/* * PicoDrive * (c) Copyright Dave, 2004 * (C) notaz, 2006-2009 * * This work is licensed under the terms of MAME license. * See COPYING file in the top-level directory. */ #include #include "../pico_int.h" #include "ym2612.h" #include "sn76496.h" #include "emu2413/emu2413.h" #include "resampler.h" #include "mix.h" void (*PsndMix_32_to_16l)(s16 *dest, s32 *src, int count) = mix_32_to_16l_stereo; // master int buffer to mix to // +1 for a fill triggered by an instruction overhanging into the next scanline static s32 PsndBuffer[2*(54000+100)/50+2]; // cdda output buffer s16 cdda_out_buffer[2*1152]; // sn76496 extern int *sn76496_regs; // FM resampling polyphase FIR static resampler_t *fmresampler; static int (*PsndFMUpdate)(s32 *buffer, int length, int stereo, int is_buf_empty); // ym2413 static OPLL *opll = NULL; static OPLL old_opll; static struct { uint32_t adr; uint8_t reg[sizeof(opll->reg)]; } opll_buf; void YM2413_regWrite(unsigned data){ OPLL_writeIO(opll,0,data); } void YM2413_dataWrite(unsigned data){ OPLL_writeIO(opll,1,data); } PICO_INTERNAL void *YM2413GetRegs(void) { memcpy(opll_buf.reg, opll->reg, sizeof(opll->reg)); opll_buf.adr = opll->adr; return &opll_buf; } PICO_INTERNAL void YM2413UnpackState(void) { int i; for (i = sizeof(opll->reg)-1; i >= 0; i--) { OPLL_writeIO(opll, 0, i); OPLL_writeIO(opll, 1, opll_buf.reg[i]); } opll->adr = opll_buf.adr; } PICO_INTERNAL void PsndInit(void) { opll = OPLL_new(OSC_NTSC/15, OSC_NTSC/15/72); OPLL_setChipType(opll,0); OPLL_reset(opll); } PICO_INTERNAL void PsndExit(void) { OPLL_delete(opll); opll = NULL; resampler_free(fmresampler); fmresampler = NULL; } PICO_INTERNAL void PsndReset(void) { // PsndRerate calls YM2612Init, which also resets PsndRerate(0); timers_reset(); } // FM polyphase FIR resampling #define FMFIR_TAPS 9 // resample FM from its native 53267Hz/52781Hz with polyphase FIR filter static int ymchans; static void YM2612Update(s32 *buffer, int length, int stereo) { ymchans = YM2612UpdateOne(buffer, length, stereo, 1); } static int YM2612UpdateFIR(s32 *buffer, int length, int stereo, int is_buf_empty) { resampler_update(fmresampler, buffer, length, YM2612Update); return ymchans; } // resample SMS FM from its native 49716Hz/49262Hz with polyphase FIR filter static void YM2413Update(s32 *buffer, int length, int stereo) { while (length-- > 0) { int16_t getdata = OPLL_calc(opll) * 3; *buffer++ = getdata; buffer += stereo; // only left for stereo, to be mixed to right later } } static int YM2413UpdateFIR(s32 *buffer, int length, int stereo, int is_buf_empty) { if (!is_buf_empty) memset(buffer, 0, (length << stereo) * sizeof(*buffer)); resampler_update(fmresampler, buffer, length, YM2413Update); return 0; } // FIR setup, looks for a close enough rational number matching the ratio static void YMFM_setup_FIR(int inrate, int outrate, int stereo) { int mindiff = 999; int diff, mul, div; int minmult = 22, maxmult = 55; // min,max interpolation factor // compute filter ratio with largest multiplier for smallest error for (mul = minmult; mul <= maxmult; mul++) { div = (inrate*mul + outrate/2) / outrate; diff = outrate*div/mul - inrate; if (abs(diff) < abs(mindiff)) { mindiff = diff; Pico.snd.fm_fir_mul = mul; Pico.snd.fm_fir_div = div; if (abs(mindiff) <= inrate/1000+1) break; // below error limit } } printf("FM polyphase FIR ratio=%d/%d error=%.3f%%\n", Pico.snd.fm_fir_mul, Pico.snd.fm_fir_div, 100.0*mindiff/inrate); resampler_free(fmresampler); fmresampler = resampler_new(FMFIR_TAPS, Pico.snd.fm_fir_mul, Pico.snd.fm_fir_div, 0.85, 2, 2*inrate/50, stereo); } // wrapper for the YM2612UpdateONE macro static int YM2612UpdateONE(s32 *buffer, int length, int stereo, int is_buf_empty) { return YM2612UpdateOne(buffer, length, stereo, is_buf_empty); } // to be called after changing sound rate or chips void PsndRerate(int preserve_state) { void *state = NULL; int target_fps = Pico.m.pal ? 50 : 60; int target_lines = Pico.m.pal ? 313 : 262; int sms_clock = Pico.m.pal ? OSC_PAL/15 : OSC_NTSC/15; int ym2612_clock = Pico.m.pal ? OSC_PAL/7 : OSC_NTSC/7; int ym2612_rate = YM2612_NATIVE_RATE(); int ym2413_rate = (sms_clock + 36) / 72; if (preserve_state) { state = malloc(0x204); if (state == NULL) return; ym2612_pack_state(); memcpy(state, YM2612GetRegs(), 0x204); if (opll != NULL) memcpy(&old_opll, opll, sizeof(OPLL)); // remember old state } if (PicoIn.AHW & PAHW_SMS) { OPLL_setRate(opll, ym2413_rate); OPLL_reset(opll); YMFM_setup_FIR(ym2413_rate, PicoIn.sndRate, 0); PsndFMUpdate = YM2413UpdateFIR; } else if ((PicoIn.opt & POPT_EN_FM_FILTER) && ym2612_rate != PicoIn.sndRate) { // polyphase FIR resampler, resampling directly from native to output rate YM2612Init(ym2612_clock, ym2612_rate, ((PicoIn.opt&POPT_DIS_FM_SSGEG) ? 0 : ST_SSG) | ((PicoIn.opt&POPT_EN_FM_DAC) ? ST_DAC : 0)); YMFM_setup_FIR(ym2612_rate, PicoIn.sndRate, PicoIn.opt & POPT_EN_STEREO); PsndFMUpdate = YM2612UpdateFIR; } else { YM2612Init(ym2612_clock, PicoIn.sndRate, ((PicoIn.opt&POPT_DIS_FM_SSGEG) ? 0 : ST_SSG) | ((PicoIn.opt&POPT_EN_FM_DAC) ? ST_DAC : 0)); PsndFMUpdate = YM2612UpdateONE; } if (preserve_state) { // feed it back it's own registers, just like after loading state memcpy(YM2612GetRegs(), state, 0x204); ym2612_unpack_state(); if (opll != NULL) { memcpy(&opll->adr, &old_opll.adr, sizeof(OPLL)-20); // restore old state OPLL_forceRefresh(opll); } } if (preserve_state) memcpy(state, sn76496_regs, 28*4); // remember old state SN76496_init(Pico.m.pal ? OSC_PAL/15 : OSC_NTSC/15, PicoIn.sndRate); if (preserve_state) memcpy(sn76496_regs, state, 28*4); // restore old state if (state) free(state); // calculate Pico.snd.len Pico.snd.len = PicoIn.sndRate / target_fps; Pico.snd.len_e_add = ((PicoIn.sndRate - Pico.snd.len * target_fps) << 16) / target_fps; Pico.snd.len_e_cnt = 0; // Q16 // samples per line (Q16) Pico.snd.smpl_mult = 65536LL * PicoIn.sndRate / (target_fps*target_lines); // samples per z80 clock (Q20) Pico.snd.clkl_mult = 16 * Pico.snd.smpl_mult * 15/7 / 488.5; // samples per 44.1 KHz sample Pico.snd.cdda_mult = 65536LL * 44100 / PicoIn.sndRate; Pico.snd.cdda_div = 65536LL * PicoIn.sndRate / 44100; // clear all buffers memset32(PsndBuffer, 0, sizeof(PsndBuffer)/4); memset(cdda_out_buffer, 0, sizeof(cdda_out_buffer)); if (PicoIn.sndOut) PsndClear(); // set mixer PsndMix_32_to_16l = (PicoIn.opt & POPT_EN_STEREO) ? mix_32_to_16l_stereo : mix_32_to_16_mono; mix_reset(PicoIn.opt & POPT_EN_SNDFILTER ? PicoIn.sndFilterAlpha : 0); if (PicoIn.AHW & PAHW_PICO) PicoReratePico(); } PICO_INTERNAL void PsndStartFrame(void) { // compensate for float part of Pico.snd.len Pico.snd.len_use = Pico.snd.len; Pico.snd.len_e_cnt += Pico.snd.len_e_add; if (Pico.snd.len_e_cnt >= 0x10000) { Pico.snd.len_e_cnt -= 0x10000; Pico.snd.len_use++; } } PICO_INTERNAL void PsndDoDAC(int cyc_to) { int pos, len; int dout = ym2612.dacout; // nothing to do if sound is off if (!PicoIn.sndOut) return; // number of samples to fill in buffer (Q20) len = (cyc_to * Pico.snd.clkl_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; // 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) { s16 *d = PicoIn.sndOut + pos*2; // 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 { s16 *d = PicoIn.sndOut + pos; *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 cyc_to) { int pos, len; int stereo = 0; // nothing to do if sound is off if (!PicoIn.sndOut) return; // number of samples to fill in buffer (Q20) len = (cyc_to * Pico.snd.clkl_mult) - Pico.snd.psg_pos; // update position and calculate buffer offset and length pos = (Pico.snd.psg_pos+0x80000) >> 20; Pico.snd.psg_pos += len; len = ((Pico.snd.psg_pos+0x80000) >> 20) - pos; if (len <= 0) return; if (!(PicoIn.opt & POPT_EN_PSG)) return; if (PicoIn.opt & POPT_EN_STEREO) { stereo = 1; pos <<= 1; } SN76496Update(PicoIn.sndOut + pos, len, stereo); } PICO_INTERNAL void PsndDoSMSFM(int cyc_to) { int pos, len; int stereo = 0; s32 *buf32 = PsndBuffer; s16 *buf = PicoIn.sndOut; // nothing to do if sound is off if (!PicoIn.sndOut) return; // number of samples to fill in buffer (Q20) len = (cyc_to * Pico.snd.clkl_mult) - Pico.snd.ym2413_pos; // update position and calculate buffer offset and length pos = (Pico.snd.ym2413_pos+0x80000) >> 20; Pico.snd.ym2413_pos += len; len = ((Pico.snd.ym2413_pos+0x80000) >> 20) - pos; if (len <= 0) return; if (!(PicoIn.opt & POPT_EN_YM2413)) return; if (PicoIn.opt & POPT_EN_STEREO) { stereo = 1; pos <<= 1; } if (Pico.m.hardware & PMS_HW_FMUSED) { buf += pos; PsndFMUpdate(buf32, len, 0, 0); while (len--) { *buf++ += *buf32++; buf += stereo; } } } PICO_INTERNAL void PsndDoFM(int cyc_to) { int pos, len; int stereo = 0; // nothing to do if sound is off if (!PicoIn.sndOut) return; // Q20, number of samples since last call len = (cyc_to * Pico.snd.clkl_mult) - Pico.snd.fm_pos; // update position and calculate buffer offset and length pos = (Pico.snd.fm_pos+0x80000) >> 20; Pico.snd.fm_pos += len; len = ((Pico.snd.fm_pos+0x80000) >> 20) - pos; if (len <= 0) return; // fill buffer if (PicoIn.opt & POPT_EN_STEREO) { stereo = 1; pos <<= 1; } if (PicoIn.opt & POPT_EN_FM) PsndFMUpdate(PsndBuffer + pos, len, stereo, 1); } PICO_INTERNAL void PsndDoPCM(int cyc_to) { int pos, len; int stereo = 0; // nothing to do if sound is off if (!PicoIn.sndOut) return; // Q20, number of samples since last call len = (cyc_to * Pico.snd.clkl_mult) - Pico.snd.pcm_pos; // update position and calculate buffer offset and length pos = (Pico.snd.pcm_pos+0x80000) >> 20; Pico.snd.pcm_pos += len; len = ((Pico.snd.pcm_pos+0x80000) >> 20) - pos; if (len <= 0) return; // fill buffer if (PicoIn.opt & POPT_EN_STEREO) { stereo = 1; pos <<= 1; } PicoPicoPCMUpdate(PicoIn.sndOut + pos, len, stereo); } // cdda static void cdda_raw_update(s32 *buffer, int length, int stereo) { int ret, cdda_bytes; cdda_bytes = (length * Pico.snd.cdda_mult >> 16) * 4; ret = pm_read_audio(cdda_out_buffer, cdda_bytes, Pico_mcd->cdda_stream); if (ret < cdda_bytes) { memset((char *)cdda_out_buffer + ret, 0, cdda_bytes - ret); Pico_mcd->cdda_stream = NULL; return; } // now mix if (stereo) switch (Pico.snd.cdda_mult) { case 0x10000: mix_16h_to_32(buffer, cdda_out_buffer, length*2); break; case 0x20000: mix_16h_to_32_s1(buffer, cdda_out_buffer, length*2); break; case 0x40000: mix_16h_to_32_s2(buffer, cdda_out_buffer, length*2); break; default: mix_16h_to_32_resample_stereo(buffer, cdda_out_buffer, length, Pico.snd.cdda_mult); } else mix_16h_to_32_resample_mono(buffer, cdda_out_buffer, length, Pico.snd.cdda_mult); } void cdda_start_play(int lba_base, int lba_offset, int lb_len) { if (Pico_mcd->cdda_type == CT_MP3) { int pos1024 = 0; if (lba_offset) pos1024 = lba_offset * 1024 / lb_len; mp3_start_play(Pico_mcd->cdda_stream, pos1024); return; } pm_seek(Pico_mcd->cdda_stream, (lba_base + lba_offset) * 2352, SEEK_SET); if (Pico_mcd->cdda_type == CT_WAV) { // skip headers, assume it's 44kHz stereo uncompressed pm_seek(Pico_mcd->cdda_stream, 44, SEEK_CUR); } } PICO_INTERNAL void PsndClear(void) { int len = Pico.snd.len; if (Pico.snd.len_e_add) len++; // drop pos remainder to avoid rounding errors (not entirely correct though) Pico.snd.dac_pos = Pico.snd.fm_pos = Pico.snd.psg_pos = Pico.snd.ym2413_pos = Pico.snd.pcm_pos = 0; if (!PicoIn.sndOut) return; if (PicoIn.opt & POPT_EN_STEREO) memset32((int *) PicoIn.sndOut, 0, len); // assume PicoIn.sndOut to be aligned else { s16 *out = PicoIn.sndOut; if ((uintptr_t)out & 2) { *out++ = 0; len--; } memset32((int *) out, 0, len/2); if (len & 1) out[len-1] = 0; } if (!(PicoIn.opt & POPT_EN_FM)) memset32(PsndBuffer, 0, PicoIn.opt & POPT_EN_STEREO ? len*2 : len); } static int PsndRender(int offset, int length) { s32 *buf32; int stereo = (PicoIn.opt & 8) >> 3; int fmlen = ((Pico.snd.fm_pos+0x80000) >> 20); int daclen = ((Pico.snd.dac_pos+0x80000) >> 20); int psglen = ((Pico.snd.psg_pos+0x80000) >> 20); int pcmlen = ((Pico.snd.pcm_pos+0x80000) >> 20); buf32 = PsndBuffer+(offset< 0 && PicoIn.sndOut) { s16 *psgbuf = PicoIn.sndOut + (psglen << stereo); Pico.snd.psg_pos += (length-psglen) << 20; if (PicoIn.opt & POPT_EN_PSG) SN76496Update(psgbuf, length-psglen, stereo); } if (PicoIn.AHW & PAHW_PICO) { // always need to render sound for interrupts s16 *buf16 = PicoIn.sndOut ? PicoIn.sndOut + (pcmlen< 0 && PicoIn.sndOut) { s16 *dacbuf = PicoIn.sndOut + (daclen << stereo); Pico.snd.dac_pos += (length-daclen) << 20; *dacbuf++ += Pico.snd.dac_val2; if (stereo) dacbuf++; for (daclen++; length-daclen > 0; daclen++) { *dacbuf++ += Pico.snd.dac_val; if (stereo) dacbuf++; } Pico.snd.dac_val2 = Pico.snd.dac_val; } // Add in parts of the FM buffer not yet done if (length-fmlen > 0 && PicoIn.sndOut) { s32 *fmbuf = buf32 + ((fmlen-offset) << stereo); Pico.snd.fm_pos += (length-fmlen) << 20; if (PicoIn.opt & POPT_EN_FM) PsndFMUpdate(fmbuf, length-fmlen, stereo, 1); } // CD: PCM sound if (PicoIn.AHW & PAHW_MCD) { pcd_pcm_update(buf32, length-offset, stereo); } // CD: CDDA audio // CD mode, cdda enabled, not data track, CDC is reading if ((PicoIn.AHW & PAHW_MCD) && (PicoIn.opt & POPT_EN_MCD_CDDA) && Pico_mcd->cdda_stream != NULL && !(Pico_mcd->s68k_regs[0x36] & 1)) { if (Pico_mcd->cdda_type == CT_MP3) mp3_update(buf32, length-offset, stereo); else cdda_raw_update(buf32, length-offset, stereo); } if ((PicoIn.AHW & PAHW_32X) && (PicoIn.opt & POPT_EN_PWM)) p32x_pwm_update(buf32, length-offset, stereo); // convert + limit to normal 16bit output if (PicoIn.sndOut) PsndMix_32_to_16l(PicoIn.sndOut+(offset<> 3; int psglen = ((Pico.snd.psg_pos+0x80000) >> 20); int ym2413len = ((Pico.snd.ym2413_pos+0x80000) >> 20); if (!PicoIn.sndOut) return length; pprof_start(sound); // Add in parts of the PSG output not yet done if (length-psglen > 0) { s16 *psgbuf = PicoIn.sndOut + (psglen << stereo); Pico.snd.psg_pos += (length-psglen) << 20; if (PicoIn.opt & POPT_EN_PSG) SN76496Update(psgbuf, length-psglen, stereo); } if (length-ym2413len > 0) { s16 *ym2413buf = PicoIn.sndOut + (ym2413len << stereo); Pico.snd.ym2413_pos += (length-ym2413len) << 20; int len = (length-ym2413len); if (Pico.m.hardware & PMS_HW_FMUSED) { PsndFMUpdate(buf32, len, 0, 0); while (len--) { *ym2413buf++ += *buf32++; ym2413buf += stereo; } } } // upmix to "stereo" if needed if (PicoIn.opt & POPT_EN_STEREO) { int i; s16 *p; for (i = length, p = (s16 *)PicoIn.sndOut; i > 0; i--, p+=2) *(p + 1) = *p; } pprof_end(sound); return length; } PICO_INTERNAL void PsndGetSamplesMS(int y) { static int curr_pos = 0; curr_pos = PsndRenderMS(0, Pico.snd.len_use); if (PicoIn.writeSound != NULL && PicoIn.sndOut) PicoIn.writeSound(curr_pos * ((PicoIn.opt & POPT_EN_STEREO) ? 4 : 2)); PsndClear(); } // vim:shiftwidth=2:ts=2:expandtab