cmd.exe: Add support for cmd.exe /u (Unicode from internal pgms).

[wine/wine-kai.git] / dlls / dsound / 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., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
 */

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

#define NONAMELESSSTRUCT
#define NONAMELESSUNION
#include "windef.h"
#include "winbase.h"
#include "winuser.h"
#include "mmsystem.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=%d Pan=%d\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 = %x, right = %x\n", volpan->dwTotalLeftAmpFactor, volpan->dwTotalRightAmpFactor);
}

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

    TRACE("left=%x, right=%x\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=%d Pan=%d\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;
}

/**
 * Check for application callback requests for when the play position
 * reaches certain points.
 *
 * The offsets that will be triggered will be those between the recorded
 * "last played" position for the buffer (i.e. dsb->playpos) and "len" bytes
 * beyond that position.
 */
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 = %d, playpos = %d, 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 %d, 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;
}

/**
 * Copy a single frame from the given input buffer to the given output buffer.
 * Translate 8 <-> 16 bits and mono <-> stereo
 */
static inline void cp_fields(const IDirectSoundBufferImpl *dsb, const BYTE *ibuf, BYTE *obuf )
{
        DirectSoundDevice * device = dsb->device;
        INT fl,fr;

        if (dsb->pwfx->wBitsPerSample == 8)  {
                if (device->pwfx->wBitsPerSample == 8 &&
                    device->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 = *((const INT16 *)ibuf);
                fr = (dsb->pwfx->nChannels==2 ? *(((const INT16 *)ibuf) + 1)  : fl);
        }

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

/**
 * Mix at most the given amount of data into the given device buffer from the
 * given secondary buffer, starting from the dsb's first currently unmixed
 * frame (buf_mixpos), translating frequency (pitch), stereo/mono and
 * bits-per-sample. The secondary buffer sample is looped if it is not
 * long enough and it is a looping buffer.
 * (Doesn't perform any mixing - this is a straight copy operation).
 *
 * Now with PerfectPitch (tm) technology
 *
 * dsb = the secondary buffer
 * buf = the device buffer
 * len = number of bytes to store in the device buffer
 *
 * Returns: the number of bytes read from the secondary buffer
 *   (ie. len, adjusted for frequency, number of channels and sample size,
 *    and limited by buffer length for non-looping buffers)
 */
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->device->pwfx->nBlockAlign;

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

        TRACE("(%p, %p, %p), buf_mixpos=%d\n", dsb, ibp, obp, dsb->buf_mixpos);
        /* Check for the best case */
        if ((dsb->freq == dsb->device->pwfx->nSamplesPerSec) &&
            (dsb->pwfx->wBitsPerSample == dsb->device->pwfx->wBitsPerSample) &&
            (dsb->pwfx->nChannels == dsb->device->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->device->pwfx->nSamplesPerSec) {
                TRACE("(%p) Same sample rate %d = primary %d\n", dsb,
                        dsb->freq, dsb->device->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->device->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 = %x, right = %x\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->device->pwfx->wBitsPerSample) {
        case 8:
                /* 8-bit WAV is unsigned, but we need to operate */
                /* on signed data for this to work properly */
                switch (dsb->device->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->device->pwfx->nChannels);
                        break;
                }
                break;
        case 16:
                /* 16-bit WAV is signed -- much better */
                switch (dsb->device->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->device->pwfx->nChannels);
                        break;
                }
                break;
        default:
                FIXME("doesn't support %d bit samples\n", dsb->device->pwfx->wBitsPerSample);
                break;
        }
}

/**
 * Make sure the device's tmp_buffer is at least the given size. Return a
 * pointer to it.
 */
static LPBYTE DSOUND_tmpbuffer(DirectSoundDevice *device, DWORD len)
{
    TRACE("(%p,%d)\n", device, len);

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

        device->tmp_buffer_len = len;
    }

    return device->tmp_buffer;
}

/**
 * Mix (at most) the given number of bytes into the given position of the
 * device buffer, from the secondary buffer "dsb" (starting at the current
 * mix position for that buffer).
 *
 * Returns the number of bytes actually mixed into the device buffer. This
 * will match fraglen unless the end of the secondary buffer is reached
 * (and it is not looping).
 *
 * dsb  = the secondary buffer to mix from
 * writepos = position (offset) in device buffer to write at
 * fraglen = number of bytes to mix
 */
static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen)
{
        INT     i, len, ilen, field, todo;
        BYTE    *buf, *ibuf;

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

        len = fraglen;
        if (!(dsb->playflags & DSBPLAY_LOOPING)) {
                /* This buffer is not looping, so make sure the requested
                 * length will not take us past the end of the buffer */
                int secondary_remainder = dsb->buflen - dsb->buf_mixpos;
                int adjusted_remainder = MulDiv(dsb->device->pwfx->nAvgBytesPerSec, secondary_remainder, dsb->nAvgBytesPerSec);
                assert(adjusted_remainder >= 0);
                        /* The adjusted remainder must be at least one sample,
                         * otherwise we will never reach the end of the
                         * secondary buffer, as there will perpetually be a
                         * fractional remainder */
                TRACE("secondary_remainder = %d, adjusted_remainder = %d, len = %d\n", secondary_remainder, adjusted_remainder, len);
                if (adjusted_remainder < len) {
                        TRACE("clipping len to remainder of secondary buffer\n");
                        len = adjusted_remainder;
                }
                if (len == 0)
                        return 0;
        }

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

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

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

        /* first, copy the data from the DirectSoundBuffer into the temporary
           buffer, translating frequency/bits-per-sample/number-of-channels
           to match the device settings */
        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);

        /* Now mix the temporary buffer into the devices main buffer */
        if (dsb->device->pwfx->wBitsPerSample == 8) {
                BYTE    *obuf = dsb->device->buffer + writepos;

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

                if ((writepos + len) <= dsb->device->buflen)
                        todo = len / 2;
                else
                        todo = (dsb->device->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->device->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 */
                } else if (dsb->buf_mixpos > dsb->buflen) {
                        ERR("Mixpos (%u) past buflen (%u), capping...\n", dsb->buf_mixpos, dsb->buflen);
                        dsb->buf_mixpos = dsb->buflen;
                }
        }

        return len;
}

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

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

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

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

        TRACE("PhaseCancel (%p) len = %d, dest = %d\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->device->pwfx->wBitsPerSample == 8) {
                BYTE    *obuf = dsb->device->buffer + writepos;

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

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

                        for (i = 0; i < todo; i++) {
                                /* 8-bit WAV is unsigned */
                                field = (*obuf - 128);
                                field -= (*ibuf++ - 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->device->buffer + writepos);

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

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

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

                        for (i = 0; i < todo; i++) {
                                /* 16-bit WAV is signed */
                                field = *obufs;
                                field -= *ibufs++;
                                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->device->pwfx->nBlockAlign;
        /* determine amount of premixed data to cancel */
        DWORD primary_done =
                ((dsb->primary_mixpos < writepos) ? dsb->device->buflen : 0) +
                dsb->primary_mixpos - writepos;

        TRACE("(%p, %d), buf_mixpos=%d\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->device->pwfx->nBlockAlign;
                writepos =
                        ((dsb->primary_mixpos < nlen) ? dsb->device->buflen : 0) +
                        dsb->primary_mixpos - nlen;
        }

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

        TRACE("new buf_mixpos=%d, primary_mixpos=%d (len=%d)\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->device->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, %d), buf_mixpos=%d\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->device->need_remix = TRUE;
}

void DSOUND_ForceRemix(IDirectSoundBufferImpl *dsb)
{
        TRACE("(%p)\n",dsb);
        EnterCriticalSection(&dsb->lock);
        if (dsb->state == STATE_PLAYING)
                dsb->device->need_remix = TRUE;
        LeaveCriticalSection(&dsb->lock);
}

/**
 * Calculate the distance between two buffer offsets, taking wraparound
 * into account.
 */
static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
{
        if (ptr1 >= ptr2) {
                return ptr1 - ptr2;
        } else {
                return buflen + ptr1 - ptr2;
        }
}

/**
 * Mix some frames from the given secondary buffer "dsb" into the device
 * primary buffer.
 *
 * dsb = the secondary buffer
 * playpos = the current play position in the device buffer (primary buffer)
 * writepos = the current safe-to-write position in the device buffer
 * mixlen = the maximum number of bytes in the primary buffer to mix, from the
 *          current writepos.
 *
 * Returns: the number of bytes beyond the writepos that were mixed.
 */
static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, DWORD playpos, DWORD writepos, DWORD mixlen)
{
        /* The buffer's primary_mixpos may be before or after the the device
         * buffer's mixpos, but both must be ahead of writepos. */

        DWORD len, slen;
        /* determine this buffer's write position */
        DWORD buf_writepos = DSOUND_CalcPlayPosition(dsb, writepos, writepos);
        /* determine how much already-mixed data exists */
        DWORD buf_done = DSOUND_BufPtrDiff(dsb->buflen, dsb->buf_mixpos, buf_writepos);
        DWORD primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);
        DWORD adv_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->device->mixpos, writepos);
        DWORD played = DSOUND_BufPtrDiff(dsb->buflen, buf_writepos, dsb->playpos);
        DWORD buf_left = dsb->buflen - buf_writepos;
        int still_behind;

        TRACE("(%p,%d,%d,%d)\n",dsb,playpos,writepos,mixlen);
        TRACE("buf_writepos=%d, primary_writepos=%d\n", buf_writepos, writepos);
        TRACE("buf_done=%d, primary_done=%d\n", buf_done, primary_done);
        TRACE("buf_mixpos=%d, primary_mixpos=%d, mixlen=%d\n", dsb->buf_mixpos, dsb->primary_mixpos,
              mixlen);
        TRACE("looping=%d, startpos=%d, leadin=%d\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=%d\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=%d, probably_valid_left=%d\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->device->pwfx->nBlockAlign;
                        /* check whether to clip mix_len */
                        if (probably_valid_left < mixlen) {
                                TRACE("clipping to probably_valid_left=%d\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=%d < primary_done=%d)\n", mixlen, primary_done);
                return 0;
        }
        len = mixlen - primary_done;
        TRACE("remaining mixlen=%d\n", len);

        if (len < dsb->device->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->device->buflen - dsb->primary_mixpos;
                if (slen > len) slen = len;
                slen = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, slen);

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

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

                if ((dsb->state == STATE_STOPPED) || !slen) break;
        }
        TRACE("new primary_mixpos=%d, primary_advbase=%d\n", dsb->primary_mixpos, dsb->device->mixpos);
        TRACE("mixed data len=%d, 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;
                dsb->need_remix = 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 = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, dsb->device->mixpos);
        if (slen > mixlen) {
                /* the primary_done and still_behind checks above should have worked */
                FIXME("problem with advancement calculation (advlen=%d > mixlen=%d)\n", slen, mixlen);
                slen = 0;
        }
        return slen;
}

/**
 * For a DirectSoundDevice, go through all the currently playing buffers and
 * mix them in to the device buffer.
 *
 * playpos = the current play position in the primary buffer
 * writepos = the current safe-to-write position in the primary buffer
 * mixlen = the maximum amount to mix into the primary buffer
 *          (beyond the current writepos)
 * recover = true if the sound device may have been reset and the write
 *           position in the device buffer changed
 *
 * Returns:  the length beyond the writepos that was mixed to.
 */
static DWORD DSOUND_MixToPrimary(const DirectSoundDevice *device, DWORD playpos, DWORD writepos,
                                 DWORD mixlen, BOOL recover)
{
        INT                     i, len, maxlen = 0;
        IDirectSoundBufferImpl  *dsb;

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

                if (dsb->buflen && dsb->state && !dsb->hwbuf) {
                        TRACE("Checking %p, mixlen=%d\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(DirectSoundDevice *device, DWORD writepos)
{
        INT                     i;
        IDirectSoundBufferImpl  *dsb;
        int nfiller;

        TRACE("(%p,%d)\n", device, writepos);

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

        /* reset all buffer mix positions */
        for (i = 0; i < device->nrofbuffers; i++) {
                dsb = device->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 (device->mixpos < writepos) {
                FillMemory(device->buffer + writepos, device->buflen - writepos, nfiller);
                FillMemory(device->buffer, device->mixpos, nfiller);
        } else {
                FillMemory(device->buffer + writepos, device->mixpos - writepos, nfiller);
        }

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

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

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

/* #define SYNC_CALLBACK */

/**
 * Perform mixing for a Direct Sound device. That is, go through all the
 * secondary buffers (the sound bites currently playing) and mix them in
 * to the primary buffer (the device buffer).
 */
static void DSOUND_PerformMix(DirectSoundDevice *device)
{
        int nfiller;
        BOOL forced;
        HRESULT hres;

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

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

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

        if (device->priolevel != DSSCL_WRITEPRIMARY) {
                BOOL paused = ((device->state == STATE_STOPPED) || (device->state == STATE_STARTING));
                /* FIXME: document variables */
                DWORD playpos, writepos, inq, maxq, frag;
                if (device->hwbuf) {
                        hres = IDsDriverBuffer_GetPosition(device->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 += device->writelead;
                                writepos %= device->buflen;
                        } else writepos = playpos;
                } else {
                        playpos = device->pwplay * device->fraglen;
                        writepos = playpos;
                        if (!paused) {
                                writepos += ds_hel_margin * device->fraglen;
                                writepos %= device->buflen;
                        }
                }
                TRACE("primary playpos=%d, writepos=%d, clrpos=%d, mixpos=%d, buflen=%d\n",
                      playpos,writepos,device->playpos,device->mixpos,device->buflen);
                assert(device->playpos < device->buflen);
                /* wipe out just-played sound data */
                if (playpos < device->playpos) {
                        FillMemory(device->buffer + device->playpos, device->buflen - device->playpos, nfiller);
                        FillMemory(device->buffer, playpos, nfiller);
                } else {
                        FillMemory(device->buffer + device->playpos, playpos - device->playpos, nfiller);
                }
                device->playpos = playpos;

                EnterCriticalSection(&(device->mixlock));

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

                /* check how much prebuffering is left */
                inq = DSOUND_BufPtrDiff(device->buflen, device->mixpos, writepos);

                /* find the maximum we can prebuffer */
                if (!paused)
                        maxq = DSOUND_BufPtrDiff(device->buflen, playpos, writepos);
                /* If we get the whole buffer, difference is 0, so we need to set whole buffer then */
                if (paused || !maxq)
                        maxq = device->buflen;

                /* clip maxq to device->prebuf */
                frag = device->prebuf * device->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=%d, maxq=%d)\n", inq, maxq);
                        inq = 0;
                        /* stop the playback now, to allow buffers to refill */
                        if (device->state == STATE_PLAYING) {
                                device->state = STATE_STARTING;
                        }
                        else if (device->state == STATE_STOPPING) {
                                device->state = STATE_STOPPED;
                        }
                        else {
                                /* how can we have an underrun if we aren't playing? */
                                WARN("unexpected primary state (%d)\n", device->state);
                        }
#ifdef SYNC_CALLBACK
                        /* DSOUND_callback may need this lock */
                        LeaveCriticalSection(&(device->mixlock));
#endif
                        if (DSOUND_PrimaryStop(device) != DS_OK)
                                WARN("DSOUND_PrimaryStop failed\n");
#ifdef SYNC_CALLBACK
                        EnterCriticalSection(&(device->mixlock));
#endif
                        if (device->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(device->hwbuf, &playpos, NULL);
                                if (hres) {
                                        LeaveCriticalSection(&(device->mixlock));
                                        WARN("IDsDriverBuffer_GetPosition failed\n");
                                        return;
                                }
                        } else {
                                playpos = device->pwplay * device->fraglen;
                        }
                        writepos = playpos;
                        device->playpos = playpos;
                        device->mixpos = writepos;
                        inq = 0;
                        maxq = device->buflen;
                        if (maxq > frag) maxq = frag;
                        FillMemory(device->buffer, device->buflen, nfiller);
                        paused = TRUE;
                }

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

                if (frag) {
                        /* buffers have been filled, restart playback */
                        if (device->state == STATE_STARTING) {
                                device->state = STATE_PLAYING;
                        }
                        else if (device->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 */
                                device->state = STATE_STOPPING;
                        }
                        LeaveCriticalSection(&(device->mixlock));
                        if (paused) {
                                if (DSOUND_PrimaryPlay(device) != DS_OK)
                                        WARN("DSOUND_PrimaryPlay failed\n");
                                else
                                        TRACE("starting playback\n");
                        }
                }
                else
                        LeaveCriticalSection(&(device->mixlock));
        } else {
                /* in the DSSCL_WRITEPRIMARY mode, the app is totally in charge... */
                if (device->state == STATE_STARTING) {
                        if (DSOUND_PrimaryPlay(device) != DS_OK)
                                WARN("DSOUND_PrimaryPlay failed\n");
                        else
                                device->state = STATE_PLAYING;
                }
                else if (device->state == STATE_STOPPING) {
                        if (DSOUND_PrimaryStop(device) != DS_OK)
                                WARN("DSOUND_PrimaryStop failed\n");
                        else
                                device->state = STATE_STOPPED;
                }
        }
}

void CALLBACK DSOUND_timer(UINT timerID, UINT msg, DWORD_PTR dwUser,
                           DWORD_PTR dw1, DWORD_PTR dw2)
{
        DirectSoundDevice * device = (DirectSoundDevice*)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 %d\n", start_time);

        if (DSOUND_renderer[device->drvdesc.dnDevNode] != device) {
                ERR("dsound died without killing us?\n");
                timeKillEvent(timerID);
                timeEndPeriod(DS_TIME_RES);
                return;
        }

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

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

        RtlReleaseResource(&(device->buffer_list_lock));

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

void CALLBACK DSOUND_callback(HWAVEOUT hwo, UINT msg, DWORD dwUser, DWORD dw1, DWORD dw2)
{
        DirectSoundDevice * device = (DirectSoundDevice*)dwUser;
        TRACE("(%p,%x,%x,%x,%x)\n",hwo,msg,dwUser,dw1,dw2);
        TRACE("entering at %d, 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 (device->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(&(device->mixlock));
#endif
                /* retrieve current values */
                fraglen = device->fraglen;
                buflen = device->buflen;
                pwplay = device->pwplay;
                playpos = pwplay * fraglen;
                mixpos = device->mixpos;
                /* check remaining mixed data */
                inq = DSOUND_BufPtrDiff(buflen, mixpos, playpos);
                mixq = inq / fraglen;
                if ((inq - (mixq * fraglen)) > 0) mixq++;
                /* complete the playing buffer */
                TRACE("done playing primary pos=%d\n", playpos);
                pwplay++;
                if (pwplay >= DS_HEL_FRAGS) pwplay = 0;
                /* write new values */
                device->pwplay = pwplay;
                device->pwqueue--;
                /* queue new buffer if we have data for it */
                if (inq>1) DSOUND_WaveQueue(device, inq-1);
#ifdef SYNC_CALLBACK
                LeaveCriticalSection(&(device->mixlock));
#endif
        }
        TRACE("completed\n");
}