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mixer.c

/*                DirectSound
 *
 * Copyright 1998 Marcus Meissner
 * Copyright 1998 Rob Riggs
 * Copyright 2000-2002 TransGaming Technologies, Inc.
 *
 * This library is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * This library is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with this library; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */

#include <assert.h>
#include <stdarg.h>
#include <math.h> /* Insomnia - pow() function */

#define NONAMELESSSTRUCT
#define NONAMELESSUNION
#include "windef.h"
#include "winbase.h"
#include "mmsystem.h"
#include "winreg.h"
#include "winternl.h"
#include "wine/debug.h"
#include "dsound.h"
#include "dsdriver.h"
#include "dsound_private.h"

WINE_DEFAULT_DEBUG_CHANNEL(dsound);

void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan)
{
      double temp;
      TRACE("(%p)\n",volpan);

      TRACE("Vol=%ld Pan=%ld\n", volpan->lVolume, volpan->lPan);
      /* the AmpFactors are expressed in 16.16 fixed point */
      volpan->dwVolAmpFactor = (ULONG) (pow(2.0, volpan->lVolume / 600.0) * 0xffff);
      /* FIXME: dwPan{Left|Right}AmpFactor */

      /* FIXME: use calculated vol and pan ampfactors */
      temp = (double) (volpan->lVolume - (volpan->lPan > 0 ? volpan->lPan : 0));
      volpan->dwTotalLeftAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
      temp = (double) (volpan->lVolume + (volpan->lPan < 0 ? volpan->lPan : 0));
      volpan->dwTotalRightAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);

      TRACE("left = %lx, right = %lx\n", volpan->dwTotalLeftAmpFactor, volpan->dwTotalRightAmpFactor);
}

void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan)
{
    double left,right;
    TRACE("(%p)\n",volpan);

    TRACE("left=%lx, right=%lx\n",volpan->dwTotalLeftAmpFactor,volpan->dwTotalRightAmpFactor);
    if (volpan->dwTotalLeftAmpFactor==0)
        left=-10000;
    else
        left=600 * log(((double)volpan->dwTotalLeftAmpFactor) / 0xffff) / log(2);
    if (volpan->dwTotalRightAmpFactor==0)
        right=-10000;
    else
        right=600 * log(((double)volpan->dwTotalRightAmpFactor) / 0xffff) / log(2);
    if (left<right)
    {
        volpan->lVolume=right;
        volpan->dwVolAmpFactor=volpan->dwTotalRightAmpFactor;
    }
    else
    {
        volpan->lVolume=left;
        volpan->dwVolAmpFactor=volpan->dwTotalLeftAmpFactor;
    }
    if (volpan->lVolume < -10000)
        volpan->lVolume=-10000;
    volpan->lPan=right-left;
    if (volpan->lPan < -10000)
        volpan->lPan=-10000;

    TRACE("Vol=%ld Pan=%ld\n", volpan->lVolume, volpan->lPan);
}

void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
{
      TRACE("(%p)\n",dsb);

      /* calculate the 10ms write lead */
      dsb->writelead = (dsb->freq / 100) * dsb->pwfx->nBlockAlign;
}

void DSOUND_CheckEvent(IDirectSoundBufferImpl *dsb, int len)
{
      int               i;
      DWORD             offset;
      LPDSBPOSITIONNOTIFY     event;
      TRACE("(%p,%d)\n",dsb,len);

      if (dsb->nrofnotifies == 0)
            return;

      TRACE("(%p) buflen = %ld, playpos = %ld, len = %d\n",
            dsb, dsb->buflen, dsb->playpos, len);
      for (i = 0; i < dsb->nrofnotifies ; i++) {
            event = dsb->notifies + i;
            offset = event->dwOffset;
            TRACE("checking %d, position %ld, event = %p\n",
                  i, offset, event->hEventNotify);
            /* DSBPN_OFFSETSTOP has to be the last element. So this is */
            /* OK. [Inside DirectX, p274] */
            /*  */
            /* This also means we can't sort the entries by offset, */
            /* because DSBPN_OFFSETSTOP == -1 */
            if (offset == DSBPN_OFFSETSTOP) {
                  if (dsb->state == STATE_STOPPED) {
                        SetEvent(event->hEventNotify);
                        TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
                        return;
                  } else
                        return;
            }
            if ((dsb->playpos + len) >= dsb->buflen) {
                  if ((offset < ((dsb->playpos + len) % dsb->buflen)) ||
                      (offset >= dsb->playpos)) {
                        TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
                        SetEvent(event->hEventNotify);
                  }
            } else {
                  if ((offset >= dsb->playpos) && (offset < (dsb->playpos + len))) {
                        TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
                        SetEvent(event->hEventNotify);
                  }
            }
      }
}

/* WAV format info can be found at:
 *
 *    http://www.cwi.nl/ftp/audio/AudioFormats.part2
 *    ftp://ftp.cwi.nl/pub/audio/RIFF-format
 *
 * Import points to remember:
 *    8-bit WAV is unsigned
 *    16-bit WAV is signed
 */
 /* Use the same formulas as pcmconverter.c */
static inline INT16 cvtU8toS16(BYTE b)
{
    return (short)((b+(b << 8))-32768);
}

static inline BYTE cvtS16toU8(INT16 s)
{
    return (s >> 8) ^ (unsigned char)0x80;
}

static inline void cp_fields(const IDirectSoundBufferImpl *dsb, BYTE *ibuf, BYTE *obuf )
{
        INT fl,fr;

        if (dsb->pwfx->wBitsPerSample == 8)  {
                if (dsb->dsound->pwfx->wBitsPerSample == 8 &&
                    dsb->dsound->pwfx->nChannels == dsb->pwfx->nChannels) {
                        /* avoid needless 8->16->8 conversion */
                        *obuf=*ibuf;
                        if (dsb->pwfx->nChannels==2)
                                *(obuf+1)=*(ibuf+1);
                        return;
                }
                fl = cvtU8toS16(*ibuf);
                fr = (dsb->pwfx->nChannels==2 ? cvtU8toS16(*(ibuf + 1)) : fl);
        } else {
                fl = *((INT16 *)ibuf);
                fr = (dsb->pwfx->nChannels==2 ? *(((INT16 *)ibuf) + 1)  : fl);
        }

        if (dsb->dsound->pwfx->nChannels == 2) {
                if (dsb->dsound->pwfx->wBitsPerSample == 8) {
                        *obuf = cvtS16toU8(fl);
                        *(obuf + 1) = cvtS16toU8(fr);
                        return;
                }
                if (dsb->dsound->pwfx->wBitsPerSample == 16) {
                        *((INT16 *)obuf) = fl;
                        *(((INT16 *)obuf) + 1) = fr;
                        return;
                }
        }
        if (dsb->dsound->pwfx->nChannels == 1) {
                fl = (fl + fr) >> 1;
                if (dsb->dsound->pwfx->wBitsPerSample == 8) {
                        *obuf = cvtS16toU8(fl);
                        return;
                }
                if (dsb->dsound->pwfx->wBitsPerSample == 16) {
                        *((INT16 *)obuf) = fl;
                        return;
                }
        }
}

/* Now with PerfectPitch (tm) technology */
static INT DSOUND_MixerNorm(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len)
{
      INT   i, size, ipos, ilen;
      BYTE  *ibp, *obp;
      INT   iAdvance = dsb->pwfx->nBlockAlign;
      INT   oAdvance = dsb->dsound->pwfx->nBlockAlign;

      ibp = dsb->buffer->memory + dsb->buf_mixpos;
      obp = buf;

      TRACE("(%p, %p, %p), buf_mixpos=%ld\n", dsb, ibp, obp, dsb->buf_mixpos);
      /* Check for the best case */
      if ((dsb->freq == dsb->dsound->pwfx->nSamplesPerSec) &&
          (dsb->pwfx->wBitsPerSample == dsb->dsound->pwfx->wBitsPerSample) &&
          (dsb->pwfx->nChannels == dsb->dsound->pwfx->nChannels)) {
              INT bytesleft = dsb->buflen - dsb->buf_mixpos;
            TRACE("(%p) Best case\n", dsb);
            if (len <= bytesleft )
                  CopyMemory(obp, ibp, len);
            else { /* wrap */
                  CopyMemory(obp, ibp, bytesleft);
                  CopyMemory(obp + bytesleft, dsb->buffer->memory, len - bytesleft);
            }
            return len;
      }

      /* Check for same sample rate */
      if (dsb->freq == dsb->dsound->pwfx->nSamplesPerSec) {
            TRACE("(%p) Same sample rate %ld = primary %ld\n", dsb,
                  dsb->freq, dsb->dsound->pwfx->nSamplesPerSec);
            ilen = 0;
            for (i = 0; i < len; i += oAdvance) {
                  cp_fields(dsb, ibp, obp );
                  ibp += iAdvance;
                  ilen += iAdvance;
                  obp += oAdvance;
                  if (ibp >= (BYTE *)(dsb->buffer->memory + dsb->buflen))
                        ibp = dsb->buffer->memory;    /* wrap */
            }
            return (ilen);
      }

      /* Mix in different sample rates */
      /* */
      /* New PerfectPitch(tm) Technology (c) 1998 Rob Riggs */
      /* Patent Pending :-] */

      /* Patent enhancements (c) 2000 Ove Kåven,
       * TransGaming Technologies Inc. */

      /* FIXME("(%p) Adjusting frequency: %ld -> %ld (need optimization)\n",
         dsb, dsb->freq, dsb->dsound->pwfx->nSamplesPerSec); */

      size = len / oAdvance;
      ilen = 0;
      ipos = dsb->buf_mixpos;
      for (i = 0; i < size; i++) {
                cp_fields(dsb, (dsb->buffer->memory + ipos), obp);
            obp += oAdvance;
            dsb->freqAcc += dsb->freqAdjust;
            if (dsb->freqAcc >= (1<<DSOUND_FREQSHIFT)) {
                  ULONG adv = (dsb->freqAcc>>DSOUND_FREQSHIFT) * iAdvance;
                  dsb->freqAcc &= (1<<DSOUND_FREQSHIFT)-1;
                  ipos += adv; ilen += adv;
                  ipos %= dsb->buflen;
            }
      }
      return ilen;
}

static void DSOUND_MixerVol(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len)
{
      INT   i;
      BYTE  *bpc = buf;
      INT16 *bps = (INT16 *) buf;

      TRACE("(%p,%p,%d)\n",dsb,buf,len);
      TRACE("left = %lx, right = %lx\n", dsb->cvolpan.dwTotalLeftAmpFactor, 
            dsb->cvolpan.dwTotalRightAmpFactor);

      if ((!(dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) || (dsb->cvolpan.lPan == 0)) &&
          (!(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) || (dsb->cvolpan.lVolume == 0)) &&
          !(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
            return;           /* Nothing to do */

      /* If we end up with some bozo coder using panning or 3D sound */
      /* with a mono primary buffer, it could sound very weird using */
      /* this method. Oh well, tough patooties. */

      switch (dsb->dsound->pwfx->wBitsPerSample) {
      case 8:
            /* 8-bit WAV is unsigned, but we need to operate */
            /* on signed data for this to work properly */
            switch (dsb->dsound->pwfx->nChannels) {
            case 1:
                  for (i = 0; i < len; i++) {
                        INT val = *bpc - 128;
                        val = (val * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
                        *bpc = val + 128;
                        bpc++;
                  }
                  break;
            case 2:
                  for (i = 0; i < len; i+=2) {
                        INT val = *bpc - 128;
                        val = (val * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
                        *bpc++ = val + 128;
                        val = *bpc - 128;
                        val = (val * dsb->cvolpan.dwTotalRightAmpFactor) >> 16;
                        *bpc = val + 128;
                        bpc++;
                  }
                  break;
            default:
                  FIXME("doesn't support %d channels\n", dsb->dsound->pwfx->nChannels);
                  break;
            }
            break;
      case 16:
            /* 16-bit WAV is signed -- much better */
            switch (dsb->dsound->pwfx->nChannels) {
            case 1:
                  for (i = 0; i < len; i += 2) {
                        *bps = (*bps * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
                        bps++;
                  }
                  break;
            case 2:
                  for (i = 0; i < len; i += 4) {
                        *bps = (*bps * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
                        bps++;
                        *bps = (*bps * dsb->cvolpan.dwTotalRightAmpFactor) >> 16;
                        bps++;
                  }
                  break;
            default:
                  FIXME("doesn't support %d channels\n", dsb->dsound->pwfx->nChannels);
                  break;
            }
            break;
      default:
            FIXME("doesn't support %d bit samples\n", dsb->dsound->pwfx->wBitsPerSample);
            break;
      }
}

static LPBYTE DSOUND_tmpbuffer(IDirectSoundImpl *dsound, DWORD len)
{
    TRACE("(%p,%ld)\n",dsound,len);

    if (len > dsound->tmp_buffer_len) {
        if (dsound->tmp_buffer)
            dsound->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsound->tmp_buffer, len);
        else
            dsound->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, len);

        dsound->tmp_buffer_len = len;
    }

    return dsound->tmp_buffer;
}

static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen)
{
      INT   i, len, ilen, field, todo;
      BYTE  *buf, *ibuf;

      TRACE("(%p,%ld,%ld)\n",dsb,writepos,fraglen);

      len = fraglen;
      if (!(dsb->playflags & DSBPLAY_LOOPING)) {
            INT temp = MulDiv(dsb->dsound->pwfx->nAvgBytesPerSec, dsb->buflen,
                  dsb->nAvgBytesPerSec) -
                  MulDiv(dsb->dsound->pwfx->nAvgBytesPerSec, dsb->buf_mixpos,
                  dsb->nAvgBytesPerSec);
            len = min(len, temp);
      }

      if (len % dsb->dsound->pwfx->nBlockAlign) {
            INT nBlockAlign = dsb->dsound->pwfx->nBlockAlign;
            len = (len / nBlockAlign) * nBlockAlign;  /* data alignment */
            ERR("length not a multiple of block size, len = %d, block size = %d\n", len, nBlockAlign);
      }

      if (len == 0) {
            /* This should only happen if we aren't looping and temp < nBlockAlign */
            return 0;
      }

      if ((buf = ibuf = DSOUND_tmpbuffer(dsb->dsound, len)) == NULL)
            return 0;

      TRACE("MixInBuffer (%p) len = %d, dest = %ld\n", dsb, len, writepos);

      ilen = DSOUND_MixerNorm(dsb, ibuf, len);
      if ((dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) ||
          (dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) ||
          (dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
            DSOUND_MixerVol(dsb, ibuf, len);

      if (dsb->dsound->pwfx->wBitsPerSample == 8) {
            BYTE  *obuf = dsb->dsound->buffer + writepos;

            if ((writepos + len) <= dsb->dsound->buflen)
                  todo = len;
            else
                  todo = dsb->dsound->buflen - writepos;

            for (i = 0; i < todo; i++) {
                  /* 8-bit WAV is unsigned */
                  field = (*ibuf++ - 128);
                  field += (*obuf - 128);
                  if (field > 127) field = 127;
                  else if (field < -128) field = -128;
                  *obuf++ = field + 128;
            }
 
            if (todo < len) {
                  todo = len - todo;
                  obuf = dsb->dsound->buffer;

                  for (i = 0; i < todo; i++) {
                        /* 8-bit WAV is unsigned */
                        field = (*ibuf++ - 128);
                        field += (*obuf - 128);
                        if (field > 127) field = 127;
                        else if (field < -128) field = -128;
                        *obuf++ = field + 128;
                  }
            }
        } else {
            INT16 *ibufs, *obufs;

            ibufs = (INT16 *) ibuf;
            obufs = (INT16 *)(dsb->dsound->buffer + writepos);

            if ((writepos + len) <= dsb->dsound->buflen)
                  todo = len / 2;
            else
                  todo = (dsb->dsound->buflen - writepos) / 2;

            for (i = 0; i < todo; i++) {
                  /* 16-bit WAV is signed */
                  field = *ibufs++;
                  field += *obufs;
                  if (field > 32767) field = 32767;
                  else if (field < -32768) field = -32768;
                  *obufs++ = field;
            }

            if (todo < (len / 2)) {
                  todo = (len / 2) - todo;
                  obufs = (INT16 *)dsb->dsound->buffer;

                  for (i = 0; i < todo; i++) {
                        /* 16-bit WAV is signed */
                        field = *ibufs++;
                        field += *obufs;
                        if (field > 32767) field = 32767;
                        else if (field < -32768) field = -32768;
                        *obufs++ = field;
                  }
            }
        }

      if (dsb->leadin && (dsb->startpos > dsb->buf_mixpos) && (dsb->startpos <= dsb->buf_mixpos + ilen)) {
            /* HACK... leadin should be reset when the PLAY position reaches the startpos,
             * not the MIX position... but if the sound buffer is bigger than our prebuffering
             * (which must be the case for the streaming buffers that need this hack anyway)
             * plus DS_HEL_MARGIN or equivalent, then this ought to work anyway. */
            dsb->leadin = FALSE;
      }

      dsb->buf_mixpos += ilen;

      if (dsb->buf_mixpos >= dsb->buflen) {
            if (dsb->playflags & DSBPLAY_LOOPING) {
                  /* wrap */
                  dsb->buf_mixpos %= dsb->buflen;
                  if (dsb->leadin && (dsb->startpos <= dsb->buf_mixpos))
                        dsb->leadin = FALSE; /* HACK: see above */
            }
      }

      return len;
}

static void DSOUND_PhaseCancel(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD len)
{
      INT     ilen, field;
      UINT    i, todo;
      BYTE  *buf, *ibuf;

      TRACE("(%p,%ld,%ld)\n",dsb,writepos,len);

      if (len % dsb->dsound->pwfx->nBlockAlign) {
            INT nBlockAlign = dsb->dsound->pwfx->nBlockAlign;
            len = (len / nBlockAlign) * nBlockAlign;  /* data alignment */
            ERR("length not a multiple of block size, len = %ld, block size = %d\n", len, nBlockAlign);
      }

      if ((buf = ibuf = DSOUND_tmpbuffer(dsb->dsound, len)) == NULL)
            return;

      TRACE("PhaseCancel (%p) len = %ld, dest = %ld\n", dsb, len, writepos);

      ilen = DSOUND_MixerNorm(dsb, ibuf, len);
      if ((dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) ||
          (dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) ||
          (dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
            DSOUND_MixerVol(dsb, ibuf, len);

      /* subtract instead of add, to phase out premixed data */
      if (dsb->dsound->pwfx->wBitsPerSample == 8) {
            BYTE  *obuf = dsb->dsound->buffer + writepos;

            if ((writepos + len) <= dsb->dsound->buflen)
                  todo = len;
            else
                  todo = dsb->dsound->buflen - writepos;

            for (i = 0; i < todo; i++) {
                  /* 8-bit WAV is unsigned */
                  field = (*ibuf++ - 128);
                  field -= (*obuf - 128);
                  if (field > 127) field = 127;
                  else if (field < -128) field = -128;
                  *obuf++ = field + 128;
            }
 
            if (todo < len) {
                  todo = len - todo;
                  obuf = dsb->dsound->buffer;

                  for (i = 0; i < todo; i++) {
                        /* 8-bit WAV is unsigned */
                        field = (*ibuf++ - 128);
                        field -= (*obuf - 128);
                        if (field > 127) field = 127;
                        else if (field < -128) field = -128;
                        *obuf++ = field + 128;
                  }
            }
        } else {
            INT16 *ibufs, *obufs;

            ibufs = (INT16 *) ibuf;
            obufs = (INT16 *)(dsb->dsound->buffer + writepos);

            if ((writepos + len) <= dsb->dsound->buflen)
                  todo = len / 2;
            else
                  todo = (dsb->dsound->buflen - writepos) / 2;

            for (i = 0; i < todo; i++) {
                  /* 16-bit WAV is signed */
                  field = *ibufs++;
                  field -= *obufs;
                  if (field > 32767) field = 32767;
                  else if (field < -32768) field = -32768;
                  *obufs++ = field;
            }

            if (todo < (len / 2)) {
                  todo = (len / 2) - todo;
                  obufs = (INT16 *)dsb->dsound->buffer;

                  for (i = 0; i < todo; i++) {
                        /* 16-bit WAV is signed */
                        field = *ibufs++;
                        field -= *obufs;
                        if (field > 32767) field = 32767;
                        else if (field < -32768) field = -32768;
                        *obufs++ = field;
                  }
            }
        }
}

static void DSOUND_MixCancel(IDirectSoundBufferImpl *dsb, DWORD writepos, BOOL cancel)
{
      DWORD   size, flen, len, npos, nlen;
      INT   iAdvance = dsb->pwfx->nBlockAlign;
      INT   oAdvance = dsb->dsound->pwfx->nBlockAlign;
      /* determine amount of premixed data to cancel */
      DWORD primary_done =
            ((dsb->primary_mixpos < writepos) ? dsb->dsound->buflen : 0) +
            dsb->primary_mixpos - writepos;

      TRACE("(%p, %ld), buf_mixpos=%ld\n", dsb, writepos, dsb->buf_mixpos);

      /* backtrack the mix position */
      size = primary_done / oAdvance;
      flen = size * dsb->freqAdjust;
      len = (flen >> DSOUND_FREQSHIFT) * iAdvance;
      flen &= (1<<DSOUND_FREQSHIFT)-1;
      while (dsb->freqAcc < flen) {
            len += iAdvance;
            dsb->freqAcc += 1<<DSOUND_FREQSHIFT;
      }
      len %= dsb->buflen;
      npos = ((dsb->buf_mixpos < len) ? dsb->buflen : 0) +
            dsb->buf_mixpos - len;
      if (dsb->leadin && (dsb->startpos > npos) && (dsb->startpos <= npos + len)) {
            /* stop backtracking at startpos */
            npos = dsb->startpos;
            len = ((dsb->buf_mixpos < npos) ? dsb->buflen : 0) +
                  dsb->buf_mixpos - npos;
            flen = dsb->freqAcc;
            nlen = len / dsb->pwfx->nBlockAlign;
            nlen = ((nlen << DSOUND_FREQSHIFT) + flen) / dsb->freqAdjust;
            nlen *= dsb->dsound->pwfx->nBlockAlign;
            writepos =
                  ((dsb->primary_mixpos < nlen) ? dsb->dsound->buflen : 0) +
                  dsb->primary_mixpos - nlen;
      }

      dsb->freqAcc -= flen;
      dsb->buf_mixpos = npos;
      dsb->primary_mixpos = writepos;

      TRACE("new buf_mixpos=%ld, primary_mixpos=%ld (len=%ld)\n",
            dsb->buf_mixpos, dsb->primary_mixpos, len);

      if (cancel) DSOUND_PhaseCancel(dsb, writepos, len);
}

void DSOUND_MixCancelAt(IDirectSoundBufferImpl *dsb, DWORD buf_writepos)
{
#if 0
      DWORD   i, size, flen, len, npos, nlen;
      INT   iAdvance = dsb->pwfx->nBlockAlign;
      INT   oAdvance = dsb->dsound->pwfx->nBlockAlign;
      /* determine amount of premixed data to cancel */
      DWORD buf_done =
            ((dsb->buf_mixpos < buf_writepos) ? dsb->buflen : 0) +
            dsb->buf_mixpos - buf_writepos;
#endif

      WARN("(%p, %ld), buf_mixpos=%ld\n", dsb, buf_writepos, dsb->buf_mixpos);
      /* since this is not implemented yet, just cancel *ALL* prebuffering for now
       * (which is faster anyway when there's only a single secondary buffer) */
      dsb->dsound->need_remix = TRUE;
}

void DSOUND_ForceRemix(IDirectSoundBufferImpl *dsb)
{
      TRACE("(%p)\n",dsb);
      EnterCriticalSection(&dsb->lock);
      if (dsb->state == STATE_PLAYING) {
#if 0 /* this may not be quite reliable yet */
            dsb->need_remix = TRUE;
#else
            dsb->dsound->need_remix = TRUE;
#endif
      }
      LeaveCriticalSection(&dsb->lock);
}

static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, DWORD playpos, DWORD writepos, DWORD mixlen)
{
      DWORD len, slen;
      /* determine this buffer's write position */
      DWORD buf_writepos = DSOUND_CalcPlayPosition(dsb, dsb->state & dsb->dsound->state, writepos,
                                         writepos, dsb->primary_mixpos, dsb->buf_mixpos);
      /* determine how much already-mixed data exists */
      DWORD buf_done =
            ((dsb->buf_mixpos < buf_writepos) ? dsb->buflen : 0) +
            dsb->buf_mixpos - buf_writepos;
      DWORD primary_done =
            ((dsb->primary_mixpos < writepos) ? dsb->dsound->buflen : 0) +
            dsb->primary_mixpos - writepos;
      DWORD adv_done =
            ((dsb->dsound->mixpos < writepos) ? dsb->dsound->buflen : 0) +
            dsb->dsound->mixpos - writepos;
      DWORD played =
            ((buf_writepos < dsb->playpos) ? dsb->buflen : 0) +
            buf_writepos - dsb->playpos;
      DWORD buf_left = dsb->buflen - buf_writepos;
      int still_behind;

      TRACE("(%p,%ld,%ld,%ld)\n",dsb,playpos,writepos,mixlen);
      TRACE("buf_writepos=%ld, primary_writepos=%ld\n", buf_writepos, writepos);
      TRACE("buf_done=%ld, primary_done=%ld\n", buf_done, primary_done);
      TRACE("buf_mixpos=%ld, primary_mixpos=%ld, mixlen=%ld\n", dsb->buf_mixpos, dsb->primary_mixpos,
            mixlen);
      TRACE("looping=%ld, startpos=%ld, leadin=%ld\n", dsb->playflags, dsb->startpos, dsb->leadin);

      /* check for notification positions */
      if (dsb->dsbd.dwFlags & DSBCAPS_CTRLPOSITIONNOTIFY &&
          dsb->state != STATE_STARTING) {
            DSOUND_CheckEvent(dsb, played);
      }

      /* save write position for non-GETCURRENTPOSITION2... */
      dsb->playpos = buf_writepos;

      /* check whether CalcPlayPosition detected a mixing underrun */
      if ((buf_done == 0) && (dsb->primary_mixpos != writepos)) {
            /* it did, but did we have more to play? */
            if ((dsb->playflags & DSBPLAY_LOOPING) ||
                (dsb->buf_mixpos < dsb->buflen)) {
                  /* yes, have to recover */
                  ERR("underrun on sound buffer %p\n", dsb);
                  TRACE("recovering from underrun: primary_mixpos=%ld\n", writepos);
            }
            dsb->primary_mixpos = writepos;
            primary_done = 0;
      }
      /* determine how far ahead we should mix */
      if (((dsb->playflags & DSBPLAY_LOOPING) ||
           (dsb->leadin && (dsb->probably_valid_to != 0))) &&
          !(dsb->dsbd.dwFlags & DSBCAPS_STATIC)) {
            /* if this is a streaming buffer, it typically means that
             * we should defer mixing past probably_valid_to as long
             * as we can, to avoid unnecessary remixing */
            /* the heavy-looking calculations shouldn't be that bad,
             * as any game isn't likely to be have more than 1 or 2
             * streaming buffers in use at any time anyway... */
            DWORD probably_valid_left =
                  (dsb->probably_valid_to == (DWORD)-1) ? dsb->buflen :
                  ((dsb->probably_valid_to < buf_writepos) ? dsb->buflen : 0) +
                  dsb->probably_valid_to - buf_writepos;
            /* check for leadin condition */
            if ((probably_valid_left == 0) &&
                (dsb->probably_valid_to == dsb->startpos) &&
                dsb->leadin)
                  probably_valid_left = dsb->buflen;
            TRACE("streaming buffer probably_valid_to=%ld, probably_valid_left=%ld\n",
                  dsb->probably_valid_to, probably_valid_left);
            /* check whether the app's time is already up */
            if (probably_valid_left < dsb->writelead) {
                  WARN("probably_valid_to now within writelead, possible streaming underrun\n");
                  /* once we pass the point of no return,
                   * no reason to hold back anymore */
                  dsb->probably_valid_to = (DWORD)-1;
                  /* we just have to go ahead and mix what we have,
                   * there's no telling what the app is thinking anyway */
            } else {
                  /* adjust for our frequency and our sample size */
                  probably_valid_left = MulDiv(probably_valid_left,
                                         1 << DSOUND_FREQSHIFT,
                                         dsb->pwfx->nBlockAlign * dsb->freqAdjust) *
                                      dsb->dsound->pwfx->nBlockAlign;
                  /* check whether to clip mix_len */
                  if (probably_valid_left < mixlen) {
                        TRACE("clipping to probably_valid_left=%ld\n", probably_valid_left);
                        mixlen = probably_valid_left;
                  }
            }
      }
      /* cut mixlen with what's already been mixed */
      if (mixlen < primary_done) {
            /* huh? and still CalcPlayPosition didn't
             * detect an underrun? */
            FIXME("problem with underrun detection (mixlen=%ld < primary_done=%ld)\n", mixlen, primary_done);
            return 0;
      }
      len = mixlen - primary_done;
      TRACE("remaining mixlen=%ld\n", len);

      if (len < dsb->dsound->fraglen) {
            /* smaller than a fragment, wait until it gets larger
             * before we take the mixing overhead */
            TRACE("mixlen not worth it, deferring mixing\n");
            still_behind = 1;
            goto post_mix;
      }

      /* ok, we know how much to mix, let's go */
      still_behind = (adv_done > primary_done);
      while (len) {
            slen = dsb->dsound->buflen - dsb->primary_mixpos;
            if (slen > len) slen = len;
            slen = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, slen);

            if ((dsb->primary_mixpos < dsb->dsound->mixpos) &&
                (dsb->primary_mixpos + slen >= dsb->dsound->mixpos))
                  still_behind = FALSE;

            dsb->primary_mixpos += slen; len -= slen;
            dsb->primary_mixpos %= dsb->dsound->buflen;

            if ((dsb->state == STATE_STOPPED) || !slen) break;
      }
      TRACE("new primary_mixpos=%ld, primary_advbase=%ld\n", dsb->primary_mixpos, dsb->dsound->mixpos);
      TRACE("mixed data len=%ld, still_behind=%d\n", mixlen-len, still_behind);

post_mix:
      /* check if buffer should be considered complete */
      if (buf_left < dsb->writelead &&
          !(dsb->playflags & DSBPLAY_LOOPING)) {
            dsb->state = STATE_STOPPED;
            dsb->playpos = 0;
            dsb->last_playpos = 0;
            dsb->buf_mixpos = 0;
            dsb->leadin = FALSE;
            DSOUND_CheckEvent(dsb, buf_left);
      }

      /* return how far we think the primary buffer can
       * advance its underrun detector...*/
      if (still_behind) return 0;
      if ((mixlen - len) < primary_done) return 0;
      slen = ((dsb->primary_mixpos < dsb->dsound->mixpos) ?
            dsb->dsound->buflen : 0) + dsb->primary_mixpos -
            dsb->dsound->mixpos;
      if (slen > mixlen) {
            /* the primary_done and still_behind checks above should have worked */
            FIXME("problem with advancement calculation (advlen=%ld > mixlen=%ld)\n", slen, mixlen);
            slen = 0;
      }
      return slen;
}

static DWORD DSOUND_MixToPrimary(IDirectSoundImpl *dsound, DWORD playpos, DWORD writepos, DWORD mixlen, BOOL recover)
{
      INT               i, len, maxlen = 0;
      IDirectSoundBufferImpl  *dsb;

      TRACE("(%ld,%ld,%ld,%d)\n", playpos, writepos, mixlen, recover);
      for (i = 0; i < dsound->nrofbuffers; i++) {
            dsb = dsound->buffers[i];

            if (dsb->buflen && dsb->state && !dsb->hwbuf) {
                  TRACE("Checking %p, mixlen=%ld\n", dsb, mixlen);
                  EnterCriticalSection(&(dsb->lock));
                  if (dsb->state == STATE_STOPPING) {
                        DSOUND_MixCancel(dsb, writepos, TRUE);
                        dsb->state = STATE_STOPPED;
                        DSOUND_CheckEvent(dsb, 0);
                  } else {
                        if ((dsb->state == STATE_STARTING) || recover) {
                              dsb->primary_mixpos = writepos;
                              dsb->cvolpan = dsb->volpan;
                              dsb->need_remix = FALSE;
                        }
                        else if (dsb->need_remix) {
                              DSOUND_MixCancel(dsb, writepos, TRUE);
                              dsb->cvolpan = dsb->volpan;
                              dsb->need_remix = FALSE;
                        }
                        len = DSOUND_MixOne(dsb, playpos, writepos, mixlen);
                        if (dsb->state == STATE_STARTING)
                              dsb->state = STATE_PLAYING;
                        maxlen = (len > maxlen) ? len : maxlen;
                  }
                  LeaveCriticalSection(&(dsb->lock));
            }
      }

      return maxlen;
}

static void DSOUND_MixReset(IDirectSoundImpl *dsound, DWORD writepos)
{
      INT               i;
      IDirectSoundBufferImpl  *dsb;
      int nfiller;

      TRACE("(%ld)\n", writepos);

      /* the sound of silence */
      nfiller = dsound->pwfx->wBitsPerSample == 8 ? 128 : 0;

      /* reset all buffer mix positions */
      for (i = 0; i < dsound->nrofbuffers; i++) {
            dsb = dsound->buffers[i];

            if (dsb->buflen && dsb->state && !dsb->hwbuf) {
                  TRACE("Resetting %p\n", dsb);
                  EnterCriticalSection(&(dsb->lock));
                  if (dsb->state == STATE_STOPPING) {
                        dsb->state = STATE_STOPPED;
                  }
                  else if (dsb->state == STATE_STARTING) {
                        /* nothing */
                  } else {
                        DSOUND_MixCancel(dsb, writepos, FALSE);
                        dsb->cvolpan = dsb->volpan;
                        dsb->need_remix = FALSE;
                  }
                  LeaveCriticalSection(&(dsb->lock));
            }
      }

      /* wipe out premixed data */
      if (dsound->mixpos < writepos) {
            FillMemory(dsound->buffer + writepos, dsound->buflen - writepos, nfiller);
            FillMemory(dsound->buffer, dsound->mixpos, nfiller);
      } else {
            FillMemory(dsound->buffer + writepos, dsound->mixpos - writepos, nfiller);
      }

      /* reset primary mix position */
      dsound->mixpos = writepos;
}

static void DSOUND_CheckReset(IDirectSoundImpl *dsound, DWORD writepos)
{
      TRACE("(%p,%ld)\n",dsound,writepos);
      if (dsound->need_remix) {
            DSOUND_MixReset(dsound, writepos);
            dsound->need_remix = FALSE;
            /* maximize Half-Life performance */
            dsound->prebuf = ds_snd_queue_min;
            dsound->precount = 0;
      } else {
            dsound->precount++;
            if (dsound->precount >= 4) {
                  if (dsound->prebuf < ds_snd_queue_max)
                        dsound->prebuf++;
                  dsound->precount = 0;
            }
      }
      TRACE("premix adjust: %d\n", dsound->prebuf);
}

void DSOUND_WaveQueue(IDirectSoundImpl *dsound, DWORD mixq)
{
      TRACE("(%p,%ld)\n",dsound,mixq);
      if (mixq + dsound->pwqueue > ds_hel_queue) mixq = ds_hel_queue - dsound->pwqueue;
      TRACE("queueing %ld buffers, starting at %d\n", mixq, dsound->pwwrite);
      for (; mixq; mixq--) {
            waveOutWrite(dsound->hwo, dsound->pwave[dsound->pwwrite], sizeof(WAVEHDR));
            dsound->pwwrite++;
            if (dsound->pwwrite >= DS_HEL_FRAGS) dsound->pwwrite = 0;
            dsound->pwqueue++;
      }
}

/* #define SYNC_CALLBACK */

void DSOUND_PerformMix(IDirectSoundImpl *dsound)
{
      int nfiller;
      BOOL forced;
      HRESULT hres;

      TRACE("(%p)\n", dsound);

      /* the sound of silence */
      nfiller = dsound->pwfx->wBitsPerSample == 8 ? 128 : 0;

      /* whether the primary is forced to play even without secondary buffers */
      forced = ((dsound->state == STATE_PLAYING) || (dsound->state == STATE_STARTING));

      if (dsound->priolevel != DSSCL_WRITEPRIMARY) {
            BOOL paused = ((dsound->state == STATE_STOPPED) || (dsound->state == STATE_STARTING));
            /* FIXME: document variables */
            DWORD playpos, writepos, inq, maxq, frag;
            if (dsound->hwbuf) {
                  hres = IDsDriverBuffer_GetPosition(dsound->hwbuf, &playpos, &writepos);
                  if (hres) {
                      WARN("IDsDriverBuffer_GetPosition failed\n");
                      return;
                  }
                  /* Well, we *could* do Just-In-Time mixing using the writepos,
                   * but that's a little bit ambitious and unnecessary... */
                  /* rather add our safety margin to the writepos, if we're playing */
                  if (!paused) {
                        writepos += dsound->writelead;
                        writepos %= dsound->buflen;
                  } else writepos = playpos;
            } else {
                  playpos = dsound->pwplay * dsound->fraglen;
                  writepos = playpos;
                  if (!paused) {
                        writepos += ds_hel_margin * dsound->fraglen;
                        writepos %= dsound->buflen;
                  }
            }
            TRACE("primary playpos=%ld, writepos=%ld, clrpos=%ld, mixpos=%ld, buflen=%ld\n",
                  playpos,writepos,dsound->playpos,dsound->mixpos,dsound->buflen);
            assert(dsound->playpos < dsound->buflen);
            /* wipe out just-played sound data */
            if (playpos < dsound->playpos) {
                  FillMemory(dsound->buffer + dsound->playpos, dsound->buflen - dsound->playpos, nfiller);
                  FillMemory(dsound->buffer, playpos, nfiller);
            } else {
                  FillMemory(dsound->buffer + dsound->playpos, playpos - dsound->playpos, nfiller);
            }
            dsound->playpos = playpos;

            EnterCriticalSection(&(dsound->mixlock));

            /* reset mixing if necessary */
            DSOUND_CheckReset(dsound, writepos);

            /* check how much prebuffering is left */
            inq = dsound->mixpos;
            if (inq < writepos)
                  inq += dsound->buflen;
            inq -= writepos;

            /* find the maximum we can prebuffer */
            if (!paused) {
                  maxq = playpos;
                  if (maxq < writepos)
                        maxq += dsound->buflen;
                  maxq -= writepos;
            } else maxq = dsound->buflen;

            /* clip maxq to dsound->prebuf */
            frag = dsound->prebuf * dsound->fraglen;
            if (maxq > frag) maxq = frag;

            /* check for consistency */
            if (inq > maxq) {
                  /* the playback position must have passed our last
                   * mixed position, i.e. it's an underrun, or we have
                   * nothing more to play */
                  TRACE("reached end of mixed data (inq=%ld, maxq=%ld)\n", inq, maxq);
                  inq = 0;
                  /* stop the playback now, to allow buffers to refill */
                  if (dsound->state == STATE_PLAYING) {
                        dsound->state = STATE_STARTING;
                  }
                  else if (dsound->state == STATE_STOPPING) {
                        dsound->state = STATE_STOPPED;
                  }
                  else {
                        /* how can we have an underrun if we aren't playing? */
                        WARN("unexpected primary state (%ld)\n", dsound->state);
                  }
#ifdef SYNC_CALLBACK
                  /* DSOUND_callback may need this lock */
                  LeaveCriticalSection(&(dsound->mixlock));
#endif
                  if (DSOUND_PrimaryStop(dsound) != DS_OK)
                        WARN("DSOUND_PrimaryStop failed\n");
#ifdef SYNC_CALLBACK
                  EnterCriticalSection(&(dsound->mixlock));
#endif
                  if (dsound->hwbuf) {
                        /* the Stop is supposed to reset play position to beginning of buffer */
                        /* unfortunately, OSS is not able to do so, so get current pointer */
                        hres = IDsDriverBuffer_GetPosition(dsound->hwbuf, &playpos, NULL);
                        if (hres) {
                              LeaveCriticalSection(&(dsound->mixlock));
                              WARN("IDsDriverBuffer_GetPosition failed\n");
                              return;
                        }
                  } else {
                        playpos = dsound->pwplay * dsound->fraglen;
                  }
                  writepos = playpos;
                  dsound->playpos = playpos;
                  dsound->mixpos = writepos;
                  inq = 0;
                  maxq = dsound->buflen;
                  if (maxq > frag) maxq = frag;
                  FillMemory(dsound->buffer, dsound->buflen, nfiller);
                  paused = TRUE;
            }

            /* do the mixing */
            frag = DSOUND_MixToPrimary(dsound, playpos, writepos, maxq, paused);
            if (forced) frag = maxq - inq;
            dsound->mixpos += frag;
            dsound->mixpos %= dsound->buflen;

            if (frag) {
                  /* buffers have been filled, restart playback */
                  if (dsound->state == STATE_STARTING) {
                        dsound->state = STATE_PLAYING;
                  }
                  else if (dsound->state == STATE_STOPPED) {
                        /* the dsound is supposed to play if there's something to play
                         * even if it is reported as stopped, so don't let this confuse you */
                        dsound->state = STATE_STOPPING;
                  }
                  LeaveCriticalSection(&(dsound->mixlock));
                  if (paused) {
                        if (DSOUND_PrimaryPlay(dsound) != DS_OK)
                              WARN("DSOUND_PrimaryPlay failed\n");
                        else
                              TRACE("starting playback\n");
                  }
            }
            else
                  LeaveCriticalSection(&(dsound->mixlock));
      } else {
            /* in the DSSCL_WRITEPRIMARY mode, the app is totally in charge... */
            if (dsound->state == STATE_STARTING) {
                  if (DSOUND_PrimaryPlay(dsound) != DS_OK)
                        WARN("DSOUND_PrimaryPlay failed\n");
                  else
                        dsound->state = STATE_PLAYING;
            }
            else if (dsound->state == STATE_STOPPING) {
                  if (DSOUND_PrimaryStop(dsound) != DS_OK)
                        WARN("DSOUND_PrimaryStop failed\n");
                  else
                        dsound->state = STATE_STOPPED;
            }
      }
}

void CALLBACK DSOUND_timer(UINT timerID, UINT msg, DWORD dwUser, DWORD dw1, DWORD dw2)
{
        IDirectSoundImpl* This = (IDirectSoundImpl*)dwUser;
      DWORD start_time =  GetTickCount();
        DWORD end_time;
      TRACE("(%d,%d,0x%lx,0x%lx,0x%lx)\n",timerID,msg,dwUser,dw1,dw2);
        TRACE("entering at %ld\n", start_time);

      if (dsound != This) {
            ERR("dsound died without killing us?\n");
            timeKillEvent(timerID);
            timeEndPeriod(DS_TIME_RES);
            return;
      }

      RtlAcquireResourceShared(&(This->buffer_list_lock), TRUE);

      if (This->ref)
            DSOUND_PerformMix(This);

      RtlReleaseResource(&(This->buffer_list_lock));

      end_time = GetTickCount();
      TRACE("completed processing at %ld, duration = %ld\n", end_time, end_time - start_time);
}

void CALLBACK DSOUND_callback(HWAVEOUT hwo, UINT msg, DWORD dwUser, DWORD dw1, DWORD dw2)
{
        IDirectSoundImpl* This = (IDirectSoundImpl*)dwUser;
      TRACE("(%p,%x,%lx,%lx,%lx)\n",hwo,msg,dwUser,dw1,dw2);
      TRACE("entering at %ld, msg=%08x(%s)\n", GetTickCount(), msg, 
            msg==MM_WOM_DONE ? "MM_WOM_DONE" : msg==MM_WOM_CLOSE ? "MM_WOM_CLOSE" : 
            msg==MM_WOM_OPEN ? "MM_WOM_OPEN" : "UNKNOWN");
      if (msg == MM_WOM_DONE) {
            DWORD inq, mixq, fraglen, buflen, pwplay, playpos, mixpos;
            if (This->pwqueue == (DWORD)-1) {
                  TRACE("completed due to reset\n");
                  return;
            }
/* it could be a bad idea to enter critical section here... if there's lock contention,
 * the resulting scheduling delays might obstruct the winmm player thread */
#ifdef SYNC_CALLBACK
            EnterCriticalSection(&(This->mixlock));
#endif
            /* retrieve current values */
            fraglen = This->fraglen;
            buflen = This->buflen;
            pwplay = This->pwplay;
            playpos = pwplay * fraglen;
            mixpos = This->mixpos;
            /* check remaining mixed data */
            inq = ((mixpos < playpos) ? buflen : 0) + mixpos - playpos;
            mixq = inq / fraglen;
            if ((inq - (mixq * fraglen)) > 0) mixq++;
            /* complete the playing buffer */
            TRACE("done playing primary pos=%ld\n", playpos);
            pwplay++;
            if (pwplay >= DS_HEL_FRAGS) pwplay = 0;
            /* write new values */
            This->pwplay = pwplay;
            This->pwqueue--;
            /* queue new buffer if we have data for it */
            if (inq>1) DSOUND_WaveQueue(This, inq-1);
#ifdef SYNC_CALLBACK
            LeaveCriticalSection(&(This->mixlock));
#endif
      }
      TRACE("completed\n");
}

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