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SDL/src/audio/SDL_audio.c
Ryan C. Gordon e5d49c2033 Added a GetPendingBytes method to the audio backend. 
This will (eventually) make SDL_GetQueuedAudioSize() more accurate, and thus
reduce latency. Right now this isn't implemented anywhere, so we assume data
fed to the audio callback is consumed by the hardware and immediately played
to completion.
2014-07-30 11:08:31 -04:00

1522 lines
45 KiB
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/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 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_audiomem.h"
#include "SDL_sysaudio.h"
#define _THIS SDL_AudioDevice *_this
static SDL_AudioDriver current_audio;
static SDL_AudioDevice *open_devices[16];
/* !!! FIXME: These are wordy and unlocalized... */
#define DEFAULT_OUTPUT_DEVNAME "System audio output device"
#define DEFAULT_INPUT_DEVNAME "System audio capture device"
/*
* Not all of these will be compiled and linked in, but it's convenient
* to have a complete list here and saves yet-another block of #ifdefs...
* Please see bootstrap[], below, for the actual #ifdef mess.
*/
extern AudioBootStrap BSD_AUDIO_bootstrap;
extern AudioBootStrap DSP_bootstrap;
extern AudioBootStrap ALSA_bootstrap;
extern AudioBootStrap PULSEAUDIO_bootstrap;
extern AudioBootStrap QSAAUDIO_bootstrap;
extern AudioBootStrap SUNAUDIO_bootstrap;
extern AudioBootStrap ARTS_bootstrap;
extern AudioBootStrap ESD_bootstrap;
#if SDL_AUDIO_DRIVER_NACL
extern AudioBootStrap NACLAUD_bootstrap;
#endif
extern AudioBootStrap NAS_bootstrap;
extern AudioBootStrap XAUDIO2_bootstrap;
extern AudioBootStrap DSOUND_bootstrap;
extern AudioBootStrap WINMM_bootstrap;
extern AudioBootStrap PAUDIO_bootstrap;
extern AudioBootStrap HAIKUAUDIO_bootstrap;
extern AudioBootStrap COREAUDIO_bootstrap;
extern AudioBootStrap SNDMGR_bootstrap;
extern AudioBootStrap DISKAUD_bootstrap;
extern AudioBootStrap DUMMYAUD_bootstrap;
extern AudioBootStrap DCAUD_bootstrap;
extern AudioBootStrap DART_bootstrap;
extern AudioBootStrap NDSAUD_bootstrap;
extern AudioBootStrap FUSIONSOUND_bootstrap;
extern AudioBootStrap ANDROIDAUD_bootstrap;
extern AudioBootStrap PSPAUD_bootstrap;
extern AudioBootStrap SNDIO_bootstrap;
/* 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_BSD
&BSD_AUDIO_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
&NACLAUD_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_NAS
&NAS_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_XAUDIO2
&XAUDIO2_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_DISK
&DISKAUD_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_DUMMY
&DUMMYAUD_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_FUSIONSOUND
&FUSIONSOUND_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_ANDROID
&ANDROIDAUD_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_PSP
&PSPAUD_bootstrap,
#endif
NULL
};
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(int iscapture, SDL_AddAudioDevice addfn)
{ /* no-op. */
}
static void
SDL_AudioThreadInit_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 void
SDL_AudioWaitDone_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioCloseDevice_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioDeinitialize_Default(void)
{ /* no-op. */
}
static int
SDL_AudioOpenDevice_Default(_THIS, const char *devname, int iscapture)
{
return -1;
}
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
finalize_audio_entry_points(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; \
}
FILL_STUB(DetectDevices);
FILL_STUB(OpenDevice);
FILL_STUB(ThreadInit);
FILL_STUB(WaitDevice);
FILL_STUB(PlayDevice);
FILL_STUB(GetPendingBytes);
FILL_STUB(GetDeviceBuf);
FILL_STUB(WaitDone);
FILL_STUB(CloseDevice);
FILL_STUB(LockDevice);
FILL_STUB(UnlockDevice);
FILL_STUB(Deinitialize);
#undef FILL_STUB
}
#if 0 /* !!! FIXME: rewrite/remove this streamer code. */
/* Streaming functions (for when the input and output buffer sizes are different) */
/* Write [length] bytes from buf into the streamer */
static void
SDL_StreamWrite(SDL_AudioStreamer * stream, Uint8 * buf, int length)
{
int i;
for (i = 0; i < length; ++i) {
stream->buffer[stream->write_pos] = buf[i];
++stream->write_pos;
}
}
/* Read [length] bytes out of the streamer into buf */
static void
SDL_StreamRead(SDL_AudioStreamer * stream, Uint8 * buf, int length)
{
int i;
for (i = 0; i < length; ++i) {
buf[i] = stream->buffer[stream->read_pos];
++stream->read_pos;
}
}
static int
SDL_StreamLength(SDL_AudioStreamer * stream)
{
return (stream->write_pos - stream->read_pos) % stream->max_len;
}
/* Initialize the stream by allocating the buffer and setting the read/write heads to the beginning */
#if 0
static int
SDL_StreamInit(SDL_AudioStreamer * stream, int max_len, Uint8 silence)
{
/* First try to allocate the buffer */
stream->buffer = (Uint8 *) SDL_malloc(max_len);
if (stream->buffer == NULL) {
return -1;
}
stream->max_len = max_len;
stream->read_pos = 0;
stream->write_pos = 0;
/* Zero out the buffer */
SDL_memset(stream->buffer, silence, max_len);
return 0;
}
#endif
/* Deinitialize the stream simply by freeing the buffer */
static void
SDL_StreamDeinit(SDL_AudioStreamer * stream)
{
SDL_free(stream->buffer);
}
#endif
/* buffer queueing support... */
/* this expects that you managed thread safety elsewhere. */
static void
free_audio_queue(SDL_AudioBufferQueue *buffer)
{
while (buffer) {
SDL_AudioBufferQueue *next = buffer->next;
SDL_free(buffer);
buffer = next;
}
}
static void SDLCALL
SDL_BufferQueueDrainCallback(void *userdata, Uint8 *stream, int _len)
{
/* this function always holds the mixer lock before being called. */
Uint32 len = (Uint32) _len;
SDL_AudioDevice *device = (SDL_AudioDevice *) userdata;
SDL_AudioBufferQueue *buffer;
SDL_assert(device != NULL); /* this shouldn't ever happen, right?! */
SDL_assert(_len >= 0); /* this shouldn't ever happen, right?! */
while ((len > 0) && ((buffer = device->buffer_queue_head) != NULL)) {
const Uint32 avail = buffer->datalen - buffer->startpos;
const Uint32 cpy = SDL_min(len, avail);
SDL_assert(device->queued_bytes >= avail);
SDL_memcpy(stream, buffer->data + buffer->startpos, cpy);
buffer->startpos += cpy;
stream += cpy;
device->queued_bytes -= cpy;
len -= cpy;
if (buffer->startpos == buffer->datalen) { /* packet is done, put it in the pool. */
device->buffer_queue_head = buffer->next;
SDL_assert((buffer->next != NULL) || (buffer == device->buffer_queue_tail));
buffer->next = device->buffer_queue_pool;
device->buffer_queue_pool = buffer;
}
}
SDL_assert((device->buffer_queue_head != NULL) == (device->queued_bytes != 0));
if (len > 0) { /* fill any remaining space in the stream with silence. */
SDL_assert(device->buffer_queue_head == NULL);
SDL_memset(stream, device->spec.silence, len);
}
if (device->buffer_queue_head == NULL) {
device->buffer_queue_tail = NULL; /* in case we drained the queue entirely. */
}
}
int
SDL_QueueAudio(SDL_AudioDeviceID devid, const void *_data, Uint32 len)
{
SDL_AudioDevice *device = get_audio_device(devid);
const Uint8 *data = (const Uint8 *) _data;
SDL_AudioBufferQueue *orighead;
SDL_AudioBufferQueue *origtail;
Uint32 origlen;
Uint32 datalen;
if (!device) {
return -1; /* get_audio_device() will have set the error state */
}
if (device->spec.callback != SDL_BufferQueueDrainCallback) {
return SDL_SetError("Audio device has a callback, queueing not allowed");
}
current_audio.impl.LockDevice(device);
orighead = device->buffer_queue_head;
origtail = device->buffer_queue_tail;
origlen = origtail ? origtail->datalen : 0;
while (len > 0) {
SDL_AudioBufferQueue *packet = device->buffer_queue_tail;
SDL_assert(!packet || (packet->datalen <= SDL_AUDIOBUFFERQUEUE_PACKETLEN));
if (!packet || (packet->datalen >= SDL_AUDIOBUFFERQUEUE_PACKETLEN)) {
/* tail packet missing or completely full; we need a new packet. */
packet = device->buffer_queue_pool;
if (packet != NULL) {
/* we have one available in the pool. */
device->buffer_queue_pool = packet->next;
} else {
/* Have to allocate a new one! */
packet = (SDL_AudioBufferQueue *) SDL_malloc(sizeof (SDL_AudioBufferQueue));
if (packet == NULL) {
/* uhoh, reset so we've queued nothing new, free what we can. */
if (!origtail) {
packet = device->buffer_queue_head; /* whole queue. */
} else {
packet = origtail->next; /* what we added to existing queue. */
origtail->next = NULL;
origtail->datalen = origlen;
}
device->buffer_queue_head = orighead;
device->buffer_queue_tail = origtail;
device->buffer_queue_pool = NULL;
current_audio.impl.UnlockDevice(device);
free_audio_queue(packet); /* give back what we can. */
return SDL_OutOfMemory();
}
}
packet->datalen = 0;
packet->startpos = 0;
packet->next = NULL;
SDL_assert((device->buffer_queue_head != NULL) == (device->queued_bytes != 0));
if (device->buffer_queue_tail == NULL) {
device->buffer_queue_head = packet;
} else {
device->buffer_queue_tail->next = packet;
}
device->buffer_queue_tail = packet;
}
datalen = SDL_min(len, SDL_AUDIOBUFFERQUEUE_PACKETLEN - packet->datalen);
SDL_memcpy(packet->data + packet->datalen, data, datalen);
data += datalen;
len -= datalen;
packet->datalen += datalen;
device->queued_bytes += datalen;
}
current_audio.impl.UnlockDevice(device);
return 0;
}
Uint32
SDL_GetQueuedAudioSize(SDL_AudioDeviceID devid)
{
/* this happens to work for non-queueing devices, since we memset()
the device to zero at init time, and these devices should return 0. */
Uint32 retval = 0;
SDL_AudioDevice *device = get_audio_device(devid);
if (device) {
current_audio.impl.LockDevice(device);
retval = device->queued_bytes + current_audio.impl.GetPendingBytes(device);
current_audio.impl.UnlockDevice(device);
}
return retval;
}
void
SDL_ClearQueuedAudio(SDL_AudioDeviceID devid)
{
SDL_AudioDevice *device = get_audio_device(devid);
SDL_AudioBufferQueue *buffer = NULL;
if (!device) {
return; /* nothing to do. */
}
/* Blank out the device and release the mutex. Free it afterwards. */
current_audio.impl.LockDevice(device);
buffer = device->buffer_queue_head;
device->buffer_queue_tail = NULL;
device->buffer_queue_head = NULL;
device->queued_bytes = 0;
current_audio.impl.UnlockDevice(device);
free_audio_queue(buffer);
}
#if defined(__ANDROID__)
#include <android/log.h>
#endif
/* The general mixing thread function */
int SDLCALL
SDL_RunAudio(void *devicep)
{
SDL_AudioDevice *device = (SDL_AudioDevice *) devicep;
Uint8 *stream;
int stream_len;
void *udata;
void (SDLCALL * fill) (void *userdata, Uint8 * stream, int len);
Uint32 delay;
#if 0 /* !!! FIXME: rewrite/remove this streamer code. */
/* For streaming when the buffer sizes don't match up */
Uint8 *istream;
int istream_len = 0;
#endif
/* 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);
/* Set up the mixing function */
fill = device->spec.callback;
udata = device->spec.userdata;
/* By default do not stream */
device->use_streamer = 0;
if (device->convert.needed) {
#if 0 /* !!! FIXME: I took len_div out of the structure. Use rate_incr instead? */
/* If the result of the conversion alters the length, i.e. resampling is being used, use the streamer */
if (device->convert.len_mult != 1 || device->convert.len_div != 1) {
/* The streamer's maximum length should be twice whichever is larger: spec.size or len_cvt */
stream_max_len = 2 * device->spec.size;
if (device->convert.len_mult > device->convert.len_div) {
stream_max_len *= device->convert.len_mult;
stream_max_len /= device->convert.len_div;
}
if (SDL_StreamInit(&device->streamer, stream_max_len, silence) <
0)
return -1;
device->use_streamer = 1;
/* istream_len should be the length of what we grab from the callback and feed to conversion,
so that we get close to spec_size. I.e. we want device.spec_size = istream_len * u / d
*/
istream_len =
device->spec.size * device->convert.len_div /
device->convert.len_mult;
}
#endif
stream_len = device->convert.len;
} else {
stream_len = device->spec.size;
}
/* Calculate the delay while paused */
delay = ((device->spec.samples * 1000) / device->spec.freq);
/* Determine if the streamer is necessary here */
#if 0 /* !!! FIXME: rewrite/remove this streamer code. */
if (device->use_streamer == 1) {
/* This code is almost the same as the old code. The difference is, instead of reading
directly from the callback into "stream", then converting and sending the audio off,
we go: callback -> "istream" -> (conversion) -> streamer -> stream -> device.
However, reading and writing with streamer are done separately:
- We only call the callback and write to the streamer when the streamer does not
contain enough samples to output to the device.
- We only read from the streamer and tell the device to play when the streamer
does have enough samples to output.
This allows us to perform resampling in the conversion step, where the output of the
resampling process can be any number. We will have to see what a good size for the
stream's maximum length is, but I suspect 2*max(len_cvt, stream_len) is a good figure.
*/
while (device->enabled) {
if (device->paused) {
SDL_Delay(delay);
continue;
}
/* Only read in audio if the streamer doesn't have enough already (if it does not have enough samples to output) */
if (SDL_StreamLength(&device->streamer) < stream_len) {
/* Set up istream */
if (device->convert.needed) {
if (device->convert.buf) {
istream = device->convert.buf;
} else {
continue;
}
} else {
/* FIXME: Ryan, this is probably wrong. I imagine we don't want to get
* a device buffer both here and below in the stream output.
*/
istream = current_audio.impl.GetDeviceBuf(device);
if (istream == NULL) {
istream = device->fake_stream;
}
}
/* Read from the callback into the _input_ stream */
SDL_LockMutex(device->mixer_lock);
(*fill) (udata, istream, istream_len);
SDL_UnlockMutex(device->mixer_lock);
/* Convert the audio if necessary and write to the streamer */
if (device->convert.needed) {
SDL_ConvertAudio(&device->convert);
if (istream == NULL) {
istream = device->fake_stream;
}
/* SDL_memcpy(istream, device->convert.buf, device->convert.len_cvt); */
SDL_StreamWrite(&device->streamer, device->convert.buf,
device->convert.len_cvt);
} else {
SDL_StreamWrite(&device->streamer, istream, istream_len);
}
}
/* Only output audio if the streamer has enough to output */
if (SDL_StreamLength(&device->streamer) >= stream_len) {
/* Set up the output stream */
if (device->convert.needed) {
if (device->convert.buf) {
stream = device->convert.buf;
} else {
continue;
}
} else {
stream = current_audio.impl.GetDeviceBuf(device);
if (stream == NULL) {
stream = device->fake_stream;
}
}
/* Now read from the streamer */
SDL_StreamRead(&device->streamer, stream, stream_len);
/* Ready current buffer for play and change current buffer */
if (stream != device->fake_stream) {
current_audio.impl.PlayDevice(device);
/* Wait for an audio buffer to become available */
current_audio.impl.WaitDevice(device);
} else {
SDL_Delay(delay);
}
}
}
} else
#endif
{
/* Otherwise, do not use the streamer. This is the old code. */
const int silence = (int) device->spec.silence;
/* Loop, filling the audio buffers */
while (device->enabled) {
/* Fill the current buffer with sound */
if (device->convert.needed) {
if (device->convert.buf) {
stream = device->convert.buf;
} else {
continue;
}
} else {
stream = current_audio.impl.GetDeviceBuf(device);
if (stream == NULL) {
stream = device->fake_stream;
}
}
SDL_LockMutex(device->mixer_lock);
if (device->paused) {
SDL_memset(stream, silence, stream_len);
} else {
(*fill) (udata, stream, stream_len);
}
SDL_UnlockMutex(device->mixer_lock);
/* Convert the audio if necessary */
if (device->convert.needed) {
SDL_ConvertAudio(&device->convert);
stream = current_audio.impl.GetDeviceBuf(device);
if (stream == NULL) {
stream = device->fake_stream;
}
SDL_memcpy(stream, device->convert.buf,
device->convert.len_cvt);
}
/* Ready current buffer for play and change current buffer */
if (stream != device->fake_stream) {
current_audio.impl.PlayDevice(device);
/* Wait for an audio buffer to become available */
current_audio.impl.WaitDevice(device);
} else {
SDL_Delay(delay);
}
}
}
/* Wait for the audio to drain.. */
current_audio.impl.WaitDone(device);
/* If necessary, deinit the streamer */
#if 0 /* !!! FIXME: rewrite/remove this streamer code. */
if (device->use_streamer == 1)
SDL_StreamDeinit(&device->streamer);
#endif
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_memset(&current_audio, '\0', sizeof(current_audio));
SDL_memset(open_devices, '\0', sizeof(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_memset(&current_audio, 0, sizeof(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_memset(&current_audio, 0, sizeof(current_audio));
return (-1); /* No driver was available, so fail. */
}
finalize_audio_entry_points();
return (0);
}
/*
* Get the current audio driver name
*/
const char *
SDL_GetCurrentAudioDriver()
{
return current_audio.name;
}
static void
free_device_list(char ***devices, int *devCount)
{
int i = *devCount;
if ((i > 0) && (*devices != NULL)) {
while (i--) {
SDL_free((*devices)[i]);
}
}
SDL_free(*devices);
*devices = NULL;
*devCount = 0;
}
static
void SDL_AddCaptureAudioDevice(const char *_name)
{
char *name = NULL;
void *ptr = SDL_realloc(current_audio.inputDevices,
(current_audio.inputDeviceCount+1) * sizeof(char*));
if (ptr == NULL) {
return; /* oh well. */
}
current_audio.inputDevices = (char **) ptr;
name = SDL_strdup(_name); /* if this returns NULL, that's okay. */
current_audio.inputDevices[current_audio.inputDeviceCount++] = name;
}
static
void SDL_AddOutputAudioDevice(const char *_name)
{
char *name = NULL;
void *ptr = SDL_realloc(current_audio.outputDevices,
(current_audio.outputDeviceCount+1) * sizeof(char*));
if (ptr == NULL) {
return; /* oh well. */
}
current_audio.outputDevices = (char **) ptr;
name = SDL_strdup(_name); /* if this returns NULL, that's okay. */
current_audio.outputDevices[current_audio.outputDeviceCount++] = name;
}
int
SDL_GetNumAudioDevices(int iscapture)
{
int retval = 0;
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
return -1;
}
if ((iscapture) && (!current_audio.impl.HasCaptureSupport)) {
return 0;
}
if ((iscapture) && (current_audio.impl.OnlyHasDefaultInputDevice)) {
return 1;
}
if ((!iscapture) && (current_audio.impl.OnlyHasDefaultOutputDevice)) {
return 1;
}
if (iscapture) {
free_device_list(&current_audio.inputDevices,
&current_audio.inputDeviceCount);
current_audio.impl.DetectDevices(iscapture, SDL_AddCaptureAudioDevice);
retval = current_audio.inputDeviceCount;
} else {
free_device_list(&current_audio.outputDevices,
&current_audio.outputDeviceCount);
current_audio.impl.DetectDevices(iscapture, SDL_AddOutputAudioDevice);
retval = current_audio.outputDeviceCount;
}
return retval;
}
const char *
SDL_GetAudioDeviceName(int index, int iscapture)
{
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) {
goto no_such_device;
}
if ((iscapture) && (current_audio.impl.OnlyHasDefaultInputDevice)) {
if (index > 0) {
goto no_such_device;
}
return DEFAULT_INPUT_DEVNAME;
}
if ((!iscapture) && (current_audio.impl.OnlyHasDefaultOutputDevice)) {
if (index > 0) {
goto no_such_device;
}
return DEFAULT_OUTPUT_DEVNAME;
}
if (iscapture) {
if (index >= current_audio.inputDeviceCount) {
goto no_such_device;
}
return current_audio.inputDevices[index];
} else {
if (index >= current_audio.outputDeviceCount) {
goto no_such_device;
}
return current_audio.outputDevices[index];
}
no_such_device:
SDL_SetError("No such device");
return NULL;
}
static void
close_audio_device(SDL_AudioDevice * device)
{
device->enabled = 0;
if (device->thread != NULL) {
SDL_WaitThread(device->thread, NULL);
}
if (device->mixer_lock != NULL) {
SDL_DestroyMutex(device->mixer_lock);
}
SDL_FreeAudioMem(device->fake_stream);
if (device->convert.needed) {
SDL_FreeAudioMem(device->convert.buf);
}
if (device->opened) {
current_audio.impl.CloseDevice(device);
device->opened = 0;
}
free_audio_queue(device->buffer_queue_head);
free_audio_queue(device->buffer_queue_pool);
SDL_FreeAudioMem(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)
{
SDL_AudioDeviceID id = 0;
SDL_AudioSpec _obtained;
SDL_AudioDevice *device;
SDL_bool build_cvt;
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;
}
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.OnlyHasDefaultInputDevice)) {
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;
}
}
}
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;
}
}
}
device = (SDL_AudioDevice *) SDL_AllocAudioMem(sizeof(SDL_AudioDevice));
if (device == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_zerop(device);
device->spec = *obtained;
device->enabled = 1;
device->paused = 1;
device->iscapture = iscapture;
/* Create a semaphore 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;
}
}
/* force a device detection if we haven't done one yet. */
if ( ((iscapture) && (current_audio.inputDevices == NULL)) ||
((!iscapture) && (current_audio.outputDevices == NULL)) ) {
SDL_GetNumAudioDevices(iscapture);
}
if (current_audio.impl.OpenDevice(device, devname, iscapture) < 0) {
close_audio_device(device);
return 0;
}
device->opened = 1;
/* Allocate a fake audio memory buffer */
device->fake_stream = (Uint8 *)SDL_AllocAudioMem(device->spec.size);
if (device->fake_stream == NULL) {
close_audio_device(device);
SDL_OutOfMemory();
return 0;
}
/* See if we need to do any conversion */
build_cvt = SDL_FALSE;
if (obtained->freq != device->spec.freq) {
if (allowed_changes & SDL_AUDIO_ALLOW_FREQUENCY_CHANGE) {
obtained->freq = device->spec.freq;
} else {
build_cvt = SDL_TRUE;
}
}
if (obtained->format != device->spec.format) {
if (allowed_changes & SDL_AUDIO_ALLOW_FORMAT_CHANGE) {
obtained->format = device->spec.format;
} else {
build_cvt = SDL_TRUE;
}
}
if (obtained->channels != device->spec.channels) {
if (allowed_changes & SDL_AUDIO_ALLOW_CHANNELS_CHANGE) {
obtained->channels = device->spec.channels;
} else {
build_cvt = SDL_TRUE;
}
}
/* If the audio driver changes the buffer size, accept it.
This needs to be done after the format is modified above,
otherwise it might not have the correct buffer size.
*/
if (device->spec.samples != obtained->samples) {
obtained->samples = device->spec.samples;
SDL_CalculateAudioSpec(obtained);
}
if (build_cvt) {
/* Build an audio conversion block */
if (SDL_BuildAudioCVT(&device->convert,
obtained->format, obtained->channels,
obtained->freq,
device->spec.format, device->spec.channels,
device->spec.freq) < 0) {
close_audio_device(device);
return 0;
}
if (device->convert.needed) {
device->convert.len = (int) (((double) device->spec.size) /
device->convert.len_ratio);
device->convert.buf =
(Uint8 *) SDL_AllocAudioMem(device->convert.len *
device->convert.len_mult);
if (device->convert.buf == NULL) {
close_audio_device(device);
SDL_OutOfMemory();
return 0;
}
}
}
if (device->spec.callback == NULL) { /* use buffer queueing? */
/* pool a few packets to start. Enough for two callbacks. */
const int packetlen = SDL_AUDIOBUFFERQUEUE_PACKETLEN;
const int wantbytes = ((device->convert.needed) ? device->convert.len : device->spec.size) * 2;
const int wantpackets = (wantbytes / packetlen) + ((wantbytes % packetlen) ? packetlen : 0);
for (i = 0; i < wantpackets; i++) {
SDL_AudioBufferQueue *packet = (SDL_AudioBufferQueue *) SDL_malloc(sizeof (SDL_AudioBufferQueue));
if (packet) { /* don't care if this fails, we'll deal later. */
packet->datalen = 0;
packet->startpos = 0;
packet->next = device->buffer_queue_pool;
device->buffer_queue_pool = packet;
}
}
device->spec.callback = SDL_BufferQueueDrainCallback;
device->spec.userdata = device;
}
/* Find an available device ID and store the structure... */
for (id = min_id - 1; id < SDL_arraysize(open_devices); id++) {
if (open_devices[id] == NULL) {
open_devices[id] = device;
break;
}
}
if (id == SDL_arraysize(open_devices)) {
SDL_SetError("Too many open audio devices");
close_audio_device(device);
return 0;
}
/* Start the audio thread if necessary */
if (!current_audio.impl.ProvidesOwnCallbackThread) {
/* Start the audio thread */
char name[64];
SDL_snprintf(name, sizeof (name), "SDLAudioDev%d", (int) (id + 1));
/* !!! FIXME: this is nasty. */
#if defined(__WIN32__) && !defined(HAVE_LIBC)
#undef SDL_CreateThread
#if SDL_DYNAMIC_API
device->thread = SDL_CreateThread_REAL(SDL_RunAudio, name, device, NULL, NULL);
#else
device->thread = SDL_CreateThread(SDL_RunAudio, name, device, NULL, NULL);
#endif
#else
device->thread = SDL_CreateThread(SDL_RunAudio, name, device);
#endif
if (device->thread == NULL) {
SDL_CloseAudioDevice(id + 1);
SDL_SetError("Couldn't create audio thread");
return 0;
}
}
return id + 1;
}
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 {
id = open_audio_device(NULL, 0, desired, NULL, 0, 1);
}
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 && device->enabled) {
if (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);
device->paused = pause_on;
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)
{
SDL_AudioDevice *device = get_audio_device(devid);
if (device) {
close_audio_device(device);
open_devices[devid - 1] = NULL;
}
}
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++) {
if (open_devices[i] != NULL) {
SDL_CloseAudioDevice(i+1);
}
}
/* Free the driver data */
current_audio.impl.Deinitialize();
free_device_list(&current_audio.outputDevices,
&current_audio.outputDeviceCount);
free_device_list(&current_audio.inputDevices,
&current_audio.inputDeviceCount);
SDL_memset(&current_audio, '\0', sizeof(current_audio));
SDL_memset(open_devices, '\0', sizeof(open_devices));
}
#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_AudioFormat format;
if (device->convert.needed) {
format = device->convert.src_format;
} else {
format = device->spec.format;
}
SDL_MixAudioFormat(dst, src, format, len, volume);
}
}
/* vi: set ts=4 sw=4 expandtab: */
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