/*

Simple DirectMedia Layer

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_config.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;

extern AudioBootStrap NAS_bootstrap;

extern AudioBootStrap PAUDIO_bootstrap;

extern AudioBootStrap BEOSAUDIO_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;

/* 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_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_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_BEOSAUDIO

&BEOSAUDIO_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_NDS

&NDSAUD_bootstrap,

#endif

#if SDL_AUDIO_DRIVER_FUSIONSOUND

&FUSIONSOUND_bootstrap,

#endif

#if SDL_AUDIO_DRIVER_ANDROID

&ANDROIDAUD_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 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 0;

}

static void

SDL_AudioLockDevice_Default(SDL_AudioDevice * device)

{

if (device->thread && (SDL_ThreadID() == device->threadid)) {

return;

}

SDL_mutexP(device->mixer_lock);

}

static void

SDL_AudioUnlockDevice_Default(SDL_AudioDevice * device)

{

if (device->thread && (SDL_ThreadID() == device->threadid)) {

return;

}

SDL_mutexV(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(GetDeviceBuf);

FILL_STUB(WaitDone);

FILL_STUB(CloseDevice);

FILL_STUB(LockDevice);

FILL_STUB(UnlockDevice);

FILL_STUB(Deinitialize);

#undef FILL_STUB

}

/* 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)

{

if (stream->buffer != NULL) {

SDL_free(stream->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;

/* For streaming when the buffer sizes don't match up */

Uint8 *istream;

int istream_len = 0;

/* 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; */

stream_len = device->spec.size;

} 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 (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_mutexP(device->mixer_lock);

(*fill) (udata, istream, istream_len);

SDL_mutexV(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 {

/* Otherwise, do not use the streamer. This is the old code. */

/* Loop, filling the audio buffers */

while (device->enabled) {

if (device->paused) {

SDL_Delay(delay);

continue;

}

/* 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_mutexP(device->mixer_lock);

(*fill) (udata, stream, stream_len);

SDL_mutexV(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 (device->use_streamer == 1)

SDL_StreamDeinit(&device->streamer);

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_strcasecmp(backend->name, driver_name))) ||

((!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]);

}

}

if (*devices != NULL) {

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) {

}

if ((iscapture) && (current_audio.impl.OnlyHasDefaultInputDevice)) {

return DEFAULT_INPUT_DEVNAME;

}

if ((!iscapture) && (current_audio.impl.OnlyHasDefaultOutputDevice)) {

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);

}

if (device->fake_stream != NULL) {

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;

}

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->callback == NULL) {

SDL_SetError("SDL_OpenAudio() passed a NULL callback");

return 0;

}

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_memset(device, '\0', sizeof(SDL_AudioDevice));

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)) {

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;

}

/* If the audio driver changes the buffer size, accept it */

if (device->spec.samples != obtained->samples) {

obtained->samples = device->spec.samples;

SDL_CalculateAudioSpec(obtained);

}

/* 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 (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) obtained->size) /

device->convert.len_ratio);

device->convert.buf =