void
sgi_audio_loopback(audio_desc_t ad, int gain)
{
long pvbuf[4];
int pvcnt;
UNUSED(ad); assert(audio_fd > 0);
pvcnt = 2;
pvbuf[0] = AL_MONITOR_CTL;
pvbuf[1] = AL_MONITOR_OFF;
if (gain) {
pvcnt = 6;
pvbuf[1] = AL_MONITOR_ON;
pvbuf[2] = AL_LEFT_MONITOR_ATTEN;
pvbuf[3] = 255 - RAT_TO_SGI_DEVICE(gain);
pvbuf[4] = AL_RIGHT_MONITOR_ATTEN;
pvbuf[5] = pvbuf[3];
}
if (ALsetparams(AL_DEFAULT_DEVICE, pvbuf, pvcnt) != 0) {
debug_msg("loopback failed\n");
}
}
static int
open_audio_port (AudioContext return_ac, AudioContext desc)
{
ALconfig config = ALnewconfig();
long params[2];
adjust_audio_volume (& desc->device);
return_ac->ac_left_speaker_gain = desc->ac_left_speaker_gain;
return_ac->ac_right_speaker_gain = desc->ac_right_speaker_gain;
params[0] = AL_OUTPUT_RATE;
params[1] = desc->ac_output_rate;
ALsetparams (desc->ac_device, params, 2);
return_ac->ac_output_rate = desc->ac_output_rate;
if (set_channels (config, desc->ac_nchan)==-1)
return -1;
return_ac->ac_nchan = desc->ac_nchan;
if (set_output_format (config, desc->ac_format)==-1)
return -1;
return_ac->ac_format = desc->ac_format;
ALsetqueuesize (config, (long) CHUNKSIZE);
return_ac->ac_port = ALopenport("XEmacs audio output", "w", config);
ALfreeconfig (config);
if (return_ac->ac_port==0)
{
report_file_error ("Opening audio output port", Qnil);
return -1;
}
return 0;
}
int sound_init()
{
long pvbuf[2];
pvbuf[0] = AL_OUTPUT_COUNT;
pvbuf[2] = AL_MONITOR_CTL;
if (ALgetparams(AL_DEFAULT_DEVICE, pvbuf, 4) < 0)
{
fprintf(stderr,"sound driver : No audio hardware\n");
return 0;
}
pvbuf[0] = AL_OUTPUT_RATE;
pvbuf[1] = 11025;
if (ALsetparams(AL_DEFAULT_DEVICE,pvbuf,4) < 0)
{
fprintf(stderr,"sound driver : Could not set sample rate\n");
return 0;
}
ALseterrorhandler(0);
audioconfig = ALnewconfig();
if (!audioconfig)
{
fprintf(stderr,"failed to create audio config\n");
return 0;
}
else if (ALsetchannels(audioconfig,AL_MONO))
{ fprintf(stderr,"sound driver : could not set audio channels\n");
return 0;
}
else if (ALsetqueuesize(audioconfig,BUF_SIZE))
{
fprintf(stderr,"sound driver : could not set audio que size\n");
ALfreeconfig(audioconfig);
return 0;
} else if (ALsetwidth (audioconfig, AL_SAMPLE_8))
{
fprintf(stderr,"sound driver :could not set 8 bit samples\n");
ALfreeconfig(audioconfig);
return 0;
}
audioport=ALopenport("Abuse sound driver","w",audioconfig);
if (!audioport)
{
fprintf(stderr,"sound driver : could not open audio port\n");
ALfreeconfig(audioconfig);
return 0;
}
int i=0,j;
uchar *vd=volume_table;
for (;i<32;i++)
{
for (j=0;j<256;j++,vd++)
*vd=(j-128)*i/31+128;
}
return 1;
}
static void adjust_audio_volume(AudioDevice device)
{
long params[4];
params[0] = AL_LEFT_SPEAKER_GAIN;
params[1] = device->left_speaker_gain;
params[2] = AL_RIGHT_SPEAKER_GAIN;
params[3] = device->right_speaker_gain;
ALsetparams(device->device, params, 4);
}
static Lisp_Object restore_audio_port(Lisp_Object closure)
{
Lisp_Object *contents = XVECTOR_DATA(closure);
saved_device_state[1] = XINT(contents[0]);
saved_device_state[3] = XINT(contents[1]);
saved_device_state[5] = XINT(contents[2]);
ALsetparams(AL_DEFAULT_DEVICE, saved_device_state, 6);
return Qnil;
}
/*
* Set the volume level.
*/
static void sfxSetGain(float l, float r) {
long pvbuf[4];
pvbuf[0] = AL_LEFT_SPEAKER_GAIN;
pvbuf[1] = l * currLeftGain;
pvbuf[2] = AL_RIGHT_SPEAKER_GAIN;
pvbuf[3] = r * currRightGain;
ALsetparams(AL_DEFAULT_DEVICE, pvbuf, 4L);
}
int esd_audio_write(void *buffer, int buf_size)
{
ALsetparams(AL_DEFAULT_DEVICE, rate_params, 2);
if (ALwritesamps(outaudioport, buffer, buf_size / 2) == 0) {
ALsetfillpoint(outaudioport, ESD_BUF_SIZE);
return buf_size;
}
else
return 0;
}
int esd_audio_read(void *buffer, int buf_size)
{
ALsetparams(AL_DEFAULT_DEVICE, (rate_params + 2), 2);
if (ALreadsamps(inaudioport, buffer, buf_size / 2) == 0) {
ALsetfillpoint(inaudioport, ESD_BUF_SIZE);
return buf_size;
}
else
return 0;
}
static Bool audriv_al_set_rate(ALport port, unsigned long rate)
{
long pv[2];
pv[0] = (port == out ? AL_OUTPUT_RATE : AL_INPUT_RATE);
pv[1] = rate;
if(ALsetparams(AL_DEFAULT_DEVICE, pv, 2) < 0)
return False;
return True;
}
void
sgi_audio_set_igain(audio_desc_t ad, int gain)
{
long cmd[4];
UNUSED(ad); assert(audio_fd > 0);
cmd[0] = AL_LEFT_INPUT_ATTEN;
cmd[1] = 255 - RAT_TO_SGI_DEVICE(gain);
cmd[2] = AL_RIGHT_INPUT_ATTEN;
cmd[3] = cmd[1];
ALsetparams(AL_DEFAULT_DEVICE, cmd, 4L);
}
void
sgi_audio_set_ogain(audio_desc_t ad, int vol)
{
long cmd[4];
UNUSED(ad); assert(audio_fd > 0);
cmd[0] = AL_LEFT_SPEAKER_GAIN;
cmd[1] = RAT_TO_SGI_DEVICE(vol);
cmd[2] = AL_RIGHT_SPEAKER_GAIN;
cmd[3] = cmd[1];
ALsetparams(AL_DEFAULT_DEVICE, cmd, 4L);
}
/*
* Reset the audio hardware to where we found it when we started.
*/
static void sfxResetAudioHw(void) {
long pvbuf[6];
pvbuf[0] = AL_LEFT_SPEAKER_GAIN;
pvbuf[1] = origLeftGain;
pvbuf[2] = AL_RIGHT_SPEAKER_GAIN;
pvbuf[3] = origRightGain;
pvbuf[4] = AL_OUTPUT_RATE;
pvbuf[5] = origOutputRate;
ALsetparams(AL_DEFAULT_DEVICE, pvbuf, 6L);
return;
}
void
sgi_audio_iport_set(audio_desc_t ad, audio_port_t port)
{
long pvbuf[2];
UNUSED(ad); assert(audio_fd > 0);
switch(port) {
case SGI_IPORT_LINE_IN:
pvbuf[0] = AL_INPUT_SOURCE;
pvbuf[1] = AL_INPUT_LINE;
ALsetparams(AL_DEFAULT_DEVICE, pvbuf, 2);
break;
case SGI_IPORT_MICROPHONE:
default:
pvbuf[0] = AL_INPUT_SOURCE;
pvbuf[1] = AL_INPUT_MIC;
ALsetparams(AL_DEFAULT_DEVICE, pvbuf, 2);
break;
case SGI_IPORT_APANEL:
break;
}
iport = port;
}
static int sgi_init(const char *param, int *speed, int *fragsize, int *fragnr, int *channels)
{
long chpars[] = { AL_OUTPUT_RATE, 0 };
int st;
/* No stereo capability. */
*channels = 1;
ALseterrorhandler(sgi_errorhandler);
chpars[1] = *speed;
st = ALsetparams(AL_DEFAULT_DEVICE, chpars, 2);
if (st < 0) {
return 1;
}
st = ALgetparams(AL_DEFAULT_DEVICE, chpars, 2);
if (st < 0) {
return 1;
}
*speed = chpars[1];
sgi_audioconfig = ALnewconfig();
if (!sgi_audioconfig) {
return 1;
}
st = ALsetchannels(sgi_audioconfig, AL_MONO);
if (st < 0) {
goto fail;
}
st = ALsetwidth(sgi_audioconfig, AL_SAMPLE_16);
if (st < 0) {
goto fail;
}
st = ALsetqueuesize(sgi_audioconfig, *fragsize * *fragnr);
if (st < 0) {
goto fail;
}
sgi_audioport = ALopenport("outport", "w", sgi_audioconfig);
if (!sgi_audioport) {
goto fail;
}
return 0;
fail:
ALfreeconfig(sgi_audioconfig);
sgi_audioconfig = NULL;
return 1;
}
FileDescriptor InitAudio(ALport *alpp) {
/* initialize audio driver
for 16-bit 44100kHz monophonic sample source to DACs */
ALconfig alc;
alc = ALnewconfig();
ALsetwidth (alc, AL_SAMPLE_16);
ALsetqueuesize (alc, OUTPUTQUEUESIZE);
ALsetchannels(alc, (long)1);
*alpp = ALopenport("obuf", "w", alc);
{
long pvbuf[2];
long pvlen=2;
pvbuf[0] = AL_OUTPUT_RATE;
pvbuf[1] = AL_RATE_44100;
ALsetparams(AL_DEFAULT_DEVICE, pvbuf, pvlen);
the_sample_rate = 44100.0f;
}
/* obtain a file descriptor associated with sound output port */
return ALgetfd(*alpp);
}
int esd_audio_open()
{
ALconfig audioconfig;
audioconfig = ALnewconfig();
rate_params[1] = esd_audio_rate;
rate_params[3] = esd_audio_rate;
if (!audioconfig) {
printf( "Couldn't initialize new audio config\n" );
esd_audio_fd = -1;
return esd_audio_fd;
} else {
long pvbuf[] = { AL_OUTPUT_COUNT, 0,
AL_MONITOR_CTL, 0,
AL_OUTPUT_RATE, 0 };
if (ALgetparams(AL_DEFAULT_DEVICE, pvbuf, 6) < 0)
if (oserror() == AL_BAD_DEVICE_ACCESS) {
esd_audio_fd = -1;
return esd_audio_fd;
}
if (pvbuf[1] == 0 && pvbuf[3] == AL_MONITOR_OFF) {
ALsetparams(AL_DEFAULT_DEVICE, rate_params, 2);
} else
if (pvbuf[5] != esd_audio_rate) {
printf("audio device is already in use with wrong sample output rate\n");
esd_audio_fd = -1;
return esd_audio_fd;
}
/* ALsetsampfmt(audioconfig, AL_SAMPFMT_TWOSCOMP); this is the default */
/* ALsetwidth(audioconfig, AL_SAMPLE_16); this is the default */
if ( (esd_audio_format & ESD_MASK_CHAN) == ESD_MONO)
ALsetchannels(audioconfig, AL_MONO);
/* else ALsetchannels(audioconfig, AL_STEREO); this is the default */
ALsetqueuesize(audioconfig, ESD_BUF_SIZE * 2);
outaudioport = ALopenport("esd", "w", audioconfig);
if (outaudioport == (ALport) 0) {
switch (oserror()) {
case AL_BAD_NO_PORTS:
printf( "system is out of ports\n");
esd_audio_fd = -1;
return esd_audio_fd;
break;
case AL_BAD_DEVICE_ACCESS:
printf("couldn't access audio device\n");
esd_audio_fd = -1;
return esd_audio_fd;
break;
case AL_BAD_OUT_OF_MEM:
printf("out of memory\n");
esd_audio_fd = -1;
return esd_audio_fd;
break;
}
/* don't know how we got here, but it must be bad */
esd_audio_fd = -1;
return esd_audio_fd;
}
ALsetfillpoint(outaudioport, ESD_BUF_SIZE);
esd_audio_fd = ALgetfd(outaudioport);
/*
* If we are recording, open a second port to read from
* and return that fd instead
*/
if ( (esd_audio_format & ESD_MASK_FUNC) == ESD_RECORD ) {
inaudioport = ALopenport("esd", "r", audioconfig);
if (inaudioport == (ALport) 0) {
switch (oserror()) {
case AL_BAD_NO_PORTS:
printf( "system is out of ports\n");
esd_audio_fd = -1;
return esd_audio_fd;
break;
case AL_BAD_DEVICE_ACCESS:
printf("couldn't access audio device\n");
esd_audio_fd = -1;
return esd_audio_fd;
break;
case AL_BAD_OUT_OF_MEM:
printf("out of memory\n");
esd_audio_fd = -1;
return esd_audio_fd;
break;
default:
printf( "Unknown error opening port\n" );
}
/* don't know how we got here, but it must be bad */
esd_audio_fd = -1;
return esd_audio_fd;
}
ALsetfillpoint(inaudioport, ESD_BUF_SIZE);
ALsetparams(AL_DEFAULT_DEVICE, (rate_params + 2), 2);
esd_audio_fd = ALgetfd(inaudioport);
}
}
return esd_audio_fd;
}
int
sgi_audio_open(audio_desc_t ad, audio_format* ifmt, audio_format *ofmt)
{
ALconfig c;
long cmd[8];
if (audio_fd != -1) {
sgi_audio_close(ad);
}
if (ifmt->encoding != DEV_S16) return FALSE;
if ((c = ALnewconfig()) == NULL) {
fprintf(stderr, "ALnewconfig error\n");
exit(1);
}
switch(ifmt->channels) {
case 1:
ALsetchannels(c, AL_MONO); break;
case 2:
ALsetchannels(c, AL_STEREO); break;
default:
sgi_audio_close(ad);
}
ALsetwidth(c, AL_SAMPLE_16);
ALsetqueuesize(c, QSIZE);
ALsetsampfmt(c, AL_SAMPFMT_TWOSCOMP);
if ((wp = ALopenport("RAT write", "w", c)) == NULL) {
fprintf(stderr, "ALopenport (write) error\n");
sgi_audio_close(ad);
return FALSE;
}
if ((rp = ALopenport("RAT read", "r", c)) == NULL) {
fprintf(stderr, "ALopenport (read) error\n");
sgi_audio_close(ad);
return FALSE;
}
cmd[0] = AL_OUTPUT_RATE;
cmd[1] = ofmt->sample_rate;
cmd[2] = AL_INPUT_RATE;
cmd[3] = ifmt->sample_rate;
cmd[4] = AL_MONITOR_CTL;
cmd[5] = AL_MONITOR_OFF;
/*cmd[6] = AL_INPUT_SOURCE;*/
/*cmd[7] = AL_INPUT_MIC;*/
if (ALsetparams(AL_DEFAULT_DEVICE, cmd, 6L/*was 8L*/) == -1) {
fprintf(stderr, "audio_open/ALsetparams error\n");
sgi_audio_close(ad);
}
/* Get the file descriptor to use in select */
audio_fd = ALgetfd(rp);
if (ALsetfillpoint(rp, ifmt->bytes_per_block) < 0) {
debug_msg("ALsetfillpoint failed (%d samples)\n", ifmt->bytes_per_block);
}
bytes_per_block = ifmt->bytes_per_block;
/* We probably should free the config here... */
return TRUE;
}
/*
* Sound handler.
*/
static void sfxSoundHandler(void *arg) {
ALport *ap = (ALport *)arg;
Sample samp[MAX_AUDIO_PORTS];
int idx, nap;
int treated;
int nSounds = 1;
int nextPort = 0;
long maxSampsPerPass;
long sampCount;
long pvbuf[2];
struct _sfx ss;
struct pollfd pf;
prctl(PR_TERMCHILD, 0);
(void) signal(SIGHUP, sfxDieGracefully);
maxSampsPerPass = 1600;
for (idx=0; idx < nAudioPorts; idx++) {
samp[idx].id = -1;
samp[idx].sample = NULL;
samp[idx].sampsToPlay = 0;
samp[idx].repeat = 0;
}
/*
* Set sample rate for output device.
*/
pvbuf[0] = AL_OUTPUT_RATE;
pvbuf[1] = AL_RATE_16000;
(void) ALsetparams(AL_DEFAULT_DEVICE, pvbuf, 2L);
/*
* Prepare to read from pipe.
*/
pf.fd = spigot[0];
pf.events = POLLIN | POLLRDNORM | POLLRDBAND;
#define EVER ;;
for (EVER) {
if (nSounds == 0 || (idx=poll(&pf, 1, 0)) > 0) {
(void) read(spigot[0], &ss, sizeof(ss));
treated = 0;
if (ss.loop == 1 && ss.repeat == 0) {
treated = 1;
for (idx=0; idx < nAudioPorts; idx++) {
if (samp[idx].id == ss.id) {
samp[idx].id = -1;
samp[idx].repeat = 0;
samp[idx].sampsToPlay = 0;
samp[idx].sample = NULL;
}
}
} else if (ss.loop == 1 && ss.count > 1) {
for (idx=0; idx < nAudioPorts; idx++) {
if (samp[idx].id == ss.id) {
treated = 1;
samp[idx].repeat = ss.repeat;
samp[idx].sampsToPlay = (long)ss.soundSize[ss.pitch];
samp[idx].sampsPlayed = 0;
samp[idx].sample = ss.soundData[ss.pitch];
}
}
}
if (!treated) {
for (idx=0; idx < nAudioPorts; idx++) {
nextPort = (nextPort+1) % nAudioPorts;
if (samp[nextPort].repeat == 0)
break;
}
samp[nextPort].sample = ss.soundData[ss.pitch];
samp[nextPort].sampsToPlay = (long)ss.soundSize[ss.pitch];
samp[nextPort].sampsPlayed = 0;
samp[nextPort].repeat = ss.repeat;
samp[nextPort].id = ss.id;
}
} else if (idx < 0)
(void) fprintf(stderr, "panic: input poll failed: %s\n", strerror(errno));
nSounds = 0;
nap = 0;
for (idx=0; idx < nAudioPorts; idx++) {
if (samp[idx].sampsToPlay > 0) {
nSounds++;
if (ALgetfilled(ap[idx]) > 4000) {
nap++;
continue;
}
}
if (samp[idx].sampsToPlay >= maxSampsPerPass) {
(void) ALwritesamps(ap[idx],
samp[idx].sample + samp[idx].sampsPlayed, maxSampsPerPass);
samp[idx].sampsPlayed += maxSampsPerPass;
samp[idx].sampsToPlay -= maxSampsPerPass;
} else if (samp[idx].sampsToPlay > 0) {
if ((samp[idx].sampsToPlay%2) == 1) {
samp[idx].sampsToPlay -= 1;
samp[idx].sampsPlayed += 1;
}
if (samp[idx].sampsToPlay > 0)
(void) ALwritesamps(ap[idx],
samp[idx].sample+samp[idx].sampsPlayed, samp[idx].sampsToPlay);
if (samp[idx].repeat) {
sampCount = maxSampsPerPass - samp[idx].sampsToPlay;
samp[idx].sampsToPlay += samp[idx].sampsPlayed - sampCount;
samp[idx].sampsPlayed = sampCount;
(void) ALwritesamps(ap[idx], samp[idx].sample, sampCount);
} else
samp[idx].sampsToPlay = 0;
}
}
if (nap == nSounds)
sginap(1);
}
return;
}
static BOOL SGI_Init(void)
{
long chpars[] = { AL_OUTPUT_RATE, AL_RATE_22050 };
switch(md_mixfreq) {
case 8000:
chpars[1] = AL_RATE_8000;
break;
case 11025:
chpars[1] = AL_RATE_11025;
break;
case 16000:
chpars[1] = AL_RATE_16000;
break;
case 22050:
chpars[1] = AL_RATE_22050;
break;
case 32000:
chpars[1] = AL_RATE_32000;
break;
case 44100:
chpars[1] = AL_RATE_44100;
break;
case 48000:
chpars[1] = AL_RATE_48000;
break;
default:
_mm_errno=MMERR_SGI_SPEED;
return 1;
}
ALseterrorhandler(0);
ALsetparams(AL_DEFAULT_DEVICE, chpars, 2);
if (!(sgi_config=ALnewconfig())) {
_mm_errno=MMERR_OPENING_AUDIO;
return 1;
}
if (md_mode&DMODE_16BITS) {
if (ALsetwidth(sgi_config,AL_SAMPLE_16)<0) {
_mm_errno=MMERR_SGI_16BIT;
return 1;
}
sample_factor = 1;
} else {
if (ALsetwidth(sgi_config,AL_SAMPLE_8)<0) {
_mm_errno=MMERR_SGI_8BIT;
return 1;
}
sample_factor = 0;
}
if (md_mode&DMODE_STEREO) {
if (ALsetchannels(sgi_config,AL_STEREO)<0) {
_mm_errno=MMERR_SGI_STEREO;
return 1;
}
} else {
if (ALsetchannels(sgi_config,AL_MONO)<0) {
_mm_errno=MMERR_SGI_MONO;
return 1;
}
}
if ((getenv("MM_SGI_FRAGSIZE"))&&(sgi_fragsize!=DEFAULT_SGI_FRAGSIZE))
sgi_fragsize=atol(getenv("MM_SGI_FRAGSIZE"));
if (!sgi_fragsize) sgi_fragsize=DEFAULT_SGI_FRAGSIZE;
if ((getenv("MM_SGI_BUFSIZE"))&&(sgi_bufsize!=DEFAULT_SGI_BUFSIZE))
sgi_bufsize=atol(getenv("MM_SGI_BUFSIZE"));
if (!sgi_bufsize) sgi_fragsize=DEFAULT_SGI_BUFSIZE;
ALsetqueuesize(sgi_config, sgi_bufsize);
if (!(sgi_port=ALopenport("libmikmod","w",sgi_config))) {
_mm_errno=MMERR_OPENING_AUDIO;
return 1;
}
if(!(audiobuffer=(SBYTE*)_mm_malloc(sgi_fragsize))) return 1;
return VC_Init();
}
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);
}
/*--------------------------------------------------------------------------------------*/
PaError PaHost_OpenStream(internalPortAudioStream *past)
{
PaError result = paNoError;
PaHostSoundControl *pahsc;
unsigned int minNumBuffers;
internalPortAudioDevice *padIN, *padOUT; /* For looking up native AL-numbers. */
ALconfig sgiALconfig = NULL; /* IRIX-datatype. */
long pvbuf[8]; /* To get/set hardware configs. */
long sr, ALqsize;
DBUG(("PaHost_OpenStream() called.\n")); /* Alloc FASTMEM and init host data. */
if (!past)
{
ERR_RPT(("Streampointer NULL!\n"));
result = paBadStreamPtr; goto done;
}
pahsc = (PaHostSoundControl*)PaHost_AllocateFastMemory(sizeof(PaHostSoundControl));
if (pahsc == NULL)
{
ERR_RPT(("FAST Memory allocation failed.\n")); /* Pass trough some ERR_RPT-exit- */
result = paInsufficientMemory; goto done; /* code (nothing will be freed). */
}
memset(pahsc, 0, sizeof(PaHostSoundControl));
/* pahsc->pahsc_threadPID = -1; Should pahsc_threadPID be inited to */
past->past_DeviceData = (void*)pahsc; /* -1 instead of 0 ?? */
/*--------------------------------------------------- Allocate native buffers: --------*/
pahsc->pahsc_SamplesPerInputBuffer = past->past_FramesPerUserBuffer * /* Needed by the */
past->past_NumInputChannels; /* audio-thread. */
pahsc->pahsc_BytesPerInputBuffer = pahsc->pahsc_SamplesPerInputBuffer * sizeof(short);
if (past->past_NumInputChannels > 0) /* Assumes short = 16 bits! */
{
pahsc->pahsc_NativeInputBuffer = (short*)PaHost_AllocateFastMemory(pahsc->pahsc_BytesPerInputBuffer);
if( pahsc->pahsc_NativeInputBuffer == NULL )
{
ERR_RPT(("Fast memory allocation for input-buffer failed.\n"));
result = paInsufficientMemory; goto done;
}
}
pahsc->pahsc_SamplesPerOutputBuffer = past->past_FramesPerUserBuffer * /* Needed by the */
past->past_NumOutputChannels; /* audio-thread. */
pahsc->pahsc_BytesPerOutputBuffer = pahsc->pahsc_SamplesPerOutputBuffer * sizeof(short);
if (past->past_NumOutputChannels > 0) /* Assumes short = 16 bits! */
{
pahsc->pahsc_NativeOutputBuffer = (short*)PaHost_AllocateFastMemory(pahsc->pahsc_BytesPerOutputBuffer);
if (pahsc->pahsc_NativeOutputBuffer == NULL)
{
ERR_RPT(("Fast memory allocation for output-buffer failed.\n"));
result = paInsufficientMemory; goto done;
}
}
/*------------------------------------------ Manipulate hardware if necessary and allowed: --*/
ALseterrorhandler(0); /* 0 = turn off the default error handler. */
pvbuf[0] = AL_INPUT_RATE;
pvbuf[2] = AL_INPUT_COUNT;
pvbuf[4] = AL_OUTPUT_RATE; /* TO FIX: rates may be logically, not always in Hz! */
pvbuf[6] = AL_OUTPUT_COUNT;
sr = (long)(past->past_SampleRate + 0.5); /* Common for input and output :-) */
if (past->past_NumInputChannels > 0) /* We need to lookup the corre- */
{ /* sponding native AL-number(s). */
padIN = Pa_GetInternalDevice(past->past_InputDeviceID);
if (!padIN)
{
ERR_RPT(("Pa_GetInternalDevice() for input failed.\n"));
result = paHostError; goto done;
}
if (ALgetparams(padIN->pad_ALdevice, &pvbuf[0], 4)) /* Although input and output will both be on */
goto sgiError; /* the same AL-device, the AL-library might */
if (pvbuf[1] != sr) /* contain more than AL_DEFAULT_DEVICE in */
{ /* Rate different from current harware-rate? the future. Therefore 2 seperate queries. */
if (pvbuf[3] > 0) /* Means, there's other clients using AL-input-ports */
{
ERR_RPT(("Sorry, not allowed to switch input-hardware to %ld Hz because \
another process is currently using input at %ld kHz.\n", sr, pvbuf[1]));
result = paHostError; goto done;
}
pvbuf[1] = sr; /* Then set input-rate. */
if (ALsetparams(padIN->pad_ALdevice, &pvbuf[0], 2))
goto sgiError; /* WHETHER THIS SAMPLERATE WAS REALLY PRESENT IN OUR ARRAY OF RATES, */
} /* IS NOT CHECKED, AT LEAST NOT BY ME, WITHIN THIS FILE! Does PA do? */
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;
}
Error SoundCardPMO::Init(OutputInfo * info)
{
ALconfig config = ALnewconfig();
long pvbuffer[2];
m_properlyInitialized = false;
if (!info)
{
info = myInfo;
}
else
{
m_iDataSize = info->max_buffer_size;
}
pvbuffer[0] = AL_OUTPUT_RATE;
pvbuffer[1] = info->samples_per_second;
if (ALsetparams(AL_DEFAULT_DEVICE, pvbuffer, 2) < 0)
{
ReportError("Cannot set the soundcard's sampling speed.");
return (Error) pmoError_IOCTL_SNDCTL_DSP_SPEED;
}
int ret = -1;
if (info->bits_per_sample > 8)
ret = ALsetwidth(config, AL_SAMPLE_16);
else
ret = ALsetwidth(config, AL_SAMPLE_8);
if (ret < 0)
{
ReportError("The soundcard does not support 16 bit sample size.");
return (Error) pmoError_IOCTL_SNDCTL_DSP_SAMPLESIZE;
}
ret = -1;
if (info->number_of_channels == 2)
ret = ALsetchannels(config, AL_STEREO);
else
ret = ALsetchannels(config, AL_MONO);
if (ret < 0)
{
ReportError("Cannot set the soundcard to stereo.");
return (Error) pmoError_IOCTL_SNDCTL_DSP_STEREO;
}
ALsetqueuesize(config, m_iDataSize);
outaudioport = ALopenport("freeamp", "w", config);
if (outaudioport == (ALport)0)
{
ReportError("Couldn't open port.");
return (Error) pmoError_IOCTL_SNDCTL_DSP_STEREO;
}
audio_fd = ALgetfd(outaudioport);
channels = info->number_of_channels;
// configure the device:
int play_precision = info->bits_per_sample;
int play_stereo = channels - 1;
int play_sample_rate = info->samples_per_second;
memcpy(myInfo, info, sizeof(OutputInfo));
m_properlyInitialized = true;
// PORTING: The GETOSPACE ioctl determines how much space the kernel's
// output buffer has. Your OS may not have this.
m_iTotalFragments = 2048; /* arbitrary blah */
m_iOutputBufferSize = play_precision * m_iTotalFragments;
m_iBytesPerSample = info->number_of_channels * (info->bits_per_sample / 8);
return kError_NoErr;
}
Bool audriv_set_play_volume(int volume)
/* 演奏音量を 0 〜 255 の範囲内で設定します．0 は無音，255 は最大音量．
* 0 未満は 0，255 を超える値は 255 に等価．
* 成功した場合は True を，失敗した場合は False を返します．
*/
{
#ifndef SGI_OLDAL
double gain;
ALfixed lrgain[2];
ALpv pv;
int resource;
if(volume < 0)
volume = 0;
else if(volume > 255)
volume = 255;
if(volume == 0)
{
if(out_ginfo.specialVals & AL_NEG_INFINITY_BIT)
gain = alFixedToDouble(AL_NEG_INFINITY);
else
gain = alFixedToDouble(out_ginfo.min.ll);
}
else if(volume == 255)
{
gain = alFixedToDouble(out_ginfo.max.ll);
}
else
{
double min, max;
min = alFixedToDouble(out_ginfo.min.ll);
max = alFixedToDouble(out_ginfo.max.ll);
gain = min + (max - min) * (volume - 1) * (1.0/255);
if(gain < min)
gain = min;
else if(gain > max)
gain = max;
}
if(out == NULL)
resource = AL_DEFAULT_OUTPUT;
else
resource = alGetResource(out);
lrgain[0] = lrgain[1] = alDoubleToFixed(gain);
pv.param = AL_GAIN;
pv.value.ptr = lrgain;
pv.sizeIn = 2;
if(alSetParams(resource, &pv, 1) < 0)
{
audriv_err(ALERROR);
return False;
}
return True;
#else
long gain[4];
if(volume < 0)
volume = 0;
else if(volume > 255)
volume = 255;
gain[0] = AL_LEFT_SPEAKER_GAIN;
gain[1] = volume;
gain[2] = AL_RIGHT_SPEAKER_GAIN;
gain[3] = volume;
if(ALsetparams(AL_DEFAULT_DEVICE, gain, 4) < 0)
{
audriv_err(ALERROR);
return False;
}
return True;
#endif /* SGI_OLDAL */
}
void *Audio_Irix::open (AudioConfig& cfg, const char *)
{
// Copy input parameters. May later be replaced with driver defaults.
_settings = cfg;
_config = ALnewconfig();
// Set sample format
ALsetsampfmt(_config, AL_SAMPFMT_TWOSCOMP);
// stereo or mono mode
_settings.channels = cfg.channels >= 2 ? 2 : 1;
if (_settings.channels == 2)
ALsetchannels(_config, AL_STEREO);
else
ALsetchannels(_config, AL_MONO);
// 16 or 8 bit sample
_settings.precision = cfg.precision >= 16 ? 16 : 8;
if (_settings.precision == 16)
ALsetwidth(_config, AL_SAMPLE_16);
else
ALsetwidth(_config, AL_SAMPLE_8);
// Frequency
long chpars[] = {AL_OUTPUT_RATE, 0};
if (cfg.frequency > 48000)
chpars[1] = AL_RATE_48000;
else if (cfg.frequency > 44100)
chpars[1] = AL_RATE_44100;
else if (cfg.frequency > 32000)
chpars[1] = AL_RATE_32000;
else if (cfg.frequency > 22050)
chpars[1] = AL_RATE_22050;
else if (cfg.frequency > 16000)
chpars[1] = AL_RATE_16000;
else
chpars[1] = AL_RATE_11025;
ALsetparams(AL_DEFAULT_DEVICE, chpars, 2);
ALgetparams(AL_DEFAULT_DEVICE, chpars, 2);
_settings.frequency = (uint_least32_t) chpars[1];
// Allocate sound buffers and set audio queue
ALsetqueuesize(_config, chpars[1]);
// open audio device
_audio = ALopenport("SIDPLAY2 sound", "w", _config);
if (_audio == NULL)
{
perror("AUDIO:");
_errorString = "ERROR: Could not open audio device.\n See standard error output.";
ALfreeconfig(_config);
return 0;
}
// Setup internal Config
_settings.encoding = AUDIO_SIGNED_PCM;
_settings.bufSize = (uint_least32_t) chpars[1];
// Update the users settings
getConfig (cfg);
// Allocate memory same size as buffer
#ifdef HAVE_EXCEPTIONS
_sampleBuffer = new(std::nothrow) int_least8_t[chpars[1]];
#else
_sampleBuffer = new int_least8_t[chpars[1]];
#endif
_errorString = "OK";
return (void *) _sampleBuffer;
}
static int setaudio(struct xmp_options *o)
{
int bsize = 32 * 1024;
ALconfig config;
long pvbuffer[2];
char *token, **parm;
int i;
parm_init();
chkparm1("buffer", bsize = strtoul(token, NULL, 0));
parm_end();
if ((config = ALnewconfig()) == 0)
return XMP_ERR_DINIT;
/*
* Set sampling rate
*/
pvbuffer[0] = AL_OUTPUT_RATE;
#if 0 /* DOESN'T WORK */
for (i = 0; srate[i]; i++) {
if (srate[i] <= o->freq)
pvbuffer[1] = o->freq = srate[i];
}
#endif /* DOESN'T WORK */
/*
* This was flawed as far as I can tell - it just progressively lowered
* the sample rate to the lowest possible!
*
* o->freq = 44100
*
* i = 0 / if (48000 <= 44100)
* i = 1 / if (44100 <= 44100)
* then pvbuffer[1] = o->freq = 44100
* i = 2 / if (32000 <= 44100)
* then pvbuffer[1] = o->freq = 32000
* i = 3 / if (22050 <= 32000)
* then pvbuffer[1] = o->freq = 22050
* etc...
*
* Below is my attempt to write a new one. It picks the next highest
* rate available up to the maximum. This seems a lot more reasonable.
*
* - 19990706 bdowning
*/
for (i = 0; srate[i]; i++) ; /* find the end of the array */
while (i-- > 0) {
if (srate[i] >= o->freq) {
pvbuffer[1] = o->freq = srate[i];
break;
}
}
if (i == 0)
pvbuffer[1] = o->freq = srate[0]; /* 48 kHz. Wow! */
if (ALsetparams(AL_DEFAULT_DEVICE, pvbuffer, 2) < 0)
return XMP_ERR_DINIT;
/*
* Set sample format to signed integer
*/
if (ALsetsampfmt(config, AL_SAMPFMT_TWOSCOMP) < 0)
return XMP_ERR_DINIT;
/*
* Set sample width; 24 bit samples are not currently supported by xmp
*/
if (o->resol > 8) {
if (ALsetwidth(config, AL_SAMPLE_16) < 0) {
if (ALsetwidth(config, AL_SAMPLE_8) < 0)
return XMP_ERR_DINIT;
o->resol = 8;
} else
al_sample_16 = 1;
} else {
if (ALsetwidth(config, AL_SAMPLE_8) < 0) {
if (ALsetwidth(config, AL_SAMPLE_16) < 0)
return XMP_ERR_DINIT;
o->resol = 16;
} else
al_sample_16 = 0;
}
/*
* Set number of channels; 4 channel output is not currently supported
*/
if (o->outfmt & XMP_FMT_MONO) {
if (ALsetchannels(config, AL_MONO) < 0) {
if (ALsetchannels(config, AL_STEREO) < 0)
return XMP_ERR_DINIT;
o->outfmt &= ~XMP_FMT_MONO;
}
} else {
if (ALsetchannels(config, AL_STEREO) < 0) {
if (ALsetchannels(config, AL_MONO) < 0)
return XMP_ERR_DINIT;
o->outfmt |= XMP_FMT_MONO;
}
}
/*
* Set buffer size
*/
if (ALsetqueuesize(config, bsize) < 0)
return XMP_ERR_DINIT;
/*
* Open the audio port
*/
if ((audio_port = ALopenport("xmp", "w", config)) == 0)
return XMP_ERR_DINIT;
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 AudioContext
initialize_audio_port (AudioContext desc)
{
/* we can't use the same port for mono and stereo */
static AudioContextRec mono_port_state
= { { 0, 0, 0, 0 },
{ (ALport) 0, AFunknown, 1, 0 },
{ (void *) 0, (unsigned long) 0 } };
#if HAVE_STEREO
static AudioContextRec stereo_port_state
= { { 0, 0, 0, 0 },
{ (ALport) 0, AFunknown, 2, 0 },
{ (void *) 0, (unsigned long) 0 } };
static AudioContext return_ac;
switch (desc->ac_nchan)
{
case 1: return_ac = & mono_port_state; break;
case 2: return_ac = & stereo_port_state; break;
default: return (AudioContext) 0;
}
#else /* not HAVE_STEREO */
static AudioContext return_ac = & mono_port_state;
#endif /* not HAVE_STEREO */
return_ac->device = desc->device;
return_ac->buffer = desc->buffer;
return_ac->ac_format = desc->ac_format;
return_ac->ac_queue_size = desc->ac_queue_size;
if (return_ac->ac_port==(ALport) 0)
{
if ((open_audio_port (return_ac, desc))==-1)
{
report_file_error ("Open audio port", Qnil);
return (AudioContext) 0;
}
}
else
{
ALconfig config = ALgetconfig (return_ac->ac_port);
int changed = 0;
long params[2];
params[0] = AL_OUTPUT_RATE;
ALgetparams (return_ac->ac_device, params, 2);
return_ac->ac_output_rate = params[1];
if (return_ac->ac_output_rate != desc->ac_output_rate)
{
return_ac->ac_output_rate = params[1] = desc->ac_output_rate;
ALsetparams (return_ac->ac_device, params, 2);
}
if ((changed = set_output_format (config, return_ac->ac_format))==-1)
return (AudioContext) 0;
return_ac->ac_format = desc->ac_format;
if (changed)
ALsetconfig (return_ac->ac_port, config);
}
return_ac->ac_write_chunk_function = desc->ac_write_chunk_function;
get_current_volumes (& return_ac->device);
if (return_ac->ac_left_speaker_gain != desc->ac_left_speaker_gain
|| return_ac->ac_right_speaker_gain != desc->ac_right_speaker_gain)
adjust_audio_volume (& desc->device);
return return_ac;
}
