3 * Copyright 1998 Marcus Meissner
4 * Copyright 1998 Rob Riggs
5 * Copyright 2000-2002 TransGaming Technologies, Inc.
6 * Copyright 2007 Peter Dons Tychsen
7 * Copyright 2007 Maarten Lankhorst
9 * This library is free software; you can redistribute it and/or
10 * modify it under the terms of the GNU Lesser General Public
11 * License as published by the Free Software Foundation; either
12 * version 2.1 of the License, or (at your option) any later version.
14 * This library is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17 * Lesser General Public License for more details.
19 * You should have received a copy of the GNU Lesser General Public
20 * License along with this library; if not, write to the Free Software
21 * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
26 #include <math.h> /* Insomnia - pow() function */
28 #define NONAMELESSSTRUCT
29 #define NONAMELESSUNION
34 #include "wine/debug.h"
37 #include "dsound_private.h"
39 WINE_DEFAULT_DEBUG_CHANNEL(dsound);
41 void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan)
44 TRACE("(%p)\n",volpan);
46 TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
47 /* the AmpFactors are expressed in 16.16 fixed point */
48 volpan->dwVolAmpFactor = (ULONG) (pow(2.0, volpan->lVolume / 600.0) * 0xffff);
49 /* FIXME: dwPan{Left|Right}AmpFactor */
51 /* FIXME: use calculated vol and pan ampfactors */
52 temp = (double) (volpan->lVolume - (volpan->lPan > 0 ? volpan->lPan : 0));
53 volpan->dwTotalLeftAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
54 temp = (double) (volpan->lVolume + (volpan->lPan < 0 ? volpan->lPan : 0));
55 volpan->dwTotalRightAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
57 TRACE("left = %x, right = %x\n", volpan->dwTotalLeftAmpFactor, volpan->dwTotalRightAmpFactor);
60 void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan)
63 TRACE("(%p)\n",volpan);
65 TRACE("left=%x, right=%x\n",volpan->dwTotalLeftAmpFactor,volpan->dwTotalRightAmpFactor);
66 if (volpan->dwTotalLeftAmpFactor==0)
69 left=600 * log(((double)volpan->dwTotalLeftAmpFactor) / 0xffff) / log(2);
70 if (volpan->dwTotalRightAmpFactor==0)
73 right=600 * log(((double)volpan->dwTotalRightAmpFactor) / 0xffff) / log(2);
76 volpan->lVolume=right;
77 volpan->dwVolAmpFactor=volpan->dwTotalRightAmpFactor;
82 volpan->dwVolAmpFactor=volpan->dwTotalLeftAmpFactor;
84 if (volpan->lVolume < -10000)
85 volpan->lVolume=-10000;
86 volpan->lPan=right-left;
87 if (volpan->lPan < -10000)
90 TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
93 /** Convert a primary buffer position to a pointer position for device->mix_buffer
94 * device: DirectSoundDevice for which to calculate
95 * pos: Primary buffer position to converts
96 * Returns: Offset for mix_buffer
98 DWORD DSOUND_bufpos_to_mixpos(const DirectSoundDevice* device, DWORD pos)
100 DWORD ret = pos * 32 / device->pwfx->wBitsPerSample;
101 if (device->pwfx->wBitsPerSample == 32)
106 /* NOTE: Not all secpos have to always be mapped to a bufpos, other way around is always the case
107 * DWORD64 is used here because a single DWORD wouldn't be big enough to fit the freqAcc for big buffers
109 /** This function converts a 'native' sample pointer to a resampled pointer that fits for primary
110 * secmixpos is used to decide which freqAcc is needed
111 * overshot tells what the 'actual' secpos is now (optional)
113 DWORD DSOUND_secpos_to_bufpos(const IDirectSoundBufferImpl *dsb, DWORD secpos, DWORD secmixpos, DWORD* overshot)
115 DWORD64 framelen = secpos / dsb->pwfx->nBlockAlign;
116 DWORD64 freqAdjust = dsb->freqAdjust;
117 DWORD64 acc, freqAcc;
119 if (secpos < secmixpos)
120 freqAcc = dsb->freqAccNext;
121 else freqAcc = dsb->freqAcc;
122 acc = (framelen << DSOUND_FREQSHIFT) + (freqAdjust - 1 - freqAcc);
126 DWORD64 oshot = acc * freqAdjust + freqAcc;
127 assert(oshot >= framelen << DSOUND_FREQSHIFT);
128 oshot -= framelen << DSOUND_FREQSHIFT;
129 *overshot = (DWORD)oshot;
130 assert(*overshot < dsb->freqAdjust);
132 return (DWORD)acc * dsb->device->pwfx->nBlockAlign;
135 /** Convert a resampled pointer that fits for primary to a 'native' sample pointer
136 * freqAccNext is used here rather than freqAcc: In case the app wants to fill up to
137 * the play position it won't overwrite it
139 static DWORD DSOUND_bufpos_to_secpos(const IDirectSoundBufferImpl *dsb, DWORD bufpos)
141 DWORD oAdv = dsb->device->pwfx->nBlockAlign, iAdv = dsb->pwfx->nBlockAlign, pos;
145 framelen = bufpos/oAdv;
146 acc = framelen * (DWORD64)dsb->freqAdjust + (DWORD64)dsb->freqAccNext;
147 acc = acc >> DSOUND_FREQSHIFT;
148 pos = (DWORD)acc * iAdv;
149 if (pos >= dsb->buflen)
150 /* Because of differences between freqAcc and freqAccNext, this might happen */
151 pos = dsb->buflen - iAdv;
152 TRACE("Converted %d/%d to %d/%d\n", bufpos, dsb->tmp_buffer_len, pos, dsb->buflen);
157 * Move freqAccNext to freqAcc, and find new values for buffer length and freqAccNext
159 static void DSOUND_RecalcFreqAcc(IDirectSoundBufferImpl *dsb)
161 if (!dsb->freqneeded) return;
162 dsb->freqAcc = dsb->freqAccNext;
163 dsb->tmp_buffer_len = DSOUND_secpos_to_bufpos(dsb, dsb->buflen, 0, &dsb->freqAccNext);
164 TRACE("New freqadjust: %04x, new buflen: %d\n", dsb->freqAccNext, dsb->tmp_buffer_len);
168 * Recalculate the size for temporary buffer, and new writelead
169 * Should be called when one of the following things occur:
170 * - Primary buffer format is changed
171 * - This buffer format (frequency) is changed
173 * After this, DSOUND_MixToTemporary(dsb, 0, dsb->buflen) should
174 * be called to refill the temporary buffer with data.
176 void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
178 BOOL needremix = TRUE, needresample = (dsb->freq != dsb->device->pwfx->nSamplesPerSec);
179 DWORD bAlign = dsb->pwfx->nBlockAlign, pAlign = dsb->device->pwfx->nBlockAlign;
183 /* calculate the 10ms write lead */
184 dsb->writelead = (dsb->freq / 100) * dsb->pwfx->nBlockAlign;
186 if ((dsb->pwfx->wBitsPerSample == dsb->device->pwfx->wBitsPerSample) &&
187 (dsb->pwfx->nChannels == dsb->device->pwfx->nChannels) && !needresample)
189 HeapFree(GetProcessHeap(), 0, dsb->tmp_buffer);
190 dsb->tmp_buffer = NULL;
191 dsb->max_buffer_len = dsb->freqAcc = dsb->freqAccNext = 0;
192 dsb->freqneeded = needresample;
194 dsb->convert = convertbpp[dsb->pwfx->wBitsPerSample/8 - 1][dsb->device->pwfx->wBitsPerSample/8 - 1];
199 DSOUND_RecalcFreqAcc(dsb);
201 dsb->tmp_buffer_len = dsb->buflen / bAlign * pAlign;
202 dsb->max_buffer_len = dsb->tmp_buffer_len;
203 dsb->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, dsb->max_buffer_len);
204 FillMemory(dsb->tmp_buffer, dsb->tmp_buffer_len, dsb->device->pwfx->wBitsPerSample == 8 ? 128 : 0);
206 else dsb->max_buffer_len = dsb->tmp_buffer_len = dsb->buflen;
207 dsb->buf_mixpos = DSOUND_secpos_to_bufpos(dsb, dsb->sec_mixpos, 0, NULL);
211 * Check for application callback requests for when the play position
212 * reaches certain points.
214 * The offsets that will be triggered will be those between the recorded
215 * "last played" position for the buffer (i.e. dsb->playpos) and "len" bytes
216 * beyond that position.
218 void DSOUND_CheckEvent(const IDirectSoundBufferImpl *dsb, DWORD playpos, int len)
222 LPDSBPOSITIONNOTIFY event;
223 TRACE("(%p,%d)\n",dsb,len);
225 if (dsb->nrofnotifies == 0)
228 TRACE("(%p) buflen = %d, playpos = %d, len = %d\n",
229 dsb, dsb->buflen, playpos, len);
230 for (i = 0; i < dsb->nrofnotifies ; i++) {
231 event = dsb->notifies + i;
232 offset = event->dwOffset;
233 TRACE("checking %d, position %d, event = %p\n",
234 i, offset, event->hEventNotify);
235 /* DSBPN_OFFSETSTOP has to be the last element. So this is */
236 /* OK. [Inside DirectX, p274] */
238 /* This also means we can't sort the entries by offset, */
239 /* because DSBPN_OFFSETSTOP == -1 */
240 if (offset == DSBPN_OFFSETSTOP) {
241 if (dsb->state == STATE_STOPPED) {
242 SetEvent(event->hEventNotify);
243 TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
248 if ((playpos + len) >= dsb->buflen) {
249 if ((offset < ((playpos + len) % dsb->buflen)) ||
250 (offset >= playpos)) {
251 TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
252 SetEvent(event->hEventNotify);
255 if ((offset >= playpos) && (offset < (playpos + len))) {
256 TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
257 SetEvent(event->hEventNotify);
264 * Copy a single frame from the given input buffer to the given output buffer.
265 * Translate 8 <-> 16 bits and mono <-> stereo
267 static inline void cp_fields(const IDirectSoundBufferImpl *dsb, const BYTE *ibuf, BYTE *obuf )
269 DirectSoundDevice *device = dsb->device;
270 INT istep = dsb->pwfx->wBitsPerSample / 8, ostep = device->pwfx->wBitsPerSample / 8;
272 if (device->pwfx->nChannels == dsb->pwfx->nChannels) {
273 dsb->convert(ibuf, obuf);
274 if (device->pwfx->nChannels == 2)
275 dsb->convert(ibuf + istep, obuf + ostep);
278 if (device->pwfx->nChannels == 1 && dsb->pwfx->nChannels == 2)
280 dsb->convert(ibuf, obuf);
283 if (device->pwfx->nChannels == 2 && dsb->pwfx->nChannels == 1)
285 dsb->convert(ibuf, obuf);
286 dsb->convert(ibuf, obuf + ostep);
291 * Calculate the distance between two buffer offsets, taking wraparound
294 static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
296 /* If these asserts fail, the problem is not here, but in the underlying code */
297 assert(ptr1 < buflen);
298 assert(ptr2 < buflen);
302 return buflen + ptr1 - ptr2;
306 * Mix at most the given amount of data into the allocated temporary buffer
307 * of the given secondary buffer, starting from the dsb's first currently
308 * unsampled frame (writepos), translating frequency (pitch), stereo/mono
309 * and bits-per-sample so that it is ideal for the primary buffer.
310 * Doesn't perform any mixing - this is a straight copy/convert operation.
312 * dsb = the secondary buffer
313 * writepos = Starting position of changed buffer
314 * len = number of bytes to resample from writepos
316 * NOTE: writepos + len <= buflen, This function doesn't loop!
318 void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD len)
321 BYTE *ibp, *obp, *ibp_begin, *obp_begin;
322 INT iAdvance = dsb->pwfx->nBlockAlign;
323 INT oAdvance = dsb->device->pwfx->nBlockAlign;
324 DWORD freqAcc, target_writepos, overshot;
326 if (!dsb->tmp_buffer)
327 /* Nothing to do, already ideal format */
330 ibp = dsb->buffer->memory + writepos;
331 ibp_begin = dsb->buffer->memory;
332 obp_begin = dsb->tmp_buffer;
334 TRACE("(%p, %p)\n", dsb, ibp);
335 /* Check for the best case */
336 if ((dsb->freq == dsb->device->pwfx->nSamplesPerSec) &&
337 (dsb->pwfx->wBitsPerSample == dsb->device->pwfx->wBitsPerSample) &&
338 (dsb->pwfx->nChannels == dsb->device->pwfx->nChannels)) {
339 obp = dsb->tmp_buffer + writepos;
340 /* Why would we need a temporary buffer if we do best case? */
341 FIXME("(%p) Why do we resample for best case??? Bad!!\n", dsb);
342 CopyMemory(obp, ibp, len);
346 /* Check for same sample rate */
347 if (dsb->freq == dsb->device->pwfx->nSamplesPerSec) {
348 TRACE("(%p) Same sample rate %d = primary %d\n", dsb,
349 dsb->freq, dsb->device->pwfx->nSamplesPerSec);
350 obp = dsb->tmp_buffer + writepos/iAdvance*oAdvance;
351 for (i = 0; i < len; i += iAdvance) {
352 cp_fields(dsb, ibp, obp);
359 /* Mix in different sample rates */
360 TRACE("(%p) Adjusting frequency: %d -> %d\n", dsb, dsb->freq, dsb->device->pwfx->nSamplesPerSec);
361 size = len / iAdvance;
363 target_writepos = DSOUND_secpos_to_bufpos(dsb, writepos, dsb->sec_mixpos, &freqAcc);
364 overshot = freqAcc >> DSOUND_FREQSHIFT;
367 if (overshot >= size)
370 writepos += overshot * iAdvance;
371 if (writepos >= dsb->buflen)
373 ibp = dsb->buffer->memory + writepos;
374 freqAcc &= (1 << DSOUND_FREQSHIFT) - 1;
375 TRACE("Overshot: %d, freqAcc: %04x\n", overshot, freqAcc);
378 obp = dsb->tmp_buffer + target_writepos;
379 /* FIXME: Small problem here when we're overwriting buf_mixpos, it then STILL uses old freqAcc, not sure if it matters or not */
381 cp_fields(dsb, ibp, obp);
383 freqAcc += dsb->freqAdjust;
384 if (freqAcc >= (1<<DSOUND_FREQSHIFT)) {
385 ULONG adv = (freqAcc>>DSOUND_FREQSHIFT);
386 freqAcc &= (1<<DSOUND_FREQSHIFT)-1;
387 ibp += adv * iAdvance;
393 /** Apply volume to the given soundbuffer from (primary) position writepos and length len
394 * Returns: NULL if no volume needs to be applied
395 * or else a memory handle that holds 'len' volume adjusted buffer */
396 static LPBYTE DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, DWORD writepos, INT len)
402 INT nChannels = dsb->device->pwfx->nChannels;
403 LPBYTE mem = (dsb->tmp_buffer ? dsb->tmp_buffer : dsb->buffer->memory)+writepos;
405 TRACE("(%p,%d)\n",dsb,len);
406 TRACE("left = %x, right = %x\n", dsb->volpan.dwTotalLeftAmpFactor,
407 dsb->volpan.dwTotalRightAmpFactor);
409 if ((!(dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) || (dsb->volpan.lPan == 0)) &&
410 (!(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) || (dsb->volpan.lVolume == 0)) &&
411 !(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
412 return NULL; /* Nothing to do */
414 if (nChannels != 1 && nChannels != 2)
416 FIXME("There is no support for %d channels\n", nChannels);
420 if (dsb->device->pwfx->wBitsPerSample != 8 && dsb->device->pwfx->wBitsPerSample != 16)
422 FIXME("There is no support for %d bpp\n", dsb->device->pwfx->wBitsPerSample);
426 if (dsb->device->tmp_buffer_len < len || !dsb->device->tmp_buffer)
428 dsb->device->tmp_buffer_len = len;
429 if (dsb->device->tmp_buffer)
430 dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, len);
432 dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, len);
434 bpc = dsb->device->tmp_buffer;
437 vLeft = dsb->volpan.dwTotalLeftAmpFactor;
439 vRight = dsb->volpan.dwTotalRightAmpFactor;
443 switch (dsb->device->pwfx->wBitsPerSample) {
445 /* 8-bit WAV is unsigned, but we need to operate */
446 /* on signed data for this to work properly */
447 for (i = 0; i < len; i+=2) {
448 *(bpc++) = (((*(mem++) - 128) * vLeft) >> 16) + 128;
449 *(bpc++) = (((*(mem++) - 128) * vRight) >> 16) + 128;
451 if (len % 2 == 1 && nChannels == 1)
452 *(bpc++) = (((*(mem++) - 128) * vLeft) >> 16) + 128;
455 /* 16-bit WAV is signed -- much better */
456 for (i = 0; i < len; i += 4) {
457 *(bps++) = (*(mems++) * vLeft) >> 16;
458 *(bps++) = (*(mems++) * vRight) >> 16;
460 if (len % 4 == 2 && nChannels == 1)
461 *(bps++) = ((INT)*(mems++) * vLeft) >> 16;
464 return dsb->device->tmp_buffer;
468 * Mix (at most) the given number of bytes into the given position of the
469 * device buffer, from the secondary buffer "dsb" (starting at the current
470 * mix position for that buffer).
472 * Returns the number of bytes actually mixed into the device buffer. This
473 * will match fraglen unless the end of the secondary buffer is reached
474 * (and it is not looping).
476 * dsb = the secondary buffer to mix from
477 * writepos = position (offset) in device buffer to write at
478 * fraglen = number of bytes to mix
480 static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen)
482 INT len = fraglen, ilen;
483 BYTE *ibuf = (dsb->tmp_buffer ? dsb->tmp_buffer : dsb->buffer->memory) + dsb->buf_mixpos, *volbuf;
484 DWORD oldpos, mixbufpos;
486 TRACE("buf_mixpos=%d/%d sec_mixpos=%d/%d\n", dsb->buf_mixpos, dsb->tmp_buffer_len, dsb->sec_mixpos, dsb->buflen);
487 TRACE("(%p,%d,%d)\n",dsb,writepos,fraglen);
489 assert(dsb->buf_mixpos + len <= dsb->tmp_buffer_len);
491 if (len % dsb->device->pwfx->nBlockAlign) {
492 INT nBlockAlign = dsb->device->pwfx->nBlockAlign;
493 ERR("length not a multiple of block size, len = %d, block size = %d\n", len, nBlockAlign);
494 len -= len % nBlockAlign; /* data alignment */
497 /* Apply volume if needed */
498 volbuf = DSOUND_MixerVol(dsb, dsb->buf_mixpos, len);
502 mixbufpos = DSOUND_bufpos_to_mixpos(dsb->device, writepos);
503 /* Now mix the temporary buffer into the devices main buffer */
504 if ((writepos + len) <= dsb->device->buflen)
505 dsb->device->mixfunction(ibuf, dsb->device->mix_buffer + mixbufpos, len);
508 DWORD todo = dsb->device->buflen - writepos;
509 dsb->device->mixfunction(ibuf, dsb->device->mix_buffer + mixbufpos, todo);
510 dsb->device->mixfunction(ibuf + todo, dsb->device->mix_buffer, len - todo);
513 oldpos = dsb->sec_mixpos;
514 dsb->buf_mixpos += len;
516 if (dsb->buf_mixpos >= dsb->tmp_buffer_len) {
517 if (dsb->buf_mixpos > dsb->tmp_buffer_len)
518 ERR("Mixpos (%u) past buflen (%u), capping...\n", dsb->buf_mixpos, dsb->tmp_buffer_len);
519 if (dsb->playflags & DSBPLAY_LOOPING) {
520 dsb->buf_mixpos -= dsb->tmp_buffer_len;
521 } else if (dsb->buf_mixpos >= dsb->tmp_buffer_len) {
522 dsb->buf_mixpos = dsb->sec_mixpos = 0;
523 dsb->state = STATE_STOPPED;
525 DSOUND_RecalcFreqAcc(dsb);
528 dsb->sec_mixpos = DSOUND_bufpos_to_secpos(dsb, dsb->buf_mixpos);
529 ilen = DSOUND_BufPtrDiff(dsb->buflen, dsb->sec_mixpos, oldpos);
530 /* check for notification positions */
531 if (dsb->dsbd.dwFlags & DSBCAPS_CTRLPOSITIONNOTIFY &&
532 dsb->state != STATE_STARTING) {
533 DSOUND_CheckEvent(dsb, oldpos, ilen);
536 /* increase mix position */
537 dsb->primary_mixpos += len;
538 if (dsb->primary_mixpos >= dsb->device->buflen)
539 dsb->primary_mixpos -= dsb->device->buflen;
544 * Mix some frames from the given secondary buffer "dsb" into the device
547 * dsb = the secondary buffer
548 * playpos = the current play position in the device buffer (primary buffer)
549 * writepos = the current safe-to-write position in the device buffer
550 * mixlen = the maximum number of bytes in the primary buffer to mix, from the
553 * Returns: the number of bytes beyond the writepos that were mixed.
555 static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD mixlen)
557 /* The buffer's primary_mixpos may be before or after the device
558 * buffer's mixpos, but both must be ahead of writepos. */
561 TRACE("(%p,%d,%d)\n",dsb,writepos,mixlen);
562 TRACE("writepos=%d, buf_mixpos=%d, primary_mixpos=%d, mixlen=%d\n", writepos, dsb->buf_mixpos, dsb->primary_mixpos, mixlen);
563 TRACE("looping=%d, leadin=%d, buflen=%d\n", dsb->playflags, dsb->leadin, dsb->tmp_buffer_len);
565 /* If leading in, only mix about 20 ms, and 'skip' mixing the rest, for more fluid pointer advancement */
566 if (dsb->leadin && dsb->state == STATE_STARTING)
568 if (mixlen > 2 * dsb->device->fraglen)
570 dsb->primary_mixpos += mixlen - 2 * dsb->device->fraglen;
571 dsb->primary_mixpos %= dsb->device->buflen;
576 /* calculate how much pre-buffering has already been done for this buffer */
577 primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);
580 if(mixlen < primary_done)
582 /* Should *NEVER* happen */
583 ERR("Fatal error. Under/Overflow? primary_done=%d, mixpos=%d/%d (%d/%d), primary_mixpos=%d, writepos=%d, mixlen=%d\n", primary_done,dsb->buf_mixpos,dsb->tmp_buffer_len,dsb->sec_mixpos, dsb->buflen, dsb->primary_mixpos, writepos, mixlen);
587 /* take into acount already mixed data */
588 mixlen -= primary_done;
590 TRACE("primary_done=%d, mixlen (primary) = %i\n", primary_done, mixlen);
595 /* First try to mix to the end of the buffer if possible
596 * Theoretically it would allow for better optimization
598 if (mixlen + dsb->buf_mixpos >= dsb->tmp_buffer_len)
600 DWORD newmixed, mixfirst = dsb->tmp_buffer_len - dsb->buf_mixpos;
601 newmixed = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixfirst);
604 if (dsb->playflags & DSBPLAY_LOOPING)
605 while (newmixed && mixlen)
607 mixfirst = (dsb->tmp_buffer_len < mixlen ? dsb->tmp_buffer_len : mixlen);
608 newmixed = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixfirst);
612 else DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixlen);
614 /* re-calculate the primary done */
615 primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);
617 TRACE("new primary_mixpos=%d, total mixed data=%d\n", dsb->primary_mixpos, primary_done);
619 /* Report back the total prebuffered amount for this buffer */
624 * For a DirectSoundDevice, go through all the currently playing buffers and
625 * mix them in to the device buffer.
627 * writepos = the current safe-to-write position in the primary buffer
628 * mixlen = the maximum amount to mix into the primary buffer
629 * (beyond the current writepos)
630 * mustlock = Do we have to fight for lock because we otherwise risk an underrun?
631 * recover = true if the sound device may have been reset and the write
632 * position in the device buffer changed
633 * all_stopped = reports back if all buffers have stopped
635 * Returns: the length beyond the writepos that was mixed to.
638 static DWORD DSOUND_MixToPrimary(const DirectSoundDevice *device, DWORD writepos, DWORD mixlen, BOOL mustlock, BOOL recover, BOOL *all_stopped)
642 IDirectSoundBufferImpl *dsb;
645 /* unless we find a running buffer, all have stopped */
648 TRACE("(%d,%d,%d)\n", writepos, mixlen, recover);
649 for (i = 0; i < device->nrofbuffers; i++) {
650 dsb = device->buffers[i];
652 TRACE("MixToPrimary for %p, state=%d\n", dsb, dsb->state);
654 if (dsb->buflen && dsb->state && !dsb->hwbuf) {
655 TRACE("Checking %p, mixlen=%d\n", dsb, mixlen);
656 if (!RtlAcquireResourceShared(&dsb->lock, mustlock))
661 /* if buffer is stopping it is stopped now */
662 if (dsb->state == STATE_STOPPING) {
663 dsb->state = STATE_STOPPED;
664 DSOUND_CheckEvent(dsb, 0, 0);
665 } else if (dsb->state != STATE_STOPPED) {
667 /* if recovering, reset the mix position */
668 if ((dsb->state == STATE_STARTING) || recover) {
669 dsb->primary_mixpos = writepos;
672 /* mix next buffer into the main buffer */
673 len = DSOUND_MixOne(dsb, writepos, mixlen);
675 /* if the buffer was starting, it must be playing now */
676 if (dsb->state == STATE_STARTING)
677 dsb->state = STATE_PLAYING;
679 if (!minlen) minlen = len;
681 /* record the minimum length mixed from all buffers */
682 /* we only want to return the length which *all* buffers have mixed */
683 else if (len) minlen = (len < minlen) ? len : minlen;
685 *all_stopped = FALSE;
687 RtlReleaseResource(&dsb->lock);
691 TRACE("Mixed at least %d from all buffers\n", minlen);
692 if (!gotall) return 0;
697 * Add buffers to the emulated wave device system.
699 * device = The current dsound playback device
700 * force = If TRUE, the function will buffer up as many frags as possible,
701 * even though and will ignore the actual state of the primary buffer.
706 static void DSOUND_WaveQueue(DirectSoundDevice *device, BOOL force)
708 DWORD prebuf_frags, wave_writepos, wave_fragpos, i;
709 TRACE("(%p)\n", device);
711 /* calculate the current wave frag position */
712 wave_fragpos = (device->pwplay + device->pwqueue) % device->helfrags;
714 /* calculte the current wave write position */
715 wave_writepos = wave_fragpos * device->fraglen;
717 TRACE("wave_fragpos = %i, wave_writepos = %i, pwqueue = %i, prebuf = %i\n",
718 wave_fragpos, wave_writepos, device->pwqueue, device->prebuf);
722 /* check remaining prebuffered frags */
723 prebuf_frags = device->mixpos / device->fraglen;
724 if (prebuf_frags == device->helfrags)
726 TRACE("wave_fragpos = %d, mixpos_frags = %d\n", wave_fragpos, prebuf_frags);
727 if (prebuf_frags < wave_fragpos)
728 prebuf_frags += device->helfrags;
729 prebuf_frags -= wave_fragpos;
730 TRACE("wanted prebuf_frags = %d\n", prebuf_frags);
733 /* buffer the maximum amount of frags */
734 prebuf_frags = device->prebuf;
736 /* limit to the queue we have left */
737 if ((prebuf_frags + device->pwqueue) > device->prebuf)
738 prebuf_frags = device->prebuf - device->pwqueue;
740 TRACE("prebuf_frags = %i\n", prebuf_frags);
743 device->pwqueue += prebuf_frags;
745 /* get out of CS when calling the wave system */
746 LeaveCriticalSection(&(device->mixlock));
749 /* queue up the new buffers */
750 for(i=0; i<prebuf_frags; i++){
751 TRACE("queueing wave buffer %i\n", wave_fragpos);
752 waveOutWrite(device->hwo, &device->pwave[wave_fragpos], sizeof(WAVEHDR));
754 wave_fragpos %= device->helfrags;
758 EnterCriticalSection(&(device->mixlock));
760 TRACE("queue now = %i\n", device->pwqueue);
764 * Perform mixing for a Direct Sound device. That is, go through all the
765 * secondary buffers (the sound bites currently playing) and mix them in
766 * to the primary buffer (the device buffer).
768 static void DSOUND_PerformMix(DirectSoundDevice *device)
770 TRACE("(%p)\n", device);
773 EnterCriticalSection(&(device->mixlock));
775 if (device->priolevel != DSSCL_WRITEPRIMARY) {
776 BOOL recover = FALSE, all_stopped = FALSE;
777 DWORD playpos, writepos, writelead, maxq, frag, prebuff_max, prebuff_left, size1, size2, mixplaypos, mixplaypos2;
779 BOOL lock = (device->hwbuf && !(device->drvdesc.dwFlags & DSDDESC_DONTNEEDPRIMARYLOCK));
780 BOOL mustlock = FALSE;
783 /* the sound of silence */
784 nfiller = device->pwfx->wBitsPerSample == 8 ? 128 : 0;
786 /* get the position in the primary buffer */
787 if (DSOUND_PrimaryGetPosition(device, &playpos, &writepos) != 0){
788 LeaveCriticalSection(&(device->mixlock));
792 TRACE("primary playpos=%d, writepos=%d, clrpos=%d, mixpos=%d, buflen=%d\n",
793 playpos,writepos,device->playpos,device->mixpos,device->buflen);
794 assert(device->playpos < device->buflen);
796 mixplaypos = DSOUND_bufpos_to_mixpos(device, device->playpos);
797 mixplaypos2 = DSOUND_bufpos_to_mixpos(device, playpos);
798 /* wipe out just-played sound data */
799 if (playpos < device->playpos) {
800 buf1 = device->buffer + device->playpos;
801 buf2 = device->buffer;
802 size1 = device->buflen - device->playpos;
804 FillMemory(device->mix_buffer + mixplaypos, device->mix_buffer_len - mixplaypos, 0);
805 FillMemory(device->mix_buffer, mixplaypos2, 0);
807 IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, device->playpos, size1+size2, 0);
808 FillMemory(buf1, size1, nfiller);
809 if (playpos && (!buf2 || !size2))
810 FIXME("%d: (%d, %d)=>(%d, %d) There should be an additional buffer here!!\n", __LINE__, device->playpos, device->mixpos, playpos, writepos);
811 FillMemory(buf2, size2, nfiller);
813 IDsDriverBuffer_Unlock(device->hwbuf, buf1, size1, buf2, size2);
815 buf1 = device->buffer + device->playpos;
817 size1 = playpos - device->playpos;
819 FillMemory(device->mix_buffer + mixplaypos, mixplaypos2 - mixplaypos, 0);
821 IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, device->playpos, size1+size2, 0);
822 FillMemory(buf1, size1, nfiller);
825 FIXME("%d: There should be no additional buffer here!!\n", __LINE__);
826 FillMemory(buf2, size2, nfiller);
829 IDsDriverBuffer_Unlock(device->hwbuf, buf1, size1, buf2, size2);
831 device->playpos = playpos;
833 /* calc maximum prebuff */
834 prebuff_max = (device->prebuf * device->fraglen);
835 if (!device->hwbuf && playpos + prebuff_max >= device->helfrags * device->fraglen)
836 prebuff_max += device->buflen - device->helfrags * device->fraglen;
838 /* check how close we are to an underrun. It occurs when the writepos overtakes the mixpos */
839 prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);
840 writelead = DSOUND_BufPtrDiff(device->buflen, writepos, playpos);
842 /* find the maximum we can prebuffer from current write position */
843 maxq = (writelead < prebuff_max) ? (prebuff_max - writelead) : 0;
845 TRACE("prebuff_left = %d, prebuff_max = %dx%d=%d, writelead=%d\n",
846 prebuff_left, device->prebuf, device->fraglen, prebuff_max, writelead);
848 /* check for underrun. underrun occurs when the write position passes the mix position */
849 if((prebuff_left > prebuff_max) || (device->state == STATE_STOPPED) || (device->state == STATE_STARTING)){
850 if (device->state == STATE_STOPPING || device->state == STATE_PLAYING)
851 WARN("Probable buffer underrun\n");
852 else TRACE("Buffer starting or buffer underrun\n");
854 /* recover mixing for all buffers */
857 /* reset mix position to write position */
858 device->mixpos = writepos;
861 /* Do we risk an 'underrun' if we don't advance pointer? */
862 if (writelead/device->fraglen <= ds_snd_queue_min || recover)
866 IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, writepos, maxq, 0);
869 frag = DSOUND_MixToPrimary(device, writepos, maxq, mustlock, recover, &all_stopped);
871 if (frag + writepos > device->buflen)
873 DWORD todo = device->buflen - writepos;
874 device->normfunction(device->mix_buffer + DSOUND_bufpos_to_mixpos(device, writepos), device->buffer + writepos, todo);
875 device->normfunction(device->mix_buffer, device->buffer, frag - todo);
878 device->normfunction(device->mix_buffer + DSOUND_bufpos_to_mixpos(device, writepos), device->buffer + writepos, frag);
880 /* update the mix position, taking wrap-around into acount */
881 device->mixpos = writepos + frag;
882 device->mixpos %= device->buflen;
886 DWORD frag2 = (frag > size1 ? frag - size1 : 0);
890 FIXME("Buffering too much! (%d, %d, %d, %d)\n", maxq, frag, size2, frag2 - size2);
893 IDsDriverBuffer_Unlock(device->hwbuf, buf1, frag, buf2, frag2);
896 /* update prebuff left */
897 prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);
899 /* check if have a whole fragment */
900 if (prebuff_left >= device->fraglen){
902 /* update the wave queue if using wave system */
904 DSOUND_WaveQueue(device, FALSE);
906 /* buffers are full. start playing if applicable */
907 if(device->state == STATE_STARTING){
908 TRACE("started primary buffer\n");
909 if(DSOUND_PrimaryPlay(device) != DS_OK){
910 WARN("DSOUND_PrimaryPlay failed\n");
913 /* we are playing now */
914 device->state = STATE_PLAYING;
918 /* buffers are full. start stopping if applicable */
919 if(device->state == STATE_STOPPED){
920 TRACE("restarting primary buffer\n");
921 if(DSOUND_PrimaryPlay(device) != DS_OK){
922 WARN("DSOUND_PrimaryPlay failed\n");
925 /* start stopping again. as soon as there is no more data, it will stop */
926 device->state = STATE_STOPPING;
931 /* if device was stopping, its for sure stopped when all buffers have stopped */
932 else if((all_stopped == TRUE) && (device->state == STATE_STOPPING)){
933 TRACE("All buffers have stopped. Stopping primary buffer\n");
934 device->state = STATE_STOPPED;
936 /* stop the primary buffer now */
937 DSOUND_PrimaryStop(device);
942 /* update the wave queue if using wave system */
944 DSOUND_WaveQueue(device, TRUE);
946 /* Keep alsa happy, which needs GetPosition called once every 10 ms */
947 IDsDriverBuffer_GetPosition(device->hwbuf, NULL, NULL);
949 /* in the DSSCL_WRITEPRIMARY mode, the app is totally in charge... */
950 if (device->state == STATE_STARTING) {
951 if (DSOUND_PrimaryPlay(device) != DS_OK)
952 WARN("DSOUND_PrimaryPlay failed\n");
954 device->state = STATE_PLAYING;
956 else if (device->state == STATE_STOPPING) {
957 if (DSOUND_PrimaryStop(device) != DS_OK)
958 WARN("DSOUND_PrimaryStop failed\n");
960 device->state = STATE_STOPPED;
964 LeaveCriticalSection(&(device->mixlock));
968 void CALLBACK DSOUND_timer(UINT timerID, UINT msg, DWORD_PTR dwUser,
969 DWORD_PTR dw1, DWORD_PTR dw2)
971 DirectSoundDevice * device = (DirectSoundDevice*)dwUser;
972 DWORD start_time = GetTickCount();
974 TRACE("(%d,%d,0x%lx,0x%lx,0x%lx)\n",timerID,msg,dwUser,dw1,dw2);
975 TRACE("entering at %d\n", start_time);
977 if (DSOUND_renderer[device->drvdesc.dnDevNode] != device) {
978 ERR("dsound died without killing us?\n");
979 timeKillEvent(timerID);
980 timeEndPeriod(DS_TIME_RES);
984 RtlAcquireResourceShared(&(device->buffer_list_lock), TRUE);
987 DSOUND_PerformMix(device);
989 RtlReleaseResource(&(device->buffer_list_lock));
991 end_time = GetTickCount();
992 TRACE("completed processing at %d, duration = %d\n", end_time, end_time - start_time);
995 void CALLBACK DSOUND_callback(HWAVEOUT hwo, UINT msg, DWORD dwUser, DWORD dw1, DWORD dw2)
997 DirectSoundDevice * device = (DirectSoundDevice*)dwUser;
998 TRACE("(%p,%x,%x,%x,%x)\n",hwo,msg,dwUser,dw1,dw2);
999 TRACE("entering at %d, msg=%08x(%s)\n", GetTickCount(), msg,
1000 msg==MM_WOM_DONE ? "MM_WOM_DONE" : msg==MM_WOM_CLOSE ? "MM_WOM_CLOSE" :
1001 msg==MM_WOM_OPEN ? "MM_WOM_OPEN" : "UNKNOWN");
1003 /* check if packet completed from wave driver */
1004 if (msg == MM_WOM_DONE) {
1007 EnterCriticalSection(&(device->mixlock));
1009 TRACE("done playing primary pos=%d\n", device->pwplay * device->fraglen);
1011 /* update playpos */
1013 device->pwplay %= device->helfrags;
1016 if(device->pwqueue == 0){
1017 ERR("Wave queue corrupted!\n");
1023 LeaveCriticalSection(&(device->mixlock));
1026 TRACE("completed\n");