dsound: Fix locking in mixer.

[wine/wine64.git] / dlls / dsound / mixer.c

/*                      DirectSound
 *
 * Copyright 1998 Marcus Meissner
 * Copyright 1998 Rob Riggs
 * Copyright 2000-2002 TransGaming Technologies, Inc.
 * Copyright 2007 Peter Dons Tychsen
 *
 * 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, DWORD playpos, 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, 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 ((playpos + len) >= dsb->buflen) {
                        if ((offset < ((playpos + len) % dsb->buflen)) ||
                            (offset >= playpos)) {
                                TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
                                SetEvent(event->hEventNotify);
                        }
                } else {
                        if ((offset >= playpos) && (offset < (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 K�ven,
         * 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->volpan.dwTotalLeftAmpFactor,
                dsb->volpan.dwTotalRightAmpFactor);

        if ((!(dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) || (dsb->volpan.lPan == 0)) &&
            (!(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) || (dsb->volpan.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->volpan.dwTotalLeftAmpFactor) >> 16;
                                *bpc = val + 128;
                                bpc++;
                        }
                        break;
                case 2:
                        for (i = 0; i < len; i+=2) {
                                INT val = *bpc - 128;
                                val = (val * dsb->volpan.dwTotalLeftAmpFactor) >> 16;
                                *bpc++ = val + 128;
                                val = *bpc - 128;
                                val = (val * dsb->volpan.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->volpan.dwTotalLeftAmpFactor) >> 16;
                                bps++;
                        }
                        break;
                case 2:
                        for (i = 0; i < len; i += 4) {
                                *bps = (*bps * dsb->volpan.dwTotalLeftAmpFactor) >> 16;
                                bps++;
                                *bps = (*bps * dsb->volpan.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);
                adjusted_remainder -= adjusted_remainder % dsb->device->pwfx->nBlockAlign; /* data alignment */
                        /* 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; /* data alignment */
        }

        /* Create temp buffer to hold actual resulting data */
        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);

        /* then apply the correct volume, if necessary */
        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;
        }

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

        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;
                }
        }

        /* increase mix position */
        dsb->primary_mixpos += len;
        dsb->primary_mixpos %= dsb->device->buflen;
        return len;
}

/**
 * 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
 * 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 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 primary_done, buflen = dsb->buflen / dsb->pwfx->nBlockAlign * dsb->device->pwfx->nBlockAlign;

        TRACE("(%p,%d,%d)\n",dsb,writepos,mixlen);
        TRACE("writepos=%d, buf_mixpos=%d, primary_mixpos=%d, mixlen=%d\n", writepos, dsb->buf_mixpos, dsb->primary_mixpos, mixlen);
        TRACE("looping=%d, startpos=%d, leadin=%d, buflen=%d\n", dsb->playflags, dsb->startpos, dsb->leadin, dsb->buflen);

        /* calculate how much pre-buffering has already been done for this buffer */
        primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);

        /* sanity */
        if(mixlen < primary_done)
        {
                /* Should *NEVER* happen */
                ERR("Fatal error. Under/Overflow? primary_done=%d, mixpos=%d, primary_mixpos=%d, writepos=%d, mixlen=%d\n", primary_done,dsb->buf_mixpos,dsb->primary_mixpos, writepos, mixlen);
                return 0;
        }

        /* take into acount already mixed data */
        mixlen -= primary_done;

        TRACE("primary_done=%d, mixlen (primary) = %i\n", primary_done, mixlen);

        if ((dsb->playflags & DSBPLAY_LOOPING) && mixlen > buflen)
        {
                while (mixlen > buflen)
                {
                        DWORD mixedlength = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, buflen);
                        mixlen -= buflen;
                        if (!mixedlength)
                        {
                                mixlen = 0;
                                break;
                        }
                }

        }

        /* clip to valid length */
        mixlen = (buflen < mixlen) ? buflen : mixlen;
        TRACE("mixlen (buffer)=%d\n", mixlen);

        if (mixlen)
                /* mix more data */
                mixlen = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixlen);

        TRACE("new primary_mixpos=%d, mixed data len=%d, buffer left = %d\n",
                dsb->primary_mixpos, mixlen, (dsb->buflen - dsb->buf_mixpos));

        /* re-calculate the primary done */
        primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);

        /* check if buffer should be considered complete */
        if (((dsb->buflen - dsb->buf_mixpos) < dsb->writelead) &&
            !(dsb->playflags & DSBPLAY_LOOPING)) {

                TRACE("Buffer reached end. Stopped\n");

                dsb->state = STATE_STOPPED;
                dsb->buf_mixpos = 0;
                dsb->leadin = FALSE;
        }

        /* Report back the total prebuffered amount for this buffer */
        return primary_done;
}

/**
 * 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
 * all_stopped = reports back if all buffers have stopped
 *
 * Returns:  the length beyond the writepos that was mixed to.
 */

static DWORD DSOUND_MixToPrimary(const DirectSoundDevice *device, DWORD writepos,
                                 DWORD mixlen, BOOL recover, BOOL *all_stopped)
{
        INT i, len;
        DWORD minlen = 0;
        IDirectSoundBufferImpl  *dsb;

        /* unless we find a running buffer, all have stopped */
        *all_stopped = TRUE;

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

                TRACE("MixToPrimary for %p, state=%d\n", dsb, dsb->state);

                if (dsb->buflen && dsb->state && !dsb->hwbuf) {
                        TRACE("Checking %p, mixlen=%d\n", dsb, mixlen);
                        RtlAcquireResourceShared(&dsb->lock, TRUE);

                        /* if buffer is stopping it is stopped now */
                        if (dsb->state == STATE_STOPPING) {
                                dsb->state = STATE_STOPPED;
                                DSOUND_CheckEvent(dsb, 0, 0);
                        } else {

                                /* if recovering, reset the mix position */
                                if ((dsb->state == STATE_STARTING) || recover) {
                                        dsb->primary_mixpos = writepos;
                                }

                                /* mix next buffer into the main buffer */
                                len = DSOUND_MixOne(dsb, writepos, mixlen);

                                /* if the buffer was starting, it must be playing now */
                                if (dsb->state == STATE_STARTING)
                                        dsb->state = STATE_PLAYING;

                                /* check if min-len should be initialized */
                                if(minlen == 0) minlen = len;

                                /* record the minimum length mixed from all buffers */
                                /* we only want to return the length which *all* buffers have mixed */
                                if(len != 0) minlen = (len < minlen) ? len : minlen;
                        }

                        if(dsb->state != STATE_STOPPED){
                                *all_stopped = FALSE;
                        }

                        RtlReleaseResource(&dsb->lock);
                }
        }

        TRACE("Mixed at least %d from all buffers\n", minlen);

        return minlen;
}

/**
 * Add buffers to the emulated wave device system.
 *
 * device = The current dsound playback device
 * force = If TRUE, the function will buffer up as many frags as possible,
 *         even though and will ignore the actual state of the primary buffer.
 *
 * Returns:  None
 */

static void DSOUND_WaveQueue(DirectSoundDevice *device, BOOL force)
{
        DWORD prebuf_frags, wave_writepos, wave_fragpos, i;
        TRACE("(%p)\n", device);

        /* calculate the current wave frag position */
        wave_fragpos = (device->pwplay + device->pwqueue) % DS_HEL_FRAGS;

        /* calculte the current wave write position */
        wave_writepos = wave_fragpos * device->fraglen;

        TRACE("wave_fragpos = %i, wave_writepos = %i, pwqueue = %i, prebuf = %i\n",
                wave_fragpos, wave_writepos, device->pwqueue, device->prebuf);

        if(force == FALSE){
                /* check remaining prebuffered frags */
                prebuf_frags = DSOUND_BufPtrDiff(device->buflen, device->mixpos, wave_writepos);
                prebuf_frags = prebuf_frags / device->fraglen;
        }
        else{
                /* buffer the maximum amount of frags */
                prebuf_frags = device->prebuf;
        }

        /* limit to the queue we have left */
        if((prebuf_frags + device->pwqueue) > device->prebuf)
                prebuf_frags = device->prebuf - device->pwqueue;

        TRACE("prebuf_frags = %i\n", prebuf_frags);

        /* adjust queue */
        device->pwqueue += prebuf_frags;

        /* get out of CS when calling the wave system */
        LeaveCriticalSection(&(device->mixlock));
        /* **** */

        /* queue up the new buffers */
        for(i=0; i<prebuf_frags; i++){
                TRACE("queueing wave buffer %i\n", wave_fragpos);
                waveOutWrite(device->hwo, device->pwave[wave_fragpos], sizeof(WAVEHDR));
                wave_fragpos++;
                wave_fragpos %= DS_HEL_FRAGS;
        }

        /* **** */
        EnterCriticalSection(&(device->mixlock));

        TRACE("queue now = %i\n", device->pwqueue);
}

/**
 * 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)
{

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

        /* **** */
        EnterCriticalSection(&(device->mixlock));

        if (device->priolevel != DSSCL_WRITEPRIMARY) {
                BOOL recover = FALSE, all_stopped = FALSE;
                DWORD playpos, writepos, writelead, maxq, frag, prebuff_max, prebuff_left, size1, size2;
                LPVOID buf1, buf2;
                BOOL lock = (device->hwbuf && !(device->drvdesc.dwFlags & DSDDESC_DONTNEEDPRIMARYLOCK));
                int nfiller;

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

                /* get the position in the primary buffer */
                if (DSOUND_PrimaryGetPosition(device, &playpos, &writepos) != 0){
                        LeaveCriticalSection(&(device->mixlock));
                        return;
                }

                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) {
                        buf1 = device->buffer + device->playpos;
                        buf2 = device->buffer;
                        size1 = device->buflen - device->playpos;
                        size2 = playpos;
                        if (lock)
                                IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, device->playpos, size1+size2, 0);
                        FillMemory(buf1, size1, nfiller);
                        if (playpos && (!buf2 || !size2))
                                FIXME("%d: (%d, %d)=>(%d, %d) There should be an additional buffer here!!\n", __LINE__, device->playpos, device->mixpos, playpos, writepos);
                        FillMemory(buf2, size2, nfiller);
                        if (lock)
                                IDsDriverBuffer_Unlock(device->hwbuf, buf1, size1, buf2, size2);
                } else {
                        buf1 = device->buffer + device->playpos;
                        buf2 = NULL;
                        size1 = playpos - device->playpos;
                        size2 = 0;
                        if (lock)
                                IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, device->playpos, size1+size2, 0);
                        FillMemory(buf1, size1, nfiller);
                        if (buf2 && size2)
                        {
                                FIXME("%d: There should be no additional buffer here!!\n", __LINE__);
                                FillMemory(buf2, size2, nfiller);
                        }
                        if (lock)
                                IDsDriverBuffer_Unlock(device->hwbuf, buf1, size1, buf2, size2);
                }
                device->playpos = playpos;

                /* calc maximum prebuff */
                prebuff_max = (device->prebuf * device->fraglen);

                /* check how close we are to an underrun. It occurs when the writepos overtakes the mixpos */
                prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);

                writelead = DSOUND_BufPtrDiff(device->buflen, writepos, playpos);

                /* find the maximum we can prebuffer from current write position */
                maxq = (writelead < prebuff_max) ? (prebuff_max - writelead) : 0;

                TRACE("prebuff_left = %d, prebuff_max = %dx%d=%d, writelead=%d\n",
                        prebuff_left, device->prebuf, device->fraglen, prebuff_max, writelead);

                /* check for underrun. underrun occurs when the write position passes the mix position */
                if((prebuff_left > prebuff_max) || (device->state == STATE_STOPPED) || (device->state == STATE_STARTING)){
                        if (device->state == STATE_STOPPING || device->state == STATE_PLAYING)
                                WARN("Probable buffer underrun\n");
                        else TRACE("Buffer starting or buffer underrun\n");

                        /* recover mixing for all buffers */
                        recover = TRUE;

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

                if (lock)
                        IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, writepos, maxq, 0);

                /* do the mixing */
                frag = DSOUND_MixToPrimary(device, writepos, maxq, recover, &all_stopped);

                /* update the mix position, taking wrap-around into acount */
                device->mixpos = writepos + frag;
                device->mixpos %= device->buflen;

                if (lock)
                {
                        DWORD frag2 = (frag > size1 ? frag - size1 : 0);
                        frag -= frag2;
                        if (frag2 > size2)
                        {
                                FIXME("Buffering too much! (%d, %d, %d, %d)\n", maxq, frag, size2, frag2 - size2);
                                frag2 = size2;
                        }
                        IDsDriverBuffer_Unlock(device->hwbuf, buf1, frag, buf2, frag2);
                }

                /* update prebuff left */
                prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);

                /* check if have a whole fragment */
                if (prebuff_left >= device->fraglen){

                        /* update the wave queue if using wave system */
                        if(device->hwbuf == NULL){
                                DSOUND_WaveQueue(device,TRUE);
                        }

                        /* buffers are full. start playing if applicable */
                        if(device->state == STATE_STARTING){
                                TRACE("started primary buffer\n");
                                if(DSOUND_PrimaryPlay(device) != DS_OK){
                                        WARN("DSOUND_PrimaryPlay failed\n");
                                }
                                else{
                                        /* we are playing now */
                                        device->state = STATE_PLAYING;
                                }
                        }

                        /* buffers are full. start stopping if applicable */
                        if(device->state == STATE_STOPPED){
                                TRACE("restarting primary buffer\n");
                                if(DSOUND_PrimaryPlay(device) != DS_OK){
                                        WARN("DSOUND_PrimaryPlay failed\n");
                                }
                                else{
                                        /* start stopping again. as soon as there is no more data, it will stop */
                                        device->state = STATE_STOPPING;
                                }
                        }
                }

                /* if device was stopping, its for sure stopped when all buffers have stopped */
                else if((all_stopped == TRUE) && (device->state == STATE_STOPPING)){
                        TRACE("All buffers have stopped. Stopping primary buffer\n");
                        device->state = STATE_STOPPED;

                        /* stop the primary buffer now */
                        DSOUND_PrimaryStop(device);
                }

        } else {

                /* update the wave queue if using wave system */
                if(device->hwbuf == NULL)
                        DSOUND_WaveQueue(device, TRUE);
                else
                        /* Keep alsa happy, which needs GetPosition called once every 10 ms */
                        IDsDriverBuffer_GetPosition(device->hwbuf, NULL, NULL);

                /* 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;
                }
        }

        LeaveCriticalSection(&(device->mixlock));
        /* **** */
}

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");

        /* check if packet completed from wave driver */
        if (msg == MM_WOM_DONE) {

                /* **** */
                EnterCriticalSection(&(device->mixlock));

                TRACE("done playing primary pos=%d\n", device->pwplay * device->fraglen);

                /* update playpos */
                device->pwplay++;
                device->pwplay %= DS_HEL_FRAGS;

                /* sanity */
                if(device->pwqueue == 0){
                        ERR("Wave queue corrupted!\n");
                }

                /* update queue */
                device->pwqueue--;

                LeaveCriticalSection(&(device->mixlock));
                /* **** */
        }
        TRACE("completed\n");
}