tomato-testing/test/testautomation_audio.c
Green Sky 4d48f9d237 Squashed 'external/sdl/SDL/' changes from b8d91252c6..ec0042081e
ec0042081e Add .gitattributes file
a5d9db0cd0 cmake: build tests for UWP
b7889a7389 winrt: use windowsio in non-libc mode
ea8757a748 Make testaudiostreamdynamicresample compatible with emscripten
1a7a74fb2e cmake: build emscripten tests as html page
64d570f027 Add minimal http server for emscripten test apps
8e898c4a21 SDL_test does not parse --samples argument
91cd5478be audio: Fix resampler overflowing input buffer.
f290c85b22 testaudiocapture: Make sure we convert captured audio to output format.
b75c751dfc rwlock: Make generic implmentations work on single-threaded platforms.
80850af7ce The controller update complete events are no longer disabled by default
3f486224a9 Fixed refresh rate calculation for KMSDRM
342ec51131 Fix overflow when doing SDL_sscanf("%hd", ...)
9129e1d557 Fixed crash when setting the default cursor twice
8e99a4f4f5 Undo variable rename
be67f0de10 Fixed crashes related to the default cursor on WinRT and KMSDRM
94b3f78c44 Fix out of bound read of 'has_hat' array
94f48f19b0 Use more specific build destinations when creating an xcframework
dabd45997e Back out change supporting multiple names for binding elements
efe15588d5 Relabel back paddles as left or right
be884f0c95 ci: disable visionos.yml by renaming the file
ac094d00f5 ci: add workflow_dispatch event to visionos workflow
9be9e2292b build: Consistently use pathlib APIs in cmake/xxd.py
a9f6950657 Fixed deadlock shutting down Android sensors
d9f09e77f2 Actually make the sensors magical!
690eae7d22 Implement visionOS support
e385d6da0a Fixed build warning
6b93e788fa Improved sensor thread-safety
4ee0e5a984 Fixed thread-safety warnings
12deed91f8 Added information on how to enable thread-safety analysis
5735d2b03b coreaudio: Fixed assertion when device fails/quits mid-iteration.
1022fd6e04 testaudio: the test framework opens an audio device at startup; close it.
0714da37a4 audio: Fix audio stream callback calculations when future buffer has space.
917e036f6f MSVC has __declspec(deprecated)
279ff8909f Changed example code to avoid potential divide by zero
8a1afc9b10 Fixed Android not sending controller event timestamps
463c456b98 Fill the correct member with the joystick ID in SDL_EVENT_JOYSTICK_UPDATE_COMPLETE
55cf1abaa6 test: Don't flag testsurround as suitable for non-interactive use
a2d594269c Fixed pixel format compatibility with SDL2
79a190aa23 Fixed setting invalid bpp for FOURCC formats in SDL_GetMasksForPixelFormatEnum()
8fdebdd3e0 Sync SDL3 wiki -> header
b903ccf945 SDL_rwops read/write functions return size_t again
c03f5b4b69 Fixed rounding up in SDL_PrintFloat
75a020aa6b Only query serial number and firmware versions from Sony PS5 controllers
fa189d302e Added the Victrix Pro FS for PS4/PS5 to the controller list
26205b659d Fixed PS4/PS5 touchpad for third party controllers
6af0448af9 include: fixed a typo in SDL_RenderGetMetalCommandEncoder docs.
f3cb46b083 SDL_thread.h: do not conflict with sdl2-compat::sdl3_include_wrapper.h
080b1dfbdb Revert "Improved fallback for SDL_COMPILE_TIME_ASSERT() (thanks @icculus!)"
9d453daa23 Improved fallback for SDL_COMPILE_TIME_ASSERT() (thanks @icculus!)
1fb2419882 Removed reference to renamed function
e7d56dd0b2 audio: Renamed new API SDL_UnpauseAudioDevice to SDL_ResumeAudioDevice.
2b0c0f5b6b Don't pass NULL to strncmp
778e8185cd Fix size of memcpy in SDL_AudioDeviceFormatChangedAlreadyLocked And add diagnostic that allows to find this kind of issue in clang-tidy
4bb426abad Sync SDL3 wiki -> header
3a752ce650 Reapply "Changed 'freesrc' parameter from int to SDL_bool" to SDL_wave.c
2ba03b4db0 fix build after previous commit.
0026adffd4 apply force_align_arg_pointer attribute to correct version of SDL_RunApp
77446e2029 Unaligned stacks on i686-w64-mingw32 may lead to crashes
d3bcc3f057 Fixed build errors when OpenGL isn't enabled
35ad68e126 Sync SDL3 wiki -> header
70323a8350 Add a function to display the system menu for a window
be5f66c84e testaudio: Fixed soundboard icon, which had a colorkey issue.
c0a88930bf Sync SDL3 wiki -> header
18c59cc969 Merge the SDL3 audio subsystem redesign!
99b0e31788 The Steam Controller D-Pad is only pressed when the button is pressed down
103073d694 Set NSBluetoothAlwaysUsageDescription for testcontroller
ca02bb6c8c We don't need testdropfile-Info.plist
e063f662e9 Enable the controller update complete events
06bea1eb55 Added a gamepad mapping for the G-Shark GS-GP702
5ca3c50bf0 testaudio: Fix compiler warning.
1b1f02c5aa testaudio: Apparently compilers don't like this possibly being NULL now...?
2de9253b6c test: Added testaudio
fb3ab3f113 SDL_video.c: move ngage video before offscreen.
843572d993 Don't mark autorelease keys as virtual
648de4f9b8 Fixed duplicate key press/release events on iOS
a8abe612ed Only pass keypresses up the responder chain when text input is active
c3288d113e Synchronize on-screen keyboard state with text input active state
5fb92ef2f7 Fixed whitespace
f5ea6ae18d Revert "Stop beep when running iOS apps on ARM-based Macs"
546508b9b4 Allow test programs to run at full resolution on iPads
68a4bb01e0 Allocate displays as an array of pointers instead of an array of objects
07578fde3d Fixed crash if a display is enumerated twice
a509771a87 fix ios CI workflow after commit e4460e897f
72ce76905a The scheme isn't always the same as the framework name (e.g. xmp_lite vs xmp-lite)
e4460e897f By default Xcode expects the framework target name to be the name of the project.
ac683773dc Added missing tests to the "All" target
7dd56eaafe Removed unnecessary reference to testoverlay-Info.plist
e1c7f524ef Reduce the number of times SDL3 is duplicated in the xcframework script
65538011ca Make Xcode targets more specific
efe114c300 Revert "Renamed the xcframework target from "SDL.xcframework" to "xcframework""
73ed1d21a9 Renamed the xcframework target from "SDL.xcframework" to "xcframework"
76b4d8a0d8 Build the Framework instead of a static library for iOS and tvOS
d1bf979160 Removed unnecessary setting from the "Create DMG" target
c94cb3a5d8 Simplified the Xcode project to a single Framework target
ea60474c65 cmake: don't build SDL3-static Apple framework
8f00d7856d Sync SDL3 wiki -> header
d4a867a256 Rename SDL_GetPath to SDL_GetUserFolder
71099149b8 Fall back to Xlib if XRandR isn't available
b7f32f74ce Note the removal of the SDL_RENDERER_TARGETTEXTURE flag
0eda582160 testaudiostreamdynamicresample: Load sample.wav correctly.
87eae9a0a1 aaudio: We need a mixbuf on capture devices, too.
fb68e84646 wayland: Fix memory leaks
b0edd23c00 testsurround: Log available audio output devices at the start.
ae3090c387 androidaudio: Move Init/bootstrap code to bottom of source code.
18fc0db9e5 aaudio: Rearranged source code to match other backends.
2507c1d68b aaudio: Disconnect playing devices if error callback fires.
32a3fc3783 aaudio: Use the callback interface.
b49ce86765 audio: Fixed compiler warning on Android NDK.
1c074e8d97 android: Fixed audio device detection.
82ce05ad01 pulseaudio: Be more aggressive with hotplug thread synchronization.
5cbdf1168e androidaudio: Fixed incorrect JNI call (thanks, @madebr!)
660054f3dc include: Correct comment about audio device hotplug events.
ab68428a64 aaudio: Fixed for older SDKs and Android releases.
5ff87c6d4a android: Reworked audio backends for SDL3 audio API.
54af687210 testautomation_audio.c: Patched to compile.  :/
5e82090662 testautomation_audio.c: Apparently we aren't updating test code for C99 atm.
7f4488f625 wasapi: More fixes for Clang warnings.
29a0c689c9 wasapi: Patched to compile with Clang.
4aa95c21bc pspaudio: Patched to compile.
9a2a0a1463 ps2audio: Delete errant character that got inserted before previous commit.
2c578bd0d5 qnxaudio: Rewrite for SDL3 audio APIs.
455eef4cd9 audio: Use AtomicAdd for device counts, don't treat as a refcount.
095ea57f94 pspaudio: Patched to compile.
d7cf63db67 ps2audio: Patched to compile.
027b9e8787 coreaudio: (maybe) patched to compile on iOS.
4836c2db07 pspaudio: Patched to compile.
86ca412436 n3dsaudio: Patched to compile.
66bcee2ca9 testaudiostreamdynamicresample.c: Fixed MSVC compiler warning.
dbf993d358 vitaaudio: patched to compile.
5707e14716 audio: Fix up some things that broke when rebasing the branch against main.
6567285eae SDL_migration.cocci: Fix up SDL_(Pause|Unpause)Audio.
0b6255551e test: Fixed incorrect SDL_OpenAudioDevice call in testautomation.
107fd941cd vitaaudio: Clean up correctly in CloseDevice.
9fa4a6ef87 netbsdaudio: Minor fix.
b0d89868c6 n3dsaudio: Updated (but untested!) for SDL3 audio API.
ba27176106 vitaaudio: Untested attempt to move Vita audio to SDL3's audio API.
0b58e96d9e wasapi: Patched WinRT to compile.
d6b4f48488 visualc: Turn on multiprocessor compilation.
c58d95c343 wasapi: Reworked for new SDL3 audio API, other win32 fixes.
dc04f85646 audio: whoops, that should be an int.
be0dc630b7 audio: Fixed incorrect assertion
77b3fb06ee directsound: First shot at updating for SDL3 audio API.
4399b71715 audio: Generalize how backends can lookup an SDL_AudioDevice.
2fb122fe46 audio: backends now "find" instead of "obtain" devices by handle.
c3f5a5fc72 dummyaudio: SDL3ify style
7d65ff86e2 diskaudio: Adjusted for later SDL3 audio API redesign changes.
4ba9c2eade dummyaudio: Configurable delay, other SDL3 API fixes.
fb395d3ad7 sndio: Updated to the SDL3 audio API.
1a55282051 dsp: Some minor logic fixes
6bc85577d7 netbsdaudio: Updated for SDL3 audio API.
0f6e59312b netbsdaudio: Removed email address from source code.
51ae78c0af haikuaudio: Updated for SDL3 audio API.
fc7ed18ca1 emscriptenaudio: don't forget to finalize the audio thread
4233c41ce2 pulseaudio: Removed unnecessary variable.
a0528cd5ed emscriptenaudio: Updated for SDL3 audio API.
79cc29ba35 wave: Don't check if format->channels > INT_MAX, it's a Uint16.
1bfe97c235 pspaudio: Updated for SDL3 audio API.
121a2dce15 audio: Make sure `device->hidden` is NULL after CloseDevice
3d6ba0cafd ps2audio: Removed free of buffer that hasn't been allocated yet.
00ed6f8827 test: Fixed compiler warnings for unused vars.
6f12f68ec9 ps2audio: SDL3ified the style
4993743a02 ps2audio: Renamed `_this` to `device`
74568cdb2b ps2audio: Updated (but untested) for SDL3 audio API.
c83b68ef26 jack: renamed `_this` to `device`.
3f4f004794 audio: Remove an assertion that no longer makes sense.
86243b2589 jack: Use ProvidesOwnCallbackThread.
18906a32b8 jack: First shot at updating for SDL3 audio API.
a2b488359e dsp: Removed debug logging
6fd71185cd dsp: Updated for new SDL3 audio API.
3482d1215a alsa: Don't ever block in CaptureFromDevice.
65d296ef1a audio: Use SDL_powerof2 instead of reinventing it.
409b544505 alsa: Updated for new SDL3 audio API
0999a090a7 audio: More tweaking of `device->thread_alive`
f94ffd6092 audio: Fixed logic error
4deb2970c9 alsa: Renamed `_this` to `device`
0fb9e4baae audio: Remove no-longer-used SupportsNonPow2Samples
c653e57768 coreaudio: rewritten for SDL3 audio redesign!
533777eff5 audio: SDL_sysaudio.h comment conversion.
8473e522e0 audio: unify device thread naming.
258bc9efed audio: PlayDevice now passes the buffer, too, for convenience.
e518149d14 audio: Fixed locking in SDL_AudioDeviceDisconnected
22afa5735f audio: FreeDeviceHandle should pass the whole device, for convenience.
9e3c5f93e0 coreaudio: Change `_this` to `device`
e969160de0 audio: unset a freed variable to NULL
1fc01b0300 audio: Try to definitely have a default device set up.
b60a56d368 audio: take first reported device if no default was specified.
a8323ebe68 audio: Better handling of ProvidesOwnCallbackThread backends.
1dffb72c1d pipewire: Hooked up default device change notifications.
a93fcf2444 audio: fixed flushed stream reporting bytes but not being able to get them.
ad6c1781fc pulseaudio: Minor cleanups.
cfc8a0d17d pipewire: First shot at moving to the new SDL3 audio interfaces.
13202642a3 aaudio: Fixed capitialization, plus some minor cleanups.
3e9991b535 audio: Make sure we don't write to a NULL pointer.
943351affb pulseaudio: GetDefaultAudioInfo isn't a thing anymore.
11dfc4d737 test: Update testautomation_audio for SDL3 audio API.
29afc2e42b test: Update testresample for SDL3 audio API.
3a02eecced test: Update testsurround for SDL3 audio API.
e1c78718d4 test: testaudiocapture is updated for the SDL3 audio API.
f48cb716c2 pulseaudio: a couple minor tweaks.
dac25fe9eb audio: Seperate audio capture into Wait/Read operations.
3e10c0005d audio: Capture devices should respect logical device pausing.
7e700531c5 audio: Allow SDL_OpenAudioDevice to accept a NULL spec.
bb1cbbd33a test: Update testaudioinfo for SDL3 audio API.
883aee32c5 audio: Let default formats differ for output and capture devices.
62cf24eeb9 pulseaudio: Listen for server events in addition to sources and sinks.
924f370bd7 pulseaudio: Fix deadlock in HotplugThread.
5d4e9e5f80 test: Updated testaudiostreamdynamicresample to SDL3 audio API.
f883b9fc64 test: Updated testaudiohotplug to SDL3 audio API.
2be5f726d4 audio: Removed debug logging.
323ecce123 docs: Added migration note about SDL_AUDIODEVICEREMOVED.
47b0321ebf test: Removed loopwavequeue.c; obsolete in SDL3.
0e5a1d4f29 pulseaudio: Removed debug logging.
f598626e46 test: loopwave shouldn't use an audiostream callback.
eee407caf8 docs: migration guide note that SDL_LoadWAV has a different return type.
b03c493fc4 test: Updated testmultiaudio to new SDL3 audio API
fe1daf6fb5 audio: Mark disconnected default devices as "zombies".
cdd2ba81de audio: Fixed adding new physical devices to a double-linked list.
db39cbf208 audio: Allow SDL_GetAudioDeviceFormat() to query the default devices.
ee10bab3cd audio: An enormous amount of work on managing default devices.
c7a44eea83 audio: Fixed logic error.
089cd87cb5 audio: Make sure device count stays correct as hardware disconnects.
e50cb72eb6 docs: Note that audio opening doesn't implicitly init SDL now.
97b2f747d0 docs: Corrections to audio section of README-migration.md
464640440f audio: Added SDL_GetAudioStreamBinding.
01f7b53865 audio: Readded (logical) device pausing.
fd4c9f4e11 audio: documentation improvements.
4b78b789a7 audio: Switch SDL_audio.c and SDL_audiocvt.c to C99-ish syntax.
d96a1db7d7 audio: Opening via a logical device ID should also track default device.
b2e020958f audio: Wrap device access in opening of logical devices.
7ee2459927 audio: Check for unlikely failure case in WAV loaded.
3d65a2cefe audio: Made SDL_LoadWAV a real function, not just a macro.
26525f5fd3 audio: Readd SDL_AudioSpec, but just with format/channels/freq fields.
e6aaed7d79 include: Audio is not, and has not been, a raw mixing buffer for a long time.
56b1bc2198 audio: SDL_AudioStream now has callbacks for Get and Put operations.
905c4fff5b audio: First shot at the SDL3 audio subsystem redesign!
b221b59995 cmake: add SDL_REVISION option
0500fca00c Add missing break
d3f2de7f29 fixed typo in prev. patch.
12b35c6a46 test/testnativecocoa.m: fixed deprecation warnings.
e24b3e2fa4 cmake: rename SDL_TEST -> SDL_TEST_LIBRARY
da5016d336 cmake: use pkg-config + test compile instead of Find module for detecting rpi
deec574ff6 cmake: fix SDL_HIDAPI_LIBUSB
f2ae00c1ad Sync SDL3 wiki -> header
41a96c8133 doc: document building of SDL tests with CMake
3174d0b970 Sorted controller list
27b8abb056 Add Steam Deck controller mapping to database.
41d436f0fe Use SetWindowPos to show windows when SDL_HINT_WINDOW_ACTIVATE_WHEN_SHOWN is set to avoid activating the parent window when showing a child window
0dc85f3078 Improved the documentation for the gamepad paddle buttons
2fff999a41 Try to create the dummy mouse cursor after video backend initialization
d086d9874d Sync SDL3 wiki -> header
bce598addd SDL_pixels.c: Fixed compiler warning on Android NDK.
ad0c0d3cde Sync SDL3 wiki -> header
f8e8dff7ee tests: Fix automated window grab and positioning tests under Wayland/XWayland
4cffbc3644 Add VS code directory to gitignore
666f81bace Add more endian-specific aliases for 32 bit pixelformats
4749df0a63 Just disable the 4214 warning instead of trying to change the structure definition

git-subtree-dir: external/sdl/SDL
git-subtree-split: ec0042081ea104d5dd0ee291105210e00a4fe3d9
2023-08-12 20:17:29 +02:00

1003 lines
37 KiB
C

/**
* Original code: automated SDL audio test written by Edgar Simo "bobbens"
* New/updated tests: aschiffler at ferzkopp dot net
*/
/* quiet windows compiler warnings */
#if defined(_MSC_VER) && !defined(_CRT_SECURE_NO_WARNINGS)
#define _CRT_SECURE_NO_WARNINGS
#endif
#include <math.h>
#include <stdio.h>
#include <SDL3/SDL.h>
#include <SDL3/SDL_test.h>
#include "testautomation_suites.h"
/* ================= Test Case Implementation ================== */
/* Fixture */
static void audioSetUp(void *arg)
{
/* Start SDL audio subsystem */
int ret = SDL_InitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_InitSubSystem(SDL_INIT_AUDIO)");
SDLTest_AssertCheck(ret == 0, "Check result from SDL_InitSubSystem(SDL_INIT_AUDIO)");
if (ret != 0) {
SDLTest_LogError("%s", SDL_GetError());
}
}
static void audioTearDown(void *arg)
{
/* Remove a possibly created file from SDL disk writer audio driver; ignore errors */
(void)remove("sdlaudio.raw");
SDLTest_AssertPass("Cleanup of test files completed");
}
#if 0 /* !!! FIXME: maybe update this? */
/* Global counter for callback invocation */
static int g_audio_testCallbackCounter;
/* Global accumulator for total callback length */
static int g_audio_testCallbackLength;
/* Test callback function */
static void SDLCALL audio_testCallback(void *userdata, Uint8 *stream, int len)
{
/* track that callback was called */
g_audio_testCallbackCounter++;
g_audio_testCallbackLength += len;
}
#endif
static SDL_AudioDeviceID g_audio_id = -1;
/* Test case functions */
/**
* \brief Stop and restart audio subsystem
*
* \sa SDL_QuitSubSystem
* \sa SDL_InitSubSystem
*/
static int audio_quitInitAudioSubSystem(void *arg)
{
/* Stop SDL audio subsystem */
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO)");
/* Restart audio again */
audioSetUp(NULL);
return TEST_COMPLETED;
}
/**
* \brief Start and stop audio directly
*
* \sa SDL_InitAudio
* \sa SDL_QuitAudio
*/
static int audio_initQuitAudio(void *arg)
{
int result;
int i, iMax;
const char *audioDriver;
/* Stop SDL audio subsystem */
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO)");
/* Loop over all available audio drivers */
iMax = SDL_GetNumAudioDrivers();
SDLTest_AssertPass("Call to SDL_GetNumAudioDrivers()");
SDLTest_AssertCheck(iMax > 0, "Validate number of audio drivers; expected: >0 got: %d", iMax);
for (i = 0; i < iMax; i++) {
audioDriver = SDL_GetAudioDriver(i);
SDLTest_AssertPass("Call to SDL_GetAudioDriver(%d)", i);
SDLTest_Assert(audioDriver != NULL, "Audio driver name is not NULL");
SDLTest_AssertCheck(audioDriver[0] != '\0', "Audio driver name is not empty; got: %s", audioDriver); /* NOLINT(clang-analyzer-core.NullDereference): Checked for NULL above */
/* Call Init */
SDL_SetHint("SDL_AUDIO_DRIVER", audioDriver);
result = SDL_InitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_InitSubSystem(SDL_INIT_AUDIO) with driver='%s'", audioDriver);
SDLTest_AssertCheck(result == 0, "Validate result value; expected: 0 got: %d", result);
/* Call Quit */
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO)");
}
/* NULL driver specification */
audioDriver = NULL;
/* Call Init */
SDL_SetHint("SDL_AUDIO_DRIVER", audioDriver);
result = SDL_InitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_AudioInit(NULL)");
SDLTest_AssertCheck(result == 0, "Validate result value; expected: 0 got: %d", result);
/* Call Quit */
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO)");
/* Restart audio again */
audioSetUp(NULL);
return TEST_COMPLETED;
}
/**
* \brief Start, open, close and stop audio
*
* \sa SDL_InitAudio
* \sa SDL_OpenAudioDevice
* \sa SDL_CloseAudioDevice
* \sa SDL_QuitAudio
*/
static int audio_initOpenCloseQuitAudio(void *arg)
{
int result;
int i, iMax, j, k;
const char *audioDriver;
SDL_AudioSpec desired;
/* Stop SDL audio subsystem */
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO)");
/* Loop over all available audio drivers */
iMax = SDL_GetNumAudioDrivers();
SDLTest_AssertPass("Call to SDL_GetNumAudioDrivers()");
SDLTest_AssertCheck(iMax > 0, "Validate number of audio drivers; expected: >0 got: %d", iMax);
for (i = 0; i < iMax; i++) {
audioDriver = SDL_GetAudioDriver(i);
SDLTest_AssertPass("Call to SDL_GetAudioDriver(%d)", i);
SDLTest_Assert(audioDriver != NULL, "Audio driver name is not NULL");
SDLTest_AssertCheck(audioDriver[0] != '\0', "Audio driver name is not empty; got: %s", audioDriver); /* NOLINT(clang-analyzer-core.NullDereference): Checked for NULL above */
/* Change specs */
for (j = 0; j < 2; j++) {
/* Call Init */
SDL_SetHint("SDL_AUDIO_DRIVER", audioDriver);
result = SDL_InitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_InitSubSystem(SDL_INIT_AUDIO) with driver='%s'", audioDriver);
SDLTest_AssertCheck(result == 0, "Validate result value; expected: 0 got: %d", result);
/* Set spec */
SDL_memset(&desired, 0, sizeof(desired));
switch (j) {
case 0:
/* Set standard desired spec */
desired.freq = 22050;
desired.format = SDL_AUDIO_S16SYS;
desired.channels = 2;
case 1:
/* Set custom desired spec */
desired.freq = 48000;
desired.format = SDL_AUDIO_F32SYS;
desired.channels = 2;
break;
}
/* Call Open (maybe multiple times) */
for (k = 0; k <= j; k++) {
result = SDL_OpenAudioDevice(SDL_AUDIO_DEVICE_DEFAULT_OUTPUT, &desired);
if (k == 0) {
g_audio_id = result;
}
SDLTest_AssertPass("Call to SDL_OpenAudioDevice(SDL_AUDIO_DEVICE_DEFAULT_OUTPUT, desired_spec_%d), call %d", j, k + 1);
SDLTest_AssertCheck(result > 0, "Verify return value; expected: > 0, got: %d", result);
}
/* Call Close (maybe multiple times) */
for (k = 0; k <= j; k++) {
SDL_CloseAudioDevice(g_audio_id);
SDLTest_AssertPass("Call to SDL_CloseAudioDevice(), call %d", k + 1);
}
/* Call Quit (maybe multiple times) */
for (k = 0; k <= j; k++) {
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO), call %d", k + 1);
}
} /* spec loop */
} /* driver loop */
/* Restart audio again */
audioSetUp(NULL);
return TEST_COMPLETED;
}
/**
* \brief Pause and unpause audio
*
* \sa SDL_PauseAudioDevice
* \sa SDL_PlayAudioDevice
*/
static int audio_pauseUnpauseAudio(void *arg)
{
int iMax;
int i, j /*, k, l*/;
int result;
const char *audioDriver;
SDL_AudioSpec desired;
/* Stop SDL audio subsystem */
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO)");
/* Loop over all available audio drivers */
iMax = SDL_GetNumAudioDrivers();
SDLTest_AssertPass("Call to SDL_GetNumAudioDrivers()");
SDLTest_AssertCheck(iMax > 0, "Validate number of audio drivers; expected: >0 got: %d", iMax);
for (i = 0; i < iMax; i++) {
audioDriver = SDL_GetAudioDriver(i);
SDLTest_AssertPass("Call to SDL_GetAudioDriver(%d)", i);
SDLTest_Assert(audioDriver != NULL, "Audio driver name is not NULL");
SDLTest_AssertCheck(audioDriver[0] != '\0', "Audio driver name is not empty; got: %s", audioDriver); /* NOLINT(clang-analyzer-core.NullDereference): Checked for NULL above */
/* Change specs */
for (j = 0; j < 2; j++) {
/* Call Init */
SDL_SetHint("SDL_AUDIO_DRIVER", audioDriver);
result = SDL_InitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_InitSubSystem(SDL_INIT_AUDIO) with driver='%s'", audioDriver);
SDLTest_AssertCheck(result == 0, "Validate result value; expected: 0 got: %d", result);
/* Set spec */
SDL_memset(&desired, 0, sizeof(desired));
switch (j) {
case 0:
/* Set standard desired spec */
desired.freq = 22050;
desired.format = SDL_AUDIO_S16SYS;
desired.channels = 2;
break;
case 1:
/* Set custom desired spec */
desired.freq = 48000;
desired.format = SDL_AUDIO_F32SYS;
desired.channels = 2;
break;
}
/* Call Open */
g_audio_id = SDL_OpenAudioDevice(SDL_AUDIO_DEVICE_DEFAULT_OUTPUT, &desired);
result = g_audio_id;
SDLTest_AssertPass("Call to SDL_OpenAudioDevice(SDL_AUDIO_DEVICE_DEFAULT_OUTPUT, desired_spec_%d)", j);
SDLTest_AssertCheck(result > 0, "Verify return value; expected > 0 got: %d", result);
#if 0 /* !!! FIXME: maybe update this? */
/* Start and stop audio multiple times */
for (l = 0; l < 3; l++) {
SDLTest_Log("Pause/Unpause iteration: %d", l + 1);
/* Reset callback counters */
g_audio_testCallbackCounter = 0;
g_audio_testCallbackLength = 0;
/* Un-pause audio to start playing (maybe multiple times) */
for (k = 0; k <= j; k++) {
SDL_PlayAudioDevice(g_audio_id);
SDLTest_AssertPass("Call to SDL_PlayAudioDevice(g_audio_id), call %d", k + 1);
}
/* Wait for callback */
int totalDelay = 0;
do {
SDL_Delay(10);
totalDelay += 10;
} while (g_audio_testCallbackCounter == 0 && totalDelay < 1000);
SDLTest_AssertCheck(g_audio_testCallbackCounter > 0, "Verify callback counter; expected: >0 got: %d", g_audio_testCallbackCounter);
SDLTest_AssertCheck(g_audio_testCallbackLength > 0, "Verify callback length; expected: >0 got: %d", g_audio_testCallbackLength);
/* Pause audio to stop playing (maybe multiple times) */
for (k = 0; k <= j; k++) {
const int pause_on = (k == 0) ? 1 : SDLTest_RandomIntegerInRange(99, 9999);
if (pause_on) {
SDL_PauseAudioDevice(g_audio_id);
SDLTest_AssertPass("Call to SDL_PauseAudioDevice(g_audio_id), call %d", k + 1);
} else {
SDL_PlayAudioDevice(g_audio_id);
SDLTest_AssertPass("Call to SDL_PlayAudioDevice(g_audio_id), call %d", k + 1);
}
}
/* Ensure callback is not called again */
const int originalCounter = g_audio_testCallbackCounter;
SDL_Delay(totalDelay + 10);
SDLTest_AssertCheck(originalCounter == g_audio_testCallbackCounter, "Verify callback counter; expected: %d, got: %d", originalCounter, g_audio_testCallbackCounter);
}
#endif
/* Call Close */
SDL_CloseAudioDevice(g_audio_id);
SDLTest_AssertPass("Call to SDL_CloseAudioDevice()");
/* Call Quit */
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDLTest_AssertPass("Call to SDL_QuitSubSystem(SDL_INIT_AUDIO)");
} /* spec loop */
} /* driver loop */
/* Restart audio again */
audioSetUp(NULL);
return TEST_COMPLETED;
}
/**
* \brief Enumerate and name available audio devices (output and capture).
*
* \sa SDL_GetNumAudioDevices
* \sa SDL_GetAudioDeviceName
*/
static int audio_enumerateAndNameAudioDevices(void *arg)
{
int t;
int i, n;
char *name;
SDL_AudioDeviceID *devices = NULL;
/* Iterate over types: t=0 output device, t=1 input/capture device */
for (t = 0; t < 2; t++) {
/* Get number of devices. */
devices = (t) ? SDL_GetAudioCaptureDevices(&n) : SDL_GetAudioOutputDevices(&n);
SDLTest_AssertPass("Call to SDL_GetAudio%sDevices(%i)", (t) ? "Capture" : "Output", t);
SDLTest_Log("Number of %s devices < 0, reported as %i", (t) ? "capture" : "output", n);
SDLTest_AssertCheck(n >= 0, "Validate result is >= 0, got: %i", n);
/* List devices. */
if (n > 0) {
SDLTest_AssertCheck(devices != NULL, "Validate devices is not NULL if n > 0");
for (i = 0; i < n; i++) {
name = SDL_GetAudioDeviceName(devices[i]);
SDLTest_AssertPass("Call to SDL_GetAudioDeviceName(%i)", i);
SDLTest_AssertCheck(name != NULL, "Verify result from SDL_GetAudioDeviceName(%i) is not NULL", i);
if (name != NULL) {
SDLTest_AssertCheck(name[0] != '\0', "verify result from SDL_GetAudioDeviceName(%i) is not empty, got: '%s'", i, name);
SDL_free(name);
}
}
}
SDL_free(devices);
}
return TEST_COMPLETED;
}
/**
* \brief Negative tests around enumeration and naming of audio devices.
*
* \sa SDL_GetNumAudioDevices
* \sa SDL_GetAudioDeviceName
*/
static int audio_enumerateAndNameAudioDevicesNegativeTests(void *arg)
{
return TEST_COMPLETED; /* nothing in here atm since these interfaces changed in SDL3. */
}
/**
* \brief Checks available audio driver names.
*
* \sa SDL_GetNumAudioDrivers
* \sa SDL_GetAudioDriver
*/
static int audio_printAudioDrivers(void *arg)
{
int i, n;
const char *name;
/* Get number of drivers */
n = SDL_GetNumAudioDrivers();
SDLTest_AssertPass("Call to SDL_GetNumAudioDrivers()");
SDLTest_AssertCheck(n >= 0, "Verify number of audio drivers >= 0, got: %i", n);
/* List drivers. */
if (n > 0) {
for (i = 0; i < n; i++) {
name = SDL_GetAudioDriver(i);
SDLTest_AssertPass("Call to SDL_GetAudioDriver(%i)", i);
SDLTest_AssertCheck(name != NULL, "Verify returned name is not NULL");
if (name != NULL) {
SDLTest_AssertCheck(name[0] != '\0', "Verify returned name is not empty, got: '%s'", name);
}
}
}
return TEST_COMPLETED;
}
/**
* \brief Checks current audio driver name with initialized audio.
*
* \sa SDL_GetCurrentAudioDriver
*/
static int audio_printCurrentAudioDriver(void *arg)
{
/* Check current audio driver */
const char *name = SDL_GetCurrentAudioDriver();
SDLTest_AssertPass("Call to SDL_GetCurrentAudioDriver()");
SDLTest_AssertCheck(name != NULL, "Verify returned name is not NULL");
if (name != NULL) {
SDLTest_AssertCheck(name[0] != '\0', "Verify returned name is not empty, got: '%s'", name);
}
return TEST_COMPLETED;
}
/* Definition of all formats, channels, and frequencies used to test audio conversions */
static SDL_AudioFormat g_audioFormats[] = { SDL_AUDIO_S8, SDL_AUDIO_U8, SDL_AUDIO_S16LSB, SDL_AUDIO_S16MSB, SDL_AUDIO_S16SYS, SDL_AUDIO_S16,
SDL_AUDIO_S32LSB, SDL_AUDIO_S32MSB, SDL_AUDIO_S32SYS, SDL_AUDIO_S32,
SDL_AUDIO_F32LSB, SDL_AUDIO_F32MSB, SDL_AUDIO_F32SYS, SDL_AUDIO_F32 };
static const char *g_audioFormatsVerbose[] = { "SDL_AUDIO_S8", "SDL_AUDIO_U8", "SDL_AUDIO_S16LSB", "SDL_AUDIO_S16MSB", "SDL_AUDIO_S16SYS", "SDL_AUDIO_S16",
"SDL_AUDIO_S32LSB", "SDL_AUDIO_S32MSB", "SDL_AUDIO_S32SYS", "SDL_AUDIO_S32",
"SDL_AUDIO_F32LSB", "SDL_AUDIO_F32MSB", "SDL_AUDIO_F32SYS", "SDL_AUDIO_F32" };
static const int g_numAudioFormats = SDL_arraysize(g_audioFormats);
static Uint8 g_audioChannels[] = { 1, 2, 4, 6 };
static const int g_numAudioChannels = SDL_arraysize(g_audioChannels);
static int g_audioFrequencies[] = { 11025, 22050, 44100, 48000 };
static const int g_numAudioFrequencies = SDL_arraysize(g_audioFrequencies);
/**
* \brief Builds various audio conversion structures
*
* \sa SDL_CreateAudioStream
*/
static int audio_buildAudioStream(void *arg)
{
SDL_AudioStream *stream;
SDL_AudioSpec spec1;
SDL_AudioSpec spec2;
int i, ii, j, jj, k, kk;
/* No conversion needed */
spec1.format = SDL_AUDIO_S16LSB;
spec1.channels = 2;
spec1.freq = 22050;
stream = SDL_CreateAudioStream(&spec1, &spec1);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(spec1 ==> spec1)");
SDLTest_AssertCheck(stream != NULL, "Verify stream value; expected: != NULL, got: %p", (void *)stream);
SDL_DestroyAudioStream(stream);
/* Typical conversion */
spec1.format = SDL_AUDIO_S8;
spec1.channels = 1;
spec1.freq = 22050;
spec2.format = SDL_AUDIO_S16LSB;
spec2.channels = 2;
spec2.freq = 44100;
stream = SDL_CreateAudioStream(&spec1, &spec2);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(spec1 ==> spec2)");
SDLTest_AssertCheck(stream != NULL, "Verify stream value; expected: != NULL, got: %p", (void *)stream);
SDL_DestroyAudioStream(stream);
/* All source conversions with random conversion targets, allow 'null' conversions */
for (i = 0; i < g_numAudioFormats; i++) {
for (j = 0; j < g_numAudioChannels; j++) {
for (k = 0; k < g_numAudioFrequencies; k++) {
spec1.format = g_audioFormats[i];
spec1.channels = g_audioChannels[j];
spec1.freq = g_audioFrequencies[k];
ii = SDLTest_RandomIntegerInRange(0, g_numAudioFormats - 1);
jj = SDLTest_RandomIntegerInRange(0, g_numAudioChannels - 1);
kk = SDLTest_RandomIntegerInRange(0, g_numAudioFrequencies - 1);
spec2.format = g_audioFormats[ii];
spec2.channels = g_audioChannels[jj];
spec2.freq = g_audioFrequencies[kk];
stream = SDL_CreateAudioStream(&spec1, &spec2);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(format[%i]=%s(%i),channels[%i]=%i,freq[%i]=%i ==> format[%i]=%s(%i),channels[%i]=%i,freq[%i]=%i)",
i, g_audioFormatsVerbose[i], spec1.format, j, spec1.channels, k, spec1.freq, ii, g_audioFormatsVerbose[ii], spec2.format, jj, spec2.channels, kk, spec2.freq);
SDLTest_AssertCheck(stream != NULL, "Verify stream value; expected: != NULL, got: %p", (void *)stream);
if (stream == NULL) {
SDLTest_LogError("%s", SDL_GetError());
}
SDL_DestroyAudioStream(stream);
}
}
}
return TEST_COMPLETED;
}
/**
* \brief Checks calls with invalid input to SDL_CreateAudioStream
*
* \sa SDL_CreateAudioStream
*/
static int audio_buildAudioStreamNegative(void *arg)
{
const char *error;
SDL_AudioStream *stream;
SDL_AudioSpec spec1;
SDL_AudioSpec spec2;
int i;
char message[256];
/* Valid format */
spec1.format = SDL_AUDIO_S8;
spec1.channels = 1;
spec1.freq = 22050;
spec2.format = SDL_AUDIO_S16LSB;
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_S16LSB;
spec2.channels = 2;
spec2.freq = 44100;
SDL_ClearError();
SDLTest_AssertPass("Call to SDL_ClearError()");
/* Set various invalid format inputs */
SDL_strlcpy(message, "Invalid: ", 256);
if (i & 1) {
SDL_strlcat(message, " spec1.format", 256);
spec1.format = 0;
}
if (i & 2) {
SDL_strlcat(message, " spec1.channels", 256);
spec1.channels = 0;
}
if (i & 4) {
SDL_strlcat(message, " spec1.freq", 256);
spec1.freq = 0;
}
if (i & 8) {
SDL_strlcat(message, " spec2.format", 256);
spec2.format = 0;
}
if (i & 16) {
SDL_strlcat(message, " spec2.channels", 256);
spec2.channels = 0;
}
if (i & 32) {
SDL_strlcat(message, " spec2.freq", 256);
spec2.freq = 0;
}
SDLTest_Log("%s", message);
stream = SDL_CreateAudioStream(&spec1, &spec2);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(spec1 ==> spec2)");
SDLTest_AssertCheck(stream == NULL, "Verify stream value; expected: NULL, got: %p", (void *)stream);
error = SDL_GetError();
SDLTest_AssertPass("Call to SDL_GetError()");
SDLTest_AssertCheck(error != NULL && error[0] != '\0', "Validate that error message was not NULL or empty");
SDL_DestroyAudioStream(stream);
}
SDL_ClearError();
SDLTest_AssertPass("Call to SDL_ClearError()");
return TEST_COMPLETED;
}
/**
* \brief Checks current audio status.
*
* \sa SDL_GetAudioDeviceStatus
*/
static int audio_getAudioStatus(void *arg)
{
return TEST_COMPLETED; /* no longer a thing in SDL3. */
}
/**
* \brief Opens, checks current audio status, and closes a device.
*
* \sa SDL_GetAudioStatus
*/
static int audio_openCloseAndGetAudioStatus(void *arg)
{
return TEST_COMPLETED; /* not a thing in SDL3. */
}
/**
* \brief Locks and unlocks open audio device.
*
* \sa SDL_LockAudioDevice
* \sa SDL_UnlockAudioDevice
*/
static int audio_lockUnlockOpenAudioDevice(void *arg)
{
return TEST_COMPLETED; /* not a thing in SDL3 */
}
/**
* \brief Convert audio using various conversion structures
*
* \sa SDL_CreateAudioStream
*/
static int audio_convertAudio(void *arg)
{
SDL_AudioStream *stream;
SDL_AudioSpec spec1;
SDL_AudioSpec spec2;
int c;
char message[128];
int i, ii, j, jj, k, kk;
/* Iterate over bitmask that determines which parameters are modified in the conversion */
for (c = 1; c < 8; c++) {
SDL_strlcpy(message, "Changing:", 128);
if (c & 1) {
SDL_strlcat(message, " Format", 128);
}
if (c & 2) {
SDL_strlcat(message, " Channels", 128);
}
if (c & 4) {
SDL_strlcat(message, " Frequencies", 128);
}
SDLTest_Log("%s", message);
/* All source conversions with random conversion targets */
for (i = 0; i < g_numAudioFormats; i++) {
for (j = 0; j < g_numAudioChannels; j++) {
for (k = 0; k < g_numAudioFrequencies; k++) {
spec1.format = g_audioFormats[i];
spec1.channels = g_audioChannels[j];
spec1.freq = g_audioFrequencies[k];
/* Ensure we have a different target format */
do {
if (c & 1) {
ii = SDLTest_RandomIntegerInRange(0, g_numAudioFormats - 1);
} else {
ii = 1;
}
if (c & 2) {
jj = SDLTest_RandomIntegerInRange(0, g_numAudioChannels - 1);
} else {
jj = j;
}
if (c & 4) {
kk = SDLTest_RandomIntegerInRange(0, g_numAudioFrequencies - 1);
} else {
kk = k;
}
} while ((i == ii) && (j == jj) && (k == kk));
spec2.format = g_audioFormats[ii];
spec2.channels = g_audioChannels[jj];
spec2.freq = g_audioFrequencies[kk];
stream = SDL_CreateAudioStream(&spec1, &spec2);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(format[%i]=%s(%i),channels[%i]=%i,freq[%i]=%i ==> format[%i]=%s(%i),channels[%i]=%i,freq[%i]=%i)",
i, g_audioFormatsVerbose[i], spec1.format, j, spec1.channels, k, spec1.freq, ii, g_audioFormatsVerbose[ii], spec2.format, jj, spec2.channels, kk, spec2.freq);
SDLTest_AssertCheck(stream != NULL, "Verify stream value; expected: != NULL, got: %p", (void *)stream);
if (stream == NULL) {
SDLTest_LogError("%s", SDL_GetError());
} else {
Uint8 *dst_buf = NULL, *src_buf = NULL;
int dst_len = 0, src_len = 0, real_dst_len = 0;
int l = 64;
int src_samplesize, dst_samplesize;
src_samplesize = (SDL_AUDIO_BITSIZE(spec1.format) / 8) * spec1.channels;
dst_samplesize = (SDL_AUDIO_BITSIZE(spec2.format) / 8) * spec2.channels;
/* Create some random data to convert */
src_len = l * src_samplesize;
SDLTest_Log("Creating dummy sample buffer of %i length (%i bytes)", l, src_len);
src_buf = (Uint8 *)SDL_malloc(src_len);
SDLTest_AssertCheck(dst_buf != NULL, "Check src data buffer to convert is not NULL");
if (src_buf == NULL) {
return TEST_ABORTED;
}
src_len = src_len & ~(src_samplesize - 1);
dst_len = dst_samplesize * (src_len / src_samplesize);
if (spec1.freq < spec2.freq) {
const double mult = ((double)spec2.freq) / ((double)spec1.freq);
dst_len *= (int) SDL_ceil(mult);
}
dst_len = dst_len & ~(dst_samplesize - 1);
dst_buf = (Uint8 *)SDL_calloc(1, dst_len);
SDLTest_AssertCheck(dst_buf != NULL, "Check dst data buffer to convert is not NULL");
if (dst_buf == NULL) {
return TEST_ABORTED;
}
/* Run the audio converter */
if (SDL_PutAudioStreamData(stream, src_buf, src_len) < 0 ||
SDL_FlushAudioStream(stream) < 0) {
return TEST_ABORTED;
}
real_dst_len = SDL_GetAudioStreamData(stream, dst_buf, dst_len);
SDLTest_AssertCheck(real_dst_len > 0, "Verify result value; expected: > 0; got: %i", real_dst_len);
if (real_dst_len < 0) {
return TEST_ABORTED;
}
SDL_DestroyAudioStream(stream);
/* Free converted buffer */
SDL_free(src_buf);
SDL_free(dst_buf);
}
}
}
}
}
return TEST_COMPLETED;
}
/**
* \brief Opens, checks current connected status, and closes a device.
*
* \sa SDL_AudioDeviceConnected
*/
static int audio_openCloseAudioDeviceConnected(void *arg)
{
return TEST_COMPLETED; /* not a thing in SDL3. */
}
static double sine_wave_sample(const Sint64 idx, const Sint64 rate, const Sint64 freq, const double phase)
{
/* Using integer modulo to avoid precision loss caused by large floating
* point numbers. Sint64 is needed for the large integer multiplication.
* The integers are assumed to be non-negative so that modulo is always
* non-negative.
* sin(i / rate * freq * 2 * PI + phase)
* = sin(mod(i / rate * freq, 1) * 2 * PI + phase)
* = sin(mod(i * freq, rate) / rate * 2 * PI + phase) */
return SDL_sin(((double)(idx * freq % rate)) / ((double)rate) * (SDL_PI_D * 2) + phase);
}
/**
* \brief Check signal-to-noise ratio and maximum error of audio resampling.
*
* \sa https://wiki.libsdl.org/SDL_CreateAudioStream
* \sa https://wiki.libsdl.org/SDL_DestroyAudioStream
* \sa https://wiki.libsdl.org/SDL_PutAudioStreamData
* \sa https://wiki.libsdl.org/SDL_FlushAudioStream
* \sa https://wiki.libsdl.org/SDL_GetAudioStreamData
*/
static int audio_resampleLoss(void *arg)
{
/* Note: always test long input time (>= 5s from experience) in some test
* cases because an improper implementation may suffer from low resampling
* precision with long input due to e.g. doing subtraction with large floats. */
struct test_spec_t {
int time;
int freq;
double phase;
int rate_in;
int rate_out;
double signal_to_noise;
double max_error;
} test_specs[] = {
{ 50, 440, 0, 44100, 48000, 60, 0.0025 },
{ 50, 5000, SDL_PI_D / 2, 20000, 10000, 65, 0.0010 },
{ 0 }
};
int spec_idx = 0;
for (spec_idx = 0; test_specs[spec_idx].time > 0; ++spec_idx) {
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 * (int)sizeof(float);
const int len_target = frames_target * (int)sizeof(float);
SDL_AudioSpec tmpspec1, tmpspec2;
Uint64 tick_beg = 0;
Uint64 tick_end = 0;
int i = 0;
int ret = 0;
SDL_AudioStream *stream = NULL;
float *buf_in = NULL;
float *buf_out = NULL;
int len_out = 0;
double max_error = 0;
double sum_squared_error = 0;
double sum_squared_value = 0;
double signal_to_noise = 0;
SDLTest_AssertPass("Test resampling of %i s %i Hz %f phase sine wave from sampling rate of %i Hz to %i Hz",
spec->time, spec->freq, spec->phase, spec->rate_in, spec->rate_out);
tmpspec1.format = SDL_AUDIO_F32;
tmpspec1.channels = 1;
tmpspec1.freq = spec->rate_in;
tmpspec2.format = SDL_AUDIO_F32;
tmpspec2.channels = 1;
tmpspec2.freq = spec->rate_out;
stream = SDL_CreateAudioStream(&tmpspec1, &tmpspec2);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(SDL_AUDIO_F32, 1, %i, SDL_AUDIO_F32, 1, %i)", spec->rate_in, spec->rate_out);
SDLTest_AssertCheck(stream != NULL, "Expected SDL_CreateAudioStream to succeed.");
if (stream == NULL) {
return TEST_ABORTED;
}
buf_in = (float *)SDL_malloc(len_in);
SDLTest_AssertCheck(buf_in != NULL, "Expected input buffer to be created.");
if (buf_in == NULL) {
SDL_DestroyAudioStream(stream);
return TEST_ABORTED;
}
for (i = 0; i < frames_in; ++i) {
*(buf_in + i) = (float)sine_wave_sample(i, spec->rate_in, spec->freq, spec->phase);
}
tick_beg = SDL_GetPerformanceCounter();
ret = SDL_PutAudioStreamData(stream, buf_in, len_in);
SDLTest_AssertPass("Call to SDL_PutAudioStreamData(stream, buf_in, %i)", len_in);
SDLTest_AssertCheck(ret == 0, "Expected SDL_PutAudioStreamData to succeed.");
SDL_free(buf_in);
if (ret != 0) {
SDL_DestroyAudioStream(stream);
return TEST_ABORTED;
}
ret = SDL_FlushAudioStream(stream);
SDLTest_AssertPass("Call to SDL_FlushAudioStream(stream)");
SDLTest_AssertCheck(ret == 0, "Expected SDL_FlushAudioStream to succeed");
if (ret != 0) {
SDL_DestroyAudioStream(stream);
return TEST_ABORTED;
}
buf_out = (float *)SDL_malloc(len_target);
SDLTest_AssertCheck(buf_out != NULL, "Expected output buffer to be created.");
if (buf_out == NULL) {
SDL_DestroyAudioStream(stream);
return TEST_ABORTED;
}
len_out = SDL_GetAudioStreamData(stream, buf_out, len_target);
SDLTest_AssertPass("Call to SDL_GetAudioStreamData(stream, buf_out, %i)", len_target);
/** !!! FIXME: SDL_AudioStream does not return output of the same length as
** !!! FIXME: the input even if SDL_FlushAudioStream is called. */
SDLTest_AssertCheck(len_out <= len_target, "Expected output length to be no larger than %i, got %i.",
len_target, len_out);
SDL_DestroyAudioStream(stream);
if (len_out > len_target) {
SDL_free(buf_out);
return TEST_ABORTED;
}
tick_end = SDL_GetPerformanceCounter();
SDLTest_Log("Resampling used %f seconds.", ((double)(tick_end - tick_beg)) / SDL_GetPerformanceFrequency());
for (i = 0; i < len_out / (int)sizeof(float); ++i) {
const float output = *(buf_out + i);
const double target = sine_wave_sample(i, spec->rate_out, spec->freq, spec->phase);
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);
}
return TEST_COMPLETED;
}
/* ================= Test Case References ================== */
/* Audio test cases */
static const SDLTest_TestCaseReference audioTest1 = {
audio_enumerateAndNameAudioDevices, "audio_enumerateAndNameAudioDevices", "Enumerate and name available audio devices (output and capture)", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest2 = {
audio_enumerateAndNameAudioDevicesNegativeTests, "audio_enumerateAndNameAudioDevicesNegativeTests", "Negative tests around enumeration and naming of audio devices.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest3 = {
audio_printAudioDrivers, "audio_printAudioDrivers", "Checks available audio driver names.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest4 = {
audio_printCurrentAudioDriver, "audio_printCurrentAudioDriver", "Checks current audio driver name with initialized audio.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest5 = {
audio_buildAudioStream, "audio_buildAudioStream", "Builds various audio conversion structures.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest6 = {
audio_buildAudioStreamNegative, "audio_buildAudioStreamNegative", "Checks calls with invalid input to SDL_CreateAudioStream", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest7 = {
audio_getAudioStatus, "audio_getAudioStatus", "Checks current audio status.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest8 = {
audio_openCloseAndGetAudioStatus, "audio_openCloseAndGetAudioStatus", "Opens and closes audio device and get audio status.", TEST_ENABLED
};
static const SDLTest_TestCaseReference audioTest9 = {
audio_lockUnlockOpenAudioDevice, "audio_lockUnlockOpenAudioDevice", "Locks and unlocks an open audio device.", TEST_ENABLED
};
/* TODO: enable test when SDL_ConvertAudio segfaults on cygwin have been fixed.
* TODO: re-check, since this was changer to AudioStream */
/* For debugging, test case can be run manually using --filter audio_convertAudio */
static const SDLTest_TestCaseReference audioTest10 = {
audio_convertAudio, "audio_convertAudio", "Convert audio using available formats.", TEST_DISABLED
};
/* 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
};
/* Sequence of Audio test cases */
static const SDLTest_TestCaseReference *audioTests[] = {
&audioTest1, &audioTest2, &audioTest3, &audioTest4, &audioTest5, &audioTest6,
&audioTest7, &audioTest8, &audioTest9, &audioTest10, &audioTest11,
&audioTest12, &audioTest13, &audioTest14, &audioTest15, &audioTest16, NULL
};
/* Audio test suite (global) */
SDLTest_TestSuiteReference audioTestSuite = {
"Audio",
audioSetUp,
audioTests,
audioTearDown
};