📄 mixer.c
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if (len % dsb->dsound->device->pwfx->nBlockAlign) {
INT nBlockAlign = dsb->dsound->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->dsound->device, len)) == NULL)
return 0;
TRACE("MixInBuffer (%p) len = %d, dest = %ld\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);
if (dsb->dsound->device->pwfx->wBitsPerSample == 8) {
BYTE *obuf = dsb->dsound->device->buffer + writepos;
if ((writepos + len) <= dsb->dsound->device->buflen)
todo = len;
else
todo = dsb->dsound->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->dsound->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->dsound->device->buffer + writepos);
if ((writepos + len) <= dsb->dsound->device->buflen)
todo = len / 2;
else
todo = (dsb->dsound->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->dsound->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 (%lu) past buflen (%lu), 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,%ld,%ld)\n",dsb,writepos,len);
if (len % dsb->dsound->device->pwfx->nBlockAlign) {
INT nBlockAlign = dsb->dsound->device->pwfx->nBlockAlign;
ERR("length not a multiple of block size, len = %ld, block size = %d\n", len, nBlockAlign);
len = (len / nBlockAlign) * nBlockAlign; /* data alignment */
}
if ((buf = ibuf = DSOUND_tmpbuffer(dsb->dsound->device, len)) == NULL)
return;
TRACE("PhaseCancel (%p) len = %ld, dest = %ld\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->dsound->device->pwfx->wBitsPerSample == 8) {
BYTE *obuf = dsb->dsound->device->buffer + writepos;
if ((writepos + len) <= dsb->dsound->device->buflen)
todo = len;
else
todo = dsb->dsound->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->dsound->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->dsound->device->buffer + writepos);
if ((writepos + len) <= dsb->dsound->device->buflen)
todo = len / 2;
else
todo = (dsb->dsound->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->dsound->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;
}
}
}
}
static void DSOUND_MixCancel(IDirectSoundBufferImpl *dsb, DWORD writepos, BOOL cancel)
{
DWORD size, flen, len, npos, nlen;
INT iAdvance = dsb->pwfx->nBlockAlign;
INT oAdvance = dsb->dsound->device->pwfx->nBlockAlign;
/* determine amount of premixed data to cancel */
DWORD primary_done =
((dsb->primary_mixpos < writepos) ? dsb->dsound->device->buflen : 0) +
dsb->primary_mixpos - writepos;
TRACE("(%p, %ld), buf_mixpos=%ld\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->dsound->device->pwfx->nBlockAlign;
writepos =
((dsb->primary_mixpos < nlen) ? dsb->dsound->device->buflen : 0) +
dsb->primary_mixpos - nlen;
}
dsb->freqAcc -= flen;
dsb->buf_mixpos = npos;
dsb->primary_mixpos = writepos;
TRACE("new buf_mixpos=%ld, primary_mixpos=%ld (len=%ld)\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->dsound->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, %ld), buf_mixpos=%ld\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->dsound->device->need_remix = TRUE;
}
void DSOUND_ForceRemix(IDirectSoundBufferImpl *dsb)
{
TRACE("(%p)\n",dsb);
EnterCriticalSection(&dsb->lock);
if (dsb->state == STATE_PLAYING)
dsb->dsound->device->need_remix = TRUE;
LeaveCriticalSection(&dsb->lock);
}
static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, DWORD playpos, DWORD writepos, DWORD mixlen)
{
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 =
((dsb->buf_mixpos < buf_writepos) ? dsb->buflen : 0) +
dsb->buf_mixpos - buf_writepos;
DWORD primary_done =
((dsb->primary_mixpos < writepos) ? dsb->dsound->device->buflen : 0) +
dsb->primary_mixpos - writepos;
DWORD adv_done =
((dsb->dsound->device->mixpos < writepos) ? dsb->dsound->device->buflen : 0) +
dsb->dsound->device->mixpos - writepos;
DWORD played =
((buf_writepos < dsb->playpos) ? dsb->buflen : 0) +
buf_writepos - dsb->playpos;
DWORD buf_left = dsb->buflen - buf_writepos;
int still_behind;
TRACE("(%p,%ld,%ld,%ld)\n",dsb,playpos,writepos,mixlen);
TRACE("buf_writepos=%ld, primary_writepos=%ld\n", buf_writepos, writepos);
TRACE("buf_done=%ld, primary_done=%ld\n", buf_done, primary_done);
TRACE("buf_mixpos=%ld, primary_mixpos=%ld, mixlen=%ld\n", dsb->buf_mixpos, dsb->primary_mixpos,
mixlen);
TRACE("looping=%ld, startpos=%ld, leadin=%ld\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=%ld\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=%ld, probably_valid_left=%ld\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->dsound->device->pwfx->nBlockAlign;
/* check whether to clip mix_len */
if (probably_valid_left < mixlen) {
TRACE("clipping to probably_valid_left=%ld\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=%ld < primary_done=%ld)\n", mixlen, primary_done);
return 0;
}
len = mixlen - primary_done;
TRACE("remaining mixlen=%ld\n", len);
if (len < dsb->dsound->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->dsound->device->buflen - dsb->primary_mixpos;
if (slen > len) slen = len;
slen = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, slen);
if ((dsb->primary_mixpos < dsb->dsound->device->mixpos) &&
(dsb->primary_mixpos + slen >= dsb->dsound->device->mixpos))
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