/** * 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 #include #include #include #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 = -1; /* Test case functions */ /** * \brief 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; } /** * \brief 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; /* 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 */ /* 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; } /** * \brief 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; /* 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 */ /* 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; 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; } /** * \brief 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; /* 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 */ /* 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; } /** * \brief 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; } /** * \brief 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. */ } /** * \brief 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; } /** * \brief 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_S16, SDL_AUDIO_S32LE, SDL_AUDIO_S32BE, SDL_AUDIO_S32, SDL_AUDIO_F32LE, SDL_AUDIO_F32BE, SDL_AUDIO_F32 }; static const char *g_audioFormatsVerbose[] = { "SDL_AUDIO_S8", "SDL_AUDIO_U8", "SDL_AUDIO_S16LE", "SDL_AUDIO_S16BE", "SDL_AUDIO_S16", "SDL_AUDIO_S32LE", "SDL_AUDIO_S32BE", "SDL_AUDIO_S32", "SDL_AUDIO_F32LE", "SDL_AUDIO_F32BE", "SDL_AUDIO_F32" }; 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)); /** * \brief 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; /* 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); } } } return TEST_COMPLETED; } /** * \brief 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; } /** * \brief Checks current audio status. * * \sa SDL_GetAudioDeviceStatus */ static int audio_getAudioStatus(void *arg) { return TEST_COMPLETED; /* no longer a thing in SDL3. */ } /** * \brief 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. */ } /** * \brief 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 */ } /** * \brief 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; } /** * \brief 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); } /** * \brief 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, 0.0009 }, { 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; int ret = 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, 1, %i, SDL_AUDIO_F32, 1, %i)", spec->rate_in, 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(); 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) { SDL_DestroyAudioStream(stream); 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_target, len_out); SDL_DestroyAudioStream(stream); if (len_out > len_target) { SDL_free(buf_out); 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; } /** * \brief 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; } /** * \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_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 };