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sdl2_frt/src/audio/SDL_audio.c
Sam Lantinga e3cc5b2c6b Updated copyright for 2018
2018-01-03 10:03:25 -08:00

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/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../SDL_internal.h"
/* Allow access to a raw mixing buffer */
#include "SDL.h"
#include "SDL_audio.h"
#include "SDL_audio_c.h"
#include "SDL_sysaudio.h"
#include "../thread/SDL_systhread.h"
#define _THIS SDL_AudioDevice *_this
static SDL_AudioDriver current_audio;
static SDL_AudioDevice *open_devices[16];
/* Available audio drivers */
static const AudioBootStrap *const bootstrap[] = {
#if SDL_AUDIO_DRIVER_PULSEAUDIO
&PULSEAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_ALSA
&ALSA_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_SNDIO
&SNDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_NETBSD
&NETBSDAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_OSS
&DSP_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_QSA
&QSAAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_SUNAUDIO
&SUNAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_ARTS
&ARTS_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_ESD
&ESD_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_NACL
&NACLAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_NAS
&NAS_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_WASAPI
&WASAPI_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_DSOUND
&DSOUND_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_WINMM
&WINMM_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_PAUDIO
&PAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_HAIKU
&HAIKUAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_COREAUDIO
&COREAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_FUSIONSOUND
&FUSIONSOUND_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_ANDROID
&ANDROIDAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_PSP
&PSPAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_EMSCRIPTEN
&EMSCRIPTENAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_JACK
&JACK_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_DISK
&DISKAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_DUMMY
&DUMMYAUDIO_bootstrap,
#endif
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;
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;
static SDL_bool
LoadLibSampleRate(void)
{
const char *hint = SDL_GetHint(SDL_HINT_AUDIO_RESAMPLING_MODE);
SRC_available = SDL_FALSE;
SRC_converter = 0;
if (!hint || *hint == '0' || SDL_strcasecmp(hint, "default") == 0) {
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) {
SRC_converter = SRC_SINC_MEDIUM_QUALITY;
} 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. */
}
#ifdef SDL_LIBSAMPLERATE_DYNAMIC
SDL_assert(SRC_lib == NULL);
SRC_lib = SDL_LoadObject(SDL_LIBSAMPLERATE_DYNAMIC);
if (!SRC_lib) {
SDL_ClearError();
return SDL_FALSE;
}
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");
if (!SRC_src_new || !SRC_src_process || !SRC_src_reset || !SRC_src_delete || !SRC_src_strerror) {
SDL_UnloadObject(SRC_lib);
SRC_lib = NULL;
return SDL_FALSE;
}
#else
SRC_src_new = src_new;
SRC_src_process = src_process;
SRC_src_reset = src_reset;
SRC_src_delete = src_delete;
SRC_src_strerror = src_strerror;
#endif
SRC_available = SDL_TRUE;
return SDL_TRUE;
}
static void
UnloadLibSampleRate(void)
{
#ifdef SDL_LIBSAMPLERATE_DYNAMIC
if (SRC_lib != NULL) {
SDL_UnloadObject(SRC_lib);
}
SRC_lib = NULL;
#endif
SRC_available = SDL_FALSE;
SRC_src_new = NULL;
SRC_src_process = NULL;
SRC_src_reset = NULL;
SRC_src_delete = NULL;
SRC_src_strerror = NULL;
}
#endif
static SDL_AudioDevice *
get_audio_device(SDL_AudioDeviceID id)
{
id--;
if ((id >= SDL_arraysize(open_devices)) || (open_devices[id] == NULL)) {
SDL_SetError("Invalid audio device ID");
return NULL;
}
return open_devices[id];
}
/* stubs for audio drivers that don't need a specific entry point... */
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, (void *) ((size_t) 0x1));
if (current_audio.impl.HasCaptureSupport) {
SDL_AddAudioDevice(SDL_TRUE, DEFAULT_INPUT_DEVNAME, (void *) ((size_t) 0x2));
}
}
static void
SDL_AudioThreadInit_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioThreadDeinit_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioBeginLoopIteration_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioWaitDevice_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioPlayDevice_Default(_THIS)
{ /* no-op. */
}
static int
SDL_AudioGetPendingBytes_Default(_THIS)
{
return 0;
}
static Uint8 *
SDL_AudioGetDeviceBuf_Default(_THIS)
{
return NULL;
}
static int
SDL_AudioCaptureFromDevice_Default(_THIS, void *buffer, int buflen)
{
return -1; /* just fail immediately. */
}
static void
SDL_AudioFlushCapture_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioPrepareToClose_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioCloseDevice_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioDeinitialize_Default(void)
{ /* no-op. */
}
static void
SDL_AudioFreeDeviceHandle_Default(void *handle)
{ /* no-op. */
}
static int
SDL_AudioOpenDevice_Default(_THIS, void *handle, const char *devname, int iscapture)
{
return SDL_Unsupported();
}
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,
tries to lock the thread that we've already locked from the
device thread...just in case we only have non-recursive mutexes. */
if (device->thread && (SDL_ThreadID() == device->threadid)) {
return SDL_TRUE;
}
return SDL_FALSE;
}
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)
{
if (!is_in_audio_device_thread(device)) {
SDL_UnlockMutex(device->mixer_lock);
}
}
static void
SDL_AudioLockOrUnlockDeviceWithNoMixerLock(SDL_AudioDevice * device)
{
}
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.
*/
if (current_audio.impl.SkipMixerLock) {
if (current_audio.impl.LockDevice == NULL) {
current_audio.impl.LockDevice = SDL_AudioLockOrUnlockDeviceWithNoMixerLock;
}
if (current_audio.impl.UnlockDevice == NULL) {
current_audio.impl.UnlockDevice = SDL_AudioLockOrUnlockDeviceWithNoMixerLock;
}
}
#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);
FILL_STUB(ThreadDeinit);
FILL_STUB(BeginLoopIteration);
FILL_STUB(WaitDevice);
FILL_STUB(PlayDevice);
FILL_STUB(GetPendingBytes);
FILL_STUB(GetDeviceBuf);
FILL_STUB(CaptureFromDevice);
FILL_STUB(FlushCapture);
FILL_STUB(PrepareToClose);
FILL_STUB(CloseDevice);
FILL_STUB(LockDevice);
FILL_STUB(UnlockDevice);
FILL_STUB(FreeDeviceHandle);
FILL_STUB(Deinitialize);
#undef FILL_STUB
}
/* device hotplug support... */
static int
add_audio_device(const char *name, void *handle, SDL_AudioDeviceItem **devices, int *devCount)
{
int retval = -1;
const size_t size = sizeof (SDL_AudioDeviceItem) + SDL_strlen(name) + 1;
SDL_AudioDeviceItem *item = (SDL_AudioDeviceItem *) SDL_malloc(size);
if (item == NULL) {
return -1;
}
SDL_assert(handle != NULL); /* we reserve NULL, audio backends can't use it. */
item->handle = handle;
SDL_strlcpy(item->name, name, size - sizeof (SDL_AudioDeviceItem));
SDL_LockMutex(current_audio.detectionLock);
item->next = *devices;
*devices = item;
retval = (*devCount)++;
SDL_UnlockMutex(current_audio.detectionLock);
return retval;
}
static SDL_INLINE int
add_capture_device(const char *name, void *handle)
{
SDL_assert(current_audio.impl.HasCaptureSupport);
return add_audio_device(name, handle, &current_audio.inputDevices, &current_audio.inputDeviceCount);
}
static SDL_INLINE int
add_output_device(const char *name, void *handle)
{
return add_audio_device(name, handle, &current_audio.outputDevices, &current_audio.outputDeviceCount);
}
static void
free_device_list(SDL_AudioDeviceItem **devices, int *devCount)
{
SDL_AudioDeviceItem *item, *next;
for (item = *devices; item != NULL; item = next) {
next = item->next;
if (item->handle != NULL) {
current_audio.impl.FreeDeviceHandle(item->handle);
}
SDL_free(item);
}
*devices = NULL;
*devCount = 0;
}
/* The audio backends call this when a new device is plugged in. */
void
SDL_AddAudioDevice(const int iscapture, const char *name, void *handle)
{
const int device_index = iscapture ? add_capture_device(name, handle) : add_output_device(name, handle);
if (device_index != -1) {
/* Post the event, if desired */
if (SDL_GetEventState(SDL_AUDIODEVICEADDED) == SDL_ENABLE) {
SDL_Event event;
SDL_zero(event);
event.adevice.type = SDL_AUDIODEVICEADDED;
event.adevice.which = device_index;
event.adevice.iscapture = iscapture;
SDL_PushEvent(&event);
}
}
}
/* The audio backends call this when a currently-opened device is lost. */
void SDL_OpenedAudioDeviceDisconnected(SDL_AudioDevice *device)
{
SDL_assert(get_audio_device(device->id) == device);
if (!SDL_AtomicGet(&device->enabled)) {
return;
}
/* Ends the audio callback and mark the device as STOPPED, but the
app still needs to close the device to free resources. */
current_audio.impl.LockDevice(device);
SDL_AtomicSet(&device->enabled, 0);
current_audio.impl.UnlockDevice(device);
/* Post the event, if desired */
if (SDL_GetEventState(SDL_AUDIODEVICEREMOVED) == SDL_ENABLE) {
SDL_Event event;
SDL_zero(event);
event.adevice.type = SDL_AUDIODEVICEREMOVED;
event.adevice.which = device->id;
event.adevice.iscapture = device->iscapture ? 1 : 0;
SDL_PushEvent(&event);
}
}
static void
mark_device_removed(void *handle, SDL_AudioDeviceItem *devices, SDL_bool *removedFlag)
{
SDL_AudioDeviceItem *item;
SDL_assert(handle != NULL);
for (item = devices; item != NULL; item = item->next) {
if (item->handle == handle) {
item->handle = NULL;
*removedFlag = SDL_TRUE;
return;
}
}
}
/* The audio backends call this when a device is removed from the system. */
void
SDL_RemoveAudioDevice(const int iscapture, void *handle)
{
int device_index;
SDL_AudioDevice *device = NULL;
SDL_LockMutex(current_audio.detectionLock);
if (iscapture) {
mark_device_removed(handle, current_audio.inputDevices, &current_audio.captureDevicesRemoved);
} else {
mark_device_removed(handle, current_audio.outputDevices, &current_audio.outputDevicesRemoved);
}
for (device_index = 0; device_index < SDL_arraysize(open_devices); device_index++)
{
device = open_devices[device_index];
if (device != NULL && device->handle == handle)
{
SDL_OpenedAudioDeviceDisconnected(device);
break;
}
}
SDL_UnlockMutex(current_audio.detectionLock);
current_audio.impl.FreeDeviceHandle(handle);
}
/* buffer queueing support... */
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;
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?! */
dequeued = SDL_ReadFromDataQueue(device->buffer_queue, stream, len);
stream += dequeued;
len -= (int) dequeued;
if (len > 0) { /* fill any remaining space in the stream with silence. */
SDL_assert(SDL_CountDataQueue(device->buffer_queue) == 0);
SDL_memset(stream, device->spec.silence, 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_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
later, but you probably have bigger problems in this case anyhow. */
SDL_WriteToDataQueue(device->buffer_queue, stream, 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 */
} else if (device->iscapture) {
return SDL_SetError("This is a capture device, queueing not allowed");
} else if (device->callbackspec.callback != SDL_BufferQueueDrainCallback) {
return SDL_SetError("Audio device has a callback, queueing not allowed");
}
if (len > 0) {
current_audio.impl.LockDevice(device);
rc = SDL_WriteToDataQueue(device->buffer_queue, data, len);
current_audio.impl.UnlockDevice(device);
}
return rc;
}
Uint32
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. */
}
current_audio.impl.LockDevice(device);
rc = (Uint32) SDL_ReadFromDataQueue(device->buffer_queue, data, len);
current_audio.impl.UnlockDevice(device);
return rc;
}
Uint32
SDL_GetQueuedAudioSize(SDL_AudioDeviceID devid)
{
Uint32 retval = 0;
SDL_AudioDevice *device = get_audio_device(devid);
if (!device) {
return 0;
}
/* Nothing to do unless we're set up for queueing. */
if (device->callbackspec.callback == SDL_BufferQueueDrainCallback) {
current_audio.impl.LockDevice(device);
retval = ((Uint32) SDL_CountDataQueue(device->buffer_queue)) + current_audio.impl.GetPendingBytes(device);
current_audio.impl.UnlockDevice(device);
} else if (device->callbackspec.callback == SDL_BufferQueueFillCallback) {
current_audio.impl.LockDevice(device);
retval = (Uint32) SDL_CountDataQueue(device->buffer_queue);
current_audio.impl.UnlockDevice(device);
}
return retval;
}
void
SDL_ClearQueuedAudio(SDL_AudioDeviceID devid)
{
SDL_AudioDevice *device = get_audio_device(devid);
if (!device) {
return; /* nothing to do. */
}
/* Blank out the device and release the mutex. Free it afterwards. */
current_audio.impl.LockDevice(device);
/* Keep up to two packets in the pool to reduce future malloc pressure. */
SDL_ClearDataQueue(device->buffer_queue, SDL_AUDIOBUFFERQUEUE_PACKETLEN * 2);
current_audio.impl.UnlockDevice(device);
}
/* The general mixing thread function */
static int SDLCALL
SDL_RunAudio(void *devicep)
{
SDL_AudioDevice *device = (SDL_AudioDevice *) devicep;
void *udata = device->callbackspec.userdata;
SDL_AudioCallback callback = device->callbackspec.callback;
int data_len = 0;
Uint8 *data;
SDL_assert(!device->iscapture);
/* The audio mixing is always a high priority thread */
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_HIGH);
/* Perform any thread setup */
device->threadid = SDL_ThreadID();
current_audio.impl.ThreadInit(device);
/* Loop, filling the audio buffers */
while (!SDL_AtomicGet(&device->shutdown)) {
current_audio.impl.BeginLoopIteration(device);
data_len = device->callbackspec.size;
/* Fill the current buffer with sound */
if (!device->stream && SDL_AtomicGet(&device->enabled)) {
SDL_assert(data_len == device->spec.size);
data = current_audio.impl.GetDeviceBuf(device);
} else {
/* if the device isn't enabled, we still write to the
work_buffer, so the app's callback will fire with
a regular frequency, in case they depend on that
for timing or progress. They can use hotplug
now to know if the device failed.
Streaming playback uses work_buffer, too. */
data = NULL;
}
if (data == NULL) {
data = device->work_buffer;
}
/* !!! FIXME: this should be LockDevice. */
SDL_LockMutex(device->mixer_lock);
if (SDL_AtomicGet(&device->paused)) {
SDL_memset(data, device->spec.silence, data_len);
} else {
callback(udata, data, data_len);
}
SDL_UnlockMutex(device->mixer_lock);
if (device->stream) {
/* Stream available audio to device, converting/resampling. */
/* 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)) {
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... */
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? */
} else {
if (got != device->spec.size) {
SDL_memset(data, device->spec.silence, device->spec.size);
}
current_audio.impl.PlayDevice(device);
current_audio.impl.WaitDevice(device);
}
}
} else if (data == device->work_buffer) {
/* 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. */
/* queue this buffer and wait for it to finish playing. */
current_audio.impl.PlayDevice(device);
current_audio.impl.WaitDevice(device);
}
}
current_audio.impl.PrepareToClose(device);
/* Wait for the audio to drain. */
SDL_Delay(((device->spec.samples * 1000) / device->spec.freq) * 2);
current_audio.impl.ThreadDeinit(device);
return 0;
}
/* !!! FIXME: this needs to deal with device spec changes. */
/* The general capture thread function */
static int SDLCALL
SDL_CaptureAudio(void *devicep)
{
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;
void *udata = device->callbackspec.userdata;
SDL_AudioCallback callback = device->callbackspec.callback;
SDL_assert(device->iscapture);
/* The audio mixing is always a high priority thread */
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_HIGH);
/* Perform any thread setup */
device->threadid = SDL_ThreadID();
current_audio.impl.ThreadInit(device);
/* Loop, filling the audio buffers */
while (!SDL_AtomicGet(&device->shutdown)) {
int still_need;
Uint8 *ptr;
current_audio.impl.BeginLoopIteration(device);
if (SDL_AtomicGet(&device->paused)) {
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. */
continue;
}
/* Fill the current buffer with sound */
still_need = data_len;
/* Use the work_buffer to hold data read from the device. */
data = device->work_buffer;
SDL_assert(data != NULL);
ptr = data;
/* We still read from the device when "paused" to keep the state sane,
and block when there isn't data so this thread isn't eating CPU.
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. */
} 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. :) */
if (rc > 0) {
still_need -= rc;
ptr += rc;
} else { /* uhoh, device failed for some reason! */
SDL_OpenedAudioDeviceDisconnected(device);
break;
}
}
}
if (still_need > 0) {
/* Keep any data we already read, silence the rest. */
SDL_memset(ptr, silence, still_need);
}
if (device->stream) {
/* if this fails...oh well. */
SDL_AudioStreamPut(device->stream, data, data_len);
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) {
SDL_memset(device->work_buffer, device->spec.silence, device->callbackspec.size);
}
/* !!! FIXME: this should be LockDevice. */
SDL_LockMutex(device->mixer_lock);
if (!SDL_AtomicGet(&device->paused)) {
callback(udata, device->work_buffer, device->callbackspec.size);
}
SDL_UnlockMutex(device->mixer_lock);
}
} else { /* feeding user callback directly without streaming. */
/* !!! FIXME: this should be LockDevice. */
SDL_LockMutex(device->mixer_lock);
if (!SDL_AtomicGet(&device->paused)) {
callback(udata, data, device->callbackspec.size);
}
SDL_UnlockMutex(device->mixer_lock);
}
}
current_audio.impl.FlushCapture(device);
current_audio.impl.ThreadDeinit(device);
return 0;
}
static SDL_AudioFormat
SDL_ParseAudioFormat(const char *string)
{
#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);
CHECK_FMT_STRING(S16LSB);
CHECK_FMT_STRING(U16MSB);
CHECK_FMT_STRING(S16MSB);
CHECK_FMT_STRING(U16SYS);
CHECK_FMT_STRING(S16SYS);
CHECK_FMT_STRING(U16);
CHECK_FMT_STRING(S16);
CHECK_FMT_STRING(S32LSB);
CHECK_FMT_STRING(S32MSB);
CHECK_FMT_STRING(S32SYS);
CHECK_FMT_STRING(S32);
CHECK_FMT_STRING(F32LSB);
CHECK_FMT_STRING(F32MSB);
CHECK_FMT_STRING(F32SYS);
CHECK_FMT_STRING(F32);
#undef CHECK_FMT_STRING
return 0;
}
int
SDL_GetNumAudioDrivers(void)
{
return SDL_arraysize(bootstrap) - 1;
}
const char *
SDL_GetAudioDriver(int index)
{
if (index >= 0 && index < SDL_GetNumAudioDrivers()) {
return bootstrap[index]->name;
}
return NULL;
}
int
SDL_AudioInit(const char *driver_name)
{
int i = 0;
int initialized = 0;
int tried_to_init = 0;
if (SDL_WasInit(SDL_INIT_AUDIO)) {
SDL_AudioQuit(); /* shutdown driver if already running. */
}
SDL_zero(current_audio);
SDL_zero(open_devices);
/* Select the proper audio driver */
if (driver_name == NULL) {
driver_name = SDL_getenv("SDL_AUDIODRIVER");
}
for (i = 0; (!initialized) && (bootstrap[i]); ++i) {
/* make sure we should even try this driver before doing so... */
const AudioBootStrap *backend = bootstrap[i];
if ((driver_name && (SDL_strncasecmp(backend->name, driver_name, SDL_strlen(driver_name)) != 0)) ||
(!driver_name && backend->demand_only)) {
continue;
}
tried_to_init = 1;
SDL_zero(current_audio);
current_audio.name = backend->name;
current_audio.desc = backend->desc;
initialized = backend->init(&current_audio.impl);
}
if (!initialized) {
/* specific drivers will set the error message if they fail... */
if (!tried_to_init) {
if (driver_name) {
SDL_SetError("Audio target '%s' not available", driver_name);
} else {
SDL_SetError("No available audio device");
}
}
SDL_zero(current_audio);
return -1; /* No driver was available, so fail. */
}
current_audio.detectionLock = SDL_CreateMutex();
finish_audio_entry_points_init();
/* Make sure we have a list of devices available at startup. */
current_audio.impl.DetectDevices();
#ifdef HAVE_LIBSAMPLERATE_H
LoadLibSampleRate();
#endif
return 0;
}
/*
* Get the current audio driver name
*/
const char *
SDL_GetCurrentAudioDriver()
{
return current_audio.name;
}
/* 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)
{
SDL_AudioDeviceItem *item = *devices;
SDL_AudioDeviceItem *prev = NULL;
int total = 0;
while (item) {
SDL_AudioDeviceItem *next = item->next;
if (item->handle != NULL) {
total++;
prev = item;
} else {
if (prev) {
prev->next = next;
} else {
*devices = next;
}
SDL_free(item);
}
item = next;
}
*devCount = total;
*removedFlag = SDL_FALSE;
}
int
SDL_GetNumAudioDevices(int iscapture)
{
int retval = 0;
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
return -1;
}
SDL_LockMutex(current_audio.detectionLock);
if (iscapture && current_audio.captureDevicesRemoved) {
clean_out_device_list(&current_audio.inputDevices, &current_audio.inputDeviceCount, &current_audio.captureDevicesRemoved);
}
if (!iscapture && current_audio.outputDevicesRemoved) {
clean_out_device_list(&current_audio.outputDevices, &current_audio.outputDeviceCount, &current_audio.outputDevicesRemoved);
current_audio.outputDevicesRemoved = SDL_FALSE;
}
retval = iscapture ? current_audio.inputDeviceCount : current_audio.outputDeviceCount;
SDL_UnlockMutex(current_audio.detectionLock);
return retval;
}
const char *
SDL_GetAudioDeviceName(int index, int iscapture)
{
const char *retval = NULL;
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
SDL_SetError("Audio subsystem is not initialized");
return NULL;
}
if ((iscapture) && (!current_audio.impl.HasCaptureSupport)) {
SDL_SetError("No capture support");
return NULL;
}
if (index >= 0) {
SDL_AudioDeviceItem *item;
int i;
SDL_LockMutex(current_audio.detectionLock);
item = iscapture ? current_audio.inputDevices : current_audio.outputDevices;
i = iscapture ? current_audio.inputDeviceCount : current_audio.outputDeviceCount;
if (index < i) {
for (i--; i > index; i--, item = item->next) {
SDL_assert(item != NULL);
}
SDL_assert(item != NULL);
retval = item->name;
}
SDL_UnlockMutex(current_audio.detectionLock);
}
if (retval == NULL) {
SDL_SetError("No such device");
}
return retval;
}
static void
close_audio_device(SDL_AudioDevice * device)
{
if (!device) {
return;
}
if (device->id > 0) {
SDL_AudioDevice *opendev = open_devices[device->id - 1];
SDL_assert((opendev == device) || (opendev == NULL));
if (opendev == device) {
open_devices[device->id - 1] = NULL;
}
}
SDL_AtomicSet(&device->shutdown, 1);
SDL_AtomicSet(&device->enabled, 0);
if (device->thread != NULL) {
SDL_WaitThread(device->thread, NULL);
}
if (device->mixer_lock != NULL) {
SDL_DestroyMutex(device->mixer_lock);
}
SDL_free(device->work_buffer);
SDL_FreeAudioStream(device->stream);
if (device->hidden != NULL) {
current_audio.impl.CloseDevice(device);
}
SDL_FreeDataQueue(device->buffer_queue);
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)
{
SDL_memcpy(prepared, orig, sizeof(SDL_AudioSpec));
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 */
}
}
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 */
}
}
switch (orig->channels) {
case 0:{
const char *env = SDL_getenv("SDL_AUDIO_CHANNELS");
if ((!env) || ((prepared->channels = (Uint8) SDL_atoi(env)) == 0)) {
prepared->channels = 2; /* a reasonable default */
}
break;
}
case 1: /* Mono */
case 2: /* Stereo */
case 4: /* surround */
case 6: /* surround with center and lfe */
break;
default:
SDL_SetError("Unsupported number of audio channels.");
return 0;
}
if (orig->samples == 0) {
const char *env = SDL_getenv("SDL_AUDIO_SAMPLES");
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;
int power2 = 1;
while (power2 < samples) {
power2 *= 2;
}
prepared->samples = power2;
}
}
/* Calculate the silence and size of the audio specification */
SDL_CalculateAudioSpec(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)
{
const SDL_bool is_internal_thread = (desired->callback == NULL);
SDL_AudioDeviceID id = 0;
SDL_AudioSpec _obtained;
SDL_AudioDevice *device;
SDL_bool build_stream;
void *handle = NULL;
int i = 0;
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
SDL_SetError("Audio subsystem is not initialized");
return 0;
}
if ((iscapture) && (!current_audio.impl.HasCaptureSupport)) {
SDL_SetError("No capture support");
return 0;
}
/* !!! FIXME: there is a race condition here if two devices open from two threads at once. */
/* Find an available device ID... */
for (id = min_id - 1; id < SDL_arraysize(open_devices); id++) {
if (open_devices[id] == NULL) {
break;
}
}
if (id == SDL_arraysize(open_devices)) {
SDL_SetError("Too many open audio devices");
return 0;
}
if (!obtained) {
obtained = &_obtained;
}
if (!prepare_audiospec(desired, obtained)) {
return 0;
}
/* If app doesn't care about a specific device, let the user override. */
if (devname == NULL) {
devname = SDL_getenv("SDL_AUDIO_DEVICE_NAME");
}
/*
* Catch device names at the high level for the simple case...
* This lets us have a basic "device enumeration" for systems that
* don't have multiple devices, but makes sure the device name is
* always NULL when it hits the low level.
*
* Also make sure that the simple case prevents multiple simultaneous
* opens of the default system device.
*/
if ((iscapture) && (current_audio.impl.OnlyHasDefaultCaptureDevice)) {
if ((devname) && (SDL_strcmp(devname, DEFAULT_INPUT_DEVNAME) != 0)) {
SDL_SetError("No such device");
return 0;
}
devname = NULL;
for (i = 0; i < SDL_arraysize(open_devices); i++) {
if ((open_devices[i]) && (open_devices[i]->iscapture)) {
SDL_SetError("Audio device already open");
return 0;
}
}
} else if ((!iscapture) && (current_audio.impl.OnlyHasDefaultOutputDevice)) {
if ((devname) && (SDL_strcmp(devname, DEFAULT_OUTPUT_DEVNAME) != 0)) {
SDL_SetError("No such device");
return 0;
}
devname = NULL;
for (i = 0; i < SDL_arraysize(open_devices); i++) {
if ((open_devices[i]) && (!open_devices[i]->iscapture)) {
SDL_SetError("Audio device already open");
return 0;
}
}
} else if (devname != NULL) {
/* if the app specifies an exact string, we can pass the backend
an actual device handle thingey, which saves them the effort of
figuring out what device this was (such as, reenumerating
everything again to find the matching human-readable name).
It might still need to open a device based on the string for,
say, a network audio server, but this optimizes some cases. */
SDL_AudioDeviceItem *item;
SDL_LockMutex(current_audio.detectionLock);
for (item = iscapture ? current_audio.inputDevices : current_audio.outputDevices; item; item = item->next) {
if ((item->handle != NULL) && (SDL_strcmp(item->name, devname) == 0)) {
handle = item->handle;
break;
}
}
SDL_UnlockMutex(current_audio.detectionLock);
}
if (!current_audio.impl.AllowsArbitraryDeviceNames) {
/* has to be in our device list, or the default device. */
if ((handle == NULL) && (devname != NULL)) {
SDL_SetError("No such device.");
return 0;
}
}
device = (SDL_AudioDevice *) SDL_calloc(1, sizeof (SDL_AudioDevice));
if (device == NULL) {
SDL_OutOfMemory();
return 0;
}
device->id = id + 1;
device->spec = *obtained;
device->iscapture = iscapture ? SDL_TRUE : SDL_FALSE;
device->handle = handle;
SDL_AtomicSet(&device->shutdown, 0); /* just in case. */
SDL_AtomicSet(&device->paused, 1);
SDL_AtomicSet(&device->enabled, 1);
/* Create a mutex for locking the sound buffers */
if (!current_audio.impl.SkipMixerLock) {
device->mixer_lock = SDL_CreateMutex();
if (device->mixer_lock == NULL) {
close_audio_device(device);
SDL_SetError("Couldn't create mixer lock");
return 0;
}
}
if (current_audio.impl.OpenDevice(device, handle, devname, iscapture) < 0) {
close_audio_device(device);
return 0;
}
/* if your target really doesn't need it, set it to 0x1 or something. */
/* otherwise, close_audio_device() won't call impl.CloseDevice(). */
SDL_assert(device->hidden != NULL);
/* See if we need to do any conversion */
build_stream = SDL_FALSE;
if (obtained->freq != device->spec.freq) {
if (allowed_changes & SDL_AUDIO_ALLOW_FREQUENCY_CHANGE) {
obtained->freq = device->spec.freq;
} else {
build_stream = SDL_TRUE;
}
}
if (obtained->format != device->spec.format) {
if (allowed_changes & SDL_AUDIO_ALLOW_FORMAT_CHANGE) {
obtained->format = device->spec.format;
} else {
build_stream = SDL_TRUE;
}
}
if (obtained->channels != device->spec.channels) {
if (allowed_changes & SDL_AUDIO_ALLOW_CHANNELS_CHANGE) {
obtained->channels = device->spec.channels;
} else {
build_stream = SDL_TRUE;
}
}
/* !!! FIXME in 2.1: add SDL_AUDIO_ALLOW_SAMPLES_CHANGE flag?
As of 2.0.6, we will build a stream to buffer the difference between
what the app wants to feed and the device wants to eat, so everyone
gets their way. In prior releases, SDL would force the callback to
feed at the rate the device requested, adjusted for resampling.
*/
if (device->spec.samples != obtained->samples) {
build_stream = SDL_TRUE;
}
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);
} else {
device->stream = SDL_NewAudioStream(obtained->format, obtained->channels,
obtained->freq, device->spec.format,
device->spec.channels, device->spec.freq);
}
if (!device->stream) {
close_audio_device(device);
return 0;
}
}
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) {
close_audio_device(device);
SDL_SetError("Couldn't create audio buffer queue");
return 0;
}
device->callbackspec.callback = iscapture ? SDL_BufferQueueFillCallback : SDL_BufferQueueDrainCallback;
device->callbackspec.userdata = device;
}
/* Allocate a scratch audio buffer */
device->work_buffer_len = build_stream ? device->callbackspec.size : 0;
if (device->spec.size > device->work_buffer_len) {
device->work_buffer_len = device->spec.size;
}
SDL_assert(device->work_buffer_len > 0);
device->work_buffer = (Uint8 *) SDL_malloc(device->work_buffer_len);
if (device->work_buffer == NULL) {
close_audio_device(device);
SDL_OutOfMemory();
return 0;
}
open_devices[id] = device; /* add it to our list of open devices. */
/* Start the audio thread if necessary */
if (!current_audio.impl.ProvidesOwnCallbackThread) {
/* Start the audio thread */
/* !!! FIXME: we don't force the audio thread stack size here if it calls into user code, but maybe we should? */
/* buffer queueing callback only needs a few bytes, so make the stack tiny. */
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);
device->thread = SDL_CreateThreadInternal(iscapture ? SDL_CaptureAudio : SDL_RunAudio, threadname, stacksize, device);
if (device->thread == NULL) {
close_audio_device(device);
SDL_SetError("Couldn't create audio thread");
return 0;
}
}
return device->id;
}
int
SDL_OpenAudio(SDL_AudioSpec * desired, SDL_AudioSpec * obtained)
{
SDL_AudioDeviceID id = 0;
/* Start up the audio driver, if necessary. This is legacy behaviour! */
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) {
return -1;
}
}
/* SDL_OpenAudio() is legacy and can only act on Device ID #1. */
if (open_devices[0] != NULL) {
SDL_SetError("Audio device is already opened");
return -1;
}
if (obtained) {
id = open_audio_device(NULL, 0, desired, obtained,
SDL_AUDIO_ALLOW_ANY_CHANGE, 1);
} else {
SDL_AudioSpec _obtained;
SDL_zero(_obtained);
id = open_audio_device(NULL, 0, desired, &_obtained, 0, 1);
/* On successful open, copy calculated values into 'desired'. */
if (id > 0) {
desired->size = _obtained.size;
desired->silence = _obtained.silence;
}
}
SDL_assert((id == 0) || (id == 1));
return (id == 0) ? -1 : 0;
}
SDL_AudioDeviceID
SDL_OpenAudioDevice(const char *device, int iscapture,
const SDL_AudioSpec * desired, SDL_AudioSpec * obtained,
int allowed_changes)
{
return open_audio_device(device, iscapture, desired, obtained,
allowed_changes, 2);
}
SDL_AudioStatus
SDL_GetAudioDeviceStatus(SDL_AudioDeviceID devid)
{
SDL_AudioDevice *device = get_audio_device(devid);
SDL_AudioStatus status = SDL_AUDIO_STOPPED;
if (device && SDL_AtomicGet(&device->enabled)) {
if (SDL_AtomicGet(&device->paused)) {
status = SDL_AUDIO_PAUSED;
} else {
status = SDL_AUDIO_PLAYING;
}
}
return status;
}
SDL_AudioStatus
SDL_GetAudioStatus(void)
{
return SDL_GetAudioDeviceStatus(1);
}
void
SDL_PauseAudioDevice(SDL_AudioDeviceID devid, int pause_on)
{
SDL_AudioDevice *device = get_audio_device(devid);
if (device) {
current_audio.impl.LockDevice(device);
SDL_AtomicSet(&device->paused, pause_on ? 1 : 0);
current_audio.impl.UnlockDevice(device);
}
}
void
SDL_PauseAudio(int pause_on)
{
SDL_PauseAudioDevice(1, pause_on);
}
void
SDL_LockAudioDevice(SDL_AudioDeviceID devid)
{
/* Obtain a lock on the mixing buffers */
SDL_AudioDevice *device = get_audio_device(devid);
if (device) {
current_audio.impl.LockDevice(device);
}
}
void
SDL_LockAudio(void)
{
SDL_LockAudioDevice(1);
}
void
SDL_UnlockAudioDevice(SDL_AudioDeviceID devid)
{
/* Obtain a lock on the mixing buffers */
SDL_AudioDevice *device = get_audio_device(devid);
if (device) {
current_audio.impl.UnlockDevice(device);
}
}
void
SDL_UnlockAudio(void)
{
SDL_UnlockAudioDevice(1);
}
void
SDL_CloseAudioDevice(SDL_AudioDeviceID devid)
{
close_audio_device(get_audio_device(devid));
}
void
SDL_CloseAudio(void)
{
SDL_CloseAudioDevice(1);
}
void
SDL_AudioQuit(void)
{
SDL_AudioDeviceID i;
if (!current_audio.name) { /* not initialized?! */
return;
}
for (i = 0; i < SDL_arraysize(open_devices); i++) {
close_audio_device(open_devices[i]);
}
free_device_list(&current_audio.outputDevices, &current_audio.outputDeviceCount);
free_device_list(&current_audio.inputDevices, &current_audio.inputDeviceCount);
/* Free the driver data */
current_audio.impl.Deinitialize();
SDL_DestroyMutex(current_audio.detectionLock);
SDL_zero(current_audio);
SDL_zero(open_devices);
#ifdef HAVE_LIBSAMPLERATE_H
UnloadLibSampleRate();
#endif
SDL_FreeResampleFilter();
}
#define NUM_FORMATS 10
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},
};
SDL_AudioFormat
SDL_FirstAudioFormat(SDL_AudioFormat format)
{
for (format_idx = 0; format_idx < NUM_FORMATS; ++format_idx) {
if (format_list[format_idx][0] == format) {
break;
}
}
format_idx_sub = 0;
return SDL_NextAudioFormat();
}
SDL_AudioFormat
SDL_NextAudioFormat(void)
{
if ((format_idx == NUM_FORMATS) || (format_idx_sub == NUM_FORMATS)) {
return 0;
}
return format_list[format_idx][format_idx_sub++];
}
void
SDL_CalculateAudioSpec(SDL_AudioSpec * spec)
{
switch (spec->format) {
case AUDIO_U8:
spec->silence = 0x80;
break;
default:
spec->silence = 0x00;
break;
}
spec->size = SDL_AUDIO_BITSIZE(spec->format) / 8;
spec->size *= spec->channels;
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)
{
/* Mix the user-level audio format */
SDL_AudioDevice *device = get_audio_device(1);
if (device != NULL) {
SDL_MixAudioFormat(dst, src, device->callbackspec.format, len, volume);
}
}
/* vi: set ts=4 sw=4 expandtab: */
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