[picodrive.git] / pico / sound / sound.c

/*\r
 * PicoDrive\r
 * (c) Copyright Dave, 2004\r
 * (C) notaz, 2006-2009\r
 * (C) irixxxx, 2019-2024\r
 *\r
 * This work is licensed under the terms of MAME license.\r
 * See COPYING file in the top-level directory.\r
 */\r
\r
#include <string.h>\r
#include "../pico_int.h"\r
#include "ym2612.h"\r
#include "sn76496.h"\r
#include "emu2413/emu2413.h"\r
#include "resampler.h"\r
#include "mix.h"\r
\r
#define YM2612_CH6PAN   0x1b6   // panning register for channel 6 (used for DAC)\r
\r
void (*PsndMix_32_to_16)(s16 *dest, s32 *src, int count) = mix_32_to_16_stereo;\r
\r
// master int buffer to mix to\r
// +1 for a fill triggered by an instruction overhanging into the next scanline\r
static s32 PsndBuffer[2*(54000+100)/50+2];\r
\r
// cdda output buffer\r
s16 cdda_out_buffer[2*1152];\r
\r
// sn76496\r
extern int *sn76496_regs;\r
\r
// FM resampling polyphase FIR\r
static resampler_t *fmresampler;\r
static int (*PsndFMUpdate)(s32 *buffer, int length, int stereo, int is_buf_empty);\r
\r
// ym2413\r
static OPLL *opll = NULL;\r
static struct {\r
  uint32_t adr;\r
  uint8_t reg[sizeof(opll->reg)];\r
} opll_buf;\r
\r
\r
void YM2413_regWrite(unsigned data){\r
  OPLL_writeIO(opll,0,data);\r
}\r
void YM2413_dataWrite(unsigned data){\r
  OPLL_writeIO(opll,1,data);\r
}\r
\r
PICO_INTERNAL void *YM2413GetRegs(void)\r
{\r
  memcpy(opll_buf.reg, opll->reg, sizeof(opll->reg));\r
  opll_buf.adr = opll->adr;\r
  return &opll_buf;\r
}\r
\r
PICO_INTERNAL void YM2413UnpackState(void)\r
{\r
  int i;\r
\r
  for (i = sizeof(opll->reg)-1; i >= 0; i--) {\r
    OPLL_writeIO(opll, 0, i);\r
    OPLL_writeIO(opll, 1, opll_buf.reg[i]);\r
  }\r
  opll->adr = opll_buf.adr;\r
}\r
\r
PICO_INTERNAL void PsndInit(void)\r
{\r
  opll = OPLL_new(OSC_NTSC/15, OSC_NTSC/15/72);\r
  OPLL_setChipType(opll,0);\r
  OPLL_reset(opll);\r
}\r
\r
PICO_INTERNAL void PsndExit(void)\r
{\r
  OPLL_delete(opll);\r
  opll = NULL;\r
\r
  resampler_free(fmresampler); fmresampler = NULL;\r
}\r
\r
PICO_INTERNAL void PsndReset(void)\r
{\r
  // PsndRerate calls YM2612Init, which also resets\r
  PsndRerate(0);\r
  timers_reset();\r
}\r
\r
// FM polyphase FIR resampling\r
#define FMFIR_TAPS	9\r
\r
// resample FM from its native 53267Hz/52781Hz with polyphase FIR filter\r
static int ymchans;\r
static void YM2612Update(s32 *buffer, int length, int stereo)\r
{\r
  ymchans = YM2612UpdateOne(buffer, length, stereo, 1);\r
}\r
\r
static int YM2612UpdateFIR(s32 *buffer, int length, int stereo, int is_buf_empty)\r
{\r
  resampler_update(fmresampler, buffer, length, YM2612Update);\r
  return ymchans;\r
}\r
\r
// resample SMS FM from its native 49716Hz/49262Hz with polyphase FIR filter\r
static void YM2413Update(s32 *buffer, int length, int stereo)\r
{\r
  while (length-- > 0) {\r
    int16_t getdata = OPLL_calc(opll) * 3;\r
    *buffer++ = getdata;\r
    buffer += stereo; // only left for stereo, to be mixed to right later\r
  }\r
}\r
\r
static int YM2413UpdateFIR(s32 *buffer, int length, int stereo, int is_buf_empty)\r
{\r
  if (!is_buf_empty) memset(buffer, 0, (length << stereo) * sizeof(*buffer));\r
  resampler_update(fmresampler, buffer, length, YM2413Update);\r
  return 0;\r
}\r
\r
// FIR setup, looks for a close enough rational number matching the ratio\r
static void YMFM_setup_FIR(int inrate, int outrate, int stereo)\r
{\r
  int mindiff = 999;\r
  int diff, mul, div;\r
  int minmult = 22, maxmult = 55; // min,max interpolation factor\r
\r
  // compute filter ratio with largest multiplier for smallest error\r
  for (mul = minmult; mul <= maxmult; mul++) {\r
    div = (inrate*mul + outrate/2) / outrate;\r
    diff = outrate*div/mul - inrate;\r
    if (abs(diff) < abs(mindiff)) {\r
      mindiff = diff;\r
      Pico.snd.fm_fir_mul = mul;\r
      Pico.snd.fm_fir_div = div;\r
      if (abs(mindiff) <= inrate/1000+1) break; // below error limit\r
    }\r
  }\r
  printf("FM polyphase FIR ratio=%d/%d error=%.3f%%\n",\r
        Pico.snd.fm_fir_mul, Pico.snd.fm_fir_div, 100.0*mindiff/inrate);\r
\r
  resampler_free(fmresampler);\r
  fmresampler = resampler_new(FMFIR_TAPS, Pico.snd.fm_fir_mul, Pico.snd.fm_fir_div,\r
        0.85, 2, 2*inrate/50, stereo);\r
}\r
\r
// wrapper for the YM2612UpdateONE macro\r
static int YM2612UpdateONE(s32 *buffer, int length, int stereo, int is_buf_empty)\r
{\r
  return YM2612UpdateOne(buffer, length, stereo, is_buf_empty);\r
}\r
\r
static int ymclock;\r
static int ymrate;\r
static int ymopts;\r
\r
// to be called after changing sound rate or chips\r
void PsndRerate(int preserve_state)\r
{\r
  void *state = NULL;\r
  int target_fps = Pico.m.pal ? 50 : 60;\r
  int target_lines = Pico.m.pal ? 313 : 262;\r
  int sms_clock = Pico.m.pal ? OSC_PAL/15 : OSC_NTSC/15;\r
  int ym2413_rate = (sms_clock + 36) / 72;\r
  int ym2612_clock = Pico.m.pal ? OSC_PAL/7 : OSC_NTSC/7;\r
  int ym2612_rate = YM2612_NATIVE_RATE();\r
  int ym2612_init = !preserve_state;\r
\r
  // don't init YM2612 if preserve_state and no parameter changes\r
  ym2612_init |= ymclock != ym2612_clock || ymopts != (PicoIn.opt & (POPT_DIS_FM_SSGEG|POPT_EN_FM_DAC));\r
  ym2612_init |= ymrate != (PicoIn.opt & POPT_EN_FM_FILTER ? ym2612_rate : PicoIn.sndRate);\r
  ymclock = ym2612_clock;\r
  ymrate = (PicoIn.opt & POPT_EN_FM_FILTER ? ym2612_rate : PicoIn.sndRate);\r
  ymopts = PicoIn.opt & (POPT_DIS_FM_SSGEG|POPT_EN_FM_DAC);\r
\r
  if (preserve_state && ym2612_init) {\r
    state = malloc(0x204);\r
    if (state == NULL) return;\r
    ym2612_pack_state();\r
    memcpy(state, YM2612GetRegs(), 0x204);\r
  }\r
\r
  if (PicoIn.AHW & PAHW_SMS) {\r
    OPLL_setRate(opll, ym2413_rate);\r
    if (!preserve_state)\r
      OPLL_reset(opll);\r
    YMFM_setup_FIR(ym2413_rate, PicoIn.sndRate, 0);\r
    PsndFMUpdate = YM2413UpdateFIR;\r
  } else if ((PicoIn.opt & POPT_EN_FM_FILTER) && ym2612_rate != PicoIn.sndRate) {\r
    // polyphase FIR resampler, resampling directly from native to output rate\r
    if (ym2612_init)\r
      YM2612Init(ym2612_clock, ym2612_rate,\r
        ((PicoIn.opt&POPT_DIS_FM_SSGEG) ? 0 : ST_SSG) |\r
        ((PicoIn.opt&POPT_EN_FM_DAC)    ? ST_DAC : 0));\r
    YMFM_setup_FIR(ym2612_rate, PicoIn.sndRate, PicoIn.opt & POPT_EN_STEREO);\r
    PsndFMUpdate = YM2612UpdateFIR;\r
  } else {\r
    if (ym2612_init)\r
      YM2612Init(ym2612_clock, PicoIn.sndRate,\r
        ((PicoIn.opt&POPT_DIS_FM_SSGEG) ? 0 : ST_SSG) |\r
        ((PicoIn.opt&POPT_EN_FM_DAC)    ? ST_DAC : 0));\r
    PsndFMUpdate = YM2612UpdateONE;\r
  }\r
\r
  if (preserve_state && ym2612_init) {\r
    // feed it back it's own registers, just like after loading state\r
    memcpy(YM2612GetRegs(), state, 0x204);\r
    ym2612_unpack_state();\r
    free(state);\r
  }\r
\r
  if (preserve_state)\r
    SN76496_set_clockrate(Pico.m.pal ? OSC_PAL/15 : OSC_NTSC/15, PicoIn.sndRate);\r
  else\r
    SN76496_init(Pico.m.pal ? OSC_PAL/15 : OSC_NTSC/15, PicoIn.sndRate);\r
\r
  // calculate Pico.snd.len\r
  Pico.snd.len = PicoIn.sndRate / target_fps;\r
  Pico.snd.len_e_add = ((PicoIn.sndRate - Pico.snd.len * target_fps) << 16) / target_fps;\r
  Pico.snd.len_e_cnt = 0; // Q16\r
\r
  // samples per line (Q16)\r
  Pico.snd.smpl_mult = 65536LL * PicoIn.sndRate / (target_fps*target_lines);\r
  // samples per z80 clock (Q20)\r
  Pico.snd.clkl_mult = 16 * Pico.snd.smpl_mult * 15/7 / 488.5;\r
  // samples per 44.1 KHz sample\r
  Pico.snd.cdda_mult = 65536LL * 44100 / PicoIn.sndRate;\r
  Pico.snd.cdda_div  = 65536LL * PicoIn.sndRate / 44100;\r
\r
  // clear all buffers\r
  memset32(PsndBuffer, 0, sizeof(PsndBuffer)/4);\r
  memset(cdda_out_buffer, 0, sizeof(cdda_out_buffer));\r
  if (PicoIn.sndOut)\r
    PsndClear();\r
\r
  // set mixer\r
  PsndMix_32_to_16 = (PicoIn.opt & POPT_EN_STEREO) ? mix_32_to_16_stereo : mix_32_to_16_mono;\r
  mix_reset(PicoIn.opt & POPT_EN_SNDFILTER ? PicoIn.sndFilterAlpha : 0);\r
\r
  if (PicoIn.AHW & PAHW_PICO)\r
    PicoReratePico();\r
}\r
\r
\r
PICO_INTERNAL void PsndStartFrame(void)\r
{\r
  // compensate for float part of Pico.snd.len\r
  Pico.snd.len_use = Pico.snd.len;\r
  Pico.snd.len_e_cnt += Pico.snd.len_e_add;\r
  if (Pico.snd.len_e_cnt >= 0x10000) {\r
    Pico.snd.len_e_cnt -= 0x10000;\r
    Pico.snd.len_use++;\r
  }\r
}\r
\r
PICO_INTERNAL void PsndDoDAC(int cyc_to)\r
{\r
  int pos, len;\r
  int dout = ym2612.dacout;\r
\r
  // nothing to do if sound is off\r
  if (!PicoIn.sndOut) return;\r
\r
  // number of samples to fill in buffer (Q20)\r
  len = (cyc_to * Pico.snd.clkl_mult) - Pico.snd.dac_pos;\r
\r
  // update position and calculate buffer offset and length\r
  pos = (Pico.snd.dac_pos+0x80000) >> 20;\r
  Pico.snd.dac_pos += len;\r
  len = ((Pico.snd.dac_pos+0x80000) >> 20) - pos;\r
\r
  // avoid loss of the 1st sample of a new block (Q rounding issues)\r
  if (pos+len == 0)\r
    len = 1, Pico.snd.dac_pos += 0x80000;\r
  if (len <= 0)\r
    return;\r
\r
  // fill buffer, applying a rather weak order 1 bessel IIR on the way\r
  // y[n] = (x[n] + x[n-1])*(1/2) (3dB cutoff at 11025 Hz, no gain)\r
  // 1 sample delay for correct IIR filtering over audio frame boundaries\r
  if (PicoIn.opt & POPT_EN_STEREO) {\r
    s16 *d = PicoIn.sndOut + pos*2;\r
    int pan = ym2612.REGS[YM2612_CH6PAN];\r
    int l = pan & 0x80 ? Pico.snd.dac_val : 0;\r
    int r = pan & 0x40 ? Pico.snd.dac_val : 0;\r
    *d++ += pan & 0x80 ? Pico.snd.dac_val2 : 0;\r
    *d++ += pan & 0x40 ? Pico.snd.dac_val2 : 0;\r
    while (--len) *d++ += l, *d++ += r;\r
  } else {\r
    s16 *d = PicoIn.sndOut + pos;\r
    *d++ += Pico.snd.dac_val2;\r
    while (--len) *d++ += Pico.snd.dac_val;\r
  }\r
  Pico.snd.dac_val2 = (Pico.snd.dac_val + dout) >> 1;\r
  Pico.snd.dac_val = dout;\r
}\r
\r
PICO_INTERNAL void PsndDoPSG(int cyc_to)\r
{\r
  int pos, len;\r
  int stereo = 0;\r
\r
  // nothing to do if sound is off\r
  if (!PicoIn.sndOut) return;\r
\r
  // number of samples to fill in buffer (Q20)\r
  len = (cyc_to * Pico.snd.clkl_mult) - Pico.snd.psg_pos;\r
\r
  // update position and calculate buffer offset and length\r
  pos = (Pico.snd.psg_pos+0x80000) >> 20;\r
  Pico.snd.psg_pos += len;\r
  len = ((Pico.snd.psg_pos+0x80000) >> 20) - pos;\r
\r
  if (len <= 0)\r
    return;\r
  if (!(PicoIn.opt & POPT_EN_PSG))\r
    return;\r
\r
  if (PicoIn.opt & POPT_EN_STEREO) {\r
    stereo = 1;\r
    pos <<= 1;\r
  }\r
  SN76496Update(PicoIn.sndOut + pos, len, stereo);\r
}\r
\r
PICO_INTERNAL void PsndDoSMSFM(int cyc_to)\r
{\r
  int pos, len;\r
  int stereo = 0;\r
  s32 *buf32 = PsndBuffer;\r
  s16 *buf = PicoIn.sndOut;\r
\r
  // nothing to do if sound is off\r
  if (!PicoIn.sndOut) return;\r
\r
  // number of samples to fill in buffer (Q20)\r
  len = (cyc_to * Pico.snd.clkl_mult) - Pico.snd.ym2413_pos;\r
\r
  // update position and calculate buffer offset and length\r
  pos = (Pico.snd.ym2413_pos+0x80000) >> 20;\r
  Pico.snd.ym2413_pos += len;\r
  len = ((Pico.snd.ym2413_pos+0x80000) >> 20) - pos;\r
\r
  if (len <= 0)\r
    return;\r
  if (!(PicoIn.opt & POPT_EN_YM2413))\r
    return;\r
\r
  if (PicoIn.opt & POPT_EN_STEREO) {\r
    stereo = 1;\r
    pos <<= 1;\r
  }\r
\r
  if (Pico.m.hardware & PMS_HW_FMUSED) {\r
    buf += pos;\r
    PsndFMUpdate(buf32, len, 0, 0);\r
    if (stereo) \r
      while (len--) {\r
        *buf++ += *buf32;\r
        *buf++ += *buf32++;\r
      }\r
    else\r
      while (len--) {\r
        *buf++ += *buf32++;\r
      }\r
  }\r
}\r
\r
PICO_INTERNAL void PsndDoFM(int cyc_to)\r
{\r
  int pos, len;\r
  int stereo = 0;\r
\r
  // nothing to do if sound is off\r
  if (!PicoIn.sndOut) return;\r
\r
  // Q20, number of samples since last call\r
  len = (cyc_to * Pico.snd.clkl_mult) - Pico.snd.fm_pos;\r
\r
  // update position and calculate buffer offset and length\r
  pos = (Pico.snd.fm_pos+0x80000) >> 20;\r
  Pico.snd.fm_pos += len;\r
  len = ((Pico.snd.fm_pos+0x80000) >> 20) - pos;\r
  if (len <= 0)\r
    return;\r
\r
  // fill buffer\r
  if (PicoIn.opt & POPT_EN_STEREO) {\r
    stereo = 1;\r
    pos <<= 1;\r
  }\r
  if (PicoIn.opt & POPT_EN_FM)\r
    PsndFMUpdate(PsndBuffer + pos, len, stereo, 1);\r
}\r
\r
PICO_INTERNAL void PsndDoPCM(int cyc_to)\r
{\r
  int pos, len;\r
  int stereo = 0;\r
\r
  // nothing to do if sound is off\r
  if (!PicoIn.sndOut) return;\r
\r
  // Q20, number of samples since last call\r
  len = (cyc_to * Pico.snd.clkl_mult) - Pico.snd.pcm_pos;\r
\r
  // update position and calculate buffer offset and length\r
  pos = (Pico.snd.pcm_pos+0x80000) >> 20;\r
  Pico.snd.pcm_pos += len;\r
  len = ((Pico.snd.pcm_pos+0x80000) >> 20) - pos;\r
  if (len <= 0)\r
    return;\r
\r
  // fill buffer\r
  if (PicoIn.opt & POPT_EN_STEREO) {\r
    stereo = 1;\r
    pos <<= 1;\r
  }\r
  PicoPicoPCMUpdate(PicoIn.sndOut + pos, len, stereo);\r
}\r
\r
// cdda\r
static void cdda_raw_update(s32 *buffer, int length, int stereo)\r
{\r
  int ret, cdda_bytes;\r
\r
  cdda_bytes = (length * Pico.snd.cdda_mult >> 16) * 4;\r
\r
  ret = pm_read_audio(cdda_out_buffer, cdda_bytes, Pico_mcd->cdda_stream);\r
  if (ret < cdda_bytes) {\r
    memset((char *)cdda_out_buffer + ret, 0, cdda_bytes - ret);\r
    Pico_mcd->cdda_stream = NULL;\r
    return;\r
  }\r
\r
  // now mix\r
  if (stereo) switch (Pico.snd.cdda_mult) {\r
    case 0x10000: mix_16h_to_32(buffer, cdda_out_buffer, length*2);     break;\r
    case 0x20000: mix_16h_to_32_s1(buffer, cdda_out_buffer, length*2);  break;\r
    case 0x40000: mix_16h_to_32_s2(buffer, cdda_out_buffer, length*2);  break;\r
    default: mix_16h_to_32_resample_stereo(buffer, cdda_out_buffer, length, Pico.snd.cdda_mult);\r
  } else\r
    mix_16h_to_32_resample_mono(buffer, cdda_out_buffer, length, Pico.snd.cdda_mult);\r
}\r
\r
void cdda_start_play(int lba_base, int lba_offset, int lb_len)\r
{\r
  if (Pico_mcd->cdda_type == CT_MP3)\r
  {\r
    int pos1024 = 0;\r
\r
    if (lba_offset)\r
      pos1024 = lba_offset * 1024 / lb_len;\r
\r
    mp3_start_play(Pico_mcd->cdda_stream, pos1024);\r
    return;\r
  }\r
\r
  pm_seek(Pico_mcd->cdda_stream, (lba_base + lba_offset) * 2352, SEEK_SET);\r
  if (Pico_mcd->cdda_type == CT_WAV)\r
  {\r
    // skip headers, assume it's 44kHz stereo uncompressed\r
    pm_seek(Pico_mcd->cdda_stream, 44, SEEK_CUR);\r
  }\r
}\r
\r
\r
PICO_INTERNAL void PsndClear(void)\r
{\r
  int len = Pico.snd.len;\r
  if (Pico.snd.len_e_add) len++;\r
\r
  // drop pos remainder to avoid rounding errors (not entirely correct though)\r
  Pico.snd.dac_pos = Pico.snd.fm_pos = Pico.snd.psg_pos = Pico.snd.ym2413_pos = Pico.snd.pcm_pos = 0;\r
  if (!PicoIn.sndOut) return;\r
\r
  if (PicoIn.opt & POPT_EN_STEREO)\r
    memset32((int *) PicoIn.sndOut, 0, len); // assume PicoIn.sndOut to be aligned\r
  else {\r
    s16 *out = PicoIn.sndOut;\r
    if ((uintptr_t)out & 2) { *out++ = 0; len--; }\r
    memset32((int *) out, 0, len/2);\r
    if (len & 1) out[len-1] = 0;\r
  }\r
  if (!(PicoIn.opt & POPT_EN_FM))\r
    memset32(PsndBuffer, 0, PicoIn.opt & POPT_EN_STEREO ? len*2 : len);\r
}\r
\r
\r
static int PsndRender(int offset, int length)\r
{\r
  s32 *buf32;\r
  int stereo = (PicoIn.opt & 8) >> 3;\r
  int fmlen = ((Pico.snd.fm_pos+0x80000) >> 20);\r
  int daclen = ((Pico.snd.dac_pos+0x80000) >> 20);\r
  int psglen = ((Pico.snd.psg_pos+0x80000) >> 20);\r
  int pcmlen = ((Pico.snd.pcm_pos+0x80000) >> 20);\r
\r
  buf32 = PsndBuffer+(offset<<stereo);\r
\r
  pprof_start(sound);\r
\r
  // Add in parts of the PSG output not yet done\r
  if (length-psglen > 0 && PicoIn.sndOut) {\r
    s16 *psgbuf = PicoIn.sndOut + (psglen << stereo);\r
    Pico.snd.psg_pos += (length-psglen) << 20;\r
    if (PicoIn.opt & POPT_EN_PSG)\r
      SN76496Update(psgbuf, length-psglen, stereo);\r
  }\r
\r
  if (PicoIn.AHW & PAHW_PICO) {\r
    // always need to render sound for interrupts\r
    s16 *buf16 = PicoIn.sndOut ? PicoIn.sndOut + (pcmlen<<stereo) : NULL;\r
    PicoPicoPCMUpdate(buf16, length-pcmlen, stereo);\r
    return length;\r
  }\r
\r
  // Fill up DAC output in case of missing samples (Q rounding errors)\r
  if (length-daclen > 0 && PicoIn.sndOut) {\r
    Pico.snd.dac_pos += (length-daclen) << 20;\r
    if (PicoIn.opt & POPT_EN_STEREO) {\r
      s16 *d = PicoIn.sndOut + daclen*2;\r
      int pan = ym2612.REGS[YM2612_CH6PAN];\r
      int l = pan & 0x80 ? Pico.snd.dac_val : 0;\r
      int r = pan & 0x40 ? Pico.snd.dac_val : 0;\r
      *d++ += pan & 0x80 ? Pico.snd.dac_val2 : 0;\r
      *d++ += pan & 0x40 ? Pico.snd.dac_val2 : 0;\r
      if (l|r) for (daclen++; length-daclen > 0; daclen++)\r
          *d++ += l, *d++ += r;\r
    } else {\r
      s16 *d = PicoIn.sndOut + daclen;\r
      *d++ += Pico.snd.dac_val2;\r
      if (Pico.snd.dac_val) for (daclen++; length-daclen > 0; daclen++)\r
        *d++ += Pico.snd.dac_val;\r
    }\r
    Pico.snd.dac_val2 = Pico.snd.dac_val;\r
  }\r
\r
  // Add in parts of the FM buffer not yet done\r
  if (length-fmlen > 0 && PicoIn.sndOut) {\r
    s32 *fmbuf = buf32 + ((fmlen-offset) << stereo);\r
    Pico.snd.fm_pos += (length-fmlen) << 20;\r
    if (PicoIn.opt & POPT_EN_FM)\r
      PsndFMUpdate(fmbuf, length-fmlen, stereo, 1);\r
  }\r
\r
  // CD: PCM sound\r
  if (PicoIn.AHW & PAHW_MCD) {\r
    pcd_pcm_update(buf32, length-offset, stereo);\r
  }\r
\r
  // CD: CDDA audio\r
  // CD mode, cdda enabled, not data track, CDC is reading\r
  if ((PicoIn.AHW & PAHW_MCD) && (PicoIn.opt & POPT_EN_MCD_CDDA)\r
      && Pico_mcd->cdda_stream != NULL\r
      && !(Pico_mcd->s68k_regs[0x36] & 1))\r
  {\r
    if (Pico_mcd->cdda_type == CT_MP3)\r
      mp3_update(buf32, length-offset, stereo);\r
    else\r
      cdda_raw_update(buf32, length-offset, stereo);\r
  }\r
\r
  if ((PicoIn.AHW & PAHW_32X) && (PicoIn.opt & POPT_EN_PWM))\r
    p32x_pwm_update(buf32, length-offset, stereo);\r
\r
  // convert + limit to normal 16bit output\r
  if (PicoIn.sndOut)\r
    PsndMix_32_to_16(PicoIn.sndOut+(offset<<stereo), buf32, length-offset);\r
\r
  pprof_end(sound);\r
\r
  return length;\r
}\r
\r
PICO_INTERNAL void PsndGetSamples(int y)\r
{\r
  static int curr_pos = 0;\r
\r
  curr_pos  = PsndRender(0, Pico.snd.len_use);\r
\r
  if (PicoIn.writeSound && PicoIn.sndOut)\r
    PicoIn.writeSound(curr_pos * ((PicoIn.opt & POPT_EN_STEREO) ? 4 : 2));\r
  // clear sound buffer\r
  PsndClear();\r
}\r
\r
static int PsndRenderMS(int offset, int length)\r
{\r
  s32 *buf32 = PsndBuffer;\r
  int stereo = (PicoIn.opt & 8) >> 3;\r
  int psglen = ((Pico.snd.psg_pos+0x80000) >> 20);\r
  int ym2413len = ((Pico.snd.ym2413_pos+0x80000) >> 20);\r
\r
  if (!PicoIn.sndOut)\r
    return length;\r
\r
  pprof_start(sound);\r
\r
  // Add in parts of the PSG output not yet done\r
  if (length-psglen > 0) {\r
    s16 *psgbuf = PicoIn.sndOut + (psglen << stereo);\r
    Pico.snd.psg_pos += (length-psglen) << 20;\r
    if (PicoIn.opt & POPT_EN_PSG)\r
      SN76496Update(psgbuf, length-psglen, stereo);\r
  }\r
\r
  if (length-ym2413len > 0) {\r
    s16 *ym2413buf = PicoIn.sndOut + (ym2413len << stereo);\r
    Pico.snd.ym2413_pos += (length-ym2413len) << 20;\r
    int len = (length-ym2413len);\r
    if (Pico.m.hardware & PMS_HW_FMUSED) {\r
      PsndFMUpdate(buf32, len, 0, 0);\r
      if (stereo)\r
        while (len--) {\r
          *ym2413buf++ += *buf32;\r
          *ym2413buf++ += *buf32++;\r
        }\r
      else\r
        while (len--) {\r
          *ym2413buf++ += *buf32++;\r
        }\r
    }\r
  }\r
\r
  pprof_end(sound);\r
\r
  return length;\r
}\r
\r
PICO_INTERNAL void PsndGetSamplesMS(int y)\r
{\r
  static int curr_pos = 0;\r
\r
  curr_pos  = PsndRenderMS(0, Pico.snd.len_use);\r
\r
  if (PicoIn.writeSound != NULL && PicoIn.sndOut)\r
    PicoIn.writeSound(curr_pos * ((PicoIn.opt & POPT_EN_STEREO) ? 4 : 2));\r
  PsndClear();\r
}\r
\r
// vim:shiftwidth=2:ts=2:expandtab\r