dsound: Reimplement rendering devices on mmdevapi.

[wine/multimedia.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
 * Copyright 2007 Maarten Lankhorst
 * Copyright 2011 Owen Rudge for CodeWeavers
 *
 * 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 COBJMACROS
#define NONAMELESSSTRUCT
#define NONAMELESSUNION
#include "windef.h"
#include "winbase.h"
#include "mmsystem.h"
#include "wingdi.h"
#include "mmreg.h"
#include "winternl.h"
#include "wine/debug.h"
#include "dsound.h"
#include "ks.h"
#include "ksmedia.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);
}

/** Convert a primary buffer position to a pointer position for device->mix_buffer
 * device: DirectSoundDevice for which to calculate
 * pos: Primary buffer position to converts
 * Returns: Offset for mix_buffer
 */
DWORD DSOUND_bufpos_to_mixpos(const DirectSoundDevice* device, DWORD pos)
{
    DWORD ret = pos * 32 / device->pwfx->wBitsPerSample;
    if (device->pwfx->wBitsPerSample == 32)
        ret *= 2;
    return ret;
}

/* NOTE: Not all secpos have to always be mapped to a bufpos, other way around is always the case
 * DWORD64 is used here because a single DWORD wouldn't be big enough to fit the freqAcc for big buffers
 */
/** This function converts a 'native' sample pointer to a resampled pointer that fits for primary
 * secmixpos is used to decide which freqAcc is needed
 * overshot tells what the 'actual' secpos is now (optional)
 */
DWORD DSOUND_secpos_to_bufpos(const IDirectSoundBufferImpl *dsb, DWORD secpos, DWORD secmixpos, DWORD* overshot)
{
        DWORD64 framelen = secpos / dsb->pwfx->nBlockAlign;
        DWORD64 freqAdjust = dsb->freqAdjust;
        DWORD64 acc, freqAcc;

        if (secpos < secmixpos)
                freqAcc = dsb->freqAccNext;
        else freqAcc = dsb->freqAcc;
        acc = (framelen << DSOUND_FREQSHIFT) + (freqAdjust - 1 - freqAcc);
        acc /= freqAdjust;
        if (overshot)
        {
                DWORD64 oshot = acc * freqAdjust + freqAcc;
                assert(oshot >= framelen << DSOUND_FREQSHIFT);
                oshot -= framelen << DSOUND_FREQSHIFT;
                *overshot = (DWORD)oshot;
                assert(*overshot < dsb->freqAdjust);
        }
        return (DWORD)acc * dsb->device->pwfx->nBlockAlign;
}

/** Convert a resampled pointer that fits for primary to a 'native' sample pointer
 * freqAccNext is used here rather than freqAcc: In case the app wants to fill up to
 * the play position it won't overwrite it
 */
static DWORD DSOUND_bufpos_to_secpos(const IDirectSoundBufferImpl *dsb, DWORD bufpos)
{
        DWORD oAdv = dsb->device->pwfx->nBlockAlign, iAdv = dsb->pwfx->nBlockAlign, pos;
        DWORD64 framelen;
        DWORD64 acc;

        framelen = bufpos/oAdv;
        acc = framelen * (DWORD64)dsb->freqAdjust + (DWORD64)dsb->freqAccNext;
        acc = acc >> DSOUND_FREQSHIFT;
        pos = (DWORD)acc * iAdv;
        if (pos >= dsb->buflen)
                /* Because of differences between freqAcc and freqAccNext, this might happen */
                pos = dsb->buflen - iAdv;
        TRACE("Converted %d/%d to %d/%d\n", bufpos, dsb->tmp_buffer_len, pos, dsb->buflen);
        return pos;
}

/**
 * Move freqAccNext to freqAcc, and find new values for buffer length and freqAccNext
 */
static void DSOUND_RecalcFreqAcc(IDirectSoundBufferImpl *dsb)
{
        if (!dsb->freqneeded) return;
        dsb->freqAcc = dsb->freqAccNext;
        dsb->tmp_buffer_len = DSOUND_secpos_to_bufpos(dsb, dsb->buflen, 0, &dsb->freqAccNext);
        TRACE("New freqadjust: %04x, new buflen: %d\n", dsb->freqAccNext, dsb->tmp_buffer_len);
}

/**
 * Recalculate the size for temporary buffer, and new writelead
 * Should be called when one of the following things occur:
 * - Primary buffer format is changed
 * - This buffer format (frequency) is changed
 *
 * After this, DSOUND_MixToTemporary(dsb, 0, dsb->buflen) should
 * be called to refill the temporary buffer with data.
 */
void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
{
        BOOL needremix = TRUE, needresample = (dsb->freq != dsb->device->pwfx->nSamplesPerSec);
        DWORD bAlign = dsb->pwfx->nBlockAlign, pAlign = dsb->device->pwfx->nBlockAlign;
        WAVEFORMATEXTENSIBLE *pwfxe;
        BOOL ieee = FALSE;

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

        pwfxe = (WAVEFORMATEXTENSIBLE *) dsb->pwfx;

        if ((pwfxe->Format.wFormatTag == WAVE_FORMAT_IEEE_FLOAT) || ((pwfxe->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE)
            && (IsEqualGUID(&pwfxe->SubFormat, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT))))
                ieee = TRUE;

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

        if ((dsb->pwfx->wBitsPerSample == dsb->device->pwfx->wBitsPerSample) &&
            (dsb->pwfx->nChannels == dsb->device->pwfx->nChannels) && !needresample && !ieee)
                needremix = FALSE;
        HeapFree(GetProcessHeap(), 0, dsb->tmp_buffer);
        dsb->tmp_buffer = NULL;
        dsb->max_buffer_len = dsb->freqAcc = dsb->freqAccNext = 0;
        dsb->freqneeded = needresample;

        if (ieee)
                dsb->convert = convertbpp[4][dsb->device->pwfx->wBitsPerSample/8 - 1];
        else
                dsb->convert = convertbpp[dsb->pwfx->wBitsPerSample/8 - 1][dsb->device->pwfx->wBitsPerSample/8 - 1];

        dsb->resampleinmixer = FALSE;

        if (needremix)
        {
                if (needresample)
                        DSOUND_RecalcFreqAcc(dsb);
                else
                        dsb->tmp_buffer_len = dsb->buflen / bAlign * pAlign;
                dsb->max_buffer_len = dsb->tmp_buffer_len;
                if ((dsb->max_buffer_len <= dsb->device->buflen || dsb->max_buffer_len < ds_snd_shadow_maxsize * 1024 * 1024) && ds_snd_shadow_maxsize >= 0)
                        dsb->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, dsb->max_buffer_len);
                if (dsb->tmp_buffer)
                        FillMemory(dsb->tmp_buffer, dsb->tmp_buffer_len, dsb->device->pwfx->wBitsPerSample == 8 ? 128 : 0);
                else
                        dsb->resampleinmixer = TRUE;
        }
        else dsb->max_buffer_len = dsb->tmp_buffer_len = dsb->buflen;
        dsb->buf_mixpos = DSOUND_secpos_to_bufpos(dsb, dsb->sec_mixpos, 0, NULL);
}

/**
 * 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(const 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] */
                /* Windows does not seem to enforce this, and some apps rely */
                /* on that, so we can't stop there. */
                /*  */
                /* 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);
                        }
                        continue;
                }
                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);
                        }
                }
        }
}

/**
 * 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,
        UINT istride, UINT ostride, UINT count, UINT freqAcc, UINT adj)
{
    DirectSoundDevice *device = dsb->device;
    INT istep = dsb->pwfx->wBitsPerSample / 8, ostep = device->pwfx->wBitsPerSample / 8;

    if (device->pwfx->nChannels == dsb->pwfx->nChannels ||
        (device->pwfx->nChannels == 2 && dsb->pwfx->nChannels == 6) ||
        (device->pwfx->nChannels == 8 && dsb->pwfx->nChannels == 2) ||
        (device->pwfx->nChannels == 6 && dsb->pwfx->nChannels == 2)) {
        dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
        if (device->pwfx->nChannels == 2 || dsb->pwfx->nChannels == 2)
            dsb->convert(ibuf + istep, obuf + ostep, istride, ostride, count, freqAcc, adj);
        return;
    }

    if (device->pwfx->nChannels == 1 && dsb->pwfx->nChannels == 2)
    {
        dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
        return;
    }

    if (device->pwfx->nChannels == 2 && dsb->pwfx->nChannels == 1)
    {
        dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
        dsb->convert(ibuf, obuf + ostep, istride, ostride, count, freqAcc, adj);
        return;
    }

    WARN("Unable to remap channels: device=%u, buffer=%u\n", device->pwfx->nChannels,
            dsb->pwfx->nChannels);
}

/**
 * Calculate the distance between two buffer offsets, taking wraparound
 * into account.
 */
static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
{
/* If these asserts fail, the problem is not here, but in the underlying code */
        assert(ptr1 < buflen);
        assert(ptr2 < buflen);
        if (ptr1 >= ptr2) {
                return ptr1 - ptr2;
        } else {
                return buflen + ptr1 - ptr2;
        }
}
/**
 * Mix at most the given amount of data into the allocated temporary buffer
 * of the given secondary buffer, starting from the dsb's first currently
 * unsampled frame (writepos), translating frequency (pitch), stereo/mono
 * and bits-per-sample so that it is ideal for the primary buffer.
 * Doesn't perform any mixing - this is a straight copy/convert operation.
 *
 * dsb = the secondary buffer
 * writepos = Starting position of changed buffer
 * len = number of bytes to resample from writepos
 *
 * NOTE: writepos + len <= buflen. When called by mixer, MixOne makes sure of this.
 */
void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD len, BOOL inmixer)
{
        INT     size;
        BYTE    *ibp, *obp, *obp_begin;
        INT     iAdvance = dsb->pwfx->nBlockAlign;
        INT     oAdvance = dsb->device->pwfx->nBlockAlign;
        DWORD freqAcc, target_writepos = 0, overshot, maxlen;

        /* We resample only when needed */
        if ((dsb->tmp_buffer && inmixer) || (!dsb->tmp_buffer && !inmixer) || dsb->resampleinmixer != inmixer)
                return;

        assert(writepos + len <= dsb->buflen);
        if (inmixer && writepos + len < dsb->buflen)
                len += dsb->pwfx->nBlockAlign;

        maxlen = DSOUND_secpos_to_bufpos(dsb, len, 0, NULL);

        ibp = dsb->buffer->memory + writepos;
        if (!inmixer)
                obp_begin = dsb->tmp_buffer;
        else if (dsb->device->tmp_buffer_len < maxlen || !dsb->device->tmp_buffer)
        {
                dsb->device->tmp_buffer_len = maxlen;
                if (dsb->device->tmp_buffer)
                        dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, maxlen);
                else
                        dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, maxlen);
                obp_begin = dsb->device->tmp_buffer;
        }
        else
                obp_begin = dsb->device->tmp_buffer;

        TRACE("(%p, %p)\n", dsb, ibp);
        size = len / iAdvance;

        /* 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);
                obp = obp_begin;
                if (!inmixer)
                         obp += writepos/iAdvance*oAdvance;

                cp_fields(dsb, ibp, obp, iAdvance, oAdvance, size, 0, 1 << DSOUND_FREQSHIFT);
                return;
        }

        /* Mix in different sample rates */
        TRACE("(%p) Adjusting frequency: %d -> %d\n", dsb, dsb->freq, dsb->device->pwfx->nSamplesPerSec);

        target_writepos = DSOUND_secpos_to_bufpos(dsb, writepos, dsb->sec_mixpos, &freqAcc);
        overshot = freqAcc >> DSOUND_FREQSHIFT;
        if (overshot)
        {
                if (overshot >= size)
                        return;
                size -= overshot;
                writepos += overshot * iAdvance;
                if (writepos >= dsb->buflen)
                        return;
                ibp = dsb->buffer->memory + writepos;
                freqAcc &= (1 << DSOUND_FREQSHIFT) - 1;
                TRACE("Overshot: %d, freqAcc: %04x\n", overshot, freqAcc);
        }

        if (!inmixer)
                obp = obp_begin + target_writepos;
        else obp = obp_begin;

        /* FIXME: Small problem here when we're overwriting buf_mixpos, it then STILL uses old freqAcc, not sure if it matters or not */
        cp_fields(dsb, ibp, obp, iAdvance, oAdvance, size, freqAcc, dsb->freqAdjust);
}

/** Apply volume to the given soundbuffer from (primary) position writepos and length len
 * Returns: NULL if no volume needs to be applied
 * or else a memory handle that holds 'len' volume adjusted buffer */
static LPBYTE DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, INT len)
{
        INT     i;
        BYTE    *bpc;
        INT16   *bps, *mems;
        DWORD vLeft, vRight;
        INT nChannels = dsb->device->pwfx->nChannels;
        LPBYTE mem = (dsb->tmp_buffer ? dsb->tmp_buffer : dsb->buffer->memory) + dsb->buf_mixpos;

        if (dsb->resampleinmixer)
                mem = dsb->device->tmp_buffer;

        TRACE("(%p,%d)\n",dsb,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 NULL; /* Nothing to do */

        if (nChannels != 1 && nChannels != 2)
        {
                FIXME("There is no support for %d channels\n", nChannels);
                return NULL;
        }

        if (dsb->device->pwfx->wBitsPerSample != 8 && dsb->device->pwfx->wBitsPerSample != 16)
        {
                FIXME("There is no support for %d bpp\n", dsb->device->pwfx->wBitsPerSample);
                return NULL;
        }

        if (dsb->device->tmp_buffer_len < len || !dsb->device->tmp_buffer)
        {
                /* If we just resampled in DSOUND_MixToTemporary, we shouldn't need to resize here */
                assert(!dsb->resampleinmixer);
                dsb->device->tmp_buffer_len = len;
                if (dsb->device->tmp_buffer)
                        dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, len);
                else
                        dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, len);
        }

        bpc = dsb->device->tmp_buffer;
        bps = (INT16 *)bpc;
        mems = (INT16 *)mem;
        vLeft = dsb->volpan.dwTotalLeftAmpFactor;
        if (nChannels > 1)
                vRight = dsb->volpan.dwTotalRightAmpFactor;
        else
                vRight = vLeft;

        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 */
                for (i = 0; i < len-1; i+=2) {
                        *(bpc++) = (((*(mem++) - 128) * vLeft) >> 16) + 128;
                        *(bpc++) = (((*(mem++) - 128) * vRight) >> 16) + 128;
                }
                if (len % 2 == 1 && nChannels == 1)
                        *(bpc++) = (((*(mem++) - 128) * vLeft) >> 16) + 128;
                break;
        case 16:
                /* 16-bit WAV is signed -- much better */
                for (i = 0; i < len-3; i += 4) {
                        *(bps++) = (*(mems++) * vLeft) >> 16;
                        *(bps++) = (*(mems++) * vRight) >> 16;
                }
                if (len % 4 == 2 && nChannels == 1)
                        *(bps++) = ((INT)*(mems++) * vLeft) >> 16;
                break;
        }
        return dsb->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 len = fraglen, ilen;
        BYTE *ibuf = (dsb->tmp_buffer ? dsb->tmp_buffer : dsb->buffer->memory) + dsb->buf_mixpos, *volbuf;
        DWORD oldpos, mixbufpos;

        TRACE("buf_mixpos=%d/%d sec_mixpos=%d/%d\n", dsb->buf_mixpos, dsb->tmp_buffer_len, dsb->sec_mixpos, dsb->buflen);
        TRACE("(%p,%d,%d)\n",dsb,writepos,fraglen);

        assert(dsb->buf_mixpos + len <= dsb->tmp_buffer_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; /* data alignment */
        }

        /* Resample buffer to temporary buffer specifically allocated for this purpose, if needed */
        DSOUND_MixToTemporary(dsb, dsb->sec_mixpos, DSOUND_bufpos_to_secpos(dsb, dsb->buf_mixpos+len) - dsb->sec_mixpos, TRUE);
        if (dsb->resampleinmixer)
                ibuf = dsb->device->tmp_buffer;

        /* Apply volume if needed */
        volbuf = DSOUND_MixerVol(dsb, len);
        if (volbuf)
                ibuf = volbuf;

        mixbufpos = DSOUND_bufpos_to_mixpos(dsb->device, writepos);
        /* Now mix the temporary buffer into the devices main buffer */
        if ((writepos + len) <= dsb->device->buflen)
                dsb->device->mixfunction(ibuf, dsb->device->mix_buffer + mixbufpos, len);
        else
        {
                DWORD todo = dsb->device->buflen - writepos;
                dsb->device->mixfunction(ibuf, dsb->device->mix_buffer + mixbufpos, todo);
                dsb->device->mixfunction(ibuf + todo, dsb->device->mix_buffer, len - todo);
        }

        oldpos = dsb->sec_mixpos;
        dsb->buf_mixpos += len;

        if (dsb->buf_mixpos >= dsb->tmp_buffer_len) {
                if (dsb->buf_mixpos > dsb->tmp_buffer_len)
                        ERR("Mixpos (%u) past buflen (%u), capping...\n", dsb->buf_mixpos, dsb->tmp_buffer_len);
                if (dsb->playflags & DSBPLAY_LOOPING) {
                        dsb->buf_mixpos -= dsb->tmp_buffer_len;
                } else if (dsb->buf_mixpos >= dsb->tmp_buffer_len) {
                        dsb->buf_mixpos = dsb->sec_mixpos = 0;
                        dsb->state = STATE_STOPPED;
                }
                DSOUND_RecalcFreqAcc(dsb);
        }

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

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

/**
 * 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 writepos, DWORD mixlen)
{
        /* The buffer's primary_mixpos may be before or after the device
         * buffer's mixpos, but both must be ahead of writepos. */
        DWORD primary_done;

        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, leadin=%d, buflen=%d\n", dsb->playflags, dsb->leadin, dsb->tmp_buffer_len);

        /* If leading in, only mix about 20 ms, and 'skip' mixing the rest, for more fluid pointer advancement */
        if (dsb->leadin && dsb->state == STATE_STARTING)
        {
                if (mixlen > 2 * dsb->device->fraglen)
                {
                        dsb->primary_mixpos += mixlen - 2 * dsb->device->fraglen;
                        dsb->primary_mixpos %= dsb->device->buflen;
                }
        }
        dsb->leadin = FALSE;

        /* 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/%d (%d/%d), primary_mixpos=%d, writepos=%d, mixlen=%d\n", primary_done,dsb->buf_mixpos,dsb->tmp_buffer_len,dsb->sec_mixpos, dsb->buflen, dsb->primary_mixpos, writepos, mixlen);
                dsb->primary_mixpos = writepos + mixlen;
                dsb->primary_mixpos %= dsb->device->buflen;
                return mixlen;
        }

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

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

        if (!mixlen)
                return primary_done;

        /* First try to mix to the end of the buffer if possible
         * Theoretically it would allow for better optimization
        */
        if (mixlen + dsb->buf_mixpos >= dsb->tmp_buffer_len)
        {
                DWORD newmixed, mixfirst = dsb->tmp_buffer_len - dsb->buf_mixpos;
                newmixed = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixfirst);
                mixlen -= newmixed;

                if (dsb->playflags & DSBPLAY_LOOPING)
                        while (newmixed && mixlen)
                        {
                                mixfirst = (dsb->tmp_buffer_len < mixlen ? dsb->tmp_buffer_len : mixlen);
                                newmixed = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixfirst);
                                mixlen -= newmixed;
                        }
        }
        else DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixlen);

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

        TRACE("new primary_mixpos=%d, total mixed data=%d\n", dsb->primary_mixpos, primary_done);

        /* 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.
 *
 * 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) {
                        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 (dsb->state != STATE_STOPPED) {

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

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

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

                                if (!minlen) minlen = len;

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

                                *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_frames, buf_offs_bytes, wave_fragpos;
        int prebuf_frags;
        BYTE *buffer;
        HRESULT hr;

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

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

        /* calculate the current wave write position */
        buf_offs_bytes = wave_fragpos * device->fraglen;

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

        if (!force)
        {
                /* check remaining prebuffered frags */
                prebuf_frags = device->mixpos / device->fraglen;
                if (prebuf_frags == device->helfrags)
                        --prebuf_frags;
                TRACE("wave_fragpos = %d, mixpos_frags = %d\n", wave_fragpos, prebuf_frags);
                if (prebuf_frags < wave_fragpos)
                        prebuf_frags += device->helfrags;
                prebuf_frags -= wave_fragpos;
                TRACE("wanted prebuf_frags = %d\n", prebuf_frags);
        }
        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);

        if(!prebuf_frags)
                return;

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

        prebuf_frames = ((prebuf_frags + wave_fragpos > device->helfrags) ?
                        (prebuf_frags + wave_fragpos - device->helfrags) :
                        (prebuf_frags)) * device->fraglen / device->pwfx->nBlockAlign;

        hr = IAudioRenderClient_GetBuffer(device->render, prebuf_frames, &buffer);
        if(FAILED(hr)){
                WARN("GetBuffer failed: %08x\n", hr);
                return;
        }

        memcpy(buffer, device->buffer + buf_offs_bytes,
                        prebuf_frames * device->pwfx->nBlockAlign);

        hr = IAudioRenderClient_ReleaseBuffer(device->render, prebuf_frames, 0);
        if(FAILED(hr)){
                WARN("ReleaseBuffer failed: %08x\n", hr);
                return;
        }

        /* check if anything wrapped */
        prebuf_frags = prebuf_frags + wave_fragpos - device->helfrags;
        if(prebuf_frags > 0){
                prebuf_frames = prebuf_frags * device->fraglen / device->pwfx->nBlockAlign;

                hr = IAudioRenderClient_GetBuffer(device->render, prebuf_frames, &buffer);
                if(FAILED(hr)){
                        WARN("GetBuffer failed: %08x\n", hr);
                        return;
                }

                memcpy(buffer, device->buffer, prebuf_frames * device->pwfx->nBlockAlign);

                hr = IAudioRenderClient_ReleaseBuffer(device->render, prebuf_frames, 0);
                if(FAILED(hr)){
                        WARN("ReleaseBuffer failed: %08x\n", hr);
                        return;
                }
        }

        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)
{
        UINT64 clock_pos, clock_freq, pos_bytes;
        UINT delta_frags;
        HRESULT hr;

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

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

        hr = IAudioClock_GetFrequency(device->clock, &clock_freq);
        if(FAILED(hr)){
                WARN("GetFrequency failed: %08x\n", hr);
        LeaveCriticalSection(&device->mixlock);
                return;
        }

        hr = IAudioClock_GetPosition(device->clock, &clock_pos, NULL);
        if(FAILED(hr)){
                WARN("GetCurrentPadding failed: %08x\n", hr);
        LeaveCriticalSection(&device->mixlock);
                return;
        }

        pos_bytes = (clock_pos / (double)clock_freq) * device->pwfx->nSamplesPerSec *
                device->pwfx->nBlockAlign;

        delta_frags = (pos_bytes - device->last_pos_bytes) / device->fraglen;
        if(delta_frags > 0){
                device->pwplay += delta_frags;
                device->pwplay %= device->helfrags;
                device->pwqueue -= delta_frags;
                device->last_pos_bytes = pos_bytes - (pos_bytes % device->fraglen);
        }

        if (device->priolevel != DSSCL_WRITEPRIMARY) {
                BOOL recover = FALSE, all_stopped = FALSE;
                DWORD playpos, writepos, writelead, maxq, frag, prebuff_max, prebuff_left, size1, size2, mixplaypos, mixplaypos2;
                LPVOID buf1, buf2;
                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);

                mixplaypos = DSOUND_bufpos_to_mixpos(device, device->playpos);
                mixplaypos2 = DSOUND_bufpos_to_mixpos(device, playpos);

                /* calc maximum prebuff */
                prebuff_max = (device->prebuf * device->fraglen);
                if (playpos + prebuff_max >= device->helfrags * device->fraglen)
                        prebuff_max += device->buflen - device->helfrags * 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);

                /* check for underrun. underrun occurs when the write position passes the mix position
                 * also wipe out just-played sound data */
                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;

                        ZeroMemory(device->mix_buffer, device->mix_buffer_len);
                        ZeroMemory(device->buffer, device->buflen);
                } else if (playpos < device->playpos) {
                        buf1 = device->buffer + device->playpos;
                        buf2 = device->buffer;
                        size1 = device->buflen - device->playpos;
                        size2 = playpos;
                        FillMemory(device->mix_buffer + mixplaypos, device->mix_buffer_len - mixplaypos, 0);
                        FillMemory(device->mix_buffer, mixplaypos2, 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);
                } else {
                        buf1 = device->buffer + device->playpos;
                        buf2 = NULL;
                        size1 = playpos - device->playpos;
                        size2 = 0;
                        FillMemory(device->mix_buffer + mixplaypos, mixplaypos2 - mixplaypos, 0);
                        FillMemory(buf1, size1, nfiller);
                        if (buf2 && size2)
                        {
                                FIXME("%d: There should be no additional buffer here!!\n", __LINE__);
                                FillMemory(buf2, size2, nfiller);
                        }
                }
                device->playpos = 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);

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

                if (frag + writepos > device->buflen)
                {
                        DWORD todo = device->buflen - writepos;
                        device->normfunction(device->mix_buffer + DSOUND_bufpos_to_mixpos(device, writepos), device->buffer + writepos, todo);
                        device->normfunction(device->mix_buffer, device->buffer, frag - todo);
                }
                else
                        device->normfunction(device->mix_buffer + DSOUND_bufpos_to_mixpos(device, writepos), device->buffer + writepos, frag);

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

                /* 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 */
                        DSOUND_WaveQueue(device, FALSE);

                        /* 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 {

                DSOUND_WaveQueue(device, TRUE);

                /* 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);

        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);
}