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SDL_audio.c
1187 lines (1031 loc) · 34 KB
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/*
SDL - Simple DirectMedia Layer
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Copyright (C) 1997-2006 Sam Lantinga
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This library is free software; you can redistribute it and/or
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modify it under the terms of the GNU Lesser General Public
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License as published by the Free Software Foundation; either
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version 2.1 of the License, or (at your option) any later version.
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This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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Lesser General Public License for more details.
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You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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Sam Lantinga
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slouken@libsdl.org
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*/
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#include "SDL_config.h"
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/* Allow access to a raw mixing buffer */
#include "SDL.h"
#include "SDL_audio_c.h"
#include "SDL_audiomem.h"
#include "SDL_sysaudio.h"
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#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 DMA_bootstrap;
extern AudioBootStrap ALSA_bootstrap;
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extern AudioBootStrap PULSEAUDIO_bootstrap;
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extern AudioBootStrap QNXNTOAUDIO_bootstrap;
extern AudioBootStrap SUNAUDIO_bootstrap;
extern AudioBootStrap DMEDIA_bootstrap;
extern AudioBootStrap ARTS_bootstrap;
extern AudioBootStrap ESD_bootstrap;
extern AudioBootStrap NAS_bootstrap;
extern AudioBootStrap DSOUND_bootstrap;
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extern AudioBootStrap WINWAVEOUT_bootstrap;
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extern AudioBootStrap PAUDIO_bootstrap;
extern AudioBootStrap BEOSAUDIO_bootstrap;
extern AudioBootStrap COREAUDIO_bootstrap;
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extern AudioBootStrap COREAUDIOIPHONE_bootstrap;
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extern AudioBootStrap SNDMGR_bootstrap;
extern AudioBootStrap MINTAUDIO_GSXB_bootstrap;
extern AudioBootStrap MINTAUDIO_MCSN_bootstrap;
extern AudioBootStrap MINTAUDIO_STFA_bootstrap;
extern AudioBootStrap MINTAUDIO_XBIOS_bootstrap;
extern AudioBootStrap MINTAUDIO_DMA8_bootstrap;
extern AudioBootStrap DISKAUD_bootstrap;
extern AudioBootStrap DUMMYAUD_bootstrap;
extern AudioBootStrap DCAUD_bootstrap;
extern AudioBootStrap MMEAUDIO_bootstrap;
extern AudioBootStrap DART_bootstrap;
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extern AudioBootStrap NDSAUD_bootstrap;
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/* Available audio drivers */
static AudioBootStrap *bootstrap[] = {
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#if SDL_AUDIO_DRIVER_BSD
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&BSD_AUDIO_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_OSS
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&DSP_bootstrap,
&DMA_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_ALSA
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&ALSA_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_PULSEAUDIO
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&PULSEAUDIO_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_QNXNTO
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&QNXNTOAUDIO_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_SUNAUDIO
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&SUNAUDIO_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_DMEDIA
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&DMEDIA_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_ARTS
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&ARTS_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_ESD
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&ESD_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_NAS
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&NAS_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_DSOUND
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&DSOUND_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_WINWAVEOUT
&WINWAVEOUT_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_PAUDIO
&PAUDIO_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_BEOSAUDIO
&BEOSAUDIO_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_COREAUDIO
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&COREAUDIO_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_COREAUDIOIPHONE
&COREAUDIOIPHONE_bootstrap,
#endif
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#if SDL_AUDIO_DRIVER_SNDMGR
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&SNDMGR_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_MINT
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&MINTAUDIO_GSXB_bootstrap,
&MINTAUDIO_MCSN_bootstrap,
&MINTAUDIO_STFA_bootstrap,
&MINTAUDIO_XBIOS_bootstrap,
&MINTAUDIO_DMA8_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_DISK
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&DISKAUD_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_DUMMY
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&DUMMYAUD_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_DC
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&DCAUD_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_MMEAUDIO
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&MMEAUDIO_bootstrap,
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#endif
#if SDL_AUDIO_DRIVER_DART
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&DART_bootstrap,
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#endif
#if SDL_AUDIO_DRIVER_NDS
&NDSAUD_bootstrap,
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#endif
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NULL
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};
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static SDL_AudioDevice *
get_audio_device(SDL_AudioDeviceID id)
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{
id--;
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if ((id >= SDL_arraysize(open_devices)) || (open_devices[id] == NULL)) {
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SDL_SetError("Invalid audio device ID");
return NULL;
}
return open_devices[id];
}
/* stubs for audio drivers that don't need a specific entry point... */
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static int
SDL_AudioDetectDevices_Default(int iscapture)
{
return -1;
}
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static void
SDL_AudioThreadInit_Default(_THIS)
{ /* no-op. */
}
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static void
SDL_AudioWaitDevice_Default(_THIS)
{ /* no-op. */
}
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static void
SDL_AudioPlayDevice_Default(_THIS)
{ /* no-op. */
}
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static Uint8 *
SDL_AudioGetDeviceBuf_Default(_THIS)
{
return NULL;
}
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static void
SDL_AudioWaitDone_Default(_THIS)
{ /* no-op. */
}
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static void
SDL_AudioCloseDevice_Default(_THIS)
{ /* no-op. */
}
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static void
SDL_AudioDeinitialize_Default(void)
{ /* no-op. */
}
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static int
SDL_AudioOpenDevice_Default(_THIS, const char *devname, int iscapture)
{
return 0;
}
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static const char *
SDL_AudioGetDeviceName_Default(int index, int iscapture)
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{
SDL_SetError("No such device");
return NULL;
}
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);
}
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static void
finalize_audio_entry_points(void)
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{
/*
* Fill in stub functions for unused driver entry points. This lets us
* blindly call them without having to check for validity first.
*/
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#define FILL_STUB(x) \
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if (current_audio.impl.x == NULL) { \
current_audio.impl.x = SDL_Audio##x##_Default; \
}
FILL_STUB(DetectDevices);
FILL_STUB(GetDeviceName);
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);
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#undef FILL_STUB
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}
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/* Streaming functions (for when the input and output buffer sizes are different) */
/* Write [length] bytes from buf into the streamer */
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 */
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;
}
}
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 */
int
SDL_StreamInit(SDL_AudioStreamer * stream, int max_len, Uint8 silence)
{
/* First try to allocate the buffer */
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stream->buffer = (Uint8 *) SDL_malloc(max_len);
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if (stream->buffer == NULL) {
return -1;
}
stream->max_len = max_len;
stream->read_pos = 0;
stream->write_pos = 0;
/* Zero out the buffer */
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SDL_memset(stream->buffer, silence, max_len);
return 0;
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}
/* Deinitialize the stream simply by freeing the buffer */
void
SDL_StreamDeinit(SDL_AudioStreamer * stream)
{
if (stream->buffer != NULL) {
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SDL_free(stream->buffer);
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}
}
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/* The general mixing thread function */
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int SDLCALL
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SDL_RunAudio(void *devicep)
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{
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SDL_AudioDevice *device = (SDL_AudioDevice *) devicep;
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Uint8 *stream;
int stream_len;
void *udata;
void (SDLCALL * fill) (void *userdata, Uint8 * stream, int len);
int silence;
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int stream_max_len;
/* For streaming when the buffer sizes don't match up */
Uint8 *istream;
int istream_len;
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/* Perform any thread setup */
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device->threadid = SDL_ThreadID();
current_audio.impl.ThreadInit(device);
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/* Set up the mixing function */
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fill = device->spec.callback;
udata = device->spec.userdata;
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/* By default do not stream */
device->use_streamer = 0;
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if (device->convert.needed) {
if (device->convert.src_format == AUDIO_U8) {
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silence = 0x80;
} else {
silence = 0;
}
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/* 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;
}
/* stream_len = device->convert.len; */
stream_len = device->spec.size;
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} else {
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silence = device->spec.silence;
stream_len = device->spec.size;
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}
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/* 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 reding
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) {
/* 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 {
istream = current_audio.impl.GetDeviceBuf(device);
if (istream == NULL) {
istream = device->fake_stream;
}
}
/* Read from the callback into the _input_ stream */
if (!device->paused) {
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);
}
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}
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/* 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 */
if (stream == device->fake_stream) {
SDL_Delay((device->spec.samples * 1000) /
device->spec.freq);
} else {
current_audio.impl.WaitDevice(device);
}
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}
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}
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} else {
/* Otherwise, do not use the streamer. This is the old code. */
/* 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;
}
}
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if (!device->paused) {
SDL_mutexP(device->mixer_lock);
(*fill) (udata, stream, stream_len);
SDL_mutexV(device->mixer_lock);
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}
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/* 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);
}
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/* 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 */
if (stream == device->fake_stream) {
SDL_Delay((device->spec.samples * 1000) / device->spec.freq);
} else {
current_audio.impl.WaitDevice(device);
}
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}
}
/* Wait for the audio to drain.. */
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current_audio.impl.WaitDone(device);
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/* If necessary, deinit the streamer */
if (device->use_streamer == 1)
SDL_StreamDeinit(&device->streamer);
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return (0);
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}
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static SDL_AudioFormat
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SDL_ParseAudioFormat(const char *string)
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{
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#define CHECK_FMT_STRING(x) if (SDL_strcmp(string, #x) == 0) return AUDIO_##x
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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);
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#undef CHECK_FMT_STRING
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return 0;
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}
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int
SDL_GetNumAudioDrivers(void)
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{
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return (SDL_arraysize(bootstrap) - 1);
}
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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)
{
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int i = 0;
int initialized = 0;
int tried_to_init = 0;
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if (SDL_WasInit(SDL_INIT_AUDIO)) {
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SDL_AudioQuit(); /* shutdown driver if already running. */
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}
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SDL_memset(¤t_audio, '\0', sizeof(current_audio));
SDL_memset(open_devices, '\0', sizeof(open_devices));
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/* Select the proper audio driver */
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if (driver_name == NULL) {
driver_name = SDL_getenv("SDL_AUDIODRIVER");
}
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for (i = 0; (!initialized) && (bootstrap[i]); ++i) {
/* make sure we should even try this driver before doing so... */
const AudioBootStrap *backend = bootstrap[i];
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if (((driver_name) && (SDL_strcasecmp(backend->name, driver_name))) ||
((!driver_name) && (backend->demand_only))) {
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continue;
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}
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tried_to_init = 1;
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SDL_memset(¤t_audio, 0, sizeof(current_audio));
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current_audio.name = backend->name;
current_audio.desc = backend->desc;
initialized = backend->init(¤t_audio.impl);
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}
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if (!initialized) {
/* specific drivers will set the error message if they fail... */
if (!tried_to_init) {
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if (driver_name) {
SDL_SetError("%s not available", driver_name);
} else {
SDL_SetError("No available audio device");
}
}
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SDL_memset(¤t_audio, 0, sizeof(current_audio));
return (-1); /* No driver was available, so fail. */
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}
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finalize_audio_entry_points();
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return (0);
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}
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/*
* Get the current audio driver name
*/
const char *
SDL_GetCurrentAudioDriver()
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{
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return current_audio.name;
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}
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int
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SDL_GetNumAudioDevices(int iscapture)
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{
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if (!SDL_WasInit(SDL_INIT_AUDIO)) {
return -1;
}
if ((iscapture) && (!current_audio.impl.HasCaptureSupport)) {
return 0;
}
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if ((iscapture) && (current_audio.impl.OnlyHasDefaultInputDevice)) {
return 1;
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}
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if ((!iscapture) && (current_audio.impl.OnlyHasDefaultOutputDevice)) {
return 1;
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}
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return current_audio.impl.DetectDevices(iscapture);
}
const char *
SDL_GetAudioDeviceName(int index, int iscapture)
{
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
SDL_SetError("Audio subsystem is not initialized");
return NULL;
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}
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if ((iscapture) && (!current_audio.impl.HasCaptureSupport)) {
SDL_SetError("No capture support");
return NULL;
682
}
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
if (index < 0) {
SDL_SetError("No such device");
return NULL;
}
if ((iscapture) && (current_audio.impl.OnlyHasDefaultInputDevice)) {
return DEFAULT_INPUT_DEVNAME;
}
if ((!iscapture) && (current_audio.impl.OnlyHasDefaultOutputDevice)) {
return DEFAULT_OUTPUT_DEVNAME;
}
return current_audio.impl.GetDeviceName(index, iscapture);
}
static void
702
close_audio_device(SDL_AudioDevice * device)
703
704
705
706
707
708
709
{
device->enabled = 0;
if (device->thread != NULL) {
SDL_WaitThread(device->thread, NULL);
}
if (device->mixer_lock != NULL) {
SDL_DestroyMutex(device->mixer_lock);
710
}
711
712
if (device->fake_stream != NULL) {
SDL_FreeAudioMem(device->fake_stream);
713
}
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
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
731
prepare_audiospec(const SDL_AudioSpec * orig, SDL_AudioSpec * prepared)
732
{
733
SDL_memcpy(prepared, orig, sizeof(SDL_AudioSpec));
734
735
736
737
738
739
740
741
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");
742
743
if ((!env) || ((prepared->freq = SDL_atoi(env)) == 0)) {
prepared->freq = 22050; /* a reasonable default */
744
745
}
}
746
747
748
749
if (orig->format == 0) {
const char *env = SDL_getenv("SDL_AUDIO_FORMAT");
if ((!env) || ((prepared->format = SDL_ParseAudioFormat(env)) == 0)) {
750
prepared->format = AUDIO_S16; /* a reasonable default */
751
752
753
754
}
}
switch (orig->channels) {
755
756
case 0:{
const char *env = SDL_getenv("SDL_AUDIO_CHANNELS");
757
if ((!env) || ((prepared->channels = (Uint8) SDL_atoi(env)) == 0)) {
758
759
760
prepared->channels = 2; /* a reasonable default */
}
break;
761
}
762
763
764
765
766
767
case 1: /* Mono */
case 2: /* Stereo */
case 4: /* surround */
case 6: /* surround with center and lfe */
break;
default:
768
769
SDL_SetError("Unsupported number of audio channels.");
return 0;
770
}
771
772
773
if (orig->samples == 0) {
const char *env = SDL_getenv("SDL_AUDIO_SAMPLES");
774
if ((!env) || ((prepared->samples = (Uint16) SDL_atoi(env)) == 0)) {
775
776
777
778
779
780
781
782
/* 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;
783
784
}
}
785
786
787
788
789
790
791
792
793
794
/* 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,
795
796
const SDL_AudioSpec * _desired, SDL_AudioSpec * obtained,
int min_id)
797
798
799
800
801
802
803
804
805
{
SDL_AudioDeviceID id = 0;
SDL_AudioSpec desired;
SDL_AudioDevice *device;
int i = 0;
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
SDL_SetError("Audio subsystem is not initialized");
return 0;
806
}
807
808
809
810
if ((iscapture) && (!current_audio.impl.HasCaptureSupport)) {
SDL_SetError("No capture support");
return 0;
811
}
812
813
814
815
if (!prepare_audiospec(_desired, &desired)) {
return 0;
}
816
817
818
819
820
/* If app doesn't care about a specific device, let the user override. */
if (devname == NULL) {
devname = SDL_getenv("SDL_AUDIO_DEVICE_NAME");
}
821
822
823
824
825
826
827
828
829
830
/*
* 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.
*/
831
832
833
834
835
836
837
if ((iscapture) && (current_audio.impl.OnlyHasDefaultInputDevice)) {
if ((devname) && (SDL_strcmp(devname, DEFAULT_INPUT_DEVNAME) != 0)) {
SDL_SetError("No such device");
return 0;
}
devname = NULL;
838
839
840
841
842
843
844
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;
}
}
845
846
}
847
848
849
850
851
852
if ((!iscapture) && (current_audio.impl.OnlyHasDefaultOutputDevice)) {
if ((devname) && (SDL_strcmp(devname, DEFAULT_OUTPUT_DEVNAME) != 0)) {
SDL_SetError("No such device");
return 0;
}
devname = NULL;
853
854
855
856
857
858
859
860
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;
}
}
}
861
862
device = (SDL_AudioDevice *) SDL_AllocAudioMem(sizeof(SDL_AudioDevice));
863
864
865
if (device == NULL) {
SDL_OutOfMemory();
return 0;
866
}
867
868
SDL_memset(device, '\0', sizeof(SDL_AudioDevice));
SDL_memcpy(&device->spec, &desired, sizeof(SDL_AudioSpec));
869
870
871
device->enabled = 1;
device->paused = 1;
device->iscapture = iscapture;
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
/* 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;
}
}
if (!current_audio.impl.OpenDevice(device, devname, iscapture)) {
close_audio_device(device);
return 0;
}
device->opened = 1;
888
889
/* If the audio driver changes the buffer size, accept it */
890
891
892
if (device->spec.samples != desired.samples) {
desired.samples = device->spec.samples;
SDL_CalculateAudioSpec(&device->spec);
893
894
895
}
/* Allocate a fake audio memory buffer */
896
897
898
device->fake_stream = SDL_AllocAudioMem(device->spec.size);
if (device->fake_stream == NULL) {
close_audio_device(device);
899
SDL_OutOfMemory();
900
return 0;
901
902
903
904
}
/* See if we need to do any conversion */
if (obtained != NULL) {
905
906
907
908
SDL_memcpy(obtained, &device->spec, sizeof(SDL_AudioSpec));
} else if (desired.freq != device->spec.freq ||
desired.format != device->spec.format ||
desired.channels != device->spec.channels) {
909
/* Build an audio conversion block */
910
911
912
913
914
915
916
if (SDL_BuildAudioCVT(&device->convert,
desired.format, desired.channels,
desired.freq,
device->spec.format, device->spec.channels,
device->spec.freq) < 0) {
close_audio_device(device);
return 0;
917
}
918
if (device->convert.needed) {
919
920
device->convert.len = (int) (((double) desired.size) /
device->convert.len_ratio);
921
922
923
924
925
926
device->convert.buf =
(Uint8 *) SDL_AllocAudioMem(device->convert.len *
device->convert.len_mult);
if (device->convert.buf == NULL) {
close_audio_device(device);
927
SDL_OutOfMemory();
928
return 0;
929
930
931
}
}
}
932
933
/* Find an available device ID and store the structure... */
934
for (id = min_id - 1; id < SDL_arraysize(open_devices); id++) {
935
936
937
938
939
940
941
942
943
944
945
946
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;
}
947
/* Start the audio thread if necessary */
948
if (!current_audio.impl.ProvidesOwnCallbackThread) {
949
/* Start the audio thread */
950
/* !!! FIXME: this is nasty. */
951
#if (defined(__WIN32__) && !defined(_WIN32_WCE)) && !defined(HAVE_LIBC)
952
#undef SDL_CreateThread
953
device->thread = SDL_CreateThread(SDL_RunAudio, device, NULL, NULL);
954
#else
955
device->thread = SDL_CreateThread(SDL_RunAudio, device);
956
#endif
957
if (device->thread == NULL) {
958
SDL_CloseAudioDevice(id + 1);
959
SDL_SetError("Couldn't create audio thread");
960
961
962
963
return 0;
}
}
964
return id + 1;
965
966
967
968
969
970
971
972
973
974
975
}
int
SDL_OpenAudio(const 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) {
976
977
return (-1);
}
978
}
979
980
981
982
983
/* 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);
984
}
985
986
id = open_audio_device(NULL, 0, desired, obtained, 1);
987
if (id > 1) { /* this should never happen in theory... */
988
SDL_CloseAudioDevice(id);
989
SDL_SetError("Internal error"); /* MUST be Device ID #1! */
990
991
return (-1);
}
992
993
return ((id == 0) ? -1 : 0);
994
995
}
996
997
SDL_AudioDeviceID
SDL_OpenAudioDevice(const char *device, int iscapture,
998
const SDL_AudioSpec * desired, SDL_AudioSpec * obtained)
999
{
1000
return open_audio_device(device, iscapture, desired, obtained, 2);