static int DCAUD_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
switch(spec->format&0xff) {
case 8: spec->format = AUDIO_S8; break;
case 16: spec->format = AUDIO_S16LSB; break;
default:
SDL_SetError("Unsupported audio format");
return(-1);
}
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(spec);
/* Allocate mixing buffer */
this->hidden->mixlen = spec->size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if ( this->hidden->mixbuf == NULL ) {
return(-1);
}
memset(this->hidden->mixbuf, spec->silence, spec->size);
this->hidden->leftpos = 0x11000;
this->hidden->rightpos = 0x11000+spec->size;
this->hidden->playing = 0;
this->hidden->nextbuf = 0;
/* We're ready to rock and roll. :-) */
return(0);
}
static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
SDL_MintAudio_device = this;
if (Mint_CheckAudio(this, spec)==-1) {
return -1;
}
SDL_CalculateAudioSpec(spec);
DEBUG_PRINT((DEBUG_NAME "buffer size=%d\n", spec->size));
SDL_MintAudio_audiobuf[0] = Atari_SysMalloc(spec->size *2, MX_STRAM);
if (SDL_MintAudio_audiobuf[0]==NULL) {
SDL_SetError("MINT_OpenAudio: Not enough memory for audio buffer");
return (-1);
}
SDL_MintAudio_audiobuf[1] = SDL_MintAudio_audiobuf[0] + spec->size ;
SDL_MintAudio_numbuf=0;
SDL_memset(SDL_MintAudio_audiobuf[0], spec->silence, spec->size *2);
SDL_MintAudio_audiosize = spec->size;
SDL_MintAudio_mutex = 0;
DEBUG_PRINT((DEBUG_NAME "buffer 0 at 0x%08x\n", SDL_MintAudio_audiobuf[0]));
DEBUG_PRINT((DEBUG_NAME "buffer 1 at 0x%08x\n", SDL_MintAudio_audiobuf[1]));
SDL_MintAudio_CheckFpu();
Mint_InitAudio(this, spec);
return(1);
}
static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
SDL_MintAudio_device = this;
/* Check audio capabilities */
if (Mint_CheckAudio(this, spec)==-1) {
return -1;
}
SDL_CalculateAudioSpec(spec);
/* Allocate memory for audio buffers in DMA-able RAM */
DEBUG_PRINT((DEBUG_NAME "buffer size=%d\n", spec->size));
SDL_MintAudio_audiobuf[0] = Atari_SysMalloc(spec->size *2, MX_STRAM);
if (SDL_MintAudio_audiobuf[0]==NULL) {
SDL_SetError("MINT_OpenAudio: Not enough memory for audio buffer");
return (-1);
}
SDL_MintAudio_audiobuf[1] = SDL_MintAudio_audiobuf[0] + spec->size ;
SDL_MintAudio_numbuf=0;
SDL_memset(SDL_MintAudio_audiobuf[0], spec->silence, spec->size *2);
SDL_MintAudio_audiosize = spec->size;
SDL_MintAudio_mutex = 0;
DEBUG_PRINT((DEBUG_NAME "buffer 0 at 0x%08x\n", SDL_MintAudio_audiobuf[0]));
DEBUG_PRINT((DEBUG_NAME "buffer 1 at 0x%08x\n", SDL_MintAudio_audiobuf[1]));
SDL_MintAudio_CheckFpu();
/* Setup audio hardware */
Mint_InitAudio(this, spec);
return(1); /* We don't use threaded audio */
}
static int
AndroidAUD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
SDL_AudioFormat test_format;
if (iscapture) {
/* TODO: implement capture */
return SDL_SetError("Capture not supported on Android");
}
if (audioDevice != NULL) {
return SDL_SetError("Only one audio device at a time please!");
}
audioDevice = this;
this->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, (sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
test_format = SDL_FirstAudioFormat(this->spec.format);
while (test_format != 0) { /* no "UNKNOWN" constant */
if ((test_format == AUDIO_U8) || (test_format == AUDIO_S16LSB)) {
this->spec.format = test_format;
break;
}
test_format = SDL_NextAudioFormat();
}
if (test_format == 0) {
/* Didn't find a compatible format :( */
return SDL_SetError("No compatible audio format!");
}
if (this->spec.channels > 1) {
this->spec.channels = 2;
} else {
this->spec.channels = 1;
}
if (this->spec.freq < 8000) {
this->spec.freq = 8000;
}
if (this->spec.freq > 48000) {
this->spec.freq = 48000;
}
/* TODO: pass in/return a (Java) device ID, also whether we're opening for input or output */
this->spec.samples = Android_JNI_OpenAudioDevice(this->spec.freq, this->spec.format == AUDIO_U8 ? 0 : 1, this->spec.channels, this->spec.samples);
SDL_CalculateAudioSpec(&this->spec);
if (this->spec.samples == 0) {
/* Init failed? */
return SDL_SetError("Java-side initialization failed!");
}
return 0;
}
static int
ANDROIDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
SDL_AudioFormat test_format;
SDL_assert((captureDevice == NULL) || !iscapture);
SDL_assert((audioDevice == NULL) || iscapture);
if (iscapture) {
captureDevice = this;
} else {
audioDevice = this;
}
this->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, (sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
test_format = SDL_FirstAudioFormat(this->spec.format);
while (test_format != 0) { /* no "UNKNOWN" constant */
if ((test_format == AUDIO_U8) || (test_format == AUDIO_S16LSB)) {
this->spec.format = test_format;
break;
}
test_format = SDL_NextAudioFormat();
}
if (test_format == 0) {
/* Didn't find a compatible format :( */
return SDL_SetError("No compatible audio format!");
}
if (this->spec.channels > 1) {
this->spec.channels = 2;
} else {
this->spec.channels = 1;
}
if (this->spec.freq < 8000) {
this->spec.freq = 8000;
}
if (this->spec.freq > 48000) {
this->spec.freq = 48000;
}
/* TODO: pass in/return a (Java) device ID */
this->spec.samples = Android_JNI_OpenAudioDevice(iscapture, this->spec.freq, this->spec.format == AUDIO_U8 ? 0 : 1, this->spec.channels, this->spec.samples);
if (this->spec.samples == 0) {
/* Init failed? */
return SDL_SetError("Java-side initialization failed!");
}
SDL_CalculateAudioSpec(&this->spec);
return 0;
}
static int DMA_ReopenAudio(_THIS, const char *audiodev, int format, int stereo,
SDL_AudioSpec *spec)
{
int frag_spec;
int value;
/* Close and then reopen the audio device */
close(audio_fd);
audio_fd = open(audiodev, O_RDWR, 0);
if ( audio_fd < 0 ) {
SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
return(-1);
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(spec);
/* Determine the power of two of the fragment size */
for ( frag_spec = 0; (0x01<<frag_spec) < spec->size; ++frag_spec );
if ( (0x01<<frag_spec) != spec->size ) {
SDL_SetError("Fragment size must be a power of two");
return(-1);
}
/* Set the audio buffering parameters */
if ( ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_spec) < 0 ) {
SDL_SetError("Couldn't set audio fragment spec");
return(-1);
}
/* Set the audio format */
value = format;
if ( (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) ||
(value != format) ) {
SDL_SetError("Couldn't set audio format");
return(-1);
}
/* Set mono or stereo audio */
value = (spec->channels > 1);
if ( (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) < 0) ||
(value != stereo) ) {
SDL_SetError("Couldn't set audio channels");
return(-1);
}
/* Set the DSP frequency */
value = spec->freq;
if ( ioctl(audio_fd, SNDCTL_DSP_SPEED, &value) < 0 ) {
SDL_SetError("Couldn't set audio frequency");
return(-1);
}
spec->freq = value;
/* We successfully re-opened the audio */
return(0);
}
static int
PSPAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
int format, mixlen, i;
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc(sizeof(*this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_zerop(this->hidden);
switch (this->spec.format & 0xff) {
case 8:
case 16:
this->spec.format = AUDIO_S16LSB;
break;
default:
return SDL_SetError("Unsupported audio format");
}
/* The sample count must be a multiple of 64. */
this->spec.samples = PSP_AUDIO_SAMPLE_ALIGN(this->spec.samples);
this->spec.freq = 44100;
/* Update the fragment size as size in bytes. */
SDL_CalculateAudioSpec(&this->spec);
/* Allocate the mixing buffer. Its size and starting address must
be a multiple of 64 bytes. Our sample count is already a multiple of
64, so spec->size should be a multiple of 64 as well. */
mixlen = this->spec.size * NUM_BUFFERS;
this->hidden->rawbuf = (Uint8 *) memalign(64, mixlen);
if (this->hidden->rawbuf == NULL) {
return SDL_SetError("Couldn't allocate mixing buffer");
}
/* Setup the hardware channel. */
if (this->spec.channels == 1) {
format = PSP_AUDIO_FORMAT_MONO;
} else {
format = PSP_AUDIO_FORMAT_STEREO;
}
this->hidden->channel = sceAudioChReserve(PSP_AUDIO_NEXT_CHANNEL, this->spec.samples, format);
if (this->hidden->channel < 0) {
free(this->hidden->rawbuf);
this->hidden->rawbuf = NULL;
return SDL_SetError("Couldn't reserve hardware channel");
}
memset(this->hidden->rawbuf, 0, mixlen);
for (i = 0; i < NUM_BUFFERS; i++) {
this->hidden->mixbufs[i] = &this->hidden->rawbuf[i * this->spec.size];
}
this->hidden->next_buffer = 0;
return 0;
}
static int ESD_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
esd_format_t format;
/* Convert audio spec to the ESD audio format */
format = (ESD_STREAM | ESD_PLAY);
switch ( spec->format & 0xFF ) {
case 8:
format |= ESD_BITS8;
break;
case 16:
format |= ESD_BITS16;
break;
default:
SDL_SetError("Unsupported ESD audio format");
return(-1);
}
if ( spec->channels == 1 ) {
format |= ESD_MONO;
} else {
format |= ESD_STEREO;
}
#if 0
spec->samples = ESD_BUF_SIZE; /* Darn, no way to change this yet */
#endif
/* Open a connection to the ESD audio server */
audio_fd = SDL_NAME(esd_play_stream)(format, spec->freq, NULL, get_progname());
if ( audio_fd < 0 ) {
SDL_SetError("Couldn't open ESD connection");
return(-1);
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(spec);
frame_ticks = (float)(spec->samples*1000)/spec->freq;
next_frame = SDL_GetTicks()+frame_ticks;
/* Allocate mixing buffer */
mixlen = spec->size;
mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
if ( mixbuf == NULL ) {
return(-1);
}
memset(mixbuf, spec->silence, spec->size);
/* Get the parent process id (we're the parent of the audio thread) */
parent = getpid();
/* We're ready to rock and roll. :-) */
return(0);
}
static int
MINTSTFA_OpenDevice(_THIS, const char *devname, int iscapture)
{
SDL_MintAudio_device = this;
/* Check audio capabilities */
if (MINTSTFA_CheckAudio(this) == -1) {
return 0;
}
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
SDL_CalculateAudioSpec(&this->spec);
/* Allocate memory for audio buffers in DMA-able RAM */
DEBUG_PRINT((DEBUG_NAME "buffer size=%d\n", this->spec.size));
SDL_MintAudio_audiobuf[0] =
Atari_SysMalloc(this->spec.size * 2, MX_STRAM);
if (SDL_MintAudio_audiobuf[0] == NULL) {
SDL_OutOfMemory();
SDL_free(this->hidden);
this->hidden = NULL;
return 0;
}
SDL_MintAudio_audiobuf[1] = SDL_MintAudio_audiobuf[0] + this->spec.size;
SDL_MintAudio_numbuf = 0;
SDL_memset(SDL_MintAudio_audiobuf[0], this->spec.silence,
this->spec.size * 2);
SDL_MintAudio_audiosize = this->spec.size;
SDL_MintAudio_mutex = 0;
DEBUG_PRINT((DEBUG_NAME "buffer 0 at 0x%08x\n",
SDL_MintAudio_audiobuf[0]));
DEBUG_PRINT((DEBUG_NAME "buffer 1 at 0x%08x\n",
SDL_MintAudio_audiobuf[1]));
SDL_MintAudio_CheckFpu();
/* Setup audio hardware */
MINTSTFA_InitAudio(this);
return 1; /* good to go. */
}
static int ANDROIDAUD_OpenAudio(_THIS, const char *devname, int iscapture)
{
SDL_AudioSpec *audioFormat = &this->spec;
int bytesPerSample;
JNIEnv * jniEnv = NULL;
this->hidden = (struct SDL_PrivateAudioData *) SDL_malloc((sizeof *this->hidden));
if ( this->hidden == NULL ) {
SDL_OutOfMemory();
return(-1);
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
if( ! (this->spec.format == AUDIO_S8 || this->spec.format == AUDIO_S16) )
{
__android_log_print(ANDROID_LOG_ERROR, "libSDL", "Application requested unsupported audio format - only S8 and S16 are supported");
return (-1); // TODO: enable format conversion? Don't know how to do that in SDL
}
bytesPerSample = (audioFormat->format & 0xFF) / 8;
audioFormat->format = ( bytesPerSample == 2 ) ? AUDIO_S16 : AUDIO_S8;
(*jniVM)->AttachCurrentThread(jniVM, &jniEnv, NULL);
if( !jniEnv )
{
__android_log_print(ANDROID_LOG_ERROR, "libSDL", "ANDROIDAUD_OpenAudio: Java VM AttachCurrentThread() failed");
return (-1); // TODO: enable format conversion? Don't know how to do that in SDL
}
audioBufferSize = (*jniEnv)->CallIntMethod( jniEnv, JavaAudioThread, JavaInitAudio,
(jint)audioFormat->freq, (jint)audioFormat->channels,
(jint)(( bytesPerSample == 2 ) ? 1 : 0), (jint)audioFormat->size);
if( audioBufferSize == 0 )
{
__android_log_print(ANDROID_LOG_INFO, "libSDL", "ANDROIDAUD_OpenAudio(): failed to get audio buffer from JNI");
ANDROIDAUD_CloseAudio(this);
return(-1);
}
/* We cannot call DetachCurrentThread() from main thread or we'll crash */
/* (*jniVM)->DetachCurrentThread(jniVM); */
audioFormat->samples = audioBufferSize / bytesPerSample / audioFormat->channels;
audioFormat->size = audioBufferSize;
SDL_CalculateAudioSpec(&this->spec);
return(1);
}
static int DUMMYAUD_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
sDevice = this;
float bytes_per_sec = 0.0f;
/*
/* Allocate mixing buffer */
//this->hidden->mixlen = spec->size;
//this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
//if ( this->hidden->mixbuf == NULL ) {
// return(-1);
//}
//SDL_memset(this->hidden->mixbuf, spec->silence, spec->size);
//bytes_per_sec = (float) (((spec->format & 0xFF) / 8) * spec->channels * spec->freq);
/*
* We try to make this request more audio at the correct rate for
* a given audio spec, so timing stays fairly faithful.
* Also, we have it not block at all for the first two calls, so
* it seems like we're filling two audio fragments right out of the
* gate, like other SDL drivers tend to do.
*/
//this->hidden->initial_calls = 2;
//this->hidden->write_delay = (Uint32) ((((float) spec->size) / bytes_per_sec) * 1000.0f);
int bps = 2;
switch(spec->format) {
case AUDIO_U8: this->hidden->info.fmt = AUDIO_FMT_U8; bps=1; break;
case AUDIO_S8: this->hidden->info.fmt = AUDIO_FMT_S8; bps=1; break;
case AUDIO_U16: this->hidden->info.fmt = AUDIO_FMT_U16; break;
case AUDIO_S16: this->hidden->info.fmt = AUDIO_FMT_S16; break;
default: spec->format = AUDIO_S16; this->hidden->info.fmt = AUDIO_FMT_S16; break;
}
if(spec->channels>2) spec->channels=2;
SDL_CalculateAudioSpec(spec);
this->hidden->info.numChannels = spec->channels;
this->hidden->info.sampleRate = spec->freq;
this->hidden->info.bufferSize = spec->samples*bps*spec->channels;
this->hidden->info.buffer = malloc(this->hidden->info.bufferSize);
maAudioBufferInit(&this->hidden->info);
/* We're ready to rock and roll. :-) */
return (1);
}
static int
ANDROIDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
SDL_AudioFormat test_format;
SDL_assert((captureDevice == NULL) || !iscapture);
SDL_assert((audioDevice == NULL) || iscapture);
if (iscapture) {
captureDevice = this;
} else {
audioDevice = this;
}
this->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, (sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
test_format = SDL_FirstAudioFormat(this->spec.format);
while (test_format != 0) { /* no "UNKNOWN" constant */
if ((test_format == AUDIO_U8) ||
(test_format == AUDIO_S16) ||
(test_format == AUDIO_F32)) {
this->spec.format = test_format;
break;
}
test_format = SDL_NextAudioFormat();
}
if (test_format == 0) {
/* Didn't find a compatible format :( */
return SDL_SetError("No compatible audio format!");
}
if (Android_JNI_OpenAudioDevice(iscapture, &this->spec) < 0) {
return -1;
}
SDL_CalculateAudioSpec(&this->spec);
return 0;
}
static int NAS_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
AuElement elms[3];
int buffer_size;
Uint16 test_format, format;
this->hidden->mixbuf = NULL;
/* Try for a closest match on audio format */
format = 0;
for ( test_format = SDL_FirstAudioFormat(spec->format);
! format && test_format; ) {
format = sdlformat_to_auformat(test_format);
if (format == AuNone) {
test_format = SDL_NextAudioFormat();
}
}
if ( format == 0 ) {
SDL_SetError("Couldn't find any hardware audio formats");
return(-1);
}
spec->format = test_format;
this->hidden->aud = AuOpenServer("", 0, NULL, 0, NULL, NULL);
if (this->hidden->aud == 0)
{
SDL_SetError("Couldn't open connection to NAS server");
return (-1);
}
this->hidden->dev = find_device(this, spec->channels);
if ((this->hidden->dev == AuNone) || (!(this->hidden->flow = AuCreateFlow(this->hidden->aud, NULL)))) {
AuCloseServer(this->hidden->aud);
this->hidden->aud = 0;
SDL_SetError("Couldn't find a fitting playback device on NAS server");
return (-1);
}
buffer_size = spec->freq;
if (buffer_size < 4096)
buffer_size = 4096;
if (buffer_size > 32768)
buffer_size = 32768; /* So that the buffer won't get unmanageably big. */
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(spec);
this2 = this->hidden;
AuMakeElementImportClient(elms, spec->freq, format, spec->channels, AuTrue,
buffer_size, buffer_size / 4, 0, NULL);
AuMakeElementExportDevice(elms+1, 0, this->hidden->dev, spec->freq,
AuUnlimitedSamples, 0, NULL);
AuSetElements(this->hidden->aud, this->hidden->flow, AuTrue, 2, elms, NULL);
AuRegisterEventHandler(this->hidden->aud, AuEventHandlerIDMask, 0, this->hidden->flow,
event_handler, (AuPointer) NULL);
AuStartFlow(this->hidden->aud, this->hidden->flow, NULL);
/* Allocate mixing buffer */
this->hidden->mixlen = spec->size;
this->hidden->mixbuf = (Uint8 *)SDL_AllocAudioMem(this->hidden->mixlen);
if ( this->hidden->mixbuf == NULL ) {
return(-1);
}
memset(this->hidden->mixbuf, spec->silence, spec->size);
/* Get the parent process id (we're the parent of the audio thread) */
this->hidden->parent = getpid();
/* We're ready to rock and roll. :-) */
return(0);
}
static int
BSDAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
{
const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
SDL_AudioFormat format = 0;
audio_info_t info;
/* We don't care what the devname is...we'll try to open anything. */
/* ...but default to first name in the list... */
if (devname == NULL) {
devname = SDL_GetAudioDeviceName(0, iscapture);
if (devname == NULL) {
SDL_SetError("No such audio device");
return 0;
}
}
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Open the audio device */
this->hidden->audio_fd = open(devname, flags, 0);
if (this->hidden->audio_fd < 0) {
SDL_SetError("Couldn't open %s: %s", devname, strerror(errno));
return 0;
}
AUDIO_INITINFO(&info);
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
/* Set to play mode */
info.mode = AUMODE_PLAY;
if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) < 0) {
BSDAUDIO_CloseDevice(this);
SDL_SetError("Couldn't put device into play mode");
return 0;
}
AUDIO_INITINFO(&info);
for (format = SDL_FirstAudioFormat(this->spec.format);
format; format = SDL_NextAudioFormat()) {
switch (format) {
case AUDIO_U8:
info.play.encoding = AUDIO_ENCODING_ULINEAR;
info.play.precision = 8;
break;
case AUDIO_S8:
info.play.encoding = AUDIO_ENCODING_SLINEAR;
info.play.precision = 8;
break;
case AUDIO_S16LSB:
info.play.encoding = AUDIO_ENCODING_SLINEAR_LE;
info.play.precision = 16;
break;
case AUDIO_S16MSB:
info.play.encoding = AUDIO_ENCODING_SLINEAR_BE;
info.play.precision = 16;
break;
case AUDIO_U16LSB:
info.play.encoding = AUDIO_ENCODING_ULINEAR_LE;
info.play.precision = 16;
break;
case AUDIO_U16MSB:
info.play.encoding = AUDIO_ENCODING_ULINEAR_BE;
info.play.precision = 16;
break;
default:
continue;
}
if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) == 0) {
break;
}
}
if (!format) {
BSDAUDIO_CloseDevice(this);
SDL_SetError("No supported encoding for 0x%x", this->spec.format);
return 0;
}
this->spec.format = format;
AUDIO_INITINFO(&info);
info.play.channels = this->spec.channels;
if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) == -1) {
this->spec.channels = 1;
}
AUDIO_INITINFO(&info);
info.play.sample_rate = this->spec.freq;
info.blocksize = this->spec.size;
info.hiwat = 5;
info.lowat = 3;
(void) ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info);
(void) ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info);
this->spec.freq = info.play.sample_rate;
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
BSDAUDIO_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
BSDAUDIO_Status(this);
/* We're ready to rock and roll. :-) */
return (0);
}
static int
WINMM_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
int valid_datatype = 0;
MMRESULT result;
WAVEFORMATEX waveformat;
UINT devId = WAVE_MAPPER; /* WAVE_MAPPER == choose system's default */
UINT i;
if (handle != NULL) { /* specific device requested? */
/* -1 because we increment the original value to avoid NULL. */
const size_t val = ((size_t) handle) - 1;
devId = (UINT) val;
}
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Initialize the wavebuf structures for closing */
for (i = 0; i < NUM_BUFFERS; ++i)
this->hidden->wavebuf[i].dwUser = 0xFFFF;
if (this->spec.channels > 2)
this->spec.channels = 2; /* !!! FIXME: is this right? */
while ((!valid_datatype) && (test_format)) {
switch (test_format) {
case AUDIO_U8:
case AUDIO_S16:
case AUDIO_S32:
case AUDIO_F32:
this->spec.format = test_format;
if (PrepWaveFormat(this, devId, &waveformat, iscapture)) {
valid_datatype = 1;
} else {
test_format = SDL_NextAudioFormat();
}
break;
default:
test_format = SDL_NextAudioFormat();
break;
}
}
if (!valid_datatype) {
WINMM_CloseDevice(this);
return SDL_SetError("Unsupported audio format");
}
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* Open the audio device */
if (iscapture) {
result = waveInOpen(&this->hidden->hin, devId, &waveformat,
(DWORD_PTR) CaptureSound, (DWORD_PTR) this,
CALLBACK_FUNCTION);
} else {
result = waveOutOpen(&this->hidden->hout, devId, &waveformat,
(DWORD_PTR) FillSound, (DWORD_PTR) this,
CALLBACK_FUNCTION);
}
if (result != MMSYSERR_NOERROR) {
WINMM_CloseDevice(this);
return SetMMerror("waveOutOpen()", result);
}
#ifdef SOUND_DEBUG
/* Check the sound device we retrieved */
{
WAVEOUTCAPS caps;
result = waveOutGetDevCaps((UINT) this->hidden->hout,
&caps, sizeof(caps));
if (result != MMSYSERR_NOERROR) {
WINMM_CloseDevice(this);
return SetMMerror("waveOutGetDevCaps()", result);
}
printf("Audio device: %s\n", caps.szPname);
}
#endif
/* Create the audio buffer semaphore */
this->hidden->audio_sem =
CreateSemaphore(NULL, NUM_BUFFERS - 1, NUM_BUFFERS, NULL);
if (this->hidden->audio_sem == NULL) {
WINMM_CloseDevice(this);
return SDL_SetError("Couldn't create semaphore");
}
/* Create the sound buffers */
this->hidden->mixbuf =
(Uint8 *) SDL_malloc(NUM_BUFFERS * this->spec.size);
if (this->hidden->mixbuf == NULL) {
WINMM_CloseDevice(this);
return SDL_OutOfMemory();
}
for (i = 0; i < NUM_BUFFERS; ++i) {
SDL_memset(&this->hidden->wavebuf[i], 0,
sizeof(this->hidden->wavebuf[i]));
this->hidden->wavebuf[i].dwBufferLength = this->spec.size;
this->hidden->wavebuf[i].dwFlags = WHDR_DONE;
this->hidden->wavebuf[i].lpData =
(LPSTR) & this->hidden->mixbuf[i * this->spec.size];
result = waveOutPrepareHeader(this->hidden->hout,
&this->hidden->wavebuf[i],
sizeof(this->hidden->wavebuf[i]));
if (result != MMSYSERR_NOERROR) {
WINMM_CloseDevice(this);
return SetMMerror("waveOutPrepareHeader()", result);
}
}
return 0; /* Ready to go! */
}
static int
XAUDIO2_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
HRESULT result = S_OK;
WAVEFORMATEX waveformat;
int valid_format = 0;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
IXAudio2 *ixa2 = NULL;
IXAudio2SourceVoice *source = NULL;
#if defined(SDL_XAUDIO2_WIN8)
LPCWSTR devId = NULL;
#else
UINT32 devId = 0; /* 0 == system default device. */
#endif
static IXAudio2VoiceCallbackVtbl callbacks_vtable = {
VoiceCBOnVoiceProcessPassStart,
VoiceCBOnVoiceProcessPassEnd,
VoiceCBOnStreamEnd,
VoiceCBOnBufferStart,
VoiceCBOnBufferEnd,
VoiceCBOnLoopEnd,
VoiceCBOnVoiceError
};
static IXAudio2VoiceCallback callbacks = { &callbacks_vtable };
#if defined(SDL_XAUDIO2_WIN8)
/* !!! FIXME: hook up hotplugging. */
#else
if (handle != NULL) { /* specific device requested? */
/* -1 because we increment the original value to avoid NULL. */
const size_t val = ((size_t) handle) - 1;
devId = (UINT32) val;
}
#endif
if (XAudio2Create(&ixa2, 0, XAUDIO2_DEFAULT_PROCESSOR) != S_OK) {
return SDL_SetError("XAudio2: XAudio2Create() failed at open.");
}
/*
XAUDIO2_DEBUG_CONFIGURATION debugConfig;
debugConfig.TraceMask = XAUDIO2_LOG_ERRORS; //XAUDIO2_LOG_WARNINGS | XAUDIO2_LOG_DETAIL | XAUDIO2_LOG_FUNC_CALLS | XAUDIO2_LOG_TIMING | XAUDIO2_LOG_LOCKS | XAUDIO2_LOG_MEMORY | XAUDIO2_LOG_STREAMING;
debugConfig.BreakMask = XAUDIO2_LOG_ERRORS; //XAUDIO2_LOG_WARNINGS;
debugConfig.LogThreadID = TRUE;
debugConfig.LogFileline = TRUE;
debugConfig.LogFunctionName = TRUE;
debugConfig.LogTiming = TRUE;
ixa2->SetDebugConfiguration(&debugConfig);
*/
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
IXAudio2_Release(ixa2);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
this->hidden->ixa2 = ixa2;
this->hidden->semaphore = SDL_CreateSemaphore(1);
if (this->hidden->semaphore == NULL) {
XAUDIO2_CloseDevice(this);
return SDL_SetError("XAudio2: CreateSemaphore() failed!");
}
while ((!valid_format) && (test_format)) {
switch (test_format) {
case AUDIO_U8:
case AUDIO_S16:
case AUDIO_S32:
case AUDIO_F32:
this->spec.format = test_format;
valid_format = 1;
break;
}
test_format = SDL_NextAudioFormat();
}
if (!valid_format) {
XAUDIO2_CloseDevice(this);
return SDL_SetError("XAudio2: Unsupported audio format");
}
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* We feed a Source, it feeds the Mastering, which feeds the device. */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_malloc(2 * this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
XAUDIO2_CloseDevice(this);
return SDL_OutOfMemory();
}
this->hidden->nextbuf = this->hidden->mixbuf;
SDL_memset(this->hidden->mixbuf, 0, 2 * this->hidden->mixlen);
/* We use XAUDIO2_DEFAULT_CHANNELS instead of this->spec.channels. On
Xbox360, this means 5.1 output, but on Windows, it means "figure out
what the system has." It might be preferable to let XAudio2 blast
stereo output to appropriate surround sound configurations
instead of clamping to 2 channels, even though we'll configure the
Source Voice for whatever number of channels you supply. */
#if SDL_XAUDIO2_WIN8
result = IXAudio2_CreateMasteringVoice(ixa2, &this->hidden->mastering,
XAUDIO2_DEFAULT_CHANNELS,
this->spec.freq, 0, devId, NULL, AudioCategory_GameEffects);
#else
result = IXAudio2_CreateMasteringVoice(ixa2, &this->hidden->mastering,
XAUDIO2_DEFAULT_CHANNELS,
this->spec.freq, 0, devId, NULL);
#endif
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
return SDL_SetError("XAudio2: Couldn't create mastering voice");
}
SDL_zero(waveformat);
if (SDL_AUDIO_ISFLOAT(this->spec.format)) {
waveformat.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
} else {
waveformat.wFormatTag = WAVE_FORMAT_PCM;
}
waveformat.wBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
waveformat.nChannels = this->spec.channels;
waveformat.nSamplesPerSec = this->spec.freq;
waveformat.nBlockAlign =
waveformat.nChannels * (waveformat.wBitsPerSample / 8);
waveformat.nAvgBytesPerSec =
waveformat.nSamplesPerSec * waveformat.nBlockAlign;
waveformat.cbSize = sizeof(waveformat);
#ifdef __WINRT__
// DLudwig: for now, make XAudio2 do sample rate conversion, just to
// get the loopwave test to work.
//
// TODO, WinRT: consider removing WinRT-specific source-voice creation code from SDL_xaudio2.c
result = IXAudio2_CreateSourceVoice(ixa2, &source, &waveformat,
0,
1.0f, &callbacks, NULL, NULL);
#else
result = IXAudio2_CreateSourceVoice(ixa2, &source, &waveformat,
XAUDIO2_VOICE_NOSRC |
XAUDIO2_VOICE_NOPITCH,
1.0f, &callbacks, NULL, NULL);
#endif
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
return SDL_SetError("XAudio2: Couldn't create source voice");
}
this->hidden->source = source;
/* Start everything playing! */
result = IXAudio2_StartEngine(ixa2);
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
return SDL_SetError("XAudio2: Couldn't start engine");
}
result = IXAudio2SourceVoice_Start(source, 0, XAUDIO2_COMMIT_NOW);
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
return SDL_SetError("XAudio2: Couldn't start source voice");
}
return 0; /* good to go. */
}
static int
SNDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
struct sio_par par;
int status;
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc(sizeof(*this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, sizeof(*this->hidden));
this->hidden->mixlen = this->spec.size;
/* !!! FIXME: SIO_DEVANY can be a specific device... */
if ((this->hidden->dev = SNDIO_sio_open(SIO_DEVANY, SIO_PLAY, 0)) == NULL) {
SNDIO_CloseDevice(this);
return SDL_SetError("sio_open() failed");
}
SNDIO_sio_initpar(&par);
par.rate = this->spec.freq;
par.pchan = this->spec.channels;
par.round = this->spec.samples;
par.appbufsz = par.round * 2;
/* Try for a closest match on audio format */
status = -1;
while (test_format && (status < 0)) {
if (!SDL_AUDIO_ISFLOAT(test_format)) {
par.le = SDL_AUDIO_ISLITTLEENDIAN(test_format) ? 1 : 0;
par.sig = SDL_AUDIO_ISSIGNED(test_format) ? 1 : 0;
par.bits = SDL_AUDIO_BITSIZE(test_format);
if (SNDIO_sio_setpar(this->hidden->dev, &par) == 0) {
continue;
}
if (SNDIO_sio_getpar(this->hidden->dev, &par) == 0) {
SNDIO_CloseDevice(this);
return SDL_SetError("sio_getpar() failed");
}
if (par.bps != SIO_BPS(par.bits)) {
continue;
}
if ((par.bits == 8 * par.bps) || (par.msb)) {
status = 0;
break;
}
}
test_format = SDL_NextAudioFormat();
}
if (status < 0) {
SNDIO_CloseDevice(this);
return SDL_SetError("sndio: Couldn't find any hardware audio formats");
}
if ((par.bps == 4) && (par.sig) && (par.le))
this->spec.format = AUDIO_S32LSB;
else if ((par.bps == 4) && (par.sig) && (!par.le))
this->spec.format = AUDIO_S32MSB;
else if ((par.bps == 2) && (par.sig) && (par.le))
this->spec.format = AUDIO_S16LSB;
else if ((par.bps == 2) && (par.sig) && (!par.le))
this->spec.format = AUDIO_S16MSB;
else if ((par.bps == 2) && (!par.sig) && (par.le))
this->spec.format = AUDIO_U16LSB;
else if ((par.bps == 2) && (!par.sig) && (!par.le))
this->spec.format = AUDIO_U16MSB;
else if ((par.bps == 1) && (par.sig))
this->spec.format = AUDIO_S8;
else if ((par.bps == 1) && (!par.sig))
this->spec.format = AUDIO_U8;
else {
SNDIO_CloseDevice(this);
return SDL_SetError("sndio: Got unsupported hardware audio format.");
}
this->spec.freq = par.rate;
this->spec.channels = par.pchan;
this->spec.samples = par.round;
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
SNDIO_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->hidden->mixlen);
if (!SNDIO_sio_start(this->hidden->dev)) {
return SDL_SetError("sio_start() failed");
}
/* We're ready to rock and roll. :-) */
return 0;
}
static int
PULSEAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
{
struct SDL_PrivateAudioData *h = NULL;
Uint16 test_format = 0;
pa_sample_spec paspec;
pa_buffer_attr paattr;
pa_channel_map pacmap;
pa_stream_flags_t flags = 0;
int state = 0;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
h = this->hidden;
paspec.format = PA_SAMPLE_INVALID;
/* Try for a closest match on audio format */
for (test_format = SDL_FirstAudioFormat(this->spec.format);
(paspec.format == PA_SAMPLE_INVALID) && test_format;) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
switch (test_format) {
case AUDIO_U8:
paspec.format = PA_SAMPLE_U8;
break;
case AUDIO_S16LSB:
paspec.format = PA_SAMPLE_S16LE;
break;
case AUDIO_S16MSB:
paspec.format = PA_SAMPLE_S16BE;
break;
case AUDIO_S32LSB:
paspec.format = PA_SAMPLE_S32LE;
break;
case AUDIO_S32MSB:
paspec.format = PA_SAMPLE_S32BE;
break;
case AUDIO_F32LSB:
paspec.format = PA_SAMPLE_FLOAT32LE;
break;
case AUDIO_F32MSB:
paspec.format = PA_SAMPLE_FLOAT32BE;
break;
default:
paspec.format = PA_SAMPLE_INVALID;
break;
}
if (paspec.format == PA_SAMPLE_INVALID) {
test_format = SDL_NextAudioFormat();
}
}
if (paspec.format == PA_SAMPLE_INVALID) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
/* Calculate the final parameters for this audio specification */
#ifdef PA_STREAM_ADJUST_LATENCY
this->spec.samples /= 2; /* Mix in smaller chunck to avoid underruns */
#endif
SDL_CalculateAudioSpec(&this->spec);
/* Allocate mixing buffer */
h->mixlen = this->spec.size;
h->mixbuf = (Uint8 *) SDL_AllocAudioMem(h->mixlen);
if (h->mixbuf == NULL) {
PULSEAUDIO_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(h->mixbuf, this->spec.silence, this->spec.size);
paspec.channels = this->spec.channels;
paspec.rate = this->spec.freq;
/* Reduced prebuffering compared to the defaults. */
#ifdef PA_STREAM_ADJUST_LATENCY
/* 2x original requested bufsize */
paattr.tlength = h->mixlen * 4;
paattr.prebuf = -1;
paattr.maxlength = -1;
/* -1 can lead to pa_stream_writable_size() >= mixlen never being true */
paattr.minreq = h->mixlen;
flags = PA_STREAM_ADJUST_LATENCY;
#else
paattr.tlength = h->mixlen*2;
paattr.prebuf = h->mixlen*2;
paattr.maxlength = h->mixlen*2;
paattr.minreq = h->mixlen;
#endif
/* The SDL ALSA output hints us that we use Windows' channel mapping */
/* http://bugzilla.libsdl.org/show_bug.cgi?id=110 */
PULSEAUDIO_pa_channel_map_init_auto(&pacmap, this->spec.channels,
PA_CHANNEL_MAP_WAVEEX);
/* Set up a new main loop */
if (!(h->mainloop = PULSEAUDIO_pa_mainloop_new())) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("pa_mainloop_new() failed");
}
h->mainloop_api = PULSEAUDIO_pa_mainloop_get_api(h->mainloop);
h->context = PULSEAUDIO_pa_context_new(h->mainloop_api, getAppName());
if (!h->context) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("pa_context_new() failed");
}
/* Connect to the PulseAudio server */
if (PULSEAUDIO_pa_context_connect(h->context, NULL, 0, NULL) < 0) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Could not setup connection to PulseAudio");
}
do {
if (PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("pa_mainloop_iterate() failed");
}
state = PULSEAUDIO_pa_context_get_state(h->context);
if (!PA_CONTEXT_IS_GOOD(state)) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Could not connect to PulseAudio");
}
} while (state != PA_CONTEXT_READY);
h->stream = PULSEAUDIO_pa_stream_new(
h->context,
"Simple DirectMedia Layer", /* stream description */
&paspec, /* sample format spec */
&pacmap /* channel map */
);
if (h->stream == NULL) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Could not set up PulseAudio stream");
}
if (PULSEAUDIO_pa_stream_connect_playback(h->stream, NULL, &paattr, flags,
NULL, NULL) < 0) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Could not connect PulseAudio stream");
}
do {
if (PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("pa_mainloop_iterate() failed");
}
state = PULSEAUDIO_pa_stream_get_state(h->stream);
if (!PA_STREAM_IS_GOOD(state)) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Could not create to PulseAudio stream");
}
} while (state != PA_STREAM_READY);
/* We're ready to rock and roll. :-) */
return 0;
}
static int
prepare_audiounit(_THIS, const char *devname, int iscapture,
const AudioStreamBasicDescription * strdesc)
{
OSStatus result = noErr;
AURenderCallbackStruct callback;
AudioComponentDescription desc;
AudioComponent comp = NULL;
UInt32 enableIO = 0;
const AudioUnitElement output_bus = 0;
const AudioUnitElement input_bus = 1;
const AudioUnitElement bus = ((iscapture) ? input_bus : output_bus);
const AudioUnitScope scope = ((iscapture) ? kAudioUnitScope_Output :
kAudioUnitScope_Input);
SDL_memset(&desc, '\0', sizeof(AudioComponentDescription));
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_RemoteIO;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
comp = AudioComponentFindNext(NULL, &desc);
if (comp == NULL) {
fprintf(stderr, "Couldn't find requested CoreAudio component");
return 0;
}
/* Open & initialize the audio unit */
/*
AudioComponentInstanceNew only available on iPhone OS 2.0 and Mac OS X 10.6
We can't use OpenAComponent on iPhone because it is not present
*/
result = AudioComponentInstanceNew(comp, &this->hidden->audioUnit);
CHECK_RESULT("AudioComponentInstanceNew");
this->hidden->audioUnitOpened = 1;
// !!! FIXME: this is wrong?
enableIO = ((iscapture) ? 1 : 0);
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input, input_bus,
&enableIO, sizeof(enableIO));
CHECK_RESULT("AudioUnitSetProperty (kAudioUnitProperty_EnableIO input)");
// !!! FIXME: this is wrong?
enableIO = ((iscapture) ? 0 : 1);
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output, output_bus,
&enableIO, sizeof(enableIO));
CHECK_RESULT("AudioUnitSetProperty (kAudioUnitProperty_EnableIO output)");
/*result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioOutputUnitProperty_CurrentDevice,
kAudioUnitScope_Global, 0,
&this->hidden->deviceID,
sizeof(AudioDeviceID));
CHECK_RESULT("AudioUnitSetProperty (kAudioOutputUnitProperty_CurrentDevice)"); */
/* Set the data format of the audio unit. */
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioUnitProperty_StreamFormat,
scope, bus, strdesc, sizeof(*strdesc));
CHECK_RESULT("AudioUnitSetProperty (kAudioUnitProperty_StreamFormat)");
/* Set the audio callback */
SDL_memset(&callback, '\0', sizeof(AURenderCallbackStruct));
callback.inputProc = ((iscapture) ? inputCallback : outputCallback);
callback.inputProcRefCon = this;
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioUnitProperty_SetRenderCallback,
scope, bus, &callback, sizeof(callback));
CHECK_RESULT
("AudioUnitSetProperty (kAudioUnitProperty_SetInputCallback)");
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
/* Allocate a sample buffer */
this->hidden->bufferOffset = this->hidden->bufferSize = this->spec.size;
this->hidden->buffer = SDL_malloc(this->hidden->bufferSize);
result = AudioUnitInitialize(this->hidden->audioUnit);
CHECK_RESULT("AudioUnitInitialize");
/* Finally, start processing of the audio unit */
result = AudioOutputUnitStart(this->hidden->audioUnit);
CHECK_RESULT("AudioOutputUnitStart");
/* We're running! */
return 1;
}
static int
NAS_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
AuElement elms[3];
int buffer_size;
SDL_AudioFormat test_format, format;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Try for a closest match on audio format */
format = 0;
for (test_format = SDL_FirstAudioFormat(this->spec.format);
!format && test_format;) {
format = sdlformat_to_auformat(test_format);
if (format == AuNone) {
test_format = SDL_NextAudioFormat();
}
}
if (format == 0) {
NAS_CloseDevice(this);
return SDL_SetError("NAS: Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
this->hidden->aud = NAS_AuOpenServer("", 0, NULL, 0, NULL, NULL);
if (this->hidden->aud == 0) {
NAS_CloseDevice(this);
return SDL_SetError("NAS: Couldn't open connection to NAS server");
}
this->hidden->dev = find_device(this, this->spec.channels);
if ((this->hidden->dev == AuNone)
|| (!(this->hidden->flow = NAS_AuCreateFlow(this->hidden->aud, 0)))) {
NAS_CloseDevice(this);
return SDL_SetError("NAS: Couldn't find a fitting device on NAS server");
}
buffer_size = this->spec.freq;
if (buffer_size < 4096)
buffer_size = 4096;
if (buffer_size > 32768)
buffer_size = 32768; /* So that the buffer won't get unmanageably big. */
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
this2 = this->hidden;
AuMakeElementImportClient(elms, this->spec.freq, format,
this->spec.channels, AuTrue, buffer_size,
buffer_size / 4, 0, NULL);
AuMakeElementExportDevice(elms + 1, 0, this->hidden->dev, this->spec.freq,
AuUnlimitedSamples, 0, NULL);
NAS_AuSetElements(this->hidden->aud, this->hidden->flow, AuTrue, 2, elms,
NULL);
NAS_AuRegisterEventHandler(this->hidden->aud, AuEventHandlerIDMask, 0,
this->hidden->flow, event_handler,
(AuPointer) NULL);
NAS_AuStartFlow(this->hidden->aud, this->hidden->flow, NULL);
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
NAS_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/* We're ready to rock and roll. :-) */
return 0;
}
static int
PAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
{
const char *workaround = SDL_getenv("SDL_DSP_NOSELECT");
char audiodev[1024];
const char *err = NULL;
int format;
int bytes_per_sample;
SDL_AudioFormat test_format;
audio_init paud_init;
audio_buffer paud_bufinfo;
audio_status paud_status;
audio_control paud_control;
audio_change paud_change;
int fd = -1;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Open the audio device */
fd = OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
this->hidden->audio_fd = fd;
if (fd < 0) {
PAUDIO_CloseDevice(this);
return SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
}
/*
* We can't set the buffer size - just ask the device for the maximum
* that we can have.
*/
if (ioctl(fd, AUDIO_BUFFER, &paud_bufinfo) < 0) {
PAUDIO_CloseDevice(this);
return SDL_SetError("Couldn't get audio buffer information");
}
if (this->spec.channels > 1)
this->spec.channels = 2;
else
this->spec.channels = 1;
/*
* Fields in the audio_init structure:
*
* Ignored by us:
*
* paud.loadpath[LOAD_PATH]; * DSP code to load, MWave chip only?
* paud.slot_number; * slot number of the adapter
* paud.device_id; * adapter identification number
*
* Input:
*
* paud.srate; * the sampling rate in Hz
* paud.bits_per_sample; * 8, 16, 32, ...
* paud.bsize; * block size for this rate
* paud.mode; * ADPCM, PCM, MU_LAW, A_LAW, SOURCE_MIX
* paud.channels; * 1=mono, 2=stereo
* paud.flags; * FIXED - fixed length data
* * LEFT_ALIGNED, RIGHT_ALIGNED (var len only)
* * TWOS_COMPLEMENT - 2's complement data
* * SIGNED - signed? comment seems wrong in sys/audio.h
* * BIG_ENDIAN
* paud.operation; * PLAY, RECORD
*
* Output:
*
* paud.flags; * PITCH - pitch is supported
* * INPUT - input is supported
* * OUTPUT - output is supported
* * MONITOR - monitor is supported
* * VOLUME - volume is supported
* * VOLUME_DELAY - volume delay is supported
* * BALANCE - balance is supported
* * BALANCE_DELAY - balance delay is supported
* * TREBLE - treble control is supported
* * BASS - bass control is supported
* * BESTFIT_PROVIDED - best fit returned
* * LOAD_CODE - DSP load needed
* paud.rc; * NO_PLAY - DSP code can't do play requests
* * NO_RECORD - DSP code can't do record requests
* * INVALID_REQUEST - request was invalid
* * CONFLICT - conflict with open's flags
* * OVERLOADED - out of DSP MIPS or memory
* paud.position_resolution; * smallest increment for position
*/
paud_init.srate = this->spec.freq;
paud_init.mode = PCM;
paud_init.operation = PLAY;
paud_init.channels = this->spec.channels;
/* Try for a closest match on audio format */
format = 0;
for (test_format = SDL_FirstAudioFormat(this->spec.format);
!format && test_format;) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
switch (test_format) {
case AUDIO_U8:
bytes_per_sample = 1;
paud_init.bits_per_sample = 8;
paud_init.flags = TWOS_COMPLEMENT | FIXED;
format = 1;
break;
case AUDIO_S8:
bytes_per_sample = 1;
paud_init.bits_per_sample = 8;
paud_init.flags = SIGNED | TWOS_COMPLEMENT | FIXED;
format = 1;
break;
case AUDIO_S16LSB:
bytes_per_sample = 2;
paud_init.bits_per_sample = 16;
paud_init.flags = SIGNED | TWOS_COMPLEMENT | FIXED;
format = 1;
break;
case AUDIO_S16MSB:
bytes_per_sample = 2;
paud_init.bits_per_sample = 16;
paud_init.flags = BIG_ENDIAN | SIGNED | TWOS_COMPLEMENT | FIXED;
format = 1;
break;
case AUDIO_U16LSB:
bytes_per_sample = 2;
paud_init.bits_per_sample = 16;
paud_init.flags = TWOS_COMPLEMENT | FIXED;
format = 1;
break;
case AUDIO_U16MSB:
bytes_per_sample = 2;
paud_init.bits_per_sample = 16;
paud_init.flags = BIG_ENDIAN | TWOS_COMPLEMENT | FIXED;
format = 1;
break;
default:
break;
}
if (!format) {
test_format = SDL_NextAudioFormat();
}
}
if (format == 0) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Couldn't find any hardware audio formats\n");
#endif
PAUDIO_CloseDevice(this);
return SDL_SetError("Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
/*
* We know the buffer size and the max number of subsequent writes
* that can be pending. If more than one can pend, allow the application
* to do something like double buffering between our write buffer and
* the device's own buffer that we are filling with write() anyway.
*
* We calculate this->spec.samples like this because
* SDL_CalculateAudioSpec() will give put paud_bufinfo.write_buf_cap
* (or paud_bufinfo.write_buf_cap/2) into this->spec.size in return.
*/
if (paud_bufinfo.request_buf_cap == 1) {
this->spec.samples = paud_bufinfo.write_buf_cap
/ bytes_per_sample / this->spec.channels;
} else {
this->spec.samples = paud_bufinfo.write_buf_cap
/ bytes_per_sample / this->spec.channels / 2;
}
paud_init.bsize = bytes_per_sample * this->spec.channels;
SDL_CalculateAudioSpec(&this->spec);
/*
* The AIX paud device init can't modify the values of the audio_init
* structure that we pass to it. So we don't need any recalculation
* of this stuff and no reinit call as in linux dsp code.
*
* /dev/paud supports all of the encoding formats, so we don't need
* to do anything like reopening the device, either.
*/
if (ioctl(fd, AUDIO_INIT, &paud_init) < 0) {
switch (paud_init.rc) {
case 1:
err = "Couldn't set audio format: DSP can't do play requests";
break;
case 2:
err = "Couldn't set audio format: DSP can't do record requests";
break;
case 4:
err = "Couldn't set audio format: request was invalid";
break;
case 5:
err = "Couldn't set audio format: conflict with open's flags";
break;
case 6:
err = "Couldn't set audio format: out of DSP MIPS or memory";
break;
default:
err = "Couldn't set audio format: not documented in sys/audio.h";
break;
}
}
if (err != NULL) {
PAUDIO_CloseDevice(this);
return SDL_SetError("Paudio: %s", err);
}
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
PAUDIO_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/*
* Set some paramters: full volume, first speaker that we can find.
* Ignore the other settings for now.
*/
paud_change.input = AUDIO_IGNORE; /* the new input source */
paud_change.output = OUTPUT_1; /* EXTERNAL_SPEAKER,INTERNAL_SPEAKER,OUTPUT_1 */
paud_change.monitor = AUDIO_IGNORE; /* the new monitor state */
paud_change.volume = 0x7fffffff; /* volume level [0-0x7fffffff] */
paud_change.volume_delay = AUDIO_IGNORE; /* the new volume delay */
paud_change.balance = 0x3fffffff; /* the new balance */
paud_change.balance_delay = AUDIO_IGNORE; /* the new balance delay */
paud_change.treble = AUDIO_IGNORE; /* the new treble state */
paud_change.bass = AUDIO_IGNORE; /* the new bass state */
paud_change.pitch = AUDIO_IGNORE; /* the new pitch state */
paud_control.ioctl_request = AUDIO_CHANGE;
paud_control.request_info = (char *) &paud_change;
if (ioctl(fd, AUDIO_CONTROL, &paud_control) < 0) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Can't change audio display settings\n");
#endif
}
/*
* Tell the device to expect data. Actual start will wait for
* the first write() call.
*/
paud_control.ioctl_request = AUDIO_START;
paud_control.position = 0;
if (ioctl(fd, AUDIO_CONTROL, &paud_control) < 0) {
PAUDIO_CloseDevice(this);
#ifdef DEBUG_AUDIO
fprintf(stderr, "Can't start audio play\n");
#endif
return SDL_SetError("Can't start audio play");
}
/* Check to see if we need to use select() workaround */
if (workaround != NULL) {
this->hidden->frame_ticks = (float) (this->spec.samples * 1000) /
this->spec.freq;
this->hidden->next_frame = SDL_GetTicks() + this->hidden->frame_ticks;
}
/* We're ready to rock and roll. :-) */
return 0;
}
static int Mac_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
SndDoubleBufferHeader2 audio_dbh;
int i;
long initOptions;
int sample_bits;
SndDoubleBackUPP doubleBackProc;
/* Check to make sure double-buffered audio is available */
if ( ! DoubleBufferAudio_Available() ) {
SDL_SetError("Sound manager doesn't support double-buffering");
return(-1);
}
/* Very few conversions are required, but... */
switch (spec->format) {
case AUDIO_S8:
spec->format = AUDIO_U8;
break;
case AUDIO_U16LSB:
spec->format = AUDIO_S16LSB;
break;
case AUDIO_U16MSB:
spec->format = AUDIO_S16MSB;
break;
}
SDL_CalculateAudioSpec(spec);
/* initialize the double-back header */
memset(&audio_dbh, 0, sizeof(audio_dbh));
doubleBackProc = NewSndDoubleBackProc (sndDoubleBackProc);
sample_bits = spec->size / spec->samples / spec->channels * 8;
audio_dbh.dbhNumChannels = spec->channels;
audio_dbh.dbhSampleSize = sample_bits;
audio_dbh.dbhCompressionID = 0;
audio_dbh.dbhPacketSize = 0;
audio_dbh.dbhSampleRate = spec->freq << 16;
audio_dbh.dbhDoubleBack = doubleBackProc;
audio_dbh.dbhFormat = 0;
/* Note that we install the 16bitLittleEndian Converter if needed. */
if ( spec->format == 0x8010 ) {
audio_dbh.dbhCompressionID = fixedCompression;
audio_dbh.dbhFormat = k16BitLittleEndianFormat;
}
/* allocate the 2 double-back buffers */
for ( i=0; i<2; ++i ) {
audio_buf[i] = calloc(1, sizeof(SndDoubleBuffer)+spec->size);
if ( audio_buf[i] == NULL ) {
SDL_OutOfMemory();
return(-1);
}
audio_buf[i]->dbNumFrames = spec->samples;
audio_buf[i]->dbFlags = dbBufferReady;
audio_buf[i]->dbUserInfo[0] = (long)this;
audio_dbh.dbhBufferPtr[i] = audio_buf[i];
}
/* Create the sound manager channel */
channel = (SndChannelPtr)malloc(sizeof(*channel));
if ( channel == NULL ) {
SDL_OutOfMemory();
return(-1);
}
if ( spec->channels >= 2 ) {
initOptions = initStereo;
} else {
initOptions = initMono;
}
channel->userInfo = 0;
channel->qLength = 128;
if ( SndNewChannel(&channel, sampledSynth, initOptions, 0L) != noErr ) {
SDL_SetError("Unable to create audio channel");
free(channel);
channel = NULL;
return(-1);
}
/* Start playback */
if ( SndPlayDoubleBuffer(channel, (SndDoubleBufferHeaderPtr)&audio_dbh)
!= noErr ) {
SDL_SetError("Unable to play double buffered audio");
return(-1);
}
return 1;
}
static int
DART_OpenDevice(_THIS, const char *devname, int iscapture)
{
SDL_AudioFormat test_format = SDL_FirstAudioFormat(_this->spec.format);
int valid_datatype = 0;
MCI_AMP_OPEN_PARMS AmpOpenParms;
int iDeviceOrd = 0; // Default device to be used
int bOpenShared = 1; // Try opening it shared
int iBits = 16; // Default is 16 bits signed
int iFreq = 44100; // Default is 44KHz
int iChannels = 2; // Default is 2 channels (Stereo)
int iNumBufs = 2; // Number of audio buffers: 2
int iBufSize;
int iOpenMode;
int iSilence;
int rc;
/* Initialize all variables that we clean on shutdown */
_this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *_this->hidden));
if (_this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(_this->hidden, 0, (sizeof *_this->hidden));
// First thing is to try to open a given DART device!
SDL_memset(&AmpOpenParms, 0, sizeof(MCI_AMP_OPEN_PARMS));
// pszDeviceType should contain the device type in low word, and device ordinal in high word!
AmpOpenParms.pszDeviceType =
(PSZ) (MCI_DEVTYPE_AUDIO_AMPMIX | (iDeviceOrd << 16));
iOpenMode = MCI_WAIT | MCI_OPEN_TYPE_ID;
if (bOpenShared)
iOpenMode |= MCI_OPEN_SHAREABLE;
rc = mciSendCommand(0, MCI_OPEN, iOpenMode, (PVOID) & AmpOpenParms, 0);
if (rc != MCIERR_SUCCESS) { // No audio available??
DART_CloseDevice(_this);
SDL_SetError("DART: Couldn't open audio device.");
return 0;
}
// Save the device ID we got from DART!
// We will use this in the next calls!
_this->hidden->iCurrDeviceOrd = iDeviceOrd = AmpOpenParms.usDeviceID;
// Determine the audio parameters from the AudioSpec
if (_this->spec.channels > 4)
_this->spec.channels = 4;
while ((!valid_datatype) && (test_format)) {
_this->spec.format = test_format;
valid_datatype = 1;
switch (test_format) {
case AUDIO_U8:
// Unsigned 8 bit audio data
iSilence = 0x80;
_this->hidden->iCurrBits = iBits = 8;
break;
case AUDIO_S16LSB:
// Signed 16 bit audio data
iSilence = 0x00;
_this->hidden->iCurrBits = iBits = 16;
break;
// !!! FIXME: int32?
default:
valid_datatype = 0;
test_format = SDL_NextAudioFormat();
break;
}
}
if (!valid_datatype) { // shouldn't happen, but just in case...
// Close DART, and exit with error code!
DART_CloseDevice(_this);
SDL_SetError("Unsupported audio format");
return 0;
}
_this->hidden->iCurrFreq = iFreq = _this->spec.freq;
_this->hidden->iCurrChannels = iChannels = _this->spec.channels;
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&_this->spec);
_this->hidden->iCurrBufSize = iBufSize = _this->spec.size;
// Now query this device if it supports the given freq/bits/channels!
SDL_memset(&(_this->hidden->MixSetupParms), 0,
sizeof(MCI_MIXSETUP_PARMS));
_this->hidden->MixSetupParms.ulBitsPerSample = iBits;
_this->hidden->MixSetupParms.ulFormatTag = MCI_WAVE_FORMAT_PCM;
_this->hidden->MixSetupParms.ulSamplesPerSec = iFreq;
_this->hidden->MixSetupParms.ulChannels = iChannels;
_this->hidden->MixSetupParms.ulFormatMode = MCI_PLAY;
_this->hidden->MixSetupParms.ulDeviceType = MCI_DEVTYPE_WAVEFORM_AUDIO;
_this->hidden->MixSetupParms.pmixEvent = DARTEventFunc;
rc = mciSendCommand(iDeviceOrd, MCI_MIXSETUP,
MCI_WAIT | MCI_MIXSETUP_QUERYMODE,
&(_this->hidden->MixSetupParms), 0);
if (rc != MCIERR_SUCCESS) { // The device cannot handle this format!
// Close DART, and exit with error code!
DART_CloseDevice(_this);
SDL_SetError("Audio device doesn't support requested audio format");
return 0;
}
// The device can handle this format, so initialize!
rc = mciSendCommand(iDeviceOrd, MCI_MIXSETUP,
MCI_WAIT | MCI_MIXSETUP_INIT,
&(_this->hidden->MixSetupParms), 0);
if (rc != MCIERR_SUCCESS) { // The device could not be opened!
// Close DART, and exit with error code!
DART_CloseDevice(_this);
SDL_SetError("Audio device could not be set up");
return 0;
}
// Ok, the device is initialized.
// Now we should allocate buffers. For this, we need a place where
// the buffer descriptors will be:
_this->hidden->pMixBuffers =
(MCI_MIX_BUFFER *) SDL_malloc(sizeof(MCI_MIX_BUFFER) * iNumBufs);
if (!(_this->hidden->pMixBuffers)) { // Not enough memory!
// Close DART, and exit with error code!
DART_CloseDevice(_this);
SDL_OutOfMemory();
return 0;
}
// Now that we have the place for buffer list, we can ask DART for the
// buffers!
_this->hidden->BufferParms.ulNumBuffers = iNumBufs; // Number of buffers
_this->hidden->BufferParms.ulBufferSize = iBufSize; // each with this size
_this->hidden->BufferParms.pBufList = _this->hidden->pMixBuffers; // getting descriptorts into this list
// Allocate buffers!
rc = mciSendCommand(iDeviceOrd, MCI_BUFFER,
MCI_WAIT | MCI_ALLOCATE_MEMORY,
&(_this->hidden->BufferParms), 0);
if ((rc != MCIERR_SUCCESS)
|| (iNumBufs != _this->hidden->BufferParms.ulNumBuffers)
|| (_this->hidden->BufferParms.ulBufferSize == 0)) { // Could not allocate memory!
// Close DART, and exit with error code!
DART_CloseDevice(_this);
SDL_SetError("DART could not allocate buffers");
return 0;
}
_this->hidden->iCurrNumBufs = iNumBufs;
// Ok, we have all the buffers allocated, let's mark them!
{
int i;
for (i = 0; i < iNumBufs; i++) {
pMixBufferDesc pBufferDesc =
(pMixBufferDesc) SDL_malloc(sizeof(tMixBufferDesc));;
// Check if this buffer was really allocated by DART
if ((!(_this->hidden->pMixBuffers[i].pBuffer))
|| (!pBufferDesc)) { // Wrong buffer!
DART_CloseDevice(_this);
SDL_SetError("Error at internal buffer check");
return 0;
}
pBufferDesc->iBufferUsage = BUFFER_EMPTY;
pBufferDesc->pSDLAudioDevice = _this;
_this->hidden->pMixBuffers[i].ulBufferLength =
_this->hidden->BufferParms.ulBufferSize;
_this->hidden->pMixBuffers[i].ulUserParm = (ULONG) pBufferDesc; // User parameter: Description of buffer
_this->hidden->pMixBuffers[i].ulFlags = 0; // Some stuff should be flagged here for DART, like end of
// audio data, but as we will continously send
// audio data, there will be no end.:)
SDL_memset(_this->hidden->pMixBuffers[i].pBuffer, iSilence,
iBufSize);
}
}
_this->hidden->iNextFreeBuffer = 0;
_this->hidden->iLastPlayedBuf = -1;
// Create event semaphore
if (DosCreateEventSem
(NULL, &(_this->hidden->hevAudioBufferPlayed), 0, FALSE) != NO_ERROR)
{
DART_CloseDevice(_this);
SDL_SetError("Could not create event semaphore");
return 0;
}
return 1;
}
static int PULSE_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
int state;
Uint16 test_format;
pa_sample_spec paspec;
pa_buffer_attr paattr;
pa_channel_map pacmap;
pa_stream_flags_t flags = 0;
paspec.format = PA_SAMPLE_INVALID;
for ( test_format = SDL_FirstAudioFormat(spec->format); test_format; ) {
switch ( test_format ) {
case AUDIO_U8:
paspec.format = PA_SAMPLE_U8;
break;
case AUDIO_S16LSB:
paspec.format = PA_SAMPLE_S16LE;
break;
case AUDIO_S16MSB:
paspec.format = PA_SAMPLE_S16BE;
break;
}
if ( paspec.format != PA_SAMPLE_INVALID )
break;
test_format = SDL_NextAudioFormat();
}
if (paspec.format == PA_SAMPLE_INVALID ) {
SDL_SetError("Couldn't find any suitable audio formats");
return(-1);
}
spec->format = test_format;
paspec.channels = spec->channels;
paspec.rate = spec->freq;
#ifdef PA_STREAM_ADJUST_LATENCY
spec->samples /= 2;
#endif
SDL_CalculateAudioSpec(spec);
mixlen = spec->size;
mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
if ( mixbuf == NULL ) {
return(-1);
}
SDL_memset(mixbuf, spec->silence, spec->size);
#ifdef PA_STREAM_ADJUST_LATENCY
paattr.tlength = mixlen * 4;
paattr.prebuf = -1;
paattr.maxlength = -1;
paattr.minreq = mixlen;
flags = PA_STREAM_ADJUST_LATENCY;
#else
paattr.tlength = mixlen*2;
paattr.prebuf = mixlen*2;
paattr.maxlength = mixlen*2;
paattr.minreq = mixlen;
#endif
SDL_NAME(pa_channel_map_init_auto)(
&pacmap, spec->channels, PA_CHANNEL_MAP_WAVEEX);
if (!(mainloop = SDL_NAME(pa_mainloop_new)())) {
PULSE_CloseAudio(this);
SDL_SetError("pa_mainloop_new() failed");
return(-1);
}
if (this->hidden->caption == NULL) {
char *title = NULL;
SDL_WM_GetCaption(&title, NULL);
PULSE_SetCaption(this, title);
}
mainloop_api = SDL_NAME(pa_mainloop_get_api)(mainloop);
if (!(context = SDL_NAME(pa_context_new)(mainloop_api,
this->hidden->caption))) {
PULSE_CloseAudio(this);
SDL_SetError("pa_context_new() failed");
return(-1);
}
if (SDL_NAME(pa_context_connect)(context, NULL, 0, NULL) < 0) {
PULSE_CloseAudio(this);
SDL_SetError("Could not setup connection to PulseAudio");
return(-1);
}
do {
if (SDL_NAME(pa_mainloop_iterate)(mainloop, 1, NULL) < 0) {
PULSE_CloseAudio(this);
SDL_SetError("pa_mainloop_iterate() failed");
return(-1);
}
state = SDL_NAME(pa_context_get_state)(context);
if (!PA_CONTEXT_IS_GOOD(state)) {
PULSE_CloseAudio(this);
SDL_SetError("Could not connect to PulseAudio");
return(-1);
}
} while (state != PA_CONTEXT_READY);
stream = SDL_NAME(pa_stream_new)(
context,
"Simple DirectMedia Layer",
&paspec,
&pacmap
);
if ( stream == NULL ) {
PULSE_CloseAudio(this);
SDL_SetError("Could not setup PulseAudio stream");
return(-1);
}
if (SDL_NAME(pa_stream_connect_playback)(stream, NULL, &paattr, flags,
NULL, NULL) < 0) {
PULSE_CloseAudio(this);
SDL_SetError("Could not connect PulseAudio stream");
return(-1);
}
do {
if (SDL_NAME(pa_mainloop_iterate)(mainloop, 1, NULL) < 0) {
PULSE_CloseAudio(this);
SDL_SetError("pa_mainloop_iterate() failed");
return(-1);
}
state = SDL_NAME(pa_stream_get_state)(stream);
if (!PA_STREAM_IS_GOOD(state)) {
PULSE_CloseAudio(this);
SDL_SetError("Could not create to PulseAudio stream");
return(-1);
}
} while (state != PA_STREAM_READY);
return(0);
}
static int DSP_ReopenAudio(_THIS, const char *audiodev, int format,
SDL_AudioSpec *spec)
{
int frag_spec;
int value;
/* Close and then reopen the audio device */
close(audio_fd);
audio_fd = open(audiodev, O_WRONLY, 0);
if ( audio_fd < 0 ) {
SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
return(-1);
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(spec);
/* Determine the power of two of the fragment size */
for ( frag_spec = 0; (0x01<<frag_spec) < spec->size; ++frag_spec );
if ( (0x01<<frag_spec) != spec->size ) {
SDL_SetError("Fragment size must be a power of two");
return(-1);
}
frag_spec |= 0x00020000; /* two fragments, for low latency */
/* Set the audio buffering parameters */
#ifdef DEBUG_AUDIO
fprintf(stderr, "Requesting %d fragments of size %d\n",
(frag_spec >> 16), 1<<(frag_spec&0xFFFF));
#endif
if ( ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_spec) < 0 ) {
fprintf(stderr, "Warning: Couldn't set audio fragment size\n");
}
#ifdef DEBUG_AUDIO
{ audio_buf_info info;
ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &info);
fprintf(stderr, "fragments = %d\n", info.fragments);
fprintf(stderr, "fragstotal = %d\n", info.fragstotal);
fprintf(stderr, "fragsize = %d\n", info.fragsize);
fprintf(stderr, "bytes = %d\n", info.bytes);
}
#endif
/* Set the audio format */
value = format;
if ( (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) ||
(value != format) ) {
SDL_SetError("Couldn't set audio format");
return(-1);
}
/* Set the number of channels of output */
value = spec->channels;
#ifdef SNDCTL_DSP_CHANNELS
if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &value) < 0 ) {
#endif
value = (spec->channels > 1);
ioctl(audio_fd, SNDCTL_DSP_STEREO, &value);
value = (value ? 2 : 1);
#ifdef SNDCTL_DSP_CHANNELS
}
#endif
if ( value != spec->channels ) {
SDL_SetError("Couldn't set audio channels");
return(-1);
}
/* Set the DSP frequency */
value = spec->freq;
if ( ioctl(audio_fd, SOUND_PCM_WRITE_RATE, &value) < 0 ) {
SDL_SetError("Couldn't set audio frequency");
return(-1);
}
spec->freq = value;
/* We successfully re-opened the audio */
return(0);
}
static int
XAUDIO2_OpenDevice(_THIS, const char *devname, int iscapture)
{
HRESULT result = S_OK;
WAVEFORMATEX waveformat;
int valid_format = 0;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
IXAudio2 *ixa2 = NULL;
IXAudio2SourceVoice *source = NULL;
UINT32 devId = 0; /* 0 == system default device. */
static IXAudio2VoiceCallbackVtbl callbacks_vtable = {
VoiceCBOnVoiceProcessPassStart,
VoiceCBOnVoiceProcessPassEnd,
VoiceCBOnStreamEnd,
VoiceCBOnBufferStart,
VoiceCBOnBufferEnd,
VoiceCBOnLoopEnd,
VoiceCBOnVoiceError
};
static IXAudio2VoiceCallback callbacks = { &callbacks_vtable };
// add WIN_CoInitialize() and WIN_CoUninitialize here;
// to avoid XAudio2Create return hr 0x800401f0 ипн╢╣Всц CoInitialize;
if (iscapture) {
SDL_SetError("XAudio2: capture devices unsupported.");
return 0;
} else if (XAudio2Create(&ixa2, 0, XAUDIO2_DEFAULT_PROCESSOR) != S_OK) {
SDL_SetError("XAudio2: XAudio2Create() failed.");
return 0;
}
if (devname != NULL) {
UINT32 devcount = 0;
UINT32 i = 0;
if (IXAudio2_GetDeviceCount(ixa2, &devcount) != S_OK) {
IXAudio2_Release(ixa2);
SDL_SetError("XAudio2: IXAudio2_GetDeviceCount() failed.");
return 0;
}
for (i = 0; i < devcount; i++) {
XAUDIO2_DEVICE_DETAILS details;
if (IXAudio2_GetDeviceDetails(ixa2, i, &details) == S_OK) {
char *str = utf16_to_utf8(details.DisplayName);
if (str != NULL) {
const int match = (SDL_strcmp(str, devname) == 0);
SDL_free(str);
if (match) {
devId = i;
break;
}
}
}
}
if (i == devcount) {
IXAudio2_Release(ixa2);
SDL_SetError("XAudio2: Requested device not found.");
return 0;
}
}
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
IXAudio2_Release(ixa2);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
this->hidden->ixa2 = ixa2;
this->hidden->semaphore = CreateSemaphore(NULL, 1, 2, NULL);
if (this->hidden->semaphore == NULL) {
XAUDIO2_CloseDevice(this);
SDL_SetError("XAudio2: CreateSemaphore() failed!");
return 0;
}
while ((!valid_format) && (test_format)) {
switch (test_format) {
case AUDIO_U8:
case AUDIO_S16:
case AUDIO_S32:
case AUDIO_F32:
this->spec.format = test_format;
valid_format = 1;
break;
}
test_format = SDL_NextAudioFormat();
}
if (!valid_format) {
XAUDIO2_CloseDevice(this);
SDL_SetError("XAudio2: Unsupported audio format");
return 0;
}
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* We feed a Source, it feeds the Mastering, which feeds the device. */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_malloc(2 * this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
XAUDIO2_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
this->hidden->nextbuf = this->hidden->mixbuf;
SDL_memset(this->hidden->mixbuf, 0, 2 * this->hidden->mixlen);
/* We use XAUDIO2_DEFAULT_CHANNELS instead of this->spec.channels. On
Xbox360, this means 5.1 output, but on Windows, it means "figure out
what the system has." It might be preferable to let XAudio2 blast
stereo output to appropriate surround sound configurations
instead of clamping to 2 channels, even though we'll configure the
Source Voice for whatever number of channels you supply. */
result = IXAudio2_CreateMasteringVoice(ixa2, &this->hidden->mastering,
XAUDIO2_DEFAULT_CHANNELS,
this->spec.freq, 0, devId, NULL);
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
SDL_SetError("XAudio2: Couldn't create mastering voice");
return 0;
}
SDL_zero(waveformat);
if (SDL_AUDIO_ISFLOAT(this->spec.format)) {
waveformat.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
} else {
waveformat.wFormatTag = WAVE_FORMAT_PCM;
}
waveformat.wBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
waveformat.nChannels = this->spec.channels;
waveformat.nSamplesPerSec = this->spec.freq;
waveformat.nBlockAlign =
waveformat.nChannels * (waveformat.wBitsPerSample / 8);
waveformat.nAvgBytesPerSec =
waveformat.nSamplesPerSec * waveformat.nBlockAlign;
result = IXAudio2_CreateSourceVoice(ixa2, &source, &waveformat,
XAUDIO2_VOICE_NOSRC |
XAUDIO2_VOICE_NOPITCH,
1.0f, &callbacks, NULL, NULL);
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
SDL_SetError("XAudio2: Couldn't create source voice");
return 0;
}
this->hidden->source = source;
/* Start everything playing! */
result = IXAudio2_StartEngine(ixa2);
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
SDL_SetError("XAudio2: Couldn't start engine");
return 0;
}
result = IXAudio2SourceVoice_Start(source, 0, XAUDIO2_COMMIT_NOW);
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
SDL_SetError("XAudio2: Couldn't start source voice");
return 0;
}
return 1; /* good to go. */
}
int DIB_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
MMRESULT result;
int i;
WAVEFORMATEX waveformat;
/* Initialize the wavebuf structures for closing */
sound = NULL;
audio_sem = NULL;
for ( i = 0; i < NUM_BUFFERS; ++i )
wavebuf[i].dwUser = 0xFFFF;
mixbuf = NULL;
/* Set basic WAVE format parameters */
SDL_memset(&waveformat, 0, sizeof(waveformat));
waveformat.wFormatTag = WAVE_FORMAT_PCM;
/* Determine the audio parameters from the AudioSpec */
switch ( spec->format & 0xFF ) {
case 8:
/* Unsigned 8 bit audio data */
spec->format = AUDIO_U8;
waveformat.wBitsPerSample = 8;
break;
case 16:
/* Signed 16 bit audio data */
spec->format = AUDIO_S16;
waveformat.wBitsPerSample = 16;
break;
default:
SDL_SetError("Unsupported audio format");
return(-1);
}
waveformat.nChannels = spec->channels;
waveformat.nSamplesPerSec = spec->freq;
waveformat.nBlockAlign =
waveformat.nChannels * (waveformat.wBitsPerSample/8);
waveformat.nAvgBytesPerSec =
waveformat.nSamplesPerSec * waveformat.nBlockAlign;
/* Check the buffer size -- minimum of 1/4 second (word aligned) */
if ( spec->samples < (spec->freq/4) )
spec->samples = ((spec->freq/4)+3)&~3;
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(spec);
/* Open the audio device */
result = waveOutOpen(&sound, WAVE_MAPPER, &waveformat,
(DWORD_PTR)FillSound, (DWORD_PTR)this, CALLBACK_FUNCTION);
if ( result != MMSYSERR_NOERROR ) {
SetMMerror("waveOutOpen()", result);
return(-1);
}
#ifdef SOUND_DEBUG
/* Check the sound device we retrieved */
{
WAVEOUTCAPS caps;
result = waveOutGetDevCaps((UINT)sound, &caps, sizeof(caps));
if ( result != MMSYSERR_NOERROR ) {
SetMMerror("waveOutGetDevCaps()", result);
return(-1);
}
printf("Audio device: %s\n", caps.szPname);
}
#endif
/* Create the audio buffer semaphore */
#if defined(_WIN32_WCE) && (_WIN32_WCE < 300)
audio_sem = CreateSemaphoreCE(NULL, NUM_BUFFERS-1, NUM_BUFFERS, NULL);
#else
audio_sem = CreateSemaphore(NULL, NUM_BUFFERS-1, NUM_BUFFERS, NULL);
#endif
if ( audio_sem == NULL ) {
SDL_SetError("Couldn't create semaphore");
return(-1);
}
/* Create the sound buffers */
mixbuf = (Uint8 *)SDL_malloc(NUM_BUFFERS*spec->size);
if ( mixbuf == NULL ) {
SDL_SetError("Out of memory");
return(-1);
}
for ( i = 0; i < NUM_BUFFERS; ++i ) {
SDL_memset(&wavebuf[i], 0, sizeof(wavebuf[i]));
wavebuf[i].lpData = (LPSTR) &mixbuf[i*spec->size];
wavebuf[i].dwBufferLength = spec->size;
wavebuf[i].dwFlags = WHDR_DONE;
result = waveOutPrepareHeader(sound, &wavebuf[i],
sizeof(wavebuf[i]));
if ( result != MMSYSERR_NOERROR ) {
SetMMerror("waveOutPrepareHeader()", result);
return(-1);
}
}
/* Ready to go! */
next_buffer = 0;
return(0);
}
static int
QSA_OpenDevice(_THIS, const char *devname, int iscapture)
{
int status = 0;
int format = 0;
SDL_AudioFormat test_format = 0;
int found = 0;
snd_pcm_channel_setup_t csetup;
snd_pcm_channel_params_t cparams;
/* Initialize all variables that we clean on shutdown */
this->hidden =
(struct SDL_PrivateAudioData *) SDL_calloc(1,
(sizeof
(struct
SDL_PrivateAudioData)));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, sizeof(struct SDL_PrivateAudioData));
/* Initialize channel transfer parameters to default */
QSA_InitAudioParams(&cparams);
/* Initialize channel direction: capture or playback */
this->hidden->iscapture = iscapture;
/* Find deviceid and cardid by device name for playback */
if ((!this->hidden->iscapture) && (devname != NULL)) {
uint32_t device;
int32_t status;
/* Search in the playback devices */
device = 0;
do {
status = SDL_strcmp(qsa_playback_device[device].name, devname);
if (status == 0) {
/* Found requested device */
this->hidden->deviceno = qsa_playback_device[device].deviceno;
this->hidden->cardno = qsa_playback_device[device].cardno;
break;
}
device++;
if (device >= qsa_playback_devices) {
QSA_CloseDevice(this);
return SDL_SetError("No such playback device");
}
} while (1);
}
/* Find deviceid and cardid by device name for capture */
if ((this->hidden->iscapture) && (devname != NULL)) {
/* Search in the capture devices */
uint32_t device;
int32_t status;
/* Searching in the playback devices */
device = 0;
do {
status = SDL_strcmp(qsa_capture_device[device].name, devname);
if (status == 0) {
/* Found requested device */
this->hidden->deviceno = qsa_capture_device[device].deviceno;
this->hidden->cardno = qsa_capture_device[device].cardno;
break;
}
device++;
if (device >= qsa_capture_devices) {
QSA_CloseDevice(this);
return SDL_SetError("No such capture device");
}
} while (1);
}
/* Check if SDL requested default audio device */
if (devname == NULL) {
/* Open system default audio device */
if (!this->hidden->iscapture) {
status = snd_pcm_open_preferred(&this->hidden->audio_handle,
&this->hidden->cardno,
&this->hidden->deviceno,
SND_PCM_OPEN_PLAYBACK);
} else {
status = snd_pcm_open_preferred(&this->hidden->audio_handle,
&this->hidden->cardno,
&this->hidden->deviceno,
SND_PCM_OPEN_CAPTURE);
}
} else {
/* Open requested audio device */
if (!this->hidden->iscapture) {
status =
snd_pcm_open(&this->hidden->audio_handle,
this->hidden->cardno, this->hidden->deviceno,
SND_PCM_OPEN_PLAYBACK);
} else {
status =
snd_pcm_open(&this->hidden->audio_handle,
this->hidden->cardno, this->hidden->deviceno,
SND_PCM_OPEN_CAPTURE);
}
}
/* Check if requested device is opened */
if (status < 0) {
this->hidden->audio_handle = NULL;
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_open", status);
}
if (!QSA_CheckBuggyCards(this, QSA_MMAP_WORKAROUND)) {
/* Disable QSA MMAP plugin for buggy audio drivers */
status =
snd_pcm_plugin_set_disable(this->hidden->audio_handle,
PLUGIN_DISABLE_MMAP);
if (status < 0) {
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_plugin_set_disable", status);
}
}
/* Try for a closest match on audio format */
format = 0;
/* can't use format as SND_PCM_SFMT_U8 = 0 in qsa */
found = 0;
for (test_format = SDL_FirstAudioFormat(this->spec.format); !found;) {
/* if match found set format to equivalent QSA format */
switch (test_format) {
case AUDIO_U8:
{
format = SND_PCM_SFMT_U8;
found = 1;
}
break;
case AUDIO_S8:
{
format = SND_PCM_SFMT_S8;
found = 1;
}
break;
case AUDIO_S16LSB:
{
format = SND_PCM_SFMT_S16_LE;
found = 1;
}
break;
case AUDIO_S16MSB:
{
format = SND_PCM_SFMT_S16_BE;
found = 1;
}
break;
case AUDIO_U16LSB:
{
format = SND_PCM_SFMT_U16_LE;
found = 1;
}
break;
case AUDIO_U16MSB:
{
format = SND_PCM_SFMT_U16_BE;
found = 1;
}
break;
case AUDIO_S32LSB:
{
format = SND_PCM_SFMT_S32_LE;
found = 1;
}
break;
case AUDIO_S32MSB:
{
format = SND_PCM_SFMT_S32_BE;
found = 1;
}
break;
case AUDIO_F32LSB:
{
format = SND_PCM_SFMT_FLOAT_LE;
found = 1;
}
break;
case AUDIO_F32MSB:
{
format = SND_PCM_SFMT_FLOAT_BE;
found = 1;
}
break;
default:
{
break;
}
}
if (!found) {
test_format = SDL_NextAudioFormat();
}
}
/* assumes test_format not 0 on success */
if (test_format == 0) {
QSA_CloseDevice(this);
return SDL_SetError("QSA: Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
/* Set the audio format */
cparams.format.format = format;
/* Set mono/stereo/4ch/6ch/8ch audio */
cparams.format.voices = this->spec.channels;
/* Set rate */
cparams.format.rate = this->spec.freq;
/* Setup the transfer parameters according to cparams */
status = snd_pcm_plugin_params(this->hidden->audio_handle, &cparams);
if (status < 0) {
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_channel_params", status);
}
/* Make sure channel is setup right one last time */
SDL_memset(&csetup, 0, sizeof(csetup));
if (!this->hidden->iscapture) {
csetup.channel = SND_PCM_CHANNEL_PLAYBACK;
} else {
csetup.channel = SND_PCM_CHANNEL_CAPTURE;
}
/* Setup an audio channel */
if (snd_pcm_plugin_setup(this->hidden->audio_handle, &csetup) < 0) {
QSA_CloseDevice(this);
return SDL_SetError("QSA: Unable to setup channel");
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
this->hidden->pcm_len = this->spec.size;
if (this->hidden->pcm_len == 0) {
this->hidden->pcm_len =
csetup.buf.block.frag_size * this->spec.channels *
(snd_pcm_format_width(format) / 8);
}
/*
* Allocate memory to the audio buffer and initialize with silence
* (Note that buffer size must be a multiple of fragment size, so find
* closest multiple)
*/
this->hidden->pcm_buf =
(Uint8 *) SDL_AllocAudioMem(this->hidden->pcm_len);
if (this->hidden->pcm_buf == NULL) {
QSA_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->pcm_buf, this->spec.silence,
this->hidden->pcm_len);
/* get the file descriptor */
if (!this->hidden->iscapture) {
this->hidden->audio_fd =
snd_pcm_file_descriptor(this->hidden->audio_handle,
SND_PCM_CHANNEL_PLAYBACK);
} else {
this->hidden->audio_fd =
snd_pcm_file_descriptor(this->hidden->audio_handle,
SND_PCM_CHANNEL_CAPTURE);
}
if (this->hidden->audio_fd < 0) {
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_file_descriptor", status);
}
/* Prepare an audio channel */
if (!this->hidden->iscapture) {
/* Prepare audio playback */
status =
snd_pcm_plugin_prepare(this->hidden->audio_handle,
SND_PCM_CHANNEL_PLAYBACK);
} else {
/* Prepare audio capture */
status =
snd_pcm_plugin_prepare(this->hidden->audio_handle,
SND_PCM_CHANNEL_CAPTURE);
}
if (status < 0) {
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_plugin_prepare", status);
}
/* We're really ready to rock and roll. :-) */
return 0;
}
static int AL_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
ALconfig audio_config;
#ifdef OLD_IRIX_AUDIO
long audio_param[2];
#else
ALpv audio_param;
#endif
int width;
/* Determine the audio parameters from the AudioSpec */
switch ( spec->format & 0xFF ) {
case 8: { /* Signed 8 bit audio data */
spec->format = AUDIO_S8;
width = AL_SAMPLE_8;
}
break;
case 16: { /* Signed 16 bit audio data */
spec->format = AUDIO_S16MSB;
width = AL_SAMPLE_16;
}
break;
default: {
SDL_SetError("Unsupported audio format");
return(-1);
}
}
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(spec);
/* Set output frequency */
#ifdef OLD_IRIX_AUDIO
audio_param[0] = AL_OUTPUT_RATE;
audio_param[1] = spec->freq;
if( ALsetparams(AL_DEFAULT_DEVICE, audio_param, 2) < 0 ) {
#else
audio_param.param = AL_RATE;
audio_param.value.i = spec->freq;
if( alSetParams(AL_DEFAULT_OUTPUT, &audio_param, 1) < 0 ) {
#endif
SDL_SetError("alSetParams failed");
return(-1);
}
/* Open the audio port with the requested frequency */
audio_port = NULL;
audio_config = alNewConfig();
if ( audio_config &&
(alSetSampFmt(audio_config, AL_SAMPFMT_TWOSCOMP) >= 0) &&
(alSetWidth(audio_config, width) >= 0) &&
(alSetQueueSize(audio_config, spec->samples*2) >= 0) &&
(alSetChannels(audio_config, spec->channels) >= 0) ) {
audio_port = alOpenPort("SDL audio", "w", audio_config);
}
alFreeConfig(audio_config);
if( audio_port == NULL ) {
SDL_SetError("Unable to open audio port");
return(-1);
}
/* Allocate mixing buffer */
mixbuf = (Uint8 *)SDL_AllocAudioMem(spec->size);
if ( mixbuf == NULL ) {
SDL_OutOfMemory();
return(-1);
}
memset(mixbuf, spec->silence, spec->size);
/* We're ready to rock and roll. :-) */
return(0);
}
static int Mac_OpenAudio(_THIS, SDL_AudioSpec *spec) {
SndCallBackUPP callback;
int sample_bits;
int i;
long initOptions;
/* Very few conversions are required, but... */
switch (spec->format) {
case AUDIO_S8:
spec->format = AUDIO_U8;
break;
case AUDIO_U16LSB:
spec->format = AUDIO_S16LSB;
break;
case AUDIO_U16MSB:
spec->format = AUDIO_S16MSB;
break;
}
SDL_CalculateAudioSpec(spec);
/* initialize bufferCmd header */
memset (&header, 0, sizeof(header));
callback = NewSndCallBackUPP (callBackProc);
sample_bits = spec->size / spec->samples / spec->channels * 8;
#ifdef DEBUG_AUDIO
fprintf(stderr,
"Audio format 0x%x, channels = %d, sample_bits = %d, frequency = %d\n",
spec->format, spec->channels, sample_bits, spec->freq);
#endif /* DEBUG_AUDIO */
header.numChannels = spec->channels;
header.sampleSize = sample_bits;
header.sampleRate = spec->freq << 16;
header.numFrames = spec->samples;
header.encode = cmpSH;
/* Note that we install the 16bitLittleEndian Converter if needed. */
if ( spec->format == 0x8010 ) {
header.compressionID = fixedCompression;
header.format = k16BitLittleEndianFormat;
}
/* allocate 2 buffers */
for (i=0; i<2; i++) {
buffer[i] = (UInt8*)malloc (sizeof(UInt8) * spec->size);
if (buffer[i] == NULL) {
SDL_OutOfMemory();
return (-1);
}
memset (buffer[i], 0, spec->size);
}
/* Create the sound manager channel */
channel = (SndChannelPtr)malloc(sizeof(*channel));
if ( channel == NULL ) {
SDL_OutOfMemory();
return(-1);
}
if ( spec->channels >= 2 ) {
initOptions = initStereo;
} else {
initOptions = initMono;
}
channel->userInfo = (long)this;
channel->qLength = 128;
if ( SndNewChannel(&channel, sampledSynth, initOptions, callback) != noErr ) {
SDL_SetError("Unable to create audio channel");
free(channel);
channel = NULL;
return(-1);
}
/* start playback */
{
SndCommand cmd;
cmd.cmd = callBackCmd;
cmd.param2 = 0;
running = 1;
SndDoCommand (channel, &cmd, 0);
}
return 1;
}
