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SDL_audio.c
1209 lines (1051 loc) · 34.6 KB
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/*
SDL - Simple DirectMedia Layer
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Copyright (C) 1997-2011 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"
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#include "SDL_audio.h"
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#include "SDL_audio_c.h"
#include "SDL_audiomem.h"
#include "SDL_sysaudio.h"
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#define _THIS SDL_AudioDevice *_this
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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 QSAAUDIO_bootstrap;
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extern AudioBootStrap SUNAUDIO_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 DISKAUD_bootstrap;
extern AudioBootStrap DUMMYAUD_bootstrap;
extern AudioBootStrap DCAUD_bootstrap;
extern AudioBootStrap DART_bootstrap;
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extern AudioBootStrap NDSAUD_bootstrap;
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extern AudioBootStrap FUSIONSOUND_bootstrap;
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extern AudioBootStrap ANDROIDAUD_bootstrap;
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/* Available audio drivers */
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static const AudioBootStrap *const bootstrap[] = {
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#if SDL_AUDIO_DRIVER_PULSEAUDIO
&PULSEAUDIO_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_BSD
&BSD_AUDIO_bootstrap,
#endif
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#if SDL_AUDIO_DRIVER_OSS
&DSP_bootstrap,
&DMA_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_QSA
&QSAAUDIO_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_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_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_NDS
&NDSAUD_bootstrap,
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#endif
#if SDL_AUDIO_DRIVER_FUSIONSOUND
&FUSIONSOUND_bootstrap,
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#endif
#if SDL_AUDIO_DRIVER_ANDROID
&ANDROIDAUD_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 */
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static void
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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 */
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static void
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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;
}
}
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static int
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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 */
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#if 0
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static int
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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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}
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#endif
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/* Deinitialize the stream simply by freeing the buffer */
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static void
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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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#if defined(ANDROID)
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#include <android/log.h>
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#endif
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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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Uint32 delay;
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/* For streaming when the buffer sizes don't match up */
Uint8 *istream;
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int istream_len = 0;
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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 0 /* !!! FIXME: I took len_div out of the structure. Use rate_incr instead? */
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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;
}
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#endif
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/* 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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/* Calculate the delay while paused */
delay = ((device->spec.samples * 1000) / device->spec.freq);
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/* Determine if the streamer is necessary here */
if (device->use_streamer == 1) {
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/* This code is almost the same as the old code. The difference is, instead of reading
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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) {
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if (device->paused) {
SDL_Delay(delay);
continue;
}
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/* 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 {
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/* 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.
*/
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istream = current_audio.impl.GetDeviceBuf(device);
if (istream == NULL) {
istream = device->fake_stream;
}
}
/* Read from the callback into the _input_ stream */
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SDL_mutexP(device->mixer_lock);
(*fill) (udata, istream, istream_len);
SDL_mutexV(device->mixer_lock);
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/* 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 */
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if (stream != device->fake_stream) {
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current_audio.impl.PlayDevice(device);
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/* Wait for an audio buffer to become available */
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current_audio.impl.WaitDevice(device);
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} else {
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SDL_Delay(delay);
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}
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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) {
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if (device->paused) {
SDL_Delay(delay);
continue;
}
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/* 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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SDL_mutexP(device->mixer_lock);
(*fill) (udata, stream, stream_len);
SDL_mutexV(device->mixer_lock);
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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 */
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if (stream != device->fake_stream) {
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current_audio.impl.PlayDevice(device);
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/* Wait for an audio buffer to become available */
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current_audio.impl.WaitDevice(device);
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} else {
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SDL_Delay(delay);
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}
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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;
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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) {
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SDL_SetError("Audio target '%s' not available", driver_name);
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} 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;
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}
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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
695
close_audio_device(SDL_AudioDevice * device)
696
697
698
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700
701
702
{
device->enabled = 0;
if (device->thread != NULL) {
SDL_WaitThread(device->thread, NULL);
}
if (device->mixer_lock != NULL) {
SDL_DestroyMutex(device->mixer_lock);
703
}
704
705
if (device->fake_stream != NULL) {
SDL_FreeAudioMem(device->fake_stream);
706
}
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709
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711
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713
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716
717
718
719
720
721
722
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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
724
prepare_audiospec(const SDL_AudioSpec * orig, SDL_AudioSpec * prepared)
725
{
726
SDL_memcpy(prepared, orig, sizeof(SDL_AudioSpec));
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731
732
733
734
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");
735
736
if ((!env) || ((prepared->freq = SDL_atoi(env)) == 0)) {
prepared->freq = 22050; /* a reasonable default */
737
738
}
}
739
740
741
742
if (orig->format == 0) {
const char *env = SDL_getenv("SDL_AUDIO_FORMAT");
if ((!env) || ((prepared->format = SDL_ParseAudioFormat(env)) == 0)) {
743
prepared->format = AUDIO_S16; /* a reasonable default */
744
745
746
747
}
}
switch (orig->channels) {
748
749
case 0:{
const char *env = SDL_getenv("SDL_AUDIO_CHANNELS");
750
if ((!env) || ((prepared->channels = (Uint8) SDL_atoi(env)) == 0)) {
751
752
753
prepared->channels = 2; /* a reasonable default */
}
break;
754
}
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759
760
case 1: /* Mono */
case 2: /* Stereo */
case 4: /* surround */
case 6: /* surround with center and lfe */
break;
default:
761
762
SDL_SetError("Unsupported number of audio channels.");
return 0;
763
}
764
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766
if (orig->samples == 0) {
const char *env = SDL_getenv("SDL_AUDIO_SAMPLES");
767
if ((!env) || ((prepared->samples = (Uint16) SDL_atoi(env)) == 0)) {
768
769
770
771
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774
775
/* 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;
776
777
}
}
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784
785
786
787
/* 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,
788
789
const SDL_AudioSpec * desired, SDL_AudioSpec * obtained,
int allowed_changes, int min_id)
790
791
{
SDL_AudioDeviceID id = 0;
792
SDL_AudioSpec _obtained;
793
SDL_AudioDevice *device;
794
SDL_bool build_cvt;
795
796
797
798
799
int i = 0;
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
SDL_SetError("Audio subsystem is not initialized");
return 0;
800
}
801
802
803
804
if ((iscapture) && (!current_audio.impl.HasCaptureSupport)) {
SDL_SetError("No capture support");
return 0;
805
}
806
807
808
809
810
if (!obtained) {
obtained = &_obtained;
}
if (!prepare_audiospec(desired, obtained)) {
811
812
return 0;
}
813
814
815
816
817
/* If app doesn't care about a specific device, let the user override. */
if (devname == NULL) {
devname = SDL_getenv("SDL_AUDIO_DEVICE_NAME");
}
818
819
820
821
822
823
824
825
826
827
/*
* 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.
*/
828
829
830
831
832
833
834
if ((iscapture) && (current_audio.impl.OnlyHasDefaultInputDevice)) {
if ((devname) && (SDL_strcmp(devname, DEFAULT_INPUT_DEVNAME) != 0)) {
SDL_SetError("No such device");
return 0;
}
devname = NULL;
835
836
837
838
839
840
841
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;
}
}
842
843
}
844
845
846
847
848
849
if ((!iscapture) && (current_audio.impl.OnlyHasDefaultOutputDevice)) {
if ((devname) && (SDL_strcmp(devname, DEFAULT_OUTPUT_DEVNAME) != 0)) {
SDL_SetError("No such device");
return 0;
}
devname = NULL;
850
851
852
853
854
855
856
857
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;
}
}
}
858
859
device = (SDL_AudioDevice *) SDL_AllocAudioMem(sizeof(SDL_AudioDevice));
860
861
862
if (device == NULL) {
SDL_OutOfMemory();
return 0;
863
}
864
SDL_memset(device, '\0', sizeof(SDL_AudioDevice));
865
device->spec = *obtained;
866
867
868
device->enabled = 1;
device->paused = 1;
device->iscapture = iscapture;
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
/* 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;
885
886
/* Allocate a fake audio memory buffer */
887
device->fake_stream = (Uint8 *)SDL_AllocAudioMem(device->spec.size);
888
889
if (device->fake_stream == NULL) {
close_audio_device(device);
890
SDL_OutOfMemory();
891
return 0;
892
893
}
894
895
896
897
898
899
/* If the audio driver changes the buffer size, accept it */
if (device->spec.samples != obtained->samples) {
obtained->samples = device->spec.samples;
SDL_CalculateAudioSpec(obtained);
}
900
/* See if we need to do any conversion */
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
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) {
924
/* Build an audio conversion block */
925
if (SDL_BuildAudioCVT(&device->convert,
926
927
obtained->format, obtained->channels,
obtained->freq,
928
929
930
931
device->spec.format, device->spec.channels,
device->spec.freq) < 0) {
close_audio_device(device);
return 0;
932
}
933
if (device->convert.needed) {
934
device->convert.len = (int) (((double) obtained->size) /
935
device->convert.len_ratio);
936
937
938
939
940
941
device->convert.buf =
(Uint8 *) SDL_AllocAudioMem(device->convert.len *
device->convert.len_mult);
if (device->convert.buf == NULL) {
close_audio_device(device);
942
SDL_OutOfMemory();
943
return 0;
944
945
946
}
}
}
947
948
/* Find an available device ID and store the structure... */
949
for (id = min_id - 1; id < SDL_arraysize(open_devices); id++) {
950
951
952
953
954
955
956
957
958
959
960
961
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;
}
962
/* Start the audio thread if necessary */
963
if (!current_audio.impl.ProvidesOwnCallbackThread) {
964
/* Start the audio thread */
965
/* !!! FIXME: this is nasty. */
966
#if (defined(__WIN32__) && !defined(_WIN32_WCE)) && !defined(HAVE_LIBC)
967
#undef SDL_CreateThread
968
device->thread = SDL_CreateThread(SDL_RunAudio, device, NULL, NULL);
969
#else
970
device->thread = SDL_CreateThread(SDL_RunAudio, device);
971
#endif
972
if (device->thread == NULL) {
973
SDL_CloseAudioDevice(id + 1);
974
SDL_SetError("Couldn't create audio thread");
975
976
977
978
return 0;
}
}
979
return id + 1;
980
981
982
983
}
int
984
SDL_OpenAudio(SDL_AudioSpec * desired, SDL_AudioSpec * obtained)
985
986
987
988
989
990
{
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) {
991
992
return (-1);
}
993
}
994
995
996
997
998
/* 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);
999
}
1000