int sound_update(unsigned int cycles)
{
int delta, preamp, time, l, r, *ptr;
/* Run PSG & FM chips until end of frame */
SN76489_Update(cycles);
fm_update(cycles);
/* FM output pre-amplification */
preamp = config.fm_preamp;
/* FM frame initial timestamp */
time = fm_cycles_start;
/* Restore last FM outputs from previous frame */
l = fm_last[0];
r = fm_last[1];
/* FM buffer start pointer */
ptr = fm_buffer;
/* flush FM samples */
if (config.hq_fm)
{
/* high-quality Band-Limited synthesis */
do
{
/* left channel */
delta = ((*ptr++ * preamp) / 100) - l;
l += delta;
blip_add_delta(snd.blips[0][0], time, delta);
/* right channel */
delta = ((*ptr++ * preamp) / 100) - r;
r += delta;
blip_add_delta(snd.blips[0][1], time, delta);
/* increment time counter */
time += fm_cycles_ratio;
}
while (time < cycles);
}
else
{
/* faster Linear Interpolation */
do
{
/* left channel */
delta = ((*ptr++ * preamp) / 100) - l;
l += delta;
blip_add_delta_fast(snd.blips[0][0], time, delta);
/* right channel */
delta = ((*ptr++ * preamp) / 100) - r;
r += delta;
blip_add_delta_fast(snd.blips[0][1], time, delta);
/* increment time counter */
time += fm_cycles_ratio;
}
while (time < cycles);
}
/* reset FM buffer pointer */
fm_ptr = fm_buffer;
/* save last FM output for next frame */
fm_last[0] = l;
fm_last[1] = r;
/* adjust FM cycle counters for next frame */
fm_cycles_count = fm_cycles_start = time - cycles;
/* end of blip buffers time frame */
blip_end_frame(snd.blips[0][0], cycles);
blip_end_frame(snd.blips[0][1], cycles);
/* return number of available samples */
return blip_samples_avail(snd.blips[0][0]);
}
void cdd_read_audio(unsigned int samples)
{
/* previous audio outputs */
int16 l = cdd.audio[0];
int16 r = cdd.audio[1];
/* get number of internal clocks (samples) needed */
samples = blip_clocks_needed(blip[0], samples);
/* audio track playing ? */
if (!Pico_mcd->s68k_regs[0x36+0] && cdd.toc.tracks[cdd.index].fd)
{
int i, mul, delta;
/* current CD-DA fader volume */
int curVol = cdd.volume;
/* CD-DA fader volume setup (0-1024) */
int endVol = Pico_mcd->regs[0x34>>1].w >> 4;
/* read samples from current block */
#ifdef USE_LIBTREMOR
if (cdd.toc.tracks[cdd.index].vf.datasource)
{
int len, done = 0;
int16 *ptr = (int16 *) (cdc.ram);
samples = samples * 4;
while (done < samples)
{
len = ov_read(&cdd.toc.tracks[cdd.index].vf, (char *)(cdc.ram + done), samples - done, 0);
if (len <= 0)
{
done = samples;
break;
}
done += len;
}
samples = done / 4;
/* process 16-bit (host-endian) stereo samples */
for (i=0; i<samples; i++)
{
/* CD-DA fader multiplier (cf. LC7883 datasheet) */
/* (MIN) 0,1,2,3,4,8,12,16,20...,1020,1024 (MAX) */
mul = (curVol & 0x7fc) ? (curVol & 0x7fc) : (curVol & 0x03);
/* left channel */
delta = ((ptr[0] * mul) / 1024) - l;
ptr++;
l += delta;
blip_add_delta_fast(blip[0], i, delta);
/* right channel */
delta = ((ptr[0] * mul) / 1024) - r;
ptr++;
r += delta;
blip_add_delta_fast(blip[1], i, delta);
/* update CD-DA fader volume (one step/sample) */
if (curVol < endVol)
{
/* fade-in */
curVol++;
}
else if (curVol > endVol)
{
/* fade-out */
curVol--;
}
else if (!curVol)
{
/* audio will remain muted until next setup */
break;
}
}
}
else
#endif
{
#ifdef LSB_FIRST
int16 *ptr = (int16 *) (cdc.ram);
#else
uint8 *ptr = cdc.ram;
#endif
fread(cdc.ram, 1, samples * 4, cdd.toc.tracks[cdd.index].fd);
/* process 16-bit (little-endian) stereo samples */
for (i=0; i<samples; i++)
{
/* CD-DA fader multiplier (cf. LC7883 datasheet) */
/* (MIN) 0,1,2,3,4,8,12,16,20...,1020,1024 (MAX) */
mul = (curVol & 0x7fc) ? (curVol & 0x7fc) : (curVol & 0x03);
/* left channel */
#ifdef LSB_FIRST
delta = ((ptr[0] * mul) / 1024) - l;
ptr++;
#else
delta = (((int16)((ptr[0] + ptr[1]*256)) * mul) / 1024) - l;
ptr += 2;
#endif
l += delta;
blip_add_delta_fast(blip[0], i, delta);
/* right channel */
#ifdef LSB_FIRST
delta = ((ptr[0] * mul) / 1024) - r;
ptr++;
#else
delta = (((int16)((ptr[0] + ptr[1]*256)) * mul) / 1024) - r;
ptr += 2;
#endif
r += delta;
blip_add_delta_fast(blip[1], i, delta);
/* update CD-DA fader volume (one step/sample) */
if (curVol < endVol)
{
/* fade-in */
curVol++;
}
else if (curVol > endVol)
{
/* fade-out */
curVol--;
}
else if (!curVol)
{
/* audio will remain muted until next setup */
break;
}
}
}
/* save current CD-DA fader volume */
cdd.volume = curVol;
/* save last audio output for next frame */
cdd.audio[0] = l;
cdd.audio[1] = r;
}
void pcm_run(unsigned int length)
{
#ifdef LOG_PCM
error("[%d][%d]run %d PCM samples (from %d)\n", v_counter, s68k.cycles, length, pcm.cycles);
#endif
/* check if PCM chip is running */
if (pcm.enabled)
{
int i, j, l, r;
/* generate PCM samples */
for (i = 0; i != length; i++)
{
/* clear output */
l = r = 0;
/* run eight PCM channels */
for (j = 0; j != 8; j++)
{
/* check if channel is enabled */
if (pcm.status & (1 << j))
{
/* read from current WAVE RAM address */
short data = pcm.ram[(pcm.chan[j].addr >> 11) & 0xffff];
/* loop data ? */
if (data == 0xff)
{
/* reset WAVE RAM address */
pcm.chan[j].addr = pcm.chan[j].ls.w << 11;
/* read again from WAVE RAM address */
data = pcm.ram[pcm.chan[j].ls.w];
}
else
{
/* increment WAVE RAM address */
pcm.chan[j].addr += pcm.chan[j].fd.w;
}
/* infinite loop should not output any data */
if (data != 0xff)
{
/* check sign bit (output centered around 0) */
if (data & 0x80)
{
/* PCM data is positive */
data = data & 0x7f;
}
else
{
/* PCM data is negative */
data = -(data & 0x7f);
}
/* multiply PCM data with ENV & stereo PAN data then add to L/R outputs (14.5 fixed point) */
l += ((data * pcm.chan[j].env * (pcm.chan[j].pan & 0x0F)) >> 5);
r += ((data * pcm.chan[j].env * (pcm.chan[j].pan >> 4)) >> 5);
}
}
}
/* limiter */
if (l < -32768) l = -32768;
else if (l > 32767) l = 32767;
if (r < -32768) r = -32768;
else if (r > 32767) r = 32767;
/* check if PCM left output changed */
if (pcm.out[0] != l)
{
blip_add_delta_fast(snd.blips[1][0], i, l-pcm.out[0]);
pcm.out[0] = l;
}
/* check if PCM right output changed */
if (pcm.out[1] != r)
{
blip_add_delta_fast(snd.blips[1][1], i, r-pcm.out[1]);
pcm.out[1] = r;
}
}
int psg_context_load(uint8 *state)
{
int delta[2];
int i, bufferptr = 0;
/* initialize delta with current noise channel output */
if (psg.noiseShiftValue & 1)
{
delta[0] = -psg.chanOut[3][0];
delta[1] = -psg.chanOut[3][1];
}
else
{
delta[0] = 0;
delta[1] = 0;
}
/* add current tone channels output */
for (i=0; i<3; i++)
{
if (psg.polarity[i] > 0)
{
delta[0] -= psg.chanOut[i][0];
delta[1] -= psg.chanOut[i][1];
}
}
load_param(&psg.clocks,sizeof(psg.clocks));
load_param(&psg.latch,sizeof(psg.latch));
load_param(&psg.noiseShiftValue,sizeof(psg.noiseShiftValue));
load_param(psg.regs,sizeof(psg.regs));
load_param(psg.freqInc,sizeof(psg.freqInc));
load_param(psg.freqCounter,sizeof(psg.freqCounter));
load_param(psg.polarity,sizeof(psg.polarity));
load_param(psg.chanOut,sizeof(psg.chanOut));
/* add noise channel output variation */
if (psg.noiseShiftValue & 1)
{
delta[0] += psg.chanOut[3][0];
delta[1] += psg.chanOut[3][1];
}
/* add tone channels output variation */
for (i=0; i<3; i++)
{
if (psg.polarity[i] > 0)
{
delta[0] += psg.chanOut[i][0];
delta[1] += psg.chanOut[i][1];
}
}
/* update mixed channels output */
if (config.hq_psg)
{
blip_add_delta(snd.blips[0], psg.clocks, delta[0], delta[1]);
}
else
{
blip_add_delta_fast(snd.blips[0], psg.clocks, delta[0], delta[1]);
}
return bufferptr;
}
