static int MPFAUD_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
int format;
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);
switch (spec->format & 0xff) {
case 8:format=8;break;
case 16:format=16;break;
default:
SDL_SetError("Unsupported audio format");
return -1;
}
sys_sound_init(spec->freq,format,spec->channels);
/* We're ready to rock and roll. :-) */
return(0);
}
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 DISKAUD_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
const char *fname = DISKAUD_GetOutputFilename();
/* Open the audio device */
this->hidden->audio_fd = open(fname, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR);
if ( this->hidden->audio_fd < 0 ) {
SDL_SetError("Couldn't open %s: %s", fname, strerror(errno));
return(-1);
}
fprintf(stderr, "WARNING: You are using the SDL disk writer"
" audio driver!\n Writing to file [%s].\n", fname);
/* 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);
/* We're ready to rock and roll. :-) */
return(0);
}
static int DISKAUD_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
const char *fname = DISKAUD_GetOutputFilename();
/* Open the audio device */
this->hidden->output = SDL_RWFromFile(fname, "wb");
if ( this->hidden->output == NULL ) {
return(-1);
}
#if HAVE_STDIO_H
fprintf(stderr, "WARNING: You are using the SDL disk writer"
" audio driver!\n Writing to file [%s].\n", fname);
#endif
/* 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);
/* We're ready to rock and roll. :-) */
return(0);
}
static int ANDROIDAUDIOTRACK_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
initAndroidAudio(this, spec->format, spec->freq, spec->channels, spec->size);
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);
/* We're ready to rock and roll. :-) */
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
DISKAUD_OpenDevice(_THIS, const char *devname, int iscapture)
{
const char *envr = SDL_getenv(DISKENVR_WRITEDELAY);
const char *fname = DISKAUD_GetOutputFilename(devname);
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->output = SDL_RWFromFile(fname, "wb");
if (this->hidden->output == NULL) {
DISKAUD_CloseDevice(this);
return 0;
}
/* Allocate mixing buffer */
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
DISKAUD_CloseDevice(this);
return 0;
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
this->hidden->mixlen = this->spec.size;
this->hidden->write_delay =
(envr) ? SDL_atoi(envr) : DISKDEFAULT_WRITEDELAY;
#if HAVE_STDIO_H
fprintf(stderr,
"WARNING: You are using the SDL disk writer audio driver!\n"
" Writing to file [%s].\n", fname);
#endif
/* We're ready to rock and roll. :-) */
return 1;
}
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
SUNAUDIO_OpenDevice(_THIS, void *handle, 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) {
return SDL_SetError("No such audio device");
}
}
/* 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 */
this->hidden->audio_fd = open(devname, flags, 0);
if (this->hidden->audio_fd < 0) {
return SDL_SetError("Couldn't open %s: %s", devname, strerror(errno));
}
#ifdef AUDIO_SETINFO
int enc;
#endif
int desired_freq = this->spec.freq;
/* Determine the audio parameters from the AudioSpec */
switch (SDL_AUDIO_BITSIZE(this->spec.format)) {
case 8:
{ /* Unsigned 8 bit audio data */
this->spec.format = AUDIO_U8;
#ifdef AUDIO_SETINFO
enc = AUDIO_ENCODING_LINEAR8;
#endif
}
break;
case 16:
{ /* Signed 16 bit audio data */
this->spec.format = AUDIO_S16SYS;
#ifdef AUDIO_SETINFO
enc = AUDIO_ENCODING_LINEAR;
#endif
}
break;
default:
{
/* !!! FIXME: fallback to conversion on unsupported types! */
return SDL_SetError("Unsupported audio format");
}
}
this->hidden->audio_fmt = this->spec.format;
this->hidden->ulaw_only = 0; /* modern Suns do support linear audio */
#ifdef AUDIO_SETINFO
for (;;) {
audio_info_t info;
AUDIO_INITINFO(&info); /* init all fields to "no change" */
/* Try to set the requested settings */
info.play.sample_rate = this->spec.freq;
info.play.channels = this->spec.channels;
info.play.precision = (enc == AUDIO_ENCODING_ULAW)
? 8 : this->spec.format & 0xff;
info.play.encoding = enc;
if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) == 0) {
/* Check to be sure we got what we wanted */
if (ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info) < 0) {
return SDL_SetError("Error getting audio parameters: %s",
strerror(errno));
}
if (info.play.encoding == enc
&& info.play.precision == (this->spec.format & 0xff)
&& info.play.channels == this->spec.channels) {
/* Yow! All seems to be well! */
this->spec.freq = info.play.sample_rate;
break;
}
}
switch (enc) {
case AUDIO_ENCODING_LINEAR8:
/* unsigned 8bit apparently not supported here */
enc = AUDIO_ENCODING_LINEAR;
this->spec.format = AUDIO_S16SYS;
break; /* try again */
case AUDIO_ENCODING_LINEAR:
/* linear 16bit didn't work either, resort to µ-law */
enc = AUDIO_ENCODING_ULAW;
this->spec.channels = 1;
this->spec.freq = 8000;
this->spec.format = AUDIO_U8;
this->hidden->ulaw_only = 1;
break;
default:
/* oh well... */
return SDL_SetError("Error setting audio parameters: %s",
strerror(errno));
}
}
#endif /* AUDIO_SETINFO */
this->hidden->written = 0;
/* We can actually convert on-the-fly to U-Law */
if (this->hidden->ulaw_only) {
this->spec.freq = desired_freq;
this->hidden->fragsize = (this->spec.samples * 1000) /
(this->spec.freq / 8);
this->hidden->frequency = 8;
this->hidden->ulaw_buf = (Uint8 *) SDL_malloc(this->hidden->fragsize);
if (this->hidden->ulaw_buf == NULL) {
return SDL_OutOfMemory();
}
this->spec.channels = 1;
} else {
this->hidden->fragsize = this->spec.samples;
this->hidden->frequency = this->spec.freq / 1000;
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Audio device %s U-Law only\n",
this->hidden->ulaw_only ? "is" : "is not");
fprintf(stderr, "format=0x%x chan=%d freq=%d\n",
this->spec.format, this->spec.channels, this->spec.freq);
#endif
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* Allocate mixing buffer */
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->spec.size);
if (this->hidden->mixbuf == NULL) {
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 AL_OpenAudio(_THIS, SDL_AudioSpec * spec)
{
Uint16 test_format = SDL_FirstAudioFormat(spec->format);
long width = 0;
long fmt = 0;
int valid = 0;
#ifdef OLD_IRIX_AUDIO
{
long audio_param[2];
audio_param[0] = AL_OUTPUT_RATE;
audio_param[1] = spec->freq;
valid = (ALsetparams(AL_DEFAULT_DEVICE, audio_param, 2) < 0);
}
#else
{
ALpv audio_param;
audio_param.param = AL_RATE;
audio_param.value.i = spec->freq;
valid = (alSetParams(AL_DEFAULT_OUTPUT, &audio_param, 1) < 0);
}
#endif
while ((!valid) && (test_format)) {
valid = 1;
spec->format = test_format;
switch (test_format) {
case AUDIO_S8:
width = AL_SAMPLE_8;
fmt = AL_SAMPFMT_TWOSCOMP;
break;
case AUDIO_S16SYS:
width = AL_SAMPLE_16;
fmt = AL_SAMPFMT_TWOSCOMP;
break;
default:
valid = 0;
test_format = SDL_NextAudioFormat();
break;
}
if (valid) {
ALconfig audio_config = alNewConfig();
valid = 0;
if (audio_config) {
if (alSetChannels(audio_config, spec->channels) < 0) {
if (spec->channels > 2) { /* can't handle > stereo? */
spec->channels = 2; /* try again below. */
}
}
if ((alSetSampFmt(audio_config, fmt) >= 0) &&
((!width) || (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);
if (audio_port == NULL) {
/* docs say AL_BAD_CHANNELS happens here, too. */
int err = oserror();
if (err == AL_BAD_CHANNELS) {
spec->channels = 2;
alSetChannels(audio_config, spec->channels);
audio_port = alOpenPort("SDL audio", "w",
audio_config);
}
}
if (audio_port != NULL) {
valid = 1;
}
}
alFreeConfig(audio_config);
}
}
}
if (!valid) {
SDL_SetError("Unsupported audio format");
return (-1);
}
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(spec);
/* Allocate mixing buffer */
mixbuf = (Uint8 *) SDL_AllocAudioMem(spec->size);
if (mixbuf == NULL) {
SDL_OutOfMemory();
return (-1);
}
SDL_memset(mixbuf, spec->silence, spec->size);
/* We're ready to rock and roll. :-) */
return (0);
}
static int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
char audiodev[1024];
int format;
int value;
Uint16 test_format;
/* Reset the timer synchronization flag */
frame_ticks = 0.0;
/* Open the audio device */
audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
if ( audio_fd < 0 ) {
SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
return(-1);
}
mixbuf = NULL;
#ifdef USE_BLOCKING_WRITES
/* Make the file descriptor use blocking writes with fcntl() */
{ long flags;
flags = fcntl(audio_fd, F_GETFL);
flags &= ~O_NONBLOCK;
if ( fcntl(audio_fd, F_SETFL, flags) < 0 ) {
SDL_SetError("Couldn't set audio blocking mode");
return(-1);
}
}
#endif
/* Get a list of supported hardware formats */
if ( ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &value) < 0 ) {
SDL_SetError("Couldn't get audio format list");
return(-1);
}
/* Try for a closest match on audio format */
format = 0;
for ( test_format = SDL_FirstAudioFormat(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:
if ( value & AFMT_U8 ) {
format = AFMT_U8;
}
break;
case AUDIO_S8:
if ( value & AFMT_S8 ) {
format = AFMT_S8;
}
break;
case AUDIO_S16LSB:
if ( value & AFMT_S16_LE ) {
format = AFMT_S16_LE;
}
break;
case AUDIO_S16MSB:
if ( value & AFMT_S16_BE ) {
format = AFMT_S16_BE;
}
break;
case AUDIO_U16LSB:
if ( value & AFMT_U16_LE ) {
format = AFMT_U16_LE;
}
break;
case AUDIO_U16MSB:
if ( value & AFMT_U16_BE ) {
format = AFMT_U16_BE;
}
break;
default:
format = 0;
break;
}
if ( ! format ) {
test_format = SDL_NextAudioFormat();
}
}
if ( format == 0 ) {
SDL_SetError("Couldn't find any hardware audio formats");
return(-1);
}
spec->format = test_format;
/* 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
spec->channels = value;
/* Because some drivers don't allow setting the buffer size
after setting the format, we must re-open the audio device
once we know what format and channels are supported
*/
if ( DSP_ReopenAudio(this, audiodev, format, spec) < 0 ) {
/* Error is set by DSP_ReopenAudio() */
return(-1);
}
/* Allocate mixing buffer */
mixlen = spec->size;
mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
if ( mixbuf == NULL ) {
return(-1);
}
memset(mixbuf, spec->silence, spec->size);
#ifndef USE_BLOCKING_WRITES
/* Check to see if we need to use select() workaround */
{ char *workaround;
workaround = getenv("SDL_DSP_NOSELECT");
if ( workaround ) {
frame_ticks = (float)(spec->samples*1000)/spec->freq;
next_frame = SDL_GetTicks()+frame_ticks;
}
}
#endif /* !USE_BLOCKING_WRITES */
/* 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
ESD_OpenDevice(_THIS, const char *devname, int iscapture)
{
esd_format_t format = (ESD_STREAM | ESD_PLAY);
SDL_AudioFormat test_format = 0;
int found = 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));
this->hidden->audio_fd = -1;
/* Convert audio spec to the ESD audio format */
/* Try for a closest match on audio format */
for (test_format = SDL_FirstAudioFormat(this->spec.format);
!found && test_format; test_format = SDL_NextAudioFormat()) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
found = 1;
switch (test_format) {
case AUDIO_U8:
format |= ESD_BITS8;
break;
case AUDIO_S16SYS:
format |= ESD_BITS16;
break;
default:
found = 0;
break;
}
}
if (!found) {
ESD_CloseDevice(this);
return SDL_SetError("Couldn't find any hardware audio formats");
}
if (this->spec.channels == 1) {
format |= ESD_MONO;
} else {
format |= ESD_STEREO;
}
#if 0
this->spec.samples = ESD_BUF_SIZE; /* Darn, no way to change this yet */
#endif
/* Open a connection to the ESD audio server */
this->hidden->audio_fd =
SDL_NAME(esd_play_stream) (format, this->spec.freq, NULL,
get_progname());
if (this->hidden->audio_fd < 0) {
ESD_CloseDevice(this);
return SDL_SetError("Couldn't open ESD connection");
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
this->hidden->frame_ticks =
(float) (this->spec.samples * 1000) / this->spec.freq;
this->hidden->next_frame = SDL_GetTicks() + this->hidden->frame_ticks;
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
ESD_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->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
SDL_FS_OpenDevice(_THIS, const char *devname, int iscapture)
{
int bytes;
SDL_AudioFormat test_format = 0, format = 0;
FSSampleFormat fs_format;
FSStreamDescription desc;
DirectResult ret;
/* 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 */
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:
fs_format = FSSF_U8;
bytes = 1;
format = 1;
break;
case AUDIO_S16SYS:
fs_format = FSSF_S16;
bytes = 2;
format = 1;
break;
case AUDIO_S32SYS:
fs_format = FSSF_S32;
bytes = 4;
format = 1;
break;
case AUDIO_F32SYS:
fs_format = FSSF_FLOAT;
bytes = 4;
format = 1;
break;
default:
format = 0;
break;
}
if (!format) {
test_format = SDL_NextAudioFormat();
}
}
if (format == 0) {
SDL_FS_CloseDevice(this);
return SDL_SetError("Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
/* Retrieve the main sound interface. */
ret = SDL_NAME(FusionSoundCreate) (&this->hidden->fs);
if (ret) {
SDL_FS_CloseDevice(this);
return SDL_SetError("Unable to initialize FusionSound: %d", ret);
}
this->hidden->mixsamples = this->spec.size / bytes / this->spec.channels;
/* Fill stream description. */
desc.flags = FSSDF_SAMPLERATE | FSSDF_BUFFERSIZE |
FSSDF_CHANNELS | FSSDF_SAMPLEFORMAT | FSSDF_PREBUFFER;
desc.samplerate = this->spec.freq;
desc.buffersize = this->spec.size * FUSION_BUFFERS;
desc.channels = this->spec.channels;
desc.prebuffer = 10;
desc.sampleformat = fs_format;
ret =
this->hidden->fs->CreateStream(this->hidden->fs, &desc,
&this->hidden->stream);
if (ret) {
SDL_FS_CloseDevice(this);
return SDL_SetError("Unable to create FusionSoundStream: %d", ret);
}
/* See what we got */
desc.flags = FSSDF_SAMPLERATE | FSSDF_BUFFERSIZE |
FSSDF_CHANNELS | FSSDF_SAMPLEFORMAT;
ret = this->hidden->stream->GetDescription(this->hidden->stream, &desc);
this->spec.freq = desc.samplerate;
this->spec.size =
desc.buffersize / FUSION_BUFFERS * bytes * desc.channels;
this->spec.channels = desc.channels;
/* 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) {
SDL_FS_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;
}
int
DSP_OpenAudio(_THIS, SDL_AudioSpec * spec)
{
char audiodev[1024];
#ifdef AUDIO_SETINFO
int enc;
#endif
int desired_freq = spec->freq;
/* Initialize our freeable variables, in case we fail */
audio_fd = -1;
mixbuf = NULL;
ulaw_buf = NULL;
/* Determine the audio parameters from the AudioSpec */
switch (SDL_AUDIO_BITSIZE(spec->format)) {
case 8:
{ /* Unsigned 8 bit audio data */
spec->format = AUDIO_U8;
#ifdef AUDIO_SETINFO
enc = AUDIO_ENCODING_LINEAR8;
#endif
}
break;
case 16:
{ /* Signed 16 bit audio data */
spec->format = AUDIO_S16SYS;
#ifdef AUDIO_SETINFO
enc = AUDIO_ENCODING_LINEAR;
#endif
}
break;
default:
{
/* !!! FIXME: fallback to conversion on unsupported types! */
SDL_SetError("Unsupported audio format");
return (-1);
}
}
audio_fmt = spec->format;
/* Open the audio device */
audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 1);
if (audio_fd < 0) {
SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
return (-1);
}
ulaw_only = 0; /* modern Suns do support linear audio */
#ifdef AUDIO_SETINFO
for (;;) {
audio_info_t info;
AUDIO_INITINFO(&info); /* init all fields to "no change" */
/* Try to set the requested settings */
info.play.sample_rate = spec->freq;
info.play.channels = spec->channels;
info.play.precision = (enc == AUDIO_ENCODING_ULAW)
? 8 : spec->format & 0xff;
info.play.encoding = enc;
if (ioctl(audio_fd, AUDIO_SETINFO, &info) == 0) {
/* Check to be sure we got what we wanted */
if (ioctl(audio_fd, AUDIO_GETINFO, &info) < 0) {
SDL_SetError("Error getting audio parameters: %s",
strerror(errno));
return -1;
}
if (info.play.encoding == enc
&& info.play.precision == (spec->format & 0xff)
&& info.play.channels == spec->channels) {
/* Yow! All seems to be well! */
spec->freq = info.play.sample_rate;
break;
}
}
switch (enc) {
case AUDIO_ENCODING_LINEAR8:
/* unsigned 8bit apparently not supported here */
enc = AUDIO_ENCODING_LINEAR;
spec->format = AUDIO_S16SYS;
break; /* try again */
case AUDIO_ENCODING_LINEAR:
/* linear 16bit didn't work either, resort to µ-law */
enc = AUDIO_ENCODING_ULAW;
spec->channels = 1;
spec->freq = 8000;
spec->format = AUDIO_U8;
ulaw_only = 1;
break;
default:
/* oh well... */
SDL_SetError("Error setting audio parameters: %s",
strerror(errno));
return -1;
}
}
#endif /* AUDIO_SETINFO */
written = 0;
/* We can actually convert on-the-fly to U-Law */
if (ulaw_only) {
spec->freq = desired_freq;
fragsize = (spec->samples * 1000) / (spec->freq / 8);
frequency = 8;
ulaw_buf = (Uint8 *) SDL_malloc(fragsize);
if (ulaw_buf == NULL) {
SDL_OutOfMemory();
return (-1);
}
spec->channels = 1;
} else {
fragsize = spec->samples;
frequency = spec->freq / 1000;
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Audio device %s U-Law only\n",
ulaw_only ? "is" : "is not");
fprintf(stderr, "format=0x%x chan=%d freq=%d\n",
spec->format, spec->channels, spec->freq);
#endif
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(spec);
/* Allocate mixing buffer */
mixbuf = (Uint8 *) SDL_AllocAudioMem(spec->size);
if (mixbuf == NULL) {
SDL_OutOfMemory();
return (-1);
}
SDL_memset(mixbuf, spec->silence, spec->size);
/* We're ready to rock and roll. :-) */
return (0);
}
static int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
char audiodev[1024];
int format;
int value;
int frag_spec;
Uint16 test_format;
/* Open the audio device */
audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
if ( audio_fd < 0 ) {
SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
return(-1);
}
mixbuf = NULL;
/* Make the file descriptor use blocking writes with fcntl() */
{ long flags;
flags = fcntl(audio_fd, F_GETFL);
flags &= ~O_NONBLOCK;
if ( fcntl(audio_fd, F_SETFL, flags) < 0 ) {
SDL_SetError("Couldn't set audio blocking mode");
return(-1);
}
}
/* Get a list of supported hardware formats */
if ( ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &value) < 0 ) {
perror("SNDCTL_DSP_GETFMTS");
SDL_SetError("Couldn't get audio format list");
return(-1);
}
/* Try for a closest match on audio format */
format = 0;
for ( test_format = SDL_FirstAudioFormat(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:
if ( value & AFMT_U8 ) {
format = AFMT_U8;
}
break;
case AUDIO_S16LSB:
if ( value & AFMT_S16_LE ) {
format = AFMT_S16_LE;
}
break;
case AUDIO_S16MSB:
if ( value & AFMT_S16_BE ) {
format = AFMT_S16_BE;
}
break;
#if 0
/*
* These formats are not used by any real life systems so they are not
* needed here.
*/
case AUDIO_S8:
if ( value & AFMT_S8 ) {
format = AFMT_S8;
}
break;
case AUDIO_U16LSB:
if ( value & AFMT_U16_LE ) {
format = AFMT_U16_LE;
}
break;
case AUDIO_U16MSB:
if ( value & AFMT_U16_BE ) {
format = AFMT_U16_BE;
}
break;
#endif
default:
format = 0;
break;
}
if ( ! format ) {
test_format = SDL_NextAudioFormat();
}
}
if ( format == 0 ) {
SDL_SetError("Couldn't find any hardware audio formats");
return(-1);
}
spec->format = test_format;
/* Set the audio format */
value = format;
if ( (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) ||
(value != format) ) {
perror("SNDCTL_DSP_SETFMT");
SDL_SetError("Couldn't set audio format");
return(-1);
}
/* Set the number of channels of output */
value = spec->channels;
if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &value) < 0 ) {
perror("SNDCTL_DSP_CHANNELS");
SDL_SetError("Cannot set the number of channels");
return(-1);
}
spec->channels = value;
/* Set the DSP frequency */
value = spec->freq;
if ( ioctl(audio_fd, SNDCTL_DSP_SPEED, &value) < 0 ) {
perror("SNDCTL_DSP_SPEED");
SDL_SetError("Couldn't set audio frequency");
return(-1);
}
spec->freq = value;
/* 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 ) {
perror("SNDCTL_DSP_SETFRAGMENT");
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
/* 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
IRIXAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
{
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
long width = 0;
long fmt = 0;
int valid = 0;
/* !!! FIXME: Handle multiple devices and capture? */
/* 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));
#ifdef OLD_IRIX_AUDIO
{
long audio_param[2];
audio_param[0] = AL_OUTPUT_RATE;
audio_param[1] = this->spec.freq;
valid = (ALsetparams(AL_DEFAULT_DEVICE, audio_param, 2) < 0);
}
#else
{
ALpv audio_param;
audio_param.param = AL_RATE;
audio_param.value.i = this->spec.freq;
valid = (alSetParams(AL_DEFAULT_OUTPUT, &audio_param, 1) < 0);
}
#endif
while ((!valid) && (test_format)) {
valid = 1;
this->spec.format = test_format;
switch (test_format) {
case AUDIO_S8:
width = AL_SAMPLE_8;
fmt = AL_SAMPFMT_TWOSCOMP;
break;
case AUDIO_S16SYS:
width = AL_SAMPLE_16;
fmt = AL_SAMPFMT_TWOSCOMP;
break;
case AUDIO_F32SYS:
width = 0; /* not used here... */
fmt = AL_SAMPFMT_FLOAT;
break;
/* Docs say there is int24, but not int32.... */
default:
valid = 0;
test_format = SDL_NextAudioFormat();
break;
}
if (valid) {
ALconfig audio_config = alNewConfig();
valid = 0;
if (audio_config) {
if (alSetChannels(audio_config, this->spec.channels) < 0) {
if (this->spec.channels > 2) { /* can't handle > stereo? */
this->spec.channels = 2; /* try again below. */
}
}
if ((alSetSampFmt(audio_config, fmt) >= 0) &&
((!width) || (alSetWidth(audio_config, width) >= 0)) &&
(alSetQueueSize(audio_config, this->spec.samples * 2) >=
0)
&& (alSetChannels(audio_config, this->spec.channels) >=
0)) {
this->hidden->audio_port = alOpenPort("SDL audio", "w",
audio_config);
if (this->hidden->audio_port == NULL) {
/* docs say AL_BAD_CHANNELS happens here, too. */
int err = oserror();
if (err == AL_BAD_CHANNELS) {
this->spec.channels = 2;
alSetChannels(audio_config, this->spec.channels);
this->hidden->audio_port =
alOpenPort("SDL audio", "w", audio_config);
}
}
if (this->hidden->audio_port != NULL) {
valid = 1;
}
}
alFreeConfig(audio_config);
}
}
}
if (!valid) {
IRIXAUDIO_CloseDevice(this);
SDL_SetError("Unsupported audio format");
return 0;
}
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* Allocate mixing buffer */
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->spec.size);
if (this->hidden->mixbuf == NULL) {
IRIXAUDIO_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/* We're ready to rock and roll. :-) */
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;
}
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;
/* Calculate the final parameters for this audio specification */
#ifdef PA_STREAM_ADJUST_LATENCY
spec->samples /= 2; /* Mix in smaller chunck to avoid underruns */
#endif
SDL_CalculateAudioSpec(spec);
/* Allocate mixing buffer */
mixlen = spec->size;
mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
if ( mixbuf == NULL ) {
return(-1);
}
SDL_memset(mixbuf, spec->silence, spec->size);
/* Reduced prebuffering compared to the defaults. */
#ifdef PA_STREAM_ADJUST_LATENCY
paattr.tlength = mixlen * 4; /* 2x original requested bufsize */
paattr.prebuf = -1;
paattr.maxlength = -1;
paattr.minreq = mixlen; /* -1 can lead to pa_stream_writable_size()
>= mixlen never becoming true */
flags = PA_STREAM_ADJUST_LATENCY;
#else
paattr.tlength = mixlen*2;
paattr.prebuf = mixlen*2;
paattr.maxlength = mixlen*2;
paattr.minreq = mixlen;
#endif
/* The SDL ALSA output hints us that we use Windows' channel mapping */
/* http://bugzilla.libsdl.org/show_bug.cgi?id=110 */
SDL_NAME(pa_channel_map_init_auto)(
&pacmap, spec->channels, PA_CHANNEL_MAP_WAVEEX);
/* Set up a new main loop */
if (!(mainloop = SDL_NAME(pa_mainloop_new)())) {
PULSE_CloseAudio(this);
SDL_SetError("pa_mainloop_new() failed");
return(-1);
}
mainloop_api = SDL_NAME(pa_mainloop_get_api)(mainloop);
if (!(context = SDL_NAME(pa_context_new)(mainloop_api, get_progname()))) {
PULSE_CloseAudio(this);
SDL_SetError("pa_context_new() failed");
return(-1);
}
/* Connect to the PulseAudio server */
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", /* stream description */
&paspec, /* sample format spec */
&pacmap /* channel map */
);
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 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
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
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
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
OBSD_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
char audiodev[64];
Uint16 format;
audio_info_t info;
AUDIO_INITINFO(&info);
SDL_CalculateAudioSpec(spec);
#ifdef USE_TIMER_SYNC
frame_ticks = 0.0;
#endif
audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
if(audio_fd < 0) {
SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
return(-1);
}
info.mode = AUMODE_PLAY;
if(ioctl(audio_fd, AUDIO_SETINFO, &info) < 0) {
SDL_SetError("Couldn't put device into play mode");
return(-1);
}
mixbuf = NULL;
AUDIO_INITINFO(&info);
for (format = SDL_FirstAudioFormat(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(audio_fd, AUDIO_SETINFO, &info) == 0)
break;
}
if(!format) {
SDL_SetError("No supported encoding for 0x%x", spec->format);
return(-1);
}
spec->format = format;
AUDIO_INITINFO(&info);
info.play.channels = spec->channels;
if (ioctl(audio_fd, AUDIO_SETINFO, &info) == -1)
spec->channels = 1;
AUDIO_INITINFO(&info);
info.play.sample_rate = spec->freq;
info.blocksize = spec->size;
info.hiwat = 5;
info.lowat = 3;
(void)ioctl(audio_fd, AUDIO_SETINFO, &info);
(void)ioctl(audio_fd, AUDIO_GETINFO, &info);
spec->freq = info.play.sample_rate;
mixlen = spec->size;
mixbuf = (Uint8*)SDL_AllocAudioMem(mixlen);
if(mixbuf == NULL) {
return(-1);
}
SDL_memset(mixbuf, spec->silence, spec->size);
parent = getpid();
#ifdef DEBUG_AUDIO
OBSD_Status(this);
#endif
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 ARTSC_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
int bits, frag_spec;
Uint16 test_format, format;
/* Reset the timer synchronization flag */
frame_ticks = 0.0;
mixbuf = NULL;
/* Try for a closest match on audio format */
format = 0;
bits = 0;
for ( test_format = SDL_FirstAudioFormat(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:
bits = 8;
format = 1;
break;
case AUDIO_S16LSB:
bits = 16;
format = 1;
break;
default:
format = 0;
break;
}
if ( ! format ) {
test_format = SDL_NextAudioFormat();
}
}
if ( format == 0 ) {
SDL_SetError("Couldn't find any hardware audio formats");
return(-1);
}
spec->format = test_format;
stream = arts_play_stream(spec->freq, bits, spec->channels, "SDL");
/* 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 */
#ifdef ARTS_P_PACKET_SETTINGS
arts_stream_set(stream, ARTS_P_PACKET_SETTINGS, frag_spec);
#else
arts_stream_set(stream, ARTS_P_PACKET_SIZE, frag_spec&0xffff);
arts_stream_set(stream, ARTS_P_PACKET_COUNT, frag_spec>>16);
#endif
spec->size = arts_stream_get(stream, ARTS_P_PACKET_SIZE);
/* 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
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 ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
int status;
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
snd_pcm_format_t format;
snd_pcm_uframes_t frames;
Uint16 test_format;
/* Open the audio device */
/* Name of device should depend on # channels in spec */
status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if ( status < 0 ) {
SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status));
return(-1);
}
/* Figure out what the hardware is capable of */
snd_pcm_hw_params_alloca(&hwparams);
status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, hwparams);
if ( status < 0 ) {
SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status));
ALSA_CloseAudio(this);
return(-1);
}
/* SDL only uses interleaved sample output */
status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
if ( status < 0 ) {
SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status));
ALSA_CloseAudio(this);
return(-1);
}
/* Try for a closest match on audio format */
status = -1;
for ( test_format = SDL_FirstAudioFormat(spec->format);
test_format && (status < 0); ) {
switch ( test_format ) {
case AUDIO_U8:
format = SND_PCM_FORMAT_U8;
break;
case AUDIO_S8:
format = SND_PCM_FORMAT_S8;
break;
case AUDIO_S16LSB:
format = SND_PCM_FORMAT_S16_LE;
break;
case AUDIO_S16MSB:
format = SND_PCM_FORMAT_S16_BE;
break;
case AUDIO_U16LSB:
format = SND_PCM_FORMAT_U16_LE;
break;
case AUDIO_U16MSB:
format = SND_PCM_FORMAT_U16_BE;
break;
default:
format = 0;
break;
}
if ( format != 0 ) {
status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, hwparams, format);
}
if ( status < 0 ) {
test_format = SDL_NextAudioFormat();
}
}
if ( status < 0 ) {
SDL_SetError("Couldn't find any hardware audio formats");
ALSA_CloseAudio(this);
return(-1);
}
spec->format = test_format;
/* Set the number of channels */
status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, hwparams, spec->channels);
if ( status < 0 ) {
status = SDL_NAME(snd_pcm_hw_params_get_channels)(hwparams);
if ( (status <= 0) || (status > 2) ) {
SDL_SetError("Couldn't set audio channels");
ALSA_CloseAudio(this);
return(-1);
}
spec->channels = status;
}
/* Set the audio rate */
status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, spec->freq, NULL);
if ( status < 0 ) {
SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status));
ALSA_CloseAudio(this);
return(-1);
}
spec->freq = status;
/* Set the buffer size, in samples */
frames = spec->samples;
frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, frames, NULL);
spec->samples = frames;
SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, 2, NULL);
/* "set" the hardware with the desired parameters */
status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, hwparams);
if ( status < 0 ) {
SDL_SetError("Couldn't set hardware audio parameters: %s", SDL_NAME(snd_strerror)(status));
ALSA_CloseAudio(this);
return(-1);
}
/* This is useful for debugging... */
/*
{ snd_pcm_sframes_t bufsize; int fragments;
bufsize = SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams);
fragments = SDL_NAME(snd_pcm_hw_params_get_periods)(hwparams);
fprintf(stderr, "ALSA: bufsize = %ld, fragments = %d\n", bufsize, fragments);
}
*/
/* Set the software parameters */
snd_pcm_sw_params_alloca(&swparams);
status = SDL_NAME(snd_pcm_sw_params_current)(pcm_handle, swparams);
if ( status < 0 ) {
SDL_SetError("Couldn't get software config: %s", SDL_NAME(snd_strerror)(status));
ALSA_CloseAudio(this);
return(-1);
}
status = SDL_NAME(snd_pcm_sw_params_set_start_threshold)(pcm_handle, swparams, 0);
if ( status < 0 ) {
SDL_SetError("Couldn't set start threshold: %s", SDL_NAME(snd_strerror)(status));
ALSA_CloseAudio(this);
return(-1);
}
status = SDL_NAME(snd_pcm_sw_params_set_avail_min)(pcm_handle, swparams, frames);
if ( status < 0 ) {
SDL_SetError("Couldn't set avail min: %s", SDL_NAME(snd_strerror)(status));
ALSA_CloseAudio(this);
return(-1);
}
status = SDL_NAME(snd_pcm_sw_params)(pcm_handle, swparams);
if ( status < 0 ) {
SDL_SetError("Couldn't set software audio parameters: %s", SDL_NAME(snd_strerror)(status));
ALSA_CloseAudio(this);
return(-1);
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(spec);
/* Allocate mixing buffer */
mixlen = spec->size;
mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
if ( mixbuf == NULL ) {
ALSA_CloseAudio(this);
return(-1);
}
SDL_memset(mixbuf, spec->silence, spec->size);
/* Get the parent process id (we're the parent of the audio thread) */
parent = getpid();
/* Switch to blocking mode for playback */
SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0);
/* We're ready to rock and roll. :-) */
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 PULSE_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
Uint16 test_format;
pa_sample_spec paspec;
pa_buffer_attr paattr;
pa_channel_map pacmap;
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;
}
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;
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(spec);
/* Allocate mixing buffer */
mixlen = spec->size;
mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
if ( mixbuf == NULL ) {
return(-1);
}
SDL_memset(mixbuf, spec->silence, spec->size);
/* Reduced prebuffering compared to the defaults. */
paattr.tlength = mixlen;
paattr.minreq = mixlen;
paattr.fragsize = mixlen;
paattr.prebuf = mixlen;
paattr.maxlength = mixlen * 4;
/* The SDL ALSA output hints us that we use Windows' channel mapping */
/* http://bugzilla.libsdl.org/show_bug.cgi?id=110 */
SDL_NAME(pa_channel_map_init_auto)(
&pacmap, spec->channels, PA_CHANNEL_MAP_WAVEEX);
/* Connect to the PulseAudio server */
stream = SDL_NAME(pa_simple_new)(
SDL_getenv("PASERVER"), /* server */
get_progname(), /* application name */
PA_STREAM_PLAYBACK, /* playback mode */
SDL_getenv("PADEVICE"), /* device on the server */
"Simple DirectMedia Layer", /* stream description */
&paspec, /* sample format spec */
&pacmap, /* channel map */
&paattr, /* buffering attributes */
NULL /* error code */
);
if ( stream == NULL ) {
PULSE_CloseAudio(this);
SDL_SetError("Could not connect to PulseAudio");
return(-1);
}
/* Get the parent process id (we're the parent of the audio thread) */
parent = getpid();
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 NTO_OpenAudio(_THIS, SDL_AudioSpec* spec)
{
int rval;
int format;
Uint16 test_format;
int found;
audio_handle = NULL;
this->enabled = 0;
if (pcm_buf != NULL)
{
SDL_FreeAudioMem(pcm_buf);
pcm_buf = NULL;
}
/* initialize channel transfer parameters to default */
NTO_InitAudioParams(&cparams);
/* Open the audio device */
rval = snd_pcm_open_preferred(&audio_handle, &cardno, &deviceno, OPEN_FLAGS);
if (rval < 0)
{
SDL_SetError("NTO_OpenAudio(): snd_pcm_open failed: %s\n", snd_strerror(rval));
return (-1);
}
if (!NTO_CheckBuggyCards(this, QSA_MMAP_WORKAROUND))
{
/* enable count status parameter */
if ((rval = snd_pcm_plugin_set_disable(audio_handle, PLUGIN_DISABLE_MMAP)) < 0)
{
SDL_SetError("snd_pcm_plugin_set_disable failed: %s\n", snd_strerror(rval));
return (-1);
}
}
/* Try for a closest match on audio format */
format = 0;
/* can't use format as SND_PCM_SFMT_U8 = 0 in nto */
found = 0;
for (test_format=SDL_FirstAudioFormat(spec->format); !found ;)
{
/* if match found set format to equivalent ALSA 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;
default:
break;
}
if (!found)
{
test_format = SDL_NextAudioFormat();
}
}
/* assumes test_format not 0 on success */
if (test_format == 0)
{
SDL_SetError("NTO_OpenAudio(): Couldn't find any hardware audio formats");
return (-1);
}
spec->format = test_format;
/* Set the audio format */
cparams.format.format = format;
/* Set mono or stereo audio (currently only two channels supported) */
cparams.format.voices = spec->channels;
/* Set rate */
cparams.format.rate = spec->freq;
/* Setup the transfer parameters according to cparams */
rval = snd_pcm_plugin_params(audio_handle, &cparams);
if (rval < 0)
{
SDL_SetError("NTO_OpenAudio(): snd_pcm_channel_params failed: %s\n", snd_strerror(rval));
return (-1);
}
/* Make sure channel is setup right one last time */
SDL_memset(&csetup, 0x00, sizeof(csetup));
csetup.channel = SND_PCM_CHANNEL_PLAYBACK;
if (snd_pcm_plugin_setup(audio_handle, &csetup) < 0)
{
SDL_SetError("NTO_OpenAudio(): Unable to setup playback channel\n");
return -1;
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(spec);
pcm_len = spec->size;
if (pcm_len==0)
{
pcm_len = csetup.buf.block.frag_size * 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)
*/
pcm_buf = (Uint8*)SDL_AllocAudioMem(pcm_len);
if (pcm_buf == NULL)
{
SDL_SetError("NTO_OpenAudio(): pcm buffer allocation failed\n");
return (-1);
}
SDL_memset(pcm_buf, spec->silence, pcm_len);
/* get the file descriptor */
if ((audio_fd = snd_pcm_file_descriptor(audio_handle, SND_PCM_CHANNEL_PLAYBACK)) < 0)
{
SDL_SetError("NTO_OpenAudio(): snd_pcm_file_descriptor failed with error code: %s\n", snd_strerror(rval));
return (-1);
}
/* Trigger audio playback */
rval = snd_pcm_plugin_prepare(audio_handle, SND_PCM_CHANNEL_PLAYBACK);
if (rval < 0)
{
SDL_SetError("snd_pcm_plugin_prepare failed: %s\n", snd_strerror(rval));
return (-1);
}
this->enabled = 1;
/* Get the parent process id (we're the parent of the audio thread) */
parent = getpid();
/* We're really ready to rock and roll. :-) */
return (0);
}
