emulator timing fixes, VDP DMA fixes, improved DAC audio

This commit is contained in:
kub 2020-01-14 23:00:44 +01:00
parent b9bc876c9c
commit 43e1401008
10 changed files with 118 additions and 98 deletions

View file

@ -89,6 +89,8 @@ void PsndRerate(int preserve_state)
// samples per line (Q16)
Pico.snd.fm_mult = 65536LL * PicoIn.sndRate / (target_fps*target_lines);
// samples per z80 clock (Q20)
Pico.snd.dac_mult = 16 * Pico.snd.fm_mult * 15/7 / 488;
// recalculate dac info
dac_recalculate();
@ -117,34 +119,46 @@ PICO_INTERNAL void PsndStartFrame(void)
Pico.snd.len_use++;
}
Pico.snd.dac_line = Pico.snd.psg_line = 0;
Pico.snd.fm_pos = 0;
Pico.snd.psg_line = 0;
}
PICO_INTERNAL void PsndDoDAC(int line_to)
PICO_INTERNAL void PsndDoDAC(int cyc_to)
{
int pos, pos1, len;
int pos, len;
int dout = ym2612.dacout;
int line_from = Pico.snd.dac_line;
pos = dac_info[line_from];
pos1 = dac_info[line_to + 1];
len = pos1 - pos;
// number of samples to fill in buffer (Q20)
len = (cyc_to * Pico.snd.dac_mult) - Pico.snd.dac_pos;
// update position and calculate buffer offset and length
pos = (Pico.snd.dac_pos+0x80000) >> 20;
Pico.snd.dac_pos += len;
len = ((Pico.snd.dac_pos+0x80000) >> 20) - pos;
// avoid loss of the 1st sample of a new block (Q rounding issues)
if (pos+len == 0)
len = 1, Pico.snd.dac_pos += 0x80000;
if (len <= 0)
return;
Pico.snd.dac_line = line_to + 1;
if (!PicoIn.sndOut)
return;
// fill buffer, applying a rather weak order 1 bessel IIR on the way
// y[n] = (x[n] + x[n-1])*(1/2) (3dB cutoff at 11025 Hz, no gain)
// 1 sample delay for correct IIR filtering over audio frame boundaries
if (PicoIn.opt & POPT_EN_STEREO) {
short *d = PicoIn.sndOut + pos*2;
for (; len > 0; len--, d+=2) *d += dout;
// left channel only, mixed ro right channel in mixing phase
*d++ += Pico.snd.dac_val2; d++;
while (--len) *d++ += Pico.snd.dac_val, d++;
} else {
short *d = PicoIn.sndOut + pos;
for (; len > 0; len--, d++) *d += dout;
*d++ += Pico.snd.dac_val2;
while (--len) *d++ += Pico.snd.dac_val;
}
Pico.snd.dac_val2 = (Pico.snd.dac_val + dout) >> 1;
Pico.snd.dac_val = dout;
}
PICO_INTERNAL void PsndDoPSG(int line_to)
@ -258,6 +272,8 @@ PICO_INTERNAL void PsndClear(void)
}
if (!(PicoIn.opt & POPT_EN_FM))
memset32(PsndBuffer, 0, PicoIn.opt & POPT_EN_STEREO ? len*2 : len);
// drop pos remainder to avoid rounding errors (not entirely correct though)
Pico.snd.dac_pos = Pico.snd.fm_pos = 0;
}
@ -266,6 +282,7 @@ static int PsndRender(int offset, int length)
int *buf32;
int stereo = (PicoIn.opt & 8) >> 3;
int fmlen = ((Pico.snd.fm_pos+0x8000) >> 16) - offset;
int daclen = ((Pico.snd.dac_pos+0x80000) >> 20) - offset;
offset <<= stereo;
buf32 = PsndBuffer+offset;
@ -277,6 +294,15 @@ static int PsndRender(int offset, int length)
return length;
}
// Fill up DAC output in case of missing samples (Q16 rounding errors)
if (length-daclen > 0) {
short *dacbuf = PicoIn.sndOut + (daclen << stereo);
for (; length-daclen > 0; daclen++) {
*dacbuf++ += Pico.snd.dac_val;
if (stereo) dacbuf++;
}
}
// Add in parts of the FM buffer not yet done
if (length-fmlen > 0) {
int *fmbuf = buf32 + (fmlen << stereo);
@ -317,8 +343,8 @@ PICO_INTERNAL void PsndGetSamples(int y)
{
static int curr_pos = 0;
if (ym2612.dacen && Pico.snd.dac_line < y)
PsndDoDAC(y - 1);
if (ym2612.dacen)
PsndDoDAC(cycles_68k_to_z80(Pico.t.m68c_aim - Pico.t.m68c_frame_start));
PsndDoPSG(y - 1);
curr_pos = PsndRender(0, Pico.snd.len_use);
@ -327,7 +353,6 @@ PICO_INTERNAL void PsndGetSamples(int y)
PicoIn.writeSound(curr_pos * ((PicoIn.opt & POPT_EN_STEREO) ? 4 : 2));
// clear sound buffer
PsndClear();
Pico.snd.dac_line = y;
}
PICO_INTERNAL void PsndGetSamplesMS(int y)