hidclass.sys: Break the report descriptor into multiple lines.

[wine.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

#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"
#include "fir.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 */

        /* FIXME: use calculated vol and pan ampfactors */
        temp = (double) (volpan->lVolume - (volpan->lPan > 0 ? volpan->lPan : 0));
        volpan->dwTotalAmpFactor[0] = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
        temp = (double) (volpan->lVolume + (volpan->lPan < 0 ? volpan->lPan : 0));
        volpan->dwTotalAmpFactor[1] = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);

        TRACE("left = %x, right = %x\n", volpan->dwTotalAmpFactor[0], volpan->dwTotalAmpFactor[1]);
}

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

    TRACE("left=%x, right=%x\n",volpan->dwTotalAmpFactor[0],volpan->dwTotalAmpFactor[1]);
    if (volpan->dwTotalAmpFactor[0]==0)
        left=-10000;
    else
        left=600 * log(((double)volpan->dwTotalAmpFactor[0]) / 0xffff) / log(2);
    if (volpan->dwTotalAmpFactor[1]==0)
        right=-10000;
    else
        right=600 * log(((double)volpan->dwTotalAmpFactor[1]) / 0xffff) / log(2);
    if (left<right)
        volpan->lVolume=right;
    else
        volpan->lVolume=left;
    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);
}

/**
 * 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
 */
void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
{
        DWORD ichannels = dsb->pwfx->nChannels;
        DWORD ochannels = dsb->device->pwfx->nChannels;
        WAVEFORMATEXTENSIBLE *pwfxe;
        BOOL ieee = FALSE;

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

        pwfxe = (WAVEFORMATEXTENSIBLE *) dsb->pwfx;
        dsb->freqAdjustNum = dsb->freq;
        dsb->freqAdjustDen = dsb->device->pwfx->nSamplesPerSec;

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

        /**
         * Recalculate FIR step and gain.
         *
         * firstep says how many points of the FIR exist per one
         * sample in the secondary buffer. firgain specifies what
         * to multiply the FIR output by in order to attenuate it correctly.
         */
        if (dsb->freqAdjustNum / dsb->freqAdjustDen > 0) {
                /**
                 * Yes, round it a bit to make sure that the
                 * linear interpolation factor never changes.
                 */
                dsb->firstep = fir_step * dsb->freqAdjustDen / dsb->freqAdjustNum;
        } else {
                dsb->firstep = fir_step;
        }
        dsb->firgain = (float)dsb->firstep / fir_step;

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

        dsb->freqAccNum = 0;

        dsb->get_aux = ieee ? getbpp[4] : getbpp[dsb->pwfx->wBitsPerSample/8 - 1];
        dsb->put_aux = putieee32;

        dsb->get = dsb->get_aux;
        dsb->put = dsb->put_aux;

        if (ichannels == ochannels)
        {
                dsb->mix_channels = ichannels;
                if (ichannels > 32) {
                        FIXME("Copying %u channels is unsupported, limiting to first 32\n", ichannels);
                        dsb->mix_channels = 32;
                }
        }
        else if (ichannels == 1)
        {
                dsb->mix_channels = 1;

                if (ochannels == 2)
                        dsb->put = put_mono2stereo;
                else if (ochannels == 4)
                        dsb->put = put_mono2quad;
                else if (ochannels == 6)
                        dsb->put = put_mono2surround51;
        }
        else if (ochannels == 1)
        {
                dsb->mix_channels = 1;
                dsb->get = get_mono;
        }
        else if (ichannels == 2 && ochannels == 4)
        {
                dsb->mix_channels = 2;
                dsb->put = put_stereo2quad;
        }
        else if (ichannels == 2 && ochannels == 6)
        {
                dsb->mix_channels = 2;
                dsb->put = put_stereo2surround51;
        }
        else
        {
                if (ichannels > 2)
                        FIXME("Conversion from %u to %u channels is not implemented, falling back to stereo\n", ichannels, ochannels);
                dsb->mix_channels = 2;
        }
}

/**
 * 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 first, left, right, check;

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

    if(dsb->state == STATE_STOPPED){
        TRACE("Stopped...\n");
        /* DSBPN_OFFSETSTOP notifies are always at the start of the sorted array */
        for(left = 0; left < dsb->nrofnotifies; ++left){
            if(dsb->notifies[left].dwOffset != DSBPN_OFFSETSTOP)
                break;

            TRACE("Signalling %p\n", dsb->notifies[left].hEventNotify);
            SetEvent(dsb->notifies[left].hEventNotify);
        }
        return;
    }

    for(first = 0; first < dsb->nrofnotifies && dsb->notifies[first].dwOffset == DSBPN_OFFSETSTOP; ++first)
        ;

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

    check = left = first;
    right = dsb->nrofnotifies - 1;

    /* find leftmost notify that is greater than playpos */
    while(left != right){
        check = left + (right - left) / 2;
        if(dsb->notifies[check].dwOffset < playpos)
            left = check + 1;
        else if(dsb->notifies[check].dwOffset > playpos)
            right = check;
        else{
            left = check;
            break;
        }
    }

    TRACE("Not stopped: first notify: %u (%u), left notify: %u (%u), range: [%u,%u)\n",
            first, dsb->notifies[first].dwOffset,
            left, dsb->notifies[left].dwOffset,
            playpos, (playpos + len) % dsb->buflen);

    /* send notifications in range */
    if(dsb->notifies[left].dwOffset >= playpos){
        for(check = left; check < dsb->nrofnotifies; ++check){
            if(dsb->notifies[check].dwOffset >= playpos + len)
                break;

            TRACE("Signalling %p (%u)\n", dsb->notifies[check].hEventNotify, dsb->notifies[check].dwOffset);
            SetEvent(dsb->notifies[check].hEventNotify);
        }
    }

    if(playpos + len > dsb->buflen){
        for(check = first; check < left; ++check){
            if(dsb->notifies[check].dwOffset >= (playpos + len) % dsb->buflen)
                break;

            TRACE("Signalling %p (%u)\n", dsb->notifies[check].hEventNotify, dsb->notifies[check].dwOffset);
            SetEvent(dsb->notifies[check].hEventNotify);
        }
    }
}

static inline float get_current_sample(const IDirectSoundBufferImpl *dsb,
        DWORD mixpos, DWORD channel)
{
    if (mixpos >= dsb->buflen && !(dsb->playflags & DSBPLAY_LOOPING))
        return 0.0f;
    return dsb->get(dsb, mixpos % dsb->buflen, channel);
}

static UINT cp_fields_noresample(IDirectSoundBufferImpl *dsb, UINT count)
{
    UINT istride = dsb->pwfx->nBlockAlign;
    UINT ostride = dsb->device->pwfx->nChannels * sizeof(float);
    DWORD channel, i;
    for (i = 0; i < count; i++)
        for (channel = 0; channel < dsb->mix_channels; channel++)
            dsb->put(dsb, i * ostride, channel, get_current_sample(dsb,
                    dsb->sec_mixpos + i * istride, channel));
    return count;
}

static UINT cp_fields_resample(IDirectSoundBufferImpl *dsb, UINT count, LONG64 *freqAccNum)
{
    UINT i, channel;
    UINT istride = dsb->pwfx->nBlockAlign;
    UINT ostride = dsb->device->pwfx->nChannels * sizeof(float);

    LONG64 freqAcc_start = *freqAccNum;
    LONG64 freqAcc_end = freqAcc_start + count * dsb->freqAdjustNum;
    UINT dsbfirstep = dsb->firstep;
    UINT channels = dsb->mix_channels;
    UINT max_ipos = (freqAcc_start + count * dsb->freqAdjustNum) / dsb->freqAdjustDen;

    UINT fir_cachesize = (fir_len + dsbfirstep - 2) / dsbfirstep;
    UINT required_input = max_ipos + fir_cachesize;

    float* intermediate = HeapAlloc(GetProcessHeap(), 0,
            sizeof(float) * required_input * channels);

    float* fir_copy = HeapAlloc(GetProcessHeap(), 0,
            sizeof(float) * fir_cachesize);

    /* Important: this buffer MUST be non-interleaved
     * if you want -msse3 to have any effect.
     * This is good for CPU cache effects, too.
     */
    float* itmp = intermediate;
    for (channel = 0; channel < channels; channel++)
        for (i = 0; i < required_input; i++)
            *(itmp++) = get_current_sample(dsb,
                    dsb->sec_mixpos + i * istride, channel);

    for(i = 0; i < count; ++i) {
        UINT int_fir_steps = (freqAcc_start + i * dsb->freqAdjustNum) * dsbfirstep / dsb->freqAdjustDen;
        float total_fir_steps = (freqAcc_start + i * dsb->freqAdjustNum) * dsbfirstep / (float)dsb->freqAdjustDen;
        UINT ipos = int_fir_steps / dsbfirstep;

        UINT idx = (ipos + 1) * dsbfirstep - int_fir_steps - 1;
        float rem = int_fir_steps + 1.0 - total_fir_steps;

        int fir_used = 0;
        while (idx < fir_len - 1) {
            fir_copy[fir_used++] = fir[idx] * (1.0 - rem) + fir[idx + 1] * rem;
            idx += dsb->firstep;
        }

        assert(fir_used <= fir_cachesize);
        assert(ipos + fir_used <= required_input);

        for (channel = 0; channel < dsb->mix_channels; channel++) {
            int j;
            float sum = 0.0;
            float* cache = &intermediate[channel * required_input + ipos];
            for (j = 0; j < fir_used; j++)
                sum += fir_copy[j] * cache[j];
            dsb->put(dsb, i * ostride, channel, sum * dsb->firgain);
        }
    }

    *freqAccNum = freqAcc_end % dsb->freqAdjustDen;

    HeapFree(GetProcessHeap(), 0, fir_copy);
    HeapFree(GetProcessHeap(), 0, intermediate);

    return max_ipos;
}

static void cp_fields(IDirectSoundBufferImpl *dsb, UINT count, LONG64 *freqAccNum)
{
    DWORD ipos, adv;

    if (dsb->freqAdjustNum == dsb->freqAdjustDen)
        adv = cp_fields_noresample(dsb, count); /* *freqAccNum is unmodified */
    else
        adv = cp_fields_resample(dsb, count, freqAccNum);

    ipos = dsb->sec_mixpos + adv * dsb->pwfx->nBlockAlign;
    if (ipos >= dsb->buflen) {
        if (dsb->playflags & DSBPLAY_LOOPING)
            ipos %= dsb->buflen;
        else {
            ipos = 0;
            dsb->state = STATE_STOPPED;
        }
    }

    dsb->sec_mixpos = ipos;
}

/**
 * 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.
 */
static void DSOUND_MixToTemporary(IDirectSoundBufferImpl *dsb, DWORD frames)
{
        UINT size_bytes = frames * sizeof(float) * dsb->device->pwfx->nChannels;
        HRESULT hr;
        int i;

        if (dsb->device->tmp_buffer_len < size_bytes || !dsb->device->tmp_buffer)
        {
                dsb->device->tmp_buffer_len = size_bytes;
                if (dsb->device->tmp_buffer)
                        dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, size_bytes);
                else
                        dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, size_bytes);
        }

        cp_fields(dsb, frames, &dsb->freqAccNum);

        if (size_bytes > 0) {
                for (i = 0; i < dsb->num_filters; i++) {
                        if (dsb->filters[i].inplace) {
                                hr = IMediaObjectInPlace_Process(dsb->filters[i].inplace, size_bytes, (BYTE*)dsb->device->tmp_buffer, 0, DMO_INPLACE_NORMAL);

                                if (FAILED(hr))
                                        WARN("IMediaObjectInPlace_Process failed for filter %u\n", i);
                        } else
                                WARN("filter %u has no inplace object - unsupported\n", i);
                }
        }
}

static void DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, INT frames)
{
        INT     i;
        float vols[DS_MAX_CHANNELS];
        UINT channels = dsb->device->pwfx->nChannels, chan;

        TRACE("(%p,%d)\n",dsb,frames);
        TRACE("left = %x, right = %x\n", dsb->volpan.dwTotalAmpFactor[0],
                dsb->volpan.dwTotalAmpFactor[1]);

        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 (channels > DS_MAX_CHANNELS)
        {
                FIXME("There is no support for %u channels\n", channels);
                return;
        }

        for (i = 0; i < channels; ++i)
                vols[i] = dsb->volpan.dwTotalAmpFactor[i] / ((float)0xFFFF);

        for(i = 0; i < frames; ++i){
                for(chan = 0; chan < channels; ++chan){
                        dsb->device->tmp_buffer[i * channels + chan] *= vols[chan];
                }
        }
}

/**
 * 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;
        float *ibuf;
        DWORD oldpos;
        UINT frames = fraglen / dsb->device->pwfx->nBlockAlign;

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

        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 */
        oldpos = dsb->sec_mixpos;

        DSOUND_MixToTemporary(dsb, frames);
        ibuf = dsb->device->tmp_buffer;

        /* Apply volume if needed */
        DSOUND_MixerVol(dsb, frames);

        mixieee32(ibuf, dsb->device->mix_buffer, frames * dsb->device->pwfx->nChannels);

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

        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)
{
        DWORD primary_done = 0;

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

        /* If leading in, only mix about 20 ms, and 'skip' mixing the rest, for more fluid pointer advancement */
        /* FIXME: Is this needed? */
        if (dsb->leadin && dsb->state == STATE_STARTING) {
                if (mixlen > 2 * dsb->device->fraglen) {
                        primary_done = mixlen - 2 * dsb->device->fraglen;
                        mixlen = 2 * dsb->device->fraglen;
                        writepos += primary_done;
                        dsb->sec_mixpos += (primary_done / dsb->device->pwfx->nBlockAlign) *
                                dsb->pwfx->nBlockAlign * dsb->freqAdjustNum / dsb->freqAdjustDen;
                }
        }

        dsb->leadin = FALSE;

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

        /* First try to mix to the end of the buffer if possible
         * Theoretically it would allow for better optimization
        */
        primary_done += DSOUND_MixInBuffer(dsb, writepos, mixlen);

        TRACE("total mixed data=%d\n", 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 void DSOUND_MixToPrimary(const DirectSoundDevice *device, DWORD writepos, DWORD mixlen, BOOL recover, BOOL *all_stopped)
{
        INT i;
        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 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 */
                                DSOUND_MixOne(dsb, writepos, mixlen);

                                *all_stopped = FALSE;
                        }
                        RtlReleaseResource(&dsb->lock);
                }
        }
}

/**
 * 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, prebuf_bytes, read_offs_bytes;
        BYTE *buffer;
        HRESULT hr;

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

        read_offs_bytes = (device->playing_offs_bytes + device->in_mmdev_bytes) % device->buflen;

        TRACE("read_offs_bytes = %u, playing_offs_bytes = %u, in_mmdev_bytes: %u, prebuf = %u\n",
                read_offs_bytes, device->playing_offs_bytes, device->in_mmdev_bytes, device->prebuf);

        if (!force)
        {
                if(device->mixpos < device->playing_offs_bytes)
                        prebuf_bytes = device->mixpos + device->buflen - device->playing_offs_bytes;
                else
                        prebuf_bytes = device->mixpos - device->playing_offs_bytes;
        }
        else
                /* buffer the maximum amount of frags */
                prebuf_bytes = device->prebuf * device->fraglen;

        /* limit to the queue we have left */
        if(device->in_mmdev_bytes + prebuf_bytes > device->prebuf * device->fraglen)
                prebuf_bytes = device->prebuf * device->fraglen - device->in_mmdev_bytes;

        TRACE("prebuf_bytes = %u\n", prebuf_bytes);

        if(!prebuf_bytes)
                return;

        if(prebuf_bytes + read_offs_bytes > device->buflen){
                DWORD chunk_bytes = device->buflen - read_offs_bytes;
                prebuf_frames = chunk_bytes / device->pwfx->nBlockAlign;
                prebuf_bytes -= chunk_bytes;
        }else{
                prebuf_frames = prebuf_bytes / device->pwfx->nBlockAlign;
                prebuf_bytes = 0;
        }

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

        memcpy(buffer, device->buffer + read_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;
        }

        device->in_mmdev_bytes += prebuf_frames * device->pwfx->nBlockAlign;

        /* check if anything wrapped */
        if(prebuf_bytes > 0){
                prebuf_frames = prebuf_bytes / 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;
                }
                device->in_mmdev_bytes += prebuf_frames * device->pwfx->nBlockAlign;
        }

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

/**
 * 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).
 *
 * The mixing procedure goes:
 *
 * secondary->buffer (secondary format)
 *   =[Resample]=> device->tmp_buffer (float format)
 *   =[Volume]=> device->tmp_buffer (float format)
 *   =[Mix]=> device->mix_buffer (float format)
 *   =[Reformat]=> device->buffer (device format)
 */
static void DSOUND_PerformMix(DirectSoundDevice *device)
{
        UINT32 pad, to_mix_frags, to_mix_bytes;
        HRESULT hr;

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

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

        hr = IAudioClient_GetCurrentPadding(device->client, &pad);
        if(FAILED(hr)){
                WARN("GetCurrentPadding failed: %08x\n", hr);
                LeaveCriticalSection(&device->mixlock);
                return;
        }

        to_mix_frags = device->prebuf - (pad * device->pwfx->nBlockAlign + device->fraglen - 1) / device->fraglen;

        to_mix_bytes = to_mix_frags * device->fraglen;

        if(device->in_mmdev_bytes > 0){
                DWORD delta_bytes = min(to_mix_bytes, device->in_mmdev_bytes);
                device->in_mmdev_bytes -= delta_bytes;
                device->playing_offs_bytes += delta_bytes;
                device->playing_offs_bytes %= device->buflen;
        }

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

                /* 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);

                /* 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->buffer, device->buflen);
                } else if (playpos < device->playpos) {
                        buf1 = device->buffer + device->playpos;
                        buf2 = device->buffer;
                        size1 = device->buflen - device->playpos;
                        size2 = playpos;
                        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(buf1, size1, 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);

                ZeroMemory(device->mix_buffer, device->mix_buffer_len);

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

                if (maxq + writepos > device->buflen)
                {
                        DWORD todo = device->buflen - writepos;
                        DWORD offs_float = (todo / device->pwfx->nBlockAlign) * device->pwfx->nChannels;
                        device->normfunction(device->mix_buffer, device->buffer + writepos, todo);
                        device->normfunction(device->mix_buffer + offs_float, device->buffer, maxq - todo);
                }
                else
                        device->normfunction(device->mix_buffer, device->buffer + writepos, maxq);

                /* update the mix position, taking wrap-around into account */
                device->mixpos = writepos + maxq;
                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 && (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 if (device->state != STATE_STOPPED) {

                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));
        /* **** */
}

DWORD CALLBACK DSOUND_mixthread(void *p)
{
        DirectSoundDevice *dev = p;
        TRACE("(%p)\n", dev);

        while (dev->ref) {
                DWORD ret;

                /*
                 * Some audio drivers are retarded and won't fire after being
                 * stopped, add a timeout to handle this.
                 */
                ret = WaitForSingleObject(dev->sleepev, dev->sleeptime);
                if (ret == WAIT_FAILED)
                        WARN("wait returned error %u %08x!\n", GetLastError(), GetLastError());
                else if (ret != WAIT_OBJECT_0)
                        WARN("wait returned %08x!\n", ret);
                if (!dev->ref)
                        break;

                RtlAcquireResourceShared(&(dev->buffer_list_lock), TRUE);
                DSOUND_PerformMix(dev);
                RtlReleaseResource(&(dev->buffer_list_lock));
        }
        SetEvent(dev->thread_finished);
        return 0;
}