int audriv_get_play_volume(void)
/* 演奏音量を 0 〜 255 内で得ます．0 は無音，255 は最大音量．
* 失敗すると -1 を返し，そうでない場合は 0 〜 255 内の音量を返します．
*/
{
#ifndef SGI_OLDAL
ALfixed lrgain[2];
ALpv pv;
double gain, l, r, min, max;
int volume;
int resource;
min = alFixedToDouble(out_ginfo.min.ll);
max = alFixedToDouble(out_ginfo.max.ll);
pv.param = AL_GAIN;
pv.value.ptr = lrgain;
pv.sizeIn = 2;
if(out == NULL)
resource = AL_DEFAULT_OUTPUT;
else
resource = alGetResource(out);
if(alGetParams(resource, &pv, 1) < 0)
{
audriv_err(ALERROR);
return -1;
}
l = alFixedToDouble(lrgain[0]);
r = alFixedToDouble(lrgain[1]);
if(l < min) l = min; else if(l > max) l = max;
if(r < min) r = min; else if(r > max) r = max;
gain = (l + r) / 2;
volume = (gain - min) * 256 / (max - min);
if(volume < 0)
volume = 0;
else if(volume > 255)
volume = 255;
return volume;
#else
long gain[4];
int volume;
gain[0] = AL_LEFT_SPEAKER_GAIN;
gain[2] = AL_RIGHT_SPEAKER_GAIN;
if(ALgetparams(AL_DEFAULT_DEVICE, gain, 4) < 0)
{
audriv_err(ALERROR);
return -1;
}
volume = (gain[1] + gain[3]) / 2;
if(volume < 0)
volume = 0;
else if(volume > 255)
volume = 255;
return volume;
#endif /* SGI_OLDAL */
}
static Bool audriv_al_set_rate(ALport port, unsigned long rate)
{
ALpv pv;
int r;
r = alGetResource(port);
pv.param = AL_RATE;
pv.value.ll = alIntToFixed(rate);
if(alSetParams(r, &pv, 1) < 0)
return False;
return True;
}
/*
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()));
}
}
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;
}
qboolean SNDDMA_Init(void)
{
ALconfig ac = NULL;
ALpv pvbuf[2];
s_loadas8bit = Cvar_Get("s_loadas8bit", "16", CVAR_ARCHIVE);
if ((int)s_loadas8bit->value)
dma.samplebits = 8;
else
dma.samplebits = 16;
if (dma.samplebits != 16) {
Com_Printf("Don't currently support %i-bit data. Forcing 16-bit.\n",
dma.samplebits);
dma.samplebits = 16;
Cvar_SetValue( "s_loadas8bit", false );
}
s_khz = Cvar_Get("s_khz", "0", CVAR_ARCHIVE);
switch ((int)s_khz->value) {
case 48:
dma.speed = AL_RATE_48000;
break;
case 44:
dma.speed = AL_RATE_44100;
break;
case 32:
dma.speed = AL_RATE_32000;
break;
case 22:
dma.speed = AL_RATE_22050;
break;
case 16:
dma.speed = AL_RATE_16000;
break;
case 11:
dma.speed = AL_RATE_11025;
break;
case 8:
dma.speed = AL_RATE_8000;
break;
default:
dma.speed = AL_RATE_22050;
Com_Printf("Don't currently support %i kHz sample rate. Using %i.\n",
(int)s_khz->value, (int)(dma.speed/1000));
}
sndchannels = Cvar_Get("sndchannels", "2", CVAR_ARCHIVE);
dma.channels = (int)sndchannels->value;
if (dma.channels != 2)
Com_Printf("Don't currently support %i sound channels. Try 2.\n",
sndchannels);
/***********************/
ac = alNewConfig();
alSetChannels( ac, AL_STEREO );
alSetSampFmt( ac, AL_SAMPFMT_TWOSCOMP );
alSetQueueSize( ac, QSND_BUFFER_FRAMES );
if (dma.samplebits == 8)
alSetWidth( ac, AL_SAMPLE_8 );
else
alSetWidth( ac, AL_SAMPLE_16 );
sgisnd_aport = alOpenPort( "Quake", "w", ac );
if (!sgisnd_aport)
{
printf( "failed to open audio port!\n" );
}
// set desired sample rate
pvbuf[0].param = AL_MASTER_CLOCK;
pvbuf[0].value.i = AL_CRYSTAL_MCLK_TYPE;
pvbuf[1].param = AL_RATE;
pvbuf[1].value.ll = alIntToFixed( dma.speed );
alSetParams( alGetResource( sgisnd_aport ), pvbuf, 2 );
if (pvbuf[1].sizeOut < 0)
printf( "illegal sample rate %d\n", dma.speed );
sgisnd_frames_per_ns = dma.speed * 1.0e-9;
dma.samples = sizeof(dma_buffer)/(dma.samplebits/8);
dma.submission_chunk = 1;
dma.buffer = (unsigned char *)dma_buffer;
dma.samplepos = 0;
alFreeConfig( ac );
return true;
}
static int al_play (ao_instance_t * _instance, int flags, sample_t * _samples)
{
al_instance_t * instance = (al_instance_t *) _instance;
int16_t int16_samples[256*6];
int chans = -1;
#ifdef LIBDTS_DOUBLE
convert_t samples[256 * 6];
int i;
for (i = 0; i < 256 * 6; i++)
samples[i] = _samples[i];
#else
convert_t * samples = _samples;
#endif
chans = channels_multi (flags);
flags &= DTS_CHANNEL_MASK | DTS_LFE;
if (instance->set_params) {
ALconfig config;
ALpv params[2];
config = alNewConfig ();
if (!config) {
fprintf (stderr, "alNewConfig failed\n");
return 1;
}
if (alSetChannels (config, chans)) {
fprintf (stderr, "alSetChannels failed\n");
return 1;
}
if (alSetConfig (instance->port, config)) {
fprintf (stderr, "alSetConfig failed\n");
return 1;
}
alFreeConfig (config);
params[0].param = AL_MASTER_CLOCK;
params[0].value.i = AL_CRYSTAL_MCLK_TYPE;
params[1].param = AL_RATE;
params[1].value.ll = alIntToFixed (instance->sample_rate);
if (alSetParams (alGetResource (instance->port), params, 2) < 0) {
fprintf (stderr, "alSetParams failed\n");
return 1;
}
instance->flags = flags;
instance->set_params = 0;
} else if ((flags == DTS_DOLBY) && (instance->flags == DTS_STEREO)) {
fprintf (stderr, "Switching from stereo to dolby surround\n");
instance->flags = DTS_DOLBY;
} else if ((flags == DTS_STEREO) && (instance->flags == DTS_DOLBY)) {
fprintf (stderr, "Switching from dolby surround to stereo\n");
instance->flags = DTS_STEREO;
} else if (flags != instance->flags)
return 1;
convert2s16_multi (samples, int16_samples, flags);
alWriteFrames (instance->port, int16_samples, 256);
return 0;
}
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 */
}
