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Update for SDL3 coding style (#6717)

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I updated .clang-format and ran clang-format 14 over the src and test directories to standardize the code base.

In general I let clang-format have it's way, and added markup to prevent formatting of code that would break or be completely unreadable if formatted.

The script I ran for the src directory is added as build-scripts/clang-format-src.sh

This fixes:
#6592
#6593
#6594
This commit is contained in:
Sam Lantinga
2022-11-30 12:51:59 -08:00
committed by GitHub
parent 14b902faca
commit 5750bcb174
781 changed files with 51659 additions and 55763 deletions
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57
src/audio/emscripten/SDL_emscriptenaudio.c
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@@ -32,10 +32,10 @@
!!! FIXME: true always once pthread support becomes widespread. Revisit this code
!!! FIXME: at some point and see what needs to be done for that! */
static void
FeedAudioDevice(_THIS, const void *buf, const int buflen)
static void FeedAudioDevice(_THIS, const void *buf, const int buflen)
{
const int framelen = (SDL_AUDIO_BITSIZE(this->spec.format) / 8) * this->spec.channels;
/* *INDENT-OFF* */ /* clang-format off */
MAIN_THREAD_EM_ASM({
var SDL3 = Module['SDL3'];
var numChannels = SDL3.audio.currentOutputBuffer['numberOfChannels'];
@@ -50,10 +50,10 @@ FeedAudioDevice(_THIS, const void *buf, const int buflen)
}
}
}, buf, buflen / framelen);
/* *INDENT-ON* */ /* clang-format on */
}
static void
HandleAudioProcess(_THIS)
static void HandleAudioProcess(_THIS)
{
SDL_AudioCallback callback = this->callbackspec.callback;
const int stream_len = this->callbackspec.size;
@@ -69,12 +69,12 @@ HandleAudioProcess(_THIS)
return;
}
if (this->stream == NULL) { /* no conversion necessary. */
if (this->stream == NULL) { /* no conversion necessary. */
SDL_assert(this->spec.size == stream_len);
callback(this->callbackspec.userdata, this->work_buffer, stream_len);
} else { /* streaming/converting */
} else { /* streaming/converting */
int got;
while (SDL_AudioStreamAvailable(this->stream) < ((int) this->spec.size)) {
while (SDL_AudioStreamAvailable(this->stream) < ((int)this->spec.size)) {
callback(this->callbackspec.userdata, this->work_buffer, stream_len);
if (SDL_AudioStreamPut(this->stream, this->work_buffer, stream_len) == -1) {
SDL_AudioStreamClear(this->stream);
@@ -93,8 +93,7 @@ HandleAudioProcess(_THIS)
FeedAudioDevice(this, this->work_buffer, this->spec.size);
}
static void
HandleCaptureProcess(_THIS)
static void HandleCaptureProcess(_THIS)
{
SDL_AudioCallback callback = this->callbackspec.callback;
const int stream_len = this->callbackspec.size;
@@ -105,6 +104,7 @@ HandleCaptureProcess(_THIS)
return;
}
/* *INDENT-OFF* */ /* clang-format off */
MAIN_THREAD_EM_ASM({
var SDL3 = Module['SDL3'];
var numChannels = SDL3.capture.currentCaptureBuffer.numberOfChannels;
@@ -125,13 +125,14 @@ HandleCaptureProcess(_THIS)
}
}
}, this->work_buffer, (this->spec.size / sizeof (float)) / this->spec.channels);
/* *INDENT-ON* */ /* clang-format on */
/* okay, we've got an interleaved float32 array in C now. */
if (this->stream == NULL) { /* no conversion necessary. */
if (this->stream == NULL) { /* no conversion necessary. */
SDL_assert(this->spec.size == stream_len);
callback(this->callbackspec.userdata, this->work_buffer, stream_len);
} else { /* streaming/converting */
} else { /* streaming/converting */
if (SDL_AudioStreamPut(this->stream, this->work_buffer, this->spec.size) == -1) {
SDL_AtomicSet(&this->enabled, 0);
}
@@ -142,15 +143,14 @@ HandleCaptureProcess(_THIS)
if (got != stream_len) {
SDL_memset(this->work_buffer, this->callbackspec.silence, stream_len);
}
callback(this->callbackspec.userdata, this->work_buffer, stream_len); /* Send it to the app. */
callback(this->callbackspec.userdata, this->work_buffer, stream_len); /* Send it to the app. */
}
}
}
static void
EMSCRIPTENAUDIO_CloseDevice(_THIS)
static void EMSCRIPTENAUDIO_CloseDevice(_THIS)
{
/* *INDENT-OFF* */ /* clang-format off */
MAIN_THREAD_EM_ASM({
var SDL3 = Module['SDL3'];
if ($0) {
@@ -189,14 +189,14 @@ EMSCRIPTENAUDIO_CloseDevice(_THIS)
SDL3.audioContext = undefined;
}
}, this->iscapture);
/* *INDENT-ON* */ /* clang-format on */
#if 0 /* !!! FIXME: currently not used. Can we move some stuff off the SDL3 namespace? --ryan. */
#if 0 /* !!! FIXME: currently not used. Can we move some stuff off the SDL3 namespace? --ryan. */
SDL_free(this->hidden);
#endif
}
static int
EMSCRIPTENAUDIO_OpenDevice(_THIS, const char *devname)
static int EMSCRIPTENAUDIO_OpenDevice(_THIS, const char *devname)
{
SDL_AudioFormat test_format;
SDL_bool iscapture = this->iscapture;
@@ -204,6 +204,7 @@ EMSCRIPTENAUDIO_OpenDevice(_THIS, const char *devname)
/* based on parts of library_sdl.js */
/* *INDENT-OFF* */ /* clang-format off */
/* create context */
result = MAIN_THREAD_EM_ASM_INT({
if (typeof(Module['SDL3']) === 'undefined') {
@@ -228,6 +229,8 @@ EMSCRIPTENAUDIO_OpenDevice(_THIS, const char *devname)
}
return SDL3.audioContext === undefined ? -1 : 0;
}, iscapture);
/* *INDENT-ON* */ /* clang-format on */
if (result < 0) {
return SDL_SetError("Web Audio API is not available!");
}
@@ -249,7 +252,7 @@ EMSCRIPTENAUDIO_OpenDevice(_THIS, const char *devname)
this->spec.format = test_format;
/* Initialize all variables that we clean on shutdown */
#if 0 /* !!! FIXME: currently not used. Can we move some stuff off the SDL3 namespace? --ryan. */
#if 0 /* !!! FIXME: currently not used. Can we move some stuff off the SDL3 namespace? --ryan. */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
@@ -261,12 +264,13 @@ EMSCRIPTENAUDIO_OpenDevice(_THIS, const char *devname)
/* limit to native freq */
this->spec.freq = EM_ASM_INT_V({
var SDL3 = Module['SDL3'];
return SDL3.audioContext.sampleRate;
var SDL3 = Module['SDL3'];
return SDL3.audioContext.sampleRate;
});
SDL_CalculateAudioSpec(&this->spec);
/* *INDENT-OFF* */ /* clang-format off */
if (iscapture) {
/* The idea is to take the capture media stream, hook it up to an
audio graph where we can pass it through a ScriptProcessorNode
@@ -338,17 +342,16 @@ EMSCRIPTENAUDIO_OpenDevice(_THIS, const char *devname)
SDL3.audio.scriptProcessorNode['connect'](SDL3.audioContext['destination']);
}, this->spec.channels, this->spec.samples, HandleAudioProcess, this);
}
/* *INDENT-ON* */ /* clang-format on */
return 0;
}
static void
EMSCRIPTENAUDIO_LockOrUnlockDeviceWithNoMixerLock(SDL_AudioDevice * device)
static void EMSCRIPTENAUDIO_LockOrUnlockDeviceWithNoMixerLock(SDL_AudioDevice *device)
{
}
static SDL_bool
EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl * impl)
static SDL_bool EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl *impl)
{
SDL_bool available, capture_available;
@@ -362,6 +365,7 @@ EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl * impl)
impl->LockDevice = impl->UnlockDevice = EMSCRIPTENAUDIO_LockOrUnlockDeviceWithNoMixerLock;
impl->ProvidesOwnCallbackThread = SDL_TRUE;
/* *INDENT-OFF* */ /* clang-format off */
/* check availability */
available = MAIN_THREAD_EM_ASM_INT({
if (typeof(AudioContext) !== 'undefined') {
@@ -371,11 +375,13 @@ EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl * impl)
}
return false;
});
/* *INDENT-ON* */ /* clang-format on */
if (!available) {
SDL_SetError("No audio context available");
}
/* *INDENT-OFF* */ /* clang-format off */
capture_available = available && MAIN_THREAD_EM_ASM_INT({
if ((typeof(navigator.mediaDevices) !== 'undefined') && (typeof(navigator.mediaDevices.getUserMedia) !== 'undefined')) {
return true;
@@ -384,6 +390,7 @@ EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl * impl)
}
return false;
});
/* *INDENT-ON* */ /* clang-format on */
impl->HasCaptureSupport = capture_available ? SDL_TRUE : SDL_FALSE;
impl->OnlyHasDefaultCaptureDevice = capture_available ? SDL_TRUE : SDL_FALSE;