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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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415
src/audio/SDL_audio.c
View File

@@ -100,22 +100,20 @@ static const AudioBootStrap *const bootstrap[] = {
NULL
};
#ifdef HAVE_LIBSAMPLERATE_H
#ifdef SDL_LIBSAMPLERATE_DYNAMIC
static void *SRC_lib = NULL;
#endif
SDL_bool SRC_available = SDL_FALSE;
int SRC_converter = 0;
SRC_STATE* (*SRC_src_new)(int converter_type, int channels, int *error) = NULL;
SRC_STATE *(*SRC_src_new)(int converter_type, int channels, int *error) = NULL;
int (*SRC_src_process)(SRC_STATE *state, SRC_DATA *data) = NULL;
int (*SRC_src_reset)(SRC_STATE *state) = NULL;
SRC_STATE* (*SRC_src_delete)(SRC_STATE *state) = NULL;
const char* (*SRC_src_strerror)(int error) = NULL;
SRC_STATE *(*SRC_src_delete)(SRC_STATE *state) = NULL;
const char *(*SRC_src_strerror)(int error) = NULL;
int (*SRC_src_simple)(SRC_DATA *data, int converter_type, int channels) = NULL;
static SDL_bool
LoadLibSampleRate(void)
static SDL_bool LoadLibSampleRate(void)
{
const char *hint = SDL_GetHint(SDL_HINT_AUDIO_RESAMPLING_MODE);
@@ -123,7 +121,7 @@ LoadLibSampleRate(void)
SRC_converter = 0;
if (!hint || *hint == '0' || SDL_strcasecmp(hint, "default") == 0) {
return SDL_FALSE; /* don't load anything. */
return SDL_FALSE; /* don't load anything. */
} else if (*hint == '1' || SDL_strcasecmp(hint, "fast") == 0) {
SRC_converter = SRC_SINC_FASTEST;
} else if (*hint == '2' || SDL_strcasecmp(hint, "medium") == 0) {
@@ -131,7 +129,7 @@ LoadLibSampleRate(void)
} else if (*hint == '3' || SDL_strcasecmp(hint, "best") == 0) {
SRC_converter = SRC_SINC_BEST_QUALITY;
} else {
return SDL_FALSE; /* treat it like "default", don't load anything. */
return SDL_FALSE; /* treat it like "default", don't load anything. */
}
#ifdef SDL_LIBSAMPLERATE_DYNAMIC
@@ -142,12 +140,14 @@ LoadLibSampleRate(void)
return SDL_FALSE;
}
/* *INDENT-OFF* */ /* clang-format off */
SRC_src_new = (SRC_STATE* (*)(int converter_type, int channels, int *error))SDL_LoadFunction(SRC_lib, "src_new");
SRC_src_process = (int (*)(SRC_STATE *state, SRC_DATA *data))SDL_LoadFunction(SRC_lib, "src_process");
SRC_src_reset = (int(*)(SRC_STATE *state))SDL_LoadFunction(SRC_lib, "src_reset");
SRC_src_delete = (SRC_STATE* (*)(SRC_STATE *state))SDL_LoadFunction(SRC_lib, "src_delete");
SRC_src_strerror = (const char* (*)(int error))SDL_LoadFunction(SRC_lib, "src_strerror");
SRC_src_simple = (int(*)(SRC_DATA *data, int converter_type, int channels))SDL_LoadFunction(SRC_lib, "src_simple");
/* *INDENT-ON* */ /* clang-format on */
if (!SRC_src_new || !SRC_src_process || !SRC_src_reset || !SRC_src_delete || !SRC_src_strerror || !SRC_src_simple) {
SDL_UnloadObject(SRC_lib);
@@ -167,8 +167,7 @@ LoadLibSampleRate(void)
return SDL_TRUE;
}
static void
UnloadLibSampleRate(void)
static void UnloadLibSampleRate(void)
{
#ifdef SDL_LIBSAMPLERATE_DYNAMIC
if (SRC_lib != NULL) {
@@ -186,8 +185,7 @@ UnloadLibSampleRate(void)
}
#endif
static SDL_AudioDevice *
get_audio_device(SDL_AudioDeviceID id)
static SDL_AudioDevice *get_audio_device(SDL_AudioDeviceID id)
{
id--;
if ((id >= SDL_arraysize(open_devices)) || (open_devices[id] == NULL)) {
@@ -198,82 +196,67 @@ get_audio_device(SDL_AudioDeviceID id)
return open_devices[id];
}
/* stubs for audio drivers that don't need a specific entry point... */
static void
SDL_AudioDetectDevices_Default(void)
static void SDL_AudioDetectDevices_Default(void)
{
/* you have to write your own implementation if these assertions fail. */
SDL_assert(current_audio.impl.OnlyHasDefaultOutputDevice);
SDL_assert(current_audio.impl.OnlyHasDefaultCaptureDevice || !current_audio.impl.HasCaptureSupport);
SDL_AddAudioDevice(SDL_FALSE, DEFAULT_OUTPUT_DEVNAME, NULL, (void *) ((size_t) 0x1));
SDL_AddAudioDevice(SDL_FALSE, DEFAULT_OUTPUT_DEVNAME, NULL, (void *)((size_t)0x1));
if (current_audio.impl.HasCaptureSupport) {
SDL_AddAudioDevice(SDL_TRUE, DEFAULT_INPUT_DEVNAME, NULL, (void *) ((size_t) 0x2));
SDL_AddAudioDevice(SDL_TRUE, DEFAULT_INPUT_DEVNAME, NULL, (void *)((size_t)0x2));
}
}
static void
SDL_AudioThreadInit_Default(_THIS)
{ /* no-op. */
static void SDL_AudioThreadInit_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioThreadDeinit_Default(_THIS)
{ /* no-op. */
static void SDL_AudioThreadDeinit_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioWaitDevice_Default(_THIS)
{ /* no-op. */
static void SDL_AudioWaitDevice_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioPlayDevice_Default(_THIS)
{ /* no-op. */
static void SDL_AudioPlayDevice_Default(_THIS)
{ /* no-op. */
}
static Uint8 *
SDL_AudioGetDeviceBuf_Default(_THIS)
static Uint8 *SDL_AudioGetDeviceBuf_Default(_THIS)
{
return NULL;
}
static int
SDL_AudioCaptureFromDevice_Default(_THIS, void *buffer, int buflen)
static int SDL_AudioCaptureFromDevice_Default(_THIS, void *buffer, int buflen)
{
return -1; /* just fail immediately. */
return -1; /* just fail immediately. */
}
static void
SDL_AudioFlushCapture_Default(_THIS)
{ /* no-op. */
static void SDL_AudioFlushCapture_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioCloseDevice_Default(_THIS)
{ /* no-op. */
static void SDL_AudioCloseDevice_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioDeinitialize_Default(void)
{ /* no-op. */
static void SDL_AudioDeinitialize_Default(void)
{ /* no-op. */
}
static void
SDL_AudioFreeDeviceHandle_Default(void *handle)
{ /* no-op. */
static void SDL_AudioFreeDeviceHandle_Default(void *handle)
{ /* no-op. */
}
static int
SDL_AudioOpenDevice_Default(_THIS, const char *devname)
static int SDL_AudioOpenDevice_Default(_THIS, const char *devname)
{
return SDL_Unsupported();
}
static SDL_INLINE SDL_bool
is_in_audio_device_thread(SDL_AudioDevice * device)
static SDL_INLINE SDL_bool is_in_audio_device_thread(SDL_AudioDevice *device)
{
/* The device thread locks the same mutex, but not through the public API.
This check is in case the application, in the audio callback,
@@ -286,35 +269,31 @@ is_in_audio_device_thread(SDL_AudioDevice * device)
return SDL_FALSE;
}
static void
SDL_AudioLockDevice_Default(SDL_AudioDevice * device)
static void SDL_AudioLockDevice_Default(SDL_AudioDevice *device)
{
if (!is_in_audio_device_thread(device)) {
SDL_LockMutex(device->mixer_lock);
}
}
static void
SDL_AudioUnlockDevice_Default(SDL_AudioDevice * device)
static void SDL_AudioUnlockDevice_Default(SDL_AudioDevice *device)
{
if (!is_in_audio_device_thread(device)) {
SDL_UnlockMutex(device->mixer_lock);
}
}
static void
finish_audio_entry_points_init(void)
static void finish_audio_entry_points_init(void)
{
/*
* Fill in stub functions for unused driver entry points. This lets us
* blindly call them without having to check for validity first.
*/
#define FILL_STUB(x) \
if (current_audio.impl.x == NULL) { \
current_audio.impl.x = SDL_Audio##x##_Default; \
}
#define FILL_STUB(x) \
if (current_audio.impl.x == NULL) { \
current_audio.impl.x = SDL_Audio##x##_Default; \
}
FILL_STUB(DetectDevices);
FILL_STUB(OpenDevice);
FILL_STUB(ThreadInit);
@@ -332,21 +311,19 @@ finish_audio_entry_points_init(void)
#undef FILL_STUB
}
/* device hotplug support... */
static int
add_audio_device(const char *name, SDL_AudioSpec *spec, void *handle, SDL_AudioDeviceItem **devices, int *devCount)
static int add_audio_device(const char *name, SDL_AudioSpec *spec, void *handle, SDL_AudioDeviceItem **devices, int *devCount)
{
int retval = -1;
SDL_AudioDeviceItem *item;
const SDL_AudioDeviceItem *i;
int dupenum = 0;
SDL_assert(handle != NULL); /* we reserve NULL, audio backends can't use it. */
SDL_assert(handle != NULL); /* we reserve NULL, audio backends can't use it. */
SDL_assert(name != NULL);
item = (SDL_AudioDeviceItem *) SDL_malloc(sizeof (SDL_AudioDeviceItem));
item = (SDL_AudioDeviceItem *)SDL_malloc(sizeof(SDL_AudioDeviceItem));
if (!item) {
return SDL_OutOfMemory();
}
@@ -371,13 +348,13 @@ add_audio_device(const char *name, SDL_AudioSpec *spec, void *handle, SDL_AudioD
for (i = *devices; i != NULL; i = i->next) {
if (SDL_strcmp(name, i->original_name) == 0) {
dupenum = i->dupenum + 1;
break; /* stop at the highest-numbered dupe. */
break; /* stop at the highest-numbered dupe. */
}
}
if (dupenum) {
const size_t len = SDL_strlen(name) + 16;
char *replacement = (char *) SDL_malloc(len);
char *replacement = (char *)SDL_malloc(len);
if (!replacement) {
SDL_UnlockMutex(current_audio.detectionLock);
SDL_free(item->original_name);
@@ -392,28 +369,25 @@ add_audio_device(const char *name, SDL_AudioSpec *spec, void *handle, SDL_AudioD
item->next = *devices;
*devices = item;
retval = (*devCount)++; /* !!! FIXME: this should be an atomic increment */
retval = (*devCount)++; /* !!! FIXME: this should be an atomic increment */
SDL_UnlockMutex(current_audio.detectionLock);
return retval;
}
static SDL_INLINE int
add_capture_device(const char *name, SDL_AudioSpec *spec, void *handle)
static SDL_INLINE int add_capture_device(const char *name, SDL_AudioSpec *spec, void *handle)
{
SDL_assert(current_audio.impl.HasCaptureSupport);
return add_audio_device(name, spec, handle, &current_audio.inputDevices, &current_audio.inputDeviceCount);
}
static SDL_INLINE int
add_output_device(const char *name, SDL_AudioSpec *spec, void *handle)
static SDL_INLINE int add_output_device(const char *name, SDL_AudioSpec *spec, void *handle)
{
return add_audio_device(name, spec, handle, &current_audio.outputDevices, &current_audio.outputDeviceCount);
}
static void
free_device_list(SDL_AudioDeviceItem **devices, int *devCount)
static void free_device_list(SDL_AudioDeviceItem **devices, int *devCount)
{
SDL_AudioDeviceItem *item, *next;
for (item = *devices; item != NULL; item = next) {
@@ -432,10 +406,8 @@ free_device_list(SDL_AudioDeviceItem **devices, int *devCount)
*devCount = 0;
}
/* The audio backends call this when a new device is plugged in. */
void
SDL_AddAudioDevice(const SDL_bool iscapture, const char *name, SDL_AudioSpec *spec, void *handle)
void SDL_AddAudioDevice(const SDL_bool iscapture, const char *name, SDL_AudioSpec *spec, void *handle)
{
const int device_index = iscapture ? add_capture_device(name, spec, handle) : add_output_device(name, spec, handle);
if (device_index != -1) {
@@ -457,11 +429,11 @@ void SDL_OpenedAudioDeviceDisconnected(SDL_AudioDevice *device)
SDL_assert(get_audio_device(device->id) == device);
if (!SDL_AtomicGet(&device->enabled)) {
return; /* don't report disconnects more than once. */
return; /* don't report disconnects more than once. */
}
if (SDL_AtomicGet(&device->shutdown)) {
return; /* don't report disconnect if we're trying to close device. */
return; /* don't report disconnect if we're trying to close device. */
}
/* Ends the audio callback and mark the device as STOPPED, but the
@@ -481,8 +453,7 @@ void SDL_OpenedAudioDeviceDisconnected(SDL_AudioDevice *device)
}
}
static void
mark_device_removed(void *handle, SDL_AudioDeviceItem *devices, SDL_bool *removedFlag)
static void mark_device_removed(void *handle, SDL_AudioDeviceItem *devices, SDL_bool *removedFlag)
{
SDL_AudioDeviceItem *item;
SDL_assert(handle != NULL);
@@ -496,8 +467,7 @@ mark_device_removed(void *handle, SDL_AudioDeviceItem *devices, SDL_bool *remove
}
/* The audio backends call this when a device is removed from the system. */
void
SDL_RemoveAudioDevice(const SDL_bool iscapture, void *handle)
void SDL_RemoveAudioDevice(const SDL_bool iscapture, void *handle)
{
int device_index;
SDL_AudioDevice *device = NULL;
@@ -540,40 +510,36 @@ SDL_RemoveAudioDevice(const SDL_bool iscapture, void *handle)
current_audio.impl.FreeDeviceHandle(handle);
}
/* buffer queueing support... */
static void SDLCALL
SDL_BufferQueueDrainCallback(void *userdata, Uint8 *stream, int len)
static void SDLCALL SDL_BufferQueueDrainCallback(void *userdata, Uint8 *stream, int len)
{
/* this function always holds the mixer lock before being called. */
SDL_AudioDevice *device = (SDL_AudioDevice *) userdata;
SDL_AudioDevice *device = (SDL_AudioDevice *)userdata;
size_t dequeued;
SDL_assert(device != NULL); /* this shouldn't ever happen, right?! */
SDL_assert(!device->iscapture); /* this shouldn't ever happen, right?! */
SDL_assert(len >= 0); /* this shouldn't ever happen, right?! */
SDL_assert(device != NULL); /* this shouldn't ever happen, right?! */
SDL_assert(!device->iscapture); /* this shouldn't ever happen, right?! */
SDL_assert(len >= 0); /* this shouldn't ever happen, right?! */
dequeued = SDL_ReadFromDataQueue(device->buffer_queue, stream, len);
stream += dequeued;
len -= (int) dequeued;
len -= (int)dequeued;
if (len > 0) { /* fill any remaining space in the stream with silence. */
if (len > 0) { /* fill any remaining space in the stream with silence. */
SDL_assert(SDL_CountDataQueue(device->buffer_queue) == 0);
SDL_memset(stream, device->callbackspec.silence, len);
}
}
static void SDLCALL
SDL_BufferQueueFillCallback(void *userdata, Uint8 *stream, int len)
static void SDLCALL SDL_BufferQueueFillCallback(void *userdata, Uint8 *stream, int len)
{
/* this function always holds the mixer lock before being called. */
SDL_AudioDevice *device = (SDL_AudioDevice *) userdata;
SDL_AudioDevice *device = (SDL_AudioDevice *)userdata;
SDL_assert(device != NULL); /* this shouldn't ever happen, right?! */
SDL_assert(device->iscapture); /* this shouldn't ever happen, right?! */
SDL_assert(len >= 0); /* this shouldn't ever happen, right?! */
SDL_assert(device != NULL); /* this shouldn't ever happen, right?! */
SDL_assert(device->iscapture); /* this shouldn't ever happen, right?! */
SDL_assert(len >= 0); /* this shouldn't ever happen, right?! */
/* note that if this needs to allocate more space and run out of memory,
we have no choice but to quietly drop the data and hope it works out
@@ -581,14 +547,13 @@ SDL_BufferQueueFillCallback(void *userdata, Uint8 *stream, int len)
SDL_WriteToDataQueue(device->buffer_queue, stream, len);
}
int
SDL_QueueAudio(SDL_AudioDeviceID devid, const void *data, Uint32 len)
int SDL_QueueAudio(SDL_AudioDeviceID devid, const void *data, Uint32 len)
{
SDL_AudioDevice *device = get_audio_device(devid);
int rc = 0;
if (!device) {
return -1; /* get_audio_device() will have set the error state */
return -1; /* get_audio_device() will have set the error state */
} else if (device->iscapture) {
return SDL_SetError("This is a capture device, queueing not allowed");
} else if (device->callbackspec.callback != SDL_BufferQueueDrainCallback) {
@@ -610,15 +575,15 @@ SDL_DequeueAudio(SDL_AudioDeviceID devid, void *data, Uint32 len)
SDL_AudioDevice *device = get_audio_device(devid);
Uint32 rc;
if ( (len == 0) || /* nothing to do? */
(!device) || /* called with bogus device id */
(!device->iscapture) || /* playback devices can't dequeue */
(device->callbackspec.callback != SDL_BufferQueueFillCallback) ) { /* not set for queueing */
return 0; /* just report zero bytes dequeued. */
if ((len == 0) || /* nothing to do? */
(!device) || /* called with bogus device id */
(!device->iscapture) || /* playback devices can't dequeue */
(device->callbackspec.callback != SDL_BufferQueueFillCallback)) { /* not set for queueing */
return 0; /* just report zero bytes dequeued. */
}
current_audio.impl.LockDevice(device);
rc = (Uint32) SDL_ReadFromDataQueue(device->buffer_queue, data, len);
rc = (Uint32)SDL_ReadFromDataQueue(device->buffer_queue, data, len);
current_audio.impl.UnlockDevice(device);
return rc;
}
@@ -635,23 +600,21 @@ SDL_GetQueuedAudioSize(SDL_AudioDeviceID devid)
/* Nothing to do unless we're set up for queueing. */
if (device->callbackspec.callback == SDL_BufferQueueDrainCallback ||
device->callbackspec.callback == SDL_BufferQueueFillCallback)
{
device->callbackspec.callback == SDL_BufferQueueFillCallback) {
current_audio.impl.LockDevice(device);
retval = (Uint32) SDL_CountDataQueue(device->buffer_queue);
retval = (Uint32)SDL_CountDataQueue(device->buffer_queue);
current_audio.impl.UnlockDevice(device);
}
return retval;
}
void
SDL_ClearQueuedAudio(SDL_AudioDeviceID devid)
void SDL_ClearQueuedAudio(SDL_AudioDeviceID devid)
{
SDL_AudioDevice *device = get_audio_device(devid);
if (!device) {
return; /* nothing to do. */
return; /* nothing to do. */
}
/* Blank out the device and release the mutex. Free it afterwards. */
@@ -663,12 +626,10 @@ SDL_ClearQueuedAudio(SDL_AudioDeviceID devid)
current_audio.impl.UnlockDevice(device);
}
/* The general mixing thread function */
static int SDLCALL
SDL_RunAudio(void *devicep)
static int SDLCALL SDL_RunAudio(void *devicep)
{
SDL_AudioDevice *device = (SDL_AudioDevice *) devicep;
SDL_AudioDevice *device = (SDL_AudioDevice *)devicep;
void *udata = device->callbackspec.userdata;
SDL_AudioCallback callback = device->callbackspec.callback;
int data_len = 0;
@@ -727,15 +688,15 @@ SDL_RunAudio(void *devicep)
/* if this fails...oh well. We'll play silence here. */
SDL_AudioStreamPut(device->stream, data, data_len);
while (SDL_AudioStreamAvailable(device->stream) >= ((int) device->spec.size)) {
while (SDL_AudioStreamAvailable(device->stream) >= ((int)device->spec.size)) {
int got;
data = SDL_AtomicGet(&device->enabled) ? current_audio.impl.GetDeviceBuf(device) : NULL;
got = SDL_AudioStreamGet(device->stream, data ? data : device->work_buffer, device->spec.size);
SDL_assert((got <= 0) || (got == device->spec.size));
if (data == NULL) { /* device is having issues... */
if (data == NULL) { /* device is having issues... */
const Uint32 delay = ((device->spec.samples * 1000) / device->spec.freq);
SDL_Delay(delay); /* wait for as long as this buffer would have played. Maybe device recovers later? */
SDL_Delay(delay); /* wait for as long as this buffer would have played. Maybe device recovers later? */
} else {
if (got != device->spec.size) {
SDL_memset(data, device->spec.silence, device->spec.size);
@@ -748,7 +709,7 @@ SDL_RunAudio(void *devicep)
/* nothing to do; pause like we queued a buffer to play. */
const Uint32 delay = ((device->spec.samples * 1000) / device->spec.freq);
SDL_Delay(delay);
} else { /* writing directly to the device. */
} else { /* writing directly to the device. */
/* queue this buffer and wait for it to finish playing. */
current_audio.impl.PlayDevice(device);
current_audio.impl.WaitDevice(device);
@@ -765,11 +726,10 @@ SDL_RunAudio(void *devicep)
/* !!! FIXME: this needs to deal with device spec changes. */
/* The general capture thread function */
static int SDLCALL
SDL_CaptureAudio(void *devicep)
static int SDLCALL SDL_CaptureAudio(void *devicep)
{
SDL_AudioDevice *device = (SDL_AudioDevice *) devicep;
const int silence = (int) device->spec.silence;
SDL_AudioDevice *device = (SDL_AudioDevice *)devicep;
const int silence = (int)device->spec.silence;
const Uint32 delay = ((device->spec.samples * 1000) / device->spec.freq);
const int data_len = device->spec.size;
Uint8 *data;
@@ -799,11 +759,11 @@ SDL_CaptureAudio(void *devicep)
Uint8 *ptr;
if (SDL_AtomicGet(&device->paused)) {
SDL_Delay(delay); /* just so we don't cook the CPU. */
SDL_Delay(delay); /* just so we don't cook the CPU. */
if (device->stream) {
SDL_AudioStreamClear(device->stream);
}
current_audio.impl.FlushCapture(device); /* dump anything pending. */
current_audio.impl.FlushCapture(device); /* dump anything pending. */
continue;
}
@@ -821,15 +781,15 @@ SDL_CaptureAudio(void *devicep)
But we don't process it further or call the app's callback. */
if (!SDL_AtomicGet(&device->enabled)) {
SDL_Delay(delay); /* try to keep callback firing at normal pace. */
SDL_Delay(delay); /* try to keep callback firing at normal pace. */
} else {
while (still_need > 0) {
const int rc = current_audio.impl.CaptureFromDevice(device, ptr, still_need);
SDL_assert(rc <= still_need); /* device should not overflow buffer. :) */
SDL_assert(rc <= still_need); /* device should not overflow buffer. :) */
if (rc > 0) {
still_need -= rc;
ptr += rc;
} else { /* uhoh, device failed for some reason! */
} else { /* uhoh, device failed for some reason! */
SDL_OpenedAudioDeviceDisconnected(device);
break;
}
@@ -845,7 +805,7 @@ SDL_CaptureAudio(void *devicep)
/* if this fails...oh well. */
SDL_AudioStreamPut(device->stream, data, data_len);
while (SDL_AudioStreamAvailable(device->stream) >= ((int) device->callbackspec.size)) {
while (SDL_AudioStreamAvailable(device->stream) >= ((int)device->callbackspec.size)) {
const int got = SDL_AudioStreamGet(device->stream, device->work_buffer, device->callbackspec.size);
SDL_assert((got < 0) || (got == device->callbackspec.size));
if (got != device->callbackspec.size) {
@@ -859,7 +819,7 @@ SDL_CaptureAudio(void *devicep)
}
SDL_UnlockMutex(device->mixer_lock);
}
} else { /* feeding user callback directly without streaming. */
} else { /* feeding user callback directly without streaming. */
/* !!! FIXME: this should be LockDevice. */
SDL_LockMutex(device->mixer_lock);
if (!SDL_AtomicGet(&device->paused)) {
@@ -876,11 +836,11 @@ SDL_CaptureAudio(void *devicep)
return 0;
}
static SDL_AudioFormat
SDL_ParseAudioFormat(const char *string)
static SDL_AudioFormat SDL_ParseAudioFormat(const char *string)
{
#define CHECK_FMT_STRING(x) if (SDL_strcmp(string, #x) == 0) return AUDIO_##x
#define CHECK_FMT_STRING(x) \
if (SDL_strcmp(string, #x) == 0) \
return AUDIO_##x
CHECK_FMT_STRING(U8);
CHECK_FMT_STRING(S8);
CHECK_FMT_STRING(U16LSB);
@@ -903,8 +863,7 @@ SDL_ParseAudioFormat(const char *string)
return 0;
}
int
SDL_GetNumAudioDrivers(void)
int SDL_GetNumAudioDrivers(void)
{
return SDL_arraysize(bootstrap) - 1;
}
@@ -918,14 +877,13 @@ SDL_GetAudioDriver(int index)
return NULL;
}
int
SDL_AudioInit(const char *driver_name)
int SDL_AudioInit(const char *driver_name)
{
int i;
SDL_bool initialized = SDL_FALSE, tried_to_init = SDL_FALSE;
if (SDL_GetCurrentAudioDriver()) {
SDL_AudioQuit(); /* shutdown driver if already running. */
SDL_AudioQuit(); /* shutdown driver if already running. */
}
SDL_zeroa(open_devices);
@@ -998,7 +956,7 @@ SDL_AudioInit(const char *driver_name)
}
SDL_zero(current_audio);
return -1; /* No driver was available, so fail. */
return -1; /* No driver was available, so fail. */
}
current_audio.detectionLock = SDL_CreateMutex();
@@ -1025,8 +983,7 @@ SDL_GetCurrentAudioDriver()
}
/* Clean out devices that we've removed but had to keep around for stability. */
static void
clean_out_device_list(SDL_AudioDeviceItem **devices, int *devCount, SDL_bool *removedFlag)
static void clean_out_device_list(SDL_AudioDeviceItem **devices, int *devCount, SDL_bool *removedFlag)
{
SDL_AudioDeviceItem *item = *devices;
SDL_AudioDeviceItem *prev = NULL;
@@ -1057,9 +1014,7 @@ clean_out_device_list(SDL_AudioDeviceItem **devices, int *devCount, SDL_bool *re
*removedFlag = SDL_FALSE;
}
int
SDL_GetNumAudioDevices(int iscapture)
int SDL_GetNumAudioDevices(int iscapture)
{
int retval = 0;
@@ -1082,7 +1037,6 @@ SDL_GetNumAudioDevices(int iscapture)
return retval;
}
const char *
SDL_GetAudioDeviceName(int index, int iscapture)
{
@@ -1113,9 +1067,7 @@ SDL_GetAudioDeviceName(int index, int iscapture)
return retval;
}
int
SDL_GetAudioDeviceSpec(int index, int iscapture, SDL_AudioSpec *spec)
int SDL_GetAudioDeviceSpec(int index, int iscapture, SDL_AudioSpec *spec)
{
SDL_AudioDeviceItem *item;
int i, retval;
@@ -1146,9 +1098,7 @@ SDL_GetAudioDeviceSpec(int index, int iscapture, SDL_AudioSpec *spec)
return retval;
}
int
SDL_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
int SDL_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
{
if (spec == NULL) {
return SDL_InvalidParamError("spec");
@@ -1164,9 +1114,7 @@ SDL_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
return current_audio.impl.GetDefaultAudioInfo(name, spec, iscapture);
}
static void
close_audio_device(SDL_AudioDevice * device)
static void close_audio_device(SDL_AudioDevice *device)
{
if (!device) {
return;
@@ -1207,34 +1155,32 @@ close_audio_device(SDL_AudioDevice * device)
SDL_free(device);
}
/*
* Sanity check desired AudioSpec for SDL_OpenAudio() in (orig).
* Fills in a sanitized copy in (prepared).
* Returns non-zero if okay, zero on fatal parameters in (orig).
*/
static int
prepare_audiospec(const SDL_AudioSpec * orig, SDL_AudioSpec * prepared)
static int prepare_audiospec(const SDL_AudioSpec *orig, SDL_AudioSpec *prepared)
{
SDL_copyp(prepared, orig);
if (orig->freq == 0) {
const char *env = SDL_getenv("SDL_AUDIO_FREQUENCY");
if ((!env) || ((prepared->freq = SDL_atoi(env)) == 0)) {
prepared->freq = 22050; /* a reasonable default */
prepared->freq = 22050; /* a reasonable default */
}
}
if (orig->format == 0) {
const char *env = SDL_getenv("SDL_AUDIO_FORMAT");
if ((!env) || ((prepared->format = SDL_ParseAudioFormat(env)) == 0)) {
prepared->format = AUDIO_S16; /* a reasonable default */
prepared->format = AUDIO_S16; /* a reasonable default */
}
}
if (orig->channels == 0) {
const char *env = SDL_getenv("SDL_AUDIO_CHANNELS");
if ((!env) || ((prepared->channels = (Uint8) SDL_atoi(env)) == 0)) {
if ((!env) || ((prepared->channels = (Uint8)SDL_atoi(env)) == 0)) {
prepared->channels = 2; /* a reasonable default */
}
} else if (orig->channels > 8) {
@@ -1244,7 +1190,7 @@ prepare_audiospec(const SDL_AudioSpec * orig, SDL_AudioSpec * prepared)
if (orig->samples == 0) {
const char *env = SDL_getenv("SDL_AUDIO_SAMPLES");
if ((!env) || ((prepared->samples = (Uint16) SDL_atoi(env)) == 0)) {
if ((!env) || ((prepared->samples = (Uint16)SDL_atoi(env)) == 0)) {
/* Pick a default of ~46 ms at desired frequency */
/* !!! FIXME: remove this when the non-Po2 resampling is in. */
const int samples = (prepared->freq / 1000) * 46;
@@ -1262,10 +1208,9 @@ prepare_audiospec(const SDL_AudioSpec * orig, SDL_AudioSpec * prepared)
return 1;
}
static SDL_AudioDeviceID
open_audio_device(const char *devname, int iscapture,
const SDL_AudioSpec * desired, SDL_AudioSpec * obtained,
int allowed_changes, int min_id)
static SDL_AudioDeviceID open_audio_device(const char *devname, int iscapture,
const SDL_AudioSpec *desired, SDL_AudioSpec *obtained,
int allowed_changes, int min_id)
{
const SDL_bool is_internal_thread = (desired->callback == NULL);
SDL_AudioDeviceID id = 0;
@@ -1377,7 +1322,7 @@ open_audio_device(const char *devname, int iscapture,
}
}
device = (SDL_AudioDevice *) SDL_calloc(1, sizeof (SDL_AudioDevice));
device = (SDL_AudioDevice *)SDL_calloc(1, sizeof(SDL_AudioDevice));
if (device == NULL) {
SDL_OutOfMemory();
SDL_UnlockMutex(current_audio.detectionLock);
@@ -1388,7 +1333,7 @@ open_audio_device(const char *devname, int iscapture,
device->iscapture = iscapture ? SDL_TRUE : SDL_FALSE;
device->handle = handle;
SDL_AtomicSet(&device->shutdown, 0); /* just in case. */
SDL_AtomicSet(&device->shutdown, 0); /* just in case. */
SDL_AtomicSet(&device->paused, 1);
SDL_AtomicSet(&device->enabled, 1);
@@ -1451,19 +1396,19 @@ open_audio_device(const char *devname, int iscapture,
}
}
SDL_CalculateAudioSpec(obtained); /* recalc after possible changes. */
SDL_CalculateAudioSpec(obtained); /* recalc after possible changes. */
device->callbackspec = *obtained;
if (build_stream) {
if (iscapture) {
device->stream = SDL_NewAudioStream(device->spec.format,
device->spec.channels, device->spec.freq,
obtained->format, obtained->channels, obtained->freq);
device->spec.channels, device->spec.freq,
obtained->format, obtained->channels, obtained->freq);
} else {
device->stream = SDL_NewAudioStream(obtained->format, obtained->channels,
obtained->freq, device->spec.format,
device->spec.channels, device->spec.freq);
obtained->freq, device->spec.format,
device->spec.channels, device->spec.freq);
}
if (!device->stream) {
@@ -1473,7 +1418,7 @@ open_audio_device(const char *devname, int iscapture,
}
}
if (device->spec.callback == NULL) { /* use buffer queueing? */
if (device->spec.callback == NULL) { /* use buffer queueing? */
/* pool a few packets to start. Enough for two callbacks. */
device->buffer_queue = SDL_NewDataQueue(SDL_AUDIOBUFFERQUEUE_PACKETLEN, obtained->size * 2);
if (!device->buffer_queue) {
@@ -1493,7 +1438,7 @@ open_audio_device(const char *devname, int iscapture,
}
SDL_assert(device->work_buffer_len > 0);
device->work_buffer = (Uint8 *) SDL_malloc(device->work_buffer_len);
device->work_buffer = (Uint8 *)SDL_malloc(device->work_buffer_len);
if (device->work_buffer == NULL) {
close_audio_device(device);
SDL_UnlockMutex(current_audio.detectionLock);
@@ -1501,7 +1446,7 @@ open_audio_device(const char *devname, int iscapture,
return 0;
}
open_devices[id] = device; /* add it to our list of open devices. */
open_devices[id] = device; /* add it to our list of open devices. */
/* Start the audio thread if necessary */
if (!current_audio.impl.ProvidesOwnCallbackThread) {
@@ -1511,7 +1456,7 @@ open_audio_device(const char *devname, int iscapture,
const size_t stacksize = is_internal_thread ? 64 * 1024 : 0;
char threadname[64];
SDL_snprintf(threadname, sizeof (threadname), "SDLAudio%c%d", (iscapture) ? 'C' : 'P', (int) device->id);
SDL_snprintf(threadname, sizeof(threadname), "SDLAudio%c%d", (iscapture) ? 'C' : 'P', (int)device->id);
device->thread = SDL_CreateThreadInternal(iscapture ? SDL_CaptureAudio : SDL_RunAudio, threadname, stacksize, device);
if (device->thread == NULL) {
@@ -1526,9 +1471,7 @@ open_audio_device(const char *devname, int iscapture,
return device->id;
}
int
SDL_OpenAudio(SDL_AudioSpec * desired, SDL_AudioSpec * obtained)
int SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained)
{
SDL_AudioDeviceID id = 0;
@@ -1564,7 +1507,7 @@ SDL_OpenAudio(SDL_AudioSpec * desired, SDL_AudioSpec * obtained)
SDL_AudioDeviceID
SDL_OpenAudioDevice(const char *device, int iscapture,
const SDL_AudioSpec * desired, SDL_AudioSpec * obtained,
const SDL_AudioSpec *desired, SDL_AudioSpec *obtained,
int allowed_changes)
{
return open_audio_device(device, iscapture, desired, obtained,
@@ -1586,15 +1529,13 @@ SDL_GetAudioDeviceStatus(SDL_AudioDeviceID devid)
return status;
}
SDL_AudioStatus
SDL_GetAudioStatus(void)
{
return SDL_GetAudioDeviceStatus(1);
}
void
SDL_PauseAudioDevice(SDL_AudioDeviceID devid, int pause_on)
void SDL_PauseAudioDevice(SDL_AudioDeviceID devid, int pause_on)
{
SDL_AudioDevice *device = get_audio_device(devid);
if (device) {
@@ -1604,15 +1545,12 @@ SDL_PauseAudioDevice(SDL_AudioDeviceID devid, int pause_on)
}
}
void
SDL_PauseAudio(int pause_on)
void SDL_PauseAudio(int pause_on)
{
SDL_PauseAudioDevice(1, pause_on);
}
void
SDL_LockAudioDevice(SDL_AudioDeviceID devid)
void SDL_LockAudioDevice(SDL_AudioDeviceID devid)
{
/* Obtain a lock on the mixing buffers */
SDL_AudioDevice *device = get_audio_device(devid);
@@ -1621,14 +1559,12 @@ SDL_LockAudioDevice(SDL_AudioDeviceID devid)
}
}
void
SDL_LockAudio(void)
void SDL_LockAudio(void)
{
SDL_LockAudioDevice(1);
}
void
SDL_UnlockAudioDevice(SDL_AudioDeviceID devid)
void SDL_UnlockAudioDevice(SDL_AudioDeviceID devid)
{
/* Obtain a lock on the mixing buffers */
SDL_AudioDevice *device = get_audio_device(devid);
@@ -1637,30 +1573,26 @@ SDL_UnlockAudioDevice(SDL_AudioDeviceID devid)
}
}
void
SDL_UnlockAudio(void)
void SDL_UnlockAudio(void)
{
SDL_UnlockAudioDevice(1);
}
void
SDL_CloseAudioDevice(SDL_AudioDeviceID devid)
void SDL_CloseAudioDevice(SDL_AudioDeviceID devid)
{
close_audio_device(get_audio_device(devid));
}
void
SDL_CloseAudio(void)
void SDL_CloseAudio(void)
{
SDL_CloseAudioDevice(1);
}
void
SDL_AudioQuit(void)
void SDL_AudioQuit(void)
{
SDL_AudioDeviceID i;
if (!current_audio.name) { /* not initialized?! */
if (!current_audio.name) { /* not initialized?! */
return;
}
@@ -1688,26 +1620,26 @@ SDL_AudioQuit(void)
static int format_idx;
static int format_idx_sub;
static SDL_AudioFormat format_list[NUM_FORMATS][NUM_FORMATS] = {
{AUDIO_U8, AUDIO_S8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB,
AUDIO_U16MSB, AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB},
{AUDIO_S8, AUDIO_U8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB,
AUDIO_U16MSB, AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB},
{AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_S32LSB,
AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_U8, AUDIO_S8},
{AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_S32MSB,
AUDIO_S32LSB, AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_U8, AUDIO_S8},
{AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_S32LSB,
AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_U8, AUDIO_S8},
{AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_S32MSB,
AUDIO_S32LSB, AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_U8, AUDIO_S8},
{AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_S16LSB,
AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_U8, AUDIO_S8},
{AUDIO_S32MSB, AUDIO_S32LSB, AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_S16MSB,
AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_U8, AUDIO_S8},
{AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_S16LSB,
AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_U8, AUDIO_S8},
{AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_S32MSB, AUDIO_S32LSB, AUDIO_S16MSB,
AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_U8, AUDIO_S8},
{ AUDIO_U8, AUDIO_S8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB,
AUDIO_U16MSB, AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB },
{ AUDIO_S8, AUDIO_U8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB,
AUDIO_U16MSB, AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB },
{ AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_S32LSB,
AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_U8, AUDIO_S8 },
{ AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_S32MSB,
AUDIO_S32LSB, AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_U8, AUDIO_S8 },
{ AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_S32LSB,
AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_U8, AUDIO_S8 },
{ AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_S32MSB,
AUDIO_S32LSB, AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_U8, AUDIO_S8 },
{ AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_S16LSB,
AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_U8, AUDIO_S8 },
{ AUDIO_S32MSB, AUDIO_S32LSB, AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_S16MSB,
AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_U8, AUDIO_S8 },
{ AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_S16LSB,
AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_U8, AUDIO_S8 },
{ AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_S32MSB, AUDIO_S32LSB, AUDIO_S16MSB,
AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_U8, AUDIO_S8 },
};
SDL_AudioFormat
@@ -1731,26 +1663,25 @@ SDL_NextAudioFormat(void)
return format_list[format_idx][format_idx_sub++];
}
Uint8
SDL_SilenceValueForFormat(const SDL_AudioFormat format)
Uint8 SDL_SilenceValueForFormat(const SDL_AudioFormat format)
{
switch (format) {
/* !!! FIXME: 0x80 isn't perfect for U16, but we can't fit 0x8000 in a
!!! FIXME: byte for SDL_memset() use. This is actually 0.1953 percent
!!! FIXME: off from silence. Maybe just don't use U16. */
case AUDIO_U16LSB:
case AUDIO_U16MSB:
case AUDIO_U8:
return 0x80;
/* !!! FIXME: 0x80 isn't perfect for U16, but we can't fit 0x8000 in a
!!! FIXME: byte for SDL_memset() use. This is actually 0.1953 percent
!!! FIXME: off from silence. Maybe just don't use U16. */
case AUDIO_U16LSB:
case AUDIO_U16MSB:
case AUDIO_U8:
return 0x80;
default: break;
}
default:
break;
}
return 0x00;
}
void
SDL_CalculateAudioSpec(SDL_AudioSpec * spec)
void SDL_CalculateAudioSpec(SDL_AudioSpec *spec)
{
spec->silence = SDL_SilenceValueForFormat(spec->format);
spec->size = SDL_AUDIO_BITSIZE(spec->format) / 8;
@@ -1758,13 +1689,11 @@ SDL_CalculateAudioSpec(SDL_AudioSpec * spec)
spec->size *= spec->samples;
}
/*
* Moved here from SDL_mixer.c, since it relies on internals of an opened
* audio device (and is deprecated, by the way!).
*/
void
SDL_MixAudio(Uint8 * dst, const Uint8 * src, Uint32 len, int volume)
void SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume)
{
/* Mix the user-level audio format */
SDL_AudioDevice *device = get_audio_device(1);

8
src/audio/SDL_audio_c.h
View File

@@ -40,11 +40,11 @@
#include "samplerate.h"
extern SDL_bool SRC_available;
extern int SRC_converter;
extern SRC_STATE* (*SRC_src_new)(int converter_type, int channels, int *error);
extern SRC_STATE *(*SRC_src_new)(int converter_type, int channels, int *error);
extern int (*SRC_src_process)(SRC_STATE *state, SRC_DATA *data);
extern int (*SRC_src_reset)(SRC_STATE *state);
extern SRC_STATE* (*SRC_src_delete)(SRC_STATE *state);
extern const char* (*SRC_src_strerror)(int error);
extern SRC_STATE *(*SRC_src_delete)(SRC_STATE *state);
extern const char *(*SRC_src_strerror)(int error);
extern int (*SRC_src_simple)(SRC_DATA *data, int converter_type, int channels);
#endif
@@ -54,7 +54,7 @@ extern SDL_AudioFormat SDL_NextAudioFormat(void);
/* Function to calculate the size and silence for a SDL_AudioSpec */
extern Uint8 SDL_SilenceValueForFormat(const SDL_AudioFormat format);
extern void SDL_CalculateAudioSpec(SDL_AudioSpec * spec);
extern void SDL_CalculateAudioSpec(SDL_AudioSpec *spec);
/* Choose the audio filter functions below */
extern void SDL_ChooseAudioConverters(void);

579
src/audio/SDL_audio_channel_converters.h
View File

File diff suppressed because it is too large Load Diff

7
src/audio/SDL_audio_resampler_filter.h
View File

@@ -21,10 +21,10 @@
/* DO NOT EDIT, THIS FILE WAS GENERATED BY build-scripts/gen_audio_resampler_filter.c */
#define RESAMPLER_ZERO_CROSSINGS 5
#define RESAMPLER_BITS_PER_SAMPLE 16
#define RESAMPLER_ZERO_CROSSINGS 5
#define RESAMPLER_BITS_PER_SAMPLE 16
#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING (1 << ((RESAMPLER_BITS_PER_SAMPLE / 2) + 1))
#define RESAMPLER_FILTER_SIZE ((RESAMPLER_SAMPLES_PER_ZERO_CROSSING * RESAMPLER_ZERO_CROSSINGS) + 1)
#define RESAMPLER_FILTER_SIZE ((RESAMPLER_SAMPLES_PER_ZERO_CROSSING * RESAMPLER_ZERO_CROSSINGS) + 1)
static const float ResamplerFilter[RESAMPLER_FILTER_SIZE] = {
1.000000000f, 0.999993682f, 0.999974370f, 0.999941289f, 0.999894559f,
@@ -1059,4 +1059,3 @@ static const float ResamplerFilterDifference[RESAMPLER_FILTER_SIZE] = {
};
/* vi: set ts=4 sw=4 expandtab: */

560
src/audio/SDL_audiocvt.c
View File

File diff suppressed because it is too large Load Diff

22
src/audio/SDL_audiodev.c
View File

@@ -33,9 +33,9 @@
#ifndef _PATH_DEV_DSP
#if defined(__NETBSD__) || defined(__OPENBSD__)
#define _PATH_DEV_DSP "/dev/audio"
#define _PATH_DEV_DSP "/dev/audio"
#else
#define _PATH_DEV_DSP "/dev/dsp"
#define _PATH_DEV_DSP "/dev/dsp"
#endif
#endif
#ifndef _PATH_DEV_DSP24
@@ -45,8 +45,7 @@
#define _PATH_DEV_AUDIO "/dev/audio"
#endif
static void
test_device(const int iscapture, const char *fname, int flags, int (*test) (int fd))
static void test_device(const int iscapture, const char *fname, int flags, int (*test)(int fd))
{
struct stat sb;
if ((stat(fname, &sb) == 0) && (S_ISCHR(sb.st_mode))) {
@@ -64,20 +63,18 @@ test_device(const int iscapture, const char *fname, int flags, int (*test) (int
* information, making this information inaccessible at
* enumeration time
*/
SDL_AddAudioDevice(iscapture, fname, NULL, (void *) (uintptr_t) dummyhandle);
SDL_AddAudioDevice(iscapture, fname, NULL, (void *)(uintptr_t)dummyhandle);
}
}
}
}
static int
test_stub(int fd)
static int test_stub(int fd)
{
return 1;
}
static void
SDL_EnumUnixAudioDevices_Internal(const int iscapture, const int classic, int (*test)(int))
static void SDL_EnumUnixAudioDevices_Internal(const int iscapture, const int classic, int (*test)(int))
{
const int flags = iscapture ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT;
const char *audiodev;
@@ -96,9 +93,7 @@ SDL_EnumUnixAudioDevices_Internal(const int iscapture, const int classic, int (*
struct stat sb;
/* Added support for /dev/sound/\* in Linux 2.4 */
if (((stat("/dev/sound", &sb) == 0) && S_ISDIR(sb.st_mode))
&& ((stat(_PATH_DEV_DSP24, &sb) == 0)
&& S_ISCHR(sb.st_mode))) {
if (((stat("/dev/sound", &sb) == 0) && S_ISDIR(sb.st_mode)) && ((stat(_PATH_DEV_DSP24, &sb) == 0) && S_ISCHR(sb.st_mode))) {
audiodev = _PATH_DEV_DSP24;
} else {
audiodev = _PATH_DEV_DSP;
@@ -118,8 +113,7 @@ SDL_EnumUnixAudioDevices_Internal(const int iscapture, const int classic, int (*
}
}
void
SDL_EnumUnixAudioDevices(const int classic, int (*test)(int))
void SDL_EnumUnixAudioDevices(const int classic, int (*test)(int))
{
SDL_EnumUnixAudioDevices_Internal(SDL_TRUE, classic, test);
SDL_EnumUnixAudioDevices_Internal(SDL_FALSE, classic, test);

6
src/audio/SDL_audiodev_c.h
View File

@@ -30,10 +30,10 @@
#ifdef USE_BLOCKING_WRITES
#define OPEN_FLAGS_OUTPUT O_WRONLY
#define OPEN_FLAGS_INPUT O_RDONLY
#define OPEN_FLAGS_INPUT O_RDONLY
#else
#define OPEN_FLAGS_OUTPUT (O_WRONLY|O_NONBLOCK)
#define OPEN_FLAGS_INPUT (O_RDONLY|O_NONBLOCK)
#define OPEN_FLAGS_OUTPUT (O_WRONLY | O_NONBLOCK)
#define OPEN_FLAGS_INPUT (O_RDONLY | O_NONBLOCK)
#endif
extern void SDL_EnumUnixAudioDevices(const int classic, int (*test)(int));

884
src/audio/SDL_audiotypecvt.c
View File

File diff suppressed because it is too large Load Diff

406
src/audio/SDL_mixer.c
View File

@@ -78,14 +78,12 @@ static const Uint8 mix8[] = {
};
/* The volume ranges from 0 - 128 */
#define ADJUST_VOLUME(s, v) (s = (s*v)/SDL_MIX_MAXVOLUME)
#define ADJUST_VOLUME_U8(s, v) (s = (((s-128)*v)/SDL_MIX_MAXVOLUME)+128)
#define ADJUST_VOLUME_U16(s, v) (s = (((s-32768)*v)/SDL_MIX_MAXVOLUME)+32768)
#define ADJUST_VOLUME(s, v) (s = (s * v) / SDL_MIX_MAXVOLUME)
#define ADJUST_VOLUME_U8(s, v) (s = (((s - 128) * v) / SDL_MIX_MAXVOLUME) + 128)
#define ADJUST_VOLUME_U16(s, v) (s = (((s - 32768) * v) / SDL_MIX_MAXVOLUME) + 32768)
void
SDL_MixAudioFormat(Uint8 * dst, const Uint8 * src, SDL_AudioFormat format,
Uint32 len, int volume)
void SDL_MixAudioFormat(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format,
Uint32 len, int volume)
{
if (volume == 0) {
return;
@@ -94,258 +92,248 @@ SDL_MixAudioFormat(Uint8 * dst, const Uint8 * src, SDL_AudioFormat format,
switch (format) {
case AUDIO_U8:
{
Uint8 src_sample;
{
Uint8 src_sample;
while (len--) {
src_sample = *src;
ADJUST_VOLUME_U8(src_sample, volume);
*dst = mix8[*dst + src_sample];
++dst;
++src;
}
while (len--) {
src_sample = *src;
ADJUST_VOLUME_U8(src_sample, volume);
*dst = mix8[*dst + src_sample];
++dst;
++src;
}
break;
} break;
case AUDIO_S8:
{
Sint8 *dst8, *src8;
Sint8 src_sample;
int dst_sample;
const int max_audioval = SDL_MAX_SINT8;
const int min_audioval = SDL_MIN_SINT8;
{
Sint8 *dst8, *src8;
Sint8 src_sample;
int dst_sample;
const int max_audioval = SDL_MAX_SINT8;
const int min_audioval = SDL_MIN_SINT8;
src8 = (Sint8 *) src;
dst8 = (Sint8 *) dst;
while (len--) {
src_sample = *src8;
ADJUST_VOLUME(src_sample, volume);
dst_sample = *dst8 + src_sample;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*dst8 = dst_sample;
++dst8;
++src8;
src8 = (Sint8 *)src;
dst8 = (Sint8 *)dst;
while (len--) {
src_sample = *src8;
ADJUST_VOLUME(src_sample, volume);
dst_sample = *dst8 + src_sample;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*dst8 = dst_sample;
++dst8;
++src8;
}
break;
} break;
case AUDIO_S16LSB:
{
Sint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
{
Sint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
len /= 2;
while (len--) {
src1 = SDL_SwapLE16(*(Sint16 *)src);
ADJUST_VOLUME(src1, volume);
src2 = SDL_SwapLE16(*(Sint16 *)dst);
src += 2;
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(Sint16 *)dst = SDL_SwapLE16(dst_sample);
dst += 2;
len /= 2;
while (len--) {
src1 = SDL_SwapLE16(*(Sint16 *)src);
ADJUST_VOLUME(src1, volume);
src2 = SDL_SwapLE16(*(Sint16 *)dst);
src += 2;
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(Sint16 *)dst = SDL_SwapLE16(dst_sample);
dst += 2;
}
break;
} break;
case AUDIO_S16MSB:
{
Sint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
{
Sint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
len /= 2;
while (len--) {
src1 = SDL_SwapBE16(*(Sint16 *)src);
ADJUST_VOLUME(src1, volume);
src2 = SDL_SwapBE16(*(Sint16 *)dst);
src += 2;
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(Sint16 *)dst = SDL_SwapBE16(dst_sample);
dst += 2;
len /= 2;
while (len--) {
src1 = SDL_SwapBE16(*(Sint16 *)src);
ADJUST_VOLUME(src1, volume);
src2 = SDL_SwapBE16(*(Sint16 *)dst);
src += 2;
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(Sint16 *)dst = SDL_SwapBE16(dst_sample);
dst += 2;
}
break;
} break;
case AUDIO_U16LSB:
{
Uint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
{
Uint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
len /= 2;
while (len--) {
src1 = SDL_SwapLE16(*(Uint16 *)src);
ADJUST_VOLUME_U16(src1, volume);
src2 = SDL_SwapLE16(*(Uint16 *)dst);
src += 2;
dst_sample = src1 + src2 - 32768 * 2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
dst_sample += 32768;
*(Uint16 *)dst = SDL_SwapLE16(dst_sample);
dst += 2;
len /= 2;
while (len--) {
src1 = SDL_SwapLE16(*(Uint16 *)src);
ADJUST_VOLUME_U16(src1, volume);
src2 = SDL_SwapLE16(*(Uint16 *)dst);
src += 2;
dst_sample = src1 + src2 - 32768 * 2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
dst_sample += 32768;
*(Uint16 *)dst = SDL_SwapLE16(dst_sample);
dst += 2;
}
break;
} break;
case AUDIO_U16MSB:
{
Uint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
{
Uint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
len /= 2;
while (len--) {
src1 = SDL_SwapBE16(*(Uint16 *)src);
ADJUST_VOLUME_U16(src1, volume);
src2 = SDL_SwapBE16(*(Uint16 *)dst);
src += 2;
dst_sample = src1 + src2 - 32768 * 2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
dst_sample += 32768;
*(Uint16 *)dst = SDL_SwapBE16(dst_sample);
dst += 2;
len /= 2;
while (len--) {
src1 = SDL_SwapBE16(*(Uint16 *)src);
ADJUST_VOLUME_U16(src1, volume);
src2 = SDL_SwapBE16(*(Uint16 *)dst);
src += 2;
dst_sample = src1 + src2 - 32768 * 2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
dst_sample += 32768;
*(Uint16 *)dst = SDL_SwapBE16(dst_sample);
dst += 2;
}
break;
} break;
case AUDIO_S32LSB:
{
const Uint32 *src32 = (Uint32 *) src;
Uint32 *dst32 = (Uint32 *) dst;
Sint64 src1, src2;
Sint64 dst_sample;
const Sint64 max_audioval = SDL_MAX_SINT32;
const Sint64 min_audioval = SDL_MIN_SINT32;
{
const Uint32 *src32 = (Uint32 *)src;
Uint32 *dst32 = (Uint32 *)dst;
Sint64 src1, src2;
Sint64 dst_sample;
const Sint64 max_audioval = SDL_MAX_SINT32;
const Sint64 min_audioval = SDL_MIN_SINT32;
len /= 4;
while (len--) {
src1 = (Sint64) ((Sint32) SDL_SwapLE32(*src32));
src32++;
ADJUST_VOLUME(src1, volume);
src2 = (Sint64) ((Sint32) SDL_SwapLE32(*dst32));
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapLE32((Uint32) ((Sint32) dst_sample));
len /= 4;
while (len--) {
src1 = (Sint64)((Sint32)SDL_SwapLE32(*src32));
src32++;
ADJUST_VOLUME(src1, volume);
src2 = (Sint64)((Sint32)SDL_SwapLE32(*dst32));
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapLE32((Uint32)((Sint32)dst_sample));
}
break;
} break;
case AUDIO_S32MSB:
{
const Uint32 *src32 = (Uint32 *) src;
Uint32 *dst32 = (Uint32 *) dst;
Sint64 src1, src2;
Sint64 dst_sample;
const Sint64 max_audioval = SDL_MAX_SINT32;
const Sint64 min_audioval = SDL_MIN_SINT32;
{
const Uint32 *src32 = (Uint32 *)src;
Uint32 *dst32 = (Uint32 *)dst;
Sint64 src1, src2;
Sint64 dst_sample;
const Sint64 max_audioval = SDL_MAX_SINT32;
const Sint64 min_audioval = SDL_MIN_SINT32;
len /= 4;
while (len--) {
src1 = (Sint64) ((Sint32) SDL_SwapBE32(*src32));
src32++;
ADJUST_VOLUME(src1, volume);
src2 = (Sint64) ((Sint32) SDL_SwapBE32(*dst32));
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapBE32((Uint32) ((Sint32) dst_sample));
len /= 4;
while (len--) {
src1 = (Sint64)((Sint32)SDL_SwapBE32(*src32));
src32++;
ADJUST_VOLUME(src1, volume);
src2 = (Sint64)((Sint32)SDL_SwapBE32(*dst32));
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapBE32((Uint32)((Sint32)dst_sample));
}
break;
} break;
case AUDIO_F32LSB:
{
const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
const float fvolume = (float) volume;
const float *src32 = (float *) src;
float *dst32 = (float *) dst;
float src1, src2;
double dst_sample;
/* !!! FIXME: are these right? */
const double max_audioval = 3.402823466e+38F;
const double min_audioval = -3.402823466e+38F;
{
const float fmaxvolume = 1.0f / ((float)SDL_MIX_MAXVOLUME);
const float fvolume = (float)volume;
const float *src32 = (float *)src;
float *dst32 = (float *)dst;
float src1, src2;
double dst_sample;
/* !!! FIXME: are these right? */
const double max_audioval = 3.402823466e+38F;
const double min_audioval = -3.402823466e+38F;
len /= 4;
while (len--) {
src1 = ((SDL_SwapFloatLE(*src32) * fvolume) * fmaxvolume);
src2 = SDL_SwapFloatLE(*dst32);
src32++;
len /= 4;
while (len--) {
src1 = ((SDL_SwapFloatLE(*src32) * fvolume) * fmaxvolume);
src2 = SDL_SwapFloatLE(*dst32);
src32++;
dst_sample = ((double) src1) + ((double) src2);
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapFloatLE((float) dst_sample);
dst_sample = ((double)src1) + ((double)src2);
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapFloatLE((float)dst_sample);
}
break;
} break;
case AUDIO_F32MSB:
{
const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
const float fvolume = (float) volume;
const float *src32 = (float *) src;
float *dst32 = (float *) dst;
float src1, src2;
double dst_sample;
/* !!! FIXME: are these right? */
const double max_audioval = 3.402823466e+38F;
const double min_audioval = -3.402823466e+38F;
{
const float fmaxvolume = 1.0f / ((float)SDL_MIX_MAXVOLUME);
const float fvolume = (float)volume;
const float *src32 = (float *)src;
float *dst32 = (float *)dst;
float src1, src2;
double dst_sample;
/* !!! FIXME: are these right? */
const double max_audioval = 3.402823466e+38F;
const double min_audioval = -3.402823466e+38F;
len /= 4;
while (len--) {
src1 = ((SDL_SwapFloatBE(*src32) * fvolume) * fmaxvolume);
src2 = SDL_SwapFloatBE(*dst32);
src32++;
len /= 4;
while (len--) {
src1 = ((SDL_SwapFloatBE(*src32) * fvolume) * fmaxvolume);
src2 = SDL_SwapFloatBE(*dst32);
src32++;
dst_sample = ((double) src1) + ((double) src2);
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapFloatBE((float) dst_sample);
dst_sample = ((double)src1) + ((double)src2);
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapFloatBE((float)dst_sample);
}
break;
} break;
default: /* If this happens... FIXME! */
default: /* If this happens... FIXME! */
SDL_SetError("SDL_MixAudioFormat(): unknown audio format");
return;
}

41
src/audio/SDL_sysaudio.h
View File

@@ -28,11 +28,11 @@
/* !!! FIXME: These are wordy and unlocalized... */
#define DEFAULT_OUTPUT_DEVNAME "System audio output device"
#define DEFAULT_INPUT_DEVNAME "System audio capture device"
#define DEFAULT_INPUT_DEVNAME "System audio capture device"
/* The SDL audio driver */
typedef struct SDL_AudioDevice SDL_AudioDevice;
#define _THIS SDL_AudioDevice *_this
#define _THIS SDL_AudioDevice *_this
/* Audio targets should call this as devices are added to the system (such as
a USB headset being plugged in), and should also be called for
@@ -62,21 +62,21 @@ extern void SDL_OpenedAudioDeviceDisconnected(SDL_AudioDevice *device);
typedef struct SDL_AudioDriverImpl
{
void (*DetectDevices) (void);
int (*OpenDevice) (_THIS, const char *devname);
void (*ThreadInit) (_THIS); /* Called by audio thread at start */
void (*ThreadDeinit) (_THIS); /* Called by audio thread at end */
void (*WaitDevice) (_THIS);
void (*PlayDevice) (_THIS);
Uint8 *(*GetDeviceBuf) (_THIS);
int (*CaptureFromDevice) (_THIS, void *buffer, int buflen);
void (*FlushCapture) (_THIS);
void (*CloseDevice) (_THIS);
void (*LockDevice) (_THIS);
void (*UnlockDevice) (_THIS);
void (*FreeDeviceHandle) (void *handle); /**< SDL is done with handle from SDL_AddAudioDevice() */
void (*Deinitialize) (void);
int (*GetDefaultAudioInfo) (char **name, SDL_AudioSpec *spec, int iscapture);
void (*DetectDevices)(void);
int (*OpenDevice)(_THIS, const char *devname);
void (*ThreadInit)(_THIS); /* Called by audio thread at start */
void (*ThreadDeinit)(_THIS); /* Called by audio thread at end */
void (*WaitDevice)(_THIS);
void (*PlayDevice)(_THIS);
Uint8 *(*GetDeviceBuf)(_THIS);
int (*CaptureFromDevice)(_THIS, void *buffer, int buflen);
void (*FlushCapture)(_THIS);
void (*CloseDevice)(_THIS);
void (*LockDevice)(_THIS);
void (*UnlockDevice)(_THIS);
void (*FreeDeviceHandle)(void *handle); /**< SDL is done with handle from SDL_AddAudioDevice() */
void (*Deinitialize)(void);
int (*GetDefaultAudioInfo)(char **name, SDL_AudioSpec *spec, int iscapture);
/* !!! FIXME: add pause(), so we can optimize instead of mixing silence. */
@@ -89,7 +89,6 @@ typedef struct SDL_AudioDriverImpl
SDL_bool SupportsNonPow2Samples;
} SDL_AudioDriverImpl;
typedef struct SDL_AudioDeviceItem
{
void *handle;
@@ -100,7 +99,6 @@ typedef struct SDL_AudioDeviceItem
struct SDL_AudioDeviceItem *next;
} SDL_AudioDeviceItem;
typedef struct SDL_AudioDriver
{
/* * * */
@@ -123,7 +121,6 @@ typedef struct SDL_AudioDriver
SDL_AudioDeviceItem *inputDevices;
} SDL_AudioDriver;
/* Define the SDL audio driver structure */
struct SDL_AudioDevice
{
@@ -174,8 +171,8 @@ typedef struct AudioBootStrap
{
const char *name;
const char *desc;
SDL_bool (*init) (SDL_AudioDriverImpl * impl);
SDL_bool demand_only; /* 1==request explicitly, or it won't be available. */
SDL_bool (*init)(SDL_AudioDriverImpl *impl);
SDL_bool demand_only; /* 1==request explicitly, or it won't be available. */
} AudioBootStrap;
/* Not all of these are available in a given build. Use #ifdefs, etc. */

259
src/audio/SDL_wave.c
View File

@@ -41,8 +41,7 @@
* Returns 0 on success, or -1 if the multiplication overflows, in which case f1
* does not get modified.
*/
static int
SafeMult(size_t *f1, size_t f2)
static int SafeMult(size_t *f1, size_t f2)
{
if (*f1 > 0 && SIZE_MAX / *f1 <= f2) {
return -1;
@@ -64,21 +63,24 @@ typedef struct ADPCM_DecoderState
void *cstate; /* Decoding state for each channel. */
/* ADPCM data. */
struct {
struct
{
Uint8 *data;
size_t size;
size_t pos;
} input;
/* Current ADPCM block in the ADPCM data above. */
struct {
struct
{
Uint8 *data;
size_t size;
size_t pos;
} block;
/* Decoded 16-bit PCM data. */
struct {
struct
{
Sint16 *data;
size_t size;
size_t pos;
@@ -100,8 +102,7 @@ typedef struct MS_ADPCM_ChannelState
} MS_ADPCM_ChannelState;
#ifdef SDL_WAVE_DEBUG_LOG_FORMAT
static void
WaveDebugLogFormat(WaveFile *file)
static void WaveDebugLogFormat(WaveFile *file)
{
WaveFormat *format = &file->format;
const char *fmtstr = "WAVE file: %s, %u Hz, %s, %u bits, %u %s/s";
@@ -135,55 +136,60 @@ WaveDebugLogFormat(WaveFile *file)
break;
}
#define SDL_WAVE_DEBUG_CHANNELCFG(STR, CODE) case CODE: wavechannel = STR; break;
#define SDL_WAVE_DEBUG_CHANNELSTR(STR, CODE) if (format->channelmask & CODE) { \
SDL_strlcat(channelstr, channelstr[0] ? "-" STR : STR, sizeof(channelstr));}
#define SDL_WAVE_DEBUG_CHANNELCFG(STR, CODE) \
case CODE: \
wavechannel = STR; \
break;
#define SDL_WAVE_DEBUG_CHANNELSTR(STR, CODE) \
if (format->channelmask & CODE) { \
SDL_strlcat(channelstr, channelstr[0] ? "-" STR : STR, sizeof(channelstr)); \
}
if (format->formattag == EXTENSIBLE_CODE && format->channelmask > 0) {
switch (format->channelmask) {
SDL_WAVE_DEBUG_CHANNELCFG("1.0 Mono", 0x4)
SDL_WAVE_DEBUG_CHANNELCFG("1.1 Mono", 0xc)
SDL_WAVE_DEBUG_CHANNELCFG("2.0 Stereo", 0x3)
SDL_WAVE_DEBUG_CHANNELCFG("2.1 Stereo", 0xb)
SDL_WAVE_DEBUG_CHANNELCFG("3.0 Stereo", 0x7)
SDL_WAVE_DEBUG_CHANNELCFG("3.1 Stereo", 0xf)
SDL_WAVE_DEBUG_CHANNELCFG("3.0 Surround", 0x103)
SDL_WAVE_DEBUG_CHANNELCFG("3.1 Surround", 0x10b)
SDL_WAVE_DEBUG_CHANNELCFG("4.0 Quad", 0x33)
SDL_WAVE_DEBUG_CHANNELCFG("4.1 Quad", 0x3b)
SDL_WAVE_DEBUG_CHANNELCFG("4.0 Surround", 0x107)
SDL_WAVE_DEBUG_CHANNELCFG("4.1 Surround", 0x10f)
SDL_WAVE_DEBUG_CHANNELCFG("5.0", 0x37)
SDL_WAVE_DEBUG_CHANNELCFG("5.1", 0x3f)
SDL_WAVE_DEBUG_CHANNELCFG("5.0 Side", 0x607)
SDL_WAVE_DEBUG_CHANNELCFG("5.1 Side", 0x60f)
SDL_WAVE_DEBUG_CHANNELCFG("6.0", 0x137)
SDL_WAVE_DEBUG_CHANNELCFG("6.1", 0x13f)
SDL_WAVE_DEBUG_CHANNELCFG("6.0 Side", 0x707)
SDL_WAVE_DEBUG_CHANNELCFG("6.1 Side", 0x70f)
SDL_WAVE_DEBUG_CHANNELCFG("7.0", 0xf7)
SDL_WAVE_DEBUG_CHANNELCFG("7.1", 0xff)
SDL_WAVE_DEBUG_CHANNELCFG("7.0 Side", 0x6c7)
SDL_WAVE_DEBUG_CHANNELCFG("7.1 Side", 0x6cf)
SDL_WAVE_DEBUG_CHANNELCFG("7.0 Surround", 0x637)
SDL_WAVE_DEBUG_CHANNELCFG("7.1 Surround", 0x63f)
SDL_WAVE_DEBUG_CHANNELCFG("9.0 Surround", 0x5637)
SDL_WAVE_DEBUG_CHANNELCFG("9.1 Surround", 0x563f)
SDL_WAVE_DEBUG_CHANNELCFG("11.0 Surround", 0x56f7)
SDL_WAVE_DEBUG_CHANNELCFG("11.1 Surround", 0x56ff)
SDL_WAVE_DEBUG_CHANNELCFG("1.0 Mono", 0x4)
SDL_WAVE_DEBUG_CHANNELCFG("1.1 Mono", 0xc)
SDL_WAVE_DEBUG_CHANNELCFG("2.0 Stereo", 0x3)
SDL_WAVE_DEBUG_CHANNELCFG("2.1 Stereo", 0xb)
SDL_WAVE_DEBUG_CHANNELCFG("3.0 Stereo", 0x7)
SDL_WAVE_DEBUG_CHANNELCFG("3.1 Stereo", 0xf)
SDL_WAVE_DEBUG_CHANNELCFG("3.0 Surround", 0x103)
SDL_WAVE_DEBUG_CHANNELCFG("3.1 Surround", 0x10b)
SDL_WAVE_DEBUG_CHANNELCFG("4.0 Quad", 0x33)
SDL_WAVE_DEBUG_CHANNELCFG("4.1 Quad", 0x3b)
SDL_WAVE_DEBUG_CHANNELCFG("4.0 Surround", 0x107)
SDL_WAVE_DEBUG_CHANNELCFG("4.1 Surround", 0x10f)
SDL_WAVE_DEBUG_CHANNELCFG("5.0", 0x37)
SDL_WAVE_DEBUG_CHANNELCFG("5.1", 0x3f)
SDL_WAVE_DEBUG_CHANNELCFG("5.0 Side", 0x607)
SDL_WAVE_DEBUG_CHANNELCFG("5.1 Side", 0x60f)
SDL_WAVE_DEBUG_CHANNELCFG("6.0", 0x137)
SDL_WAVE_DEBUG_CHANNELCFG("6.1", 0x13f)
SDL_WAVE_DEBUG_CHANNELCFG("6.0 Side", 0x707)
SDL_WAVE_DEBUG_CHANNELCFG("6.1 Side", 0x70f)
SDL_WAVE_DEBUG_CHANNELCFG("7.0", 0xf7)
SDL_WAVE_DEBUG_CHANNELCFG("7.1", 0xff)
SDL_WAVE_DEBUG_CHANNELCFG("7.0 Side", 0x6c7)
SDL_WAVE_DEBUG_CHANNELCFG("7.1 Side", 0x6cf)
SDL_WAVE_DEBUG_CHANNELCFG("7.0 Surround", 0x637)
SDL_WAVE_DEBUG_CHANNELCFG("7.1 Surround", 0x63f)
SDL_WAVE_DEBUG_CHANNELCFG("9.0 Surround", 0x5637)
SDL_WAVE_DEBUG_CHANNELCFG("9.1 Surround", 0x563f)
SDL_WAVE_DEBUG_CHANNELCFG("11.0 Surround", 0x56f7)
SDL_WAVE_DEBUG_CHANNELCFG("11.1 Surround", 0x56ff)
default:
SDL_WAVE_DEBUG_CHANNELSTR("FL", 0x1)
SDL_WAVE_DEBUG_CHANNELSTR("FR", 0x2)
SDL_WAVE_DEBUG_CHANNELSTR("FC", 0x4)
SDL_WAVE_DEBUG_CHANNELSTR("LF", 0x8)
SDL_WAVE_DEBUG_CHANNELSTR("BL", 0x10)
SDL_WAVE_DEBUG_CHANNELSTR("BR", 0x20)
SDL_WAVE_DEBUG_CHANNELSTR("FL", 0x1)
SDL_WAVE_DEBUG_CHANNELSTR("FR", 0x2)
SDL_WAVE_DEBUG_CHANNELSTR("FC", 0x4)
SDL_WAVE_DEBUG_CHANNELSTR("LF", 0x8)
SDL_WAVE_DEBUG_CHANNELSTR("BL", 0x10)
SDL_WAVE_DEBUG_CHANNELSTR("BR", 0x20)
SDL_WAVE_DEBUG_CHANNELSTR("FLC", 0x40)
SDL_WAVE_DEBUG_CHANNELSTR("FRC", 0x80)
SDL_WAVE_DEBUG_CHANNELSTR("BC", 0x100)
SDL_WAVE_DEBUG_CHANNELSTR("SL", 0x200)
SDL_WAVE_DEBUG_CHANNELSTR("SR", 0x400)
SDL_WAVE_DEBUG_CHANNELSTR("TC", 0x800)
SDL_WAVE_DEBUG_CHANNELSTR("BC", 0x100)
SDL_WAVE_DEBUG_CHANNELSTR("SL", 0x200)
SDL_WAVE_DEBUG_CHANNELSTR("SR", 0x400)
SDL_WAVE_DEBUG_CHANNELSTR("TC", 0x800)
SDL_WAVE_DEBUG_CHANNELSTR("TFL", 0x1000)
SDL_WAVE_DEBUG_CHANNELSTR("TFC", 0x2000)
SDL_WAVE_DEBUG_CHANNELSTR("TFR", 0x4000)
@@ -224,33 +230,32 @@ WaveDebugLogFormat(WaveFile *file)
#endif
#ifdef SDL_WAVE_DEBUG_DUMP_FORMAT
static void
WaveDebugDumpFormat(WaveFile *file, Uint32 rifflen, Uint32 fmtlen, Uint32 datalen)
static void WaveDebugDumpFormat(WaveFile *file, Uint32 rifflen, Uint32 fmtlen, Uint32 datalen)
{
WaveFormat *format = &file->format;
const char *fmtstr1 = "WAVE chunk dump:\n"
"-------------------------------------------\n"
"RIFF %11u\n"
"-------------------------------------------\n"
" fmt %11u\n"
" wFormatTag 0x%04x\n"
" nChannels %11u\n"
" nSamplesPerSec %11u\n"
" nAvgBytesPerSec %11u\n"
" nBlockAlign %11u\n";
"-------------------------------------------\n"
"RIFF %11u\n"
"-------------------------------------------\n"
" fmt %11u\n"
" wFormatTag 0x%04x\n"
" nChannels %11u\n"
" nSamplesPerSec %11u\n"
" nAvgBytesPerSec %11u\n"
" nBlockAlign %11u\n";
const char *fmtstr2 = " wBitsPerSample %11u\n";
const char *fmtstr3 = " cbSize %11u\n";
const char *fmtstr4a = " wValidBitsPerSample %11u\n";
const char *fmtstr4b = " wSamplesPerBlock %11u\n";
const char *fmtstr5 = " dwChannelMask 0x%08x\n"
" SubFormat\n"
" %08x-%04x-%04x-%02x%02x%02x%02x%02x%02x%02x%02x\n";
" SubFormat\n"
" %08x-%04x-%04x-%02x%02x%02x%02x%02x%02x%02x%02x\n";
const char *fmtstr6 = "-------------------------------------------\n"
" fact\n"
" dwSampleLength %11u\n";
" fact\n"
" dwSampleLength %11u\n";
const char *fmtstr7 = "-------------------------------------------\n"
" data %11u\n"
"-------------------------------------------\n";
" data %11u\n"
"-------------------------------------------\n";
char *dumpstr;
size_t dumppos = 0;
const size_t bufsize = 1024;
@@ -315,8 +320,7 @@ WaveDebugDumpFormat(WaveFile *file, Uint32 rifflen, Uint32 fmtlen, Uint32 datale
}
#endif
static Sint64
WaveAdjustToFactValue(WaveFile *file, Sint64 sampleframes)
static Sint64 WaveAdjustToFactValue(WaveFile *file, Sint64 sampleframes)
{
if (file->fact.status == 2) {
if (file->facthint == FactStrict && sampleframes < file->fact.samplelength) {
@@ -329,8 +333,7 @@ WaveAdjustToFactValue(WaveFile *file, Sint64 sampleframes)
return sampleframes;
}
static int
MS_ADPCM_CalculateSampleFrames(WaveFile *file, size_t datalength)
static int MS_ADPCM_CalculateSampleFrames(WaveFile *file, size_t datalength)
{
WaveFormat *format = &file->format;
const size_t blockheadersize = (size_t)file->format.channels * 7;
@@ -369,8 +372,7 @@ MS_ADPCM_CalculateSampleFrames(WaveFile *file, size_t datalength)
return 0;
}
static int
MS_ADPCM_Init(WaveFile *file, size_t datalength)
static int MS_ADPCM_Init(WaveFile *file, size_t datalength)
{
WaveFormat *format = &file->format;
WaveChunk *chunk = &file->chunk;
@@ -378,7 +380,7 @@ MS_ADPCM_Init(WaveFile *file, size_t datalength)
const size_t blockdatasize = (size_t)format->blockalign - blockheadersize;
const size_t blockframebitsize = (size_t)format->bitspersample * format->channels;
const size_t blockdatasamples = (blockdatasize * 8) / blockframebitsize;
const Sint16 presetcoeffs[14] = {256, 0, 512, -256, 0, 0, 192, 64, 240, 0, 460, -208, 392, -232};
const Sint16 presetcoeffs[14] = { 256, 0, 512, -256, 0, 0, 192, 64, 240, 0, 460, -208, 392, -232 };
size_t i, coeffcount;
MS_ADPCM_CoeffData *coeffdata;
@@ -488,8 +490,7 @@ MS_ADPCM_Init(WaveFile *file, size_t datalength)
return 0;
}
static Sint16
MS_ADPCM_ProcessNibble(MS_ADPCM_ChannelState *cstate, Sint32 sample1, Sint32 sample2, Uint8 nybble)
static Sint16 MS_ADPCM_ProcessNibble(MS_ADPCM_ChannelState *cstate, Sint32 sample1, Sint32 sample2, Uint8 nybble)
{
const Sint32 max_audioval = 32767;
const Sint32 min_audioval = -32768;
@@ -525,8 +526,7 @@ MS_ADPCM_ProcessNibble(MS_ADPCM_ChannelState *cstate, Sint32 sample1, Sint32 sam
return (Sint16)new_sample;
}
static int
MS_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
static int MS_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
{
Uint8 coeffindex;
const Uint32 channels = state->channels;
@@ -586,8 +586,7 @@ MS_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
* will always contain full sample frames (same sample count for each channel).
* Incomplete sample frames are discarded.
*/
static int
MS_ADPCM_DecodeBlockData(ADPCM_DecoderState *state)
static int MS_ADPCM_DecodeBlockData(ADPCM_DecoderState *state)
{
Uint16 nybble = 0;
Sint16 sample1, sample2;
@@ -634,8 +633,7 @@ MS_ADPCM_DecodeBlockData(ADPCM_DecoderState *state)
return 0;
}
static int
MS_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
static int MS_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
{
int result;
size_t bytesleft, outputsize;
@@ -734,8 +732,7 @@ MS_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
return 0;
}
static int
IMA_ADPCM_CalculateSampleFrames(WaveFile *file, size_t datalength)
static int IMA_ADPCM_CalculateSampleFrames(WaveFile *file, size_t datalength)
{
WaveFormat *format = &file->format;
const size_t blockheadersize = (size_t)format->channels * 4;
@@ -788,8 +785,7 @@ IMA_ADPCM_CalculateSampleFrames(WaveFile *file, size_t datalength)
return 0;
}
static int
IMA_ADPCM_Init(WaveFile *file, size_t datalength)
static int IMA_ADPCM_Init(WaveFile *file, size_t datalength)
{
WaveFormat *format = &file->format;
WaveChunk *chunk = &file->chunk;
@@ -854,8 +850,7 @@ IMA_ADPCM_Init(WaveFile *file, size_t datalength)
return 0;
}
static Sint16
IMA_ADPCM_ProcessNibble(Sint8 *cindex, Sint16 lastsample, Uint8 nybble)
static Sint16 IMA_ADPCM_ProcessNibble(Sint8 *cindex, Sint16 lastsample, Uint8 nybble)
{
const Sint32 max_audioval = 32767;
const Sint32 min_audioval = -32768;
@@ -924,12 +919,11 @@ IMA_ADPCM_ProcessNibble(Sint8 *cindex, Sint16 lastsample, Uint8 nybble)
return (Sint16)sample;
}
static int
IMA_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
static int IMA_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
{
Sint16 step;
Uint32 c;
Uint8 *cstate = (Uint8 *) state->cstate;
Uint8 *cstate = (Uint8 *)state->cstate;
for (c = 0; c < state->channels; c++) {
size_t o = state->block.pos + c * 4;
@@ -947,7 +941,7 @@ IMA_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
/* Reserved byte in block header, should be 0. */
if (state->block.data[o + 3] != 0) {
/* Uh oh, corrupt data? Buggy code? */ ;
/* Uh oh, corrupt data? Buggy code? */;
}
}
@@ -964,8 +958,7 @@ IMA_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
* contains full sample frames (same sample count for each channel).
* Incomplete sample frames are discarded.
*/
static int
IMA_ADPCM_DecodeBlockData(ADPCM_DecoderState *state)
static int IMA_ADPCM_DecodeBlockData(ADPCM_DecoderState *state)
{
size_t i;
int retval = 0;
@@ -1034,8 +1027,7 @@ IMA_ADPCM_DecodeBlockData(ADPCM_DecoderState *state)
return retval;
}
static int
IMA_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
static int IMA_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
{
int result;
size_t bytesleft, outputsize;
@@ -1138,8 +1130,7 @@ IMA_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
return 0;
}
static int
LAW_Init(WaveFile *file, size_t datalength)
static int LAW_Init(WaveFile *file, size_t datalength)
{
WaveFormat *format = &file->format;
@@ -1167,8 +1158,7 @@ LAW_Init(WaveFile *file, size_t datalength)
return 0;
}
static int
LAW_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
static int LAW_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
{
#ifdef SDL_WAVE_LAW_LUT
const Sint16 alaw_lut[256] = {
@@ -1310,8 +1300,7 @@ LAW_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
return 0;
}
static int
PCM_Init(WaveFile *file, size_t datalength)
static int PCM_Init(WaveFile *file, size_t datalength)
{
WaveFormat *format = &file->format;
@@ -1354,8 +1343,7 @@ PCM_Init(WaveFile *file, size_t datalength)
return 0;
}
static int
PCM_ConvertSint24ToSint32(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
static int PCM_ConvertSint24ToSint32(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
{
WaveFormat *format = &file->format;
WaveChunk *chunk = &file->chunk;
@@ -1406,8 +1394,7 @@ PCM_ConvertSint24ToSint32(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
return 0;
}
static int
PCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
static int PCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
{
WaveFormat *format = &file->format;
WaveChunk *chunk = &file->chunk;
@@ -1449,8 +1436,7 @@ PCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
return 0;
}
static WaveRiffSizeHint
WaveGetRiffSizeHint()
static WaveRiffSizeHint WaveGetRiffSizeHint()
{
const char *hint = SDL_GetHint(SDL_HINT_WAVE_RIFF_CHUNK_SIZE);
@@ -1469,8 +1455,7 @@ WaveGetRiffSizeHint()
return RiffSizeNoHint;
}
static WaveTruncationHint
WaveGetTruncationHint()
static WaveTruncationHint WaveGetTruncationHint()
{
const char *hint = SDL_GetHint(SDL_HINT_WAVE_TRUNCATION);
@@ -1489,8 +1474,7 @@ WaveGetTruncationHint()
return TruncNoHint;
}
static WaveFactChunkHint
WaveGetFactChunkHint()
static WaveFactChunkHint WaveGetFactChunkHint()
{
const char *hint = SDL_GetHint(SDL_HINT_WAVE_FACT_CHUNK);
@@ -1509,8 +1493,7 @@ WaveGetFactChunkHint()
return FactNoHint;
}
static void
WaveFreeChunkData(WaveChunk *chunk)
static void WaveFreeChunkData(WaveChunk *chunk)
{
if (chunk->data != NULL) {
SDL_free(chunk->data);
@@ -1519,8 +1502,7 @@ WaveFreeChunkData(WaveChunk *chunk)
chunk->size = 0;
}
static int
WaveNextChunk(SDL_RWops *src, WaveChunk *chunk)
static int WaveNextChunk(SDL_RWops *src, WaveChunk *chunk)
{
Uint32 chunkheader[2];
Sint64 nextposition = chunk->position + chunk->length;
@@ -1552,8 +1534,7 @@ WaveNextChunk(SDL_RWops *src, WaveChunk *chunk)
return 0;
}
static int
WaveReadPartialChunkData(SDL_RWops *src, WaveChunk *chunk, size_t length)
static int WaveReadPartialChunkData(SDL_RWops *src, WaveChunk *chunk, size_t length)
{
WaveFreeChunkData(chunk);
@@ -1562,7 +1543,7 @@ WaveReadPartialChunkData(SDL_RWops *src, WaveChunk *chunk, size_t length)
}
if (length > 0) {
chunk->data = (Uint8 *) SDL_malloc(length);
chunk->data = (Uint8 *)SDL_malloc(length);
if (chunk->data == NULL) {
return SDL_OutOfMemory();
}
@@ -1581,30 +1562,32 @@ WaveReadPartialChunkData(SDL_RWops *src, WaveChunk *chunk, size_t length)
return 0;
}
static int
WaveReadChunkData(SDL_RWops *src, WaveChunk *chunk)
static int WaveReadChunkData(SDL_RWops *src, WaveChunk *chunk)
{
return WaveReadPartialChunkData(src, chunk, chunk->length);
}
typedef struct WaveExtensibleGUID {
typedef struct WaveExtensibleGUID
{
Uint16 encoding;
Uint8 guid[16];
} WaveExtensibleGUID;
/* Some of the GUIDs that are used by WAVEFORMATEXTENSIBLE. */
#define WAVE_FORMATTAG_GUID(tag) {(tag) & 0xff, (tag) >> 8, 0, 0, 0, 0, 16, 0, 128, 0, 0, 170, 0, 56, 155, 113}
#define WAVE_FORMATTAG_GUID(tag) \
{ \
(tag) & 0xff, (tag) >> 8, 0, 0, 0, 0, 16, 0, 128, 0, 0, 170, 0, 56, 155, 113 \
}
static WaveExtensibleGUID extensible_guids[] = {
{PCM_CODE, WAVE_FORMATTAG_GUID(PCM_CODE)},
{MS_ADPCM_CODE, WAVE_FORMATTAG_GUID(MS_ADPCM_CODE)},
{IEEE_FLOAT_CODE, WAVE_FORMATTAG_GUID(IEEE_FLOAT_CODE)},
{ALAW_CODE, WAVE_FORMATTAG_GUID(ALAW_CODE)},
{MULAW_CODE, WAVE_FORMATTAG_GUID(MULAW_CODE)},
{IMA_ADPCM_CODE, WAVE_FORMATTAG_GUID(IMA_ADPCM_CODE)}
{ PCM_CODE, WAVE_FORMATTAG_GUID(PCM_CODE) },
{ MS_ADPCM_CODE, WAVE_FORMATTAG_GUID(MS_ADPCM_CODE) },
{ IEEE_FLOAT_CODE, WAVE_FORMATTAG_GUID(IEEE_FLOAT_CODE) },
{ ALAW_CODE, WAVE_FORMATTAG_GUID(ALAW_CODE) },
{ MULAW_CODE, WAVE_FORMATTAG_GUID(MULAW_CODE) },
{ IMA_ADPCM_CODE, WAVE_FORMATTAG_GUID(IMA_ADPCM_CODE) }
};
static Uint16
WaveGetFormatGUIDEncoding(WaveFormat *format)
static Uint16 WaveGetFormatGUIDEncoding(WaveFormat *format)
{
size_t i;
for (i = 0; i < SDL_arraysize(extensible_guids); i++) {
@@ -1615,8 +1598,7 @@ WaveGetFormatGUIDEncoding(WaveFormat *format)
return UNKNOWN_CODE;
}
static int
WaveReadFormat(WaveFile *file)
static int WaveReadFormat(WaveFile *file)
{
WaveChunk *chunk = &file->chunk;
WaveFormat *format = &file->format;
@@ -1677,8 +1659,7 @@ WaveReadFormat(WaveFile *file)
return 0;
}
static int
WaveCheckFormat(WaveFile *file, size_t datalength)
static int WaveCheckFormat(WaveFile *file, size_t datalength)
{
WaveFormat *format = &file->format;
@@ -1785,8 +1766,7 @@ WaveCheckFormat(WaveFile *file, size_t datalength)
return 0;
}
static int
WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len)
static int WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len)
{
int result;
Uint32 chunkcount = 0;
@@ -2051,7 +2031,7 @@ WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioSpec *spec, Uint8 **audio_buf,
SDL_zerop(spec);
spec->freq = format->frequency;
spec->channels = (Uint8)format->channels;
spec->samples = 4096; /* Good default buffer size */
spec->samples = 4096; /* Good default buffer size */
switch (format->encoding) {
case MS_ADPCM_CODE:
@@ -2148,8 +2128,7 @@ SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_b
/* Since the WAV memory is allocated in the shared library, it must also
be freed here. (Necessary under Win32, VC++)
*/
void
SDL_FreeWAV(Uint8 *audio_buf)
void SDL_FreeWAV(Uint8 *audio_buf)
{
SDL_free(audio_buf);
}

46
src/audio/SDL_wave.h
View File

@@ -26,14 +26,14 @@
/* Define values for Microsoft WAVE format */
/*******************************************/
/* FOURCC */
#define RIFF 0x46464952 /* "RIFF" */
#define WAVE 0x45564157 /* "WAVE" */
#define FACT 0x74636166 /* "fact" */
#define LIST 0x5453494c /* "LIST" */
#define BEXT 0x74786562 /* "bext" */
#define JUNK 0x4B4E554A /* "JUNK" */
#define FMT 0x20746D66 /* "fmt " */
#define DATA 0x61746164 /* "data" */
#define RIFF 0x46464952 /* "RIFF" */
#define WAVE 0x45564157 /* "WAVE" */
#define FACT 0x74636166 /* "fact" */
#define LIST 0x5453494c /* "LIST" */
#define BEXT 0x74786562 /* "bext" */
#define JUNK 0x4B4E554A /* "JUNK" */
#define FMT 0x20746D66 /* "fmt " */
#define DATA 0x61746164 /* "data" */
/* Format tags */
#define UNKNOWN_CODE 0x0000
#define PCM_CODE 0x0001
@@ -49,13 +49,13 @@
/* Stores the WAVE format information. */
typedef struct WaveFormat
{
Uint16 formattag; /* Raw value of the first field in the fmt chunk data. */
Uint16 encoding; /* Actual encoding, possibly from the extensible header. */
Uint16 channels; /* Number of channels. */
Uint32 frequency; /* Sampling rate in Hz. */
Uint32 byterate; /* Average bytes per second. */
Uint16 blockalign; /* Bytes per block. */
Uint16 bitspersample; /* Currently supported are 8, 16, 24, 32, and 4 for ADPCM. */
Uint16 formattag; /* Raw value of the first field in the fmt chunk data. */
Uint16 encoding; /* Actual encoding, possibly from the extensible header. */
Uint16 channels; /* Number of channels. */
Uint32 frequency; /* Sampling rate in Hz. */
Uint32 byterate; /* Average bytes per second. */
Uint16 blockalign; /* Bytes per block. */
Uint16 bitspersample; /* Currently supported are 8, 16, 24, 32, and 4 for ADPCM. */
/* Extra information size. Number of extra bytes starting at byte 18 in the
* fmt chunk data. This is at least 22 for the extensible header.
@@ -66,11 +66,12 @@ typedef struct WaveFormat
Uint16 validsamplebits;
Uint32 samplesperblock; /* For compressed formats. Can be zero. Actually 16 bits in the header. */
Uint32 channelmask;
Uint8 subformat[16]; /* A format GUID. */
Uint8 subformat[16]; /* A format GUID. */
} WaveFormat;
/* Stores information on the fact chunk. */
typedef struct WaveFact {
typedef struct WaveFact
{
/* Represents the state of the fact chunk in the WAVE file.
* Set to -1 if the fact chunk is invalid.
* Set to 0 if the fact chunk is not present
@@ -101,7 +102,8 @@ typedef struct WaveChunk
} WaveChunk;
/* Controls how the size of the RIFF chunk affects the loading of a WAVE file. */
typedef enum WaveRiffSizeHint {
typedef enum WaveRiffSizeHint
{
RiffSizeNoHint,
RiffSizeForce,
RiffSizeIgnoreZero,
@@ -110,7 +112,8 @@ typedef enum WaveRiffSizeHint {
} WaveRiffSizeHint;
/* Controls how a truncated WAVE file is handled. */
typedef enum WaveTruncationHint {
typedef enum WaveTruncationHint
{
TruncNoHint,
TruncVeryStrict,
TruncStrict,
@@ -119,7 +122,8 @@ typedef enum WaveTruncationHint {
} WaveTruncationHint;
/* Controls how the fact chunk affects the loading of a WAVE file. */
typedef enum WaveFactChunkHint {
typedef enum WaveFactChunkHint
{
FactNoHint,
FactTruncate,
FactStrict,
@@ -139,7 +143,7 @@ typedef struct WaveFile
*/
Sint64 sampleframes;
void *decoderdata; /* Some decoders require extra data for a state. */
void *decoderdata; /* Some decoders require extra data for a state. */
WaveRiffSizeHint riffhint;
WaveTruncationHint trunchint;

87
src/audio/aaudio/SDL_aaudio.c
View File

@@ -28,17 +28,17 @@
/* Debug */
#if 0
# define LOGI(...) SDL_Log(__VA_ARGS__);
#define LOGI(...) SDL_Log(__VA_ARGS__);
#else
# define LOGI(...)
#define LOGI(...)
#endif
typedef struct AAUDIO_Data
{
AAudioStreamBuilder *builder;
void *handle;
#define SDL_PROC(ret,func,params) ret (*func) params;
# include "SDL_aaudiofuncs.h"
#define SDL_PROC(ret, func, params) ret(*func) params;
#include "SDL_aaudiofuncs.h"
#undef SDL_PROC
} AAUDIO_Data;
static AAUDIO_Data ctx;
@@ -48,28 +48,27 @@ static SDL_AudioDevice *captureDevice = NULL;
static int aaudio_LoadFunctions(AAUDIO_Data *data)
{
#define SDL_PROC(ret,func,params) \
do { \
data->func = SDL_LoadFunction(data->handle, #func); \
if (! data->func) { \
return SDL_SetError("Couldn't load AAUDIO function %s: %s", #func, SDL_GetError()); \
} \
#define SDL_PROC(ret, func, params) \
do { \
data->func = SDL_LoadFunction(data->handle, #func); \
if (!data->func) { \
return SDL_SetError("Couldn't load AAUDIO function %s: %s", #func, SDL_GetError()); \
} \
} while (0);
#include "SDL_aaudiofuncs.h"
#undef SDL_PROC
return 0;
}
void aaudio_errorCallback( AAudioStream *stream, void *userData, aaudio_result_t error );
void aaudio_errorCallback( AAudioStream *stream, void *userData, aaudio_result_t error )
void aaudio_errorCallback(AAudioStream *stream, void *userData, aaudio_result_t error);
void aaudio_errorCallback(AAudioStream *stream, void *userData, aaudio_result_t error)
{
LOGI( "SDL aaudio_errorCallback: %d - %s", error, ctx.AAudio_convertResultToText( error ) );
LOGI("SDL aaudio_errorCallback: %d - %s", error, ctx.AAudio_convertResultToText(error));
}
#define LIB_AAUDIO_SO "libaaudio.so"
static int
aaudio_OpenDevice(_THIS, const char *devname)
static int aaudio_OpenDevice(_THIS, const char *devname)
{
struct SDL_PrivateAudioData *private;
SDL_bool iscapture = this->iscapture;
@@ -92,7 +91,7 @@ aaudio_OpenDevice(_THIS, const char *devname)
audioDevice = this;
}
this->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, (sizeof *this->hidden));
this->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, (sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
@@ -114,11 +113,11 @@ aaudio_OpenDevice(_THIS, const char *devname)
ctx.AAudioStreamBuilder_setFormat(ctx.builder, format);
}
ctx.AAudioStreamBuilder_setErrorCallback( ctx.builder, aaudio_errorCallback, private );
ctx.AAudioStreamBuilder_setErrorCallback(ctx.builder, aaudio_errorCallback, private);
LOGI("AAudio Try to open %u hz %u bit chan %u %s samples %u",
this->spec.freq, SDL_AUDIO_BITSIZE(this->spec.format),
this->spec.channels, (this->spec.format & 0x1000) ? "BE" : "LE", this->spec.samples);
this->spec.freq, SDL_AUDIO_BITSIZE(this->spec.format),
this->spec.channels, (this->spec.format & 0x1000) ? "BE" : "LE", this->spec.samples);
res = ctx.AAudioStreamBuilder_openStream(ctx.builder, &private->stream);
if (res != AAUDIO_OK) {
@@ -138,15 +137,15 @@ aaudio_OpenDevice(_THIS, const char *devname)
}
LOGI("AAudio Try to open %u hz %u bit chan %u %s samples %u",
this->spec.freq, SDL_AUDIO_BITSIZE(this->spec.format),
this->spec.channels, (this->spec.format & 0x1000) ? "BE" : "LE", this->spec.samples);
this->spec.freq, SDL_AUDIO_BITSIZE(this->spec.format),
this->spec.channels, (this->spec.format & 0x1000) ? "BE" : "LE", this->spec.samples);
SDL_CalculateAudioSpec(&this->spec);
/* Allocate mixing buffer */
if (!iscapture) {
private->mixlen = this->spec.size;
private->mixbuf = (Uint8 *) SDL_malloc(private->mixlen);
private->mixbuf = (Uint8 *)SDL_malloc(private->mixlen);
if (private->mixbuf == NULL) {
return SDL_OutOfMemory();
}
@@ -165,8 +164,7 @@ aaudio_OpenDevice(_THIS, const char *devname)
return 0;
}
static void
aaudio_CloseDevice(_THIS)
static void aaudio_CloseDevice(_THIS)
{
struct SDL_PrivateAudioData *private = this->hidden;
aaudio_result_t res;
@@ -200,20 +198,18 @@ aaudio_CloseDevice(_THIS)
SDL_free(this->hidden);
}
static Uint8 *
aaudio_GetDeviceBuf(_THIS)
static Uint8 *aaudio_GetDeviceBuf(_THIS)
{
struct SDL_PrivateAudioData *private = this->hidden;
return private->mixbuf;
}
static void
aaudio_PlayDevice(_THIS)
static void aaudio_PlayDevice(_THIS)
{
struct SDL_PrivateAudioData *private = this->hidden;
aaudio_result_t res;
int64_t timeoutNanoseconds = 1 * 1000 * 1000; /* 8 ms */
res = ctx.AAudioStream_write(private->stream, private->mixbuf, private->mixlen / private->frame_size, timeoutNanoseconds);
res = ctx.AAudioStream_write(private->stream, private->mixbuf, private->mixlen / private->frame_size, timeoutNanoseconds);
if (res < 0) {
LOGI("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
} else {
@@ -233,13 +229,12 @@ aaudio_PlayDevice(_THIS)
#endif
}
static int
aaudio_CaptureFromDevice(_THIS, void *buffer, int buflen)
static int aaudio_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
struct SDL_PrivateAudioData *private = this->hidden;
aaudio_result_t res;
int64_t timeoutNanoseconds = 8 * 1000 * 1000; /* 8 ms */
res = ctx.AAudioStream_read(private->stream, buffer, buflen / private->frame_size, timeoutNanoseconds);
res = ctx.AAudioStream_read(private->stream, buffer, buflen / private->frame_size, timeoutNanoseconds);
if (res < 0) {
LOGI("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
return -1;
@@ -248,8 +243,7 @@ aaudio_CaptureFromDevice(_THIS, void *buffer, int buflen)
return res * private->frame_size;
}
static void
aaudio_Deinitialize(void)
static void aaudio_Deinitialize(void)
{
LOGI(__func__);
if (ctx.handle) {
@@ -267,8 +261,7 @@ aaudio_Deinitialize(void)
LOGI("End AAUDIO %s", SDL_GetError());
}
static SDL_bool
aaudio_Init(SDL_AudioDriverImpl *impl)
static SDL_bool aaudio_Init(SDL_AudioDriverImpl *impl)
{
aaudio_result_t res;
LOGI(__func__);
@@ -343,7 +336,7 @@ void aaudio_PauseDevices(void)
/* TODO: Handle multiple devices? */
struct SDL_PrivateAudioData *private;
if (audioDevice != NULL && audioDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) audioDevice->hidden;
private = (struct SDL_PrivateAudioData *)audioDevice->hidden;
if (private->stream) {
aaudio_result_t res = ctx.AAudioStream_requestPause(private->stream);
@@ -364,7 +357,7 @@ void aaudio_PauseDevices(void)
}
if (captureDevice != NULL && captureDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) captureDevice->hidden;
private = (struct SDL_PrivateAudioData *)captureDevice->hidden;
if (private->stream) {
/* Pause() isn't implemented for 'capture', use Stop() */
@@ -392,7 +385,7 @@ void aaudio_ResumeDevices(void)
/* TODO: Handle multiple devices? */
struct SDL_PrivateAudioData *private;
if (audioDevice != NULL && audioDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) audioDevice->hidden;
private = (struct SDL_PrivateAudioData *)audioDevice->hidden;
if (private->resume) {
SDL_AtomicSet(&audioDevice->paused, 0);
@@ -410,7 +403,7 @@ void aaudio_ResumeDevices(void)
}
if (captureDevice != NULL && captureDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) captureDevice->hidden;
private = (struct SDL_PrivateAudioData *)captureDevice->hidden;
if (private->resume) {
SDL_AtomicSet(&captureDevice->paused, 0);
@@ -433,24 +426,24 @@ void aaudio_ResumeDevices(void)
None of the standard state queries indicate any problem in my testing. And the error callback doesn't actually get called.
But, AAudioStream_getTimestamp() does return AAUDIO_ERROR_INVALID_STATE
*/
SDL_bool aaudio_DetectBrokenPlayState( void )
SDL_bool aaudio_DetectBrokenPlayState(void)
{
struct SDL_PrivateAudioData *private;
int64_t framePosition, timeNanoseconds;
aaudio_result_t res;
if (audioDevice == NULL || !audioDevice->hidden ) {
if (audioDevice == NULL || !audioDevice->hidden) {
return SDL_FALSE;
}
private = audioDevice->hidden;
res = ctx.AAudioStream_getTimestamp( private->stream, CLOCK_MONOTONIC, &framePosition, &timeNanoseconds );
if ( res == AAUDIO_ERROR_INVALID_STATE ) {
aaudio_stream_state_t currentState = ctx.AAudioStream_getState( private->stream );
res = ctx.AAudioStream_getTimestamp(private->stream, CLOCK_MONOTONIC, &framePosition, &timeNanoseconds);
if (res == AAUDIO_ERROR_INVALID_STATE) {
aaudio_stream_state_t currentState = ctx.AAudioStream_getState(private->stream);
/* AAudioStream_getTimestamp() will also return AAUDIO_ERROR_INVALID_STATE while the stream is still initially starting. But we only care if it silently went invalid while playing. */
if ( currentState == AAUDIO_STREAM_STATE_STARTED ) {
LOGI( "SDL aaudio_DetectBrokenPlayState: detected invalid audio device state: AAudioStream_getTimestamp result=%d, framePosition=%lld, timeNanoseconds=%lld, getState=%d", (int)res, (long long)framePosition, (long long)timeNanoseconds, (int)currentState );
if (currentState == AAUDIO_STREAM_STATE_STARTED) {
LOGI("SDL aaudio_DetectBrokenPlayState: detected invalid audio device state: AAudioStream_getTimestamp result=%d, framePosition=%lld, timeNanoseconds=%lld, getState=%d", (int)res, (long long)framePosition, (long long)timeNanoseconds, (int)currentState);
return SDL_TRUE;
}
}

5
src/audio/aaudio/SDL_aaudio.h
View File

@@ -28,7 +28,7 @@
#include <aaudio/AAudio.h>
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
@@ -42,12 +42,11 @@ struct SDL_PrivateAudioData
/* Resume device if it was paused automatically */
int resume;
};
void aaudio_ResumeDevices(void);
void aaudio_PauseDevices(void);
SDL_bool aaudio_DetectBrokenPlayState(void);
#endif /* _SDL_aaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

111
src/audio/aaudio/SDL_aaudiofuncs.h
View File

@@ -19,62 +19,61 @@
3. This notice may not be removed or altered from any source distribution.
*/
#define SDL_PROC_UNUSED(ret,func,params)
#define SDL_PROC_UNUSED(ret, func, params)
SDL_PROC(const char *, AAudio_convertResultToText, (aaudio_result_t returnCode))
SDL_PROC(const char *, AAudio_convertStreamStateToText, (aaudio_stream_state_t state))
SDL_PROC(aaudio_result_t, AAudio_createStreamBuilder, (AAudioStreamBuilder** builder))
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setDeviceId, (AAudioStreamBuilder* builder, int32_t deviceId))
SDL_PROC(void, AAudioStreamBuilder_setSampleRate, (AAudioStreamBuilder* builder, int32_t sampleRate))
SDL_PROC(void, AAudioStreamBuilder_setChannelCount, (AAudioStreamBuilder* builder, int32_t channelCount))
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setSamplesPerFrame, (AAudioStreamBuilder* builder, int32_t samplesPerFrame))
SDL_PROC(void, AAudioStreamBuilder_setFormat, (AAudioStreamBuilder* builder, aaudio_format_t format))
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setSharingMode, (AAudioStreamBuilder* builder, aaudio_sharing_mode_t sharingMode))
SDL_PROC(void, AAudioStreamBuilder_setDirection, (AAudioStreamBuilder* builder, aaudio_direction_t direction))
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setBufferCapacityInFrames, (AAudioStreamBuilder* builder, int32_t numFrames))
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setPerformanceMode, (AAudioStreamBuilder* builder, aaudio_performance_mode_t mode))
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setUsage, (AAudioStreamBuilder* builder, aaudio_usage_t usage)) /* API 28 */
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setContentType, (AAudioStreamBuilder* builder, aaudio_content_type_t contentType)) /* API 28 */
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setInputPreset, (AAudioStreamBuilder* builder, aaudio_input_preset_t inputPreset)) /* API 28 */
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setAllowedCapturePolicy, (AAudioStreamBuilder* builder, aaudio_allowed_capture_policy_t capturePolicy)) /* API 29 */
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setSessionId, (AAudioStreamBuilder* builder, aaudio_session_id_t sessionId)) /* API 28 */
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setPrivacySensitive, (AAudioStreamBuilder* builder, bool privacySensitive)) /* API 30 */
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setDataCallback, (AAudioStreamBuilder* builder, AAudioStream_dataCallback callback, void *userData))
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setFramesPerDataCallback, (AAudioStreamBuilder* builder, int32_t numFrames))
SDL_PROC(void, AAudioStreamBuilder_setErrorCallback, (AAudioStreamBuilder* builder, AAudioStream_errorCallback callback, void *userData))
SDL_PROC(aaudio_result_t , AAudioStreamBuilder_openStream, (AAudioStreamBuilder* builder, AAudioStream** stream))
SDL_PROC(aaudio_result_t , AAudioStreamBuilder_delete, (AAudioStreamBuilder* builder))
SDL_PROC_UNUSED(aaudio_result_t , AAudioStream_release, (AAudioStream* stream)) /* API 30 */
SDL_PROC(aaudio_result_t , AAudioStream_close, (AAudioStream* stream))
SDL_PROC(aaudio_result_t , AAudioStream_requestStart, (AAudioStream* stream))
SDL_PROC(aaudio_result_t , AAudioStream_requestPause, (AAudioStream* stream))
SDL_PROC_UNUSED(aaudio_result_t , AAudioStream_requestFlush, (AAudioStream* stream))
SDL_PROC(aaudio_result_t , AAudioStream_requestStop, (AAudioStream* stream))
SDL_PROC(aaudio_stream_state_t, AAudioStream_getState, (AAudioStream* stream))
SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_waitForStateChange, (AAudioStream* stream, aaudio_stream_state_t inputState, aaudio_stream_state_t *nextState, int64_t timeoutNanoseconds))
SDL_PROC(aaudio_result_t, AAudioStream_read, (AAudioStream* stream, void *buffer, int32_t numFrames, int64_t timeoutNanoseconds))
SDL_PROC(aaudio_result_t, AAudioStream_write, (AAudioStream* stream, const void *buffer, int32_t numFrames, int64_t timeoutNanoseconds))
SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_setBufferSizeInFrames, (AAudioStream* stream, int32_t numFrames))
SDL_PROC_UNUSED(int32_t, AAudioStream_getBufferSizeInFrames, (AAudioStream* stream))
SDL_PROC_UNUSED(int32_t, AAudioStream_getFramesPerBurst, (AAudioStream* stream))
SDL_PROC_UNUSED(int32_t, AAudioStream_getBufferCapacityInFrames, (AAudioStream* stream))
SDL_PROC_UNUSED(int32_t, AAudioStream_getFramesPerDataCallback, (AAudioStream* stream))
SDL_PROC(int32_t, AAudioStream_getXRunCount, (AAudioStream* stream))
SDL_PROC(int32_t, AAudioStream_getSampleRate, (AAudioStream* stream))
SDL_PROC(int32_t, AAudioStream_getChannelCount, (AAudioStream* stream))
SDL_PROC_UNUSED(int32_t, AAudioStream_getSamplesPerFrame, (AAudioStream* stream))
SDL_PROC_UNUSED(int32_t, AAudioStream_getDeviceId, (AAudioStream* stream))
SDL_PROC(aaudio_format_t, AAudioStream_getFormat, (AAudioStream* stream))
SDL_PROC_UNUSED(aaudio_sharing_mode_t, AAudioStream_getSharingMode, (AAudioStream* stream))
SDL_PROC_UNUSED(aaudio_performance_mode_t, AAudioStream_getPerformanceMode, (AAudioStream* stream))
SDL_PROC_UNUSED(aaudio_direction_t, AAudioStream_getDirection, (AAudioStream* stream))
SDL_PROC_UNUSED(int64_t, AAudioStream_getFramesWritten, (AAudioStream* stream))
SDL_PROC_UNUSED(int64_t, AAudioStream_getFramesRead, (AAudioStream* stream))
SDL_PROC_UNUSED(aaudio_session_id_t, AAudioStream_getSessionId, (AAudioStream* stream)) /* API 28 */
SDL_PROC(aaudio_result_t, AAudioStream_getTimestamp, (AAudioStream* stream, clockid_t clockid, int64_t *framePosition, int64_t *timeNanoseconds))
SDL_PROC_UNUSED(aaudio_usage_t, AAudioStream_getUsage, (AAudioStream* stream)) /* API 28 */
SDL_PROC_UNUSED(aaudio_content_type_t, AAudioStream_getContentType, (AAudioStream* stream)) /* API 28 */
SDL_PROC_UNUSED(aaudio_input_preset_t, AAudioStream_getInputPreset, (AAudioStream* stream)) /* API 28 */
SDL_PROC_UNUSED(aaudio_allowed_capture_policy_t, AAudioStream_getAllowedCapturePolicy, ( AAudioStream* stream)) /* API 29 */
SDL_PROC_UNUSED(bool, AAudioStream_isPrivacySensitive, (AAudioStream* stream)) /* API 30 */
SDL_PROC(aaudio_result_t, AAudio_createStreamBuilder, (AAudioStreamBuilder * *builder))
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setDeviceId, (AAudioStreamBuilder * builder, int32_t deviceId))
SDL_PROC(void, AAudioStreamBuilder_setSampleRate, (AAudioStreamBuilder * builder, int32_t sampleRate))
SDL_PROC(void, AAudioStreamBuilder_setChannelCount, (AAudioStreamBuilder * builder, int32_t channelCount))
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setSamplesPerFrame, (AAudioStreamBuilder * builder, int32_t samplesPerFrame))
SDL_PROC(void, AAudioStreamBuilder_setFormat, (AAudioStreamBuilder * builder, aaudio_format_t format))
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setSharingMode, (AAudioStreamBuilder * builder, aaudio_sharing_mode_t sharingMode))
SDL_PROC(void, AAudioStreamBuilder_setDirection, (AAudioStreamBuilder * builder, aaudio_direction_t direction))
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setBufferCapacityInFrames, (AAudioStreamBuilder * builder, int32_t numFrames))
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setPerformanceMode, (AAudioStreamBuilder * builder, aaudio_performance_mode_t mode))
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setUsage, (AAudioStreamBuilder * builder, aaudio_usage_t usage)) /* API 28 */
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setContentType, (AAudioStreamBuilder * builder, aaudio_content_type_t contentType)) /* API 28 */
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setInputPreset, (AAudioStreamBuilder * builder, aaudio_input_preset_t inputPreset)) /* API 28 */
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setAllowedCapturePolicy, (AAudioStreamBuilder * builder, aaudio_allowed_capture_policy_t capturePolicy)) /* API 29 */
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setSessionId, (AAudioStreamBuilder * builder, aaudio_session_id_t sessionId)) /* API 28 */
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setPrivacySensitive, (AAudioStreamBuilder * builder, bool privacySensitive)) /* API 30 */
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setDataCallback, (AAudioStreamBuilder * builder, AAudioStream_dataCallback callback, void *userData))
SDL_PROC_UNUSED(void, AAudioStreamBuilder_setFramesPerDataCallback, (AAudioStreamBuilder * builder, int32_t numFrames))
SDL_PROC(void, AAudioStreamBuilder_setErrorCallback, (AAudioStreamBuilder * builder, AAudioStream_errorCallback callback, void *userData))
SDL_PROC(aaudio_result_t, AAudioStreamBuilder_openStream, (AAudioStreamBuilder * builder, AAudioStream **stream))
SDL_PROC(aaudio_result_t, AAudioStreamBuilder_delete, (AAudioStreamBuilder * builder))
SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_release, (AAudioStream * stream)) /* API 30 */
SDL_PROC(aaudio_result_t, AAudioStream_close, (AAudioStream * stream))
SDL_PROC(aaudio_result_t, AAudioStream_requestStart, (AAudioStream * stream))
SDL_PROC(aaudio_result_t, AAudioStream_requestPause, (AAudioStream * stream))
SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_requestFlush, (AAudioStream * stream))
SDL_PROC(aaudio_result_t, AAudioStream_requestStop, (AAudioStream * stream))
SDL_PROC(aaudio_stream_state_t, AAudioStream_getState, (AAudioStream * stream))
SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_waitForStateChange, (AAudioStream * stream, aaudio_stream_state_t inputState, aaudio_stream_state_t *nextState, int64_t timeoutNanoseconds))
SDL_PROC(aaudio_result_t, AAudioStream_read, (AAudioStream * stream, void *buffer, int32_t numFrames, int64_t timeoutNanoseconds))
SDL_PROC(aaudio_result_t, AAudioStream_write, (AAudioStream * stream, const void *buffer, int32_t numFrames, int64_t timeoutNanoseconds))
SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_setBufferSizeInFrames, (AAudioStream * stream, int32_t numFrames))
SDL_PROC_UNUSED(int32_t, AAudioStream_getBufferSizeInFrames, (AAudioStream * stream))
SDL_PROC_UNUSED(int32_t, AAudioStream_getFramesPerBurst, (AAudioStream * stream))
SDL_PROC_UNUSED(int32_t, AAudioStream_getBufferCapacityInFrames, (AAudioStream * stream))
SDL_PROC_UNUSED(int32_t, AAudioStream_getFramesPerDataCallback, (AAudioStream * stream))
SDL_PROC(int32_t, AAudioStream_getXRunCount, (AAudioStream * stream))
SDL_PROC(int32_t, AAudioStream_getSampleRate, (AAudioStream * stream))
SDL_PROC(int32_t, AAudioStream_getChannelCount, (AAudioStream * stream))
SDL_PROC_UNUSED(int32_t, AAudioStream_getSamplesPerFrame, (AAudioStream * stream))
SDL_PROC_UNUSED(int32_t, AAudioStream_getDeviceId, (AAudioStream * stream))
SDL_PROC(aaudio_format_t, AAudioStream_getFormat, (AAudioStream * stream))
SDL_PROC_UNUSED(aaudio_sharing_mode_t, AAudioStream_getSharingMode, (AAudioStream * stream))
SDL_PROC_UNUSED(aaudio_performance_mode_t, AAudioStream_getPerformanceMode, (AAudioStream * stream))
SDL_PROC_UNUSED(aaudio_direction_t, AAudioStream_getDirection, (AAudioStream * stream))
SDL_PROC_UNUSED(int64_t, AAudioStream_getFramesWritten, (AAudioStream * stream))
SDL_PROC_UNUSED(int64_t, AAudioStream_getFramesRead, (AAudioStream * stream))
SDL_PROC_UNUSED(aaudio_session_id_t, AAudioStream_getSessionId, (AAudioStream * stream)) /* API 28 */
SDL_PROC(aaudio_result_t, AAudioStream_getTimestamp, (AAudioStream * stream, clockid_t clockid, int64_t *framePosition, int64_t *timeNanoseconds))
SDL_PROC_UNUSED(aaudio_usage_t, AAudioStream_getUsage, (AAudioStream * stream)) /* API 28 */
SDL_PROC_UNUSED(aaudio_content_type_t, AAudioStream_getContentType, (AAudioStream * stream)) /* API 28 */
SDL_PROC_UNUSED(aaudio_input_preset_t, AAudioStream_getInputPreset, (AAudioStream * stream)) /* API 28 */
SDL_PROC_UNUSED(aaudio_allowed_capture_policy_t, AAudioStream_getAllowedCapturePolicy, (AAudioStream * stream)) /* API 29 */
SDL_PROC_UNUSED(bool, AAudioStream_isPrivacySensitive, (AAudioStream * stream)) /* API 30 */

354
src/audio/alsa/SDL_alsa_audio.c
View File

@@ -34,7 +34,7 @@
/* Allow access to a raw mixing buffer */
#include <sys/types.h>
#include <signal.h> /* For kill() */
#include <signal.h> /* For kill() */
#include <string.h>
#include "../SDL_audio_c.h"
@@ -43,64 +43,46 @@
#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
#endif
static int (*ALSA_snd_pcm_open)
(snd_pcm_t **, const char *, snd_pcm_stream_t, int);
static int (*ALSA_snd_pcm_close) (snd_pcm_t * pcm);
static snd_pcm_sframes_t (*ALSA_snd_pcm_writei)
(snd_pcm_t *, const void *, snd_pcm_uframes_t);
static snd_pcm_sframes_t (*ALSA_snd_pcm_readi)
(snd_pcm_t *, void *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_recover) (snd_pcm_t *, int, int);
static int (*ALSA_snd_pcm_prepare) (snd_pcm_t *);
static int (*ALSA_snd_pcm_drain) (snd_pcm_t *);
static const char *(*ALSA_snd_strerror) (int);
static size_t(*ALSA_snd_pcm_hw_params_sizeof) (void);
static size_t(*ALSA_snd_pcm_sw_params_sizeof) (void);
static void (*ALSA_snd_pcm_hw_params_copy)
(snd_pcm_hw_params_t *, const snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_hw_params_any) (snd_pcm_t *, snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_hw_params_set_access)
(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_access_t);
static int (*ALSA_snd_pcm_hw_params_set_format)
(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_format_t);
static int (*ALSA_snd_pcm_hw_params_set_channels)
(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int);
static int (*ALSA_snd_pcm_hw_params_get_channels)
(const snd_pcm_hw_params_t *, unsigned int *);
static int (*ALSA_snd_pcm_hw_params_set_rate_near)
(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
static int (*ALSA_snd_pcm_hw_params_set_period_size_near)
(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *);
static int (*ALSA_snd_pcm_hw_params_get_period_size)
(const snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *);
static int (*ALSA_snd_pcm_hw_params_set_periods_min)
(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
static int (*ALSA_snd_pcm_hw_params_set_periods_first)
(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
static int (*ALSA_snd_pcm_hw_params_get_periods)
(const snd_pcm_hw_params_t *, unsigned int *, int *);
static int (*ALSA_snd_pcm_hw_params_set_buffer_size_near)
(snd_pcm_t *pcm, snd_pcm_hw_params_t *, snd_pcm_uframes_t *);
static int (*ALSA_snd_pcm_hw_params_get_buffer_size)
(const snd_pcm_hw_params_t *, snd_pcm_uframes_t *);
static int (*ALSA_snd_pcm_hw_params) (snd_pcm_t *, snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_sw_params_current) (snd_pcm_t *,
snd_pcm_sw_params_t *);
static int (*ALSA_snd_pcm_sw_params_set_start_threshold)
(snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_sw_params) (snd_pcm_t *, snd_pcm_sw_params_t *);
static int (*ALSA_snd_pcm_nonblock) (snd_pcm_t *, int);
static int (*ALSA_snd_pcm_open)(snd_pcm_t **, const char *, snd_pcm_stream_t, int);
static int (*ALSA_snd_pcm_close)(snd_pcm_t *pcm);
static snd_pcm_sframes_t (*ALSA_snd_pcm_writei)(snd_pcm_t *, const void *, snd_pcm_uframes_t);
static snd_pcm_sframes_t (*ALSA_snd_pcm_readi)(snd_pcm_t *, void *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_recover)(snd_pcm_t *, int, int);
static int (*ALSA_snd_pcm_prepare)(snd_pcm_t *);
static int (*ALSA_snd_pcm_drain)(snd_pcm_t *);
static const char *(*ALSA_snd_strerror)(int);
static size_t (*ALSA_snd_pcm_hw_params_sizeof)(void);
static size_t (*ALSA_snd_pcm_sw_params_sizeof)(void);
static void (*ALSA_snd_pcm_hw_params_copy)(snd_pcm_hw_params_t *, const snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_hw_params_any)(snd_pcm_t *, snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_hw_params_set_access)(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_access_t);
static int (*ALSA_snd_pcm_hw_params_set_format)(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_format_t);
static int (*ALSA_snd_pcm_hw_params_set_channels)(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int);
static int (*ALSA_snd_pcm_hw_params_get_channels)(const snd_pcm_hw_params_t *, unsigned int *);
static int (*ALSA_snd_pcm_hw_params_set_rate_near)(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
static int (*ALSA_snd_pcm_hw_params_set_period_size_near)(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *);
static int (*ALSA_snd_pcm_hw_params_get_period_size)(const snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *);
static int (*ALSA_snd_pcm_hw_params_set_periods_min)(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
static int (*ALSA_snd_pcm_hw_params_set_periods_first)(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
static int (*ALSA_snd_pcm_hw_params_get_periods)(const snd_pcm_hw_params_t *, unsigned int *, int *);
static int (*ALSA_snd_pcm_hw_params_set_buffer_size_near)(snd_pcm_t *pcm, snd_pcm_hw_params_t *, snd_pcm_uframes_t *);
static int (*ALSA_snd_pcm_hw_params_get_buffer_size)(const snd_pcm_hw_params_t *, snd_pcm_uframes_t *);
static int (*ALSA_snd_pcm_hw_params)(snd_pcm_t *, snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_sw_params_current)(snd_pcm_t *,
snd_pcm_sw_params_t *);
static int (*ALSA_snd_pcm_sw_params_set_start_threshold)(snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_sw_params)(snd_pcm_t *, snd_pcm_sw_params_t *);
static int (*ALSA_snd_pcm_nonblock)(snd_pcm_t *, int);
static int (*ALSA_snd_pcm_wait)(snd_pcm_t *, int);
static int (*ALSA_snd_pcm_sw_params_set_avail_min)
(snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_sw_params_set_avail_min)(snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_reset)(snd_pcm_t *);
static int (*ALSA_snd_device_name_hint) (int, const char *, void ***);
static char* (*ALSA_snd_device_name_get_hint) (const void *, const char *);
static int (*ALSA_snd_device_name_free_hint) (void **);
static int (*ALSA_snd_device_name_hint)(int, const char *, void ***);
static char *(*ALSA_snd_device_name_get_hint)(const void *, const char *);
static int (*ALSA_snd_device_name_free_hint)(void **);
static snd_pcm_sframes_t (*ALSA_snd_pcm_avail)(snd_pcm_t *);
#ifdef SND_CHMAP_API_VERSION
static snd_pcm_chmap_t* (*ALSA_snd_pcm_get_chmap) (snd_pcm_t *);
static int (*ALSA_snd_pcm_chmap_print) (const snd_pcm_chmap_t *map, size_t maxlen, char *buf);
static snd_pcm_chmap_t *(*ALSA_snd_pcm_get_chmap)(snd_pcm_t *);
static int (*ALSA_snd_pcm_chmap_print)(const snd_pcm_chmap_t *map, size_t maxlen, char *buf);
#endif
#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
@@ -110,8 +92,7 @@ static int (*ALSA_snd_pcm_chmap_print) (const snd_pcm_chmap_t *map, size_t maxle
static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC;
static void *alsa_handle = NULL;
static int
load_alsa_sym(const char *fn, void **addr)
static int load_alsa_sym(const char *fn, void **addr)
{
*addr = SDL_LoadFunction(alsa_handle, fn);
if (*addr == NULL) {
@@ -123,14 +104,14 @@ load_alsa_sym(const char *fn, void **addr)
}
/* cast funcs to char* first, to please GCC's strict aliasing rules. */
#define SDL_ALSA_SYM(x) \
if (!load_alsa_sym(#x, (void **) (char *) &ALSA_##x)) return -1
#define SDL_ALSA_SYM(x) \
if (!load_alsa_sym(#x, (void **)(char *)&ALSA_##x)) \
return -1
#else
#define SDL_ALSA_SYM(x) ALSA_##x = x
#endif
static int
load_alsa_syms(void)
static int load_alsa_syms(void)
{
SDL_ALSA_SYM(snd_pcm_open);
SDL_ALSA_SYM(snd_pcm_close);
@@ -180,8 +161,7 @@ load_alsa_syms(void)
#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
static void
UnloadALSALibrary(void)
static void UnloadALSALibrary(void)
{
if (alsa_handle != NULL) {
SDL_UnloadObject(alsa_handle);
@@ -189,8 +169,7 @@ UnloadALSALibrary(void)
}
}
static int
LoadALSALibrary(void)
static int LoadALSALibrary(void)
{
int retval = 0;
if (alsa_handle == NULL) {
@@ -210,13 +189,11 @@ LoadALSALibrary(void)
#else
static void
UnloadALSALibrary(void)
static void UnloadALSALibrary(void)
{
}
static int
LoadALSALibrary(void)
static int LoadALSALibrary(void)
{
load_alsa_syms();
return 0;
@@ -224,8 +201,7 @@ LoadALSALibrary(void)
#endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */
static const char *
get_audio_device(void *handle, const int channels)
static const char *get_audio_device(void *handle, const int channels)
{
const char *device;
@@ -234,7 +210,7 @@ get_audio_device(void *handle, const int channels)
}
/* !!! FIXME: we also check "SDL_AUDIO_DEVICE_NAME" at the higher level. */
device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */
device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */
if (device != NULL) {
return device;
}
@@ -248,49 +224,50 @@ get_audio_device(void *handle, const int channels)
return "default";
}
/* This function waits until it is possible to write a full sound buffer */
static void
ALSA_WaitDevice(_THIS)
static void ALSA_WaitDevice(_THIS)
{
#if SDL_ALSA_NON_BLOCKING
const snd_pcm_sframes_t needed = (snd_pcm_sframes_t) this->spec.samples;
const snd_pcm_sframes_t needed = (snd_pcm_sframes_t)this->spec.samples;
while (SDL_AtomicGet(&this->enabled)) {
const snd_pcm_sframes_t rc = ALSA_snd_pcm_avail(this->hidden->pcm_handle);
if ((rc < 0) && (rc != -EAGAIN)) {
/* Hmm, not much we can do - abort */
fprintf(stderr, "ALSA snd_pcm_avail failed (unrecoverable): %s\n",
ALSA_snd_strerror(rc));
ALSA_snd_strerror(rc));
SDL_OpenedAudioDeviceDisconnected(this);
return;
} else if (rc < needed) {
const Uint32 delay = ((needed - (SDL_max(rc, 0))) * 1000) / this->spec.freq;
SDL_Delay(SDL_max(delay, 10));
} else {
break; /* ready to go! */
break; /* ready to go! */
}
}
#endif
}
/* !!! FIXME: is there a channel swizzler in alsalib instead? */
/*
* https://bugzilla.libsdl.org/show_bug.cgi?id=110
* "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
* and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
*/
#define SWIZ6(T) \
static void swizzle_alsa_channels_6_##T(void *buffer, const Uint32 bufferlen) { \
T *ptr = (T *) buffer; \
Uint32 i; \
for (i = 0; i < bufferlen; i++, ptr += 6) { \
T tmp; \
tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \
tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \
} \
}
#define SWIZ6(T) \
static void swizzle_alsa_channels_6_##T(void *buffer, const Uint32 bufferlen) \
{ \
T *ptr = (T *)buffer; \
Uint32 i; \
for (i = 0; i < bufferlen; i++, ptr += 6) { \
T tmp; \
tmp = ptr[2]; \
ptr[2] = ptr[4]; \
ptr[4] = tmp; \
tmp = ptr[3]; \
ptr[3] = ptr[5]; \
ptr[5] = tmp; \
} \
}
/* !!! FIXME: is there a channel swizzler in alsalib instead? */
/* !!! FIXME: this screams for a SIMD shuffle operation. */
@@ -299,31 +276,32 @@ static void swizzle_alsa_channels_6_##T(void *buffer, const Uint32 bufferlen) {
* For Linux ALSA, this appears to be FL-FR-RL-RR-C-LFE-SL-SR
* and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-SL-SR-RL-RR"
*/
#define SWIZ8(T) \
static void swizzle_alsa_channels_8_##T(void *buffer, const Uint32 bufferlen) { \
T *ptr = (T *) buffer; \
Uint32 i; \
for (i = 0; i < bufferlen; i++, ptr += 6) { \
const T center = ptr[2]; \
const T subwoofer = ptr[3]; \
const T side_left = ptr[4]; \
const T side_right = ptr[5]; \
const T rear_left = ptr[6]; \
const T rear_right = ptr[7]; \
ptr[2] = rear_left; \
ptr[3] = rear_right; \
ptr[4] = center; \
ptr[5] = subwoofer; \
ptr[6] = side_left; \
ptr[7] = side_right; \
} \
}
#define SWIZ8(T) \
static void swizzle_alsa_channels_8_##T(void *buffer, const Uint32 bufferlen) \
{ \
T *ptr = (T *)buffer; \
Uint32 i; \
for (i = 0; i < bufferlen; i++, ptr += 6) { \
const T center = ptr[2]; \
const T subwoofer = ptr[3]; \
const T side_left = ptr[4]; \
const T side_right = ptr[5]; \
const T rear_left = ptr[6]; \
const T rear_right = ptr[7]; \
ptr[2] = rear_left; \
ptr[3] = rear_right; \
ptr[4] = center; \
ptr[5] = subwoofer; \
ptr[6] = side_left; \
ptr[7] = side_right; \
} \
}
#define CHANNEL_SWIZZLE(x) \
x(Uint64) \
x(Uint32) \
x(Uint16) \
x(Uint8)
x(Uint64) \
x(Uint32) \
x(Uint16) \
x(Uint8)
CHANNEL_SWIZZLE(SWIZ6)
CHANNEL_SWIZZLE(SWIZ8)
@@ -332,53 +310,59 @@ CHANNEL_SWIZZLE(SWIZ8)
#undef SWIZ6
#undef SWIZ8
/*
* Called right before feeding this->hidden->mixbuf to the hardware. Swizzle
* channels from Windows/Mac order to the format alsalib will want.
*/
static void
swizzle_alsa_channels(_THIS, void *buffer, Uint32 bufferlen)
static void swizzle_alsa_channels(_THIS, void *buffer, Uint32 bufferlen)
{
switch (this->spec.channels) {
#define CHANSWIZ(chans) \
case chans: \
switch ((this->spec.format & (0xFF))) { \
case 8: swizzle_alsa_channels_##chans##_Uint8(buffer, bufferlen); break; \
case 16: swizzle_alsa_channels_##chans##_Uint16(buffer, bufferlen); break; \
case 32: swizzle_alsa_channels_##chans##_Uint32(buffer, bufferlen); break; \
case 64: swizzle_alsa_channels_##chans##_Uint64(buffer, bufferlen); break; \
default: SDL_assert(!"unhandled bitsize"); break; \
} \
return;
#define CHANSWIZ(chans) \
case chans: \
switch ((this->spec.format & (0xFF))) { \
case 8: \
swizzle_alsa_channels_##chans##_Uint8(buffer, bufferlen); \
break; \
case 16: \
swizzle_alsa_channels_##chans##_Uint16(buffer, bufferlen); \
break; \
case 32: \
swizzle_alsa_channels_##chans##_Uint32(buffer, bufferlen); \
break; \
case 64: \
swizzle_alsa_channels_##chans##_Uint64(buffer, bufferlen); \
break; \
default: \
SDL_assert(!"unhandled bitsize"); \
break; \
} \
return;
CHANSWIZ(6);
CHANSWIZ(8);
#undef CHANSWIZ
default: break;
#undef CHANSWIZ
default:
break;
}
}
#ifdef SND_CHMAP_API_VERSION
/* Some devices have the right channel map, no swizzling necessary */
static void
no_swizzle(_THIS, void *buffer, Uint32 bufferlen)
static void no_swizzle(_THIS, void *buffer, Uint32 bufferlen)
{
}
#endif /* SND_CHMAP_API_VERSION */
static void
ALSA_PlayDevice(_THIS)
static void ALSA_PlayDevice(_THIS)
{
const Uint8 *sample_buf = (const Uint8 *) this->hidden->mixbuf;
const Uint8 *sample_buf = (const Uint8 *)this->hidden->mixbuf;
const int frame_size = ((SDL_AUDIO_BITSIZE(this->spec.format)) / 8) *
this->spec.channels;
snd_pcm_uframes_t frames_left = ((snd_pcm_uframes_t) this->spec.samples);
this->spec.channels;
snd_pcm_uframes_t frames_left = ((snd_pcm_uframes_t)this->spec.samples);
this->hidden->swizzle_func(this, this->hidden->mixbuf, frames_left);
while ( frames_left > 0 && SDL_AtomicGet(&this->enabled) ) {
while (frames_left > 0 && SDL_AtomicGet(&this->enabled)) {
int status = ALSA_snd_pcm_writei(this->hidden->pcm_handle,
sample_buf, frames_left);
@@ -410,29 +394,27 @@ ALSA_PlayDevice(_THIS)
}
}
static Uint8 *
ALSA_GetDeviceBuf(_THIS)
static Uint8 *ALSA_GetDeviceBuf(_THIS)
{
return this->hidden->mixbuf;
}
static int
ALSA_CaptureFromDevice(_THIS, void *buffer, int buflen)
static int ALSA_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
Uint8 *sample_buf = (Uint8 *) buffer;
Uint8 *sample_buf = (Uint8 *)buffer;
const int frame_size = ((SDL_AUDIO_BITSIZE(this->spec.format)) / 8) *
this->spec.channels;
this->spec.channels;
const int total_frames = buflen / frame_size;
snd_pcm_uframes_t frames_left = total_frames;
snd_pcm_uframes_t wait_time = frame_size / 2;
SDL_assert((buflen % frame_size) == 0);
while ( frames_left > 0 && SDL_AtomicGet(&this->enabled) ) {
while (frames_left > 0 && SDL_AtomicGet(&this->enabled)) {
int status;
status = ALSA_snd_pcm_readi(this->hidden->pcm_handle,
sample_buf, frames_left);
sample_buf, frames_left);
if (status == -EAGAIN) {
ALSA_snd_pcm_wait(this->hidden->pcm_handle, wait_time);
@@ -459,14 +441,12 @@ ALSA_CaptureFromDevice(_THIS, void *buffer, int buflen)
return (total_frames - frames_left) * frame_size;
}
static void
ALSA_FlushCapture(_THIS)
static void ALSA_FlushCapture(_THIS)
{
ALSA_snd_pcm_reset(this->hidden->pcm_handle);
}
static void
ALSA_CloseDevice(_THIS)
static void ALSA_CloseDevice(_THIS)
{
if (this->hidden->pcm_handle) {
/* Wait for the submitted audio to drain
@@ -481,8 +461,7 @@ ALSA_CloseDevice(_THIS)
SDL_free(this->hidden);
}
static int
ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params)
static int ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params)
{
int status;
snd_pcm_hw_params_t *hwparams;
@@ -496,49 +475,48 @@ ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params)
/* Attempt to match the period size to the requested buffer size */
persize = this->spec.samples;
status = ALSA_snd_pcm_hw_params_set_period_size_near(
this->hidden->pcm_handle, hwparams, &persize, NULL);
if ( status < 0 ) {
this->hidden->pcm_handle, hwparams, &persize, NULL);
if (status < 0) {
return -1;
}
/* Need to at least double buffer */
periods = 2;
status = ALSA_snd_pcm_hw_params_set_periods_min(
this->hidden->pcm_handle, hwparams, &periods, NULL);
if ( status < 0 ) {
this->hidden->pcm_handle, hwparams, &periods, NULL);
if (status < 0) {
return -1;
}
status = ALSA_snd_pcm_hw_params_set_periods_first(
this->hidden->pcm_handle, hwparams, &periods, NULL);
if ( status < 0 ) {
this->hidden->pcm_handle, hwparams, &periods, NULL);
if (status < 0) {
return -1;
}
/* "set" the hardware with the desired parameters */
status = ALSA_snd_pcm_hw_params(this->hidden->pcm_handle, hwparams);
if ( status < 0 ) {
if (status < 0) {
return -1;
}
this->spec.samples = persize;
/* This is useful for debugging */
if ( SDL_getenv("SDL_AUDIO_ALSA_DEBUG") ) {
if (SDL_getenv("SDL_AUDIO_ALSA_DEBUG")) {
snd_pcm_uframes_t bufsize;
ALSA_snd_pcm_hw_params_get_buffer_size(hwparams, &bufsize);
fprintf(stderr,
"ALSA: period size = %ld, periods = %u, buffer size = %lu\n",
persize, periods, bufsize);
"ALSA: period size = %ld, periods = %u, buffer size = %lu\n",
persize, periods, bufsize);
}
return 0;
}
static int
ALSA_OpenDevice(_THIS, const char *devname)
static int ALSA_OpenDevice(_THIS, const char *devname)
{
int status = 0;
SDL_bool iscapture = this->iscapture;
@@ -565,9 +543,9 @@ ALSA_OpenDevice(_THIS, const char *devname)
/* Open the audio device */
/* Name of device should depend on # channels in spec */
status = ALSA_snd_pcm_open(&pcm_handle,
get_audio_device(this->handle, this->spec.channels),
iscapture ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
SND_PCM_NONBLOCK);
get_audio_device(this->handle, this->spec.channels),
iscapture ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
SND_PCM_NONBLOCK);
if (status < 0) {
return SDL_SetError("ALSA: Couldn't open audio device: %s", ALSA_snd_strerror(status));
@@ -704,18 +682,18 @@ ALSA_OpenDevice(_THIS, const char *devname)
/* Allocate mixing buffer */
if (!iscapture) {
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
this->hidden->mixbuf = (Uint8 *)SDL_malloc(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->hidden->mixlen);
}
#if !SDL_ALSA_NON_BLOCKING
#if !SDL_ALSA_NON_BLOCKING
if (!iscapture) {
ALSA_snd_pcm_nonblock(pcm_handle, 0);
}
#endif
#endif
/* We're ready to rock and roll. :-) */
return 0;
@@ -728,10 +706,9 @@ typedef struct ALSA_Device
struct ALSA_Device *next;
} ALSA_Device;
static void
add_device(const int iscapture, const char *name, void *hint, ALSA_Device **pSeen)
static void add_device(const int iscapture, const char *name, void *hint, ALSA_Device **pSeen)
{
ALSA_Device *dev = SDL_malloc(sizeof (ALSA_Device));
ALSA_Device *dev = SDL_malloc(sizeof(ALSA_Device));
char *desc;
char *handle = NULL;
char *ptr;
@@ -751,7 +728,7 @@ add_device(const int iscapture, const char *name, void *hint, ALSA_Device **pSee
return;
}
} else {
desc = (char *) name;
desc = (char *)name;
}
SDL_assert(name != NULL);
@@ -788,11 +765,9 @@ add_device(const int iscapture, const char *name, void *hint, ALSA_Device **pSee
*pSeen = dev;
}
static ALSA_Device *hotplug_devices = NULL;
static void
ALSA_HotplugIteration(void)
static void ALSA_HotplugIteration(void)
{
void **hints = NULL;
ALSA_Device *dev;
@@ -807,7 +782,7 @@ ALSA_HotplugIteration(void)
int bestmatch = 0xFFFF;
size_t match_len = 0;
int defaultdev = -1;
static const char * const prefixes[] = {
static const char *const prefixes[] = {
"hw:", "sysdefault:", "default:", NULL
};
@@ -850,7 +825,7 @@ ALSA_HotplugIteration(void)
/* if we didn't find a device name prefix we like at all... */
if ((match == NULL) && (defaultdev != i)) {
continue; /* ...skip anything that isn't the default device. */
continue; /* ...skip anything that isn't the default device. */
}
name = ALSA_snd_device_name_get_hint(hints[i], "NAME");
@@ -876,7 +851,7 @@ ALSA_HotplugIteration(void)
prev = NULL;
for (dev = unseen; dev; dev = next) {
next = dev->next;
if ( (SDL_strcmp(dev->name, name) == 0) && (((isinput) && dev->iscapture) || ((isoutput) && !dev->iscapture)) ) {
if ((SDL_strcmp(dev->name, name) == 0) && (((isinput) && dev->iscapture) || ((isoutput) && !dev->iscapture))) {
if (prev) {
prev->next = next;
} else {
@@ -908,7 +883,7 @@ ALSA_HotplugIteration(void)
ALSA_snd_device_name_free_hint(hints);
hotplug_devices = seen; /* now we have a known-good list of attached devices. */
hotplug_devices = seen; /* now we have a known-good list of attached devices. */
/* report anything still in unseen as removed. */
for (dev = unseen; dev; dev = next) {
@@ -925,8 +900,7 @@ ALSA_HotplugIteration(void)
static SDL_atomic_t ALSA_hotplug_shutdown;
static SDL_Thread *ALSA_hotplug_thread;
static int SDLCALL
ALSA_HotplugThread(void *arg)
static int SDLCALL ALSA_HotplugThread(void *arg)
{
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_LOW);
@@ -937,17 +911,16 @@ ALSA_HotplugThread(void *arg)
SDL_Delay(100);
}
ALSA_HotplugIteration(); /* run the check. */
ALSA_HotplugIteration(); /* run the check. */
}
return 0;
}
#endif
static void
ALSA_DetectDevices(void)
static void ALSA_DetectDevices(void)
{
ALSA_HotplugIteration(); /* run once now before a thread continues to check. */
ALSA_HotplugIteration(); /* run once now before a thread continues to check. */
#if SDL_ALSA_HOTPLUG_THREAD
SDL_AtomicSet(&ALSA_hotplug_shutdown, 0);
@@ -956,8 +929,7 @@ ALSA_DetectDevices(void)
#endif
}
static void
ALSA_Deinitialize(void)
static void ALSA_Deinitialize(void)
{
ALSA_Device *dev;
ALSA_Device *next;
@@ -982,8 +954,7 @@ ALSA_Deinitialize(void)
UnloadALSALibrary();
}
static SDL_bool
ALSA_Init(SDL_AudioDriverImpl * impl)
static SDL_bool ALSA_Init(SDL_AudioDriverImpl *impl)
{
if (LoadALSALibrary() < 0) {
return SDL_FALSE;
@@ -1003,10 +974,9 @@ ALSA_Init(SDL_AudioDriverImpl * impl)
impl->HasCaptureSupport = SDL_TRUE;
impl->SupportsNonPow2Samples = SDL_TRUE;
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE; /* this audio target is available. */
}
AudioBootStrap ALSA_bootstrap = {
"alsa", "ALSA PCM audio", ALSA_Init, SDL_FALSE
};

2
src/audio/alsa/SDL_alsa_audio.h
View File

@@ -28,7 +28,7 @@
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{

40
src/audio/android/SDL_androidaudio.c
View File

@@ -31,11 +31,10 @@
#include <android/log.h>
static SDL_AudioDevice* audioDevice = NULL;
static SDL_AudioDevice* captureDevice = NULL;
static SDL_AudioDevice *audioDevice = NULL;
static SDL_AudioDevice *captureDevice = NULL;
static int
ANDROIDAUDIO_OpenDevice(_THIS, const char *devname)
static int ANDROIDAUDIO_OpenDevice(_THIS, const char *devname)
{
SDL_AudioFormat test_format;
SDL_bool iscapture = this->iscapture;
@@ -49,7 +48,7 @@ ANDROIDAUDIO_OpenDevice(_THIS, const char *devname)
audioDevice = this;
}
this->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, (sizeof *this->hidden));
this->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, (sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
@@ -77,32 +76,27 @@ ANDROIDAUDIO_OpenDevice(_THIS, const char *devname)
return 0;
}
static void
ANDROIDAUDIO_PlayDevice(_THIS)
static void ANDROIDAUDIO_PlayDevice(_THIS)
{
Android_JNI_WriteAudioBuffer();
}
static Uint8 *
ANDROIDAUDIO_GetDeviceBuf(_THIS)
static Uint8 *ANDROIDAUDIO_GetDeviceBuf(_THIS)
{
return Android_JNI_GetAudioBuffer();
}
static int
ANDROIDAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
static int ANDROIDAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
return Android_JNI_CaptureAudioBuffer(buffer, buflen);
}
static void
ANDROIDAUDIO_FlushCapture(_THIS)
static void ANDROIDAUDIO_FlushCapture(_THIS)
{
Android_JNI_FlushCapturedAudio();
}
static void
ANDROIDAUDIO_CloseDevice(_THIS)
static void ANDROIDAUDIO_CloseDevice(_THIS)
{
/* At this point SDL_CloseAudioDevice via close_audio_device took care of terminating the audio thread
so it's safe to terminate the Java side buffer and AudioTrack
@@ -118,8 +112,7 @@ ANDROIDAUDIO_CloseDevice(_THIS)
SDL_free(this->hidden);
}
static SDL_bool
ANDROIDAUDIO_Init(SDL_AudioDriverImpl * impl)
static SDL_bool ANDROIDAUDIO_Init(SDL_AudioDriverImpl *impl)
{
/* Set the function pointers */
impl->OpenDevice = ANDROIDAUDIO_OpenDevice;
@@ -134,7 +127,7 @@ ANDROIDAUDIO_Init(SDL_AudioDriverImpl * impl)
impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
impl->OnlyHasDefaultCaptureDevice = SDL_TRUE;
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE; /* this audio target is available. */
}
AudioBootStrap ANDROIDAUDIO_bootstrap = {
@@ -147,7 +140,7 @@ void ANDROIDAUDIO_PauseDevices(void)
/* TODO: Handle multiple devices? */
struct SDL_PrivateAudioData *private;
if (audioDevice != NULL && audioDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) audioDevice->hidden;
private = (struct SDL_PrivateAudioData *)audioDevice->hidden;
if (SDL_AtomicGet(&audioDevice->paused)) {
/* The device is already paused, leave it alone */
private->resume = SDL_FALSE;
@@ -159,7 +152,7 @@ void ANDROIDAUDIO_PauseDevices(void)
}
if (captureDevice != NULL && captureDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) captureDevice->hidden;
private = (struct SDL_PrivateAudioData *)captureDevice->hidden;
if (SDL_AtomicGet(&captureDevice->paused)) {
/* The device is already paused, leave it alone */
private->resume = SDL_FALSE;
@@ -177,7 +170,7 @@ void ANDROIDAUDIO_ResumeDevices(void)
/* TODO: Handle multiple devices? */
struct SDL_PrivateAudioData *private;
if (audioDevice != NULL && audioDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) audioDevice->hidden;
private = (struct SDL_PrivateAudioData *)audioDevice->hidden;
if (private->resume) {
SDL_AtomicSet(&audioDevice->paused, 0);
private->resume = SDL_FALSE;
@@ -186,7 +179,7 @@ void ANDROIDAUDIO_ResumeDevices(void)
}
if (captureDevice != NULL && captureDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) captureDevice->hidden;
private = (struct SDL_PrivateAudioData *)captureDevice->hidden;
if (private->resume) {
SDL_AtomicSet(&captureDevice->paused, 0);
private->resume = SDL_FALSE;
@@ -195,7 +188,7 @@ void ANDROIDAUDIO_ResumeDevices(void)
}
}
#else
#else
void ANDROIDAUDIO_ResumeDevices(void) {}
void ANDROIDAUDIO_PauseDevices(void) {}
@@ -203,4 +196,3 @@ void ANDROIDAUDIO_PauseDevices(void) {}
#endif /* SDL_AUDIO_DRIVER_ANDROID */
/* vi: set ts=4 sw=4 expandtab: */

2
src/audio/android/SDL_androidaudio.h
View File

@@ -26,7 +26,7 @@
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{

2
src/audio/coreaudio/SDL_coreaudio.h
View File

@@ -48,7 +48,7 @@
#endif
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{

269
src/audio/coreaudio/SDL_coreaudio.m
View File

@@ -32,21 +32,20 @@
#define DEBUG_COREAUDIO 0
#if DEBUG_COREAUDIO
#define CHECK_RESULT(msg) \
if (result != noErr) { \
printf("COREAUDIO: Got error %d from '%s'!\n", (int) result, msg); \
SDL_SetError("CoreAudio error (%s): %d", msg, (int) result); \
return 0; \
}
#define CHECK_RESULT(msg) \
if (result != noErr) { \
printf("COREAUDIO: Got error %d from '%s'!\n", (int)result, msg); \
SDL_SetError("CoreAudio error (%s): %d", msg, (int)result); \
return 0; \
}
#else
#define CHECK_RESULT(msg) \
if (result != noErr) { \
SDL_SetError("CoreAudio error (%s): %d", msg, (int) result); \
return 0; \
}
#define CHECK_RESULT(msg) \
if (result != noErr) { \
SDL_SetError("CoreAudio error (%s): %d", msg, (int)result); \
return 0; \
}
#endif
#if MACOSX_COREAUDIO
static const AudioObjectPropertyAddress devlist_address = {
kAudioHardwarePropertyDevices,
@@ -66,10 +65,9 @@ typedef struct AudioDeviceList
static AudioDeviceList *output_devs = NULL;
static AudioDeviceList *capture_devs = NULL;
static SDL_bool
add_to_internal_dev_list(const int iscapture, AudioDeviceID devId)
static SDL_bool add_to_internal_dev_list(const int iscapture, AudioDeviceID devId)
{
AudioDeviceList *item = (AudioDeviceList *) SDL_malloc(sizeof (AudioDeviceList));
AudioDeviceList *item = (AudioDeviceList *)SDL_malloc(sizeof(AudioDeviceList));
if (item == NULL) {
return SDL_FALSE;
}
@@ -85,16 +83,14 @@ add_to_internal_dev_list(const int iscapture, AudioDeviceID devId)
return SDL_TRUE;
}
static void
addToDevList(const char *name, SDL_AudioSpec *spec, const int iscapture, AudioDeviceID devId, void *data)
static void addToDevList(const char *name, SDL_AudioSpec *spec, const int iscapture, AudioDeviceID devId, void *data)
{
if (add_to_internal_dev_list(iscapture, devId)) {
SDL_AddAudioDevice(iscapture, name, spec, (void *) ((size_t) devId));
SDL_AddAudioDevice(iscapture, name, spec, (void *)((size_t)devId));
}
}
static void
build_device_list(int iscapture, addDevFn addfn, void *addfndata)
static void build_device_list(int iscapture, addDevFn addfn, void *addfndata)
{
OSStatus result = noErr;
UInt32 size = 0;
@@ -107,7 +103,7 @@ build_device_list(int iscapture, addDevFn addfn, void *addfndata)
if (result != kAudioHardwareNoError)
return;
devs = (AudioDeviceID *) alloca(size);
devs = (AudioDeviceID *)alloca(size);
if (devs == NULL)
return;
@@ -116,7 +112,7 @@ build_device_list(int iscapture, addDevFn addfn, void *addfndata)
if (result != kAudioHardwareNoError)
return;
max = size / sizeof (AudioDeviceID);
max = size / sizeof(AudioDeviceID);
for (i = 0; i < max; i++) {
CFStringRef cfstr = NULL;
char *ptr = NULL;
@@ -146,7 +142,7 @@ build_device_list(int iscapture, addDevFn addfn, void *addfndata)
if (result != noErr)
continue;
buflist = (AudioBufferList *) SDL_malloc(size);
buflist = (AudioBufferList *)SDL_malloc(size);
if (buflist == NULL)
continue;
@@ -166,13 +162,13 @@ build_device_list(int iscapture, addDevFn addfn, void *addfndata)
if (spec.channels == 0)
continue;
size = sizeof (sampleRate);
size = sizeof(sampleRate);
result = AudioObjectGetPropertyData(dev, &freqaddr, 0, NULL, &size, &sampleRate);
if (result == noErr) {
spec.freq = (int) sampleRate;
spec.freq = (int)sampleRate;
}
size = sizeof (CFStringRef);
size = sizeof(CFStringRef);
result = AudioObjectGetPropertyData(dev, &nameaddr, 0, NULL, &size, &cfstr);
if (result != kAudioHardwareNoError)
continue;
@@ -180,10 +176,9 @@ build_device_list(int iscapture, addDevFn addfn, void *addfndata)
len = CFStringGetMaximumSizeForEncoding(CFStringGetLength(cfstr),
kCFStringEncodingUTF8);
ptr = (char *) SDL_malloc(len + 1);
ptr = (char *)SDL_malloc(len + 1);
usable = ((ptr != NULL) &&
(CFStringGetCString
(cfstr, ptr, len + 1, kCFStringEncodingUTF8)));
(CFStringGetCString(cfstr, ptr, len + 1, kCFStringEncodingUTF8)));
CFRelease(cfstr);
@@ -202,16 +197,15 @@ build_device_list(int iscapture, addDevFn addfn, void *addfndata)
#if DEBUG_COREAUDIO
printf("COREAUDIO: Found %s device #%d: '%s' (devid %d)\n",
((iscapture) ? "capture" : "output"),
(int) i, ptr, (int) dev);
(int)i, ptr, (int)dev);
#endif
addfn(ptr, &spec, iscapture, dev, addfndata);
}
SDL_free(ptr); /* addfn() would have copied the string. */
SDL_free(ptr); /* addfn() would have copied the string. */
}
}
static void
free_audio_device_list(AudioDeviceList **list)
static void free_audio_device_list(AudioDeviceList **list)
{
AudioDeviceList *item = *list;
while (item) {
@@ -222,17 +216,15 @@ free_audio_device_list(AudioDeviceList **list)
*list = NULL;
}
static void
COREAUDIO_DetectDevices(void)
static void COREAUDIO_DetectDevices(void)
{
build_device_list(SDL_TRUE, addToDevList, NULL);
build_device_list(SDL_FALSE, addToDevList, NULL);
}
static void
build_device_change_list(const char *name, SDL_AudioSpec *spec, const int iscapture, AudioDeviceID devId, void *data)
static void build_device_change_list(const char *name, SDL_AudioSpec *spec, const int iscapture, AudioDeviceID devId, void *data)
{
AudioDeviceList **list = (AudioDeviceList **) data;
AudioDeviceList **list = (AudioDeviceList **)data;
AudioDeviceList *item;
for (item = *list; item != NULL; item = item->next) {
if (item->devid == devId) {
@@ -241,12 +233,11 @@ build_device_change_list(const char *name, SDL_AudioSpec *spec, const int iscapt
}
}
add_to_internal_dev_list(iscapture, devId); /* new device, add it. */
SDL_AddAudioDevice(iscapture, name, spec, (void *) ((size_t) devId));
add_to_internal_dev_list(iscapture, devId); /* new device, add it. */
SDL_AddAudioDevice(iscapture, name, spec, (void *)((size_t)devId));
}
static void
reprocess_device_list(const int iscapture, AudioDeviceList **list)
static void reprocess_device_list(const int iscapture, AudioDeviceList **list)
{
AudioDeviceList *item;
AudioDeviceList *prev = NULL;
@@ -263,7 +254,7 @@ reprocess_device_list(const int iscapture, AudioDeviceList **list)
if (item->alive) {
prev = item;
} else {
SDL_RemoveAudioDevice(iscapture, (void *) ((size_t) item->devid));
SDL_RemoveAudioDevice(iscapture, (void *)((size_t)item->devid));
if (prev) {
prev->next = item->next;
} else {
@@ -276,8 +267,7 @@ reprocess_device_list(const int iscapture, AudioDeviceList **list)
}
/* this is called when the system's list of available audio devices changes. */
static OSStatus
device_list_changed(AudioObjectID systemObj, UInt32 num_addr, const AudioObjectPropertyAddress *addrs, void *data)
static OSStatus device_list_changed(AudioObjectID systemObj, UInt32 num_addr, const AudioObjectPropertyAddress *addrs, void *data)
{
reprocess_device_list(SDL_TRUE, &capture_devs);
reprocess_device_list(SDL_FALSE, &output_devs);
@@ -285,7 +275,6 @@ device_list_changed(AudioObjectID systemObj, UInt32 num_addr, const AudioObjectP
}
#endif
static int open_playback_devices;
static int open_capture_devices;
static int num_open_devices;
@@ -337,16 +326,14 @@ static void interruption_begin(_THIS)
static void interruption_end(_THIS)
{
if (this != NULL && this->hidden != NULL && this->hidden->audioQueue != NULL
&& this->hidden->interrupted
&& AudioQueueStart(this->hidden->audioQueue, NULL) == AVAudioSessionErrorCodeNone) {
if (this != NULL && this->hidden != NULL && this->hidden->audioQueue != NULL && this->hidden->interrupted && AudioQueueStart(this->hidden->audioQueue, NULL) == AVAudioSessionErrorCodeNone) {
this->hidden->interrupted = SDL_FALSE;
}
}
@interface SDLInterruptionListener : NSObject
@property (nonatomic, assign) SDL_AudioDevice *device;
@property(nonatomic, assign) SDL_AudioDevice *device;
@end
@@ -354,7 +341,7 @@ static void interruption_end(_THIS)
- (void)audioSessionInterruption:(NSNotification *)note
{
@synchronized (self) {
@synchronized(self) {
NSNumber *type = note.userInfo[AVAudioSessionInterruptionTypeKey];
if (type.unsignedIntegerValue == AVAudioSessionInterruptionTypeBegan) {
interruption_begin(self.device);
@@ -366,7 +353,7 @@ static void interruption_end(_THIS)
- (void)applicationBecameActive:(NSNotification *)note
{
@synchronized (self) {
@synchronized(self) {
interruption_end(self.device);
}
}
@@ -502,9 +489,9 @@ static BOOL update_audio_session(_THIS, SDL_bool open, SDL_bool allow_playandrec
this->hidden->interruption_listener = CFBridgingRetain(listener);
} else {
SDLInterruptionListener *listener = nil;
listener = (SDLInterruptionListener *) CFBridgingRelease(this->hidden->interruption_listener);
listener = (SDLInterruptionListener *)CFBridgingRelease(this->hidden->interruption_listener);
[center removeObserver:listener];
@synchronized (listener) {
@synchronized(listener) {
listener.device = NULL;
}
}
@@ -514,25 +501,23 @@ static BOOL update_audio_session(_THIS, SDL_bool open, SDL_bool allow_playandrec
}
#endif
/* The AudioQueue callback */
static void
outputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer)
static void outputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) inUserData;
SDL_AudioDevice *this = (SDL_AudioDevice *)inUserData;
/* This flag is set before this->mixer_lock is destroyed during
shutdown, so check it before grabbing the mutex, and then check it
again _after_ in case we blocked waiting on the lock. */
if (SDL_AtomicGet(&this->shutdown)) {
return; /* don't do anything, since we don't even want to enqueue this buffer again. */
return; /* don't do anything, since we don't even want to enqueue this buffer again. */
}
SDL_LockMutex(this->mixer_lock);
if (SDL_AtomicGet(&this->shutdown)) {
SDL_UnlockMutex(this->mixer_lock);
return; /* don't do anything, since we don't even want to enqueue this buffer again. */
return; /* don't do anything, since we don't even want to enqueue this buffer again. */
}
if (!SDL_AtomicGet(&this->enabled) || SDL_AtomicGet(&this->paused)) {
@@ -540,7 +525,7 @@ outputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffe
SDL_memset(inBuffer->mAudioData, this->spec.silence, inBuffer->mAudioDataBytesCapacity);
} else if (this->stream) {
UInt32 remaining = inBuffer->mAudioDataBytesCapacity;
Uint8 *ptr = (Uint8 *) inBuffer->mAudioData;
Uint8 *ptr = (Uint8 *)inBuffer->mAudioData;
while (remaining > 0) {
if (SDL_AudioStreamAvailable(this->stream) == 0) {
@@ -566,14 +551,14 @@ outputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffe
}
} else {
UInt32 remaining = inBuffer->mAudioDataBytesCapacity;
Uint8 *ptr = (Uint8 *) inBuffer->mAudioData;
Uint8 *ptr = (Uint8 *)inBuffer->mAudioData;
while (remaining > 0) {
UInt32 len;
if (this->hidden->bufferOffset >= this->hidden->bufferSize) {
/* Generate the data */
(*this->callbackspec.callback)(this->callbackspec.userdata,
this->hidden->buffer, this->hidden->bufferSize);
this->hidden->buffer, this->hidden->bufferSize);
this->hidden->bufferOffset = 0;
}
@@ -581,8 +566,7 @@ outputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffe
if (len > remaining) {
len = remaining;
}
SDL_memcpy(ptr, (char *)this->hidden->buffer +
this->hidden->bufferOffset, len);
SDL_memcpy(ptr, (char *)this->hidden->buffer + this->hidden->bufferOffset, len);
ptr = ptr + len;
remaining -= len;
this->hidden->bufferOffset += len;
@@ -596,20 +580,19 @@ outputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffe
SDL_UnlockMutex(this->mixer_lock);
}
static void
inputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer,
const AudioTimeStamp *inStartTime, UInt32 inNumberPacketDescriptions,
const AudioStreamPacketDescription *inPacketDescs)
static void inputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer,
const AudioTimeStamp *inStartTime, UInt32 inNumberPacketDescriptions,
const AudioStreamPacketDescription *inPacketDescs)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) inUserData;
SDL_AudioDevice *this = (SDL_AudioDevice *)inUserData;
if (SDL_AtomicGet(&this->shutdown)) {
return; /* don't do anything. */
return; /* don't do anything. */
}
/* ignore unless we're active. */
if (!SDL_AtomicGet(&this->paused) && SDL_AtomicGet(&this->enabled)) {
const Uint8 *ptr = (const Uint8 *) inBuffer->mAudioData;
const Uint8 *ptr = (const Uint8 *)inBuffer->mAudioData;
UInt32 remaining = inBuffer->mAudioDataByteSize;
while (remaining > 0) {
UInt32 len = this->hidden->bufferSize - this->hidden->bufferOffset;
@@ -634,35 +617,32 @@ inputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer
AudioQueueEnqueueBuffer(this->hidden->audioQueue, inBuffer, 0, NULL);
}
#if MACOSX_COREAUDIO
static const AudioObjectPropertyAddress alive_address =
{
static const AudioObjectPropertyAddress alive_address = {
kAudioDevicePropertyDeviceIsAlive,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMain
};
static OSStatus
device_unplugged(AudioObjectID devid, UInt32 num_addr, const AudioObjectPropertyAddress *addrs, void *data)
static OSStatus device_unplugged(AudioObjectID devid, UInt32 num_addr, const AudioObjectPropertyAddress *addrs, void *data)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) data;
SDL_AudioDevice *this = (SDL_AudioDevice *)data;
SDL_bool dead = SDL_FALSE;
UInt32 isAlive = 1;
UInt32 size = sizeof (isAlive);
UInt32 size = sizeof(isAlive);
OSStatus error;
if (!SDL_AtomicGet(&this->enabled)) {
return 0; /* already known to be dead. */
return 0; /* already known to be dead. */
}
error = AudioObjectGetPropertyData(this->hidden->deviceID, &alive_address,
0, NULL, &size, &isAlive);
if (error == kAudioHardwareBadDeviceError) {
dead = SDL_TRUE; /* device was unplugged. */
dead = SDL_TRUE; /* device was unplugged. */
} else if ((error == kAudioHardwareNoError) && (!isAlive)) {
dead = SDL_TRUE; /* device died in some other way. */
dead = SDL_TRUE; /* device died in some other way. */
}
if (dead) {
@@ -673,20 +653,18 @@ device_unplugged(AudioObjectID devid, UInt32 num_addr, const AudioObjectProperty
}
/* macOS calls this when the default device changed (if we have a default device open). */
static OSStatus
default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const AudioObjectPropertyAddress *inAddresses, void *inUserData)
static OSStatus default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const AudioObjectPropertyAddress *inAddresses, void *inUserData)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) inUserData;
#if DEBUG_COREAUDIO
SDL_AudioDevice *this = (SDL_AudioDevice *)inUserData;
#if DEBUG_COREAUDIO
printf("COREAUDIO: default device changed for SDL audio device %p!\n", this);
#endif
SDL_AtomicSet(&this->hidden->device_change_flag, 1); /* let the audioqueue thread pick up on this when safe to do so. */
#endif
SDL_AtomicSet(&this->hidden->device_change_flag, 1); /* let the audioqueue thread pick up on this when safe to do so. */
return noErr;
}
#endif
static void
COREAUDIO_CloseDevice(_THIS)
static void COREAUDIO_CloseDevice(_THIS)
{
const SDL_bool iscapture = this->iscapture;
int i;
@@ -694,7 +672,7 @@ COREAUDIO_CloseDevice(_THIS)
/* !!! FIXME: what does iOS do when a bluetooth audio device vanishes? Headphones unplugged? */
/* !!! FIXME: (we only do a "default" device on iOS right now...can we do more?) */
#if MACOSX_COREAUDIO
if (this->handle != NULL) { /* we don't register this listener for default devices. */
if (this->handle != NULL) { /* we don't register this listener for default devices. */
AudioObjectRemovePropertyListener(this->hidden->deviceID, &alive_address, device_unplugged, this);
}
#endif
@@ -708,7 +686,7 @@ COREAUDIO_CloseDevice(_THIS)
}
if (this->hidden->thread) {
SDL_assert(SDL_AtomicGet(&this->shutdown) != 0); /* should have been set by SDL_audio.c */
SDL_assert(SDL_AtomicGet(&this->shutdown) != 0); /* should have been set by SDL_audio.c */
SDL_WaitThread(this->hidden->thread, NULL);
}
@@ -726,7 +704,7 @@ COREAUDIO_CloseDevice(_THIS)
if (open_devices[i] == this) {
--num_open_devices;
if (i < num_open_devices) {
SDL_memmove(&open_devices[i], &open_devices[i+1], sizeof(open_devices[i])*(num_open_devices - i));
SDL_memmove(&open_devices[i], &open_devices[i + 1], sizeof(open_devices[i]) * (num_open_devices - i));
}
break;
}
@@ -748,12 +726,11 @@ COREAUDIO_CloseDevice(_THIS)
}
#if MACOSX_COREAUDIO
static int
prepare_device(_THIS)
static int prepare_device(_THIS)
{
void *handle = this->handle;
SDL_bool iscapture = this->iscapture;
AudioDeviceID devid = (AudioDeviceID) ((size_t) handle);
AudioDeviceID devid = (AudioDeviceID)((size_t)handle);
OSStatus result = noErr;
UInt32 size = 0;
UInt32 alive = 0;
@@ -766,23 +743,20 @@ prepare_device(_THIS)
};
if (handle == NULL) {
size = sizeof (AudioDeviceID);
size = sizeof(AudioDeviceID);
addr.mSelector =
((iscapture) ? kAudioHardwarePropertyDefaultInputDevice :
kAudioHardwarePropertyDefaultOutputDevice);
((iscapture) ? kAudioHardwarePropertyDefaultInputDevice : kAudioHardwarePropertyDefaultOutputDevice);
result = AudioObjectGetPropertyData(kAudioObjectSystemObject, &addr,
0, NULL, &size, &devid);
CHECK_RESULT("AudioHardwareGetProperty (default device)");
}
addr.mSelector = kAudioDevicePropertyDeviceIsAlive;
addr.mScope = iscapture ? kAudioDevicePropertyScopeInput :
kAudioDevicePropertyScopeOutput;
addr.mScope = iscapture ? kAudioDevicePropertyScopeInput : kAudioDevicePropertyScopeOutput;
size = sizeof (alive);
size = sizeof(alive);
result = AudioObjectGetPropertyData(devid, &addr, 0, NULL, &size, &alive);
CHECK_RESULT
("AudioDeviceGetProperty (kAudioDevicePropertyDeviceIsAlive)");
CHECK_RESULT("AudioDeviceGetProperty (kAudioDevicePropertyDeviceIsAlive)");
if (!alive) {
SDL_SetError("CoreAudio: requested device exists, but isn't alive.");
@@ -790,7 +764,7 @@ prepare_device(_THIS)
}
addr.mSelector = kAudioDevicePropertyHogMode;
size = sizeof (pid);
size = sizeof(pid);
result = AudioObjectGetPropertyData(devid, &addr, 0, NULL, &size, &pid);
/* some devices don't support this property, so errors are fine here. */
@@ -803,8 +777,7 @@ prepare_device(_THIS)
return 1;
}
static int
assign_device_to_audioqueue(_THIS)
static int assign_device_to_audioqueue(_THIS)
{
const AudioObjectPropertyAddress prop = {
kAudioDevicePropertyDeviceUID,
@@ -814,7 +787,7 @@ assign_device_to_audioqueue(_THIS)
OSStatus result;
CFStringRef devuid;
UInt32 devuidsize = sizeof (devuid);
UInt32 devuidsize = sizeof(devuid);
result = AudioObjectGetPropertyData(this->hidden->deviceID, &prop, 0, NULL, &devuidsize, &devuid);
CHECK_RESULT("AudioObjectGetPropertyData (kAudioDevicePropertyDeviceUID)");
result = AudioQueueSetProperty(this->hidden->audioQueue, kAudioQueueProperty_CurrentDevice, &devuid, devuidsize);
@@ -824,8 +797,7 @@ assign_device_to_audioqueue(_THIS)
}
#endif
static int
prepare_audioqueue(_THIS)
static int prepare_audioqueue(_THIS)
{
const AudioStreamBasicDescription *strdesc = &this->hidden->strdesc;
const int iscapture = this->iscapture;
@@ -833,7 +805,8 @@ prepare_audioqueue(_THIS)
int i, numAudioBuffers = 2;
AudioChannelLayout layout;
double MINIMUM_AUDIO_BUFFER_TIME_MS;
const double msecs = (this->spec.samples / ((double) this->spec.freq)) * 1000.0;;
const double msecs = (this->spec.samples / ((double)this->spec.freq)) * 1000.0;
;
SDL_assert(CFRunLoopGetCurrent() != NULL);
@@ -845,7 +818,7 @@ prepare_audioqueue(_THIS)
CHECK_RESULT("AudioQueueNewOutput");
}
#if MACOSX_COREAUDIO
#if MACOSX_COREAUDIO
if (!assign_device_to_audioqueue(this)) {
return 0;
}
@@ -859,7 +832,7 @@ prepare_audioqueue(_THIS)
/* If this fails, oh well, we won't notice a device had an extraordinary event take place. */
AudioObjectAddPropertyListener(this->hidden->deviceID, &alive_address, device_unplugged, this);
}
#endif
#endif
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
@@ -916,12 +889,12 @@ prepare_audioqueue(_THIS)
MINIMUM_AUDIO_BUFFER_TIME_MS = 40.0;
}
#endif
if (msecs < MINIMUM_AUDIO_BUFFER_TIME_MS) { /* use more buffers if we have a VERY small sample set. */
if (msecs < MINIMUM_AUDIO_BUFFER_TIME_MS) { /* use more buffers if we have a VERY small sample set. */
numAudioBuffers = ((int)SDL_ceil(MINIMUM_AUDIO_BUFFER_TIME_MS / msecs) * 2);
}
this->hidden->numAudioBuffers = numAudioBuffers;
this->hidden->audioBuffer = SDL_calloc(1, sizeof (AudioQueueBufferRef) * numAudioBuffers);
this->hidden->audioBuffer = SDL_calloc(1, sizeof(AudioQueueBufferRef) * numAudioBuffers);
if (this->hidden->audioBuffer == NULL) {
SDL_OutOfMemory();
return 0;
@@ -948,24 +921,23 @@ prepare_audioqueue(_THIS)
return 1;
}
static int
audioqueue_thread(void *arg)
static int audioqueue_thread(void *arg)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) arg;
SDL_AudioDevice *this = (SDL_AudioDevice *)arg;
int rc;
#if MACOSX_COREAUDIO
#if MACOSX_COREAUDIO
const AudioObjectPropertyAddress default_device_address = {
this->iscapture ? kAudioHardwarePropertyDefaultInputDevice : kAudioHardwarePropertyDefaultOutputDevice,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMain
};
if (this->handle == NULL) { /* opened the default device? Register to know if the user picks a new default. */
if (this->handle == NULL) { /* opened the default device? Register to know if the user picks a new default. */
/* we don't care if this fails; we just won't change to new default devices, but we still otherwise function in this case. */
AudioObjectAddPropertyListener(kAudioObjectSystemObject, &default_device_address, default_device_changed, this);
}
#endif
#endif
rc = prepare_audioqueue(this);
if (!rc) {
@@ -982,14 +954,14 @@ audioqueue_thread(void *arg)
while (!SDL_AtomicGet(&this->shutdown)) {
CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0.10, 1);
#if MACOSX_COREAUDIO
#if MACOSX_COREAUDIO
if ((this->handle == NULL) && SDL_AtomicGet(&this->hidden->device_change_flag)) {
const AudioDeviceID prev_devid = this->hidden->deviceID;
SDL_AtomicSet(&this->hidden->device_change_flag, 0);
#if DEBUG_COREAUDIO
#if DEBUG_COREAUDIO
printf("COREAUDIO: audioqueue_thread is trying to switch to new default device!\n");
#endif
#endif
/* if any of this fails, there's not much to do but wait to see if the user gives up
and quits (flagging the audioqueue for shutdown), or toggles to some other system
@@ -1007,26 +979,25 @@ audioqueue_thread(void *arg)
}
}
}
#endif
#endif
}
if (!this->iscapture) { /* Drain off any pending playback. */
const CFTimeInterval secs = (((this->spec.size / (SDL_AUDIO_BITSIZE(this->spec.format) / 8)) / this->spec.channels) / ((CFTimeInterval) this->spec.freq)) * 2.0;
if (!this->iscapture) { /* Drain off any pending playback. */
const CFTimeInterval secs = (((this->spec.size / (SDL_AUDIO_BITSIZE(this->spec.format) / 8)) / this->spec.channels) / ((CFTimeInterval)this->spec.freq)) * 2.0;
CFRunLoopRunInMode(kCFRunLoopDefaultMode, secs, 0);
}
#if MACOSX_COREAUDIO
#if MACOSX_COREAUDIO
if (this->handle == NULL) {
/* we don't care if this fails; we just won't change to new default devices, but we still otherwise function in this case. */
AudioObjectRemovePropertyListener(kAudioObjectSystemObject, &default_device_address, default_device_changed, this);
}
#endif
#endif
return 0;
}
static int
COREAUDIO_OpenDevice(_THIS, const char *devname)
static int COREAUDIO_OpenDevice(_THIS, const char *devname)
{
AudioStreamBasicDescription *strdesc;
SDL_AudioFormat test_format;
@@ -1062,7 +1033,7 @@ COREAUDIO_OpenDevice(_THIS, const char *devname)
/* Stop CoreAudio from doing expensive audio rate conversion */
@autoreleasepool {
AVAudioSession* session = [AVAudioSession sharedInstance];
AVAudioSession *session = [AVAudioSession sharedInstance];
[session setPreferredSampleRate:this->spec.freq error:nil];
this->spec.freq = (int)session.sampleRate;
#if TARGET_OS_TV
@@ -1106,7 +1077,7 @@ COREAUDIO_OpenDevice(_THIS, const char *devname)
break;
}
if (!test_format) { /* shouldn't happen, but just in case... */
if (!test_format) { /* shouldn't happen, but just in case... */
return SDL_SetError("%s: Unsupported audio format", "coreaudio");
}
this->spec.format = test_format;
@@ -1131,7 +1102,7 @@ COREAUDIO_OpenDevice(_THIS, const char *devname)
/* This has to init in a new thread so it can get its own CFRunLoop. :/ */
this->hidden->ready_semaphore = SDL_CreateSemaphore(0);
if (!this->hidden->ready_semaphore) {
return -1; /* oh well. */
return -1; /* oh well. */
}
this->hidden->thread = SDL_CreateThreadInternal(audioqueue_thread, "AudioQueue thread", 512 * 1024, this);
@@ -1151,10 +1122,9 @@ COREAUDIO_OpenDevice(_THIS, const char *devname)
}
#if !MACOSX_COREAUDIO
static int
COREAUDIO_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
static int COREAUDIO_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
{
AVAudioSession* session = [AVAudioSession sharedInstance];
AVAudioSession *session = [AVAudioSession sharedInstance];
if (name != NULL) {
*name = NULL;
@@ -1164,9 +1134,8 @@ COREAUDIO_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
spec->channels = [session outputNumberOfChannels];
return 0;
}
#else /* MACOSX_COREAUDIO */
static int
COREAUDIO_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
#else /* MACOSX_COREAUDIO */
static int COREAUDIO_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
{
AudioDeviceID devid;
AudioBufferList *buflist;
@@ -1205,7 +1174,7 @@ COREAUDIO_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
/* Get the Device ID */
cfstr = NULL;
size = sizeof (AudioDeviceID);
size = sizeof(AudioDeviceID);
result = AudioObjectGetPropertyData(kAudioObjectSystemObject, &addr,
0, NULL, &size, &devid);
@@ -1215,7 +1184,7 @@ COREAUDIO_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
if (name != NULL) {
/* Use the Device ID to get the name */
size = sizeof (CFStringRef);
size = sizeof(CFStringRef);
result = AudioObjectGetPropertyData(devid, &nameaddr, 0, NULL, &size, &cfstr);
if (result != noErr) {
@@ -1223,8 +1192,8 @@ COREAUDIO_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
}
len = CFStringGetMaximumSizeForEncoding(CFStringGetLength(cfstr),
kCFStringEncodingUTF8);
devname = (char *) SDL_malloc(len + 1);
kCFStringEncodingUTF8);
devname = (char *)SDL_malloc(len + 1);
usable = ((devname != NULL) &&
(CFStringGetCString(cfstr, devname, len + 1, kCFStringEncodingUTF8)));
CFRelease(cfstr);
@@ -1256,13 +1225,13 @@ COREAUDIO_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
return SDL_SetError("%s: Default Device Sample Rate not found", "coreaudio");
}
spec->freq = (int) sampleRate;
spec->freq = (int)sampleRate;
result = AudioObjectGetPropertyDataSize(devid, &bufaddr, 0, NULL, &size);
if (result != noErr)
return SDL_SetError("%s: Default Device Data Size not found", "coreaudio");
buflist = (AudioBufferList *) SDL_malloc(size);
buflist = (AudioBufferList *)SDL_malloc(size);
if (buflist == NULL)
return SDL_SetError("%s: Default Device Buffer List not found", "coreaudio");
@@ -1286,8 +1255,7 @@ COREAUDIO_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
}
#endif /* MACOSX_COREAUDIO */
static void
COREAUDIO_Deinitialize(void)
static void COREAUDIO_Deinitialize(void)
{
#if MACOSX_COREAUDIO
AudioObjectRemovePropertyListener(kAudioObjectSystemObject, &devlist_address, device_list_changed, NULL);
@@ -1296,8 +1264,7 @@ COREAUDIO_Deinitialize(void)
#endif
}
static SDL_bool
COREAUDIO_Init(SDL_AudioDriverImpl * impl)
static SDL_bool COREAUDIO_Init(SDL_AudioDriverImpl *impl)
{
/* Set the function pointers */
impl->OpenDevice = COREAUDIO_OpenDevice;
@@ -1317,7 +1284,7 @@ COREAUDIO_Init(SDL_AudioDriverImpl * impl)
impl->HasCaptureSupport = SDL_TRUE;
impl->SupportsNonPow2Samples = SDL_TRUE;
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE; /* this audio target is available. */
}
AudioBootStrap COREAUDIO_bootstrap = {

136
src/audio/directsound/SDL_directsound.c
View File

@@ -41,21 +41,20 @@ static SDL_bool SupportsIMMDevice = SDL_FALSE;
#endif /* HAVE_MMDEVICEAPI_H */
/* DirectX function pointers for audio */
static void* DSoundDLL = NULL;
typedef HRESULT (WINAPI *fnDirectSoundCreate8)(LPGUID,LPDIRECTSOUND*,LPUNKNOWN);
typedef HRESULT (WINAPI *fnDirectSoundEnumerateW)(LPDSENUMCALLBACKW, LPVOID);
typedef HRESULT (WINAPI *fnDirectSoundCaptureCreate8)(LPCGUID,LPDIRECTSOUNDCAPTURE8 *,LPUNKNOWN);
typedef HRESULT (WINAPI *fnDirectSoundCaptureEnumerateW)(LPDSENUMCALLBACKW,LPVOID);
static void *DSoundDLL = NULL;
typedef HRESULT(WINAPI *fnDirectSoundCreate8)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
typedef HRESULT(WINAPI *fnDirectSoundEnumerateW)(LPDSENUMCALLBACKW, LPVOID);
typedef HRESULT(WINAPI *fnDirectSoundCaptureCreate8)(LPCGUID, LPDIRECTSOUNDCAPTURE8 *, LPUNKNOWN);
typedef HRESULT(WINAPI *fnDirectSoundCaptureEnumerateW)(LPDSENUMCALLBACKW, LPVOID);
static fnDirectSoundCreate8 pDirectSoundCreate8 = NULL;
static fnDirectSoundEnumerateW pDirectSoundEnumerateW = NULL;
static fnDirectSoundCaptureCreate8 pDirectSoundCaptureCreate8 = NULL;
static fnDirectSoundCaptureEnumerateW pDirectSoundCaptureEnumerateW = NULL;
static const GUID SDL_KSDATAFORMAT_SUBTYPE_PCM = { 0x00000001, 0x0000, 0x0010,{ 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
static const GUID SDL_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = { 0x00000003, 0x0000, 0x0010,{ 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
static const GUID SDL_KSDATAFORMAT_SUBTYPE_PCM = { 0x00000001, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
static const GUID SDL_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = { 0x00000003, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
static void
DSOUND_Unload(void)
static void DSOUND_Unload(void)
{
pDirectSoundCreate8 = NULL;
pDirectSoundEnumerateW = NULL;
@@ -68,9 +67,7 @@ DSOUND_Unload(void)
}
}
static int
DSOUND_Load(void)
static int DSOUND_Load(void)
{
int loaded = 0;
@@ -80,17 +77,19 @@ DSOUND_Load(void)
if (DSoundDLL == NULL) {
SDL_SetError("DirectSound: failed to load DSOUND.DLL");
} else {
/* Now make sure we have DirectX 8 or better... */
#define DSOUNDLOAD(f) { \
p##f = (fn##f) SDL_LoadFunction(DSoundDLL, #f); \
if (!p##f) loaded = 0; \
}
loaded = 1; /* will reset if necessary. */
/* Now make sure we have DirectX 8 or better... */
#define DSOUNDLOAD(f) \
{ \
p##f = (fn##f)SDL_LoadFunction(DSoundDLL, #f); \
if (!p##f) \
loaded = 0; \
}
loaded = 1; /* will reset if necessary. */
DSOUNDLOAD(DirectSoundCreate8);
DSOUNDLOAD(DirectSoundEnumerateW);
DSOUNDLOAD(DirectSoundCaptureCreate8);
DSOUNDLOAD(DirectSoundCaptureEnumerateW);
#undef DSOUNDLOAD
#undef DSOUNDLOAD
if (!loaded) {
SDL_SetError("DirectSound: System doesn't appear to have DX8.");
@@ -104,8 +103,7 @@ DSOUND_Load(void)
return loaded;
}
static int
SetDSerror(const char *function, int code)
static int SetDSerror(const char *function, int code)
{
const char *error;
@@ -151,14 +149,12 @@ SetDSerror(const char *function, int code)
return SDL_SetError("%s: %s (0x%x)", function, error, code);
}
static void
DSOUND_FreeDeviceHandle(void *handle)
static void DSOUND_FreeDeviceHandle(void *handle)
{
SDL_free(handle);
}
static int
DSOUND_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
static int DSOUND_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
{
#if HAVE_MMDEVICEAPI_H
if (SupportsIMMDevice) {
@@ -168,29 +164,27 @@ DSOUND_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
return SDL_Unsupported();
}
static BOOL CALLBACK
FindAllDevs(LPGUID guid, LPCWSTR desc, LPCWSTR module, LPVOID data)
static BOOL CALLBACK FindAllDevs(LPGUID guid, LPCWSTR desc, LPCWSTR module, LPVOID data)
{
const int iscapture = (int) ((size_t) data);
if (guid != NULL) { /* skip default device */
const int iscapture = (int)((size_t)data);
if (guid != NULL) { /* skip default device */
char *str = WIN_LookupAudioDeviceName(desc, guid);
if (str != NULL) {
LPGUID cpyguid = (LPGUID) SDL_malloc(sizeof (GUID));
SDL_memcpy(cpyguid, guid, sizeof (GUID));
LPGUID cpyguid = (LPGUID)SDL_malloc(sizeof(GUID));
SDL_memcpy(cpyguid, guid, sizeof(GUID));
/* Note that spec is NULL, because we are required to connect to the
* device before getting the channel mask and output format, making
* this information inaccessible at enumeration time
*/
SDL_AddAudioDevice(iscapture, str, NULL, cpyguid);
SDL_free(str); /* addfn() makes a copy of this string. */
SDL_free(str); /* addfn() makes a copy of this string. */
}
}
return TRUE; /* keep enumerating. */
return TRUE; /* keep enumerating. */
}
static void
DSOUND_DetectDevices(void)
static void DSOUND_DetectDevices(void)
{
#if HAVE_MMDEVICEAPI_H
if (SupportsIMMDevice) {
@@ -204,9 +198,7 @@ DSOUND_DetectDevices(void)
#endif /* HAVE_MMDEVICEAPI_H*/
}
static void
DSOUND_WaitDevice(_THIS)
static void DSOUND_WaitDevice(_THIS)
{
DWORD status = 0;
DWORD cursor = 0;
@@ -263,8 +255,7 @@ DSOUND_WaitDevice(_THIS)
}
}
static void
DSOUND_PlayDevice(_THIS)
static void DSOUND_PlayDevice(_THIS)
{
/* Unlock the buffer, allowing it to play */
if (this->hidden->locked_buf) {
@@ -274,8 +265,7 @@ DSOUND_PlayDevice(_THIS)
}
}
static Uint8 *
DSOUND_GetDeviceBuf(_THIS)
static Uint8 *DSOUND_GetDeviceBuf(_THIS)
{
DWORD cursor = 0;
DWORD junk = 0;
@@ -316,14 +306,13 @@ DSOUND_GetDeviceBuf(_THIS)
/* Lock the audio buffer */
result = IDirectSoundBuffer_Lock(this->hidden->mixbuf, cursor,
this->spec.size,
(LPVOID *) & this->hidden->locked_buf,
(LPVOID *)&this->hidden->locked_buf,
&rawlen, NULL, &junk, 0);
if (result == DSERR_BUFFERLOST) {
IDirectSoundBuffer_Restore(this->hidden->mixbuf);
result = IDirectSoundBuffer_Lock(this->hidden->mixbuf, cursor,
this->spec.size,
(LPVOID *) & this->
hidden->locked_buf, &rawlen, NULL,
(LPVOID *)&this->hidden->locked_buf, &rawlen, NULL,
&junk, 0);
}
if (result != DS_OK) {
@@ -333,8 +322,7 @@ DSOUND_GetDeviceBuf(_THIS)
return this->hidden->locked_buf;
}
static int
DSOUND_CaptureFromDevice(_THIS, void *buffer, int buflen)
static int DSOUND_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
struct SDL_PrivateAudioData *h = this->hidden;
DWORD junk, cursor, ptr1len, ptr2len;
@@ -343,7 +331,7 @@ DSOUND_CaptureFromDevice(_THIS, void *buffer, int buflen)
SDL_assert(buflen == this->spec.size);
while (SDL_TRUE) {
if (SDL_AtomicGet(&this->shutdown)) { /* in case the buffer froze... */
if (SDL_AtomicGet(&this->shutdown)) { /* in case the buffer froze... */
SDL_memset(buffer, this->spec.silence, buflen);
return buflen;
}
@@ -352,7 +340,7 @@ DSOUND_CaptureFromDevice(_THIS, void *buffer, int buflen)
return -1;
}
if ((cursor / this->spec.size) == h->lastchunk) {
SDL_Delay(1); /* FIXME: find out how much time is left and sleep that long */
SDL_Delay(1); /* FIXME: find out how much time is left and sleep that long */
} else {
break;
}
@@ -377,8 +365,7 @@ DSOUND_CaptureFromDevice(_THIS, void *buffer, int buflen)
return ptr1len;
}
static void
DSOUND_FlushCapture(_THIS)
static void DSOUND_FlushCapture(_THIS)
{
struct SDL_PrivateAudioData *h = this->hidden;
DWORD junk, cursor;
@@ -387,8 +374,7 @@ DSOUND_FlushCapture(_THIS)
}
}
static void
DSOUND_CloseDevice(_THIS)
static void DSOUND_CloseDevice(_THIS)
{
if (this->hidden->mixbuf != NULL) {
IDirectSoundBuffer_Stop(this->hidden->mixbuf);
@@ -411,8 +397,7 @@ DSOUND_CloseDevice(_THIS)
number of audio chunks available in the created buffer. This is for
playback devices, not capture.
*/
static int
CreateSecondary(_THIS, const DWORD bufsize, WAVEFORMATEX *wfmt)
static int CreateSecondary(_THIS, const DWORD bufsize, WAVEFORMATEX *wfmt)
{
LPDIRECTSOUND sndObj = this->hidden->sound;
LPDIRECTSOUNDBUFFER *sndbuf = &this->hidden->mixbuf;
@@ -436,14 +421,14 @@ CreateSecondary(_THIS, const DWORD bufsize, WAVEFORMATEX *wfmt)
/* Silence the initial audio buffer */
result = IDirectSoundBuffer_Lock(*sndbuf, 0, format.dwBufferBytes,
(LPVOID *) & pvAudioPtr1, &dwAudioBytes1,
(LPVOID *) & pvAudioPtr2, &dwAudioBytes2,
(LPVOID *)&pvAudioPtr1, &dwAudioBytes1,
(LPVOID *)&pvAudioPtr2, &dwAudioBytes2,
DSBLOCK_ENTIREBUFFER);
if (result == DS_OK) {
SDL_memset(pvAudioPtr1, this->spec.silence, dwAudioBytes1);
IDirectSoundBuffer_Unlock(*sndbuf,
(LPVOID) pvAudioPtr1, dwAudioBytes1,
(LPVOID) pvAudioPtr2, dwAudioBytes2);
(LPVOID)pvAudioPtr1, dwAudioBytes1,
(LPVOID)pvAudioPtr2, dwAudioBytes2);
}
/* We're ready to go */
@@ -454,8 +439,7 @@ CreateSecondary(_THIS, const DWORD bufsize, WAVEFORMATEX *wfmt)
number of audio chunks available in the created buffer. This is for
capture devices, not playback.
*/
static int
CreateCaptureBuffer(_THIS, const DWORD bufsize, WAVEFORMATEX *wfmt)
static int CreateCaptureBuffer(_THIS, const DWORD bufsize, WAVEFORMATEX *wfmt)
{
LPDIRECTSOUNDCAPTURE capture = this->hidden->capture;
LPDIRECTSOUNDCAPTUREBUFFER *capturebuf = &this->hidden->capturebuf;
@@ -463,7 +447,7 @@ CreateCaptureBuffer(_THIS, const DWORD bufsize, WAVEFORMATEX *wfmt)
HRESULT result;
SDL_zero(format);
format.dwSize = sizeof (format);
format.dwSize = sizeof(format);
format.dwFlags = DSCBCAPS_WAVEMAPPED;
format.dwBufferBytes = bufsize;
format.lpwfxFormat = wfmt;
@@ -494,15 +478,14 @@ CreateCaptureBuffer(_THIS, const DWORD bufsize, WAVEFORMATEX *wfmt)
return 0;
}
static int
DSOUND_OpenDevice(_THIS, const char *devname)
static int DSOUND_OpenDevice(_THIS, const char *devname)
{
const DWORD numchunks = 8;
HRESULT result;
SDL_bool tried_format = SDL_FALSE;
SDL_bool iscapture = this->iscapture;
SDL_AudioFormat test_format;
LPGUID guid = (LPGUID) this->handle;
LPGUID guid = (LPGUID)this->handle;
DWORD bufsize;
/* Initialize all variables that we clean on shutdown */
@@ -548,8 +531,8 @@ DSOUND_OpenDevice(_THIS, const char *devname)
bufsize = numchunks * this->spec.size;
if ((bufsize < DSBSIZE_MIN) || (bufsize > DSBSIZE_MAX)) {
SDL_SetError("Sound buffer size must be between %d and %d",
(int) ((DSBSIZE_MIN < numchunks) ? 1 : DSBSIZE_MIN / numchunks),
(int) (DSBSIZE_MAX / numchunks));
(int)((DSBSIZE_MIN < numchunks) ? 1 : DSBSIZE_MIN / numchunks),
(int)(DSBSIZE_MAX / numchunks));
} else {
int rc;
WAVEFORMATEXTENSIBLE wfmt;
@@ -565,8 +548,7 @@ DSOUND_OpenDevice(_THIS, const char *devname)
}
wfmt.Samples.wValidBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
switch (this->spec.channels)
{
switch (this->spec.channels) {
case 3: /* 3.0 (or 2.1) */
wfmt.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER;
break;
@@ -601,7 +583,7 @@ DSOUND_OpenDevice(_THIS, const char *devname)
wfmt.Format.nBlockAlign = wfmt.Format.nChannels * (wfmt.Format.wBitsPerSample / 8);
wfmt.Format.nAvgBytesPerSec = wfmt.Format.nSamplesPerSec * wfmt.Format.nBlockAlign;
rc = iscapture ? CreateCaptureBuffer(this, bufsize, (WAVEFORMATEX*) &wfmt) : CreateSecondary(this, bufsize, (WAVEFORMATEX*) &wfmt);
rc = iscapture ? CreateCaptureBuffer(this, bufsize, (WAVEFORMATEX *)&wfmt) : CreateSecondary(this, bufsize, (WAVEFORMATEX *)&wfmt);
if (rc == 0) {
this->hidden->num_buffers = numchunks;
break;
@@ -616,19 +598,17 @@ DSOUND_OpenDevice(_THIS, const char *devname)
if (!test_format) {
if (tried_format) {
return -1; /* CreateSecondary() should have called SDL_SetError(). */
return -1; /* CreateSecondary() should have called SDL_SetError(). */
}
return SDL_SetError("%s: Unsupported audio format", "directsound");
}
/* Playback buffers will auto-start playing in DSOUND_WaitDevice() */
return 0; /* good to go. */
return 0; /* good to go. */
}
static void
DSOUND_Deinitialize(void)
static void DSOUND_Deinitialize(void)
{
#if HAVE_MMDEVICEAPI_H
if (SupportsIMMDevice) {
@@ -639,9 +619,7 @@ DSOUND_Deinitialize(void)
DSOUND_Unload();
}
static SDL_bool
DSOUND_Init(SDL_AudioDriverImpl * impl)
static SDL_bool DSOUND_Init(SDL_AudioDriverImpl *impl)
{
if (!DSOUND_Load()) {
return SDL_FALSE;
@@ -667,7 +645,7 @@ DSOUND_Init(SDL_AudioDriverImpl * impl)
impl->HasCaptureSupport = SDL_TRUE;
impl->SupportsNonPow2Samples = SDL_TRUE;
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE; /* this audio target is available. */
}
AudioBootStrap DSOUND_bootstrap = {

2
src/audio/directsound/SDL_directsound.h
View File

@@ -28,7 +28,7 @@
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define _THIS SDL_AudioDevice *this
/* The DirectSound objects */
struct SDL_PrivateAudioData

62
src/audio/disk/SDL_diskaudio.c
View File

@@ -33,21 +33,19 @@
/* !!! FIXME: these should be SDL hints, not environment variables. */
/* environment variables and defaults. */
#define DISKENVR_OUTFILE "SDL_DISKAUDIOFILE"
#define DISKDEFAULT_OUTFILE "sdlaudio.raw"
#define DISKENVR_INFILE "SDL_DISKAUDIOFILEIN"
#define DISKDEFAULT_INFILE "sdlaudio-in.raw"
#define DISKENVR_IODELAY "SDL_DISKAUDIODELAY"
#define DISKENVR_OUTFILE "SDL_DISKAUDIOFILE"
#define DISKDEFAULT_OUTFILE "sdlaudio.raw"
#define DISKENVR_INFILE "SDL_DISKAUDIOFILEIN"
#define DISKDEFAULT_INFILE "sdlaudio-in.raw"
#define DISKENVR_IODELAY "SDL_DISKAUDIODELAY"
/* This function waits until it is possible to write a full sound buffer */
static void
DISKAUDIO_WaitDevice(_THIS)
static void DISKAUDIO_WaitDevice(_THIS)
{
SDL_Delay(_this->hidden->io_delay);
}
static void
DISKAUDIO_PlayDevice(_THIS)
static void DISKAUDIO_PlayDevice(_THIS)
{
const size_t written = SDL_RWwrite(_this->hidden->io,
_this->hidden->mixbuf,
@@ -62,14 +60,12 @@ DISKAUDIO_PlayDevice(_THIS)
#endif
}
static Uint8 *
DISKAUDIO_GetDeviceBuf(_THIS)
static Uint8 *DISKAUDIO_GetDeviceBuf(_THIS)
{
return _this->hidden->mixbuf;
}
static int
DISKAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
static int DISKAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
struct SDL_PrivateAudioData *h = _this->hidden;
const int origbuflen = buflen;
@@ -78,9 +74,9 @@ DISKAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
if (h->io) {
const size_t br = SDL_RWread(h->io, buffer, 1, buflen);
buflen -= (int) br;
buffer = ((Uint8 *) buffer) + br;
if (buflen > 0) { /* EOF (or error, but whatever). */
buflen -= (int)br;
buffer = ((Uint8 *)buffer) + br;
if (buflen > 0) { /* EOF (or error, but whatever). */
SDL_RWclose(h->io);
h->io = NULL;
}
@@ -92,15 +88,12 @@ DISKAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
return origbuflen;
}
static void
DISKAUDIO_FlushCapture(_THIS)
static void DISKAUDIO_FlushCapture(_THIS)
{
/* no op...we don't advance the file pointer or anything. */
}
static void
DISKAUDIO_CloseDevice(_THIS)
static void DISKAUDIO_CloseDevice(_THIS)
{
if (_this->hidden->io != NULL) {
SDL_RWclose(_this->hidden->io);
@@ -109,9 +102,7 @@ DISKAUDIO_CloseDevice(_THIS)
SDL_free(_this->hidden);
}
static const char *
get_filename(const SDL_bool iscapture, const char *devname)
static const char *get_filename(const SDL_bool iscapture, const char *devname)
{
if (devname == NULL) {
devname = SDL_getenv(iscapture ? DISKENVR_INFILE : DISKENVR_OUTFILE);
@@ -122,8 +113,7 @@ get_filename(const SDL_bool iscapture, const char *devname)
return devname;
}
static int
DISKAUDIO_OpenDevice(_THIS, const char *devname)
static int DISKAUDIO_OpenDevice(_THIS, const char *devname)
{
void *handle = _this->handle;
/* handle != NULL means "user specified the placeholder name on the fake detected device list" */
@@ -152,7 +142,7 @@ DISKAUDIO_OpenDevice(_THIS, const char *devname)
/* Allocate mixing buffer */
if (!iscapture) {
_this->hidden->mixbuf = (Uint8 *) SDL_malloc(_this->spec.size);
_this->hidden->mixbuf = (Uint8 *)SDL_malloc(_this->spec.size);
if (_this->hidden->mixbuf == NULL) {
return SDL_OutOfMemory();
}
@@ -160,24 +150,22 @@ DISKAUDIO_OpenDevice(_THIS, const char *devname)
}
SDL_LogCritical(SDL_LOG_CATEGORY_AUDIO,
"You are using the SDL disk i/o audio driver!\n");
"You are using the SDL disk i/o audio driver!\n");
SDL_LogCritical(SDL_LOG_CATEGORY_AUDIO,
" %s file [%s].\n", iscapture ? "Reading from" : "Writing to",
fname);
" %s file [%s].\n", iscapture ? "Reading from" : "Writing to",
fname);
/* We're ready to rock and roll. :-) */
return 0;
}
static void
DISKAUDIO_DetectDevices(void)
static void DISKAUDIO_DetectDevices(void)
{
SDL_AddAudioDevice(SDL_FALSE, DEFAULT_OUTPUT_DEVNAME, NULL, (void *) 0x1);
SDL_AddAudioDevice(SDL_TRUE, DEFAULT_INPUT_DEVNAME, NULL, (void *) 0x2);
SDL_AddAudioDevice(SDL_FALSE, DEFAULT_OUTPUT_DEVNAME, NULL, (void *)0x1);
SDL_AddAudioDevice(SDL_TRUE, DEFAULT_INPUT_DEVNAME, NULL, (void *)0x2);
}
static SDL_bool
DISKAUDIO_Init(SDL_AudioDriverImpl * impl)
static SDL_bool DISKAUDIO_Init(SDL_AudioDriverImpl *impl)
{
/* Set the function pointers */
impl->OpenDevice = DISKAUDIO_OpenDevice;
@@ -194,7 +182,7 @@ DISKAUDIO_Init(SDL_AudioDriverImpl * impl)
impl->HasCaptureSupport = SDL_TRUE;
impl->SupportsNonPow2Samples = SDL_TRUE;
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE; /* this audio target is available. */
}
AudioBootStrap DISKAUDIO_bootstrap = {

2
src/audio/disk/SDL_diskaudio.h
View File

@@ -26,7 +26,7 @@
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *_this
#define _THIS SDL_AudioDevice *_this
struct SDL_PrivateAudioData
{

51
src/audio/dsp/SDL_dspaudio.c
View File

@@ -24,8 +24,8 @@
/* Allow access to a raw mixing buffer */
#include <stdio.h> /* For perror() */
#include <string.h> /* For strerror() */
#include <stdio.h> /* For perror() */
#include <string.h> /* For strerror() */
#include <errno.h>
#include <unistd.h>
#include <fcntl.h>
@@ -40,16 +40,12 @@
#include "../SDL_audiodev_c.h"
#include "SDL_dspaudio.h"
static void
DSP_DetectDevices(void)
static void DSP_DetectDevices(void)
{
SDL_EnumUnixAudioDevices(0, NULL);
}
static void
DSP_CloseDevice(_THIS)
static void DSP_CloseDevice(_THIS)
{
if (this->hidden->audio_fd >= 0) {
close(this->hidden->audio_fd);
@@ -58,9 +54,7 @@ DSP_CloseDevice(_THIS)
SDL_free(this->hidden);
}
static int
DSP_OpenDevice(_THIS, const char *devname)
static int DSP_OpenDevice(_THIS, const char *devname)
{
SDL_bool iscapture = this->iscapture;
const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
@@ -203,11 +197,12 @@ DSP_OpenDevice(_THIS, const char *devname)
SDL_CalculateAudioSpec(&this->spec);
/* Determine the power of two of the fragment size */
for (frag_spec = 0; (0x01U << frag_spec) < this->spec.size; ++frag_spec);
for (frag_spec = 0; (0x01U << frag_spec) < this->spec.size; ++frag_spec)
;
if ((0x01U << frag_spec) != this->spec.size) {
return SDL_SetError("Fragment size must be a power of two");
}
frag_spec |= 0x00020000; /* two fragments, for low latency */
frag_spec |= 0x00020000; /* two fragments, for low latency */
/* Set the audio buffering parameters */
#ifdef DEBUG_AUDIO
@@ -231,7 +226,7 @@ DSP_OpenDevice(_THIS, const char *devname)
/* Allocate mixing buffer */
if (!iscapture) {
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
this->hidden->mixbuf = (Uint8 *)SDL_malloc(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
return SDL_OutOfMemory();
}
@@ -242,9 +237,7 @@ DSP_OpenDevice(_THIS, const char *devname)
return 0;
}
static void
DSP_PlayDevice(_THIS)
static void DSP_PlayDevice(_THIS)
{
struct SDL_PrivateAudioData *h = this->hidden;
if (write(h->audio_fd, h->mixbuf, h->mixlen) == -1) {
@@ -256,27 +249,24 @@ DSP_PlayDevice(_THIS)
#endif
}
static Uint8 *
DSP_GetDeviceBuf(_THIS)
static Uint8 *DSP_GetDeviceBuf(_THIS)
{
return this->hidden->mixbuf;
}
static int
DSP_CaptureFromDevice(_THIS, void *buffer, int buflen)
static int DSP_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
return (int)read(this->hidden->audio_fd, buffer, buflen);
}
static void
DSP_FlushCapture(_THIS)
static void DSP_FlushCapture(_THIS)
{
struct SDL_PrivateAudioData *h = this->hidden;
audio_buf_info info;
if (ioctl(h->audio_fd, SNDCTL_DSP_GETISPACE, &info) == 0) {
while (info.bytes > 0) {
char buf[512];
const size_t len = SDL_min(sizeof (buf), info.bytes);
const size_t len = SDL_min(sizeof(buf), info.bytes);
const ssize_t br = read(h->audio_fd, buf, len);
if (br <= 0) {
break;
@@ -287,22 +277,20 @@ DSP_FlushCapture(_THIS)
}
static SDL_bool InitTimeDevicesExist = SDL_FALSE;
static int
look_for_devices_test(int fd)
static int look_for_devices_test(int fd)
{
InitTimeDevicesExist = SDL_TRUE; /* note that _something_ exists. */
InitTimeDevicesExist = SDL_TRUE; /* note that _something_ exists. */
/* Don't add to the device list, we're just seeing if any devices exist. */
return 0;
}
static SDL_bool
DSP_Init(SDL_AudioDriverImpl * impl)
static SDL_bool DSP_Init(SDL_AudioDriverImpl *impl)
{
InitTimeDevicesExist = SDL_FALSE;
SDL_EnumUnixAudioDevices(0, look_for_devices_test);
if (!InitTimeDevicesExist) {
SDL_SetError("dsp: No such audio device");
return SDL_FALSE; /* maybe try a different backend. */
return SDL_FALSE; /* maybe try a different backend. */
}
/* Set the function pointers */
@@ -317,10 +305,9 @@ DSP_Init(SDL_AudioDriverImpl * impl)
impl->AllowsArbitraryDeviceNames = SDL_TRUE;
impl->HasCaptureSupport = SDL_TRUE;
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE; /* this audio target is available. */
}
AudioBootStrap DSP_bootstrap = {
"dsp", "OSS /dev/dsp standard audio", DSP_Init, SDL_FALSE
};

4
src/audio/dsp/SDL_dspaudio.h
View File

@@ -26,7 +26,7 @@
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
@@ -37,7 +37,7 @@ struct SDL_PrivateAudioData
Uint8 *mixbuf;
int mixlen;
};
#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
#endif /* SDL_dspaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */

16
src/audio/dummy/SDL_dummyaudio.c
View File

@@ -25,15 +25,14 @@
#include "../SDL_audio_c.h"
#include "SDL_dummyaudio.h"
static int
DUMMYAUDIO_OpenDevice(_THIS, const char *devname)
static int DUMMYAUDIO_OpenDevice(_THIS, const char *devname)
{
_this->hidden = (void *) 0x1; /* just something non-NULL */
return 0; /* always succeeds. */
_this->hidden = (void *)0x1; /* just something non-NULL */
return 0; /* always succeeds. */
}
static int
DUMMYAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
static int DUMMYAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
/* Delay to make this sort of simulate real audio input. */
SDL_Delay((_this->spec.samples * 1000) / _this->spec.freq);
@@ -43,8 +42,7 @@ DUMMYAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
return buflen;
}
static SDL_bool
DUMMYAUDIO_Init(SDL_AudioDriverImpl * impl)
static SDL_bool DUMMYAUDIO_Init(SDL_AudioDriverImpl *impl)
{
/* Set the function pointers */
impl->OpenDevice = DUMMYAUDIO_OpenDevice;
@@ -54,7 +52,7 @@ DUMMYAUDIO_Init(SDL_AudioDriverImpl * impl)
impl->OnlyHasDefaultCaptureDevice = SDL_TRUE;
impl->HasCaptureSupport = SDL_TRUE;
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE; /* this audio target is available. */
}
AudioBootStrap DUMMYAUDIO_bootstrap = {

2
src/audio/dummy/SDL_dummyaudio.h
View File

@@ -26,7 +26,7 @@
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *_this
#define _THIS SDL_AudioDevice *_this
struct SDL_PrivateAudioData
{

57
src/audio/emscripten/SDL_emscriptenaudio.c
View File

@@ -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;

2
src/audio/emscripten/SDL_emscriptenaudio.h
View File

@@ -26,7 +26,7 @@
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{

21
src/audio/haiku/SDL_haikuaudio.cc
View File

@@ -41,8 +41,7 @@ extern "C"
/* !!! FIXME: have the callback call the higher level to avoid code dupe. */
/* The Haiku callback for handling the audio buffer */
static void
FillSound(void *device, void *stream, size_t len,
static void FillSound(void *device, void *stream, size_t len,
const media_raw_audio_format & format)
{
SDL_AudioDevice *audio = (SDL_AudioDevice *) device;
@@ -84,8 +83,7 @@ FillSound(void *device, void *stream, size_t len,
SDL_UnlockMutex(audio->mixer_lock);
}
static void
HAIKUAUDIO_CloseDevice(_THIS)
static void HAIKUAUDIO_CloseDevice(_THIS)
{
if (_this->hidden->audio_obj) {
_this->hidden->audio_obj->Stop();
@@ -99,8 +97,7 @@ static const int sig_list[] = {
SIGHUP, SIGINT, SIGQUIT, SIGPIPE, SIGALRM, SIGTERM, SIGWINCH, 0
};
static inline void
MaskSignals(sigset_t * omask)
static inline void MaskSignals(sigset_t * omask)
{
sigset_t mask;
int i;
@@ -112,15 +109,13 @@ MaskSignals(sigset_t * omask)
sigprocmask(SIG_BLOCK, &mask, omask);
}
static inline void
UnmaskSignals(sigset_t * omask)
static inline void UnmaskSignals(sigset_t * omask)
{
sigprocmask(SIG_SETMASK, omask, NULL);
}
static int
HAIKUAUDIO_OpenDevice(_THIS, const char *devname)
static int HAIKUAUDIO_OpenDevice(_THIS, const char *devname)
{
media_raw_audio_format format;
SDL_AudioFormat test_format;
@@ -207,14 +202,12 @@ HAIKUAUDIO_OpenDevice(_THIS, const char *devname)
return 0;
}
static void
HAIKUAUDIO_Deinitialize(void)
static void HAIKUAUDIO_Deinitialize(void)
{
SDL_QuitBeApp();
}
static SDL_bool
HAIKUAUDIO_Init(SDL_AudioDriverImpl * impl)
static SDL_bool HAIKUAUDIO_Init(SDL_AudioDriverImpl * impl)
{
/* Initialize the Be Application, if it's not already started */
if (SDL_InitBeApp() < 0) {

2
src/audio/haiku/SDL_haikuaudio.h
View File

@@ -26,7 +26,7 @@
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *_this
#define _THIS SDL_AudioDevice *_this
struct SDL_PrivateAudioData
{

146
src/audio/jack/SDL_jackaudio.c
View File

@@ -26,36 +26,33 @@
#include "SDL_jackaudio.h"
#include "../../thread/SDL_systhread.h"
static jack_client_t * (*JACK_jack_client_open) (const char *, jack_options_t, jack_status_t *, ...);
static int (*JACK_jack_client_close) (jack_client_t *);
static void (*JACK_jack_on_shutdown) (jack_client_t *, JackShutdownCallback, void *);
static int (*JACK_jack_activate) (jack_client_t *);
static int (*JACK_jack_deactivate) (jack_client_t *);
static void * (*JACK_jack_port_get_buffer) (jack_port_t *, jack_nframes_t);
static int (*JACK_jack_port_unregister) (jack_client_t *, jack_port_t *);
static void (*JACK_jack_free) (void *);
static const char ** (*JACK_jack_get_ports) (jack_client_t *, const char *, const char *, unsigned long);
static jack_nframes_t (*JACK_jack_get_sample_rate) (jack_client_t *);
static jack_nframes_t (*JACK_jack_get_buffer_size) (jack_client_t *);
static jack_port_t * (*JACK_jack_port_register) (jack_client_t *, const char *, const char *, unsigned long, unsigned long);
static jack_port_t * (*JACK_jack_port_by_name) (jack_client_t *, const char *);
static const char * (*JACK_jack_port_name) (const jack_port_t *);
static const char * (*JACK_jack_port_type) (const jack_port_t *);
static int (*JACK_jack_connect) (jack_client_t *, const char *, const char *);
static int (*JACK_jack_set_process_callback) (jack_client_t *, JackProcessCallback, void *);
static jack_client_t *(*JACK_jack_client_open)(const char *, jack_options_t, jack_status_t *, ...);
static int (*JACK_jack_client_close)(jack_client_t *);
static void (*JACK_jack_on_shutdown)(jack_client_t *, JackShutdownCallback, void *);
static int (*JACK_jack_activate)(jack_client_t *);
static int (*JACK_jack_deactivate)(jack_client_t *);
static void *(*JACK_jack_port_get_buffer)(jack_port_t *, jack_nframes_t);
static int (*JACK_jack_port_unregister)(jack_client_t *, jack_port_t *);
static void (*JACK_jack_free)(void *);
static const char **(*JACK_jack_get_ports)(jack_client_t *, const char *, const char *, unsigned long);
static jack_nframes_t (*JACK_jack_get_sample_rate)(jack_client_t *);
static jack_nframes_t (*JACK_jack_get_buffer_size)(jack_client_t *);
static jack_port_t *(*JACK_jack_port_register)(jack_client_t *, const char *, const char *, unsigned long, unsigned long);
static jack_port_t *(*JACK_jack_port_by_name)(jack_client_t *, const char *);
static const char *(*JACK_jack_port_name)(const jack_port_t *);
static const char *(*JACK_jack_port_type)(const jack_port_t *);
static int (*JACK_jack_connect)(jack_client_t *, const char *, const char *);
static int (*JACK_jack_set_process_callback)(jack_client_t *, JackProcessCallback, void *);
static int load_jack_syms(void);
#ifdef SDL_AUDIO_DRIVER_JACK_DYNAMIC
static const char *jack_library = SDL_AUDIO_DRIVER_JACK_DYNAMIC;
static void *jack_handle = NULL;
/* !!! FIXME: this is copy/pasted in several places now */
static int
load_jack_sym(const char *fn, void **addr)
static int load_jack_sym(const char *fn, void **addr)
{
*addr = SDL_LoadFunction(jack_handle, fn);
if (*addr == NULL) {
@@ -67,11 +64,11 @@ load_jack_sym(const char *fn, void **addr)
}
/* cast funcs to char* first, to please GCC's strict aliasing rules. */
#define SDL_JACK_SYM(x) \
if (!load_jack_sym(#x, (void **) (char *) &JACK_##x)) return -1
#define SDL_JACK_SYM(x) \
if (!load_jack_sym(#x, (void **)(char *)&JACK_##x)) \
return -1
static void
UnloadJackLibrary(void)
static void UnloadJackLibrary(void)
{
if (jack_handle != NULL) {
SDL_UnloadObject(jack_handle);
@@ -79,8 +76,7 @@ UnloadJackLibrary(void)
}
}
static int
LoadJackLibrary(void)
static int LoadJackLibrary(void)
{
int retval = 0;
if (jack_handle == NULL) {
@@ -102,13 +98,11 @@ LoadJackLibrary(void)
#define SDL_JACK_SYM(x) JACK_##x = x
static void
UnloadJackLibrary(void)
static void UnloadJackLibrary(void)
{
}
static int
LoadJackLibrary(void)
static int LoadJackLibrary(void)
{
load_jack_syms();
return 0;
@@ -116,9 +110,7 @@ LoadJackLibrary(void)
#endif /* SDL_AUDIO_DRIVER_JACK_DYNAMIC */
static int
load_jack_syms(void)
static int load_jack_syms(void)
{
SDL_JACK_SYM(jack_client_open);
SDL_JACK_SYM(jack_client_close);
@@ -140,23 +132,20 @@ load_jack_syms(void)
return 0;
}
static void
jackShutdownCallback(void *arg) /* JACK went away; device is lost. */
static void jackShutdownCallback(void *arg) /* JACK went away; device is lost. */
{
SDL_AudioDevice *this = (SDL_AudioDevice *) arg;
SDL_AudioDevice *this = (SDL_AudioDevice *)arg;
SDL_OpenedAudioDeviceDisconnected(this);
SDL_SemPost(this->hidden->iosem); /* unblock the SDL thread. */
SDL_SemPost(this->hidden->iosem); /* unblock the SDL thread. */
}
// !!! FIXME: implement and register these!
//typedef int(* JackSampleRateCallback)(jack_nframes_t nframes, void *arg)
//typedef int(* JackBufferSizeCallback)(jack_nframes_t nframes, void *arg)
// typedef int(* JackSampleRateCallback)(jack_nframes_t nframes, void *arg)
// typedef int(* JackBufferSizeCallback)(jack_nframes_t nframes, void *arg)
static int
jackProcessPlaybackCallback(jack_nframes_t nframes, void *arg)
static int jackProcessPlaybackCallback(jack_nframes_t nframes, void *arg)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) arg;
SDL_AudioDevice *this = (SDL_AudioDevice *)arg;
jack_port_t **ports = this->hidden->sdlports;
const int total_channels = this->spec.channels;
const int total_frames = this->spec.samples;
@@ -168,9 +157,9 @@ jackProcessPlaybackCallback(jack_nframes_t nframes, void *arg)
}
for (channelsi = 0; channelsi < total_channels; channelsi++) {
float *dst = (float *) JACK_jack_port_get_buffer(ports[channelsi], nframes);
float *dst = (float *)JACK_jack_port_get_buffer(ports[channelsi], nframes);
if (dst) {
const float *src = ((float *) this->hidden->iobuffer) + channelsi;
const float *src = ((float *)this->hidden->iobuffer) + channelsi;
int framesi;
for (framesi = 0; framesi < total_frames; framesi++) {
*(dst++) = *src;
@@ -179,14 +168,12 @@ jackProcessPlaybackCallback(jack_nframes_t nframes, void *arg)
}
}
SDL_SemPost(this->hidden->iosem); /* tell SDL thread we're done; refill the buffer. */
return 0; /* success */
SDL_SemPost(this->hidden->iosem); /* tell SDL thread we're done; refill the buffer. */
return 0;
}
/* This function waits until it is possible to write a full sound buffer */
static void
JACK_WaitDevice(_THIS)
static void JACK_WaitDevice(_THIS)
{
if (SDL_AtomicGet(&this->enabled)) {
if (SDL_SemWait(this->hidden->iosem) == -1) {
@@ -195,27 +182,24 @@ JACK_WaitDevice(_THIS)
}
}
static Uint8 *
JACK_GetDeviceBuf(_THIS)
static Uint8 *JACK_GetDeviceBuf(_THIS)
{
return (Uint8 *) this->hidden->iobuffer;
return (Uint8 *)this->hidden->iobuffer;
}
static int
jackProcessCaptureCallback(jack_nframes_t nframes, void *arg)
static int jackProcessCaptureCallback(jack_nframes_t nframes, void *arg)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) arg;
SDL_AudioDevice *this = (SDL_AudioDevice *)arg;
if (SDL_AtomicGet(&this->enabled)) {
jack_port_t **ports = this->hidden->sdlports;
const int total_channels = this->spec.channels;
const int total_frames = this->spec.samples;
int channelsi;
for (channelsi = 0; channelsi < total_channels; channelsi++) {
const float *src = (const float *) JACK_jack_port_get_buffer(ports[channelsi], nframes);
const float *src = (const float *)JACK_jack_port_get_buffer(ports[channelsi], nframes);
if (src) {
float *dst = ((float *) this->hidden->iobuffer) + channelsi;
float *dst = ((float *)this->hidden->iobuffer) + channelsi;
int framesi;
for (framesi = 0; framesi < total_frames; framesi++) {
*dst = *(src++);
@@ -225,14 +209,13 @@ jackProcessCaptureCallback(jack_nframes_t nframes, void *arg)
}
}
SDL_SemPost(this->hidden->iosem); /* tell SDL thread we're done; new buffer is ready! */
return 0; /* success */
SDL_SemPost(this->hidden->iosem); /* tell SDL thread we're done; new buffer is ready! */
return 0;
}
static int
JACK_CaptureFromDevice(_THIS, void *buffer, int buflen)
static int JACK_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
SDL_assert(buflen == this->spec.size); /* we always fill a full buffer. */
SDL_assert(buflen == this->spec.size); /* we always fill a full buffer. */
/* Wait for JACK to fill the iobuffer */
if (SDL_SemWait(this->hidden->iosem) == -1) {
@@ -243,15 +226,12 @@ JACK_CaptureFromDevice(_THIS, void *buffer, int buflen)
return buflen;
}
static void
JACK_FlushCapture(_THIS)
static void JACK_FlushCapture(_THIS)
{
SDL_SemWait(this->hidden->iosem);
}
static void
JACK_CloseDevice(_THIS)
static void JACK_CloseDevice(_THIS)
{
if (this->hidden->client) {
JACK_jack_deactivate(this->hidden->client);
@@ -276,8 +256,7 @@ JACK_CloseDevice(_THIS)
SDL_free(this->hidden);
}
static int
JACK_OpenDevice(_THIS, const char *devname)
static int JACK_OpenDevice(_THIS, const char *devname)
{
/* Note that JACK uses "output" for capture devices (they output audio
data to us) and "input" for playback (we input audio data to them).
@@ -297,7 +276,7 @@ JACK_OpenDevice(_THIS, const char *devname)
int i;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, sizeof (*this->hidden));
this->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
@@ -325,7 +304,7 @@ JACK_OpenDevice(_THIS, const char *devname)
const char *type = JACK_jack_port_type(dport);
const int len = SDL_strlen(type);
/* See if type ends with "audio" */
if (len >= 5 && !SDL_memcmp(type+len-5, "audio", 5)) {
if (len >= 5 && !SDL_memcmp(type + len - 5, "audio", 5)) {
audio_ports[channels++] = i;
}
}
@@ -333,7 +312,6 @@ JACK_OpenDevice(_THIS, const char *devname)
return SDL_SetError("No physical JACK ports available");
}
/* !!! FIXME: docs say about buffer size: "This size may change, clients that depend on it must register a bufsize_callback so they will be notified if it does." */
/* Jack pretty much demands what it wants. */
@@ -346,23 +324,23 @@ JACK_OpenDevice(_THIS, const char *devname)
this->hidden->iosem = SDL_CreateSemaphore(0);
if (!this->hidden->iosem) {
return -1; /* error was set by SDL_CreateSemaphore */
return -1; /* error was set by SDL_CreateSemaphore */
}
this->hidden->iobuffer = (float *) SDL_calloc(1, this->spec.size);
this->hidden->iobuffer = (float *)SDL_calloc(1, this->spec.size);
if (!this->hidden->iobuffer) {
return SDL_OutOfMemory();
}
/* Build SDL's ports, which we will connect to the device ports. */
this->hidden->sdlports = (jack_port_t **) SDL_calloc(channels, sizeof (jack_port_t *));
this->hidden->sdlports = (jack_port_t **)SDL_calloc(channels, sizeof(jack_port_t *));
if (this->hidden->sdlports == NULL) {
return SDL_OutOfMemory();
}
for (i = 0; i < channels; i++) {
char portname[32];
SDL_snprintf(portname, sizeof (portname), "sdl_jack_%s_%d", sdlportstr, i);
SDL_snprintf(portname, sizeof(portname), "sdl_jack_%s_%d", sdlportstr, i);
this->hidden->sdlports[i] = JACK_jack_port_register(client, portname, JACK_DEFAULT_AUDIO_TYPE, sdlportflags, 0);
if (this->hidden->sdlports[i] == NULL) {
return SDL_SetError("jack_port_register failed");
@@ -397,14 +375,12 @@ JACK_OpenDevice(_THIS, const char *devname)
return 0;
}
static void
JACK_Deinitialize(void)
static void JACK_Deinitialize(void)
{
UnloadJackLibrary();
}
static SDL_bool
JACK_Init(SDL_AudioDriverImpl * impl)
static SDL_bool JACK_Init(SDL_AudioDriverImpl *impl)
{
if (LoadJackLibrary() < 0) {
return SDL_FALSE;
@@ -431,7 +407,7 @@ JACK_Init(SDL_AudioDriverImpl * impl)
impl->OnlyHasDefaultCaptureDevice = SDL_TRUE;
impl->HasCaptureSupport = SDL_TRUE;
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE; /* this audio target is available. */
}
AudioBootStrap JACK_bootstrap = {

61
src/audio/n3ds/SDL_n3dsaudio.c
View File

@@ -22,7 +22,6 @@
#ifdef SDL_AUDIO_DRIVER_N3DS
/* N3DS Audio driver */
#include "../SDL_sysaudio.h"
@@ -36,32 +35,27 @@ static SDL_AudioDevice *audio_device;
static void FreePrivateData(_THIS);
static int FindAudioFormat(_THIS);
static SDL_INLINE void
contextLock(_THIS)
static SDL_INLINE void contextLock(_THIS)
{
LightLock_Lock(&this->hidden->lock);
}
static SDL_INLINE void
contextUnlock(_THIS)
static SDL_INLINE void contextUnlock(_THIS)
{
LightLock_Unlock(&this->hidden->lock);
}
static void
N3DSAUD_LockAudio(_THIS)
static void N3DSAUD_LockAudio(_THIS)
{
contextLock(this);
}
static void
N3DSAUD_UnlockAudio(_THIS)
static void N3DSAUD_UnlockAudio(_THIS)
{
contextUnlock(this);
}
static void
N3DSAUD_DspHook(DSP_HookType hook)
static void N3DSAUD_DspHook(DSP_HookType hook)
{
if (hook == DSPHOOK_ONCANCEL) {
contextLock(audio_device);
@@ -72,12 +66,11 @@ N3DSAUD_DspHook(DSP_HookType hook)
}
}
static void
AudioFrameFinished(void *device)
static void AudioFrameFinished(void *device)
{
bool shouldBroadcast = false;
unsigned i;
SDL_AudioDevice *this = (SDL_AudioDevice *) device;
SDL_AudioDevice *this = (SDL_AudioDevice *)device;
contextLock(this);
@@ -95,13 +88,12 @@ AudioFrameFinished(void *device)
contextUnlock(this);
}
static int
N3DSAUDIO_OpenDevice(_THIS, const char *devname)
static int N3DSAUDIO_OpenDevice(_THIS, const char *devname)
{
Result ndsp_init_res;
Uint8 *data_vaddr;
float mix[12];
this->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, sizeof *this->hidden);
this->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof *this->hidden);
if (this->hidden == NULL) {
return SDL_OutOfMemory();
@@ -140,14 +132,14 @@ N3DSAUDIO_OpenDevice(_THIS, const char *devname)
}
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->spec.size);
this->hidden->mixbuf = (Uint8 *)SDL_malloc(this->spec.size);
if (this->hidden->mixbuf == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
data_vaddr = (Uint8 *) linearAlloc(this->hidden->mixlen * NUM_BUFFERS);
data_vaddr = (Uint8 *)linearAlloc(this->hidden->mixlen * NUM_BUFFERS);
if (data_vaddr == NULL) {
return SDL_OutOfMemory();
}
@@ -186,8 +178,7 @@ N3DSAUDIO_OpenDevice(_THIS, const char *devname)
return 0;
}
static int
N3DSAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
static int N3DSAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
/* Delay to make this sort of simulate real audio input. */
SDL_Delay((this->spec.samples * 1000) / this->spec.freq);
@@ -197,8 +188,7 @@ N3DSAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
return buflen;
}
static void
N3DSAUDIO_PlayDevice(_THIS)
static void N3DSAUDIO_PlayDevice(_THIS)
{
size_t nextbuf;
size_t sampleLen;
@@ -217,15 +207,14 @@ N3DSAUDIO_PlayDevice(_THIS)
contextUnlock(this);
memcpy((void *) this->hidden->waveBuf[nextbuf].data_vaddr,
memcpy((void *)this->hidden->waveBuf[nextbuf].data_vaddr,
this->hidden->mixbuf, sampleLen);
DSP_FlushDataCache(this->hidden->waveBuf[nextbuf].data_vaddr, sampleLen);
ndspChnWaveBufAdd(0, &this->hidden->waveBuf[nextbuf]);
}
static void
N3DSAUDIO_WaitDevice(_THIS)
static void N3DSAUDIO_WaitDevice(_THIS)
{
contextLock(this);
while (!this->hidden->isCancelled &&
@@ -235,14 +224,12 @@ N3DSAUDIO_WaitDevice(_THIS)
contextUnlock(this);
}
static Uint8 *
N3DSAUDIO_GetDeviceBuf(_THIS)
static Uint8 *N3DSAUDIO_GetDeviceBuf(_THIS)
{
return this->hidden->mixbuf;
}
static void
N3DSAUDIO_CloseDevice(_THIS)
static void N3DSAUDIO_CloseDevice(_THIS)
{
contextLock(this);
@@ -262,8 +249,7 @@ N3DSAUDIO_CloseDevice(_THIS)
FreePrivateData(this);
}
static void
N3DSAUDIO_ThreadInit(_THIS)
static void N3DSAUDIO_ThreadInit(_THIS)
{
s32 current_priority;
svcGetThreadPriority(&current_priority, CUR_THREAD_HANDLE);
@@ -273,8 +259,7 @@ N3DSAUDIO_ThreadInit(_THIS)
svcSetThreadPriority(CUR_THREAD_HANDLE, current_priority);
}
static SDL_bool
N3DSAUDIO_Init(SDL_AudioDriverImpl *impl)
static SDL_bool N3DSAUDIO_Init(SDL_AudioDriverImpl *impl)
{
/* Set the function pointers */
impl->OpenDevice = N3DSAUDIO_OpenDevice;
@@ -304,15 +289,14 @@ AudioBootStrap N3DSAUDIO_bootstrap = {
/**
* Cleans up all allocated memory, safe to call with null pointers
*/
static void
FreePrivateData(_THIS)
static void FreePrivateData(_THIS)
{
if (!this->hidden) {
return;
}
if (this->hidden->waveBuf[0].data_vaddr) {
linearFree((void *) this->hidden->waveBuf[0].data_vaddr);
linearFree((void *)this->hidden->waveBuf[0].data_vaddr);
}
if (this->hidden->mixbuf) {
@@ -324,8 +308,7 @@ FreePrivateData(_THIS)
this->hidden = NULL;
}
static int
FindAudioFormat(_THIS)
static int FindAudioFormat(_THIS)
{
SDL_bool found_valid_format = SDL_FALSE;
Uint16 test_format = SDL_FirstAudioFormat(this->spec.format);

49
src/audio/netbsd/SDL_netbsdaudio.c
View File

@@ -43,15 +43,12 @@
/* #define DEBUG_AUDIO */
static void
NETBSDAUDIO_DetectDevices(void)
static void NETBSDAUDIO_DetectDevices(void)
{
SDL_EnumUnixAudioDevices(0, NULL);
}
static void
NETBSDAUDIO_Status(_THIS)
static void NETBSDAUDIO_Status(_THIS)
{
#ifdef DEBUG_AUDIO
/* *INDENT-OFF* */ /* clang-format off */
@@ -117,12 +114,11 @@ NETBSDAUDIO_Status(_THIS)
this->spec.format,
this->spec.size);
/* *INDENT-ON* */ /* clang-format on */
#endif /* DEBUG_AUDIO */
}
static void
NETBSDAUDIO_PlayDevice(_THIS)
static void NETBSDAUDIO_PlayDevice(_THIS)
{
struct SDL_PrivateAudioData *h = this->hidden;
int written;
@@ -141,17 +137,14 @@ NETBSDAUDIO_PlayDevice(_THIS)
#endif
}
static Uint8 *
NETBSDAUDIO_GetDeviceBuf(_THIS)
static Uint8 *NETBSDAUDIO_GetDeviceBuf(_THIS)
{
return this->hidden->mixbuf;
}
static int
NETBSDAUDIO_CaptureFromDevice(_THIS, void *_buffer, int buflen)
static int NETBSDAUDIO_CaptureFromDevice(_THIS, void *_buffer, int buflen)
{
Uint8 *buffer = (Uint8 *) _buffer;
Uint8 *buffer = (Uint8 *)_buffer;
int br;
br = read(this->hidden->audio_fd, buffer, buflen);
@@ -167,30 +160,28 @@ NETBSDAUDIO_CaptureFromDevice(_THIS, void *_buffer, int buflen)
return 0;
}
static void
NETBSDAUDIO_FlushCapture(_THIS)
static void NETBSDAUDIO_FlushCapture(_THIS)
{
audio_info_t info;
size_t remain;
Uint8 buf[512];
if (ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info) < 0) {
return; /* oh well. */
return; /* oh well. */
}
remain = (size_t) (info.record.samples * (SDL_AUDIO_BITSIZE(this->spec.format) / 8));
remain = (size_t)(info.record.samples * (SDL_AUDIO_BITSIZE(this->spec.format) / 8));
while (remain > 0) {
const size_t len = SDL_min(sizeof (buf), remain);
const size_t len = SDL_min(sizeof(buf), remain);
const int br = read(this->hidden->audio_fd, buf, len);
if (br <= 0) {
return; /* oh well. */
return; /* oh well. */
}
remain -= br;
}
}
static void
NETBSDAUDIO_CloseDevice(_THIS)
static void NETBSDAUDIO_CloseDevice(_THIS)
{
if (this->hidden->audio_fd >= 0) {
close(this->hidden->audio_fd);
@@ -199,8 +190,7 @@ NETBSDAUDIO_CloseDevice(_THIS)
SDL_free(this->hidden);
}
static int
NETBSDAUDIO_OpenDevice(_THIS, const char *devname)
static int NETBSDAUDIO_OpenDevice(_THIS, const char *devname)
{
SDL_bool iscapture = this->iscapture;
SDL_AudioFormat test_format;
@@ -239,8 +229,7 @@ NETBSDAUDIO_OpenDevice(_THIS, const char *devname)
* Use the device's native sample rate so the kernel doesn't have to
* resample.
*/
this->spec.freq = iscapture ?
hwinfo.record.sample_rate : hwinfo.play.sample_rate;
this->spec.freq = iscapture ? hwinfo.record.sample_rate : hwinfo.play.sample_rate;
}
#endif
@@ -305,7 +294,7 @@ NETBSDAUDIO_OpenDevice(_THIS, const char *devname)
if (!iscapture) {
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
this->hidden->mixbuf = (Uint8 *)SDL_malloc(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
return SDL_OutOfMemory();
}
@@ -318,8 +307,7 @@ NETBSDAUDIO_OpenDevice(_THIS, const char *devname)
return 0;
}
static SDL_bool
NETBSDAUDIO_Init(SDL_AudioDriverImpl * impl)
static SDL_bool NETBSDAUDIO_Init(SDL_AudioDriverImpl *impl)
{
/* Set the function pointers */
impl->DetectDevices = NETBSDAUDIO_DetectDevices;
@@ -333,10 +321,9 @@ NETBSDAUDIO_Init(SDL_AudioDriverImpl * impl)
impl->HasCaptureSupport = SDL_TRUE;
impl->AllowsArbitraryDeviceNames = SDL_TRUE;
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE; /* this audio target is available. */
}
AudioBootStrap NETBSDAUDIO_bootstrap = {
"netbsd", "NetBSD audio", NETBSDAUDIO_Init, SDL_FALSE
};

4
src/audio/netbsd/SDL_netbsdaudio.h
View File

@@ -25,7 +25,7 @@
#include "../SDL_sysaudio.h"
#define _THIS SDL_AudioDevice *this
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
@@ -41,7 +41,7 @@ struct SDL_PrivateAudioData
float next_frame;
};
#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
#endif /* SDL_netbsdaudio_h_ */

112
src/audio/openslES/SDL_openslES.c
View File

@@ -37,9 +37,9 @@
#include <android/log.h>
#if 0
#define LOG_TAG "SDL_openslES"
#define LOGE(...) __android_log_print(ANDROID_LOG_ERROR,LOG_TAG,__VA_ARGS__)
#define LOGI(...) __android_log_print(ANDROID_LOG_INFO,LOG_TAG,__VA_ARGS__)
#define LOG_TAG "SDL_openslES"
#define LOGE(...) __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__)
#define LOGI(...) __android_log_print(ANDROID_LOG_INFO, LOG_TAG, __VA_ARGS__)
//#define LOGV(...) __android_log_print(ANDROID_LOG_VERBOSE,LOG_TAG,__VA_ARGS__)
#define LOGV(...)
#else
@@ -69,9 +69,9 @@
#define SL_SPEAKER_TOP_BACK_RIGHT ((SLuint32) 0x00020000)
*/
#define SL_ANDROID_SPEAKER_STEREO (SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT)
#define SL_ANDROID_SPEAKER_QUAD (SL_ANDROID_SPEAKER_STEREO | SL_SPEAKER_BACK_LEFT | SL_SPEAKER_BACK_RIGHT)
#define SL_ANDROID_SPEAKER_5DOT1 (SL_ANDROID_SPEAKER_QUAD | SL_SPEAKER_FRONT_CENTER | SL_SPEAKER_LOW_FREQUENCY)
#define SL_ANDROID_SPEAKER_7DOT1 (SL_ANDROID_SPEAKER_5DOT1 | SL_SPEAKER_SIDE_LEFT | SL_SPEAKER_SIDE_RIGHT)
#define SL_ANDROID_SPEAKER_QUAD (SL_ANDROID_SPEAKER_STEREO | SL_SPEAKER_BACK_LEFT | SL_SPEAKER_BACK_RIGHT)
#define SL_ANDROID_SPEAKER_5DOT1 (SL_ANDROID_SPEAKER_QUAD | SL_SPEAKER_FRONT_CENTER | SL_SPEAKER_LOW_FREQUENCY)
#define SL_ANDROID_SPEAKER_7DOT1 (SL_ANDROID_SPEAKER_5DOT1 | SL_SPEAKER_SIDE_LEFT | SL_SPEAKER_SIDE_RIGHT)
/* engine interfaces */
static SLObjectItf engineObject = NULL;
@@ -97,8 +97,8 @@ static SLAndroidSimpleBufferQueueItf recorderBufferQueue = NULL;
static const char *sldevaudiorecorderstr = "SLES Audio Recorder";
static const char *sldevaudioplayerstr = "SLES Audio Player";
#define SLES_DEV_AUDIO_RECORDER sldevaudiorecorderstr
#define SLES_DEV_AUDIO_PLAYER sldevaudioplayerstr
#define SLES_DEV_AUDIO_RECORDER sldevaudiorecorderstr
#define SLES_DEV_AUDIO_PLAYER sldevaudioplayerstr
static void openslES_DetectDevices( int iscapture )
{
LOGI( "openSLES_DetectDevices()" );
@@ -127,8 +127,7 @@ static void openslES_DestroyEngine(void)
}
}
static int
openslES_CreateEngine(void)
static int openslES_CreateEngine(void)
{
const SLInterfaceID ids[1] = { SL_IID_VOLUME };
const SLboolean req[1] = { SL_BOOLEAN_FALSE };
@@ -182,17 +181,15 @@ error:
}
/* this callback handler is called every time a buffer finishes recording */
static void
bqRecorderCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
static void bqRecorderCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
{
struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *) context;
struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *)context;
LOGV("SLES: Recording Callback");
SDL_SemPost(audiodata->playsem);
}
static void
openslES_DestroyPCMRecorder(_THIS)
static void openslES_DestroyPCMRecorder(_THIS)
{
struct SDL_PrivateAudioData *audiodata = this->hidden;
SLresult result;
@@ -223,8 +220,7 @@ openslES_DestroyPCMRecorder(_THIS)
}
}
static int
openslES_CreatePCMRecorder(_THIS)
static int openslES_CreatePCMRecorder(_THIS)
{
struct SDL_PrivateAudioData *audiodata = this->hidden;
SLDataFormat_PCM format_pcm;
@@ -251,8 +247,8 @@ openslES_CreatePCMRecorder(_THIS)
SDL_CalculateAudioSpec(&this->spec);
LOGI("Try to open %u hz %u bit chan %u %s samples %u",
this->spec.freq, SDL_AUDIO_BITSIZE(this->spec.format),
this->spec.channels, (this->spec.format & 0x1000) ? "BE" : "LE", this->spec.samples);
this->spec.freq, SDL_AUDIO_BITSIZE(this->spec.format),
this->spec.channels, (this->spec.format & 0x1000) ? "BE" : "LE", this->spec.samples);
/* configure audio source */
loc_dev.locatorType = SL_DATALOCATOR_IODEVICE;
@@ -266,13 +262,13 @@ openslES_CreatePCMRecorder(_THIS)
loc_bufq.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
loc_bufq.numBuffers = NUM_BUFFERS;
format_pcm.formatType = SL_DATAFORMAT_PCM;
format_pcm.numChannels = this->spec.channels;
format_pcm.samplesPerSec = this->spec.freq * 1000; /* / kilo Hz to milli Hz */
format_pcm.formatType = SL_DATAFORMAT_PCM;
format_pcm.numChannels = this->spec.channels;
format_pcm.samplesPerSec = this->spec.freq * 1000; /* / kilo Hz to milli Hz */
format_pcm.bitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
format_pcm.containerSize = SDL_AUDIO_BITSIZE(this->spec.format);
format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
format_pcm.channelMask = SL_SPEAKER_FRONT_CENTER;
format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
format_pcm.channelMask = SL_SPEAKER_FRONT_CENTER;
audioSnk.pLocator = &loc_bufq;
audioSnk.pFormat = &format_pcm;
@@ -322,7 +318,7 @@ openslES_CreatePCMRecorder(_THIS)
}
/* Create the sound buffers */
audiodata->mixbuff = (Uint8 *) SDL_malloc(NUM_BUFFERS * this->spec.size);
audiodata->mixbuff = (Uint8 *)SDL_malloc(NUM_BUFFERS * this->spec.size);
if (audiodata->mixbuff == NULL) {
LOGE("mixbuffer allocate - out of memory");
goto failed;
@@ -362,17 +358,15 @@ failed:
}
/* this callback handler is called every time a buffer finishes playing */
static void
bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
static void bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
{
struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *) context;
struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *)context;
LOGV("SLES: Playback Callback");
SDL_SemPost(audiodata->playsem);
}
static void
openslES_DestroyPCMPlayer(_THIS)
static void openslES_DestroyPCMPlayer(_THIS)
{
struct SDL_PrivateAudioData *audiodata = this->hidden;
SLresult result;
@@ -404,8 +398,7 @@ openslES_DestroyPCMPlayer(_THIS)
}
}
static int
openslES_CreatePCMPlayer(_THIS)
static int openslES_CreatePCMPlayer(_THIS)
{
struct SDL_PrivateAudioData *audiodata = this->hidden;
SLDataLocator_AndroidSimpleBufferQueue loc_bufq;
@@ -433,7 +426,7 @@ openslES_CreatePCMPlayer(_THIS)
if (!test_format) {
/* Didn't find a compatible format : */
LOGI( "No compatible audio format, using signed 16-bit audio" );
LOGI("No compatible audio format, using signed 16-bit audio");
test_format = AUDIO_S16SYS;
}
this->spec.format = test_format;
@@ -446,16 +439,16 @@ openslES_CreatePCMPlayer(_THIS)
SDL_CalculateAudioSpec(&this->spec);
LOGI("Try to open %u hz %s %u bit chan %u %s samples %u",
this->spec.freq, SDL_AUDIO_ISFLOAT(this->spec.format) ? "float" : "pcm", SDL_AUDIO_BITSIZE(this->spec.format),
this->spec.channels, (this->spec.format & 0x1000) ? "BE" : "LE", this->spec.samples);
this->spec.freq, SDL_AUDIO_ISFLOAT(this->spec.format) ? "float" : "pcm", SDL_AUDIO_BITSIZE(this->spec.format),
this->spec.channels, (this->spec.format & 0x1000) ? "BE" : "LE", this->spec.samples);
/* configure audio source */
loc_bufq.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
loc_bufq.numBuffers = NUM_BUFFERS;
format_pcm.formatType = SL_DATAFORMAT_PCM;
format_pcm.numChannels = this->spec.channels;
format_pcm.samplesPerSec = this->spec.freq * 1000; /* / kilo Hz to milli Hz */
format_pcm.formatType = SL_DATAFORMAT_PCM;
format_pcm.numChannels = this->spec.channels;
format_pcm.samplesPerSec = this->spec.freq * 1000; /* / kilo Hz to milli Hz */
format_pcm.bitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
format_pcm.containerSize = SDL_AUDIO_BITSIZE(this->spec.format);
@@ -465,8 +458,7 @@ openslES_CreatePCMPlayer(_THIS)
format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
}
switch (this->spec.channels)
{
switch (this->spec.channels) {
case 1:
format_pcm.channelMask = SL_SPEAKER_FRONT_LEFT;
break;
@@ -511,7 +503,7 @@ openslES_CreatePCMPlayer(_THIS)
}
audioSrc.pLocator = &loc_bufq;
audioSrc.pFormat = SDL_AUDIO_ISFLOAT(this->spec.format) ? (void*)&format_pcm_ex : (void*)&format_pcm;
audioSrc.pFormat = SDL_AUDIO_ISFLOAT(this->spec.format) ? (void *)&format_pcm_ex : (void *)&format_pcm;
/* configure audio sink */
loc_outmix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
@@ -572,7 +564,7 @@ openslES_CreatePCMPlayer(_THIS)
}
/* Create the sound buffers */
audiodata->mixbuff = (Uint8 *) SDL_malloc(NUM_BUFFERS * this->spec.size);
audiodata->mixbuff = (Uint8 *)SDL_malloc(NUM_BUFFERS * this->spec.size);
if (audiodata->mixbuff == NULL) {
LOGE("mixbuffer allocate - out of memory");
goto failed;
@@ -595,10 +587,9 @@ failed:
return -1;
}
static int
openslES_OpenDevice(_THIS, const char *devname)
static int openslES_OpenDevice(_THIS, const char *devname)
{
this->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, (sizeof *this->hidden));
this->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, (sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
@@ -624,12 +615,10 @@ openslES_OpenDevice(_THIS, const char *devname)
} else {
return SDL_SetError("Open device failed!");
}
}
}
static void
openslES_WaitDevice(_THIS)
static void openslES_WaitDevice(_THIS)
{
struct SDL_PrivateAudioData *audiodata = this->hidden;
@@ -639,8 +628,7 @@ openslES_WaitDevice(_THIS)
SDL_SemWait(audiodata->playsem);
}
static void
openslES_PlayDevice(_THIS)
static void openslES_PlayDevice(_THIS)
{
struct SDL_PrivateAudioData *audiodata = this->hidden;
SLresult result;
@@ -674,8 +662,7 @@ openslES_PlayDevice(_THIS)
/* */
/* okay.. */
static Uint8 *
openslES_GetDeviceBuf(_THIS)
static Uint8 *openslES_GetDeviceBuf(_THIS)
{
struct SDL_PrivateAudioData *audiodata = this->hidden;
@@ -683,8 +670,7 @@ openslES_GetDeviceBuf(_THIS)
return audiodata->pmixbuff[audiodata->next_buffer];
}
static int
openslES_CaptureFromDevice(_THIS, void *buffer, int buflen)
static int openslES_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
struct SDL_PrivateAudioData *audiodata = this->hidden;
SLresult result;
@@ -711,8 +697,7 @@ openslES_CaptureFromDevice(_THIS, void *buffer, int buflen)
return this->spec.size;
}
static void
openslES_CloseDevice(_THIS)
static void openslES_CloseDevice(_THIS)
{
/* struct SDL_PrivateAudioData *audiodata = this->hidden; */
@@ -727,8 +712,7 @@ openslES_CloseDevice(_THIS)
SDL_free(this->hidden);
}
static SDL_bool
openslES_Init(SDL_AudioDriverImpl * impl)
static SDL_bool openslES_Init(SDL_AudioDriverImpl *impl)
{
LOGI("openslES_Init() called");
@@ -740,13 +724,13 @@ openslES_Init(SDL_AudioDriverImpl * impl)
/* Set the function pointers */
/* impl->DetectDevices = openslES_DetectDevices; */
impl->OpenDevice = openslES_OpenDevice;
impl->WaitDevice = openslES_WaitDevice;
impl->PlayDevice = openslES_PlayDevice;
impl->GetDeviceBuf = openslES_GetDeviceBuf;
impl->OpenDevice = openslES_OpenDevice;
impl->WaitDevice = openslES_WaitDevice;
impl->PlayDevice = openslES_PlayDevice;
impl->GetDeviceBuf = openslES_GetDeviceBuf;
impl->CaptureFromDevice = openslES_CaptureFromDevice;
impl->CloseDevice = openslES_CloseDevice;
impl->Deinitialize = openslES_DestroyEngine;
impl->CloseDevice = openslES_CloseDevice;
impl->Deinitialize = openslES_DestroyEngine;
/* and the capabilities */
impl->HasCaptureSupport = SDL_TRUE;

10
src/audio/openslES/SDL_openslES.h
View File

@@ -26,15 +26,15 @@
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define _THIS SDL_AudioDevice *this
#define NUM_BUFFERS 2 /* -- Don't lower this! */
#define NUM_BUFFERS 2 /* -- Don't lower this! */
struct SDL_PrivateAudioData
{
Uint8 *mixbuff;
int next_buffer;
Uint8 *pmixbuff[NUM_BUFFERS];
Uint8 *mixbuff;
int next_buffer;
Uint8 *pmixbuff[NUM_BUFFERS];
SDL_sem *playsem;
};

287
src/audio/pipewire/SDL_pipewire.c
View File

@@ -74,7 +74,7 @@ enum PW_READY_FLAGS
{
PW_READY_FLAG_BUFFER_ADDED = 0x1,
PW_READY_FLAG_STREAM_READY = 0x2,
PW_READY_FLAG_ALL_BITS = 0x3
PW_READY_FLAG_ALL_BITS = 0x3
};
#define PW_ID_TO_HANDLE(x) (void *)((uintptr_t)x)
@@ -114,17 +114,16 @@ static struct pw_properties *(*PIPEWIRE_pw_properties_new)(const char *, ...)SPA
static int (*PIPEWIRE_pw_properties_set)(struct pw_properties *, const char *, const char *);
static int (*PIPEWIRE_pw_properties_setf)(struct pw_properties *, const char *, const char *, ...) SPA_PRINTF_FUNC(3, 4);
static int pipewire_version_major;
static int pipewire_version_minor;
static int pipewire_version_patch;
static int pipewire_version_major;
static int pipewire_version_minor;
static int pipewire_version_patch;
#ifdef SDL_AUDIO_DRIVER_PIPEWIRE_DYNAMIC
static const char *pipewire_library = SDL_AUDIO_DRIVER_PIPEWIRE_DYNAMIC;
static void *pipewire_handle = NULL;
static void *pipewire_handle = NULL;
static int
pipewire_dlsym(const char *fn, void **addr)
static int pipewire_dlsym(const char *fn, void **addr)
{
*addr = SDL_LoadFunction(pipewire_handle, fn);
if (*addr == NULL) {
@@ -140,8 +139,7 @@ pipewire_dlsym(const char *fn, void **addr)
return -1; \
}
static int
load_pipewire_library()
static int load_pipewire_library()
{
if ((pipewire_handle = SDL_LoadObject(pipewire_library))) {
return 0;
@@ -150,8 +148,7 @@ load_pipewire_library()
return -1;
}
static void
unload_pipewire_library()
static void unload_pipewire_library()
{
if (pipewire_handle) {
SDL_UnloadObject(pipewire_handle);
@@ -163,21 +160,18 @@ unload_pipewire_library()
#define SDL_PIPEWIRE_SYM(x) PIPEWIRE_##x = x
static int
load_pipewire_library()
static int load_pipewire_library()
{
return 0;
}
static void
unload_pipewire_library()
static void unload_pipewire_library()
{ /* Nothing to do */
}
#endif /* SDL_AUDIO_DRIVER_PIPEWIRE_DYNAMIC */
static int
load_pipewire_syms()
static int load_pipewire_syms()
{
SDL_PIPEWIRE_SYM(pw_get_library_version);
SDL_PIPEWIRE_SYM(pw_init);
@@ -219,8 +213,7 @@ pipewire_version_at_least(int major, int minor, int patch)
(pipewire_version_major > major || pipewire_version_minor > minor || pipewire_version_patch >= patch);
}
static int
init_pipewire_library()
static int init_pipewire_library()
{
if (!load_pipewire_library()) {
if (!load_pipewire_syms()) {
@@ -242,8 +235,7 @@ init_pipewire_library()
return -1;
}
static void
deinit_pipewire_library()
static void deinit_pipewire_library()
{
PIPEWIRE_pw_deinit();
unload_pipewire_library();
@@ -254,8 +246,8 @@ struct node_object
{
struct spa_list link;
Uint32 id;
int seq;
Uint32 id;
int seq;
SDL_bool persist;
/*
@@ -268,8 +260,8 @@ struct node_object
void *userdata;
struct pw_proxy *proxy;
struct spa_hook node_listener;
struct spa_hook core_listener;
struct spa_hook node_listener;
struct spa_hook core_listener;
};
/* A sink/source node used for stream I/O. */
@@ -277,8 +269,8 @@ struct io_node
{
struct spa_list link;
Uint32 id;
SDL_bool is_capture;
Uint32 id;
SDL_bool is_capture;
SDL_AudioSpec spec;
const char *name; /* Friendly name */
@@ -289,26 +281,25 @@ struct io_node
/* The global hotplug thread and associated objects. */
static struct pw_thread_loop *hotplug_loop;
static struct pw_core *hotplug_core;
static struct pw_context *hotplug_context;
static struct pw_registry *hotplug_registry;
static struct spa_hook hotplug_registry_listener;
static struct spa_hook hotplug_core_listener;
static struct spa_list hotplug_pending_list;
static struct spa_list hotplug_io_list;
static int hotplug_init_seq_val;
static SDL_bool hotplug_init_complete;
static SDL_bool hotplug_events_enabled;
static struct pw_core *hotplug_core;
static struct pw_context *hotplug_context;
static struct pw_registry *hotplug_registry;
static struct spa_hook hotplug_registry_listener;
static struct spa_hook hotplug_core_listener;
static struct spa_list hotplug_pending_list;
static struct spa_list hotplug_io_list;
static int hotplug_init_seq_val;
static SDL_bool hotplug_init_complete;
static SDL_bool hotplug_events_enabled;
static char *pipewire_default_sink_id = NULL;
static char *pipewire_default_sink_id = NULL;
static char *pipewire_default_source_id = NULL;
/* The active node list */
static SDL_bool
io_list_check_add(struct io_node *node)
static SDL_bool io_list_check_add(struct io_node *node)
{
struct io_node *n;
SDL_bool ret = SDL_TRUE;
SDL_bool ret = SDL_TRUE;
/* See if the node is already in the list */
spa_list_for_each (n, &hotplug_io_list, link) {
@@ -330,8 +321,7 @@ dup_found:
return ret;
}
static void
io_list_remove(Uint32 id)
static void io_list_remove(Uint32 id)
{
struct io_node *n, *temp;
@@ -351,8 +341,7 @@ io_list_remove(Uint32 id)
}
}
static void
io_list_sort()
static void io_list_sort()
{
struct io_node *default_sink = NULL, *default_source = NULL;
struct io_node *n, *temp;
@@ -377,8 +366,7 @@ io_list_sort()
}
}
static void
io_list_clear()
static void io_list_clear()
{
struct io_node *n, *temp;
@@ -388,7 +376,7 @@ io_list_clear()
}
}
static struct io_node*
static struct io_node *
io_list_get_by_id(Uint32 id)
{
struct io_node *n, *temp;
@@ -400,7 +388,7 @@ io_list_get_by_id(Uint32 id)
return NULL;
}
static struct io_node*
static struct io_node *
io_list_get_by_path(char *path)
{
struct io_node *n, *temp;
@@ -412,8 +400,7 @@ io_list_get_by_path(char *path)
return NULL;
}
static void
node_object_destroy(struct node_object *node)
static void node_object_destroy(struct node_object *node)
{
SDL_assert(node);
@@ -425,15 +412,13 @@ node_object_destroy(struct node_object *node)
}
/* The pending node list */
static void
pending_list_add(struct node_object *node)
static void pending_list_add(struct node_object *node)
{
SDL_assert(node);
spa_list_append(&hotplug_pending_list, &node->link);
}
static void
pending_list_remove(Uint32 id)
static void pending_list_remove(Uint32 id)
{
struct node_object *node, *temp;
@@ -444,8 +429,7 @@ pending_list_remove(Uint32 id)
}
}
static void
pending_list_clear()
static void pending_list_clear()
{
struct node_object *node, *temp;
@@ -454,10 +438,9 @@ pending_list_clear()
}
}
static void *
node_object_new(Uint32 id, const char *type, Uint32 version, const void *funcs, const struct pw_core_events *core_events)
static void *node_object_new(Uint32 id, const char *type, Uint32 version, const void *funcs, const struct pw_core_events *core_events)
{
struct pw_proxy *proxy;
struct pw_proxy *proxy;
struct node_object *node;
/* Create the proxy object */
@@ -470,7 +453,7 @@ node_object_new(Uint32 id, const char *type, Uint32 version, const void *funcs,
node = PIPEWIRE_pw_proxy_get_user_data(proxy);
SDL_zerop(node);
node->id = id;
node->id = id;
node->proxy = proxy;
/* Add the callbacks */
@@ -484,8 +467,7 @@ node_object_new(Uint32 id, const char *type, Uint32 version, const void *funcs,
}
/* Core sync points */
static void
core_events_hotplug_init_callback(void *object, uint32_t id, int seq)
static void core_events_hotplug_init_callback(void *object, uint32_t id, int seq)
{
if (id == PW_ID_CORE && seq == hotplug_init_seq_val) {
/* This core listener is no longer needed. */
@@ -497,11 +479,10 @@ core_events_hotplug_init_callback(void *object, uint32_t id, int seq)
}
}
static void
core_events_interface_callback(void *object, uint32_t id, int seq)
static void core_events_interface_callback(void *object, uint32_t id, int seq)
{
struct node_object *node = object;
struct io_node *io = node->userdata;
struct io_node *io = node->userdata;
if (id == PW_ID_CORE && seq == node->seq) {
/*
@@ -516,8 +497,7 @@ core_events_interface_callback(void *object, uint32_t id, int seq)
}
}
static void
core_events_metadata_callback(void *object, uint32_t id, int seq)
static void core_events_metadata_callback(void *object, uint32_t id, int seq)
{
struct node_object *node = object;
@@ -527,11 +507,10 @@ core_events_metadata_callback(void *object, uint32_t id, int seq)
}
static const struct pw_core_events hotplug_init_core_events = { PW_VERSION_CORE_EVENTS, .done = core_events_hotplug_init_callback };
static const struct pw_core_events interface_core_events = { PW_VERSION_CORE_EVENTS, .done = core_events_interface_callback };
static const struct pw_core_events metadata_core_events = { PW_VERSION_CORE_EVENTS, .done = core_events_metadata_callback };
static const struct pw_core_events interface_core_events = { PW_VERSION_CORE_EVENTS, .done = core_events_interface_callback };
static const struct pw_core_events metadata_core_events = { PW_VERSION_CORE_EVENTS, .done = core_events_metadata_callback };
static void
hotplug_core_sync(struct node_object *node)
static void hotplug_core_sync(struct node_object *node)
{
/*
* Node sync events *must* come before the hotplug init sync events or the initial
@@ -547,12 +526,11 @@ hotplug_core_sync(struct node_object *node)
}
/* Helpers for retrieving values from params */
static SDL_bool
get_range_param(const struct spa_pod *param, Uint32 key, int *def, int *min, int *max)
static SDL_bool get_range_param(const struct spa_pod *param, Uint32 key, int *def, int *min, int *max)
{
const struct spa_pod_prop *prop;
struct spa_pod *value;
Uint32 n_values, choice;
struct spa_pod *value;
Uint32 n_values, choice;
prop = spa_pod_find_prop(param, NULL, key);
@@ -581,11 +559,10 @@ get_range_param(const struct spa_pod *param, Uint32 key, int *def, int *min, int
return SDL_FALSE;
}
static SDL_bool
get_int_param(const struct spa_pod *param, Uint32 key, int *val)
static SDL_bool get_int_param(const struct spa_pod *param, Uint32 key, int *val)
{
const struct spa_pod_prop *prop;
Sint32 v;
Sint32 v;
prop = spa_pod_find_prop(param, NULL, key);
@@ -601,13 +578,12 @@ get_int_param(const struct spa_pod *param, Uint32 key, int *val)
}
/* Interface node callbacks */
static void
node_event_info(void *object, const struct pw_node_info *info)
static void node_event_info(void *object, const struct pw_node_info *info)
{
struct node_object *node = object;
struct io_node *io = node->userdata;
const char *prop_val;
Uint32 i;
struct io_node *io = node->userdata;
const char *prop_val;
Uint32 i;
if (info) {
prop_val = spa_dict_lookup(info->props, PW_KEY_AUDIO_CHANNELS);
@@ -624,11 +600,10 @@ node_event_info(void *object, const struct pw_node_info *info)
}
}
static void
node_event_param(void *object, int seq, uint32_t id, uint32_t index, uint32_t next, const struct spa_pod *param)
static void node_event_param(void *object, int seq, uint32_t id, uint32_t index, uint32_t next, const struct spa_pod *param)
{
struct node_object *node = object;
struct io_node *io = node->userdata;
struct io_node *io = node->userdata;
/* Get the default frequency */
if (io->spec.freq == 0) {
@@ -650,8 +625,7 @@ node_event_param(void *object, int seq, uint32_t id, uint32_t index, uint32_t ne
static const struct pw_node_events interface_node_events = { PW_VERSION_NODE_EVENTS, .info = node_event_info,
.param = node_event_param };
static char*
get_name_from_json(const char *json)
static char *get_name_from_json(const char *json)
{
struct spa_json parser[2];
char key[7]; /* "name" */
@@ -673,8 +647,7 @@ get_name_from_json(const char *json)
}
/* Metadata node callback */
static int
metadata_property(void *object, Uint32 subject, const char *key, const char *type, const char *value)
static int metadata_property(void *object, Uint32 subject, const char *key, const char *type, const char *value)
{
struct node_object *node = object;
@@ -700,9 +673,8 @@ metadata_property(void *object, Uint32 subject, const char *key, const char *typ
static const struct pw_metadata_events metadata_node_events = { PW_VERSION_METADATA_EVENTS, .property = metadata_property };
/* Global registry callbacks */
static void
registry_event_global_callback(void *object, uint32_t id, uint32_t permissions, const char *type, uint32_t version,
const struct spa_dict *props)
static void registry_event_global_callback(void *object, uint32_t id, uint32_t permissions, const char *type, uint32_t version,
const struct spa_dict *props)
{
struct node_object *node;
@@ -711,12 +683,12 @@ registry_event_global_callback(void *object, uint32_t id, uint32_t permissions,
const char *media_class = spa_dict_lookup(props, PW_KEY_MEDIA_CLASS);
if (media_class) {
const char *node_desc;
const char *node_path;
const char *node_desc;
const char *node_path;
struct io_node *io;
SDL_bool is_capture;
int desc_buffer_len;
int path_buffer_len;
SDL_bool is_capture;
int desc_buffer_len;
int path_buffer_len;
/* Just want sink and capture */
if (!SDL_strcasecmp(media_class, "Audio/Sink")) {
@@ -748,11 +720,11 @@ registry_event_global_callback(void *object, uint32_t id, uint32_t permissions,
}
/* Begin setting the node properties */
io->id = id;
io->is_capture = is_capture;
io->id = id;
io->is_capture = is_capture;
io->spec.format = AUDIO_F32; /* Pipewire uses floats internally, other formats require conversion. */
io->name = io->buf;
io->path = io->buf + desc_buffer_len;
io->name = io->buf;
io->path = io->buf + desc_buffer_len;
SDL_strlcpy(io->buf, node_desc, desc_buffer_len);
SDL_strlcpy(io->buf + desc_buffer_len, node_path, path_buffer_len);
@@ -772,8 +744,7 @@ registry_event_global_callback(void *object, uint32_t id, uint32_t permissions,
}
}
static void
registry_event_remove_callback(void *object, uint32_t id)
static void registry_event_remove_callback(void *object, uint32_t id)
{
io_list_remove(id);
pending_list_remove(id);
@@ -783,8 +754,7 @@ static const struct pw_registry_events registry_events = { PW_VERSION_REGISTRY_E
.global_remove = registry_event_remove_callback };
/* The hotplug thread */
static int
hotplug_loop_init()
static int hotplug_loop_init()
{
int res;
@@ -827,8 +797,7 @@ hotplug_loop_init()
return 0;
}
static void
hotplug_loop_destroy()
static void hotplug_loop_destroy()
{
if (hotplug_loop) {
PIPEWIRE_pw_thread_loop_stop(hotplug_loop);
@@ -837,7 +806,7 @@ hotplug_loop_destroy()
pending_list_clear();
io_list_clear();
hotplug_init_complete = SDL_FALSE;
hotplug_init_complete = SDL_FALSE;
hotplug_events_enabled = SDL_FALSE;
if (pipewire_default_sink_id != NULL) {
@@ -870,8 +839,7 @@ hotplug_loop_destroy()
}
}
static void
PIPEWIRE_DetectDevices()
static void PIPEWIRE_DetectDevices()
{
struct io_node *io;
@@ -913,11 +881,10 @@ static const enum spa_audio_channel PIPEWIRE_channel_map_8[] = { SPA_AUDIO_CHANN
#define COPY_CHANNEL_MAP(c) SDL_memcpy(info->position, PIPEWIRE_channel_map_##c, sizeof(PIPEWIRE_channel_map_##c))
static void
initialize_spa_info(const SDL_AudioSpec *spec, struct spa_audio_info_raw *info)
static void initialize_spa_info(const SDL_AudioSpec *spec, struct spa_audio_info_raw *info)
{
info->channels = spec->channels;
info->rate = spec->freq;
info->rate = spec->freq;
switch (spec->channels) {
case 1:
@@ -981,14 +948,13 @@ initialize_spa_info(const SDL_AudioSpec *spec, struct spa_audio_info_raw *info)
}
}
static void
output_callback(void *data)
static void output_callback(void *data)
{
struct pw_buffer *pw_buf;
struct pw_buffer *pw_buf;
struct spa_buffer *spa_buf;
Uint8 *dst;
Uint8 *dst;
_THIS = (SDL_AudioDevice *)data;
_THIS = (SDL_AudioDevice *)data;
struct pw_stream *stream = this->hidden->stream;
/* Shutting down, don't do anything */
@@ -1042,18 +1008,17 @@ output_callback(void *data)
spa_buf->datas[0].chunk->offset = 0;
spa_buf->datas[0].chunk->stride = this->hidden->stride;
spa_buf->datas[0].chunk->size = this->spec.size;
spa_buf->datas[0].chunk->size = this->spec.size;
PIPEWIRE_pw_stream_queue_buffer(stream, pw_buf);
}
static void
input_callback(void *data)
static void input_callback(void *data)
{
struct pw_buffer *pw_buf;
struct pw_buffer *pw_buf;
struct spa_buffer *spa_buf;
Uint8 *src;
_THIS = (SDL_AudioDevice *)data;
Uint8 *src;
_THIS = (SDL_AudioDevice *)data;
struct pw_stream *stream = this->hidden->stream;
/* Shutting down, don't do anything */
@@ -1075,7 +1040,7 @@ input_callback(void *data)
if (!SDL_AtomicGet(&this->paused)) {
/* Calculate the offset and data size */
const Uint32 offset = SPA_MIN(spa_buf->datas[0].chunk->offset, spa_buf->datas[0].maxsize);
const Uint32 size = SPA_MIN(spa_buf->datas[0].chunk->size, spa_buf->datas[0].maxsize - offset);
const Uint32 size = SPA_MIN(spa_buf->datas[0].chunk->size, spa_buf->datas[0].maxsize - offset);
src += offset;
@@ -1103,8 +1068,7 @@ input_callback(void *data)
PIPEWIRE_pw_stream_queue_buffer(stream, pw_buf);
}
static void
stream_add_buffer_callback(void *data, struct pw_buffer *buffer)
static void stream_add_buffer_callback(void *data, struct pw_buffer *buffer)
{
_THIS = data;
@@ -1115,7 +1079,7 @@ stream_add_buffer_callback(void *data, struct pw_buffer *buffer)
*/
if (this->spec.size > buffer->buffer->datas[0].maxsize) {
this->spec.samples = buffer->buffer->datas[0].maxsize / this->hidden->stride;
this->spec.size = buffer->buffer->datas[0].maxsize;
this->spec.size = buffer->buffer->datas[0].maxsize;
}
} else if (this->hidden->buffer == NULL) {
/*
@@ -1127,15 +1091,14 @@ stream_add_buffer_callback(void *data, struct pw_buffer *buffer)
* A packet size of 2 periods should be more than is ever needed.
*/
this->hidden->input_buffer_packet_size = SPA_MAX(this->spec.size, buffer->buffer->datas[0].maxsize) * 2;
this->hidden->buffer = SDL_NewDataQueue(this->hidden->input_buffer_packet_size, this->hidden->input_buffer_packet_size);
this->hidden->buffer = SDL_NewDataQueue(this->hidden->input_buffer_packet_size, this->hidden->input_buffer_packet_size);
}
this->hidden->stream_init_status |= PW_READY_FLAG_BUFFER_ADDED;
PIPEWIRE_pw_thread_loop_signal(this->hidden->loop, false);
}
static void
stream_state_changed_callback(void *data, enum pw_stream_state old, enum pw_stream_state state, const char *error)
static void stream_state_changed_callback(void *data, enum pw_stream_state old, enum pw_stream_state state, const char *error)
{
_THIS = data;
@@ -1150,15 +1113,14 @@ stream_state_changed_callback(void *data, enum pw_stream_state old, enum pw_stre
static const struct pw_stream_events stream_output_events = { PW_VERSION_STREAM_EVENTS,
.state_changed = stream_state_changed_callback,
.add_buffer = stream_add_buffer_callback,
.process = output_callback };
static const struct pw_stream_events stream_input_events = { PW_VERSION_STREAM_EVENTS,
.state_changed = stream_state_changed_callback,
.add_buffer = stream_add_buffer_callback,
.process = input_callback };
.add_buffer = stream_add_buffer_callback,
.process = output_callback };
static const struct pw_stream_events stream_input_events = { PW_VERSION_STREAM_EVENTS,
.state_changed = stream_state_changed_callback,
.add_buffer = stream_add_buffer_callback,
.process = input_callback };
static int
PIPEWIRE_OpenDevice(_THIS, const char *devname)
static int PIPEWIRE_OpenDevice(_THIS, const char *devname)
{
/*
* NOTE: The PW_STREAM_FLAG_RT_PROCESS flag can be set to call the stream
@@ -1169,17 +1131,17 @@ PIPEWIRE_OpenDevice(_THIS, const char *devname)
*/
static const enum pw_stream_flags STREAM_FLAGS = PW_STREAM_FLAG_AUTOCONNECT | PW_STREAM_FLAG_MAP_BUFFERS;
char thread_name[PW_THREAD_NAME_BUFFER_LENGTH];
Uint8 pod_buffer[PW_POD_BUFFER_LENGTH];
struct spa_pod_builder b = SPA_POD_BUILDER_INIT(pod_buffer, sizeof(pod_buffer));
struct spa_audio_info_raw spa_info = { 0 };
const struct spa_pod *params = NULL;
char thread_name[PW_THREAD_NAME_BUFFER_LENGTH];
Uint8 pod_buffer[PW_POD_BUFFER_LENGTH];
struct spa_pod_builder b = SPA_POD_BUILDER_INIT(pod_buffer, sizeof(pod_buffer));
struct spa_audio_info_raw spa_info = { 0 };
const struct spa_pod *params = NULL;
struct SDL_PrivateAudioData *priv;
struct pw_properties *props;
const char *app_name, *stream_name, *stream_role, *error;
Uint32 node_id = this->handle == NULL ? PW_ID_ANY : PW_HANDLE_TO_ID(this->handle);
SDL_bool iscapture = this->iscapture;
int res;
struct pw_properties *props;
const char *app_name, *stream_name, *stream_role, *error;
Uint32 node_id = this->handle == NULL ? PW_ID_ANY : PW_HANDLE_TO_ID(this->handle);
SDL_bool iscapture = this->iscapture;
int res;
/* Clamp the period size to sane values */
const int min_period = PW_MIN_SAMPLES * SPA_MAX(this->spec.freq / PW_BASE_CLOCK_RATE, 1);
@@ -1223,7 +1185,7 @@ PIPEWIRE_OpenDevice(_THIS, const char *devname)
if (this->spec.samples < min_period) {
this->spec.samples = min_period;
this->spec.size = this->spec.samples * priv->stride;
this->spec.size = this->spec.samples * priv->stride;
}
SDL_snprintf(thread_name, sizeof(thread_name), "SDLAudio%c%ld", (iscapture) ? 'C' : 'P', (long)this->handle);
@@ -1341,8 +1303,7 @@ static void PIPEWIRE_CloseDevice(_THIS)
SDL_free(this->hidden);
}
static int
PIPEWIRE_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
static int PIPEWIRE_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
{
struct io_node *node;
char *target;
@@ -1380,8 +1341,7 @@ failed:
return ret;
}
static void
PIPEWIRE_Deinitialize()
static void PIPEWIRE_Deinitialize()
{
if (pipewire_initialized) {
hotplug_loop_destroy();
@@ -1390,8 +1350,7 @@ PIPEWIRE_Deinitialize()
}
}
static SDL_bool
PIPEWIRE_Init(SDL_AudioDriverImpl *impl)
static SDL_bool PIPEWIRE_Init(SDL_AudioDriverImpl *impl)
{
if (!pipewire_initialized) {
if (init_pipewire_library() < 0) {
@@ -1407,15 +1366,15 @@ PIPEWIRE_Init(SDL_AudioDriverImpl *impl)
}
/* Set the function pointers */
impl->DetectDevices = PIPEWIRE_DetectDevices;
impl->OpenDevice = PIPEWIRE_OpenDevice;
impl->CloseDevice = PIPEWIRE_CloseDevice;
impl->Deinitialize = PIPEWIRE_Deinitialize;
impl->DetectDevices = PIPEWIRE_DetectDevices;
impl->OpenDevice = PIPEWIRE_OpenDevice;
impl->CloseDevice = PIPEWIRE_CloseDevice;
impl->Deinitialize = PIPEWIRE_Deinitialize;
impl->GetDefaultAudioInfo = PIPEWIRE_GetDefaultAudioInfo;
impl->HasCaptureSupport = SDL_TRUE;
impl->HasCaptureSupport = SDL_TRUE;
impl->ProvidesOwnCallbackThread = SDL_TRUE;
impl->SupportsNonPow2Samples = SDL_TRUE;
impl->SupportsNonPow2Samples = SDL_TRUE;
return SDL_TRUE;
}

8
src/audio/pipewire/SDL_pipewire.h
View File

@@ -33,13 +33,13 @@
struct SDL_PrivateAudioData
{
struct pw_thread_loop *loop;
struct pw_stream *stream;
struct pw_context *context;
struct SDL_DataQueue *buffer;
struct pw_stream *stream;
struct pw_context *context;
struct SDL_DataQueue *buffer;
size_t input_buffer_packet_size;
Sint32 stride; /* Bytes-per-frame */
int stream_init_status;
int stream_init_status;
};
#endif /* SDL_pipewire_h_ */

40
src/audio/ps2/SDL_ps2audio.c
View File

@@ -31,10 +31,9 @@
#include <ps2_audio_driver.h>
/* The tag name used by PS2 audio */
#define PS2AUDIO_DRIVER_NAME "ps2"
#define PS2AUDIO_DRIVER_NAME "ps2"
static int
PS2AUDIO_OpenDevice(_THIS, const char *devname)
static int PS2AUDIO_OpenDevice(_THIS, const char *devname)
{
int i, mixlen;
struct audsrv_fmt_t format;
@@ -46,21 +45,20 @@ PS2AUDIO_OpenDevice(_THIS, const char *devname)
}
SDL_zerop(this->hidden);
/* These are the native supported audio PS2 configs */
switch (this->spec.freq) {
case 11025:
case 12000:
case 22050:
case 24000:
case 32000:
case 44100:
case 48000:
this->spec.freq = this->spec.freq;
break;
default:
this->spec.freq = 48000;
break;
case 11025:
case 12000:
case 22050:
case 24000:
case 32000:
case 44100:
case 48000:
this->spec.freq = this->spec.freq;
break;
default:
this->spec.freq = 48000;
break;
}
this->spec.samples = 512;
@@ -69,8 +67,8 @@ PS2AUDIO_OpenDevice(_THIS, const char *devname)
SDL_CalculateAudioSpec(&this->spec);
format.bits = this->spec.format == AUDIO_S8 ? 8 : 16;
format.freq = this->spec.freq;
format.bits = this->spec.format == AUDIO_S8 ? 8 : 16;
format.freq = this->spec.freq;
format.channels = this->spec.channels;
this->hidden->channel = audsrv_set_format(&format);
@@ -89,7 +87,7 @@ PS2AUDIO_OpenDevice(_THIS, const char *devname)
be a multiple of 64 bytes. Our sample count is already a multiple of
64, so spec->size should be a multiple of 64 as well. */
mixlen = this->spec.size * NUM_BUFFERS;
this->hidden->rawbuf = (Uint8 *) memalign(64, mixlen);
this->hidden->rawbuf = (Uint8 *)memalign(64, mixlen);
if (this->hidden->rawbuf == NULL) {
return SDL_SetError("Couldn't allocate mixing buffer");
}
@@ -152,7 +150,7 @@ static void PS2AUDIO_Deinitialize(void)
deinit_audio_driver();
}
static SDL_bool PS2AUDIO_Init(SDL_AudioDriverImpl * impl)
static SDL_bool PS2AUDIO_Init(SDL_AudioDriverImpl *impl)
{
if (init_audio_driver() < 0) {
return SDL_FALSE;
@@ -167,7 +165,7 @@ static SDL_bool PS2AUDIO_Init(SDL_AudioDriverImpl * impl)
impl->ThreadInit = PS2AUDIO_ThreadInit;
impl->Deinitialize = PS2AUDIO_Deinitialize;
impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE; /* this audio target is available. */
}
AudioBootStrap PS2AUDIO_bootstrap = {

18
src/audio/ps2/SDL_ps2audio.h
View File

@@ -26,20 +26,20 @@
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define _THIS SDL_AudioDevice *this
#define NUM_BUFFERS 2
struct SDL_PrivateAudioData
{
/* The hardware output channel. */
int channel;
/* The raw allocated mixing buffer. */
Uint8 *rawbuf;
/* Individual mixing buffers. */
Uint8 *mixbufs[NUM_BUFFERS];
/* Index of the next available mixing buffer. */
int next_buffer;
/* The hardware output channel. */
int channel;
/* The raw allocated mixing buffer. */
Uint8 *rawbuf;
/* Individual mixing buffers. */
Uint8 *mixbufs[NUM_BUFFERS];
/* Index of the next available mixing buffer. */
int next_buffer;
};
#endif /* SDL_ps2audio_h_ */

16
src/audio/psp/SDL_pspaudio.c
View File

@@ -36,10 +36,9 @@
#include <pspthreadman.h>
/* The tag name used by PSP audio */
#define PSPAUDIO_DRIVER_NAME "psp"
#define PSPAUDIO_DRIVER_NAME "psp"
static int
PSPAUDIO_OpenDevice(_THIS, const char *devname)
static int PSPAUDIO_OpenDevice(_THIS, const char *devname)
{
int format, mixlen, i;
@@ -61,7 +60,7 @@ PSPAUDIO_OpenDevice(_THIS, const char *devname)
format = PSP_AUDIO_FORMAT_MONO;
} else {
format = PSP_AUDIO_FORMAT_STEREO;
this->spec.channels = 2; /* we're forcing the hardware to stereo. */
this->spec.channels = 2; /* we're forcing the hardware to stereo. */
}
/* PSP has some limitations with the Audio. It fully supports 44.1KHz (Mono & Stereo),
@@ -72,7 +71,7 @@ PSPAUDIO_OpenDevice(_THIS, const char *devname)
} else {
this->hidden->channel = sceAudioSRCChReserve(this->spec.samples, this->spec.freq, 2);
}
if (this->hidden->channel < 0) {
free(this->hidden->rawbuf);
this->hidden->rawbuf = NULL;
@@ -86,7 +85,7 @@ PSPAUDIO_OpenDevice(_THIS, const char *devname)
be a multiple of 64 bytes. Our sample count is already a multiple of
64, so spec->size should be a multiple of 64 as well. */
mixlen = this->spec.size * NUM_BUFFERS;
this->hidden->rawbuf = (Uint8 *) memalign(64, mixlen);
this->hidden->rawbuf = (Uint8 *)memalign(64, mixlen);
if (this->hidden->rawbuf == NULL) {
return SDL_SetError("Couldn't allocate mixing buffer");
}
@@ -155,8 +154,7 @@ static void PSPAUDIO_ThreadInit(_THIS)
}
}
static SDL_bool
PSPAUDIO_Init(SDL_AudioDriverImpl * impl)
static SDL_bool PSPAUDIO_Init(SDL_AudioDriverImpl *impl)
{
/* Set the function pointers */
impl->OpenDevice = PSPAUDIO_OpenDevice;
@@ -172,7 +170,7 @@ PSPAUDIO_Init(SDL_AudioDriverImpl * impl)
impl->HasCaptureSupport = SDL_TRUE;
impl->OnlyHasDefaultCaptureDevice = SDL_TRUE;
*/
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE; /* this audio target is available. */
}
AudioBootStrap PSPAUDIO_bootstrap = {

13
src/audio/psp/SDL_pspaudio.h
View File

@@ -25,19 +25,20 @@
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define _THIS SDL_AudioDevice *this
#define NUM_BUFFERS 2
struct SDL_PrivateAudioData {
struct SDL_PrivateAudioData
{
/* The hardware output channel. */
int channel;
int channel;
/* The raw allocated mixing buffer. */
Uint8 *rawbuf;
Uint8 *rawbuf;
/* Individual mixing buffers. */
Uint8 *mixbufs[NUM_BUFFERS];
Uint8 *mixbufs[NUM_BUFFERS];
/* Index of the next available mixing buffer. */
int next_buffer;
int next_buffer;
};
#endif /* SDL_pspaudio_h_ */

284
src/audio/pulseaudio/SDL_pulseaudio.c
View File

@@ -45,81 +45,79 @@
/* should we include monitors in the device list? Set at SDL_Init time */
static SDL_bool include_monitors = SDL_FALSE;
#if (PA_API_VERSION < 12)
/** Return non-zero if the passed state is one of the connected states */
static SDL_INLINE int PA_CONTEXT_IS_GOOD(pa_context_state_t x) {
static SDL_INLINE int PA_CONTEXT_IS_GOOD(pa_context_state_t x)
{
return x == PA_CONTEXT_CONNECTING || x == PA_CONTEXT_AUTHORIZING || x == PA_CONTEXT_SETTING_NAME || x == PA_CONTEXT_READY;
}
/** Return non-zero if the passed state is one of the connected states */
static SDL_INLINE int PA_STREAM_IS_GOOD(pa_stream_state_t x) {
static SDL_INLINE int PA_STREAM_IS_GOOD(pa_stream_state_t x)
{
return x == PA_STREAM_CREATING || x == PA_STREAM_READY;
}
#endif /* pulseaudio <= 0.9.10 */
static const char *(*PULSEAUDIO_pa_get_library_version) (void);
static pa_channel_map *(*PULSEAUDIO_pa_channel_map_init_auto) (
static const char *(*PULSEAUDIO_pa_get_library_version)(void);
static pa_channel_map *(*PULSEAUDIO_pa_channel_map_init_auto)(
pa_channel_map *, unsigned, pa_channel_map_def_t);
static const char * (*PULSEAUDIO_pa_strerror) (int);
static pa_mainloop * (*PULSEAUDIO_pa_mainloop_new) (void);
static pa_mainloop_api * (*PULSEAUDIO_pa_mainloop_get_api) (pa_mainloop *);
static int (*PULSEAUDIO_pa_mainloop_iterate) (pa_mainloop *, int, int *);
static int (*PULSEAUDIO_pa_mainloop_run) (pa_mainloop *, int *);
static void (*PULSEAUDIO_pa_mainloop_quit) (pa_mainloop *, int);
static void (*PULSEAUDIO_pa_mainloop_free) (pa_mainloop *);
static const char *(*PULSEAUDIO_pa_strerror)(int);
static pa_mainloop *(*PULSEAUDIO_pa_mainloop_new)(void);
static pa_mainloop_api *(*PULSEAUDIO_pa_mainloop_get_api)(pa_mainloop *);
static int (*PULSEAUDIO_pa_mainloop_iterate)(pa_mainloop *, int, int *);
static int (*PULSEAUDIO_pa_mainloop_run)(pa_mainloop *, int *);
static void (*PULSEAUDIO_pa_mainloop_quit)(pa_mainloop *, int);
static void (*PULSEAUDIO_pa_mainloop_free)(pa_mainloop *);
static pa_operation_state_t (*PULSEAUDIO_pa_operation_get_state) (
static pa_operation_state_t (*PULSEAUDIO_pa_operation_get_state)(
const pa_operation *);
static void (*PULSEAUDIO_pa_operation_cancel) (pa_operation *);
static void (*PULSEAUDIO_pa_operation_unref) (pa_operation *);
static void (*PULSEAUDIO_pa_operation_cancel)(pa_operation *);
static void (*PULSEAUDIO_pa_operation_unref)(pa_operation *);
static pa_context * (*PULSEAUDIO_pa_context_new) (pa_mainloop_api *,
const char *);
static int (*PULSEAUDIO_pa_context_connect) (pa_context *, const char *,
pa_context_flags_t, const pa_spawn_api *);
static pa_operation * (*PULSEAUDIO_pa_context_get_sink_info_list) (pa_context *, pa_sink_info_cb_t, void *);
static pa_operation * (*PULSEAUDIO_pa_context_get_source_info_list) (pa_context *, pa_source_info_cb_t, void *);
static pa_operation * (*PULSEAUDIO_pa_context_get_sink_info_by_index) (pa_context *, uint32_t, pa_sink_info_cb_t, void *);
static pa_operation * (*PULSEAUDIO_pa_context_get_source_info_by_index) (pa_context *, uint32_t, pa_source_info_cb_t, void *);
static pa_context_state_t (*PULSEAUDIO_pa_context_get_state) (const pa_context *);
static pa_operation * (*PULSEAUDIO_pa_context_subscribe) (pa_context *, pa_subscription_mask_t, pa_context_success_cb_t, void *);
static void (*PULSEAUDIO_pa_context_set_subscribe_callback) (pa_context *, pa_context_subscribe_cb_t, void *);
static void (*PULSEAUDIO_pa_context_disconnect) (pa_context *);
static void (*PULSEAUDIO_pa_context_unref) (pa_context *);
static pa_context *(*PULSEAUDIO_pa_context_new)(pa_mainloop_api *,
const char *);
static int (*PULSEAUDIO_pa_context_connect)(pa_context *, const char *,
pa_context_flags_t, const pa_spawn_api *);
static pa_operation *(*PULSEAUDIO_pa_context_get_sink_info_list)(pa_context *, pa_sink_info_cb_t, void *);
static pa_operation *(*PULSEAUDIO_pa_context_get_source_info_list)(pa_context *, pa_source_info_cb_t, void *);
static pa_operation *(*PULSEAUDIO_pa_context_get_sink_info_by_index)(pa_context *, uint32_t, pa_sink_info_cb_t, void *);
static pa_operation *(*PULSEAUDIO_pa_context_get_source_info_by_index)(pa_context *, uint32_t, pa_source_info_cb_t, void *);
static pa_context_state_t (*PULSEAUDIO_pa_context_get_state)(const pa_context *);
static pa_operation *(*PULSEAUDIO_pa_context_subscribe)(pa_context *, pa_subscription_mask_t, pa_context_success_cb_t, void *);
static void (*PULSEAUDIO_pa_context_set_subscribe_callback)(pa_context *, pa_context_subscribe_cb_t, void *);
static void (*PULSEAUDIO_pa_context_disconnect)(pa_context *);
static void (*PULSEAUDIO_pa_context_unref)(pa_context *);
static pa_stream * (*PULSEAUDIO_pa_stream_new) (pa_context *, const char *,
const pa_sample_spec *, const pa_channel_map *);
static int (*PULSEAUDIO_pa_stream_connect_playback) (pa_stream *, const char *,
const pa_buffer_attr *, pa_stream_flags_t, const pa_cvolume *, pa_stream *);
static int (*PULSEAUDIO_pa_stream_connect_record) (pa_stream *, const char *,
const pa_buffer_attr *, pa_stream_flags_t);
static pa_stream_state_t (*PULSEAUDIO_pa_stream_get_state) (const pa_stream *);
static size_t (*PULSEAUDIO_pa_stream_writable_size) (const pa_stream *);
static size_t (*PULSEAUDIO_pa_stream_readable_size) (const pa_stream *);
static int (*PULSEAUDIO_pa_stream_write) (pa_stream *, const void *, size_t,
pa_free_cb_t, int64_t, pa_seek_mode_t);
static pa_operation * (*PULSEAUDIO_pa_stream_drain) (pa_stream *,
pa_stream_success_cb_t, void *);
static int (*PULSEAUDIO_pa_stream_peek) (pa_stream *, const void **, size_t *);
static int (*PULSEAUDIO_pa_stream_drop) (pa_stream *);
static pa_operation * (*PULSEAUDIO_pa_stream_flush) (pa_stream *,
pa_stream_success_cb_t, void *);
static int (*PULSEAUDIO_pa_stream_disconnect) (pa_stream *);
static void (*PULSEAUDIO_pa_stream_unref) (pa_stream *);
static pa_stream *(*PULSEAUDIO_pa_stream_new)(pa_context *, const char *,
const pa_sample_spec *, const pa_channel_map *);
static int (*PULSEAUDIO_pa_stream_connect_playback)(pa_stream *, const char *,
const pa_buffer_attr *, pa_stream_flags_t, const pa_cvolume *, pa_stream *);
static int (*PULSEAUDIO_pa_stream_connect_record)(pa_stream *, const char *,
const pa_buffer_attr *, pa_stream_flags_t);
static pa_stream_state_t (*PULSEAUDIO_pa_stream_get_state)(const pa_stream *);
static size_t (*PULSEAUDIO_pa_stream_writable_size)(const pa_stream *);
static size_t (*PULSEAUDIO_pa_stream_readable_size)(const pa_stream *);
static int (*PULSEAUDIO_pa_stream_write)(pa_stream *, const void *, size_t,
pa_free_cb_t, int64_t, pa_seek_mode_t);
static pa_operation *(*PULSEAUDIO_pa_stream_drain)(pa_stream *,
pa_stream_success_cb_t, void *);
static int (*PULSEAUDIO_pa_stream_peek)(pa_stream *, const void **, size_t *);
static int (*PULSEAUDIO_pa_stream_drop)(pa_stream *);
static pa_operation *(*PULSEAUDIO_pa_stream_flush)(pa_stream *,
pa_stream_success_cb_t, void *);
static int (*PULSEAUDIO_pa_stream_disconnect)(pa_stream *);
static void (*PULSEAUDIO_pa_stream_unref)(pa_stream *);
static void (*PULSEAUDIO_pa_stream_set_write_callback)(pa_stream *, pa_stream_request_cb_t, void *);
static pa_operation * (*PULSEAUDIO_pa_context_get_server_info)(pa_context *, pa_server_info_cb_t, void *);
static pa_operation *(*PULSEAUDIO_pa_context_get_server_info)(pa_context *, pa_server_info_cb_t, void *);
static int load_pulseaudio_syms(void);
#ifdef SDL_AUDIO_DRIVER_PULSEAUDIO_DYNAMIC
static const char *pulseaudio_library = SDL_AUDIO_DRIVER_PULSEAUDIO_DYNAMIC;
static void *pulseaudio_handle = NULL;
static int
load_pulseaudio_sym(const char *fn, void **addr)
static int load_pulseaudio_sym(const char *fn, void **addr)
{
*addr = SDL_LoadFunction(pulseaudio_handle, fn);
if (*addr == NULL) {
@@ -131,11 +129,11 @@ load_pulseaudio_sym(const char *fn, void **addr)
}
/* cast funcs to char* first, to please GCC's strict aliasing rules. */
#define SDL_PULSEAUDIO_SYM(x) \
if (!load_pulseaudio_sym(#x, (void **) (char *) &PULSEAUDIO_##x)) return -1
#define SDL_PULSEAUDIO_SYM(x) \
if (!load_pulseaudio_sym(#x, (void **)(char *)&PULSEAUDIO_##x)) \
return -1
static void
UnloadPulseAudioLibrary(void)
static void UnloadPulseAudioLibrary(void)
{
if (pulseaudio_handle != NULL) {
SDL_UnloadObject(pulseaudio_handle);
@@ -143,8 +141,7 @@ UnloadPulseAudioLibrary(void)
}
}
static int
LoadPulseAudioLibrary(void)
static int LoadPulseAudioLibrary(void)
{
int retval = 0;
if (pulseaudio_handle == NULL) {
@@ -166,13 +163,11 @@ LoadPulseAudioLibrary(void)
#define SDL_PULSEAUDIO_SYM(x) PULSEAUDIO_##x = x
static void
UnloadPulseAudioLibrary(void)
static void UnloadPulseAudioLibrary(void)
{
}
static int
LoadPulseAudioLibrary(void)
static int LoadPulseAudioLibrary(void)
{
load_pulseaudio_syms();
return 0;
@@ -180,9 +175,7 @@ LoadPulseAudioLibrary(void)
#endif /* SDL_AUDIO_DRIVER_PULSEAUDIO_DYNAMIC */
static int
load_pulseaudio_syms(void)
static int load_pulseaudio_syms(void)
{
SDL_PULSEAUDIO_SYM(pa_get_library_version);
SDL_PULSEAUDIO_SYM(pa_mainloop_new);
@@ -225,15 +218,13 @@ load_pulseaudio_syms(void)
return 0;
}
static SDL_INLINE int
squashVersion(const int major, const int minor, const int patch)
static SDL_INLINE int squashVersion(const int major, const int minor, const int patch)
{
return ((major & 0xFF) << 16) | ((minor & 0xFF) << 8) | (patch & 0xFF);
}
/* Workaround for older pulse: pa_context_new() must have non-NULL appname */
static const char *
getAppName(void)
static const char *getAppName(void)
{
const char *retval = SDL_GetHint(SDL_HINT_AUDIO_DEVICE_APP_NAME);
if (retval && *retval) {
@@ -244,12 +235,12 @@ getAppName(void)
return retval;
} else {
const char *verstr = PULSEAUDIO_pa_get_library_version();
retval = "SDL Application"; /* the "oh well" default. */
retval = "SDL Application"; /* the "oh well" default. */
if (verstr != NULL) {
int maj, min, patch;
if (SDL_sscanf(verstr, "%d.%d.%d", &maj, &min, &patch) == 3) {
if (squashVersion(maj, min, patch) >= squashVersion(0, 9, 15)) {
retval = NULL; /* 0.9.15+ handles NULL correctly. */
retval = NULL; /* 0.9.15+ handles NULL correctly. */
}
}
}
@@ -257,8 +248,7 @@ getAppName(void)
return retval;
}
static void
WaitForPulseOperation(pa_mainloop *mainloop, pa_operation *o)
static void WaitForPulseOperation(pa_mainloop *mainloop, pa_operation *o)
{
/* This checks for NO errors currently. Either fix that, check results elsewhere, or do things you don't care about. */
if (mainloop && o) {
@@ -270,8 +260,7 @@ WaitForPulseOperation(pa_mainloop *mainloop, pa_operation *o)
}
}
static void
DisconnectFromPulseServer(pa_mainloop *mainloop, pa_context *context)
static void DisconnectFromPulseServer(pa_mainloop *mainloop, pa_context *context)
{
if (context) {
PULSEAUDIO_pa_context_disconnect(context);
@@ -282,8 +271,7 @@ DisconnectFromPulseServer(pa_mainloop *mainloop, pa_context *context)
}
}
static int
ConnectToPulseServer_Internal(pa_mainloop **_mainloop, pa_context **_context)
static int ConnectToPulseServer_Internal(pa_mainloop **_mainloop, pa_context **_context)
{
pa_mainloop *mainloop = NULL;
pa_context *context = NULL;
@@ -299,7 +287,7 @@ ConnectToPulseServer_Internal(pa_mainloop **_mainloop, pa_context **_context)
}
mainloop_api = PULSEAUDIO_pa_mainloop_get_api(mainloop);
SDL_assert(mainloop_api); /* this never fails, right? */
SDL_assert(mainloop_api); /* this never fails, right? */
context = PULSEAUDIO_pa_context_new(mainloop_api, getAppName());
if (context == NULL) {
@@ -331,11 +319,10 @@ ConnectToPulseServer_Internal(pa_mainloop **_mainloop, pa_context **_context)
*_context = context;
*_mainloop = mainloop;
return 0; /* connected and ready! */
return 0; /* connected and ready! */
}
static int
ConnectToPulseServer(pa_mainloop **_mainloop, pa_context **_context)
static int ConnectToPulseServer(pa_mainloop **_mainloop, pa_context **_context)
{
const int retval = ConnectToPulseServer_Internal(_mainloop, _context);
if (retval < 0) {
@@ -344,23 +331,20 @@ ConnectToPulseServer(pa_mainloop **_mainloop, pa_context **_context)
return retval;
}
/* This function waits until it is possible to write a full sound buffer */
static void
PULSEAUDIO_WaitDevice(_THIS)
static void PULSEAUDIO_WaitDevice(_THIS)
{
/* this is a no-op; we wait in PULSEAUDIO_PlayDevice now. */
}
static void WriteCallback(pa_stream *p, size_t nbytes, void *userdata)
{
struct SDL_PrivateAudioData *h = (struct SDL_PrivateAudioData *) userdata;
struct SDL_PrivateAudioData *h = (struct SDL_PrivateAudioData *)userdata;
/*printf("PULSEAUDIO WRITE CALLBACK! nbytes=%u\n", (unsigned int) nbytes);*/
h->bytes_requested += nbytes;
}
static void
PULSEAUDIO_PlayDevice(_THIS)
static void PULSEAUDIO_PlayDevice(_THIS)
{
struct SDL_PrivateAudioData *h = this->hidden;
int available = h->mixlen;
@@ -395,15 +379,12 @@ PULSEAUDIO_PlayDevice(_THIS)
/*printf("PULSEAUDIO PLAYDEVICE END! written=%d\n", written);*/
}
static Uint8 *
PULSEAUDIO_GetDeviceBuf(_THIS)
static Uint8 *PULSEAUDIO_GetDeviceBuf(_THIS)
{
return this->hidden->mixbuf;
}
static int
PULSEAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
static int PULSEAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
struct SDL_PrivateAudioData *h = this->hidden;
const void *data = NULL;
@@ -418,40 +399,40 @@ PULSEAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
h->capturelen -= cpy;
if (h->capturelen == 0) {
h->capturebuf = NULL;
PULSEAUDIO_pa_stream_drop(h->stream); /* done with this fragment. */
PULSEAUDIO_pa_stream_drop(h->stream); /* done with this fragment. */
}
return cpy; /* new data, return it. */
return cpy; /* new data, return it. */
}
if (PULSEAUDIO_pa_context_get_state(h->context) != PA_CONTEXT_READY ||
PULSEAUDIO_pa_stream_get_state(h->stream) != PA_STREAM_READY ||
PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
SDL_OpenedAudioDeviceDisconnected(this);
return -1; /* uhoh, pulse failed! */
return -1; /* uhoh, pulse failed! */
}
if (PULSEAUDIO_pa_stream_readable_size(h->stream) == 0) {
continue; /* no data available yet. */
continue; /* no data available yet. */
}
/* a new fragment is available! */
PULSEAUDIO_pa_stream_peek(h->stream, &data, &nbytes);
SDL_assert(nbytes > 0);
if (data == NULL) { /* NULL==buffer had a hole. Ignore that. */
PULSEAUDIO_pa_stream_drop(h->stream); /* drop this fragment. */
/* If data == NULL, then the buffer had a hole, ignore that */
if (data == NULL) {
PULSEAUDIO_pa_stream_drop(h->stream); /* drop this fragment. */
} else {
/* store this fragment's data, start feeding it to SDL. */
/*printf("PULSEAUDIO: captured %d new bytes\n", (int) nbytes);*/
h->capturebuf = (const Uint8 *) data;
h->capturebuf = (const Uint8 *)data;
h->capturelen = nbytes;
}
}
return -1; /* not enabled? */
return -1; /* not enabled? */
}
static void
PULSEAUDIO_FlushCapture(_THIS)
static void PULSEAUDIO_FlushCapture(_THIS)
{
struct SDL_PrivateAudioData *h = this->hidden;
const void *data = NULL;
@@ -468,21 +449,20 @@ PULSEAUDIO_FlushCapture(_THIS)
PULSEAUDIO_pa_stream_get_state(h->stream) != PA_STREAM_READY ||
PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
SDL_OpenedAudioDeviceDisconnected(this);
return; /* uhoh, pulse failed! */
return; /* uhoh, pulse failed! */
}
if (PULSEAUDIO_pa_stream_readable_size(h->stream) == 0) {
break; /* no data available, so we're done. */
break; /* no data available, so we're done. */
}
/* a new fragment is available! Just dump it. */
PULSEAUDIO_pa_stream_peek(h->stream, &data, &nbytes);
PULSEAUDIO_pa_stream_drop(h->stream); /* drop this fragment. */
PULSEAUDIO_pa_stream_drop(h->stream); /* drop this fragment. */
}
}
static void
PULSEAUDIO_CloseDevice(_THIS)
static void PULSEAUDIO_CloseDevice(_THIS)
{
if (this->hidden->stream) {
if (this->hidden->capturebuf != NULL) {
@@ -498,48 +478,44 @@ PULSEAUDIO_CloseDevice(_THIS)
SDL_free(this->hidden);
}
static void
SinkDeviceNameCallback(pa_context *c, const pa_sink_info *i, int is_last, void *data)
static void SinkDeviceNameCallback(pa_context *c, const pa_sink_info *i, int is_last, void *data)
{
if (i) {
char **devname = (char **) data;
char **devname = (char **)data;
*devname = SDL_strdup(i->name);
}
}
static void
SourceDeviceNameCallback(pa_context *c, const pa_source_info *i, int is_last, void *data)
static void SourceDeviceNameCallback(pa_context *c, const pa_source_info *i, int is_last, void *data)
{
if (i) {
char **devname = (char **) data;
char **devname = (char **)data;
*devname = SDL_strdup(i->name);
}
}
static SDL_bool
FindDeviceName(struct SDL_PrivateAudioData *h, const SDL_bool iscapture, void *handle)
static SDL_bool FindDeviceName(struct SDL_PrivateAudioData *h, const SDL_bool iscapture, void *handle)
{
const uint32_t idx = ((uint32_t) ((intptr_t) handle)) - 1;
const uint32_t idx = ((uint32_t)((intptr_t)handle)) - 1;
if (handle == NULL) { /* NULL == default device. */
if (handle == NULL) { /* NULL == default device. */
return SDL_TRUE;
}
if (iscapture) {
WaitForPulseOperation(h->mainloop,
PULSEAUDIO_pa_context_get_source_info_by_index(h->context, idx,
SourceDeviceNameCallback, &h->device_name));
PULSEAUDIO_pa_context_get_source_info_by_index(h->context, idx,
SourceDeviceNameCallback, &h->device_name));
} else {
WaitForPulseOperation(h->mainloop,
PULSEAUDIO_pa_context_get_sink_info_by_index(h->context, idx,
SinkDeviceNameCallback, &h->device_name));
PULSEAUDIO_pa_context_get_sink_info_by_index(h->context, idx,
SinkDeviceNameCallback, &h->device_name));
}
return h->device_name != NULL;
}
static int
PULSEAUDIO_OpenDevice(_THIS, const char *devname)
static int PULSEAUDIO_OpenDevice(_THIS, const char *devname)
{
struct SDL_PrivateAudioData *h = NULL;
SDL_AudioFormat test_format;
@@ -604,7 +580,7 @@ PULSEAUDIO_OpenDevice(_THIS, const char *devname)
/* Allocate mixing buffer */
if (!iscapture) {
h->mixlen = this->spec.size;
h->mixbuf = (Uint8 *) SDL_malloc(h->mixlen);
h->mixbuf = (Uint8 *)SDL_malloc(h->mixlen);
if (h->mixbuf == NULL) {
return SDL_OutOfMemory();
}
@@ -640,9 +616,9 @@ PULSEAUDIO_OpenDevice(_THIS, const char *devname)
h->stream = PULSEAUDIO_pa_stream_new(
h->context,
(name && *name) ? name : "Audio Stream", /* stream description */
&paspec, /* sample format spec */
&pacmap /* channel map */
);
&paspec, /* sample format spec */
&pacmap /* channel map */
);
if (h->stream == NULL) {
return SDL_SetError("Could not set up PulseAudio stream");
@@ -692,8 +668,7 @@ static char *default_source_name = NULL;
/* device handles are device index + 1, cast to void*, so we never pass a NULL. */
static SDL_AudioFormat
PulseFormatToSDLFormat(pa_sample_format_t format)
static SDL_AudioFormat PulseFormatToSDLFormat(pa_sample_format_t format)
{
switch (format) {
case PA_SAMPLE_U8:
@@ -716,11 +691,10 @@ PulseFormatToSDLFormat(pa_sample_format_t format)
}
/* This is called when PulseAudio adds an output ("sink") device. */
static void
SinkInfoCallback(pa_context *c, const pa_sink_info *i, int is_last, void *data)
static void SinkInfoCallback(pa_context *c, const pa_sink_info *i, int is_last, void *data)
{
SDL_AudioSpec spec;
SDL_bool add = (SDL_bool) ((intptr_t) data);
SDL_bool add = (SDL_bool)((intptr_t)data);
if (i) {
spec.freq = i->sample_spec.rate;
spec.channels = i->sample_spec.channels;
@@ -732,7 +706,7 @@ SinkInfoCallback(pa_context *c, const pa_sink_info *i, int is_last, void *data)
spec.userdata = NULL;
if (add) {
SDL_AddAudioDevice(SDL_FALSE, i->description, &spec, (void *) ((intptr_t) i->index+1));
SDL_AddAudioDevice(SDL_FALSE, i->description, &spec, (void *)((intptr_t)i->index + 1));
}
if (default_sink_path != NULL && SDL_strcmp(i->name, default_sink_path) == 0) {
@@ -745,11 +719,10 @@ SinkInfoCallback(pa_context *c, const pa_sink_info *i, int is_last, void *data)
}
/* This is called when PulseAudio adds a capture ("source") device. */
static void
SourceInfoCallback(pa_context *c, const pa_source_info *i, int is_last, void *data)
static void SourceInfoCallback(pa_context *c, const pa_source_info *i, int is_last, void *data)
{
SDL_AudioSpec spec;
SDL_bool add = (SDL_bool) ((intptr_t) data);
SDL_bool add = (SDL_bool)((intptr_t)data);
if (i) {
/* Maybe skip "monitor" sources. These are just output from other sinks. */
if (include_monitors || (i->monitor_of_sink == PA_INVALID_INDEX)) {
@@ -763,7 +736,7 @@ SourceInfoCallback(pa_context *c, const pa_source_info *i, int is_last, void *da
spec.userdata = NULL;
if (add) {
SDL_AddAudioDevice(SDL_TRUE, i->description, &spec, (void *) ((intptr_t) i->index+1));
SDL_AddAudioDevice(SDL_TRUE, i->description, &spec, (void *)((intptr_t)i->index + 1));
}
if (default_source_path != NULL && SDL_strcmp(i->name, default_source_path) == 0) {
@@ -776,8 +749,7 @@ SourceInfoCallback(pa_context *c, const pa_source_info *i, int is_last, void *da
}
}
static void
ServerInfoCallback(pa_context *c, const pa_server_info *i, void *data)
static void ServerInfoCallback(pa_context *c, const pa_server_info *i, void *data)
{
if (default_sink_path != NULL) {
SDL_free(default_sink_path);
@@ -790,14 +762,13 @@ ServerInfoCallback(pa_context *c, const pa_server_info *i, void *data)
}
/* This is called when PulseAudio has a device connected/removed/changed. */
static void
HotplugCallback(pa_context *c, pa_subscription_event_type_t t, uint32_t idx, void *data)
static void HotplugCallback(pa_context *c, pa_subscription_event_type_t t, uint32_t idx, void *data)
{
const SDL_bool added = ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_NEW);
const SDL_bool removed = ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_REMOVE);
const SDL_bool changed = ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_CHANGE);
if (added || removed || changed) { /* we only care about add/remove events. */
if (added || removed || changed) { /* we only care about add/remove events. */
const SDL_bool sink = ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SINK);
const SDL_bool source = ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SOURCE);
@@ -806,45 +777,42 @@ HotplugCallback(pa_context *c, pa_subscription_event_type_t t, uint32_t idx, voi
if (changed) {
PULSEAUDIO_pa_context_get_server_info(hotplug_context, ServerInfoCallback, NULL);
}
PULSEAUDIO_pa_context_get_sink_info_by_index(hotplug_context, idx, SinkInfoCallback, (void*) ((intptr_t) added));
PULSEAUDIO_pa_context_get_sink_info_by_index(hotplug_context, idx, SinkInfoCallback, (void *)((intptr_t)added));
} else if ((added || changed) && source) {
if (changed) {
PULSEAUDIO_pa_context_get_server_info(hotplug_context, ServerInfoCallback, NULL);
}
PULSEAUDIO_pa_context_get_source_info_by_index(hotplug_context, idx, SourceInfoCallback, (void*) ((intptr_t) added));
PULSEAUDIO_pa_context_get_source_info_by_index(hotplug_context, idx, SourceInfoCallback, (void *)((intptr_t)added));
} else if (removed && (sink || source)) {
/* removes we can handle just with the device index. */
SDL_RemoveAudioDevice(source != 0, (void *) ((intptr_t) idx+1));
SDL_RemoveAudioDevice(source != 0, (void *)((intptr_t)idx + 1));
}
}
}
/* this runs as a thread while the Pulse target is initialized to catch hotplug events. */
static int SDLCALL
HotplugThread(void *data)
static int SDLCALL HotplugThread(void *data)
{
pa_operation *o;
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_LOW);
PULSEAUDIO_pa_context_set_subscribe_callback(hotplug_context, HotplugCallback, NULL);
o = PULSEAUDIO_pa_context_subscribe(hotplug_context, PA_SUBSCRIPTION_MASK_SINK | PA_SUBSCRIPTION_MASK_SOURCE, NULL, NULL);
PULSEAUDIO_pa_operation_unref(o); /* don't wait for it, just do our thing. */
PULSEAUDIO_pa_operation_unref(o); /* don't wait for it, just do our thing. */
PULSEAUDIO_pa_mainloop_run(hotplug_mainloop, NULL);
return 0;
}
static void
PULSEAUDIO_DetectDevices()
static void PULSEAUDIO_DetectDevices()
{
WaitForPulseOperation(hotplug_mainloop, PULSEAUDIO_pa_context_get_server_info(hotplug_context, ServerInfoCallback, NULL));
WaitForPulseOperation(hotplug_mainloop, PULSEAUDIO_pa_context_get_sink_info_list(hotplug_context, SinkInfoCallback, (void*) ((intptr_t) SDL_TRUE)));
WaitForPulseOperation(hotplug_mainloop, PULSEAUDIO_pa_context_get_source_info_list(hotplug_context, SourceInfoCallback, (void*) ((intptr_t) SDL_TRUE)));
WaitForPulseOperation(hotplug_mainloop, PULSEAUDIO_pa_context_get_sink_info_list(hotplug_context, SinkInfoCallback, (void *)((intptr_t)SDL_TRUE)));
WaitForPulseOperation(hotplug_mainloop, PULSEAUDIO_pa_context_get_source_info_list(hotplug_context, SourceInfoCallback, (void *)((intptr_t)SDL_TRUE)));
/* ok, we have a sane list, let's set up hotplug notifications now... */
hotplug_thread = SDL_CreateThreadInternal(HotplugThread, "PulseHotplug", 256 * 1024, NULL);
}
static int
PULSEAUDIO_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
static int PULSEAUDIO_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
{
int i;
int numdevices;
@@ -875,8 +843,7 @@ PULSEAUDIO_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
return SDL_SetError("Could not find default PulseAudio device");
}
static void
PULSEAUDIO_Deinitialize(void)
static void PULSEAUDIO_Deinitialize(void)
{
if (hotplug_thread) {
PULSEAUDIO_pa_mainloop_quit(hotplug_mainloop, 0);
@@ -908,8 +875,7 @@ PULSEAUDIO_Deinitialize(void)
UnloadPulseAudioLibrary();
}
static SDL_bool
PULSEAUDIO_Init(SDL_AudioDriverImpl * impl)
static SDL_bool PULSEAUDIO_Init(SDL_AudioDriverImpl *impl)
{
if (LoadPulseAudioLibrary() < 0) {
return SDL_FALSE;
@@ -937,7 +903,7 @@ PULSEAUDIO_Init(SDL_AudioDriverImpl * impl)
impl->HasCaptureSupport = SDL_TRUE;
impl->SupportsNonPow2Samples = SDL_TRUE;
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE; /* this audio target is available. */
}
AudioBootStrap PULSEAUDIO_bootstrap = {

2
src/audio/pulseaudio/SDL_pulseaudio.h
View File

@@ -43,7 +43,7 @@ struct SDL_PrivateAudioData
Uint8 *mixbuf;
int mixlen;
int bytes_requested; /* bytes of data the hardware wants _now_. */
int bytes_requested; /* bytes of data the hardware wants _now_. */
const Uint8 *capturebuf;
int capturelen;

83
src/audio/sndio/SDL_sndioaudio.c
View File

@@ -50,7 +50,7 @@
#define SIO_DEVANY "default"
#endif
static struct sio_hdl * (*SNDIO_sio_open)(const char *, unsigned int, int);
static struct sio_hdl *(*SNDIO_sio_open)(const char *, unsigned int, int);
static void (*SNDIO_sio_close)(struct sio_hdl *);
static int (*SNDIO_sio_setpar)(struct sio_hdl *, struct sio_par *);
static int (*SNDIO_sio_getpar)(struct sio_hdl *, struct sio_par *);
@@ -68,8 +68,7 @@ static void (*SNDIO_sio_initpar)(struct sio_par *);
static const char *sndio_library = SDL_AUDIO_DRIVER_SNDIO_DYNAMIC;
static void *sndio_handle = NULL;
static int
load_sndio_sym(const char *fn, void **addr)
static int load_sndio_sym(const char *fn, void **addr)
{
*addr = SDL_LoadFunction(sndio_handle, fn);
if (*addr == NULL) {
@@ -81,14 +80,14 @@ load_sndio_sym(const char *fn, void **addr)
}
/* cast funcs to char* first, to please GCC's strict aliasing rules. */
#define SDL_SNDIO_SYM(x) \
if (!load_sndio_sym(#x, (void **) (char *) &SNDIO_##x)) return -1
#define SDL_SNDIO_SYM(x) \
if (!load_sndio_sym(#x, (void **)(char *)&SNDIO_##x)) \
return -1
#else
#define SDL_SNDIO_SYM(x) SNDIO_##x = x
#endif
static int
load_sndio_syms(void)
static int load_sndio_syms(void)
{
SDL_SNDIO_SYM(sio_open);
SDL_SNDIO_SYM(sio_close);
@@ -110,8 +109,7 @@ load_sndio_syms(void)
#ifdef SDL_AUDIO_DRIVER_SNDIO_DYNAMIC
static void
UnloadSNDIOLibrary(void)
static void UnloadSNDIOLibrary(void)
{
if (sndio_handle != NULL) {
SDL_UnloadObject(sndio_handle);
@@ -119,8 +117,7 @@ UnloadSNDIOLibrary(void)
}
}
static int
LoadSNDIOLibrary(void)
static int LoadSNDIOLibrary(void)
{
int retval = 0;
if (sndio_handle == NULL) {
@@ -140,13 +137,11 @@ LoadSNDIOLibrary(void)
#else
static void
UnloadSNDIOLibrary(void)
static void UnloadSNDIOLibrary(void)
{
}
static int
LoadSNDIOLibrary(void)
static int LoadSNDIOLibrary(void)
{
load_sndio_syms();
return 0;
@@ -154,24 +149,19 @@ LoadSNDIOLibrary(void)
#endif /* SDL_AUDIO_DRIVER_SNDIO_DYNAMIC */
static void
SNDIO_WaitDevice(_THIS)
static void SNDIO_WaitDevice(_THIS)
{
/* no-op; SNDIO_sio_write() blocks if necessary. */
}
static void
SNDIO_PlayDevice(_THIS)
static void SNDIO_PlayDevice(_THIS)
{
const int written = SNDIO_sio_write(this->hidden->dev,
this->hidden->mixbuf,
this->hidden->mixlen);
/* If we couldn't write, assume fatal error for now */
if ( written == 0 ) {
if (written == 0) {
SDL_OpenedAudioDeviceDisconnected(this);
}
#ifdef DEBUG_AUDIO
@@ -179,8 +169,7 @@ SNDIO_PlayDevice(_THIS)
#endif
}
static int
SNDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
static int SNDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
size_t r;
int revents;
@@ -189,8 +178,7 @@ SNDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
/* Emulate a blocking read */
r = SNDIO_sio_read(this->hidden->dev, buffer, buflen);
while (r == 0 && !SNDIO_sio_eof(this->hidden->dev)) {
if ((nfds = SNDIO_sio_pollfd(this->hidden->dev, this->hidden->pfd, POLLIN)) <= 0
|| poll(this->hidden->pfd, nfds, INFTIM) < 0) {
if ((nfds = SNDIO_sio_pollfd(this->hidden->dev, this->hidden->pfd, POLLIN)) <= 0 || poll(this->hidden->pfd, nfds, INFTIM) < 0) {
return -1;
}
revents = SNDIO_sio_revents(this->hidden->dev, this->hidden->pfd);
@@ -201,11 +189,10 @@ SNDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
break;
}
}
return (int) r;
return (int)r;
}
static void
SNDIO_FlushCapture(_THIS)
static void SNDIO_FlushCapture(_THIS)
{
char buf[512];
@@ -214,19 +201,17 @@ SNDIO_FlushCapture(_THIS)
}
}
static Uint8 *
SNDIO_GetDeviceBuf(_THIS)
static Uint8 *SNDIO_GetDeviceBuf(_THIS)
{
return this->hidden->mixbuf;
}
static void
SNDIO_CloseDevice(_THIS)
static void SNDIO_CloseDevice(_THIS)
{
if ( this->hidden->pfd != NULL ) {
if (this->hidden->pfd != NULL) {
SDL_free(this->hidden->pfd);
}
if ( this->hidden->dev != NULL ) {
if (this->hidden->dev != NULL) {
SNDIO_sio_stop(this->hidden->dev);
SNDIO_sio_close(this->hidden->dev);
}
@@ -234,8 +219,7 @@ SNDIO_CloseDevice(_THIS)
SDL_free(this->hidden);
}
static int
SNDIO_OpenDevice(_THIS, const char *devname)
static int SNDIO_OpenDevice(_THIS, const char *devname)
{
SDL_AudioFormat test_format;
struct sio_par par;
@@ -252,14 +236,14 @@ SNDIO_OpenDevice(_THIS, const char *devname)
/* Capture devices must be non-blocking for SNDIO_FlushCapture */
if ((this->hidden->dev =
SNDIO_sio_open(devname != NULL ? devname : SIO_DEVANY,
iscapture ? SIO_REC : SIO_PLAY, iscapture)) == NULL) {
SNDIO_sio_open(devname != NULL ? devname : SIO_DEVANY,
iscapture ? SIO_REC : SIO_PLAY, iscapture)) == NULL) {
return SDL_SetError("sio_open() failed");
}
/* Allocate the pollfd array for capture devices */
if (iscapture && (this->hidden->pfd =
SDL_malloc(sizeof(struct pollfd) * SNDIO_sio_nfds(this->hidden->dev))) == NULL) {
SDL_malloc(sizeof(struct pollfd) * SNDIO_sio_nfds(this->hidden->dev))) == NULL) {
return SDL_OutOfMemory();
}
@@ -325,7 +309,7 @@ SNDIO_OpenDevice(_THIS, const char *devname)
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
this->hidden->mixbuf = (Uint8 *)SDL_malloc(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
return SDL_OutOfMemory();
}
@@ -339,21 +323,18 @@ SNDIO_OpenDevice(_THIS, const char *devname)
return 0;
}
static void
SNDIO_Deinitialize(void)
static void SNDIO_Deinitialize(void)
{
UnloadSNDIOLibrary();
}
static void
SNDIO_DetectDevices(void)
static void SNDIO_DetectDevices(void)
{
SDL_AddAudioDevice(SDL_FALSE, DEFAULT_OUTPUT_DEVNAME, NULL, (void *) 0x1);
SDL_AddAudioDevice(SDL_TRUE, DEFAULT_INPUT_DEVNAME, NULL, (void *) 0x2);
SDL_AddAudioDevice(SDL_FALSE, DEFAULT_OUTPUT_DEVNAME, NULL, (void *)0x1);
SDL_AddAudioDevice(SDL_TRUE, DEFAULT_INPUT_DEVNAME, NULL, (void *)0x2);
}
static SDL_bool
SNDIO_Init(SDL_AudioDriverImpl * impl)
static SDL_bool SNDIO_Init(SDL_AudioDriverImpl *impl)
{
if (LoadSNDIOLibrary() < 0) {
return SDL_FALSE;
@@ -373,7 +354,7 @@ SNDIO_Init(SDL_AudioDriverImpl * impl)
impl->AllowsArbitraryDeviceNames = SDL_TRUE;
impl->HasCaptureSupport = SDL_TRUE;
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE; /* this audio target is available. */
}
AudioBootStrap SNDIO_bootstrap = {

2
src/audio/sndio/SDL_sndioaudio.h
View File

@@ -29,7 +29,7 @@
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{

25
src/audio/vita/SDL_vitaaudio.c
View File

@@ -36,11 +36,10 @@
#include <psp2/audioout.h>
#include <psp2/audioin.h>
#define SCE_AUDIO_SAMPLE_ALIGN(s) (((s) + 63) & ~63)
#define SCE_AUDIO_MAX_VOLUME 0x8000
#define SCE_AUDIO_SAMPLE_ALIGN(s) (((s) + 63) & ~63)
#define SCE_AUDIO_MAX_VOLUME 0x8000
static int
VITAAUD_OpenCaptureDevice(_THIS)
static int VITAAUD_OpenCaptureDevice(_THIS)
{
this->spec.freq = 16000;
this->spec.samples = 512;
@@ -48,7 +47,7 @@ VITAAUD_OpenCaptureDevice(_THIS)
SDL_CalculateAudioSpec(&this->spec);
this->hidden->port = sceAudioInOpenPort(SCE_AUDIO_IN_PORT_TYPE_VOICE , 512, 16000, SCE_AUDIO_IN_PARAM_FORMAT_S16_MONO);
this->hidden->port = sceAudioInOpenPort(SCE_AUDIO_IN_PORT_TYPE_VOICE, 512, 16000, SCE_AUDIO_IN_PARAM_FORMAT_S16_MONO);
if (this->hidden->port < 0) {
return SDL_SetError("Couldn't open audio in port: %x", this->hidden->port);
@@ -57,11 +56,10 @@ VITAAUD_OpenCaptureDevice(_THIS)
return 0;
}
static int
VITAAUD_OpenDevice(_THIS, const char *devname)
static int VITAAUD_OpenDevice(_THIS, const char *devname)
{
int format, mixlen, i, port = SCE_AUDIO_OUT_PORT_TYPE_MAIN;
int vols[2] = {SCE_AUDIO_MAX_VOLUME, SCE_AUDIO_MAX_VOLUME};
int vols[2] = { SCE_AUDIO_MAX_VOLUME, SCE_AUDIO_MAX_VOLUME };
SDL_AudioFormat test_format;
this->hidden = (struct SDL_PrivateAudioData *)
@@ -96,7 +94,7 @@ VITAAUD_OpenDevice(_THIS, const char *devname)
be a multiple of 64 bytes. Our sample count is already a multiple of
64, so spec->size should be a multiple of 64 as well. */
mixlen = this->spec.size * NUM_BUFFERS;
this->hidden->rawbuf = (Uint8 *) memalign(64, mixlen);
this->hidden->rawbuf = (Uint8 *)memalign(64, mixlen);
if (this->hidden->rawbuf == NULL) {
return SDL_SetError("Couldn't allocate mixing buffer");
}
@@ -119,7 +117,7 @@ VITAAUD_OpenDevice(_THIS, const char *devname)
return SDL_SetError("Couldn't open audio out port: %x", this->hidden->port);
}
sceAudioOutSetVolume(this->hidden->port, SCE_AUDIO_VOLUME_FLAG_L_CH|SCE_AUDIO_VOLUME_FLAG_R_CH, vols);
sceAudioOutSetVolume(this->hidden->port, SCE_AUDIO_VOLUME_FLAG_L_CH | SCE_AUDIO_VOLUME_FLAG_R_CH, vols);
SDL_memset(this->hidden->rawbuf, 0, mixlen);
for (i = 0; i < NUM_BUFFERS; i++) {
@@ -162,7 +160,7 @@ static void VITAAUD_CloseDevice(_THIS)
}
if (!this->iscapture && this->hidden->rawbuf != NULL) {
free(this->hidden->rawbuf); /* this uses memalign(), not SDL_malloc(). */
free(this->hidden->rawbuf); /* this uses memalign(), not SDL_malloc(). */
this->hidden->rawbuf = NULL;
}
}
@@ -191,8 +189,7 @@ static void VITAAUD_ThreadInit(_THIS)
}
}
static SDL_bool
VITAAUD_Init(SDL_AudioDriverImpl * impl)
static SDL_bool VITAAUD_Init(SDL_AudioDriverImpl *impl)
{
/* Set the function pointers */
impl->OpenDevice = VITAAUD_OpenDevice;
@@ -209,7 +206,7 @@ VITAAUD_Init(SDL_AudioDriverImpl * impl)
impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
impl->OnlyHasDefaultCaptureDevice = SDL_TRUE;
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE; /* this audio target is available. */
}
AudioBootStrap VITAAUD_bootstrap = {

13
src/audio/vita/SDL_vitaaudio.h
View File

@@ -25,19 +25,20 @@
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define _THIS SDL_AudioDevice *this
#define NUM_BUFFERS 2
struct SDL_PrivateAudioData {
struct SDL_PrivateAudioData
{
/* The hardware input/output port. */
int port;
int port;
/* The raw allocated mixing buffer. */
Uint8 *rawbuf;
Uint8 *rawbuf;
/* Individual mixing buffers. */
Uint8 *mixbufs[NUM_BUFFERS];
Uint8 *mixbufs[NUM_BUFFERS];
/* Index of the next available mixing buffer. */
int next_buffer;
int next_buffer;
};
#endif /* _SDL_vitaaudio_h */

160
src/audio/wasapi/SDL_wasapi.c
View File

@@ -34,7 +34,7 @@
/* These constants aren't available in older SDKs */
#ifndef AUDCLNT_STREAMFLAGS_RATEADJUST
#define AUDCLNT_STREAMFLAGS_RATEADJUST 0x00100000
#define AUDCLNT_STREAMFLAGS_RATEADJUST 0x00100000
#endif
#ifndef AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY
#define AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY 0x08000000
@@ -44,17 +44,15 @@
#endif
/* Some GUIDs we need to know without linking to libraries that aren't available before Vista. */
static const IID SDL_IID_IAudioRenderClient = { 0xf294acfc, 0x3146, 0x4483,{ 0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2 } };
static const IID SDL_IID_IAudioCaptureClient = { 0xc8adbd64, 0xe71e, 0x48a0,{ 0xa4, 0xde, 0x18, 0x5c, 0x39, 0x5c, 0xd3, 0x17 } };
static const IID SDL_IID_IAudioRenderClient = { 0xf294acfc, 0x3146, 0x4483, { 0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2 } };
static const IID SDL_IID_IAudioCaptureClient = { 0xc8adbd64, 0xe71e, 0x48a0, { 0xa4, 0xde, 0x18, 0x5c, 0x39, 0x5c, 0xd3, 0x17 } };
static void
WASAPI_DetectDevices(void)
static void WASAPI_DetectDevices(void)
{
WASAPI_EnumerateEndpoints();
}
static SDL_INLINE SDL_bool
WasapiFailed(_THIS, const HRESULT err)
static SDL_INLINE SDL_bool WasapiFailed(_THIS, const HRESULT err)
{
if (err == S_OK) {
return SDL_FALSE;
@@ -71,39 +69,38 @@ WasapiFailed(_THIS, const HRESULT err)
return SDL_TRUE;
}
static int
UpdateAudioStream(_THIS, const SDL_AudioSpec *oldspec)
static int UpdateAudioStream(_THIS, const SDL_AudioSpec *oldspec)
{
/* Since WASAPI requires us to handle all audio conversion, and our
device format might have changed, we might have to add/remove/change
the audio stream that the higher level uses to convert data, so
SDL keeps firing the callback as if nothing happened here. */
if ( (this->callbackspec.channels == this->spec.channels) &&
(this->callbackspec.format == this->spec.format) &&
(this->callbackspec.freq == this->spec.freq) &&
(this->callbackspec.samples == this->spec.samples) ) {
if ((this->callbackspec.channels == this->spec.channels) &&
(this->callbackspec.format == this->spec.format) &&
(this->callbackspec.freq == this->spec.freq) &&
(this->callbackspec.samples == this->spec.samples)) {
/* no need to buffer/convert in an AudioStream! */
SDL_FreeAudioStream(this->stream);
this->stream = NULL;
} else if ( (oldspec->channels == this->spec.channels) &&
(oldspec->format == this->spec.format) &&
(oldspec->freq == this->spec.freq) ) {
} else if ((oldspec->channels == this->spec.channels) &&
(oldspec->format == this->spec.format) &&
(oldspec->freq == this->spec.freq)) {
/* The existing audio stream is okay to keep using. */
} else {
/* replace the audiostream for new format */
SDL_FreeAudioStream(this->stream);
if (this->iscapture) {
this->stream = SDL_NewAudioStream(this->spec.format,
this->spec.channels, this->spec.freq,
this->callbackspec.format,
this->callbackspec.channels,
this->callbackspec.freq);
this->spec.channels, this->spec.freq,
this->callbackspec.format,
this->callbackspec.channels,
this->callbackspec.freq);
} else {
this->stream = SDL_NewAudioStream(this->callbackspec.format,
this->callbackspec.channels,
this->callbackspec.freq, this->spec.format,
this->spec.channels, this->spec.freq);
this->callbackspec.channels,
this->callbackspec.freq, this->spec.format,
this->spec.channels, this->spec.freq);
}
if (!this->stream) {
@@ -113,7 +110,7 @@ UpdateAudioStream(_THIS, const SDL_AudioSpec *oldspec)
/* make sure our scratch buffer can cover the new device spec. */
if (this->spec.size > this->work_buffer_len) {
Uint8 *ptr = (Uint8 *) SDL_realloc(this->work_buffer, this->spec.size);
Uint8 *ptr = (Uint8 *)SDL_realloc(this->work_buffer, this->spec.size);
if (ptr == NULL) {
return SDL_OutOfMemory();
}
@@ -124,16 +121,14 @@ UpdateAudioStream(_THIS, const SDL_AudioSpec *oldspec)
return 0;
}
static void ReleaseWasapiDevice(_THIS);
static SDL_bool
RecoverWasapiDevice(_THIS)
static SDL_bool RecoverWasapiDevice(_THIS)
{
ReleaseWasapiDevice(this); /* dump the lost device's handles. */
ReleaseWasapiDevice(this); /* dump the lost device's handles. */
if (this->hidden->default_device_generation) {
this->hidden->default_device_generation = SDL_AtomicGet(this->iscapture ? &SDL_IMMDevice_DefaultCaptureGeneration : &SDL_IMMDevice_DefaultPlaybackGeneration);
this->hidden->default_device_generation = SDL_AtomicGet(this->iscapture ? &SDL_IMMDevice_DefaultCaptureGeneration : &SDL_IMMDevice_DefaultPlaybackGeneration);
}
/* this can fail for lots of reasons, but the most likely is we had a
@@ -148,26 +143,25 @@ RecoverWasapiDevice(_THIS)
this->hidden->device_lost = SDL_FALSE;
return SDL_TRUE; /* okay, carry on with new device details! */
return SDL_TRUE; /* okay, carry on with new device details! */
}
static SDL_bool
RecoverWasapiIfLost(_THIS)
static SDL_bool RecoverWasapiIfLost(_THIS)
{
const int generation = this->hidden->default_device_generation;
SDL_bool lost = this->hidden->device_lost;
if (!SDL_AtomicGet(&this->enabled)) {
return SDL_FALSE; /* already failed. */
return SDL_FALSE; /* already failed. */
}
if (!this->hidden->client) {
return SDL_TRUE; /* still waiting for activation. */
return SDL_TRUE; /* still waiting for activation. */
}
if (!lost && (generation > 0)) { /* is a default device? */
const int newgen = SDL_AtomicGet(this->iscapture ? &SDL_IMMDevice_DefaultCaptureGeneration : &SDL_IMMDevice_DefaultPlaybackGeneration);
if (generation != newgen) { /* the desired default device was changed, jump over to it. */
if (generation != newgen) { /* the desired default device was changed, jump over to it. */
lost = SDL_TRUE;
}
}
@@ -175,33 +169,30 @@ RecoverWasapiIfLost(_THIS)
return lost ? RecoverWasapiDevice(this) : SDL_TRUE;
}
static Uint8 *
WASAPI_GetDeviceBuf(_THIS)
static Uint8 *WASAPI_GetDeviceBuf(_THIS)
{
/* get an endpoint buffer from WASAPI. */
BYTE *buffer = NULL;
while (RecoverWasapiIfLost(this) && this->hidden->render) {
if (!WasapiFailed(this, IAudioRenderClient_GetBuffer(this->hidden->render, this->spec.samples, &buffer))) {
return (Uint8 *) buffer;
return (Uint8 *)buffer;
}
SDL_assert(buffer == NULL);
}
return (Uint8 *) buffer;
return (Uint8 *)buffer;
}
static void
WASAPI_PlayDevice(_THIS)
static void WASAPI_PlayDevice(_THIS)
{
if (this->hidden->render != NULL) { /* definitely activated? */
if (this->hidden->render != NULL) { /* definitely activated? */
/* WasapiFailed() will mark the device for reacquisition or removal elsewhere. */
WasapiFailed(this, IAudioRenderClient_ReleaseBuffer(this->hidden->render, this->spec.samples, 0));
}
}
static void
WASAPI_WaitDevice(_THIS)
static void WASAPI_WaitDevice(_THIS)
{
while (RecoverWasapiIfLost(this) && this->hidden->client && this->hidden->event) {
DWORD waitResult = WaitForSingleObjectEx(this->hidden->event, 200, FALSE);
@@ -228,8 +219,7 @@ WASAPI_WaitDevice(_THIS)
}
}
static int
WASAPI_CaptureFromDevice(_THIS, void *buffer, int buflen)
static int WASAPI_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
SDL_AudioStream *stream = this->hidden->capturestream;
const int avail = SDL_AudioStreamAvailable(stream);
@@ -261,7 +251,7 @@ WASAPI_CaptureFromDevice(_THIS, void *buffer, int buflen)
if ((ret == AUDCLNT_S_BUFFER_EMPTY) || !frames) {
WASAPI_WaitDevice(this);
} else if (ret == S_OK) {
const int total = ((int) frames) * this->hidden->framesize;
const int total = ((int)frames) * this->hidden->framesize;
const int cpy = SDL_min(buflen, total);
const int leftover = total - cpy;
const SDL_bool silent = (flags & AUDCLNT_BUFFERFLAGS_SILENT) ? SDL_TRUE : SDL_FALSE;
@@ -271,15 +261,15 @@ WASAPI_CaptureFromDevice(_THIS, void *buffer, int buflen)
} else {
SDL_memcpy(buffer, ptr, cpy);
}
if (leftover > 0) {
ptr += cpy;
if (silent) {
SDL_memset(ptr, this->spec.silence, leftover); /* I guess this is safe? */
SDL_memset(ptr, this->spec.silence, leftover); /* I guess this is safe? */
}
if (SDL_AudioStreamPut(stream, ptr, leftover) == -1) {
return -1; /* uhoh, out of memory, etc. Kill device. :( */
return -1; /* uhoh, out of memory, etc. Kill device. :( */
}
}
@@ -290,36 +280,34 @@ WASAPI_CaptureFromDevice(_THIS, void *buffer, int buflen)
}
}
return -1; /* unrecoverable error. */
return -1; /* unrecoverable error. */
}
static void
WASAPI_FlushCapture(_THIS)
static void WASAPI_FlushCapture(_THIS)
{
BYTE *ptr = NULL;
UINT32 frames = 0;
DWORD flags = 0;
if (!this->hidden->capture) {
return; /* not activated yet? */
return; /* not activated yet? */
}
/* just read until we stop getting packets, throwing them away. */
while (SDL_TRUE) {
const HRESULT ret = IAudioCaptureClient_GetBuffer(this->hidden->capture, &ptr, &frames, &flags, NULL, NULL);
if (ret == AUDCLNT_S_BUFFER_EMPTY) {
break; /* no more buffered data; we're done. */
break; /* no more buffered data; we're done. */
} else if (WasapiFailed(this, ret)) {
break; /* failed for some other reason, abort. */
break; /* failed for some other reason, abort. */
} else if (WasapiFailed(this, IAudioCaptureClient_ReleaseBuffer(this->hidden->capture, frames))) {
break; /* something broke. */
break; /* something broke. */
}
}
SDL_AudioStreamClear(this->hidden->capturestream);
}
static void
ReleaseWasapiDevice(_THIS)
static void ReleaseWasapiDevice(_THIS)
{
if (this->hidden->client) {
IAudioClient_Stop(this->hidden->client);
@@ -358,20 +346,17 @@ ReleaseWasapiDevice(_THIS)
}
}
static void
WASAPI_CloseDevice(_THIS)
static void WASAPI_CloseDevice(_THIS)
{
WASAPI_UnrefDevice(this);
}
void
WASAPI_RefDevice(_THIS)
void WASAPI_RefDevice(_THIS)
{
SDL_AtomicIncRef(&this->hidden->refcount);
}
void
WASAPI_UnrefDevice(_THIS)
void WASAPI_UnrefDevice(_THIS)
{
if (!SDL_AtomicDecRef(&this->hidden->refcount)) {
return;
@@ -388,8 +373,7 @@ WASAPI_UnrefDevice(_THIS)
}
/* This is called once a device is activated, possibly asynchronously. */
int
WASAPI_PrepDevice(_THIS, const SDL_bool updatestream)
int WASAPI_PrepDevice(_THIS, const SDL_bool updatestream)
{
/* !!! FIXME: we could request an exclusive mode stream, which is lower latency;
!!! it will write into the kernel's audio buffer directly instead of
@@ -404,7 +388,7 @@ WASAPI_PrepDevice(_THIS, const SDL_bool updatestream)
!!! do in any case. */
const SDL_AudioSpec oldspec = this->spec;
const AUDCLNT_SHAREMODE sharemode = AUDCLNT_SHAREMODE_SHARED;
UINT32 bufsize = 0; /* this is in sample frames, not samples, not bytes. */
UINT32 bufsize = 0; /* this is in sample frames, not samples, not bytes. */
REFERENCE_TIME default_period = 0;
IAudioClient *client = this->hidden->client;
IAudioRenderClient *render = NULL;
@@ -435,7 +419,7 @@ WASAPI_PrepDevice(_THIS, const SDL_bool updatestream)
SDL_assert(waveformat != NULL);
this->hidden->waveformat = waveformat;
this->spec.channels = (Uint8) waveformat->nChannels;
this->spec.channels = (Uint8)waveformat->nChannels;
/* Make sure we have a valid format that we can convert to whatever WASAPI wants. */
wasapi_format = WaveFormatToSDLFormat(waveformat);
@@ -502,10 +486,10 @@ WASAPI_PrepDevice(_THIS, const SDL_bool updatestream)
if (this->iscapture) {
this->hidden->capturestream = SDL_NewAudioStream(this->spec.format, this->spec.channels, this->spec.freq, this->spec.format, this->spec.channels, this->spec.freq);
if (!this->hidden->capturestream) {
return -1; /* already set SDL_Error */
return -1; /* already set SDL_Error */
}
ret = IAudioClient_GetService(client, &SDL_IID_IAudioCaptureClient, (void**) &capture);
ret = IAudioClient_GetService(client, &SDL_IID_IAudioCaptureClient, (void **)&capture);
if (FAILED(ret)) {
return WIN_SetErrorFromHRESULT("WASAPI can't get capture client service", ret);
}
@@ -517,9 +501,9 @@ WASAPI_PrepDevice(_THIS, const SDL_bool updatestream)
return WIN_SetErrorFromHRESULT("WASAPI can't start capture", ret);
}
WASAPI_FlushCapture(this); /* MSDN says you should flush capture endpoint right after startup. */
WASAPI_FlushCapture(this); /* MSDN says you should flush capture endpoint right after startup. */
} else {
ret = IAudioClient_GetService(client, &SDL_IID_IAudioRenderClient, (void**) &render);
ret = IAudioClient_GetService(client, &SDL_IID_IAudioRenderClient, (void **)&render);
if (FAILED(ret)) {
return WIN_SetErrorFromHRESULT("WASAPI can't get render client service", ret);
}
@@ -536,14 +520,12 @@ WASAPI_PrepDevice(_THIS, const SDL_bool updatestream)
return UpdateAudioStream(this, &oldspec);
}
return 0; /* good to go. */
return 0; /* good to go. */
}
static int
WASAPI_OpenDevice(_THIS, const char *devname)
static int WASAPI_OpenDevice(_THIS, const char *devname)
{
LPCWSTR devid = (LPCWSTR) this->handle;
LPCWSTR devid = (LPCWSTR)this->handle;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
@@ -553,9 +535,9 @@ WASAPI_OpenDevice(_THIS, const char *devname)
}
SDL_zerop(this->hidden);
WASAPI_RefDevice(this); /* so CloseDevice() will unref to zero. */
WASAPI_RefDevice(this); /* so CloseDevice() will unref to zero. */
if (!devid) { /* is default device? */
if (!devid) { /* is default device? */
this->hidden->default_device_generation = SDL_AtomicGet(this->iscapture ? &SDL_IMMDevice_DefaultCaptureGeneration : &SDL_IMMDevice_DefaultPlaybackGeneration);
} else {
this->hidden->devid = SDL_wcsdup(devid);
@@ -565,7 +547,7 @@ WASAPI_OpenDevice(_THIS, const char *devname)
}
if (WASAPI_ActivateDevice(this, SDL_FALSE) == -1) {
return -1; /* already set error. */
return -1; /* already set error. */
}
/* Ready, but waiting for async device activation.
@@ -579,26 +561,22 @@ WASAPI_OpenDevice(_THIS, const char *devname)
return 0;
}
static void
WASAPI_ThreadInit(_THIS)
static void WASAPI_ThreadInit(_THIS)
{
WASAPI_PlatformThreadInit(this);
}
static void
WASAPI_ThreadDeinit(_THIS)
static void WASAPI_ThreadDeinit(_THIS)
{
WASAPI_PlatformThreadDeinit(this);
}
static void
WASAPI_Deinitialize(void)
static void WASAPI_Deinitialize(void)
{
WASAPI_PlatformDeinit();
}
static SDL_bool
WASAPI_Init(SDL_AudioDriverImpl * impl)
static SDL_bool WASAPI_Init(SDL_AudioDriverImpl *impl)
{
if (WASAPI_PlatformInit() == -1) {
return SDL_FALSE;
@@ -620,13 +598,13 @@ WASAPI_Init(SDL_AudioDriverImpl * impl)
impl->HasCaptureSupport = SDL_TRUE;
impl->SupportsNonPow2Samples = SDL_TRUE;
return SDL_TRUE; /* this audio target is available. */
return SDL_TRUE; /* this audio target is available. */
}
AudioBootStrap WASAPI_bootstrap = {
"wasapi", "WASAPI", WASAPI_Init, SDL_FALSE
};
#endif /* SDL_AUDIO_DRIVER_WASAPI */
#endif /* SDL_AUDIO_DRIVER_WASAPI */
/* vi: set ts=4 sw=4 expandtab: */

43
src/audio/wasapi/SDL_wasapi_win32.c
View File

@@ -45,26 +45,24 @@ static pfnAvSetMmThreadCharacteristicsW pAvSetMmThreadCharacteristicsW = NULL;
static pfnAvRevertMmThreadCharacteristics pAvRevertMmThreadCharacteristics = NULL;
/* Some GUIDs we need to know without linking to libraries that aren't available before Vista. */
static const IID SDL_IID_IAudioClient = { 0x1cb9ad4c, 0xdbfa, 0x4c32,{ 0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2 } };
static const IID SDL_IID_IAudioClient = { 0x1cb9ad4c, 0xdbfa, 0x4c32, { 0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2 } };
int
WASAPI_PlatformInit(void)
int WASAPI_PlatformInit(void)
{
if (SDL_IMMDevice_Init() < 0) {
return -1; /* This is set by SDL_IMMDevice_Init */
}
libavrt = LoadLibrary(TEXT("avrt.dll")); /* this library is available in Vista and later. No WinXP, so have to LoadLibrary to use it for now! */
libavrt = LoadLibrary(TEXT("avrt.dll")); /* this library is available in Vista and later. No WinXP, so have to LoadLibrary to use it for now! */
if (libavrt) {
pAvSetMmThreadCharacteristicsW = (pfnAvSetMmThreadCharacteristicsW) GetProcAddress(libavrt, "AvSetMmThreadCharacteristicsW");
pAvRevertMmThreadCharacteristics = (pfnAvRevertMmThreadCharacteristics) GetProcAddress(libavrt, "AvRevertMmThreadCharacteristics");
pAvSetMmThreadCharacteristicsW = (pfnAvSetMmThreadCharacteristicsW)GetProcAddress(libavrt, "AvSetMmThreadCharacteristicsW");
pAvRevertMmThreadCharacteristics = (pfnAvRevertMmThreadCharacteristics)GetProcAddress(libavrt, "AvRevertMmThreadCharacteristics");
}
return 0;
}
void
WASAPI_PlatformDeinit(void)
void WASAPI_PlatformDeinit(void)
{
if (libavrt) {
FreeLibrary(libavrt);
@@ -77,11 +75,10 @@ WASAPI_PlatformDeinit(void)
SDL_IMMDevice_Quit();
}
void
WASAPI_PlatformThreadInit(_THIS)
void WASAPI_PlatformThreadInit(_THIS)
{
/* this thread uses COM. */
if (SUCCEEDED(WIN_CoInitialize())) { /* can't report errors, hope it worked! */
if (SUCCEEDED(WIN_CoInitialize())) { /* can't report errors, hope it worked! */
this->hidden->coinitialized = SDL_TRUE;
}
@@ -92,8 +89,7 @@ WASAPI_PlatformThreadInit(_THIS)
}
}
void
WASAPI_PlatformThreadDeinit(_THIS)
void WASAPI_PlatformThreadDeinit(_THIS)
{
/* Set this thread back to normal priority. */
if (this->hidden->task && pAvRevertMmThreadCharacteristics) {
@@ -107,8 +103,7 @@ WASAPI_PlatformThreadDeinit(_THIS)
}
}
int
WASAPI_ActivateDevice(_THIS, const SDL_bool isrecovery)
int WASAPI_ActivateDevice(_THIS, const SDL_bool isrecovery)
{
IMMDevice *device = NULL;
HRESULT ret;
@@ -119,7 +114,7 @@ WASAPI_ActivateDevice(_THIS, const SDL_bool isrecovery)
}
/* this is not async in standard win32, yay! */
ret = IMMDevice_Activate(device, &SDL_IID_IAudioClient, CLSCTX_ALL, NULL, (void **) &this->hidden->client);
ret = IMMDevice_Activate(device, &SDL_IID_IAudioClient, CLSCTX_ALL, NULL, (void **)&this->hidden->client);
IMMDevice_Release(device);
if (FAILED(ret)) {
@@ -128,33 +123,29 @@ WASAPI_ActivateDevice(_THIS, const SDL_bool isrecovery)
}
SDL_assert(this->hidden->client != NULL);
if (WASAPI_PrepDevice(this, isrecovery) == -1) { /* not async, fire it right away. */
if (WASAPI_PrepDevice(this, isrecovery) == -1) { /* not async, fire it right away. */
return -1;
}
return 0; /* good to go. */
return 0; /* good to go. */
}
void
WASAPI_EnumerateEndpoints(void)
void WASAPI_EnumerateEndpoints(void)
{
SDL_IMMDevice_EnumerateEndpoints(SDL_FALSE);
}
int
WASAPI_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
int WASAPI_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
{
return SDL_IMMDevice_GetDefaultAudioInfo(name, spec, iscapture);
}
void
WASAPI_PlatformDeleteActivationHandler(void *handler)
void WASAPI_PlatformDeleteActivationHandler(void *handler)
{
/* not asynchronous. */
SDL_assert(!"This function should have only been called on WinRT.");
}
#endif /* SDL_AUDIO_DRIVER_WASAPI && !defined(__WINRT__) */
#endif /* SDL_AUDIO_DRIVER_WASAPI && !defined(__WINRT__) */
/* vi: set ts=4 sw=4 expandtab: */

123
src/audio/wasapi/SDL_wasapi_winrt.cpp
View File

@@ -51,13 +51,13 @@ using namespace Windows::Media::Devices;
using namespace Windows::Foundation;
using namespace Microsoft::WRL;
static Platform::String^ SDL_PKEY_AudioEngine_DeviceFormat = L"{f19f064d-082c-4e27-bc73-6882a1bb8e4c} 0";
static Platform::String ^ SDL_PKEY_AudioEngine_DeviceFormat = L"{f19f064d-082c-4e27-bc73-6882a1bb8e4c} 0";
static void WASAPI_AddDevice(const SDL_bool iscapture, const char *devname, WAVEFORMATEXTENSIBLE *fmt, LPCWSTR devid);
static void WASAPI_RemoveDevice(const SDL_bool iscapture, LPCWSTR devid);
extern "C" {
SDL_atomic_t SDL_IMMDevice_DefaultPlaybackGeneration;
SDL_atomic_t SDL_IMMDevice_DefaultCaptureGeneration;
SDL_atomic_t SDL_IMMDevice_DefaultPlaybackGeneration;
SDL_atomic_t SDL_IMMDevice_DefaultCaptureGeneration;
}
/* This is a list of device id strings we have inflight, so we have consistent pointers to the same device. */
@@ -71,20 +71,20 @@ static DevIdList *deviceid_list = NULL;
class SDL_WasapiDeviceEventHandler
{
public:
public:
SDL_WasapiDeviceEventHandler(const SDL_bool _iscapture);
~SDL_WasapiDeviceEventHandler();
void OnDeviceAdded(DeviceWatcher^ sender, DeviceInformation^ args);
void OnDeviceRemoved(DeviceWatcher^ sender, DeviceInformationUpdate^ args);
void OnDeviceUpdated(DeviceWatcher^ sender, DeviceInformationUpdate^ args);
void OnEnumerationCompleted(DeviceWatcher^ sender, Platform::Object^ args);
void OnDefaultRenderDeviceChanged(Platform::Object^ sender, DefaultAudioRenderDeviceChangedEventArgs^ args);
void OnDefaultCaptureDeviceChanged(Platform::Object^ sender, DefaultAudioCaptureDeviceChangedEventArgs^ args);
SDL_semaphore* completed;
void OnDeviceAdded(DeviceWatcher ^ sender, DeviceInformation ^ args);
void OnDeviceRemoved(DeviceWatcher ^ sender, DeviceInformationUpdate ^ args);
void OnDeviceUpdated(DeviceWatcher ^ sender, DeviceInformationUpdate ^ args);
void OnEnumerationCompleted(DeviceWatcher ^ sender, Platform::Object ^ args);
void OnDefaultRenderDeviceChanged(Platform::Object ^ sender, DefaultAudioRenderDeviceChangedEventArgs ^ args);
void OnDefaultCaptureDeviceChanged(Platform::Object ^ sender, DefaultAudioCaptureDeviceChangedEventArgs ^ args);
SDL_semaphore *completed;
private:
private:
const SDL_bool iscapture;
DeviceWatcher^ watcher;
DeviceWatcher ^ watcher;
Windows::Foundation::EventRegistrationToken added_handler;
Windows::Foundation::EventRegistrationToken removed_handler;
Windows::Foundation::EventRegistrationToken updated_handler;
@@ -93,29 +93,27 @@ private:
};
SDL_WasapiDeviceEventHandler::SDL_WasapiDeviceEventHandler(const SDL_bool _iscapture)
: iscapture(_iscapture)
, completed(SDL_CreateSemaphore(0))
: iscapture(_iscapture), completed(SDL_CreateSemaphore(0))
{
if (!completed)
return; // uhoh.
return; // uhoh.
Platform::String^ selector = _iscapture ? MediaDevice::GetAudioCaptureSelector() :
MediaDevice::GetAudioRenderSelector();
Platform::Collections::Vector<Platform::String^> properties;
Platform::String ^ selector = _iscapture ? MediaDevice::GetAudioCaptureSelector() : MediaDevice::GetAudioRenderSelector();
Platform::Collections::Vector<Platform::String ^> properties;
properties.Append(SDL_PKEY_AudioEngine_DeviceFormat);
watcher = DeviceInformation::CreateWatcher(selector, properties.GetView());
if (!watcher)
return; // uhoh.
return; // uhoh.
// !!! FIXME: this doesn't need a lambda here, I think, if I make SDL_WasapiDeviceEventHandler a proper C++/CX class. --ryan.
added_handler = watcher->Added += ref new TypedEventHandler<DeviceWatcher^, DeviceInformation^>([this](DeviceWatcher^ sender, DeviceInformation^ args) { OnDeviceAdded(sender, args); } );
removed_handler = watcher->Removed += ref new TypedEventHandler<DeviceWatcher^, DeviceInformationUpdate^>([this](DeviceWatcher^ sender, DeviceInformationUpdate^ args) { OnDeviceRemoved(sender, args); } );
updated_handler = watcher->Updated += ref new TypedEventHandler<DeviceWatcher^, DeviceInformationUpdate^>([this](DeviceWatcher^ sender, DeviceInformationUpdate^ args) { OnDeviceUpdated(sender, args); } );
completed_handler = watcher->EnumerationCompleted += ref new TypedEventHandler<DeviceWatcher^, Platform::Object^>([this](DeviceWatcher^ sender, Platform::Object^ args) { OnEnumerationCompleted(sender, args); } );
added_handler = watcher->Added += ref new TypedEventHandler<DeviceWatcher ^, DeviceInformation ^>([this](DeviceWatcher ^ sender, DeviceInformation ^ args) { OnDeviceAdded(sender, args); });
removed_handler = watcher->Removed += ref new TypedEventHandler<DeviceWatcher ^, DeviceInformationUpdate ^>([this](DeviceWatcher ^ sender, DeviceInformationUpdate ^ args) { OnDeviceRemoved(sender, args); });
updated_handler = watcher->Updated += ref new TypedEventHandler<DeviceWatcher ^, DeviceInformationUpdate ^>([this](DeviceWatcher ^ sender, DeviceInformationUpdate ^ args) { OnDeviceUpdated(sender, args); });
completed_handler = watcher->EnumerationCompleted += ref new TypedEventHandler<DeviceWatcher ^, Platform::Object ^>([this](DeviceWatcher ^ sender, Platform::Object ^ args) { OnEnumerationCompleted(sender, args); });
if (iscapture) {
default_changed_handler = MediaDevice::DefaultAudioCaptureDeviceChanged += ref new TypedEventHandler<Platform::Object^, DefaultAudioCaptureDeviceChangedEventArgs^>([this](Platform::Object^ sender, DefaultAudioCaptureDeviceChangedEventArgs^ args) { OnDefaultCaptureDeviceChanged(sender, args); } );
default_changed_handler = MediaDevice::DefaultAudioCaptureDeviceChanged += ref new TypedEventHandler<Platform::Object ^, DefaultAudioCaptureDeviceChangedEventArgs ^>([this](Platform::Object ^ sender, DefaultAudioCaptureDeviceChangedEventArgs ^ args) { OnDefaultCaptureDeviceChanged(sender, args); });
} else {
default_changed_handler = MediaDevice::DefaultAudioRenderDeviceChanged += ref new TypedEventHandler<Platform::Object^, DefaultAudioRenderDeviceChangedEventArgs^>([this](Platform::Object^ sender, DefaultAudioRenderDeviceChangedEventArgs^ args) { OnDefaultRenderDeviceChanged(sender, args); } );
default_changed_handler = MediaDevice::DefaultAudioRenderDeviceChanged += ref new TypedEventHandler<Platform::Object ^, DefaultAudioRenderDeviceChangedEventArgs ^>([this](Platform::Object ^ sender, DefaultAudioRenderDeviceChangedEventArgs ^ args) { OnDefaultRenderDeviceChanged(sender, args); });
}
watcher->Start();
}
@@ -142,17 +140,16 @@ SDL_WasapiDeviceEventHandler::~SDL_WasapiDeviceEventHandler()
}
}
void
SDL_WasapiDeviceEventHandler::OnDeviceAdded(DeviceWatcher^ sender, DeviceInformation^ info)
void SDL_WasapiDeviceEventHandler::OnDeviceAdded(DeviceWatcher ^ sender, DeviceInformation ^ info)
{
SDL_assert(sender == this->watcher);
char *utf8dev = WIN_StringToUTF8(info->Name->Data());
if (utf8dev) {
WAVEFORMATEXTENSIBLE fmt;
Platform::Object^ obj = info->Properties->Lookup(SDL_PKEY_AudioEngine_DeviceFormat);
Platform::Object ^ obj = info->Properties->Lookup(SDL_PKEY_AudioEngine_DeviceFormat);
if (obj) {
IPropertyValue^ property = (IPropertyValue^) obj;
Platform::Array<unsigned char>^ data;
IPropertyValue ^ property = (IPropertyValue ^) obj;
Platform::Array<unsigned char> ^ data;
property->GetUInt8Array(&data);
SDL_memcpy(&fmt, data->Data, SDL_min(data->Length, sizeof(WAVEFORMATEXTENSIBLE)));
} else {
@@ -164,41 +161,35 @@ SDL_WasapiDeviceEventHandler::OnDeviceAdded(DeviceWatcher^ sender, DeviceInforma
}
}
void
SDL_WasapiDeviceEventHandler::OnDeviceRemoved(DeviceWatcher^ sender, DeviceInformationUpdate^ info)
void SDL_WasapiDeviceEventHandler::OnDeviceRemoved(DeviceWatcher ^ sender, DeviceInformationUpdate ^ info)
{
SDL_assert(sender == this->watcher);
WASAPI_RemoveDevice(this->iscapture, info->Id->Data());
}
void
SDL_WasapiDeviceEventHandler::OnDeviceUpdated(DeviceWatcher^ sender, DeviceInformationUpdate^ args)
void SDL_WasapiDeviceEventHandler::OnDeviceUpdated(DeviceWatcher ^ sender, DeviceInformationUpdate ^ args)
{
SDL_assert(sender == this->watcher);
}
void
SDL_WasapiDeviceEventHandler::OnEnumerationCompleted(DeviceWatcher^ sender, Platform::Object^ args)
void SDL_WasapiDeviceEventHandler::OnEnumerationCompleted(DeviceWatcher ^ sender, Platform::Object ^ args)
{
SDL_assert(sender == this->watcher);
SDL_SemPost(this->completed);
}
void
SDL_WasapiDeviceEventHandler::OnDefaultRenderDeviceChanged(Platform::Object^ sender, DefaultAudioRenderDeviceChangedEventArgs^ args)
void SDL_WasapiDeviceEventHandler::OnDefaultRenderDeviceChanged(Platform::Object ^ sender, DefaultAudioRenderDeviceChangedEventArgs ^ args)
{
SDL_assert(this->iscapture);
SDL_AtomicAdd(&SDL_IMMDevice_DefaultPlaybackGeneration, 1);
}
void
SDL_WasapiDeviceEventHandler::OnDefaultCaptureDeviceChanged(Platform::Object^ sender, DefaultAudioCaptureDeviceChangedEventArgs^ args)
void SDL_WasapiDeviceEventHandler::OnDefaultCaptureDeviceChanged(Platform::Object ^ sender, DefaultAudioCaptureDeviceChangedEventArgs ^ args)
{
SDL_assert(!this->iscapture);
SDL_AtomicAdd(&SDL_IMMDevice_DefaultCaptureGeneration, 1);
}
static SDL_WasapiDeviceEventHandler *playback_device_event_handler;
static SDL_WasapiDeviceEventHandler *capture_device_event_handler;
@@ -238,10 +229,11 @@ void WASAPI_EnumerateEndpoints(void)
SDL_SemWait(capture_device_event_handler->completed);
}
struct SDL_WasapiActivationHandler : public RuntimeClass< RuntimeClassFlags< ClassicCom >, FtmBase, IActivateAudioInterfaceCompletionHandler >
struct SDL_WasapiActivationHandler : public RuntimeClass<RuntimeClassFlags<ClassicCom>, FtmBase, IActivateAudioInterfaceCompletionHandler>
{
SDL_WasapiActivationHandler() : device(nullptr) {}
STDMETHOD(ActivateCompleted)(IActivateAudioInterfaceAsyncOperation *operation);
STDMETHOD(ActivateCompleted)
(IActivateAudioInterfaceAsyncOperation *operation);
SDL_AudioDevice *device;
};
@@ -254,23 +246,20 @@ SDL_WasapiActivationHandler::ActivateCompleted(IActivateAudioInterfaceAsyncOpera
return S_OK;
}
void
WASAPI_PlatformDeleteActivationHandler(void *handler)
void WASAPI_PlatformDeleteActivationHandler(void *handler)
{
((SDL_WasapiActivationHandler *) handler)->Release();
((SDL_WasapiActivationHandler *)handler)->Release();
}
int
WASAPI_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
int WASAPI_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
{
return SDL_Unsupported();
}
int
WASAPI_ActivateDevice(_THIS, const SDL_bool isrecovery)
int WASAPI_ActivateDevice(_THIS, const SDL_bool isrecovery)
{
LPCWSTR devid = _this->hidden->devid;
Platform::String^ defdevid;
Platform::String ^ defdevid;
if (devid == nullptr) {
defdevid = _this->iscapture ? MediaDevice::GetDefaultAudioCaptureId(AudioDeviceRole::Default) : MediaDevice::GetDefaultAudioRenderId(AudioDeviceRole::Default);
@@ -286,11 +275,11 @@ WASAPI_ActivateDevice(_THIS, const SDL_bool isrecovery)
return SDL_SetError("Failed to allocate WASAPI activation handler");
}
handler.Get()->AddRef(); // we hold a reference after ComPtr destructs on return, causing a Release, and Release ourselves in WASAPI_PlatformDeleteActivationHandler(), etc.
handler.Get()->AddRef(); // we hold a reference after ComPtr destructs on return, causing a Release, and Release ourselves in WASAPI_PlatformDeleteActivationHandler(), etc.
handler.Get()->device = _this;
_this->hidden->activation_handler = handler.Get();
WASAPI_RefDevice(_this); /* completion handler will unref it. */
WASAPI_RefDevice(_this); /* completion handler will unref it. */
IActivateAudioInterfaceAsyncOperation *async = nullptr;
const HRESULT ret = ActivateAudioInterfaceAsync(devid, __uuidof(IAudioClient), nullptr, handler.Get(), &async);
@@ -337,22 +326,20 @@ WASAPI_ActivateDevice(_THIS, const SDL_bool isrecovery)
return 0;
}
void
WASAPI_PlatformThreadInit(_THIS)
void WASAPI_PlatformThreadInit(_THIS)
{
// !!! FIXME: set this thread to "Pro Audio" priority.
}
void
WASAPI_PlatformThreadDeinit(_THIS)
void WASAPI_PlatformThreadDeinit(_THIS)
{
// !!! FIXME: set this thread to "Pro Audio" priority.
}
/* Everything below was copied from SDL_wasapi.c, before it got moved to SDL_immdevice.c! */
static const GUID SDL_KSDATAFORMAT_SUBTYPE_PCM = { 0x00000001, 0x0000, 0x0010,{ 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
static const GUID SDL_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = { 0x00000003, 0x0000, 0x0010,{ 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
static const GUID SDL_KSDATAFORMAT_SUBTYPE_PCM = { 0x00000001, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
static const GUID SDL_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = { 0x00000003, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
extern "C" SDL_AudioFormat
WaveFormatToSDLFormat(WAVEFORMATEX *waveformat)
@@ -376,8 +363,7 @@ WaveFormatToSDLFormat(WAVEFORMATEX *waveformat)
return 0;
}
static void
WASAPI_RemoveDevice(const SDL_bool iscapture, LPCWSTR devid)
static void WASAPI_RemoveDevice(const SDL_bool iscapture, LPCWSTR devid)
{
DevIdList *i;
DevIdList *next;
@@ -399,8 +385,7 @@ WASAPI_RemoveDevice(const SDL_bool iscapture, LPCWSTR devid)
}
}
static void
WASAPI_AddDevice(const SDL_bool iscapture, const char *devname, WAVEFORMATEXTENSIBLE *fmt, LPCWSTR devid)
static void WASAPI_AddDevice(const SDL_bool iscapture, const char *devname, WAVEFORMATEXTENSIBLE *fmt, LPCWSTR devid)
{
DevIdList *devidlist;
SDL_AudioSpec spec;
@@ -410,22 +395,22 @@ WASAPI_AddDevice(const SDL_bool iscapture, const char *devname, WAVEFORMATEXTENS
phones and tablets, where you might have an internal speaker and a headphone jack and expect both to be
available and switch automatically. (!!! FIXME...?) */
/* see if we already have this one. */
/* see if we already have this one. */
for (devidlist = deviceid_list; devidlist; devidlist = devidlist->next) {
if (SDL_wcscmp(devidlist->str, devid) == 0) {
return; /* we already have this. */
return; /* we already have this. */
}
}
devidlist = (DevIdList *)SDL_malloc(sizeof(*devidlist));
if (devidlist == NULL) {
return; /* oh well. */
return; /* oh well. */
}
devid = SDL_wcsdup(devid);
if (!devid) {
SDL_free(devidlist);
return; /* oh well. */
return; /* oh well. */
}
devidlist->str = (WCHAR *)devid;
@@ -439,6 +424,6 @@ WASAPI_AddDevice(const SDL_bool iscapture, const char *devname, WAVEFORMATEXTENS
SDL_AddAudioDevice(iscapture, devname, &spec, (void *)devid);
}
#endif // SDL_AUDIO_DRIVER_WASAPI && defined(__WINRT__)
#endif // SDL_AUDIO_DRIVER_WASAPI && defined(__WINRT__)
/* vi: set ts=4 sw=4 expandtab: */