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
}
/*
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()));
}
}
/**
* Initialize global audio interface to use microphone input.
*
* @param rate [in] sampling rate in Hz
*
* @return TRUE on success, FALSE on failure.
*/
static boolean
adin_o2_setup_global(double rate)
{
int device;
ALpv setPVbuf[4];
setPVbuf[0].param = AL_INTERFACE;
setPVbuf[0].value.i = alGetResourceByName(AL_SYSTEM, "Microphone", AL_INTERFACE_TYPE);
setPVbuf[1].param = AL_MASTER_CLOCK;
setPVbuf[1].value.i = AL_CRYSTAL_MCLK_TYPE;
setPVbuf[2].param = AL_RATE;
setPVbuf[2].value.ll= alDoubleToFixed(rate);
device = alGetResourceByName(AL_SYSTEM, "Microphone", AL_DEVICE_TYPE);
if (alSetParams(device, setPVbuf, 3) < 0) {
return FALSE;
} else {
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;
}
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;
}
// 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;
}
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 */
}
