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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
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
*/
#include <assert.h>
#include <stdarg.h>
#include <math.h> /* Insomnia - pow() function */
#define NONAMELESSSTRUCT
#define NONAMELESSUNION
#include "windef.h"
#include "winbase.h"
#include "winuser.h"
#include "mmsystem.h"
#include "winternl.h"
#include "wine/debug.h"
#include "dsound.h"
#include "dsdriver.h"
#include "dsound_private.h"
WINE_DEFAULT_DEBUG_CHANNEL(dsound);
void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan)
{
double temp;
TRACE("(%p)\n",volpan);
TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
/* the AmpFactors are expressed in 16.16 fixed point */
volpan->dwVolAmpFactor = (ULONG) (pow(2.0, volpan->lVolume / 600.0) * 0xffff);
/* FIXME: dwPan{Left|Right}AmpFactor */
/* FIXME: use calculated vol and pan ampfactors */
temp = (double) (volpan->lVolume - (volpan->lPan > 0 ? volpan->lPan : 0));
volpan->dwTotalLeftAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
temp = (double) (volpan->lVolume + (volpan->lPan < 0 ? volpan->lPan : 0));
volpan->dwTotalRightAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
TRACE("left = %x, right = %x\n", volpan->dwTotalLeftAmpFactor, volpan->dwTotalRightAmpFactor);
}
void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan)
{
double left,right;
TRACE("(%p)\n",volpan);
TRACE("left=%x, right=%x\n",volpan->dwTotalLeftAmpFactor,volpan->dwTotalRightAmpFactor);
if (volpan->dwTotalLeftAmpFactor==0)
left=-10000;
else
left=600 * log(((double)volpan->dwTotalLeftAmpFactor) / 0xffff) / log(2);
if (volpan->dwTotalRightAmpFactor==0)
right=-10000;
else
right=600 * log(((double)volpan->dwTotalRightAmpFactor) / 0xffff) / log(2);
if (left<right)
{
volpan->lVolume=right;
volpan->dwVolAmpFactor=volpan->dwTotalRightAmpFactor;
}
else
{
volpan->lVolume=left;
volpan->dwVolAmpFactor=volpan->dwTotalLeftAmpFactor;
}
if (volpan->lVolume < -10000)
volpan->lVolume=-10000;
volpan->lPan=right-left;
if (volpan->lPan < -10000)
volpan->lPan=-10000;
TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
}
void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
{
TRACE("(%p)\n",dsb);
/* calculate the 10ms write lead */
dsb->writelead = (dsb->freq / 100) * dsb->pwfx->nBlockAlign;
}
/**
* Check for application callback requests for when the play position
* reaches certain points.
*
* The offsets that will be triggered will be those between the recorded
* "last played" position for the buffer (i.e. dsb->playpos) and "len" bytes
* beyond that position.
*/
void DSOUND_CheckEvent(IDirectSoundBufferImpl *dsb, DWORD playpos, int len)
{
int i;
DWORD offset;
LPDSBPOSITIONNOTIFY event;
TRACE("(%p,%d)\n",dsb,len);
if (dsb->nrofnotifies == 0)
return;
TRACE("(%p) buflen = %d, playpos = %d, len = %d\n",
dsb, dsb->buflen, playpos, len);
for (i = 0; i < dsb->nrofnotifies ; i++) {
event = dsb->notifies + i;
offset = event->dwOffset;
TRACE("checking %d, position %d, event = %p\n",
i, offset, event->hEventNotify);
/* DSBPN_OFFSETSTOP has to be the last element. So this is */
/* OK. [Inside DirectX, p274] */
/* */
/* This also means we can't sort the entries by offset, */
/* because DSBPN_OFFSETSTOP == -1 */
if (offset == DSBPN_OFFSETSTOP) {
if (dsb->state == STATE_STOPPED) {
SetEvent(event->hEventNotify);
TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
return;
} else
return;
}
if ((playpos + len) >= dsb->buflen) {
if ((offset < ((playpos + len) % dsb->buflen)) ||
(offset >= playpos)) {
TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
SetEvent(event->hEventNotify);
}
} else {
if ((offset >= playpos) && (offset < (playpos + len))) {
TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
SetEvent(event->hEventNotify);
}
}
}
}
/* WAV format info can be found at:
*
* http://www.cwi.nl/ftp/audio/AudioFormats.part2
* ftp://ftp.cwi.nl/pub/audio/RIFF-format
*
* Import points to remember:
* 8-bit WAV is unsigned
* 16-bit WAV is signed
*/
/* Use the same formulas as pcmconverter.c */
static inline INT16 cvtU8toS16(BYTE b)
{
return (short)((b+(b << 8))-32768);
}
static inline BYTE cvtS16toU8(INT16 s)
{
return (s >> 8) ^ (unsigned char)0x80;
}
/**
* Copy a single frame from the given input buffer to the given output buffer.
* Translate 8 <-> 16 bits and mono <-> stereo
*/
static inline void cp_fields(const IDirectSoundBufferImpl *dsb, const BYTE *ibuf, BYTE *obuf )
{
DirectSoundDevice * device = dsb->device;
INT fl,fr;
if (dsb->pwfx->wBitsPerSample == 8) {
if (device->pwfx->wBitsPerSample == 8 &&
device->pwfx->nChannels == dsb->pwfx->nChannels) {
/* avoid needless 8->16->8 conversion */
*obuf=*ibuf;
if (dsb->pwfx->nChannels==2)
*(obuf+1)=*(ibuf+1);
return;
}
fl = cvtU8toS16(*ibuf);
fr = (dsb->pwfx->nChannels==2 ? cvtU8toS16(*(ibuf + 1)) : fl);
} else {
fl = *((const INT16 *)ibuf);
fr = (dsb->pwfx->nChannels==2 ? *(((const INT16 *)ibuf) + 1) : fl);
}
if (device->pwfx->nChannels == 2) {
if (device->pwfx->wBitsPerSample == 8) {
*obuf = cvtS16toU8(fl);
*(obuf + 1) = cvtS16toU8(fr);
return;
}
if (device->pwfx->wBitsPerSample == 16) {
*((INT16 *)obuf) = fl;
*(((INT16 *)obuf) + 1) = fr;
return;
}
}
if (device->pwfx->nChannels == 1) {
fl = (fl + fr) >> 1;
if (device->pwfx->wBitsPerSample == 8) {
*obuf = cvtS16toU8(fl);
return;
}
if (device->pwfx->wBitsPerSample == 16) {
*((INT16 *)obuf) = fl;
return;
}
}
}
/**
* Mix at most the given amount of data into the given device buffer from the
* given secondary buffer, starting from the dsb's first currently unmixed
* frame (buf_mixpos), translating frequency (pitch), stereo/mono and
* bits-per-sample. The secondary buffer sample is looped if it is not
* long enough and it is a looping buffer.
* (Doesn't perform any mixing - this is a straight copy operation).
*
* Now with PerfectPitch (tm) technology
*
* dsb = the secondary buffer
* buf = the device buffer
* len = number of bytes to store in the device buffer
*
* Returns: the number of bytes read from the secondary buffer
* (ie. len, adjusted for frequency, number of channels and sample size,
* and limited by buffer length for non-looping buffers)
*/
static INT DSOUND_MixerNorm(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len)
{
INT i, size, ipos, ilen;
BYTE *ibp, *obp;
INT iAdvance = dsb->pwfx->nBlockAlign;
INT oAdvance = dsb->device->pwfx->nBlockAlign;
ibp = dsb->buffer->memory + dsb->buf_mixpos;
obp = buf;
TRACE("(%p, %p, %p), buf_mixpos=%d\n", dsb, ibp, obp, dsb->buf_mixpos);
/* Check for the best case */
if ((dsb->freq == dsb->device->pwfx->nSamplesPerSec) &&
(dsb->pwfx->wBitsPerSample == dsb->device->pwfx->wBitsPerSample) &&
(dsb->pwfx->nChannels == dsb->device->pwfx->nChannels)) {
INT bytesleft = dsb->buflen - dsb->buf_mixpos;
TRACE("(%p) Best case\n", dsb);
if (len <= bytesleft )
CopyMemory(obp, ibp, len);
else { /* wrap */
CopyMemory(obp, ibp, bytesleft);
CopyMemory(obp + bytesleft, dsb->buffer->memory, len - bytesleft);
}
return len;
}
/* Check for same sample rate */
if (dsb->freq == dsb->device->pwfx->nSamplesPerSec) {
TRACE("(%p) Same sample rate %d = primary %d\n", dsb,
dsb->freq, dsb->device->pwfx->nSamplesPerSec);
ilen = 0;
for (i = 0; i < len; i += oAdvance) {
cp_fields(dsb, ibp, obp );
ibp += iAdvance;
ilen += iAdvance;
obp += oAdvance;
if (ibp >= (BYTE *)(dsb->buffer->memory + dsb->buflen))
ibp = dsb->buffer->memory; /* wrap */
}
return (ilen);
}
/* Mix in different sample rates */
/* */
/* New PerfectPitch(tm) Technology (c) 1998 Rob Riggs */
/* Patent Pending :-] */
/* Patent enhancements (c) 2000 Ove K<>ven,
* TransGaming Technologies Inc. */
/* FIXME("(%p) Adjusting frequency: %ld -> %ld (need optimization)\n",
dsb, dsb->freq, dsb->device->pwfx->nSamplesPerSec); */
size = len / oAdvance;
ilen = 0;
ipos = dsb->buf_mixpos;
for (i = 0; i < size; i++) {
cp_fields(dsb, (dsb->buffer->memory + ipos), obp);
obp += oAdvance;
dsb->freqAcc += dsb->freqAdjust;
if (dsb->freqAcc >= (1<<DSOUND_FREQSHIFT)) {
ULONG adv = (dsb->freqAcc>>DSOUND_FREQSHIFT) * iAdvance;
dsb->freqAcc &= (1<<DSOUND_FREQSHIFT)-1;
ipos += adv; ilen += adv;
ipos %= dsb->buflen;
}
}
return ilen;
}
static void DSOUND_MixerVol(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len)
{
INT i;
BYTE *bpc = buf;
INT16 *bps = (INT16 *) buf;
TRACE("(%p,%p,%d)\n",dsb,buf,len);
TRACE("left = %x, right = %x\n", dsb->volpan.dwTotalLeftAmpFactor,
dsb->volpan.dwTotalRightAmpFactor);
if ((!(dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) || (dsb->volpan.lPan == 0)) &&
(!(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) || (dsb->volpan.lVolume == 0)) &&
!(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
return; /* Nothing to do */
/* If we end up with some bozo coder using panning or 3D sound */
/* with a mono primary buffer, it could sound very weird using */
/* this method. Oh well, tough patooties. */
switch (dsb->device->pwfx->wBitsPerSample) {
case 8:
/* 8-bit WAV is unsigned, but we need to operate */
/* on signed data for this to work properly */
switch (dsb->device->pwfx->nChannels) {
case 1:
for (i = 0; i < len; i++) {
INT val = *bpc - 128;
val = (val * dsb->volpan.dwTotalLeftAmpFactor) >> 16;
*bpc = val + 128;
bpc++;
}
break;
case 2:
for (i = 0; i < len; i+=2) {
INT val = *bpc - 128;
val = (val * dsb->volpan.dwTotalLeftAmpFactor) >> 16;
*bpc++ = val + 128;
val = *bpc - 128;
val = (val * dsb->volpan.dwTotalRightAmpFactor) >> 16;
*bpc = val + 128;
bpc++;
}
break;
default:
FIXME("doesn't support %d channels\n", dsb->device->pwfx->nChannels);
break;
}
break;
case 16:
/* 16-bit WAV is signed -- much better */
switch (dsb->device->pwfx->nChannels) {
case 1:
for (i = 0; i < len; i += 2) {
*bps = (*bps * dsb->volpan.dwTotalLeftAmpFactor) >> 16;
bps++;
}
break;
case 2:
for (i = 0; i < len; i += 4) {
*bps = (*bps * dsb->volpan.dwTotalLeftAmpFactor) >> 16;
bps++;
*bps = (*bps * dsb->volpan.dwTotalRightAmpFactor) >> 16;
bps++;
}
break;
default:
FIXME("doesn't support %d channels\n", dsb->device->pwfx->nChannels);
break;
}
break;
default:
FIXME("doesn't support %d bit samples\n", dsb->device->pwfx->wBitsPerSample);
break;
}
}
/**
* Make sure the device's tmp_buffer is at least the given size. Return a
* pointer to it.
*/
static LPBYTE DSOUND_tmpbuffer(DirectSoundDevice *device, DWORD len)
{
TRACE("(%p,%d)\n", device, len);
if (len > device->tmp_buffer_len) {
if (device->tmp_buffer)
device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, device->tmp_buffer, len);
else
device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, len);
device->tmp_buffer_len = len;
}
return device->tmp_buffer;
}
/**
* Mix (at most) the given number of bytes into the given position of the
* device buffer, from the secondary buffer "dsb" (starting at the current
* mix position for that buffer).
*
* Returns the number of bytes actually mixed into the device buffer. This
* will match fraglen unless the end of the secondary buffer is reached
* (and it is not looping).
*
* dsb = the secondary buffer to mix from
* writepos = position (offset) in device buffer to write at
* fraglen = number of bytes to mix
*/
static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen)
{
INT i, len, ilen, field, todo;
BYTE *buf, *ibuf;
TRACE("(%p,%d,%d)\n",dsb,writepos,fraglen);
len = fraglen;
if (!(dsb->playflags & DSBPLAY_LOOPING)) {
/* This buffer is not looping, so make sure the requested
* length will not take us past the end of the buffer */
int secondary_remainder = dsb->buflen - dsb->buf_mixpos;
int adjusted_remainder = MulDiv(dsb->device->pwfx->nAvgBytesPerSec, secondary_remainder, dsb->nAvgBytesPerSec);
assert(adjusted_remainder >= 0);
adjusted_remainder -= adjusted_remainder % dsb->device->pwfx->nBlockAlign; /* data alignment */
/* The adjusted remainder must be at least one sample,
* otherwise we will never reach the end of the
* secondary buffer, as there will perpetually be a
* fractional remainder */
TRACE("secondary_remainder = %d, adjusted_remainder = %d, len = %d\n", secondary_remainder, adjusted_remainder, len);
if (adjusted_remainder < len) {
TRACE("clipping len to remainder of secondary buffer\n");
len = adjusted_remainder;
}
if (len == 0)
return 0;
}
if (len % dsb->device->pwfx->nBlockAlign) {
INT nBlockAlign = dsb->device->pwfx->nBlockAlign;
ERR("length not a multiple of block size, len = %d, block size = %d\n", len, nBlockAlign);
len -= len % nBlockAlign; /* data alignment */
}
/* Create temp buffer to hold actual resulting data */
if ((buf = ibuf = DSOUND_tmpbuffer(dsb->device, len)) == NULL)
return 0;
TRACE("MixInBuffer (%p) len = %d, dest = %d\n", dsb, len, writepos);
/* first, copy the data from the DirectSoundBuffer into the temporary
buffer, translating frequency/bits-per-sample/number-of-channels
to match the device settings */
ilen = DSOUND_MixerNorm(dsb, ibuf, len);
/* then apply the correct volume, if necessary */
if ((dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) ||
(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) ||
(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
DSOUND_MixerVol(dsb, ibuf, len);
/* Now mix the temporary buffer into the devices main buffer */
if (dsb->device->pwfx->wBitsPerSample == 8) {
BYTE *obuf = dsb->device->buffer + writepos;
if ((writepos + len) <= dsb->device->buflen)
todo = len;
else
todo = dsb->device->buflen - writepos;
for (i = 0; i < todo; i++) {
/* 8-bit WAV is unsigned */
field = (*ibuf++ - 128);
field += (*obuf - 128);
if (field > 127) field = 127;
else if (field < -128) field = -128;
*obuf++ = field + 128;
}
if (todo < len) {
todo = len - todo;
obuf = dsb->device->buffer;
for (i = 0; i < todo; i++) {
/* 8-bit WAV is unsigned */
field = (*ibuf++ - 128);
field += (*obuf - 128);
if (field > 127) field = 127;
else if (field < -128) field = -128;
*obuf++ = field + 128;
}
}
} else {
INT16 *ibufs, *obufs;
ibufs = (INT16 *) ibuf;
obufs = (INT16 *)(dsb->device->buffer + writepos);
if ((writepos + len) <= dsb->device->buflen)
todo = len / 2;
else
todo = (dsb->device->buflen - writepos) / 2;
for (i = 0; i < todo; i++) {
/* 16-bit WAV is signed */
field = *ibufs++;
field += *obufs;
if (field > 32767) field = 32767;
else if (field < -32768) field = -32768;
*obufs++ = field;
}
if (todo < (len / 2)) {
todo = (len / 2) - todo;
obufs = (INT16 *)dsb->device->buffer;
for (i = 0; i < todo; i++) {
/* 16-bit WAV is signed */
field = *ibufs++;
field += *obufs;
if (field > 32767) field = 32767;
else if (field < -32768) field = -32768;
*obufs++ = field;
}
}
}
if (dsb->leadin && (dsb->startpos > dsb->buf_mixpos) && (dsb->startpos <= dsb->buf_mixpos + ilen)) {
/* HACK... leadin should be reset when the PLAY position reaches the startpos,
* not the MIX position... but if the sound buffer is bigger than our prebuffering
* (which must be the case for the streaming buffers that need this hack anyway)
* plus DS_HEL_MARGIN or equivalent, then this ought to work anyway. */
dsb->leadin = FALSE;
}
/* check for notification positions */
if (dsb->dsbd.dwFlags & DSBCAPS_CTRLPOSITIONNOTIFY &&
dsb->state != STATE_STARTING) {
DSOUND_CheckEvent(dsb, dsb->buf_mixpos, ilen);
}
dsb->buf_mixpos += ilen;
if (dsb->buf_mixpos >= dsb->buflen) {
if (dsb->playflags & DSBPLAY_LOOPING) {
/* wrap */
dsb->buf_mixpos %= dsb->buflen;
if (dsb->leadin && (dsb->startpos <= dsb->buf_mixpos))
dsb->leadin = FALSE; /* HACK: see above */
} else if (dsb->buf_mixpos > dsb->buflen) {
ERR("Mixpos (%u) past buflen (%u), capping...\n", dsb->buf_mixpos, dsb->buflen);
dsb->buf_mixpos = dsb->buflen;
}
}
/* increase mix position */
dsb->primary_mixpos += len;
dsb->primary_mixpos %= dsb->device->buflen;
return len;
}
/**
* Calculate the distance between two buffer offsets, taking wraparound
* into account.
*/
static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
{
if (ptr1 >= ptr2) {
return ptr1 - ptr2;
} else {
return buflen + ptr1 - ptr2;
}
}
/**
* Mix some frames from the given secondary buffer "dsb" into the device
* primary buffer.
*
* dsb = the secondary buffer
* 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 primary_done, buflen = dsb->buflen / dsb->pwfx->nBlockAlign * dsb->device->pwfx->nBlockAlign;
TRACE("(%p,%d,%d,%d)\n",dsb,playpos,writepos,mixlen);
TRACE("writepos=%d, buf_mixpos=%d, primary_mixpos=%d, mixlen=%d\n", writepos, dsb->buf_mixpos, dsb->primary_mixpos, mixlen);
TRACE("looping=%d, startpos=%d, leadin=%d, buflen=%d\n", dsb->playflags, dsb->startpos, dsb->leadin, dsb->buflen);
/* calculate how much pre-buffering has already been done for this buffer */
primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);
/* sanity */
if(mixlen < primary_done)
{
/* Should *NEVER* happen */
ERR("Fatal error. Under/Overflow? primary_done=%d, mixpos=%d, primary_mixpos=%d, writepos=%d, playpos=%d\n", primary_done,dsb->buf_mixpos,dsb->primary_mixpos, writepos, playpos);
return 0;
}
/* take into acount already mixed data */
mixlen -= primary_done;
TRACE("primary_done=%d, mixlen (primary) = %i\n", primary_done, mixlen);
if ((dsb->playflags & DSBPLAY_LOOPING) && mixlen > buflen)
{
while (mixlen > buflen)
{
DWORD mixedlength = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, buflen);
mixlen -= buflen;
if (!mixedlength)
{
mixlen = 0;
break;
}
}
}
/* clip to valid length */
mixlen = (buflen < mixlen) ? buflen : mixlen;
TRACE("mixlen (buffer)=%d\n", mixlen);
if (mixlen)
/* mix more data */
mixlen = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixlen);
TRACE("new primary_mixpos=%d, mixed data len=%d, buffer left = %d\n",
dsb->primary_mixpos, mixlen, (dsb->buflen - dsb->buf_mixpos));
/* re-calculate the primary done */
primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);
/* check if buffer should be considered complete */
if (((dsb->buflen - dsb->buf_mixpos) < dsb->writelead) &&
!(dsb->playflags & DSBPLAY_LOOPING)) {
TRACE("Buffer reached end. Stopped\n");
dsb->state = STATE_STOPPED;
dsb->buf_mixpos = 0;
dsb->leadin = FALSE;
}
/* Report back the total prebuffered amount for this buffer */
return primary_done;
}
/**
* For a DirectSoundDevice, go through all the currently playing buffers and
* mix them in to the device buffer.
*
* playpos = the current play position in the primary buffer
* writepos = the current safe-to-write position in the primary buffer
* mixlen = the maximum amount to mix into the primary buffer
* (beyond the current writepos)
* recover = true if the sound device may have been reset and the write
* position in the device buffer changed
* all_stopped = reports back if all buffers have stopped
*
* Returns: the length beyond the writepos that was mixed to.
*/
static DWORD DSOUND_MixToPrimary(const DirectSoundDevice *device, DWORD playpos, DWORD writepos,
DWORD mixlen, BOOL recover, BOOL *all_stopped)
{
INT i, len;
DWORD minlen = 0;
IDirectSoundBufferImpl *dsb;
/* unless we find a running buffer, all have stopped */
*all_stopped = TRUE;
TRACE("(%d,%d,%d,%d)\n", playpos, writepos, mixlen, recover);
for (i = 0; i < device->nrofbuffers; i++) {
dsb = device->buffers[i];
TRACE("MixToPrimary for %p, state=%d\n", dsb, dsb->state);
if (dsb->buflen && dsb->state && !dsb->hwbuf) {
TRACE("Checking %p, mixlen=%d\n", dsb, mixlen);
EnterCriticalSection(&(dsb->lock));
/* if buffer is stopping it is stopped now */
if (dsb->state == STATE_STOPPING) {
dsb->state = STATE_STOPPED;
DSOUND_CheckEvent(dsb, 0, 0);
} else {
/* if recovering, reset the mix position */
if ((dsb->state == STATE_STARTING) || recover) {
dsb->primary_mixpos = writepos;
}
/* mix next buffer into the main buffer */
len = DSOUND_MixOne(dsb, playpos, writepos, mixlen);
/* if the buffer was starting, it must be playing now */
if (dsb->state == STATE_STARTING)
dsb->state = STATE_PLAYING;
/* check if min-len should be initialized */
if(minlen == 0) minlen = len;
/* record the minimum length mixed from all buffers */
/* we only want to return the length which *all* buffers have mixed */
if(len != 0) minlen = (len < minlen) ? len : minlen;
}
if(dsb->state != STATE_STOPPED){
*all_stopped = FALSE;
}
LeaveCriticalSection(&(dsb->lock));
}
}
TRACE("Mixed at least %d from all buffers\n", minlen);
return minlen;
}
/**
* Add buffers to the emulated wave device system.
*
* device = The current dsound playback device
* force = If TRUE, the function will buffer up as many frags as possible,
* even though and will ignore the actual state of the primary buffer.
*
* Returns: None
*/
static void DSOUND_WaveQueue(DirectSoundDevice *device, BOOL force)
{
DWORD prebuf_frags, wave_writepos, wave_fragpos, i;
TRACE("(%p)\n", device);
/* calculate the current wave frag position */
wave_fragpos = (device->pwplay + device->pwqueue) % DS_HEL_FRAGS;
/* calculte the current wave write position */
wave_writepos = wave_fragpos * device->fraglen;
TRACE("wave_fragpos = %i, wave_writepos = %i, pwqueue = %i, prebuf = %i\n",
wave_fragpos, wave_writepos, device->pwqueue, device->prebuf);
if(force == FALSE){
/* check remaining prebuffered frags */
prebuf_frags = DSOUND_BufPtrDiff(device->buflen, device->mixpos, wave_writepos);
prebuf_frags = prebuf_frags / device->fraglen;
}
else{
/* buffer the maximum amount of frags */
prebuf_frags = device->prebuf;
}
/* limit to the queue we have left */
if((prebuf_frags + device->pwqueue) > device->prebuf)
prebuf_frags = device->prebuf - device->pwqueue;
TRACE("prebuf_frags = %i\n", prebuf_frags);
/* adjust queue */
device->pwqueue += prebuf_frags;
/* get out of CS when calling the wave system */
LeaveCriticalSection(&(device->mixlock));
/* **** */
/* queue up the new buffers */
for(i=0; i<prebuf_frags; i++){
TRACE("queueing wave buffer %i\n", wave_fragpos);
waveOutWrite(device->hwo, device->pwave[wave_fragpos], sizeof(WAVEHDR));
wave_fragpos++;
wave_fragpos %= DS_HEL_FRAGS;
}
/* **** */
EnterCriticalSection(&(device->mixlock));
TRACE("queue now = %i\n", device->pwqueue);
}
/**
* Perform mixing for a Direct Sound device. That is, go through all the
* secondary buffers (the sound bites currently playing) and mix them in
* to the primary buffer (the device buffer).
*/
static void DSOUND_PerformMix(DirectSoundDevice *device)
{
TRACE("(%p)\n", device);
/* **** */
EnterCriticalSection(&(device->mixlock));
if (device->priolevel != DSSCL_WRITEPRIMARY) {
BOOL recover = FALSE, all_stopped = FALSE;
DWORD playpos, writepos, writelead, maxq, frag, prebuff_max, prebuff_left, size1, size2;
LPVOID buf1, buf2;
BOOL lock = (device->hwbuf && !(device->drvdesc.dwFlags & DSDDESC_DONTNEEDPRIMARYLOCK));
int nfiller;
/* the sound of silence */
nfiller = device->pwfx->wBitsPerSample == 8 ? 128 : 0;
/* get the position in the primary buffer */
if (DSOUND_PrimaryGetPosition(device, &playpos, &writepos) != 0){
LeaveCriticalSection(&(device->mixlock));
return;
}
TRACE("primary playpos=%d, writepos=%d, clrpos=%d, mixpos=%d, buflen=%d\n",
playpos,writepos,device->playpos,device->mixpos,device->buflen);
assert(device->playpos < device->buflen);
/* wipe out just-played sound data */
if (playpos < device->playpos) {
buf1 = device->buffer + device->playpos;
buf2 = device->buffer;
size1 = device->buflen - device->playpos;
size2 = playpos;
if (lock)
IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, device->playpos, size1+size2, 0);
FillMemory(buf1, size1, nfiller);
if (playpos && (!buf2 || !size2))
FIXME("%d: (%d, %d)=>(%d, %d) There should be an additional buffer here!!\n", __LINE__, device->playpos, device->mixpos, playpos, writepos);
FillMemory(buf2, size2, nfiller);
if (lock)
IDsDriverBuffer_Unlock(device->hwbuf, buf1, size1, buf2, size2);
} else {
buf1 = device->buffer + device->playpos;
buf2 = NULL;
size1 = playpos - device->playpos;
size2 = 0;
if (lock)
IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, device->playpos, size1+size2, 0);
FillMemory(buf1, size1, nfiller);
if (buf2 && size2)
{
FIXME("%d: There should be no additional buffer here!!\n", __LINE__);
FillMemory(buf2, size2, nfiller);
}
if (lock)
IDsDriverBuffer_Unlock(device->hwbuf, buf1, size1, buf2, size2);
}
device->playpos = playpos;
/* calc maximum prebuff */
prebuff_max = (device->prebuf * device->fraglen);
/* check how close we are to an underrun. It occurs when the writepos overtakes the mixpos */
prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);
writelead = DSOUND_BufPtrDiff(device->buflen, writepos, playpos);
/* find the maximum we can prebuffer from current write position */
maxq = prebuff_max - prebuff_left;
maxq = (writelead < prebuff_max) ? (prebuff_max - writelead) : 0;
TRACE("prebuff_left = %d, prebuff_max = %dx%d=%d, writelead=%d\n",
prebuff_left, device->prebuf, device->fraglen, prebuff_max, writelead);
/* check for underrun. underrun occurs when the write position passes the mix position */
if((prebuff_left > prebuff_max) || (device->state == STATE_STOPPED) || (device->state == STATE_STARTING)){
TRACE("Buffer starting or buffer underrun\n");
/* recover mixing for all buffers */
recover = TRUE;
/* reset mix position to write position */
device->mixpos = writepos;
}
if (lock)
IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, device->mixpos, maxq, 0);
/* do the mixing */
frag = DSOUND_MixToPrimary(device, playpos, writepos, maxq, recover, &all_stopped);
/* update the mix position, taking wrap-around into acount */
device->mixpos = writepos + frag;
device->mixpos %= device->buflen;
if (lock)
{
DWORD frag2 = (frag > size1 ? frag - size1 : 0);
frag -= frag2;
if (frag2 > size2)
{
FIXME("Buffering too much! (%d, %d, %d, %d)\n", maxq, frag, size2, frag2 - size2);
frag2 = size2;
}
IDsDriverBuffer_Unlock(device->hwbuf, buf1, frag, buf2, frag2);
}
/* update prebuff left */
prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);
/* check if have a whole fragment */
if (prebuff_left >= device->fraglen){
/* update the wave queue if using wave system */
if(device->hwbuf == NULL){
DSOUND_WaveQueue(device,TRUE);
}
/* buffers are full. start playing if applicable */
if(device->state == STATE_STARTING){
TRACE("started primary buffer\n");
if(DSOUND_PrimaryPlay(device) != DS_OK){
WARN("DSOUND_PrimaryPlay failed\n");
}
else{
/* we are playing now */
device->state = STATE_PLAYING;
}
}
/* buffers are full. start stopping if applicable */
if(device->state == STATE_STOPPED){
TRACE("restarting primary buffer\n");
if(DSOUND_PrimaryPlay(device) != DS_OK){
WARN("DSOUND_PrimaryPlay failed\n");
}
else{
/* start stopping again. as soon as there is no more data, it will stop */
device->state = STATE_STOPPING;
}
}
}
/* if device was stopping, its for sure stopped when all buffers have stopped */
else if((all_stopped == TRUE) && (device->state == STATE_STOPPING)){
TRACE("All buffers have stopped. Stopping primary buffer\n");
device->state = STATE_STOPPED;
/* stop the primary buffer now */
DSOUND_PrimaryStop(device);
}
} else {
/* update the wave queue if using wave system */
if(device->hwbuf == NULL){
DSOUND_WaveQueue(device, TRUE);
}
/* in the DSSCL_WRITEPRIMARY mode, the app is totally in charge... */
if (device->state == STATE_STARTING) {
if (DSOUND_PrimaryPlay(device) != DS_OK)
WARN("DSOUND_PrimaryPlay failed\n");
else
device->state = STATE_PLAYING;
}
else if (device->state == STATE_STOPPING) {
if (DSOUND_PrimaryStop(device) != DS_OK)
WARN("DSOUND_PrimaryStop failed\n");
else
device->state = STATE_STOPPED;
}
}
LeaveCriticalSection(&(device->mixlock));
/* **** */
}
void CALLBACK DSOUND_timer(UINT timerID, UINT msg, DWORD_PTR dwUser,
DWORD_PTR dw1, DWORD_PTR dw2)
{
DirectSoundDevice * device = (DirectSoundDevice*)dwUser;
DWORD start_time = GetTickCount();
DWORD end_time;
TRACE("(%d,%d,0x%lx,0x%lx,0x%lx)\n",timerID,msg,dwUser,dw1,dw2);
TRACE("entering at %d\n", start_time);
if (DSOUND_renderer[device->drvdesc.dnDevNode] != device) {
ERR("dsound died without killing us?\n");
timeKillEvent(timerID);
timeEndPeriod(DS_TIME_RES);
return;
}
RtlAcquireResourceShared(&(device->buffer_list_lock), TRUE);
if (device->ref)
DSOUND_PerformMix(device);
RtlReleaseResource(&(device->buffer_list_lock));
end_time = GetTickCount();
TRACE("completed processing at %d, duration = %d\n", end_time, end_time - start_time);
}
void CALLBACK DSOUND_callback(HWAVEOUT hwo, UINT msg, DWORD dwUser, DWORD dw1, DWORD dw2)
{
DirectSoundDevice * device = (DirectSoundDevice*)dwUser;
TRACE("(%p,%x,%x,%x,%x)\n",hwo,msg,dwUser,dw1,dw2);
TRACE("entering at %d, msg=%08x(%s)\n", GetTickCount(), msg,
msg==MM_WOM_DONE ? "MM_WOM_DONE" : msg==MM_WOM_CLOSE ? "MM_WOM_CLOSE" :
msg==MM_WOM_OPEN ? "MM_WOM_OPEN" : "UNKNOWN");
/* check if packet completed from wave driver */
if (msg == MM_WOM_DONE) {
/* **** */
EnterCriticalSection(&(device->mixlock));
TRACE("done playing primary pos=%d\n", device->pwplay * device->fraglen);
/* update playpos */
device->pwplay++;
device->pwplay %= DS_HEL_FRAGS;
/* sanity */
if(device->pwqueue == 0){
ERR("Wave queue corrupted!\n");
}
/* update queue */
device->pwqueue--;
LeaveCriticalSection(&(device->mixlock));
/* **** */
}
TRACE("completed\n");
}
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