/**
* Device initialization: check device capability and open for recording.
*
* @param sfreq [in] required sampling frequency.
* @param dummy [in] a dummy data
*
* @return TRUE on success, FALSE on failure.
*/
boolean
adin_mic_standby(int sfreq, void *dummy)
{
long rate;
long prec = AL_SAMPLE_16;
long encd = AL_SAMPFMT_TWOSCOMP;
long chan = AL_MONO;
/* global setup */
rate = sfreq;
if (adin_o2_setup_global((double)rate) == FALSE) { /* failed */
jlog("Error: adin_o2: cannot setup microphone device (global)\n");
return(FALSE);
}
/* local parameter setup */
if ((ac = ALnewconfig()) == 0) {
jlog("Error: adin_o2: cannot config microphone device (local)\n");
return(FALSE);
}
ALsetqueuesize(ac, rate * 2 * 1); /* 2 sec. of mono. */
ALsetwidth(ac, prec);
ALsetchannels(ac, chan);
ALsetsampfmt(ac, encd);
jlog("Stat: adin_o2: local microphone port successfully initialized\n");
return(TRUE);
}
static int
set_channels (ALconfig config, unsigned int nchan)
{
switch (nchan)
{
case 1: ALsetchannels (config, AL_MONO); break;
#if HAVE_STEREO
case 2: ALsetchannels (config, AL_STEREO); break;
#endif /* HAVE_STEREO */
default:
report_file_error ("Unsupported channel count",
Fcons (make_int (nchan), Qnil));
return -1;
}
return 0;
}
/*
* Open and initialize an audio port.
*/
static ALport sfxInitAudioPort(int init) {
ALport ap;
ALconfig audioPortConfig;
long pvbuf[6];
if (init) {
pvbuf[0] = AL_LEFT_SPEAKER_GAIN;
pvbuf[2] = AL_RIGHT_SPEAKER_GAIN;
pvbuf[4] = AL_OUTPUT_RATE;
ALgetparams(AL_DEFAULT_DEVICE, pvbuf, 6L);
currLeftGain = origLeftGain = pvbuf[1];
currRightGain = origRightGain = pvbuf[3];
origOutputRate = pvbuf[5];
sfxSetGainIndex((origLeftGain+origRightGain)/2);
}
/*
* Configure and open audio port.
*/
audioPortConfig = ALnewconfig();
ALsetwidth(audioPortConfig, AL_SAMPLE_16);
ALsetchannels(audioPortConfig, AL_STEREO);
ALsetqueuesize(audioPortConfig, 16000);
ap = ALopenport("spacetrek", "w", audioPortConfig);
ALfreeconfig(audioPortConfig);
return ap;
}
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 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;
}
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;
}
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;
}
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();
}
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;
}