| /** |
| * Original code: automated SDL audio test written by Edgar Simo "bobbens" |
| * New/updated tests: aschiffler at ferzkopp dot net |
| */ |
| |
| /* quiet windows compiler warnings */ |
| #if defined(_MSC_VER) && !defined(_CRT_SECURE_NO_WARNINGS) |
| #define _CRT_SECURE_NO_WARNINGS |
| #endif |
| |
| #include <math.h> |
| #include <stdio.h> |
| |
| #include <SDL3/SDL.h> |
| #include <SDL3/SDL_test.h> |
| #include "testautomation_suites.h" |
| |
| /* ================= Test Case Implementation ================== */ |
| |
| /* Fixture */ |
| |
| static void 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) { |
| 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) || |
| !SDL_FlushAudioStream(stream)) { |
| return TEST_ABORTED; |
| } |
| |
| real_dst_len = SDL_GetAudioStreamAvailable(stream); |
| SDLTest_AssertCheck(dst_len == real_dst_len, "Verify available (post-put); expected: %i; got: %i", dst_len, real_dst_len); |
| |
| real_dst_len = SDL_GetAudioStreamData(stream, dst_buf, dst_len); |
| SDLTest_AssertCheck(dst_len == real_dst_len, "Verify result value; expected: %i; got: %i", dst_len, real_dst_len); |
| if (dst_len != real_dst_len) { |
| return TEST_ABORTED; |
| } |
| |
| real_dst_len = SDL_GetAudioStreamAvailable(stream); |
| SDLTest_AssertCheck(0 == real_dst_len, "Verify available (post-get); expected: %i; got: %i", 0, real_dst_len); |
| |
| dst_silence = SDL_GetSilenceValueForFormat(spec2.format); |
| |
| for (m = 0; m < dst_len; ++m) { |
| if (dst_buf[m] != dst_silence) { |
| SDLTest_LogError("Output buffer is not silent"); |
| return TEST_ABORTED; |
| } |
| } |
| |
| SDL_DestroyAudioStream(stream); |
| /* Free converted buffer */ |
| SDL_free(src_buf); |
| SDL_free(dst_buf); |
| } |
| } |
| } |
| } |
| } |
| |
| return TEST_COMPLETED; |
| } |
| |
| /** |
| * Opens, checks current connected status, and closes a device. |
| * |
| * \sa SDL_AudioDeviceConnected |
| */ |
| static int 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); |
| 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); |
| return TEST_ABORTED; |
| } |
| |
| tick_end = SDL_GetPerformanceCounter(); |
| SDLTest_Log("Resampling used %f seconds.", ((double)(tick_end - tick_beg)) / SDL_GetPerformanceFrequency()); |
| |
| for (i = 0; i < frames_target; ++i) { |
| const double target = sine_wave_sample(i, spec->rate_out, spec->freq, spec->phase); |
| for (j = 0; j < num_channels; ++j) { |
| const float output = *(buf_out + (i * num_channels) + j); |
| const double error = SDL_fabs(target - output); |
| max_error = SDL_max(max_error, error); |
| sum_squared_error += error * error; |
| sum_squared_value += target * target; |
| } |
| } |
| SDL_free(buf_out); |
| signal_to_noise = 10 * SDL_log10(sum_squared_value / sum_squared_error); /* decibel */ |
| SDLTest_AssertCheck(ISFINITE(sum_squared_value), "Sum of squared target should be finite."); |
| SDLTest_AssertCheck(ISFINITE(sum_squared_error), "Sum of squared error should be finite."); |
| /* Infinity is theoretically possible when there is very little to no noise */ |
| SDLTest_AssertCheck(!ISNAN(signal_to_noise), "Signal-to-noise ratio should not be NaN."); |
| SDLTest_AssertCheck(ISFINITE(max_error), "Maximum conversion error should be finite."); |
| SDLTest_AssertCheck(signal_to_noise >= spec->signal_to_noise, "Conversion signal-to-noise ratio %f dB should be no less than %f dB.", |
| signal_to_noise, spec->signal_to_noise); |
| SDLTest_AssertCheck(max_error <= spec->max_error, "Maximum conversion error %f should be no more than %f.", |
| max_error, spec->max_error); |
| |
| if (++num_channels > max_channels) { |
| num_channels = min_channels; |
| ++spec_idx; |
| } |
| } |
| |
| return TEST_COMPLETED; |
| } |
| |
| /** |
| * Check accuracy converting between audio formats. |
| * |
| * \sa SDL_ConvertAudioSamples |
| */ |
| static int 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 |
| }; |
