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wine/dlls/dsound/mixer.c
Andrew Talbot 1e4c97175e dsound: Exclude unused headers.
2007-05-03 14:23:53 +02:00

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/* DirectSound
*
* Copyright 1998 Marcus Meissner
* Copyright 1998 Rob Riggs
* Copyright 2000-2002 TransGaming Technologies, Inc.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
*/
#include <assert.h>
#include <stdarg.h>
#include <math.h> /* Insomnia - pow() function */
#define NONAMELESSSTRUCT
#define NONAMELESSUNION
#include "windef.h"
#include "winbase.h"
#include "winuser.h"
#include "mmsystem.h"
#include "winternl.h"
#include "wine/debug.h"
#include "dsound.h"
#include "dsdriver.h"
#include "dsound_private.h"
WINE_DEFAULT_DEBUG_CHANNEL(dsound);
void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan)
{
double temp;
TRACE("(%p)\n",volpan);
TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
/* the AmpFactors are expressed in 16.16 fixed point */
volpan->dwVolAmpFactor = (ULONG) (pow(2.0, volpan->lVolume / 600.0) * 0xffff);
/* FIXME: dwPan{Left|Right}AmpFactor */
/* FIXME: use calculated vol and pan ampfactors */
temp = (double) (volpan->lVolume - (volpan->lPan > 0 ? volpan->lPan : 0));
volpan->dwTotalLeftAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
temp = (double) (volpan->lVolume + (volpan->lPan < 0 ? volpan->lPan : 0));
volpan->dwTotalRightAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
TRACE("left = %x, right = %x\n", volpan->dwTotalLeftAmpFactor, volpan->dwTotalRightAmpFactor);
}
void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan)
{
double left,right;
TRACE("(%p)\n",volpan);
TRACE("left=%x, right=%x\n",volpan->dwTotalLeftAmpFactor,volpan->dwTotalRightAmpFactor);
if (volpan->dwTotalLeftAmpFactor==0)
left=-10000;
else
left=600 * log(((double)volpan->dwTotalLeftAmpFactor) / 0xffff) / log(2);
if (volpan->dwTotalRightAmpFactor==0)
right=-10000;
else
right=600 * log(((double)volpan->dwTotalRightAmpFactor) / 0xffff) / log(2);
if (left<right)
{
volpan->lVolume=right;
volpan->dwVolAmpFactor=volpan->dwTotalRightAmpFactor;
}
else
{
volpan->lVolume=left;
volpan->dwVolAmpFactor=volpan->dwTotalLeftAmpFactor;
}
if (volpan->lVolume < -10000)
volpan->lVolume=-10000;
volpan->lPan=right-left;
if (volpan->lPan < -10000)
volpan->lPan=-10000;
TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
}
void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
{
TRACE("(%p)\n",dsb);
/* calculate the 10ms write lead */
dsb->writelead = (dsb->freq / 100) * dsb->pwfx->nBlockAlign;
}
/**
* Check for application callback requests for when the play position
* reaches certain points.
*
* The offsets that will be triggered will be those between the recorded
* "last played" position for the buffer (i.e. dsb->playpos) and "len" bytes
* beyond that position.
*/
void DSOUND_CheckEvent(IDirectSoundBufferImpl *dsb, int len)
{
int i;
DWORD offset;
LPDSBPOSITIONNOTIFY event;
TRACE("(%p,%d)\n",dsb,len);
if (dsb->nrofnotifies == 0)
return;
TRACE("(%p) buflen = %d, playpos = %d, len = %d\n",
dsb, dsb->buflen, dsb->playpos, len);
for (i = 0; i < dsb->nrofnotifies ; i++) {
event = dsb->notifies + i;
offset = event->dwOffset;
TRACE("checking %d, position %d, event = %p\n",
i, offset, event->hEventNotify);
/* DSBPN_OFFSETSTOP has to be the last element. So this is */
/* OK. [Inside DirectX, p274] */
/* */
/* This also means we can't sort the entries by offset, */
/* because DSBPN_OFFSETSTOP == -1 */
if (offset == DSBPN_OFFSETSTOP) {
if (dsb->state == STATE_STOPPED) {
SetEvent(event->hEventNotify);
TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
return;
} else
return;
}
if ((dsb->playpos + len) >= dsb->buflen) {
if ((offset < ((dsb->playpos + len) % dsb->buflen)) ||
(offset >= dsb->playpos)) {
TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
SetEvent(event->hEventNotify);
}
} else {
if ((offset >= dsb->playpos) && (offset < (dsb->playpos + len))) {
TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
SetEvent(event->hEventNotify);
}
}
}
}
/* WAV format info can be found at:
*
* http://www.cwi.nl/ftp/audio/AudioFormats.part2
* ftp://ftp.cwi.nl/pub/audio/RIFF-format
*
* Import points to remember:
* 8-bit WAV is unsigned
* 16-bit WAV is signed
*/
/* Use the same formulas as pcmconverter.c */
static inline INT16 cvtU8toS16(BYTE b)
{
return (short)((b+(b << 8))-32768);
}
static inline BYTE cvtS16toU8(INT16 s)
{
return (s >> 8) ^ (unsigned char)0x80;
}
/**
* Copy a single frame from the given input buffer to the given output buffer.
* Translate 8 <-> 16 bits and mono <-> stereo
*/
static inline void cp_fields(const IDirectSoundBufferImpl *dsb, const BYTE *ibuf, BYTE *obuf )
{
DirectSoundDevice * device = dsb->device;
INT fl,fr;
if (dsb->pwfx->wBitsPerSample == 8) {
if (device->pwfx->wBitsPerSample == 8 &&
device->pwfx->nChannels == dsb->pwfx->nChannels) {
/* avoid needless 8->16->8 conversion */
*obuf=*ibuf;
if (dsb->pwfx->nChannels==2)
*(obuf+1)=*(ibuf+1);
return;
}
fl = cvtU8toS16(*ibuf);
fr = (dsb->pwfx->nChannels==2 ? cvtU8toS16(*(ibuf + 1)) : fl);
} else {
fl = *((const INT16 *)ibuf);
fr = (dsb->pwfx->nChannels==2 ? *(((const INT16 *)ibuf) + 1) : fl);
}
if (device->pwfx->nChannels == 2) {
if (device->pwfx->wBitsPerSample == 8) {
*obuf = cvtS16toU8(fl);
*(obuf + 1) = cvtS16toU8(fr);
return;
}
if (device->pwfx->wBitsPerSample == 16) {
*((INT16 *)obuf) = fl;
*(((INT16 *)obuf) + 1) = fr;
return;
}
}
if (device->pwfx->nChannels == 1) {
fl = (fl + fr) >> 1;
if (device->pwfx->wBitsPerSample == 8) {
*obuf = cvtS16toU8(fl);
return;
}
if (device->pwfx->wBitsPerSample == 16) {
*((INT16 *)obuf) = fl;
return;
}
}
}
/**
* Mix at most the given amount of data into the given device buffer from the
* given secondary buffer, starting from the dsb's first currently unmixed
* frame (buf_mixpos), translating frequency (pitch), stereo/mono and
* bits-per-sample. The secondary buffer sample is looped if it is not
* long enough and it is a looping buffer.
* (Doesn't perform any mixing - this is a straight copy operation).
*
* Now with PerfectPitch (tm) technology
*
* dsb = the secondary buffer
* buf = the device buffer
* len = number of bytes to store in the device buffer
*
* Returns: the number of bytes read from the secondary buffer
* (ie. len, adjusted for frequency, number of channels and sample size,
* and limited by buffer length for non-looping buffers)
*/
static INT DSOUND_MixerNorm(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len)
{
INT i, size, ipos, ilen;
BYTE *ibp, *obp;
INT iAdvance = dsb->pwfx->nBlockAlign;
INT oAdvance = dsb->device->pwfx->nBlockAlign;
ibp = dsb->buffer->memory + dsb->buf_mixpos;
obp = buf;
TRACE("(%p, %p, %p), buf_mixpos=%d\n", dsb, ibp, obp, dsb->buf_mixpos);
/* Check for the best case */
if ((dsb->freq == dsb->device->pwfx->nSamplesPerSec) &&
(dsb->pwfx->wBitsPerSample == dsb->device->pwfx->wBitsPerSample) &&
(dsb->pwfx->nChannels == dsb->device->pwfx->nChannels)) {
INT bytesleft = dsb->buflen - dsb->buf_mixpos;
TRACE("(%p) Best case\n", dsb);
if (len <= bytesleft )
CopyMemory(obp, ibp, len);
else { /* wrap */
CopyMemory(obp, ibp, bytesleft);
CopyMemory(obp + bytesleft, dsb->buffer->memory, len - bytesleft);
}
return len;
}
/* Check for same sample rate */
if (dsb->freq == dsb->device->pwfx->nSamplesPerSec) {
TRACE("(%p) Same sample rate %d = primary %d\n", dsb,
dsb->freq, dsb->device->pwfx->nSamplesPerSec);
ilen = 0;
for (i = 0; i < len; i += oAdvance) {
cp_fields(dsb, ibp, obp );
ibp += iAdvance;
ilen += iAdvance;
obp += oAdvance;
if (ibp >= (BYTE *)(dsb->buffer->memory + dsb->buflen))
ibp = dsb->buffer->memory; /* wrap */
}
return (ilen);
}
/* Mix in different sample rates */
/* */
/* New PerfectPitch(tm) Technology (c) 1998 Rob Riggs */
/* Patent Pending :-] */
/* Patent enhancements (c) 2000 Ove K<>ven,
* TransGaming Technologies Inc. */
/* FIXME("(%p) Adjusting frequency: %ld -> %ld (need optimization)\n",
dsb, dsb->freq, dsb->device->pwfx->nSamplesPerSec); */
size = len / oAdvance;
ilen = 0;
ipos = dsb->buf_mixpos;
for (i = 0; i < size; i++) {
cp_fields(dsb, (dsb->buffer->memory + ipos), obp);
obp += oAdvance;
dsb->freqAcc += dsb->freqAdjust;
if (dsb->freqAcc >= (1<<DSOUND_FREQSHIFT)) {
ULONG adv = (dsb->freqAcc>>DSOUND_FREQSHIFT) * iAdvance;
dsb->freqAcc &= (1<<DSOUND_FREQSHIFT)-1;
ipos += adv; ilen += adv;
ipos %= dsb->buflen;
}
}
return ilen;
}
static void DSOUND_MixerVol(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len)
{
INT i;
BYTE *bpc = buf;
INT16 *bps = (INT16 *) buf;
TRACE("(%p,%p,%d)\n",dsb,buf,len);
TRACE("left = %x, right = %x\n", dsb->cvolpan.dwTotalLeftAmpFactor,
dsb->cvolpan.dwTotalRightAmpFactor);
if ((!(dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) || (dsb->cvolpan.lPan == 0)) &&
(!(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) || (dsb->cvolpan.lVolume == 0)) &&
!(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
return; /* Nothing to do */
/* If we end up with some bozo coder using panning or 3D sound */
/* with a mono primary buffer, it could sound very weird using */
/* this method. Oh well, tough patooties. */
switch (dsb->device->pwfx->wBitsPerSample) {
case 8:
/* 8-bit WAV is unsigned, but we need to operate */
/* on signed data for this to work properly */
switch (dsb->device->pwfx->nChannels) {
case 1:
for (i = 0; i < len; i++) {
INT val = *bpc - 128;
val = (val * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
*bpc = val + 128;
bpc++;
}
break;
case 2:
for (i = 0; i < len; i+=2) {
INT val = *bpc - 128;
val = (val * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
*bpc++ = val + 128;
val = *bpc - 128;
val = (val * dsb->cvolpan.dwTotalRightAmpFactor) >> 16;
*bpc = val + 128;
bpc++;
}
break;
default:
FIXME("doesn't support %d channels\n", dsb->device->pwfx->nChannels);
break;
}
break;
case 16:
/* 16-bit WAV is signed -- much better */
switch (dsb->device->pwfx->nChannels) {
case 1:
for (i = 0; i < len; i += 2) {
*bps = (*bps * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
bps++;
}
break;
case 2:
for (i = 0; i < len; i += 4) {
*bps = (*bps * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
bps++;
*bps = (*bps * dsb->cvolpan.dwTotalRightAmpFactor) >> 16;
bps++;
}
break;
default:
FIXME("doesn't support %d channels\n", dsb->device->pwfx->nChannels);
break;
}
break;
default:
FIXME("doesn't support %d bit samples\n", dsb->device->pwfx->wBitsPerSample);
break;
}
}
/**
* Make sure the device's tmp_buffer is at least the given size. Return a
* pointer to it.
*/
static LPBYTE DSOUND_tmpbuffer(DirectSoundDevice *device, DWORD len)
{
TRACE("(%p,%d)\n", device, len);
if (len > device->tmp_buffer_len) {
if (device->tmp_buffer)
device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, device->tmp_buffer, len);
else
device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, len);
device->tmp_buffer_len = len;
}
return device->tmp_buffer;
}
/**
* Mix (at most) the given number of bytes into the given position of the
* device buffer, from the secondary buffer "dsb" (starting at the current
* mix position for that buffer).
*
* Returns the number of bytes actually mixed into the device buffer. This
* will match fraglen unless the end of the secondary buffer is reached
* (and it is not looping).
*
* dsb = the secondary buffer to mix from
* writepos = position (offset) in device buffer to write at
* fraglen = number of bytes to mix
*/
static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen)
{
INT i, len, ilen, field, todo;
BYTE *buf, *ibuf;
TRACE("(%p,%d,%d)\n",dsb,writepos,fraglen);
len = fraglen;
if (!(dsb->playflags & DSBPLAY_LOOPING)) {
/* This buffer is not looping, so make sure the requested
* length will not take us past the end of the buffer */
int secondary_remainder = dsb->buflen - dsb->buf_mixpos;
int adjusted_remainder = MulDiv(dsb->device->pwfx->nAvgBytesPerSec, secondary_remainder, dsb->nAvgBytesPerSec);
assert(adjusted_remainder >= 0);
/* The adjusted remainder must be at least one sample,
* otherwise we will never reach the end of the
* secondary buffer, as there will perpetually be a
* fractional remainder */
TRACE("secondary_remainder = %d, adjusted_remainder = %d, len = %d\n", secondary_remainder, adjusted_remainder, len);
if (adjusted_remainder < len) {
TRACE("clipping len to remainder of secondary buffer\n");
len = adjusted_remainder;
}
if (len == 0)
return 0;
}
if (len % dsb->device->pwfx->nBlockAlign) {
INT nBlockAlign = dsb->device->pwfx->nBlockAlign;
ERR("length not a multiple of block size, len = %d, block size = %d\n", len, nBlockAlign);
len = (len / nBlockAlign) * nBlockAlign; /* data alignment */
}
if ((buf = ibuf = DSOUND_tmpbuffer(dsb->device, len)) == NULL)
return 0;
TRACE("MixInBuffer (%p) len = %d, dest = %d\n", dsb, len, writepos);
/* first, copy the data from the DirectSoundBuffer into the temporary
buffer, translating frequency/bits-per-sample/number-of-channels
to match the device settings */
ilen = DSOUND_MixerNorm(dsb, ibuf, len);
if ((dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) ||
(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) ||
(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
DSOUND_MixerVol(dsb, ibuf, len);
/* Now mix the temporary buffer into the devices main buffer */
if (dsb->device->pwfx->wBitsPerSample == 8) {
BYTE *obuf = dsb->device->buffer + writepos;
if ((writepos + len) <= dsb->device->buflen)
todo = len;
else
todo = dsb->device->buflen - writepos;
for (i = 0; i < todo; i++) {
/* 8-bit WAV is unsigned */
field = (*ibuf++ - 128);
field += (*obuf - 128);
if (field > 127) field = 127;
else if (field < -128) field = -128;
*obuf++ = field + 128;
}
if (todo < len) {
todo = len - todo;
obuf = dsb->device->buffer;
for (i = 0; i < todo; i++) {
/* 8-bit WAV is unsigned */
field = (*ibuf++ - 128);
field += (*obuf - 128);
if (field > 127) field = 127;
else if (field < -128) field = -128;
*obuf++ = field + 128;
}
}
} else {
INT16 *ibufs, *obufs;
ibufs = (INT16 *) ibuf;
obufs = (INT16 *)(dsb->device->buffer + writepos);
if ((writepos + len) <= dsb->device->buflen)
todo = len / 2;
else
todo = (dsb->device->buflen - writepos) / 2;
for (i = 0; i < todo; i++) {
/* 16-bit WAV is signed */
field = *ibufs++;
field += *obufs;
if (field > 32767) field = 32767;
else if (field < -32768) field = -32768;
*obufs++ = field;
}
if (todo < (len / 2)) {
todo = (len / 2) - todo;
obufs = (INT16 *)dsb->device->buffer;
for (i = 0; i < todo; i++) {
/* 16-bit WAV is signed */
field = *ibufs++;
field += *obufs;
if (field > 32767) field = 32767;
else if (field < -32768) field = -32768;
*obufs++ = field;
}
}
}
if (dsb->leadin && (dsb->startpos > dsb->buf_mixpos) && (dsb->startpos <= dsb->buf_mixpos + ilen)) {
/* HACK... leadin should be reset when the PLAY position reaches the startpos,
* not the MIX position... but if the sound buffer is bigger than our prebuffering
* (which must be the case for the streaming buffers that need this hack anyway)
* plus DS_HEL_MARGIN or equivalent, then this ought to work anyway. */
dsb->leadin = FALSE;
}
dsb->buf_mixpos += ilen;
if (dsb->buf_mixpos >= dsb->buflen) {
if (dsb->playflags & DSBPLAY_LOOPING) {
/* wrap */
dsb->buf_mixpos %= dsb->buflen;
if (dsb->leadin && (dsb->startpos <= dsb->buf_mixpos))
dsb->leadin = FALSE; /* HACK: see above */
} else if (dsb->buf_mixpos > dsb->buflen) {
ERR("Mixpos (%u) past buflen (%u), capping...\n", dsb->buf_mixpos, dsb->buflen);
dsb->buf_mixpos = dsb->buflen;
}
}
return len;
}
static void DSOUND_PhaseCancel(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD len)
{
INT ilen, field;
UINT i, todo;
BYTE *buf, *ibuf;
TRACE("(%p,%d,%d)\n",dsb,writepos,len);
if (len % dsb->device->pwfx->nBlockAlign) {
INT nBlockAlign = dsb->device->pwfx->nBlockAlign;
ERR("length not a multiple of block size, len = %d, block size = %d\n", len, nBlockAlign);
len = (len / nBlockAlign) * nBlockAlign; /* data alignment */
}
if ((buf = ibuf = DSOUND_tmpbuffer(dsb->device, len)) == NULL)
return;
TRACE("PhaseCancel (%p) len = %d, dest = %d\n", dsb, len, writepos);
ilen = DSOUND_MixerNorm(dsb, ibuf, len);
if ((dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) ||
(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) ||
(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
DSOUND_MixerVol(dsb, ibuf, len);
/* subtract instead of add, to phase out premixed data */
if (dsb->device->pwfx->wBitsPerSample == 8) {
BYTE *obuf = dsb->device->buffer + writepos;
if ((writepos + len) <= dsb->device->buflen)
todo = len;
else
todo = dsb->device->buflen - writepos;
for (i = 0; i < todo; i++) {
/* 8-bit WAV is unsigned */
field = (*obuf - 128);
field -= (*ibuf++ - 128);
if (field > 127) field = 127;
else if (field < -128) field = -128;
*obuf++ = field + 128;
}
if (todo < len) {
todo = len - todo;
obuf = dsb->device->buffer;
for (i = 0; i < todo; i++) {
/* 8-bit WAV is unsigned */
field = (*obuf - 128);
field -= (*ibuf++ - 128);
if (field > 127) field = 127;
else if (field < -128) field = -128;
*obuf++ = field + 128;
}
}
} else {
INT16 *ibufs, *obufs;
ibufs = (INT16 *) ibuf;
obufs = (INT16 *)(dsb->device->buffer + writepos);
if ((writepos + len) <= dsb->device->buflen)
todo = len / 2;
else
todo = (dsb->device->buflen - writepos) / 2;
for (i = 0; i < todo; i++) {
/* 16-bit WAV is signed */
field = *obufs;
field -= *ibufs++;
if (field > 32767) field = 32767;
else if (field < -32768) field = -32768;
*obufs++ = field;
}
if (todo < (len / 2)) {
todo = (len / 2) - todo;
obufs = (INT16 *)dsb->device->buffer;
for (i = 0; i < todo; i++) {
/* 16-bit WAV is signed */
field = *obufs;
field -= *ibufs++;
if (field > 32767) field = 32767;
else if (field < -32768) field = -32768;
*obufs++ = field;
}
}
}
}
static void DSOUND_MixCancel(IDirectSoundBufferImpl *dsb, DWORD writepos, BOOL cancel)
{
DWORD size, flen, len, npos, nlen;
INT iAdvance = dsb->pwfx->nBlockAlign;
INT oAdvance = dsb->device->pwfx->nBlockAlign;
/* determine amount of premixed data to cancel */
DWORD primary_done =
((dsb->primary_mixpos < writepos) ? dsb->device->buflen : 0) +
dsb->primary_mixpos - writepos;
TRACE("(%p, %d), buf_mixpos=%d\n", dsb, writepos, dsb->buf_mixpos);
/* backtrack the mix position */
size = primary_done / oAdvance;
flen = size * dsb->freqAdjust;
len = (flen >> DSOUND_FREQSHIFT) * iAdvance;
flen &= (1<<DSOUND_FREQSHIFT)-1;
while (dsb->freqAcc < flen) {
len += iAdvance;
dsb->freqAcc += 1<<DSOUND_FREQSHIFT;
}
len %= dsb->buflen;
npos = ((dsb->buf_mixpos < len) ? dsb->buflen : 0) +
dsb->buf_mixpos - len;
if (dsb->leadin && (dsb->startpos > npos) && (dsb->startpos <= npos + len)) {
/* stop backtracking at startpos */
npos = dsb->startpos;
len = ((dsb->buf_mixpos < npos) ? dsb->buflen : 0) +
dsb->buf_mixpos - npos;
flen = dsb->freqAcc;
nlen = len / dsb->pwfx->nBlockAlign;
nlen = ((nlen << DSOUND_FREQSHIFT) + flen) / dsb->freqAdjust;
nlen *= dsb->device->pwfx->nBlockAlign;
writepos =
((dsb->primary_mixpos < nlen) ? dsb->device->buflen : 0) +
dsb->primary_mixpos - nlen;
}
dsb->freqAcc -= flen;
dsb->buf_mixpos = npos;
dsb->primary_mixpos = writepos;
TRACE("new buf_mixpos=%d, primary_mixpos=%d (len=%d)\n",
dsb->buf_mixpos, dsb->primary_mixpos, len);
if (cancel) DSOUND_PhaseCancel(dsb, writepos, len);
}
void DSOUND_MixCancelAt(IDirectSoundBufferImpl *dsb, DWORD buf_writepos)
{
#if 0
DWORD i, size, flen, len, npos, nlen;
INT iAdvance = dsb->pwfx->nBlockAlign;
INT oAdvance = dsb->device->pwfx->nBlockAlign;
/* determine amount of premixed data to cancel */
DWORD buf_done =
((dsb->buf_mixpos < buf_writepos) ? dsb->buflen : 0) +
dsb->buf_mixpos - buf_writepos;
#endif
WARN("(%p, %d), buf_mixpos=%d\n", dsb, buf_writepos, dsb->buf_mixpos);
/* since this is not implemented yet, just cancel *ALL* prebuffering for now
* (which is faster anyway when there's only a single secondary buffer) */
dsb->device->need_remix = TRUE;
}
void DSOUND_ForceRemix(IDirectSoundBufferImpl *dsb)
{
TRACE("(%p)\n",dsb);
EnterCriticalSection(&dsb->lock);
if (dsb->state == STATE_PLAYING)
dsb->device->need_remix = TRUE;
LeaveCriticalSection(&dsb->lock);
}
/**
* Calculate the distance between two buffer offsets, taking wraparound
* into account.
*/
static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
{
if (ptr1 >= ptr2) {
return ptr1 - ptr2;
} else {
return buflen + ptr1 - ptr2;
}
}
/**
* Mix some frames from the given secondary buffer "dsb" into the device
* primary buffer.
*
* dsb = the secondary buffer
* playpos = the current play position in the device buffer (primary buffer)
* writepos = the current safe-to-write position in the device buffer
* mixlen = the maximum number of bytes in the primary buffer to mix, from the
* current writepos.
*
* Returns: the number of bytes beyond the writepos that were mixed.
*/
static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, DWORD playpos, DWORD writepos, DWORD mixlen)
{
/* The buffer's primary_mixpos may be before or after the the device
* buffer's mixpos, but both must be ahead of writepos. */
DWORD len, slen;
/* determine this buffer's write position */
DWORD buf_writepos = DSOUND_CalcPlayPosition(dsb, writepos, writepos);
/* determine how much already-mixed data exists */
DWORD buf_done = DSOUND_BufPtrDiff(dsb->buflen, dsb->buf_mixpos, buf_writepos);
DWORD primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);
DWORD adv_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->device->mixpos, writepos);
DWORD played = DSOUND_BufPtrDiff(dsb->buflen, buf_writepos, dsb->playpos);
DWORD buf_left = dsb->buflen - buf_writepos;
int still_behind;
TRACE("(%p,%d,%d,%d)\n",dsb,playpos,writepos,mixlen);
TRACE("buf_writepos=%d, primary_writepos=%d\n", buf_writepos, writepos);
TRACE("buf_done=%d, primary_done=%d\n", buf_done, primary_done);
TRACE("buf_mixpos=%d, primary_mixpos=%d, mixlen=%d\n", dsb->buf_mixpos, dsb->primary_mixpos,
mixlen);
TRACE("looping=%d, startpos=%d, leadin=%d\n", dsb->playflags, dsb->startpos, dsb->leadin);
/* check for notification positions */
if (dsb->dsbd.dwFlags & DSBCAPS_CTRLPOSITIONNOTIFY &&
dsb->state != STATE_STARTING) {
DSOUND_CheckEvent(dsb, played);
}
/* save write position for non-GETCURRENTPOSITION2... */
dsb->playpos = buf_writepos;
/* check whether CalcPlayPosition detected a mixing underrun */
if ((buf_done == 0) && (dsb->primary_mixpos != writepos)) {
/* it did, but did we have more to play? */
if ((dsb->playflags & DSBPLAY_LOOPING) ||
(dsb->buf_mixpos < dsb->buflen)) {
/* yes, have to recover */
ERR("underrun on sound buffer %p\n", dsb);
TRACE("recovering from underrun: primary_mixpos=%d\n", writepos);
}
dsb->primary_mixpos = writepos;
primary_done = 0;
}
/* determine how far ahead we should mix */
if (((dsb->playflags & DSBPLAY_LOOPING) ||
(dsb->leadin && (dsb->probably_valid_to != 0))) &&
!(dsb->dsbd.dwFlags & DSBCAPS_STATIC)) {
/* if this is a streaming buffer, it typically means that
* we should defer mixing past probably_valid_to as long
* as we can, to avoid unnecessary remixing */
/* the heavy-looking calculations shouldn't be that bad,
* as any game isn't likely to be have more than 1 or 2
* streaming buffers in use at any time anyway... */
DWORD probably_valid_left =
(dsb->probably_valid_to == (DWORD)-1) ? dsb->buflen :
((dsb->probably_valid_to < buf_writepos) ? dsb->buflen : 0) +
dsb->probably_valid_to - buf_writepos;
/* check for leadin condition */
if ((probably_valid_left == 0) &&
(dsb->probably_valid_to == dsb->startpos) &&
dsb->leadin)
probably_valid_left = dsb->buflen;
TRACE("streaming buffer probably_valid_to=%d, probably_valid_left=%d\n",
dsb->probably_valid_to, probably_valid_left);
/* check whether the app's time is already up */
if (probably_valid_left < dsb->writelead) {
WARN("probably_valid_to now within writelead, possible streaming underrun\n");
/* once we pass the point of no return,
* no reason to hold back anymore */
dsb->probably_valid_to = (DWORD)-1;
/* we just have to go ahead and mix what we have,
* there's no telling what the app is thinking anyway */
} else {
/* adjust for our frequency and our sample size */
probably_valid_left = MulDiv(probably_valid_left,
1 << DSOUND_FREQSHIFT,
dsb->pwfx->nBlockAlign * dsb->freqAdjust) *
dsb->device->pwfx->nBlockAlign;
/* check whether to clip mix_len */
if (probably_valid_left < mixlen) {
TRACE("clipping to probably_valid_left=%d\n", probably_valid_left);
mixlen = probably_valid_left;
}
}
}
/* cut mixlen with what's already been mixed */
if (mixlen < primary_done) {
/* huh? and still CalcPlayPosition didn't
* detect an underrun? */
FIXME("problem with underrun detection (mixlen=%d < primary_done=%d)\n", mixlen, primary_done);
return 0;
}
len = mixlen - primary_done;
TRACE("remaining mixlen=%d\n", len);
if (len < dsb->device->fraglen) {
/* smaller than a fragment, wait until it gets larger
* before we take the mixing overhead */
TRACE("mixlen not worth it, deferring mixing\n");
still_behind = 1;
goto post_mix;
}
/* ok, we know how much to mix, let's go */
still_behind = (adv_done > primary_done);
while (len) {
slen = dsb->device->buflen - dsb->primary_mixpos;
if (slen > len) slen = len;
slen = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, slen);
if ((dsb->primary_mixpos < dsb->device->mixpos) &&
(dsb->primary_mixpos + slen >= dsb->device->mixpos))
still_behind = FALSE;
dsb->primary_mixpos += slen; len -= slen;
dsb->primary_mixpos %= dsb->device->buflen;
if ((dsb->state == STATE_STOPPED) || !slen) break;
}
TRACE("new primary_mixpos=%d, primary_advbase=%d\n", dsb->primary_mixpos, dsb->device->mixpos);
TRACE("mixed data len=%d, still_behind=%d\n", mixlen-len, still_behind);
post_mix:
/* check if buffer should be considered complete */
if (buf_left < dsb->writelead &&
!(dsb->playflags & DSBPLAY_LOOPING)) {
dsb->state = STATE_STOPPED;
dsb->playpos = 0;
dsb->last_playpos = 0;
dsb->buf_mixpos = 0;
dsb->leadin = FALSE;
dsb->need_remix = FALSE;
DSOUND_CheckEvent(dsb, buf_left);
}
/* return how far we think the primary buffer can
* advance its underrun detector...*/
if (still_behind) return 0;
if ((mixlen - len) < primary_done) return 0;
slen = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, dsb->device->mixpos);
if (slen > mixlen) {
/* the primary_done and still_behind checks above should have worked */
FIXME("problem with advancement calculation (advlen=%d > mixlen=%d)\n", slen, mixlen);
slen = 0;
}
return slen;
}
/**
* For a DirectSoundDevice, go through all the currently playing buffers and
* mix them in to the device buffer.
*
* playpos = the current play position in the primary buffer
* writepos = the current safe-to-write position in the primary buffer
* mixlen = the maximum amount to mix into the primary buffer
* (beyond the current writepos)
* recover = true if the sound device may have been reset and the write
* position in the device buffer changed
*
* Returns: the length beyond the writepos that was mixed to.
*/
static DWORD DSOUND_MixToPrimary(const DirectSoundDevice *device, DWORD playpos, DWORD writepos,
DWORD mixlen, BOOL recover)
{
INT i, len, maxlen = 0;
IDirectSoundBufferImpl *dsb;
TRACE("(%d,%d,%d,%d)\n", playpos, writepos, mixlen, recover);
for (i = 0; i < device->nrofbuffers; i++) {
dsb = device->buffers[i];
if (dsb->buflen && dsb->state && !dsb->hwbuf) {
TRACE("Checking %p, mixlen=%d\n", dsb, mixlen);
EnterCriticalSection(&(dsb->lock));
if (dsb->state == STATE_STOPPING) {
DSOUND_MixCancel(dsb, writepos, TRUE);
dsb->state = STATE_STOPPED;
DSOUND_CheckEvent(dsb, 0);
} else {
if ((dsb->state == STATE_STARTING) || recover) {
dsb->primary_mixpos = writepos;
dsb->cvolpan = dsb->volpan;
dsb->need_remix = FALSE;
}
else if (dsb->need_remix) {
DSOUND_MixCancel(dsb, writepos, TRUE);
dsb->cvolpan = dsb->volpan;
dsb->need_remix = FALSE;
}
len = DSOUND_MixOne(dsb, playpos, writepos, mixlen);
if (dsb->state == STATE_STARTING)
dsb->state = STATE_PLAYING;
maxlen = (len > maxlen) ? len : maxlen;
}
LeaveCriticalSection(&(dsb->lock));
}
}
return maxlen;
}
static void DSOUND_MixReset(DirectSoundDevice *device, DWORD writepos)
{
INT i;
IDirectSoundBufferImpl *dsb;
int nfiller;
TRACE("(%p,%d)\n", device, writepos);
/* the sound of silence */
nfiller = device->pwfx->wBitsPerSample == 8 ? 128 : 0;
/* reset all buffer mix positions */
for (i = 0; i < device->nrofbuffers; i++) {
dsb = device->buffers[i];
if (dsb->buflen && dsb->state && !dsb->hwbuf) {
TRACE("Resetting %p\n", dsb);
EnterCriticalSection(&(dsb->lock));
if (dsb->state == STATE_STOPPING) {
dsb->state = STATE_STOPPED;
}
else if (dsb->state == STATE_STARTING) {
/* nothing */
} else {
DSOUND_MixCancel(dsb, writepos, FALSE);
dsb->cvolpan = dsb->volpan;
dsb->need_remix = FALSE;
}
LeaveCriticalSection(&(dsb->lock));
}
}
/* wipe out premixed data */
if (device->mixpos < writepos) {
FillMemory(device->buffer + writepos, device->buflen - writepos, nfiller);
FillMemory(device->buffer, device->mixpos, nfiller);
} else {
FillMemory(device->buffer + writepos, device->mixpos - writepos, nfiller);
}
/* reset primary mix position */
device->mixpos = writepos;
}
static void DSOUND_CheckReset(DirectSoundDevice *device, DWORD writepos)
{
TRACE("(%p,%d)\n",device,writepos);
if (device->need_remix) {
DSOUND_MixReset(device, writepos);
device->need_remix = FALSE;
/* maximize Half-Life performance */
device->prebuf = ds_snd_queue_min;
device->precount = 0;
} else {
device->precount++;
if (device->precount >= 4) {
if (device->prebuf < ds_snd_queue_max)
device->prebuf++;
device->precount = 0;
}
}
TRACE("premix adjust: %d\n", device->prebuf);
}
void DSOUND_WaveQueue(DirectSoundDevice *device, DWORD mixq)
{
TRACE("(%p,%d)\n", device, mixq);
if (mixq + device->pwqueue > ds_hel_queue) mixq = ds_hel_queue - device->pwqueue;
TRACE("queueing %d buffers, starting at %d\n", mixq, device->pwwrite);
for (; mixq; mixq--) {
waveOutWrite(device->hwo, device->pwave[device->pwwrite], sizeof(WAVEHDR));
device->pwwrite++;
if (device->pwwrite >= DS_HEL_FRAGS) device->pwwrite = 0;
device->pwqueue++;
}
}
/* #define SYNC_CALLBACK */
/**
* Perform mixing for a Direct Sound device. That is, go through all the
* secondary buffers (the sound bites currently playing) and mix them in
* to the primary buffer (the device buffer).
*/
static void DSOUND_PerformMix(DirectSoundDevice *device)
{
int nfiller;
BOOL forced;
HRESULT hres;
TRACE("(%p)\n", device);
/* the sound of silence */
nfiller = device->pwfx->wBitsPerSample == 8 ? 128 : 0;
/* whether the primary is forced to play even without secondary buffers */
forced = ((device->state == STATE_PLAYING) || (device->state == STATE_STARTING));
if (device->priolevel != DSSCL_WRITEPRIMARY) {
BOOL paused = ((device->state == STATE_STOPPED) || (device->state == STATE_STARTING));
/* FIXME: document variables */
DWORD playpos, writepos, inq, maxq, frag;
if (device->hwbuf) {
hres = IDsDriverBuffer_GetPosition(device->hwbuf, &playpos, &writepos);
if (hres) {
WARN("IDsDriverBuffer_GetPosition failed\n");
return;
}
/* Well, we *could* do Just-In-Time mixing using the writepos,
* but that's a little bit ambitious and unnecessary... */
/* rather add our safety margin to the writepos, if we're playing */
if (!paused) {
writepos += device->writelead;
writepos %= device->buflen;
} else writepos = playpos;
} else {
playpos = device->pwplay * device->fraglen;
writepos = playpos;
if (!paused) {
writepos += ds_hel_margin * device->fraglen;
writepos %= device->buflen;
}
}
TRACE("primary playpos=%d, writepos=%d, clrpos=%d, mixpos=%d, buflen=%d\n",
playpos,writepos,device->playpos,device->mixpos,device->buflen);
assert(device->playpos < device->buflen);
/* wipe out just-played sound data */
if (playpos < device->playpos) {
FillMemory(device->buffer + device->playpos, device->buflen - device->playpos, nfiller);
FillMemory(device->buffer, playpos, nfiller);
} else {
FillMemory(device->buffer + device->playpos, playpos - device->playpos, nfiller);
}
device->playpos = playpos;
EnterCriticalSection(&(device->mixlock));
/* reset mixing if necessary */
DSOUND_CheckReset(device, writepos);
/* check how much prebuffering is left */
inq = DSOUND_BufPtrDiff(device->buflen, device->mixpos, writepos);
/* find the maximum we can prebuffer */
if (!paused)
maxq = DSOUND_BufPtrDiff(device->buflen, playpos, writepos);
/* If we get the whole buffer, difference is 0, so we need to set whole buffer then */
if (paused || !maxq)
maxq = device->buflen;
/* clip maxq to device->prebuf */
frag = device->prebuf * device->fraglen;
if (maxq > frag)
maxq = frag;
/* check for consistency */
if (inq > maxq) {
/* the playback position must have passed our last
* mixed position, i.e. it's an underrun, or we have
* nothing more to play */
TRACE("reached end of mixed data (inq=%d, maxq=%d)\n", inq, maxq);
inq = 0;
/* stop the playback now, to allow buffers to refill */
if (device->state == STATE_PLAYING) {
device->state = STATE_STARTING;
}
else if (device->state == STATE_STOPPING) {
device->state = STATE_STOPPED;
}
else {
/* how can we have an underrun if we aren't playing? */
WARN("unexpected primary state (%d)\n", device->state);
}
#ifdef SYNC_CALLBACK
/* DSOUND_callback may need this lock */
LeaveCriticalSection(&(device->mixlock));
#endif
if (DSOUND_PrimaryStop(device) != DS_OK)
WARN("DSOUND_PrimaryStop failed\n");
#ifdef SYNC_CALLBACK
EnterCriticalSection(&(device->mixlock));
#endif
if (device->hwbuf) {
/* the Stop is supposed to reset play position to beginning of buffer */
/* unfortunately, OSS is not able to do so, so get current pointer */
hres = IDsDriverBuffer_GetPosition(device->hwbuf, &playpos, NULL);
if (hres) {
LeaveCriticalSection(&(device->mixlock));
WARN("IDsDriverBuffer_GetPosition failed\n");
return;
}
} else {
playpos = device->pwplay * device->fraglen;
}
writepos = playpos;
device->playpos = playpos;
device->mixpos = writepos;
inq = 0;
maxq = device->buflen;
if (maxq > frag) maxq = frag;
FillMemory(device->buffer, device->buflen, nfiller);
paused = TRUE;
}
/* do the mixing */
frag = DSOUND_MixToPrimary(device, playpos, writepos, maxq, paused);
if (forced) frag = maxq - inq;
device->mixpos += frag;
device->mixpos %= device->buflen;
if (frag) {
/* buffers have been filled, restart playback */
if (device->state == STATE_STARTING) {
device->state = STATE_PLAYING;
}
else if (device->state == STATE_STOPPED) {
/* the dsound is supposed to play if there's something to play
* even if it is reported as stopped, so don't let this confuse you */
device->state = STATE_STOPPING;
}
LeaveCriticalSection(&(device->mixlock));
if (paused) {
if (DSOUND_PrimaryPlay(device) != DS_OK)
WARN("DSOUND_PrimaryPlay failed\n");
else
TRACE("starting playback\n");
}
}
else
LeaveCriticalSection(&(device->mixlock));
} else {
/* in the DSSCL_WRITEPRIMARY mode, the app is totally in charge... */
if (device->state == STATE_STARTING) {
if (DSOUND_PrimaryPlay(device) != DS_OK)
WARN("DSOUND_PrimaryPlay failed\n");
else
device->state = STATE_PLAYING;
}
else if (device->state == STATE_STOPPING) {
if (DSOUND_PrimaryStop(device) != DS_OK)
WARN("DSOUND_PrimaryStop failed\n");
else
device->state = STATE_STOPPED;
}
}
}
void CALLBACK DSOUND_timer(UINT timerID, UINT msg, DWORD_PTR dwUser,
DWORD_PTR dw1, DWORD_PTR dw2)
{
DirectSoundDevice * device = (DirectSoundDevice*)dwUser;
DWORD start_time = GetTickCount();
DWORD end_time;
TRACE("(%d,%d,0x%lx,0x%lx,0x%lx)\n",timerID,msg,dwUser,dw1,dw2);
TRACE("entering at %d\n", start_time);
if (DSOUND_renderer[device->drvdesc.dnDevNode] != device) {
ERR("dsound died without killing us?\n");
timeKillEvent(timerID);
timeEndPeriod(DS_TIME_RES);
return;
}
RtlAcquireResourceShared(&(device->buffer_list_lock), TRUE);
if (device->ref)
DSOUND_PerformMix(device);
RtlReleaseResource(&(device->buffer_list_lock));
end_time = GetTickCount();
TRACE("completed processing at %d, duration = %d\n", end_time, end_time - start_time);
}
void CALLBACK DSOUND_callback(HWAVEOUT hwo, UINT msg, DWORD dwUser, DWORD dw1, DWORD dw2)
{
DirectSoundDevice * device = (DirectSoundDevice*)dwUser;
TRACE("(%p,%x,%x,%x,%x)\n",hwo,msg,dwUser,dw1,dw2);
TRACE("entering at %d, msg=%08x(%s)\n", GetTickCount(), msg,
msg==MM_WOM_DONE ? "MM_WOM_DONE" : msg==MM_WOM_CLOSE ? "MM_WOM_CLOSE" :
msg==MM_WOM_OPEN ? "MM_WOM_OPEN" : "UNKNOWN");
if (msg == MM_WOM_DONE) {
DWORD inq, mixq, fraglen, buflen, pwplay, playpos, mixpos;
if (device->pwqueue == (DWORD)-1) {
TRACE("completed due to reset\n");
return;
}
/* it could be a bad idea to enter critical section here... if there's lock contention,
* the resulting scheduling delays might obstruct the winmm player thread */
#ifdef SYNC_CALLBACK
EnterCriticalSection(&(device->mixlock));
#endif
/* retrieve current values */
fraglen = device->fraglen;
buflen = device->buflen;
pwplay = device->pwplay;
playpos = pwplay * fraglen;
mixpos = device->mixpos;
/* check remaining mixed data */
inq = DSOUND_BufPtrDiff(buflen, mixpos, playpos);
mixq = inq / fraglen;
if ((inq - (mixq * fraglen)) > 0) mixq++;
/* complete the playing buffer */
TRACE("done playing primary pos=%d\n", playpos);
pwplay++;
if (pwplay >= DS_HEL_FRAGS) pwplay = 0;
/* write new values */
device->pwplay = pwplay;
device->pwqueue--;
/* queue new buffer if we have data for it */
if (inq>1) DSOUND_WaveQueue(device, inq-1);
#ifdef SYNC_CALLBACK
LeaveCriticalSection(&(device->mixlock));
#endif
}
TRACE("completed\n");
}
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