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Sweden-Number/dlls/dsound/mixer.c

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/* 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 == 6 && ochannels == 2)
{
dsb->mix_channels = 6;
dsb->put = put_surround512stereo;
dsb->put_aux = putieee32_sum;
}
else if (ichannels == 4 && ochannels == 2)
{
dsb->mix_channels = 4;
dsb->put = put_quad2stereo;
dsb->put_aux = putieee32_sum;
}
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, *fir_copy, *itmp;
DWORD len = required_input * channels;
len += fir_cachesize;
len *= sizeof(float);
if (!dsb->device->cp_buffer) {
dsb->device->cp_buffer = HeapAlloc(GetProcessHeap(), 0, len);
dsb->device->cp_buffer_len = len;
} else if (len > dsb->device->cp_buffer_len) {
dsb->device->cp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->cp_buffer, len);
dsb->device->cp_buffer_len = len;
}
fir_copy = dsb->device->cp_buffer;
intermediate = fir_copy + 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.
*/
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;
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);
}
if(dsb->put_aux == putieee32_sum)
memset(dsb->device->tmp_buffer, 0, dsb->device->tmp_buffer_len);
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
* fraglen = number of bytes to mix
*/
static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, float *mix_buffer, DWORD frames)
{
float *ibuf;
DWORD oldpos;
TRACE("sec_mixpos=%d/%d\n", dsb->sec_mixpos, dsb->buflen);
TRACE("(%p, frames=%d)\n",dsb,frames);
/* 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, 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 frames;
}
/**
* 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)
* frames = the maximum number of frames in the primary buffer to mix, from the
* current writepos.
*
* Returns: the number of frames beyond the writepos that were mixed.
*/
static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, float *mix_buffer, DWORD frames)
{
DWORD primary_done = 0;
TRACE("(%p, frames=%d)\n",dsb,frames);
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 (frames > 2 * dsb->device->frag_frames) {
primary_done = frames - 2 * dsb->device->frag_frames;
frames = 2 * dsb->device->frag_frames;
dsb->sec_mixpos += primary_done *
dsb->pwfx->nBlockAlign * dsb->freqAdjustNum / dsb->freqAdjustDen;
}
}
dsb->leadin = FALSE;
TRACE("frames (primary) = %i\n", frames);
/* First try to mix to the end of the buffer if possible
* Theoretically it would allow for better optimization
*/
primary_done += DSOUND_MixInBuffer(dsb, mix_buffer, frames);
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.
*
* frames = the maximum amount to mix into the primary buffer
* 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, float *mix_buffer, DWORD frames, BOOL *all_stopped)
{
INT i;
IDirectSoundBufferImpl *dsb;
/* unless we find a running buffer, all have stopped */
*all_stopped = TRUE;
TRACE("(frames %d)\n", frames);
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, frames=%d\n", dsb, frames);
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, mix_buffer, frames);
*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, LPBYTE pos, DWORD bytes)
{
BYTE *buffer;
HRESULT hr;
TRACE("(%p)\n", device);
hr = IAudioRenderClient_GetBuffer(device->render, bytes / device->pwfx->nBlockAlign, &buffer);
if(FAILED(hr)){
WARN("GetBuffer failed: %08x\n", hr);
return;
}
memcpy(buffer, pos, bytes);
hr = IAudioRenderClient_ReleaseBuffer(device->render, bytes / device->pwfx->nBlockAlign, 0);
if(FAILED(hr)) {
ERR("ReleaseBuffer failed: %08x\n", hr);
IAudioRenderClient_ReleaseBuffer(device->render, 0, 0);
return;
}
device->pad += 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)
* =[Reformat]=> device->buffer (device format, skipped on float)
*/
static void DSOUND_PerformMix(DirectSoundDevice *device)
{
DWORD block, pad_frames, pad_bytes, frames;
HRESULT hr;
TRACE("(%p)\n", device);
/* **** */
EnterCriticalSection(&device->mixlock);
hr = IAudioClient_GetCurrentPadding(device->client, &pad_frames);
if(FAILED(hr)){
WARN("GetCurrentPadding failed: %08x\n", hr);
LeaveCriticalSection(&device->mixlock);
return;
}
block = device->pwfx->nBlockAlign;
pad_bytes = pad_frames * block;
device->playpos += device->pad - pad_bytes;
device->playpos %= device->buflen;
device->pad = pad_bytes;
frames = device->ac_frames - pad_frames;
if(!frames){
/* nothing to do! */
LeaveCriticalSection(&device->mixlock);
return;
}
if (frames > device->frag_frames * 3)
frames = device->frag_frames * 3;
if (device->priolevel != DSSCL_WRITEPRIMARY) {
BOOL all_stopped = FALSE;
int nfiller;
void *buffer = NULL;
/* the sound of silence */
nfiller = device->pwfx->wBitsPerSample == 8 ? 128 : 0;
/* check for underrun. underrun occurs when the write position passes the mix position
* also wipe out just-played sound data */
if (!pad_frames)
WARN("Probable buffer underrun\n");
hr = IAudioRenderClient_GetBuffer(device->render, frames, (BYTE **)&buffer);
if(FAILED(hr)){
WARN("GetBuffer failed: %08x\n", hr);
LeaveCriticalSection(&device->mixlock);
return;
}
memset(buffer, nfiller, frames * block);
if (!device->normfunction)
DSOUND_MixToPrimary(device, buffer, frames, &all_stopped);
else {
memset(device->buffer, nfiller, device->buflen);
/* do the mixing */
DSOUND_MixToPrimary(device, (float*)device->buffer, frames, &all_stopped);
device->normfunction(device->buffer, buffer, frames * device->pwfx->nChannels);
}
hr = IAudioRenderClient_ReleaseBuffer(device->render, frames, 0);
if(FAILED(hr))
ERR("ReleaseBuffer failed: %08x\n", hr);
device->pad += frames * block;
} else if (!device->stopped) {
DWORD writepos = (device->playpos + pad_bytes) % device->buflen;
DWORD bytes = frames * block;
if (bytes > device->buflen)
bytes = device->buflen;
if (writepos + bytes > device->buflen) {
DSOUND_WaveQueue(device, device->buffer + writepos, device->buflen - writepos);
DSOUND_WaveQueue(device, device->buffer, writepos + bytes - device->buflen);
} else
DSOUND_WaveQueue(device, device->buffer + writepos, bytes);
}
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));
}
return 0;
}
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