackCounter);
}
#endif
/* Call Close */
SDL_CloseAudioDevice(g_audio_id);
SDLTest_AssertPass("Call to SDL_CloseAudioDevice()");
/* Call Quit */
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO)");
} /* spec loop */
} /* driver loop */
/* Restart audio again */
audioSetUp(NULL);
return TEST_COMPLETED;
}
/**
* Enumerate and name available audio devices (output and capture).
*
* \sa SDL_GetNumAudioDevices
* \sa SDL_GetAudioDeviceName
*/
static int audio_enumerateAndNameAudioDevices(void *arg)
{
int t;
int i, n;
char *name;
SDL_AudioDeviceID *devices = NULL;
/* Iterate over types: t=0 output device, t=1 input/capture device */
for (t = 0; t < 2; t++) {
/* Get number of devices. */
devices = (t) ? SDL_GetAudioCaptureDevices(&n) : SDL_GetAudioOutputDevices(&n);
SDLTest_AssertPass("Call to SDL_GetAudio%sDevices(%i)", (t) ? "Capture" : "Output", t);
SDLTest_Log("Number of %s devices < 0, reported as %i", (t) ? "capture" : "output", n);
SDLTest_AssertCheck(n >= 0, "Validate result is >= 0, got: %i", n);
/* List devices. */
if (n > 0) {
SDLTest_AssertCheck(devices != NULL, "Validate devices is not NULL if n > 0");
for (i = 0; i < n; i++) {
name = SDL_GetAudioDeviceName(devices[i]);
SDLTest_AssertPass("Call to SDL_GetAudioDeviceName(%i)", i);
SDLTest_AssertCheck(name != NULL, "Verify result from SDL_GetAudioDeviceName(%i) is not NULL", i);
if (name != NULL) {
SDLTest_AssertCheck(name[0] != '\0', "verify result from SDL_GetAudioDeviceName(%i) is not empty, got: '%s'", i, name);
SDL_free(name);
}
}
}
SDL_free(devices);
}
return TEST_COMPLETED;
}
/**
* Negative tests around enumeration and naming of audio devices.
*
* \sa SDL_GetNumAudioDevices
* \sa SDL_GetAudioDeviceName
*/
static int audio_enumerateAndNameAudioDevicesNegativeTests(void *arg)
{
return TEST_COMPLETED; /* nothing in here atm since these interfaces changed in SDL3. */
}
/**
* Checks available audio driver names.
*
* \sa SDL_GetNumAudioDrivers
* \sa SDL_GetAudioDriver
*/
static int audio_printAudioDrivers(void *arg)
{
int i, n;
const char *name;
/* Get number of drivers */
n = SDL_GetNumAudioDrivers();
SDLTest_AssertPass("Call to SDL_GetNumAudioDrivers()");
SDLTest_AssertCheck(n >= 0, "Verify number of audio drivers >= 0, got: %i", n);
/* List drivers. */
if (n > 0) {
for (i = 0; i < n; i++) {
name = SDL_GetAudioDriver(i);
SDLTest_AssertPass("Call to SDL_GetAudioDriver(%i)", i);
SDLTest_AssertCheck(name != NULL, "Verify returned name is not NULL");
if (name != NULL) {
SDLTest_AssertCheck(name[0] != '\0', "Verify returned name is not empty, got: '%s'", name);
}
}
}
return TEST_COMPLETED;
}
/**
* Checks current audio driver name with initialized audio.
*
* \sa SDL_GetCurrentAudioDriver
*/
static int audio_printCurrentAudioDriver(void *arg)
{
/* Check current audio driver */
const char *name = SDL_GetCurrentAudioDriver();
SDLTest_AssertPass("Call to SDL_GetCurrentAudioDriver()");
SDLTest_AssertCheck(name != NULL, "Verify returned name is not NULL");
if (name != NULL) {
SDLTest_AssertCheck(name[0] != '\0', "Verify returned name is not empty, got: '%s'", name);
}
return TEST_COMPLETED;
}
/* Definition of all formats, channels, and frequencies used to test audio conversions */
static SDL_AudioFormat g_audioFormats[] = {
SDL_AUDIO_S8, SDL_AUDIO_U8,
SDL_AUDIO_S16LE, SDL_AUDIO_S16BE,
SDL_AUDIO_S32LE, SDL_AUDIO_S32BE,
SDL_AUDIO_F32LE, SDL_AUDIO_F32BE
};
static const char *g_audioFormatsVerbose[] = {
"SDL_AUDIO_S8", "SDL_AUDIO_U8",
"SDL_AUDIO_S16LE", "SDL_AUDIO_S16BE",
"SDL_AUDIO_S32LE", "SDL_AUDIO_S32BE",
"SDL_AUDIO_F32LE", "SDL_AUDIO_F32BE"
};
static const int g_numAudioFormats = SDL_arraysize(g_audioFormats);
static Uint8 g_audioChannels[] = { 1, 2, 4, 6 };
static const int g_numAudioChannels = SDL_arraysize(g_audioChannels);
static int g_audioFrequencies[] = { 11025, 22050, 44100, 48000 };
static const int g_numAudioFrequencies = SDL_arraysize(g_audioFrequencies);
/* Verify the audio formats are laid out as expected */
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_U8_FORMAT, SDL_AUDIO_U8 == SDL_AUDIO_BITSIZE(8));
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_S8_FORMAT, SDL_AUDIO_S8 == (SDL_AUDIO_BITSIZE(8) | SDL_AUDIO_MASK_SIGNED));
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_S16LE_FORMAT, SDL_AUDIO_S16LE == (SDL_AUDIO_BITSIZE(16) | SDL_AUDIO_MASK_SIGNED));
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_S16BE_FORMAT, SDL_AUDIO_S16BE == (SDL_AUDIO_S16LE | SDL_AUDIO_MASK_BIG_ENDIAN));
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_S32LE_FORMAT, SDL_AUDIO_S32LE == (SDL_AUDIO_BITSIZE(32) | SDL_AUDIO_MASK_SIGNED));
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_S32BE_FORMAT, SDL_AUDIO_S32BE == (SDL_AUDIO_S32LE | SDL_AUDIO_MASK_BIG_ENDIAN));
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_F32LE_FORMAT, SDL_AUDIO_F32LE == (SDL_AUDIO_BITSIZE(32) | SDL_AUDIO_MASK_FLOAT | SDL_AUDIO_MASK_SIGNED));
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_F32BE_FORMAT, SDL_AUDIO_F32BE == (SDL_AUDIO_F32LE | SDL_AUDIO_MASK_BIG_ENDIAN));
/**
* Builds various audio conversion structures
*
* \sa SDL_CreateAudioStream
*/
static int audio_buildAudioStream(void *arg)
{
SDL_AudioStream *stream;
SDL_AudioSpec spec1;
SDL_AudioSpec spec2;
int i, ii, j, jj, k, kk;
/* Call Quit */
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO)");
/* No conversion needed */
spec1.format = SDL_AUDIO_S16LE;
spec1.channels = 2;
spec1.freq = 22050;
stream = SDL_CreateAudioStream(&spec1, &spec1);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(spec1 ==> spec1)");
SDLTest_AssertCheck(stream != NULL, "Verify stream value; expected: != NULL, got: %p", (void *)stream);
SDL_DestroyAudioStream(stream);
/* Typical conversion */
spec1.format = SDL_AUDIO_S8;
spec1.channels = 1;
spec1.freq = 22050;
spec2.format = SDL_AUDIO_S16LE;
spec2.channels = 2;
spec2.freq = 44100;
stream = SDL_CreateAudioStream(&spec1, &spec2);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(spec1 ==> spec2)");
SDLTest_AssertCheck(stream != NULL, "Verify stream value; expected: != NULL, got: %p", (void *)stream);
SDL_DestroyAudioStream(stream);
/* All source conversions with random conversion targets, allow 'null' conversions */
for (i = 0; i < g_numAudioFormats; i++) {
for (j = 0; j < g_numAudioChannels; j++) {
for (k = 0; k < g_numAudioFrequencies; k++) {
spec1.format = g_audioFormats[i];
spec1.channels = g_audioChannels[j];
spec1.freq = g_audioFrequencies[k];
ii = SDLTest_RandomIntegerInRange(0, g_numAudioFormats - 1);
jj = SDLTest_RandomIntegerInRange(0, g_numAudioChannels - 1);
kk = SDLTest_RandomIntegerInRange(0, g_numAudioFrequencies - 1);
spec2.format = g_audioFormats[ii];
spec2.channels = g_audioChannels[jj];
spec2.freq = g_audioFrequencies[kk];
stream = SDL_CreateAudioStream(&spec1, &spec2);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(format[%i]=%s(%i),channels[%i]=%i,freq[%i]=%i ==> format[%i]=%s(%i),channels[%i]=%i,freq[%i]=%i)",
i, g_audioFormatsVerbose[i], spec1.format, j, spec1.channels, k, spec1.freq, ii, g_audioFormatsVerbose[ii], spec2.format, jj, spec2.channels, kk, spec2.freq);
SDLTest_AssertCheck(stream != NULL, "Verify stream value; expected: != NULL, got: %p", (void *)stream);
if (stream == NULL) {
SDLTest_LogError("%s", SDL_GetError());
}
SDL_DestroyAudioStream(stream);
}
}
}
/* Restart audio again */
audioSetUp(NULL);
return TEST_COMPLETED;
}
/**
* Checks calls with invalid input to SDL_CreateAudioStream
*
* \sa SDL_CreateAudioStream
*/
static int audio_buildAudioStreamNegative(void *arg)
{
const char *error;
SDL_AudioStream *stream;
SDL_AudioSpec spec1;
SDL_AudioSpec spec2;
int i;
char message[256];
/* Valid format */
spec1.format = SDL_AUDIO_S8;
spec1.channels = 1;
spec1.freq = 22050;
spec2.format = SDL_AUDIO_S16LE;
spec2.channels = 2;
spec2.freq = 44100;
SDL_ClearError();
SDLTest_AssertPass("Call to SDL_ClearError()");
/* Invalid conversions */
for (i = 1; i < 64; i++) {
/* Valid format to start with */
spec1.format = SDL_AUDIO_S8;
spec1.channels = 1;
spec1.freq = 22050;
spec2.format = SDL_AUDIO_S16LE;
spec2.channels = 2;
spec2.freq = 44100;
SDL_ClearError();
SDLTest_AssertPass("Call to SDL_ClearError()");
/* Set various invalid format inputs */
SDL_strlcpy(message, "Invalid: ", 256);
if (i & 1) {
SDL_strlcat(message, " spec1.format", 256);
spec1.format = 0;
}
if (i & 2) {
SDL_strlcat(message, " spec1.channels", 256);
spec1.channels = 0;
}
if (i & 4) {
SDL_strlcat(message, " spec1.freq", 256);
spec1.freq = 0;
}
if (i & 8) {
SDL_strlcat(message, " spec2.format", 256);
spec2.format = 0;
}
if (i & 16) {
SDL_strlcat(message, " spec2.channels", 256);
spec2.channels = 0;
}
if (i & 32) {
SDL_strlcat(message, " spec2.freq", 256);
spec2.freq = 0;
}
SDLTest_Log("%s", message);
stream = SDL_CreateAudioStream(&spec1, &spec2);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(spec1 ==> spec2)");
SDLTest_AssertCheck(stream == NULL, "Verify stream value; expected: NULL, got: %p", (void *)stream);
error = SDL_GetError();
SDLTest_AssertPass("Call to SDL_GetError()");
SDLTest_AssertCheck(error != NULL && error[0] != '\0', "Validate that error message was not NULL or empty");
SDL_DestroyAudioStream(stream);
}
SDL_ClearError();
SDLTest_AssertPass("Call to SDL_ClearError()");
return TEST_COMPLETED;
}
/**
* Checks current audio status.
*
* \sa SDL_GetAudioDeviceStatus
*/
static int audio_getAudioStatus(void *arg)
{
return TEST_COMPLETED; /* no longer a thing in SDL3. */
}
/**
* Opens, checks current audio status, and closes a device.
*
* \sa SDL_GetAudioStatus
*/
static int audio_openCloseAndGetAudioStatus(void *arg)
{
return TEST_COMPLETED; /* not a thing in SDL3. */
}
/**
* Locks and unlocks open audio device.
*
* \sa SDL_LockAudioDevice
* \sa SDL_UnlockAudioDevice
*/
static int audio_lockUnlockOpenAudioDevice(void *arg)
{
return TEST_COMPLETED; /* not a thing in SDL3 */
}
/**
* Convert audio using various conversion structures
*
* \sa SDL_CreateAudioStream
*/
static int audio_convertAudio(void *arg)
{
SDL_AudioStream *stream;
SDL_AudioSpec spec1;
SDL_AudioSpec spec2;
int c;
char message[128];
int i, ii, j, jj, k, kk;
/* Iterate over bitmask that determines which parameters are modified in the conversion */
for (c = 1; c < 8; c++) {
SDL_strlcpy(message, "Changing:", 128);
if (c & 1) {
SDL_strlcat(message, " Format", 128);
}
if (c & 2) {
SDL_strlcat(message, " Channels", 128);
}
if (c & 4) {
SDL_strlcat(message, " Frequencies", 128);
}
SDLTest_Log("%s", message);
/* All source conversions with random conversion targets */
for (i = 0; i < g_numAudioFormats; i++) {
for (j = 0; j < g_numAudioChannels; j++) {
for (k = 0; k < g_numAudioFrequencies; k++) {
spec1.format = g_audioFormats[i];
spec1.channels = g_audioChannels[j];
spec1.freq = g_audioFrequencies[k];
/* Ensure we have a different target format */
do {
if (c & 1) {
ii = SDLTest_RandomIntegerInRange(0, g_numAudioFormats - 1);
} else {
ii = 1;
}
if (c & 2) {
jj = SDLTest_RandomIntegerInRange(0, g_numAudioChannels - 1);
} else {
jj = j;
}
if (c & 4) {
kk = SDLTest_RandomIntegerInRange(0, g_numAudioFrequencies - 1);
} else {
kk = k;
}
} while ((i == ii) && (j == jj) && (k == kk));
spec2.format = g_audioFormats[ii];
spec2.channels = g_audioChannels[jj];
spec2.freq = g_audioFrequencies[kk];
stream = SDL_CreateAudioStream(&spec1, &spec2);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(format[%i]=%s(%i),channels[%i]=%i,freq[%i]=%i ==> format[%i]=%s(%i),channels[%i]=%i,freq[%i]=%i)",
i, g_audioFormatsVerbose[i], spec1.format, j, spec1.channels, k, spec1.freq, ii, g_audioFormatsVerbose[ii], spec2.format, jj, spec2.channels, kk, spec2.freq);
SDLTest_AssertCheck(stream != NULL, "Verify stream value; expected: != NULL, got: %p", (void *)stream);
if (stream == NULL) {
SDLTest_LogError("%s", SDL_GetError());
} else {
Uint8 *dst_buf = NULL, *src_buf = NULL;
int dst_len = 0, src_len = 0, real_dst_len = 0;
int l = 64, m;
int src_framesize, dst_framesize;
int src_silence, dst_silence;
src_framesize = SDL_AUDIO_FRAMESIZE(spec1);
dst_framesize = SDL_AUDIO_FRAMESIZE(spec2);
src_len = l * src_framesize;
SDLTest_Log("Creating dummy sample buffer of %i length (%i bytes)", l, src_len);
src_buf = (Uint8 *)SDL_malloc(src_len);
SDLTest_AssertCheck(src_buf != NULL, "Check src data buffer to convert is not NULL");
if (src_buf == NULL) {
return TEST_ABORTED;
}
src_silence = SDL_GetSilenceValueForFormat(spec1.format);
SDL_memset(src_buf, src_silence, src_len);
dst_len = ((int)((((Sint64)l * spec2.freq) - 1) / spec1.freq) + 1) * dst_framesize;
dst_buf = (Uint8 *)SDL_malloc(dst_len);
SDLTest_AssertCheck(dst_buf != NULL, "Check dst data buffer to convert is not NULL");
if (dst_buf == NULL) {
return TEST_ABORTED;
}
real_dst_len = SDL_GetAudioStreamAvailable(stream);
SDLTest_AssertCheck(0 == real_dst_len, "Verify available (pre-put); expected: %i; got: %i", 0, real_dst_len);
/* Run the audio converter */
if (SDL_PutAudioStreamData(stream, src_buf, src_len) < 0 ||
SDL_FlushAudioStream(stream) < 0) {
return TEST_ABORTED;
}
real_dst_len = SDL_GetAudioStreamAvailable(stream);
SDLTest_AssertCheck(dst_len == real_dst_len, "Verify available (post-put); expected: %i; got: %i", dst_len, real_dst_len);
real_dst_len = SDL_GetAudioStreamData(stream, dst_buf, dst_len);
SDLTest_AssertCheck(dst_len == real_dst_len, "Verify result value; expected: %i; got: %i", dst_len, real_dst_len);
if (dst_len != real_dst_len) {
return TEST_ABORTED;
}
real_dst_len = SDL_GetAudioStreamAvailable(stream);
SDLTest_AssertCheck(0 == real_dst_len, "Verify available (post-get); expected: %i; got: %i", 0, real_dst_len);
dst_silence = SDL_GetSilenceValueForFormat(spec2.format);
for (m = 0; m < dst_len; ++m) {
if (dst_buf[m] != dst_silence) {
SDLTest_LogError("Output buffer is not silent");
return TEST_ABORTED;
}
}
SDL_DestroyAudioStream(stream);
/* Free converted buffer */
SDL_free(src_buf);
SDL_free(dst_buf);
}
}
}
}
}
return TEST_COMPLETED;
}
/**
* Opens, checks current connected status, and closes a device.
*
* \sa SDL_AudioDeviceConnected
*/
static int audio_openCloseAudioDeviceConnected(void *arg)
{
return TEST_COMPLETED; /* not a thing in SDL3. */
}
static double sine_wave_sample(const Sint64 idx, const Sint64 rate, const Sint64 freq, const double phase)
{
/* Using integer modulo to avoid precision loss caused by large floating
* point numbers. Sint64 is needed for the large integer multiplication.
* The integers are assumed to be non-negative so that modulo is always
* non-negative.
* sin(i / rate * freq * 2 * PI + phase)
* = sin(mod(i / rate * freq, 1) * 2 * PI + phase)
* = sin(mod(i * freq, rate) / rate * 2 * PI + phase) */
return SDL_sin(((double)(idx * freq % rate)) / ((double)rate) * (SDL_PI_D * 2) + phase);
}
/* 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.
*
* \sa https://wiki.libsdl.org/SDL_CreateAudioStream
* \sa https://wiki.libsdl.org/SDL_DestroyAudioStream
* \sa https://wiki.libsdl.org/SDL_PutAudioStreamData
* \sa https://wiki.libsdl.org/SDL_FlushAudioStream
* \sa https://wiki.libsdl.org/SDL_GetAudioStreamData
*/
static int audio_resampleLoss(void *arg)
{
/* Note: always test long input time (>= 5s from experience) in some test
* cases because an improper implementation may suffer from low resampling
* precision with long input due to e.g. doing subtraction with large floats. */
struct test_spec_t {
int time;
int freq;
double phase;
int rate_in;
int rate_out;
double signal_to_noise;
double max_error;
} test_specs[] = {
{ 50, 440, 0, 44100, 48000, 80, 0.0010 },
{ 50, 5000, SDL_PI_D / 2, 20000, 10000, 999, 0.0001 },
{ 50, 440, 0, 22050, 96000, 79, 0.0120 },
{ 50, 440, 0, 96000, 22050, 80, 0.0002 },
{ 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;) {
const struct test_spec_t *spec = &test_specs[spec_idx];
const int frames_in = spec->time * spec->rate_in;
const int frames_target = spec->time * spec->rate_out;
const int len_in = (frames_in * num_channels) * (int)sizeof(float);
const int len_target = (frames_target * num_channels) * (int)sizeof(float);
SDL_AudioSpec tmpspec1, tmpspec2;
Uint64 tick_beg = 0;
Uint64 tick_end = 0;
int i = 0;
int j = 0;
SDL_AudioStream *stream = NULL;
float *buf_in = NULL;
float *buf_out = NULL;
int len_out = 0;
double max_error = 0;
double sum_squared_error = 0;
double sum_squared_value = 0;
double signal_to_noise = 0;
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);
tmpspec1.format = SDL_AUDIO_F32;
tmpspec1.channels = num_channels;
tmpspec1.freq = spec->rate_in;
tmpspec2.format = SDL_AUDIO_F32;
tmpspec2.channels = num_channels;
tmpspec2.freq = spec->rate_out;
stream = SDL_CreateAudioStream(&tmpspec1, &tmpspec2);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(SDL_AUDIO_F32, %i, %i, SDL_AUDIO_F32, %i, %i)", num_channels, spec->rate_in, num_channels, spec->rate_out);
SDLTest_AssertCheck(stream != NULL, "Expected SDL_CreateAudioStream to succeed.");
if (stream == NULL) {
return TEST_ABORTED;
}
buf_in = (float *)SDL_malloc(len_in);
SDLTest_AssertCheck(buf_in != NULL, "Expected input buffer to be created.");
if (buf_in == NULL) {
SDL_DestroyAudioStream(stream);
return TEST_ABORTED;
}
for (i = 0; i < frames_in; ++i) {
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();
buf_out = (float *)SDL_malloc(len_target);
SDLTest_AssertCheck(buf_out != NULL, "Expected output buffer to be created.");
if (buf_out == NULL) {
SDL_DestroyAudioStream(stream);
return TEST_ABORTED;
}
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_free(buf_in);
if (len_out != len_target) {
SDL_DestroyAudioStream(stream);
return TEST_ABORTED;
}
tick_end = SDL_GetPerformanceCounter();
SDLTest_Log("Resampling used %f seconds.", ((double)(tick_end - tick_beg)) / SDL_GetPerformanceFrequency());
for (i = 0; i < frames_target; ++i) {
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);
max_error = SDL_max(max_error, error);
sum_squared_error += error * error;
sum_squared_value += target * target;
}
}
SDL_free(buf_out);
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 >= spec->signal_to_noise, "Conversion signal-to-noise ratio %f dB should be no less than %f dB.",
signal_to_noise, spec->signal_to_noise);
SDLTest_AssertCheck(max_error <= spec->max_error, "Maximum conversion error %f should be no more than %f.",
max_error, spec->max_error);
if (++num_channels > max_channels) {
num_channels = min_channels;
++spec_idx;
}
}
return TEST_COMPLETED;
}
/**
* Check accuracy converting between audio formats.
*
* \sa SDL_ConvertAudioSamples
*/
static int audio_convertAccuracy(void *arg)
{
static SDL_AudioFormat formats[] = { SDL_AUDIO_S8, SDL_AUDIO_U8, SDL_AUDIO_S16, SDL_AUDIO_S32 };
static const char* format_names[] = { "S8", "U8", "S16", "S32" };
int src_num = 65537 + 2048 + 48 + 256 + 100000;
int src_len = src_num * sizeof(float);
float* src_data = SDL_malloc(src_len);
int i, j;
SDLTest_AssertCheck(src_data != NULL, "Expected source buffer to be created.");
if (src_data == NULL) {
return TEST_ABORTED;
}
j = 0;
/* Generate a uniform range of floats between [-1.0, 1.0] */
for (i = 0; i < 65537; ++i) {
src_data[j++] = ((float)i - 32768.0f) / 32768.0f;
}
/* Generate floats close to 1.0 */
const float max_val = 16777216.0f;
for (i = 0; i < 1024; ++i) {
float f = (max_val + (float)(512 - i)) / max_val;
src_data[j++] = f;
src_data[j++] = -f;
}
for (i = 0; i < 24; ++i) {
float f = (max_val + (float)(3u << i)) / max_val;
src_data[j++] = f;
src_data[j++] = -f;
}
/* Generate floats far outside the [-1.0, 1.0] range */
for (i = 0; i < 128; ++i) {
float f = 2.0f + (float) i;
src_data[j++] = f;
src_data[j++] = -f;
}
/* Fill the rest with random floats between [-1.0, 1.0] */
for (i = 0; i < 100000; ++i) {
src_data[j++] = SDLTest_RandomSint32() / 2147483648.0f;
}
/* Shuffle the data for good measure */
for (i = src_num - 1; i > 0; --i) {
float f = src_data[i];
j = SDLTest_RandomIntegerInRange(0, i);
src_data[i] = src_data[j];
src_data[j] = f;
}
for (i = 0; i < SDL_arraysize(formats); ++i) {
SDL_AudioSpec src_spec, tmp_spec;
Uint64 convert_begin, convert_end;
Uint8 *tmp_data, *dst_data;
int tmp_len, dst_len;
int ret;
SDL_AudioFormat format = formats[i];
const char* format_name = format_names[i];
/* Formats with > 23 bits can represent every value exactly */
float min_delta = 1.0f;
float max_delta = -1.0f;
/* Subtract 1 bit to account for sign */
int bits = SDL_AUDIO_BITSIZE(format) - 1;
float target_max_delta = (bits > 23) ? 0.0f : (1.0f / (float)(1 << bits));
float target_min_delta = -target_max_delta;
src_spec.format = SDL_AUDIO_F32;
src_spec.channels = 1;
src_spec.freq = 44100;
tmp_spec.format = format;
tmp_spec.channels = 1;
tmp_spec.freq = 44100;
convert_begin = SDL_GetPerformanceCounter();
tmp_data = NULL;
tmp_len = 0;
ret = SDL_ConvertAudioSamples(&src_spec, (const Uint8*) src_data, src_len, &tmp_spec, &tmp_data, &tmp_len);
SDLTest_AssertCheck(ret == 0, "Expected SDL_ConvertAudioSamples(F32->%s) to succeed", format_name);
if (ret != 0) {
SDL_free(src_data);
return TEST_ABORTED;
}
dst_data = NULL;
dst_len = 0;
ret = SDL_ConvertAudioSamples(&tmp_spec, tmp_data, tmp_len, &src_spec, &dst_data, &dst_len);
SDLTest_AssertCheck(ret == 0, "Expected SDL_ConvertAudioSamples(%s->F32) to succeed", format_name);
if (ret != 0) {
SDL_free(tmp_data);
SDL_free(src_data);
return TEST_ABORTED;
}
convert_end = SDL_GetPerformanceCounter();
SDLTest_Log("Conversion via %s took %f seconds.", format_name, ((double)(convert_end - convert_begin)) / SDL_GetPerformanceFrequency());
SDL_free(tmp_data);
for (j = 0; j < src_num; ++j) {
float x = src_data[j];
float y = ((float*)dst_data)[j];
float d = SDL_clamp(x, -1.0f, 1.0f) - y;
min_delta = SDL_min(min_delta, d);
max_delta = SDL_max(max_delta, d);
}
SDLTest_AssertCheck(min_delta >= target_min_delta, "%s has min delta of %+f, should be >= %+f", format_name, min_delta, target_min_delta);
SDLTest_AssertCheck(max_delta <= target_max_delta, "%s has max delta of %+f, should be <= %+f", format_name, max_delta, target_max_delta);
SDL_free(dst_data);
}
SDL_free(src_data);
return TEST_COMPLETED;
}
/**
* 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_F32;
spec1.channels = 1;
spec1.freq = 20000;
spec2.format = SDL_AUDIO_F32;
spec2.channels = 1;
spec2.freq = 40000;
spec3.format = SDL_AUDIO_F32;
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 */
static const SDLTest_TestCaseReference audioTest1 = {
audio_enumerateAndNameAudioDevices, "audio_enumerateAndNameAudioDevices", "Enumerate and name available audio devices (output and capture)", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest2 = {
audio_enumerateAndNameAudioDevicesNegativeTests, "audio_enumerateAndNameAudioDevicesNegativeTests", "Negative tests around enumeration and naming of audio devices.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest3 = {
audio_printAudioDrivers, "audio_printAudioDrivers", "Checks available audio driver names.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest4 = {
audio_printCurrentAudioDriver, "audio_printCurrentAudioDriver", "Checks current audio driver name with initialized audio.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest5 = {
audio_buildAudioStream, "audio_buildAudioStream", "Builds various audio conversion structures.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest6 = {
audio_buildAudioStreamNegative, "audio_buildAudioStreamNegative", "Checks calls with invalid input to SDL_CreateAudioStream", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest7 = {
audio_getAudioStatus, "audio_getAudioStatus", "Checks current audio status.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest8 = {
audio_openCloseAndGetAudioStatus, "audio_openCloseAndGetAudioStatus", "Opens and closes audio device and get audio status.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest9 = {
audio_lockUnlockOpenAudioDevice, "audio_lockUnlockOpenAudioDevice", "Locks and unlocks an open audio device.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest10 = {
audio_convertAudio, "audio_convertAudio", "Convert audio using available formats.", TEST_ENABLED
};
/* TODO: enable test when SDL_AudioDeviceConnected has been implemented. */
static const SDLTest_TestCaseReference audioTest11 = {
audio_openCloseAudioDeviceConnected, "audio_openCloseAudioDeviceConnected", "Opens and closes audio device and get connected status.", TEST_DISABLED
};
static const SDLTest_TestCaseReference audioTest12 = {
audio_quitInitAudioSubSystem, "audio_quitInitAudioSubSystem", "Quit and re-init audio subsystem.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest13 = {
audio_initQuitAudio, "audio_initQuitAudio", "Init and quit audio drivers directly.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest14 = {
audio_initOpenCloseQuitAudio, "audio_initOpenCloseQuitAudio", "Cycle through init, open, close and quit with various audio specs.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest15 = {
audio_pauseUnpauseAudio, "audio_pauseUnpauseAudio", "Pause and Unpause audio for various audio specs while testing callback.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest16 = {
audio_resampleLoss, "audio_resampleLoss", "Check signal-to-noise ratio and maximum error of audio resampling.", TEST_ENABLED
};
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, &audioTest18, NULL
};
/* Audio test suite (global) */
SDLTest_TestSuiteReference audioTestSuite = {
"Audio",
audioSetUp,
audioTests,
audioTearDown
};
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