static int audio_set_format(struct audio_info_struct *ai, ALconfig config)
{
if (alSetSampFmt(config,AL_SAMPFMT_TWOSCOMP) < 0)
fprintf(stderr,"audio_set_format : %s\n",alGetErrorString(oserror()));
if (alSetWidth(config,AL_SAMPLE_16) < 0)
fprintf(stderr,"audio_set_format : %s\n",alGetErrorString(oserror()));
return 0;
}
static int set_format(audio_output_t *ao, ALconfig config)
{
if (alSetSampFmt(config,AL_SAMPFMT_TWOSCOMP) < 0)
error1("set_format : %s",alGetErrorString(oserror()));
if (alSetWidth(config,AL_SAMPLE_16) < 0)
error1("set_format : %s",alGetErrorString(oserror()));
return 0;
}
RD_BOOL
sgi_open(void)
{
ALparamInfo pinfo;
static int warned = 0;
#if (defined(IRIX_DEBUG))
fprintf(stderr, "sgi_open: begin\n");
#endif
if (!warned && sgi_output_device)
{
warning("device-options not supported for libao-driver\n");
warned = 1;
}
if (alGetParamInfo(AL_DEFAULT_OUTPUT, AL_GAIN, &pinfo) < 0)
{
fprintf(stderr, "sgi_open: alGetParamInfo failed: %s\n",
alGetErrorString(oserror()));
}
min_volume = alFixedToDouble(pinfo.min.ll);
max_volume = alFixedToDouble(pinfo.max.ll);
volume_range = (max_volume - min_volume);
#if (defined(IRIX_DEBUG))
fprintf(stderr, "sgi_open: minvol = %lf, maxvol= %lf, range = %lf.\n",
min_volume, max_volume, volume_range);
#endif
audioconfig = alNewConfig();
if (audioconfig == (ALconfig) 0)
{
fprintf(stderr, "sgi_open: alNewConfig failed: %s\n", alGetErrorString(oserror()));
return False;
}
output_port = alOpenPort("rdpsnd", "w", 0);
if (output_port == (ALport) 0)
{
fprintf(stderr, "sgi_open: alOpenPort failed: %s\n", alGetErrorString(oserror()));
return False;
}
#if (defined(IRIX_DEBUG))
fprintf(stderr, "sgi_open: returning\n");
#endif
return True;
}
void
wave_out_volume(uint16 left, uint16 right)
{
double gainleft, gainright;
ALpv pv[1];
ALfixed gain[8];
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_volume: begin\n");
fprintf(stderr, "left='%d', right='%d'\n", left, right);
#endif
gainleft = (double) left / IRIX_MAX_VOL;
gainright = (double) right / IRIX_MAX_VOL;
gain[0] = alDoubleToFixed(min_volume + gainleft * volume_range);
gain[1] = alDoubleToFixed(min_volume + gainright * volume_range);
pv[0].param = AL_GAIN;
pv[0].value.ptr = gain;
pv[0].sizeIn = 8;
if (alSetParams(AL_DEFAULT_OUTPUT, pv, 1) < 0)
{
fprintf(stderr, "wave_out_volume: alSetParams failed: %s\n",
alGetErrorString(oserror()));
return;
}
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_volume: returning\n");
#endif
}
BOOL
wave_out_open(void)
{
ALparamInfo pinfo;
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_open: begin\n");
#endif
if (alGetParamInfo(AL_DEFAULT_OUTPUT, AL_GAIN, &pinfo) < 0)
{
fprintf(stderr, "wave_out_open: alGetParamInfo failed: %s\n",
alGetErrorString(oserror()));
}
min_volume = alFixedToDouble(pinfo.min.ll);
max_volume = alFixedToDouble(pinfo.max.ll);
volume_range = (max_volume - min_volume);
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_open: minvol = %lf, maxvol= %lf, range = %lf.\n",
min_volume, max_volume, volume_range);
#endif
queue_lo = queue_hi = 0;
audioconfig = alNewConfig();
if (audioconfig == (ALconfig) 0)
{
fprintf(stderr, "wave_out_open: alNewConfig failed: %s\n",
alGetErrorString(oserror()));
return False;
}
output_port = alOpenPort("rdpsnd", "w", 0);
if (output_port == (ALport) 0)
{
fprintf(stderr, "wave_out_open: alOpenPort failed: %s\n",
alGetErrorString(oserror()));
return False;
}
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_open: returning\n");
#endif
return True;
}
static int set_rate(audio_output_t *ao, ALconfig config)
{
int dev = alGetDevice(config);
ALpv params[1];
/* Make sure the device is OK */
if (dev < 0)
{
error1("set_rate: %s",alGetErrorString(oserror()));
return 1;
}
params[0].param = AL_OUTPUT_RATE;
params[0].value.ll = alDoubleToFixed(ao->rate);
if (alSetParams(dev, params,1) < 0)
error1("set_rate: %s",alGetErrorString(oserror()));
return 0;
}
static int audio_set_rate(struct audio_info_struct *ai, ALconfig config)
{
int dev = alGetDevice(config);
ALpv params[1];
/* Make sure the device is OK */
if (dev < 0)
{
fprintf(stderr,"audio_set_rate : %s\n",alGetErrorString(oserror()));
return 1;
}
params[0].param = AL_OUTPUT_RATE;
params[0].value.ll = alDoubleToFixed(ai->rate);
if (alSetParams(dev, params,1) < 0)
fprintf(stderr,"audio_set_rate : %s\n",alGetErrorString(oserror()));
return 0;
}
static int set_channels(audio_output_t *ao, ALconfig config)
{
int ret;
if(ao->channels == 2)
ret = alSetChannels(config, AL_STEREO);
else
ret = alSetChannels(config, AL_MONO);
if (ret < 0)
error1("set_channels : %s",alGetErrorString(oserror()));
return 0;
}
static int audio_set_channels(struct audio_info_struct *ai, ALconfig config)
{
int ret;
if(ai->channels == 2)
ret = alSetChannels(config, AL_STEREO);
else
ret = alSetChannels(config, AL_MONO);
if (ret < 0)
fprintf(stderr,"audio_set_channels : %s\n",alGetErrorString(oserror()));
return 0;
}
static int set_format(audio_output_t *ao, ALconfig config)
{
if(ao->format == MPG123_ENC_FLOAT_32)
{
if(alSetSampFmt(config, AL_SAMPFMT_FLOAT) < 0)
{
error1("SetSampFmt: %s", alGetErrorString(oserror()));
return -1;
}
} else
{
if(alSetSampFmt(config, AL_SAMPFMT_TWOSCOMP) < 0)
{
error1("SetSampFmt: %s", alGetErrorString(oserror()));
return -1;
}
if(alSetWidth(config, AL_SAMPLE_16) < 0)
{
error1("SetWidth: %s", alGetErrorString(oserror()));
return -1;
}
}
return 0;
}
/*
Set the sample rate of an audio port.
*/
void setrate (ALport port, double rate)
{
int rv;
ALpv params;
rv = alGetResource(port);
params.param = AL_RATE;
params.value.ll = alDoubleToFixed(rate);
if (alSetParams(rv, ¶ms, 1) < 0)
{
printf("alSetParams failed: %s\n", alGetErrorString(oserror()));
}
}
void *OpenPlay(struct FFTSound *fftsound){
struct Sgiplay *sgiplay;
sgiplay=malloc(sizeof(struct Sgiplay));
sgiplay->out_c = alNewConfig();
alSetQueueSize(sgiplay->out_c, _I);
alSetSampFmt(sgiplay->out_c,AL_SAMPFMT_DOUBLE);
alSetChannels(sgiplay->out_c, fftsound->samps_per_frame);
sgiplay->out_port = alOpenPort("Ceres Play", "w", sgiplay->out_c);
if(!sgiplay->out_port){
printf("Couldn't open output port: %s\n",alGetErrorString(oserror()));
free(sgiplay);
return NULL;
}
return sgiplay;
}
BOOL
wave_out_set_format(WAVEFORMATEX * pwfx)
{
int channels;
int frameSize, channelCount;
ALpv params;
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_set_format: init...\n");
#endif
g_swapaudio = False;
if (pwfx->wBitsPerSample == 8)
width = AL_SAMPLE_8;
else if (pwfx->wBitsPerSample == 16)
{
width = AL_SAMPLE_16;
/* Do we need to swap the 16bit values? (Are we BigEndian) */
#if (defined(B_ENDIAN))
g_swapaudio = 1;
#else
g_swapaudio = 0;
#endif
}
/* Limited support to configure an opened audio port in IRIX. The
number of channels is a static setting and can not be changed after
a port is opened. So if the number of channels remains the same, we
can configure other settings; otherwise we have to reopen the audio
port, using same config. */
channels = pwfx->nChannels;
g_snd_rate = pwfx->nSamplesPerSec;
alSetSampFmt(audioconfig, AL_SAMPFMT_TWOSCOMP);
alSetWidth(audioconfig, width);
if (channels != alGetChannels(audioconfig))
{
alClosePort(output_port);
alSetChannels(audioconfig, channels);
output_port = alOpenPort("rdpsnd", "w", audioconfig);
if (output_port == (ALport) 0)
{
fprintf(stderr, "wave_out_set_format: alOpenPort failed: %s\n",
alGetErrorString(oserror()));
return False;
}
}
resource = alGetResource(output_port);
maxFillable = alGetFillable(output_port);
channelCount = alGetChannels(audioconfig);
frameSize = alGetWidth(audioconfig);
if (frameSize == 0 || channelCount == 0)
{
fprintf(stderr, "wave_out_set_format: bad frameSize or channelCount\n");
return False;
}
combinedFrameSize = frameSize * channelCount;
params.param = AL_RATE;
params.value.ll = (long long) g_snd_rate << 32;
if (alSetParams(resource, ¶ms, 1) < 0)
{
fprintf(stderr, "wave_set_format: alSetParams failed: %s\n",
alGetErrorString(oserror()));
return False;
}
if (params.sizeOut < 0)
{
fprintf(stderr, "wave_set_format: invalid rate %d\n", g_snd_rate);
return False;
}
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_set_format: returning...\n");
#endif
return True;
}
// open & setup audio device
// return: 1=success 0=fail
static int init(int rate, int channels, int format, int flags) {
int smpwidth, smpfmt;
int rv = AL_DEFAULT_OUTPUT;
smpfmt = fmt2sgial(&format, &smpwidth);
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_InitInfo, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
{ /* from /usr/share/src/dmedia/audio/setrate.c */
double frate, realrate;
ALpv x[2];
if(ao_subdevice) {
rv = alGetResourceByName(AL_SYSTEM, ao_subdevice, AL_OUTPUT_DEVICE_TYPE);
if (!rv) {
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InvalidDevice);
return 0;
}
}
frate = rate;
x[0].param = AL_RATE;
x[0].value.ll = alDoubleToFixed(rate);
x[1].param = AL_MASTER_CLOCK;
x[1].value.i = AL_CRYSTAL_MCLK_TYPE;
if (alSetParams(rv,x, 2)<0) {
mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetParms_Samplerate, alGetErrorString(oserror()));
}
if (x[0].sizeOut < 0) {
mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetAlRate);
}
if (alGetParams(rv,x, 1)<0) {
mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantGetParms, alGetErrorString(oserror()));
}
realrate = alFixedToDouble(x[0].value.ll);
if (frate != realrate) {
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_SampleRateInfo, realrate, frate);
}
sample_rate = (int)realrate;
}
bytes_per_frame = channels * smpwidth;
ao_data.samplerate = sample_rate;
ao_data.channels = channels;
ao_data.format = format;
ao_data.bps = sample_rate * bytes_per_frame;
ao_data.buffersize=131072;
ao_data.outburst = ao_data.buffersize/16;
ao_config = alNewConfig();
if (!ao_config) {
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror()));
return 0;
}
if(alSetChannels(ao_config, channels) < 0 ||
alSetWidth(ao_config, smpwidth) < 0 ||
alSetSampFmt(ao_config, smpfmt) < 0 ||
alSetQueueSize(ao_config, sample_rate) < 0 ||
alSetDevice(ao_config, rv) < 0) {
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror()));
return 0;
}
ao_port = alOpenPort("mplayer", "w", ao_config);
if (!ao_port) {
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitOpenAudioFailed, alGetErrorString(oserror()));
return 0;
}
// printf("ao_sgi, init: port %d config %d\n", ao_port, ao_config);
queue_size = alGetQueueSize(ao_config);
return 1;
}
// open & setup audio device
// return: 1=success 0=fail
static int init(int rate, int channels, int format, int flags) {
int smpwidth, smpfmt;
int rv = AL_DEFAULT_OUTPUT;
smpfmt = fmt2sgial(&format, &smpwidth);
mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] init: Samplerate: %iHz Channels: %s Format %s\n", rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
{ /* from /usr/share/src/dmedia/audio/setrate.c */
double frate, realrate;
ALpv x[2];
if(ao_subdevice) {
rv = alGetResourceByName(AL_SYSTEM, ao_subdevice, AL_OUTPUT_DEVICE_TYPE);
if (!rv) {
mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] play: invalid device.\n");
return 0;
}
}
frate = rate;
x[0].param = AL_RATE;
x[0].value.ll = alDoubleToFixed(rate);
x[1].param = AL_MASTER_CLOCK;
x[1].value.i = AL_CRYSTAL_MCLK_TYPE;
if (alSetParams(rv,x, 2)<0) {
mp_tmsg(MSGT_AO, MSGL_WARN, "[AO SGI] init: setparams failed: %s\nCould not set desired samplerate.\n", alGetErrorString(oserror()));
}
if (x[0].sizeOut < 0) {
mp_tmsg(MSGT_AO, MSGL_WARN, "[AO SGI] init: AL_RATE was not accepted on the given resource.\n");
}
if (alGetParams(rv,x, 1)<0) {
mp_tmsg(MSGT_AO, MSGL_WARN, "[AO SGI] init: getparams failed: %s\n", alGetErrorString(oserror()));
}
realrate = alFixedToDouble(x[0].value.ll);
if (frate != realrate) {
mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] init: samplerate is now %f (desired rate is %f)\n", realrate, frate);
}
sample_rate = (int)realrate;
}
bytes_per_frame = channels * smpwidth;
ao_data.samplerate = sample_rate;
ao_data.channels = channels;
ao_data.format = format;
ao_data.bps = sample_rate * bytes_per_frame;
ao_data.buffersize=131072;
ao_data.outburst = ao_data.buffersize/16;
ao_config = alNewConfig();
if (!ao_config) {
mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] init: %s\n", alGetErrorString(oserror()));
return 0;
}
if(alSetChannels(ao_config, channels) < 0 ||
alSetWidth(ao_config, smpwidth) < 0 ||
alSetSampFmt(ao_config, smpfmt) < 0 ||
alSetQueueSize(ao_config, sample_rate) < 0 ||
alSetDevice(ao_config, rv) < 0) {
mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] init: %s\n", alGetErrorString(oserror()));
return 0;
}
ao_port = alOpenPort("mplayer", "w", ao_config);
if (!ao_port) {
mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] init: Unable to open audio channel: %s\n", alGetErrorString(oserror()));
return 0;
}
// printf("ao_sgi, init: port %d config %d\n", ao_port, ao_config);
queue_size = alGetQueueSize(ao_config);
return 1;
}
main()
{
LS_DATA ls_data;
FILE *fp;
double gains[MAX_CHANNELS];
int azimuth=-10;
int elevation=14;
double *gainptr, gain;
short *inptr,*outptr;
short out[BUFFER_LENGTH][MAX_CHANNELS];
ALconfig c;
AFfilehandle fh;
ALport p;
short *in;
long frames;
int i,j,k;
int numchannels = 8; /* change this according your output device*/
int ls_set_dim = 3;
int ls_num = 8;
int ls_dirs[MAX_FIELD_AM]={-30,0, 30,0, -45,45, 45,45, -90,0,
90,0, 180,0, 180,45};
/* change these according to your loudspeaker positioning*/
/* defining loudspeaker data */
define_loudspeakers(&ls_data, ls_set_dim, ls_num, ls_dirs);
/* ls_data is a struct containing matrices etc
ls_set_dim is 2 if loudspeakers are on a (horizontal) plane
ls_set_dim is 3 if also elevated or descended loudpeakers exist
ls_num is the number of loudspeakers
ls_dirs is an array containing the angular directions of loudsp*/
/* gain factors for virtual source in direction
(int azimuth, int elevation) */
vbap(gains, &ls_data, azimuth, elevation);
/* panning monophonic stream float *in
to multiple outputs float *out[]
with gain factors float *gains */
/* input audio*/
if((fh=afOpenFile("myaiff.aiff","r",0))==NULL){
fprintf(stderr, "Could not open file myaiff.aiff\n");
exit(-1);
}
frames=AFgetframecnt(fh,AF_DEFAULT_TRACK);
if(afGetChannels(fh, AF_DEFAULT_TRACK) != 1){
fprintf(stderr, "Supports only mono aiff-files\n");
exit(-1);
}
in= malloc(frames*sizeof(short)*2);
afReadFrames(fh,AF_DEFAULT_TRACK,in,frames);
/*opening the audio port*/
c = alNewConfig();
if (!c) {
printf("Couldn't create ALconfig:%s\n", alGetErrorString(oserror()));
exit(-1);
}
alSetChannels(c,numchannels);
ALsetqueuesize(c,BUFFER_LENGTH);
p = alOpenPort("alVBAP example","w",c);
if (!p) {
printf("port open failed:%s\n", alGetErrorString(oserror()));
exit(-1);
}
fprintf(stderr,"\nPanning audio");
for(j=0;j<(frames/BUFFER_LENGTH);j++){
inptr = &(in[j*BUFFER_LENGTH]); /* audio to be panned */
outptr= out[0];
for (i=0; i<BUFFER_LENGTH; i++){ /* panning */
gainptr=gains;
for (k=0; k<numchannels; k++){
*outptr++ = (short) ((double) *inptr * *gainptr++);
}
inptr++;
}
alWriteFrames(p, out, BUFFER_LENGTH); /* write frames */
fprintf(stderr,".");
}
/*write rest samples out*/
inptr = &(in[j*BUFFER_LENGTH]); /* partial buffer */
outptr= out[0];
for (i=0; i<(frames-BUFFER_LENGTH*j); i++){ /* panning */
gainptr=gains;
for (k=0; k<numchannels; k++){
*outptr++ = (short) ((double) *inptr * *gainptr++);
}
inptr++;
}
for (;i<BUFFER_LENGTH; i++){ /* zeros */
for (k=0; k<numchannels; k++){
*outptr++ = 0;
}
}
alWriteFrames(p, out, BUFFER_LENGTH); /* write frames */
fprintf(stderr,".");
printf("\n\nDone!\n\n");
}
static int open_sgi(audio_output_t *ao)
{
int current_dev;
ALport port = NULL;
ALconfig config = alNewConfig();
ao->userptr = NULL;
/* Test for correct completion */
if(config == 0)
{
error1("open_sgi: %s", alGetErrorString(oserror()));
return -1;
}
/* Setup output device to specified device name. If there is no device name
specified in ao structure, use the default for output */
if((ao->device) != NULL)
{
current_dev = alGetResourceByName(AL_SYSTEM, ao->device, AL_OUTPUT_DEVICE_TYPE);
debug2("Dev: %s %i", ao->device, current_dev);
if(!current_dev)
{
int i, numOut;
char devname[32];
ALpv pv[1];
ALvalue *alvalues;
error2("Invalid audio resource: %s (%s)", ao->device, alGetErrorString(oserror()));
if((numOut= alQueryValues(AL_SYSTEM,AL_DEFAULT_OUTPUT,0,0,0,0))>=0)
fprintf(stderr, "There are %d output devices on this system.\n", numOut);
else
{
fprintf(stderr, "Can't find output devices. alQueryValues failed: %s\n", alGetErrorString(oserror()));
goto open_sgi_bad;
}
alvalues = malloc(sizeof(ALvalue) * numOut);
i = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, alvalues, numOut, pv, 0);
if(i == -1)
error1("alQueryValues: %s", alGetErrorString(oserror()));
else
{
for(i=0; i < numOut; i++)
{
pv[0].param = AL_NAME;
pv[0].value.ptr = devname;
pv[0].sizeIn = 32;
alGetParams(alvalues[i].i, pv, 1);
fprintf(stderr, "%i: %s\n", i, devname);
}
}
free(alvalues);
goto open_sgi_bad;
}
if(alSetDevice(config, current_dev) < 0)
{
error1("open: alSetDevice : %s",alGetErrorString(oserror()));
goto open_sgi_bad;
}
} else
current_dev = AL_DEFAULT_OUTPUT;
/* Set the device */
if(alSetDevice(config, current_dev) < 0)
{
error1("open_sgi: %s", alGetErrorString(oserror()));
goto open_sgi_bad;
}
/* Set port parameters */
if(alSetQueueSize(config, 131069) < 0)
{
error1("open_sgi: setting audio buffer failed: %s", alGetErrorString(oserror()));
goto open_sgi_bad;
}
if( set_format(ao, config) < 0
|| set_rate(ao, config) < 0
|| set_channels(ao, config) < 0 )
goto open_sgi_bad;
/* Open the audio port */
port = alOpenPort("mpg123-VSC", "w", config);
if(port == NULL)
{
error1("Unable to open audio channel: %s", alGetErrorString(oserror()));
goto open_sgi_bad;
}
ao->userptr = (void*)port;
alFreeConfig(config);
return 1;
open_sgi_bad:
/* clean up and return error */
alFreeConfig(config);
return -1;
}
/*
* public methods (static but exported through the sysdep_dsp or plugin struct)
*/
static void *
irix_dsp_create(const void *flags)
{
ALpv pvs[4];
long tempbits, tempchan;
int oldrate;
struct irix_dsp_priv_data *priv = NULL;
struct sysdep_dsp_struct *dsp = NULL;
const struct sysdep_dsp_create_params *params = flags;
ALconfig devAudioConfig;
/* allocate the dsp struct */
if (!(dsp = calloc(1, sizeof(struct sysdep_dsp_struct))))
{
fprintf(stderr, "Error: malloc failed for struct sysdep_dsp_struct\n");
return NULL;
}
/* allocate private data */
if(!(priv = calloc(1, sizeof(struct irix_dsp_priv_data))))
{
fprintf(stderr, "Error: malloc failed for struct irix_dsp_priv_data\n");
free(dsp);
return NULL;
}
/* fill in the functions and some data */
priv->port_status = -1;
dsp->_priv = priv;
dsp->get_freespace = irix_dsp_get_freespace;
dsp->write = irix_dsp_write;
dsp->destroy = irix_dsp_destroy;
dsp->hw_info.type = params->type;
dsp->hw_info.samplerate = params->samplerate;
tempchan = (dsp->hw_info.type & SYSDEP_DSP_STEREO) ? 2 : 1;
tempbits = (dsp->hw_info.type & SYSDEP_DSP_16BIT) ? 2 : 1;
#ifdef IRIX_DEBUG
fprintf(stderr, "Source Format is %dHz, %d bit, %s, with bufsize %f\n",
dsp->hw_info.samplerate,
tempbits * 8, (tempchan == 2) ? "stereo" : "mono",
params->bufsize);
#endif
/*
* Since AL wants signed data in either case, and 8-bit data from
* core xmame is unsigned, let the core xmame convert everything
* to 16-bit signed.
*/
if (tempbits == 1)
{
dsp->hw_info.type |= SYSDEP_DSP_16BIT;
tempbits = 2;
}
/*
* Get the current hardware sampling rate
*/
pvs[0].param = AL_RATE;
if (alGetParams(AL_DEFAULT_OUTPUT, pvs, 1) < 0)
{
fprintf(stderr, "alGetParams failed: %s\n", alGetErrorString(oserror()));
irix_dsp_destroy(dsp);
return NULL;
}
oldrate = pvs[0].value.i;
/*
* If requested samplerate is different than current hardware rate,
* set it.
*/
if (oldrate != dsp->hw_info.samplerate)
{
int audioHardwareRate = oldrate;
fprintf(stderr, "System sample rate was %dHz, forcing %dHz instead.\n",
oldrate, dsp->hw_info.samplerate);
/*
* If the desired rate is unsupported, most devices (such as RAD) will
* force the device rate to be as close as possible to the desired rate.
* Since close isn't going to help us here, we avoid the call entirely,
* and let core xmame audio convert to our rate.
*/
if (RateSupported(AL_DEFAULT_OUTPUT, (float) dsp->hw_info.samplerate))
{
/* Set desired sample rate */
pvs[0].param = AL_MASTER_CLOCK;
pvs[0].value.i = AL_CRYSTAL_MCLK_TYPE;
pvs[1].param = AL_RATE;
pvs[1].value.i = dsp->hw_info.samplerate;
alSetParams(AL_DEFAULT_OUTPUT, pvs, 2);
/* Get the new sample rate */
pvs[0].param = AL_RATE;
if (alGetParams(AL_DEFAULT_OUTPUT, pvs, 1) < 0)
{
fprintf(stderr, "alGetParams failed: %s\n",
alGetErrorString(oserror()));
irix_dsp_destroy(dsp);
return NULL;
}
audioHardwareRate = pvs[0].value.i;
}
if (audioHardwareRate != dsp->hw_info.samplerate)
{
fprintf(stderr, "Requested rate of %dHz is not supported by "
"the audio hardware, so forcing\n"
"playback at %dHz.\n",
dsp->hw_info.samplerate, audioHardwareRate);
dsp->hw_info.samplerate = audioHardwareRate;
}
}
/* create a config descriptor */
devAudioConfig = alNewConfig();
if (devAudioConfig == NULL) {
fprintf(stderr, "Cannot get a Descriptor. Exiting..\n");
irix_dsp_destroy(dsp);
return NULL;
}
#ifdef FORCEMONO
dsp->hw_info.type &= ~SYSDEP_DSP_STEREO;
tempchan = 1;
#endif
priv->buffer_samples = dsp->hw_info.samplerate * params->bufsize;
priv->buffer_samples *= tempchan;
priv->sampwidth = tempbits;
priv->sampchan = tempchan;
fprintf(stderr, "Setting sound to %dHz, %d bit, %s\n",
dsp->hw_info.samplerate,
tempbits * 8, (tempchan == 2) ? "stereo" : "mono");
/* source specific audio parameters */
alSetChannels(devAudioConfig, tempchan);
alSetQueueSize(devAudioConfig, priv->buffer_samples);
alSetWidth(devAudioConfig, tempbits);
alSetSampFmt(devAudioConfig, AL_SAMPFMT_TWOSCOMP);
/* Open the audio port with the parameters we setup */
priv->devAudio = alOpenPort("audio_fd", "w", devAudioConfig);
if (priv->devAudio == NULL)
{
fprintf(stderr, "Error: Cannot get an audio channel descriptor.\n");
irix_dsp_destroy(dsp);
return NULL;
}
alFreeConfig(devAudioConfig);
/*
* Since we don't use FD's with AL, we use this to inform us
* of success
*/
priv->port_status = 0;
return dsp;
}
ad_rec_t *ad_open_sps (int32 samples_per_sec)
{
// fprintf(stderr, "A/D library not implemented\n");
ad_rec_t *handle;
ALpv pv;
int device = AL_DEFAULT_INPUT;
ALconfig portconfig = alNewConfig();
ALport port;
int32 sampleRate;
long long gainValue = alDoubleToFixed(8.5);
pv.param = AL_GAIN;
pv.sizeIn = 1;
pv.value.ptr = &gainValue;
if (alSetParams(device, &pv, 1)<0) {
fprintf(stderr, "setparams failed: %s\n",alGetErrorString(oserror()));
return NULL;
}
pv.param = AL_RATE;
pv.value.ll = alDoubleToFixed(samples_per_sec);
if (alSetParams(device, &pv, 1)<0) {
fprintf(stderr, "setparams failed: %s\n",alGetErrorString(oserror()));
return NULL;
}
if (pv.sizeOut < 0) {
/*
* Not all devices will allow setting of AL_RATE (for example, digital
* inputs run only at the frequency of the external device). Check
* to see if the rate was accepted.
*/
fprintf(stderr, "AL_RATE was not accepted on the given resource\n");
return NULL;
}
if (alGetParams(device, &pv, 1)<0) {
fprintf(stderr, "getparams failed: %s\n",alGetErrorString(oserror()));
}
sampleRate = (int32)alFixedToDouble(pv.value.ll);
#if 0
printf("sample rate is set to %d\n", sampleRate);
#endif
if (alSetChannels(portconfig, 1) < 0) {
fprintf(stderr, "alSetChannels failed: %s\n",alGetErrorString(oserror()));
return NULL;
}
port = alOpenPort(" Sphinx-II input port", "r", portconfig);
if (!port) {
fprintf(stderr, "alOpenPort failed: %s\n", alGetErrorString(oserror()));
return NULL;
}
if ((handle = (ad_rec_t *) calloc (1, sizeof(ad_rec_t))) == NULL) {
fprintf(stderr, "calloc(%d) failed\n", sizeof(ad_rec_t));
abort();
}
handle->audio = port;
handle->recording = 0;
handle->sps = sampleRate;
handle->bps = sizeof(int16);
alFreeConfig(portconfig);
return handle;
}
static int open_sgi(audio_output_t *ao)
{
int dev = AL_DEFAULT_OUTPUT;
ALconfig config = alNewConfig();
ALport port = NULL;
/* Test for correct completion */
if (config == 0) {
error1("open_sgi: %s",alGetErrorString(oserror()));
return -1;
}
/* Set port parameters */
if(ao->channels == 2)
alSetChannels(config, AL_STEREO);
else
alSetChannels(config, AL_MONO);
alSetWidth(config, AL_SAMPLE_16);
alSetSampFmt(config,AL_SAMPFMT_TWOSCOMP);
alSetQueueSize(config, 131069);
/* Setup output device to specified module. If there is no module
specified in ao structure, use the default four output */
if ((ao->device) != NULL) {
char *dev_name;
dev_name=malloc((strlen(ao->device) + strlen(analog_output_res_name) + 1) *
sizeof(char));
strcpy(dev_name,ao->device);
strcat(dev_name,analog_output_res_name);
/* Find the asked device resource */
dev=alGetResourceByName(AL_SYSTEM,dev_name,AL_DEVICE_TYPE);
/* Free allocated space */
free(dev_name);
if (!dev) {
error2("Invalid audio resource: %s (%s)",dev_name, alGetErrorString(oserror()));
return -1;
}
}
/* Set the device */
if (alSetDevice(config,dev) < 0)
{
error1("open_sgi: %s",alGetErrorString(oserror()));
return -1;
}
/* Open the audio port */
port = alOpenPort("mpg123-VSC", "w", config);
if(port == NULL) {
error1("Unable to open audio channel: %s", alGetErrorString(oserror()));
return -1;
}
ao->userptr = (void*)port;
set_format(ao, config);
set_channels(ao, config);
set_rate(ao, config);
alFreeConfig(config);
return 1;
}