From 6aaedb813fa11ba0679c3051bc2eb28646b9506c Mon Sep 17 00:00:00 2001 From: 3gg <3gg@shellblade.net> Date: Sat, 30 Aug 2025 16:53:58 -0700 Subject: Update to SDL3 --- src/contrib/SDL-3.2.20/test/testautomation_audio.c | 1559 ++++++++++++++++++++ 1 file changed, 1559 insertions(+) create mode 100644 src/contrib/SDL-3.2.20/test/testautomation_audio.c (limited to 'src/contrib/SDL-3.2.20/test/testautomation_audio.c') diff --git a/src/contrib/SDL-3.2.20/test/testautomation_audio.c b/src/contrib/SDL-3.2.20/test/testautomation_audio.c new file mode 100644 index 0000000..7c141b3 --- /dev/null +++ b/src/contrib/SDL-3.2.20/test/testautomation_audio.c @@ -0,0 +1,1559 @@ +/** + * 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 SDLCALL audioSetUp(void **arg) +{ + /* Start SDL audio subsystem */ + bool ret = SDL_InitSubSystem(SDL_INIT_AUDIO); + SDLTest_AssertPass("Call to SDL_InitSubSystem(SDL_INIT_AUDIO)"); + SDLTest_AssertCheck(ret == true, "Check result from SDL_InitSubSystem(SDL_INIT_AUDIO)"); + if (!ret) { + SDLTest_LogError("%s", SDL_GetError()); + } +} + +static void SDLCALL 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 SDLCALL 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 SDLCALL 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_HINT_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 == true, "Validate result value; expected: true 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_HINT_AUDIO_DRIVER, audioDriver); + result = SDL_InitSubSystem(SDL_INIT_AUDIO); + SDLTest_AssertPass("Call to SDL_AudioInit(NULL)"); + SDLTest_AssertCheck(result == true, "Validate result value; expected: true 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 SDLCALL 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_HINT_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 == true, "Validate result value; expected: true got: %d", result); + + /* Set spec */ + SDL_zero(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_PLAYBACK, &desired); + if (k == 0) { + g_audio_id = result; + } + SDLTest_AssertPass("Call to SDL_OpenAudioDevice(SDL_AUDIO_DEVICE_DEFAULT_PLAYBACK, 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 SDLCALL 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_HINT_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 == true, "Validate result value; expected: true got: %d", result); + + /* Set spec */ + SDL_zero(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_PLAYBACK, &desired); + result = g_audio_id; + SDLTest_AssertPass("Call to SDL_OpenAudioDevice(SDL_AUDIO_DEVICE_DEFAULT_PLAYBACK, 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 (playback and recording). + * + * \sa SDL_GetNumAudioDevices + * \sa SDL_GetAudioDeviceName + */ +static int SDLCALL audio_enumerateAndNameAudioDevices(void *arg) +{ + int t; + int i, n; + const char *name; + SDL_AudioDeviceID *devices; + + /* Iterate over types: t=0 playback device, t=1 recording device */ + for (t = 0; t < 2; t++) { + /* Get number of devices. */ + devices = (t) ? SDL_GetAudioRecordingDevices(&n) : SDL_GetAudioPlaybackDevices(&n); + SDLTest_AssertPass("Call to SDL_GetAudio%sDevices(%i)", (t) ? "Recording" : "Playback", t); + SDLTest_Log("Number of %s devices < 0, reported as %i", (t) ? "recording" : "playback", 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(devices); + } + + return TEST_COMPLETED; +} + +/** + * Negative tests around enumeration and naming of audio devices. + * + * \sa SDL_GetNumAudioDevices + * \sa SDL_GetAudioDeviceName + */ +static int SDLCALL 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 SDLCALL 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 SDLCALL 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 SDL_AudioFormat g_invalidAudioFormats[] = { + (SDL_AudioFormat)SDL_DEFINE_AUDIO_FORMAT(SDL_AUDIO_MASK_SIGNED, SDL_AUDIO_MASK_BIG_ENDIAN, SDL_AUDIO_MASK_FLOAT, SDL_AUDIO_MASK_BITSIZE) +}; +static const char *g_invalidAudioFormatsVerbose[] = { + "SDL_AUDIO_UNKNOWN" +}; +static const int g_numAudioFormats = SDL_arraysize(g_audioFormats); +static const int g_numInvalidAudioFormats = SDL_arraysize(g_invalidAudioFormats); +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)); + +/** + * Call to SDL_GetAudioFormatName + * + * \sa SDL_GetAudioFormatName + */ +static int SDLCALL audio_getAudioFormatName(void *arg) +{ + const char *error; + int i; + SDL_AudioFormat format; + const char *result; + + /* audio formats */ + for (i = 0; i < g_numAudioFormats; i++) { + format = g_audioFormats[i]; + SDLTest_Log("Audio Format: %s (%d)", g_audioFormatsVerbose[i], format); + + /* Get name of format */ + result = SDL_GetAudioFormatName(format); + SDLTest_AssertPass("Call to SDL_GetAudioFormatName()"); + SDLTest_AssertCheck(result != NULL, "Verify result is not NULL"); + if (result != NULL) { + SDLTest_AssertCheck(result[0] != '\0', "Verify result is non-empty"); + SDLTest_AssertCheck(SDL_strcmp(result, g_audioFormatsVerbose[i]) == 0, + "Verify result text; expected: %s, got %s", g_audioFormatsVerbose[i], result); + } + } + + /* Negative cases */ + + /* Invalid Formats */ + SDL_ClearError(); + SDLTest_AssertPass("Call to SDL_ClearError()"); + for (i = 0; i < g_numInvalidAudioFormats; i++) { + format = g_invalidAudioFormats[i]; + result = SDL_GetAudioFormatName(format); + SDLTest_AssertPass("Call to SDL_GetAudioFormatName(%d)", format); + SDLTest_AssertCheck(result != NULL, "Verify result is not NULL"); + if (result != NULL) { + SDLTest_AssertCheck(result[0] != '\0', + "Verify result is non-empty; got: %s", result); + SDLTest_AssertCheck(SDL_strcmp(result, g_invalidAudioFormatsVerbose[i]) == 0, + "Validate name is UNKNOWN, expected: '%s', got: '%s'", g_invalidAudioFormatsVerbose[i], result); + } + error = SDL_GetError(); + SDLTest_AssertPass("Call to SDL_GetError()"); + SDLTest_AssertCheck(error == NULL || error[0] == '\0', "Validate that error message is empty"); + } + + return TEST_COMPLETED; +} + +/** + * Builds various audio conversion structures + * + * \sa SDL_CreateAudioStream + */ +static int SDLCALL audio_buildAudioStream(void *arg) +{ + SDL_AudioStream *stream; + SDL_AudioSpec spec1; + SDL_AudioSpec spec2; + int i, ii, j, jj, k, kk; + + SDL_zero(spec1); + SDL_zero(spec2); + + /* 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", 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", 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", 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 SDLCALL audio_buildAudioStreamNegative(void *arg) +{ + const char *error; + SDL_AudioStream *stream; + SDL_AudioSpec spec1; + SDL_AudioSpec spec2; + int i; + char message[256]; + + SDL_zero(spec1); + SDL_zero(spec2); + + /* 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", 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 SDLCALL 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 SDLCALL 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 SDLCALL audio_lockUnlockOpenAudioDevice(void *arg) +{ + return TEST_COMPLETED; /* not a thing in SDL3 */ +} + +/** + * Convert audio using various conversion structures + * + * \sa SDL_CreateAudioStream + */ +static int SDLCALL 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; + + SDL_zero(spec1); + SDL_zero(spec2); + + /* 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", 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) { + SDL_DestroyAudioStream(stream); + 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) { + SDL_DestroyAudioStream(stream); + SDL_free(src_buf); + 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) || + !SDL_FlushAudioStream(stream)) { + SDL_DestroyAudioStream(stream); + SDL_free(src_buf); + SDL_free(dst_buf); + 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) { + SDL_DestroyAudioStream(stream); + SDL_free(src_buf); + SDL_free(dst_buf); + 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"); + SDL_DestroyAudioStream(stream); + SDL_free(src_buf); + SDL_free(dst_buf); + 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 SDLCALL 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) { + return -1; + } + + 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) { + return -1; + } + + 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) { + return -1; + } + + 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 : -1; + } + + 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 -1; + } + + src_frame_size = SDL_AUDIO_FRAMESIZE(src_spec); + dst_frame_size = SDL_AUDIO_FRAMESIZE(dst_spec); + + while ((total_in < srclen) || (total_out < dstlen)) { + /* Make sure we put in more than the padding frames so we get non-zero output */ + const int RESAMPLER_MAX_PADDING_FRAMES = 7; /* Should match RESAMPLER_MAX_PADDING_FRAMES in SDL */ + int to_put = SDLTest_RandomIntegerInRange(RESAMPLER_MAX_PADDING_FRAMES + 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 < 0) { + 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 : -1; + } + } + } + + 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 SDLCALL 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); + const int max_target = len_target * 2; + + 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; + + SDL_zero(tmpspec1); + SDL_zero(tmpspec2); + + 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(max_target); + SDLTest_AssertCheck(buf_out != NULL, "Expected output buffer to be created."); + if (buf_out == NULL) { + SDL_DestroyAudioStream(stream); + SDL_free(buf_in); + return TEST_ABORTED; + } + + len_out = convert_audio_chunks(stream, buf_in, len_in, buf_out, max_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); + 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_DestroyAudioStream(stream); + 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 SDLCALL 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_zero(src_spec); + SDL_zero(tmp_spec); + + 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 == true, "Expected SDL_ConvertAudioSamples(F32->%s) to succeed", format_name); + if (!ret) { + 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 == true, "Expected SDL_ConvertAudioSamples(%s->F32) to succeed", format_name); + if (!ret) { + 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 SDLCALL 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 result = 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; + + SDL_zero(spec1); + SDL_zero(spec2); + SDL_zero(spec3); + + 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; + } + + result = SDL_SetAudioStreamFormat(stream, &spec1, &spec3); + if (!SDLTest_AssertCheck(result == true, "Expected SDL_SetAudioStreamFormat(spec1, spec3) to succeed")) { + goto cleanup; + } + + result = SDL_GetAudioStreamAvailable(stream); + if (!SDLTest_AssertCheck(result == 0, "Expected SDL_GetAudioStreamAvailable return 0")) { + goto cleanup; + } + + result = SDL_PutAudioStreamData(stream, buffer_1, length_1); + if (!SDLTest_AssertCheck(result == true, "Expected SDL_PutAudioStreamData(buffer_1) to succeed")) { + goto cleanup; + } + + result = SDL_FlushAudioStream(stream); + if (!SDLTest_AssertCheck(result == true, "Expected SDL_FlushAudioStream to succeed")) { + goto cleanup; + } + + result = SDL_SetAudioStreamFormat(stream, &spec2, &spec3); + if (!SDLTest_AssertCheck(result == true, "Expected SDL_SetAudioStreamFormat(spec2, spec3) to succeed")) { + goto cleanup; + } + + result = SDL_PutAudioStreamData(stream, buffer_2, length_2); + if (!SDLTest_AssertCheck(result == true, "Expected SDL_PutAudioStreamData(buffer_1) to succeed")) { + goto cleanup; + } + + result = SDL_FlushAudioStream(stream); + if (!SDLTest_AssertCheck(result == true, "Expected SDL_FlushAudioStream to succeed")) { + goto cleanup; + } + + result = SDL_GetAudioStreamAvailable(stream); + if (!SDLTest_AssertCheck(result == length_3, "Expected SDL_GetAudioStreamAvailable to return %i, got %i", length_3, result)) { + goto cleanup; + } + + result = SDL_GetAudioStreamData(stream, buffer_3, length_3); + if (!SDLTest_AssertCheck(result == length_3, "Expected SDL_GetAudioStreamData to return %i, got %i", length_3, result)) { + goto cleanup; + } + + result = SDL_GetAudioStreamAvailable(stream); + if (!SDLTest_AssertCheck(result == 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 audioTestGetAudioFormatName = { + audio_getAudioFormatName, "audio_getAudioFormatName", "Call to SDL_GetAudioFormatName", TEST_ENABLED +}; + +static const SDLTest_TestCaseReference audioTest1 = { + audio_enumerateAndNameAudioDevices, "audio_enumerateAndNameAudioDevices", "Enumerate and name available audio devices (playback and recording)", 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[] = { + &audioTestGetAudioFormatName, + &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 +}; -- cgit v1.2.3