quartz: Handle state changes in direct sound renderer correctly.
[wine/multimedia.git] / dlls / dsound / mixer.c
blob185df6e1ff65019b37aed603d8bb94938e0dc5a6
1 /* DirectSound
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
24 #include <assert.h>
25 #include <stdarg.h>
26 #include <math.h> /* Insomnia - pow() function */
28 #define NONAMELESSSTRUCT
29 #define NONAMELESSUNION
30 #include "windef.h"
31 #include "winbase.h"
32 #include "mmsystem.h"
33 #include "winternl.h"
34 #include "wine/debug.h"
35 #include "dsound.h"
36 #include "dsdriver.h"
37 #include "dsound_private.h"
39 WINE_DEFAULT_DEBUG_CHANNEL(dsound);
41 void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan)
43 double temp;
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)
62 double left,right;
63 TRACE("(%p)\n",volpan);
65 TRACE("left=%x, right=%x\n",volpan->dwTotalLeftAmpFactor,volpan->dwTotalRightAmpFactor);
66 if (volpan->dwTotalLeftAmpFactor==0)
67 left=-10000;
68 else
69 left=600 * log(((double)volpan->dwTotalLeftAmpFactor) / 0xffff) / log(2);
70 if (volpan->dwTotalRightAmpFactor==0)
71 right=-10000;
72 else
73 right=600 * log(((double)volpan->dwTotalRightAmpFactor) / 0xffff) / log(2);
74 if (left<right)
76 volpan->lVolume=right;
77 volpan->dwVolAmpFactor=volpan->dwTotalRightAmpFactor;
79 else
81 volpan->lVolume=left;
82 volpan->dwVolAmpFactor=volpan->dwTotalLeftAmpFactor;
84 if (volpan->lVolume < -10000)
85 volpan->lVolume=-10000;
86 volpan->lPan=right-left;
87 if (volpan->lPan < -10000)
88 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)
102 ret *= 2;
103 return ret;
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);
123 acc /= freqAdjust;
124 if (overshot)
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;
142 DWORD64 framelen;
143 DWORD64 acc;
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);
153 return pos;
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;
181 TRACE("(%p)\n",dsb);
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)
188 needremix = FALSE;
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];
196 dsb->resampleinmixer = FALSE;
198 if (needremix)
200 if (needresample)
201 DSOUND_RecalcFreqAcc(dsb);
202 else
203 dsb->tmp_buffer_len = dsb->buflen / bAlign * pAlign;
204 dsb->max_buffer_len = dsb->tmp_buffer_len;
205 if ((dsb->max_buffer_len <= dsb->device->buflen || dsb->max_buffer_len < ds_snd_shadow_maxsize * 1024 * 1024) && ds_snd_shadow_maxsize >= 0)
206 dsb->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, dsb->max_buffer_len);
207 if (dsb->tmp_buffer)
208 FillMemory(dsb->tmp_buffer, dsb->tmp_buffer_len, dsb->device->pwfx->wBitsPerSample == 8 ? 128 : 0);
209 else
210 dsb->resampleinmixer = TRUE;
212 else dsb->max_buffer_len = dsb->tmp_buffer_len = dsb->buflen;
213 dsb->buf_mixpos = DSOUND_secpos_to_bufpos(dsb, dsb->sec_mixpos, 0, NULL);
217 * Check for application callback requests for when the play position
218 * reaches certain points.
220 * The offsets that will be triggered will be those between the recorded
221 * "last played" position for the buffer (i.e. dsb->playpos) and "len" bytes
222 * beyond that position.
224 void DSOUND_CheckEvent(const IDirectSoundBufferImpl *dsb, DWORD playpos, int len)
226 int i;
227 DWORD offset;
228 LPDSBPOSITIONNOTIFY event;
229 TRACE("(%p,%d)\n",dsb,len);
231 if (dsb->nrofnotifies == 0)
232 return;
234 TRACE("(%p) buflen = %d, playpos = %d, len = %d\n",
235 dsb, dsb->buflen, playpos, len);
236 for (i = 0; i < dsb->nrofnotifies ; i++) {
237 event = dsb->notifies + i;
238 offset = event->dwOffset;
239 TRACE("checking %d, position %d, event = %p\n",
240 i, offset, event->hEventNotify);
241 /* DSBPN_OFFSETSTOP has to be the last element. So this is */
242 /* OK. [Inside DirectX, p274] */
243 /* Windows does not seem to enforce this, and some apps rely */
244 /* on that, so we can't stop there. */
245 /* */
246 /* This also means we can't sort the entries by offset, */
247 /* because DSBPN_OFFSETSTOP == -1 */
248 if (offset == DSBPN_OFFSETSTOP) {
249 if (dsb->state == STATE_STOPPED) {
250 SetEvent(event->hEventNotify);
251 TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
253 continue;
255 if ((playpos + len) >= dsb->buflen) {
256 if ((offset < ((playpos + len) % dsb->buflen)) ||
257 (offset >= playpos)) {
258 TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
259 SetEvent(event->hEventNotify);
261 } else {
262 if ((offset >= playpos) && (offset < (playpos + len))) {
263 TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
264 SetEvent(event->hEventNotify);
271 * Copy a single frame from the given input buffer to the given output buffer.
272 * Translate 8 <-> 16 bits and mono <-> stereo
274 static inline void cp_fields(const IDirectSoundBufferImpl *dsb, const BYTE *ibuf, BYTE *obuf,
275 UINT istride, UINT ostride, UINT count, UINT freqAcc, UINT adj)
277 DirectSoundDevice *device = dsb->device;
278 INT istep = dsb->pwfx->wBitsPerSample / 8, ostep = device->pwfx->wBitsPerSample / 8;
280 if (device->pwfx->nChannels == dsb->pwfx->nChannels ||
281 (device->pwfx->nChannels == 2 && dsb->pwfx->nChannels == 6)) {
282 dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
283 if (device->pwfx->nChannels == 2)
284 dsb->convert(ibuf + istep, obuf + ostep, istride, ostride, count, freqAcc, adj);
287 if (device->pwfx->nChannels == 1 && dsb->pwfx->nChannels == 2)
289 dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
292 if (device->pwfx->nChannels == 2 && dsb->pwfx->nChannels == 1)
294 dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
295 dsb->convert(ibuf, obuf + ostep, istride, ostride, count, freqAcc, adj);
300 * Calculate the distance between two buffer offsets, taking wraparound
301 * into account.
303 static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
305 /* If these asserts fail, the problem is not here, but in the underlying code */
306 assert(ptr1 < buflen);
307 assert(ptr2 < buflen);
308 if (ptr1 >= ptr2) {
309 return ptr1 - ptr2;
310 } else {
311 return buflen + ptr1 - ptr2;
315 * Mix at most the given amount of data into the allocated temporary buffer
316 * of the given secondary buffer, starting from the dsb's first currently
317 * unsampled frame (writepos), translating frequency (pitch), stereo/mono
318 * and bits-per-sample so that it is ideal for the primary buffer.
319 * Doesn't perform any mixing - this is a straight copy/convert operation.
321 * dsb = the secondary buffer
322 * writepos = Starting position of changed buffer
323 * len = number of bytes to resample from writepos
325 * NOTE: writepos + len <= buflen. When called by mixer, MixOne makes sure of this.
327 void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD len, BOOL inmixer)
329 INT size;
330 BYTE *ibp, *obp, *obp_begin;
331 INT iAdvance = dsb->pwfx->nBlockAlign;
332 INT oAdvance = dsb->device->pwfx->nBlockAlign;
333 DWORD freqAcc, target_writepos = 0, overshot, maxlen;
335 /* We resample only when needed */
336 if ((dsb->tmp_buffer && inmixer) || (!dsb->tmp_buffer && !inmixer) || dsb->resampleinmixer != inmixer)
337 return;
339 assert(writepos + len <= dsb->buflen);
340 if (inmixer && writepos + len < dsb->buflen)
341 len += dsb->pwfx->nBlockAlign;
343 maxlen = DSOUND_secpos_to_bufpos(dsb, len, 0, NULL);
345 ibp = dsb->buffer->memory + writepos;
346 if (!inmixer)
347 obp_begin = dsb->tmp_buffer;
348 else if (dsb->device->tmp_buffer_len < maxlen || !dsb->device->tmp_buffer)
350 dsb->device->tmp_buffer_len = maxlen;
351 if (dsb->device->tmp_buffer)
352 dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, maxlen);
353 else
354 dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, maxlen);
355 obp_begin = dsb->device->tmp_buffer;
357 else
358 obp_begin = dsb->device->tmp_buffer;
360 TRACE("(%p, %p)\n", dsb, ibp);
361 size = len / iAdvance;
363 /* Check for same sample rate */
364 if (dsb->freq == dsb->device->pwfx->nSamplesPerSec) {
365 TRACE("(%p) Same sample rate %d = primary %d\n", dsb,
366 dsb->freq, dsb->device->pwfx->nSamplesPerSec);
367 obp = obp_begin;
368 if (!inmixer)
369 obp += writepos/iAdvance*oAdvance;
371 cp_fields(dsb, ibp, obp, iAdvance, oAdvance, size, 0, 1 << DSOUND_FREQSHIFT);
372 return;
375 /* Mix in different sample rates */
376 TRACE("(%p) Adjusting frequency: %d -> %d\n", dsb, dsb->freq, dsb->device->pwfx->nSamplesPerSec);
378 target_writepos = DSOUND_secpos_to_bufpos(dsb, writepos, dsb->sec_mixpos, &freqAcc);
379 overshot = freqAcc >> DSOUND_FREQSHIFT;
380 if (overshot)
382 if (overshot >= size)
383 return;
384 size -= overshot;
385 writepos += overshot * iAdvance;
386 if (writepos >= dsb->buflen)
387 return;
388 ibp = dsb->buffer->memory + writepos;
389 freqAcc &= (1 << DSOUND_FREQSHIFT) - 1;
390 TRACE("Overshot: %d, freqAcc: %04x\n", overshot, freqAcc);
393 if (!inmixer)
394 obp = obp_begin + target_writepos;
395 else obp = obp_begin;
397 /* FIXME: Small problem here when we're overwriting buf_mixpos, it then STILL uses old freqAcc, not sure if it matters or not */
398 cp_fields(dsb, ibp, obp, iAdvance, oAdvance, size, freqAcc, dsb->freqAdjust);
401 /** Apply volume to the given soundbuffer from (primary) position writepos and length len
402 * Returns: NULL if no volume needs to be applied
403 * or else a memory handle that holds 'len' volume adjusted buffer */
404 static LPBYTE DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, INT len)
406 INT i;
407 BYTE *bpc;
408 INT16 *bps, *mems;
409 DWORD vLeft, vRight;
410 INT nChannels = dsb->device->pwfx->nChannels;
411 LPBYTE mem = (dsb->tmp_buffer ? dsb->tmp_buffer : dsb->buffer->memory) + dsb->buf_mixpos;
413 if (dsb->resampleinmixer)
414 mem = dsb->device->tmp_buffer;
416 TRACE("(%p,%d)\n",dsb,len);
417 TRACE("left = %x, right = %x\n", dsb->volpan.dwTotalLeftAmpFactor,
418 dsb->volpan.dwTotalRightAmpFactor);
420 if ((!(dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) || (dsb->volpan.lPan == 0)) &&
421 (!(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) || (dsb->volpan.lVolume == 0)) &&
422 !(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
423 return NULL; /* Nothing to do */
425 if (nChannels != 1 && nChannels != 2)
427 FIXME("There is no support for %d channels\n", nChannels);
428 return NULL;
431 if (dsb->device->pwfx->wBitsPerSample != 8 && dsb->device->pwfx->wBitsPerSample != 16)
433 FIXME("There is no support for %d bpp\n", dsb->device->pwfx->wBitsPerSample);
434 return NULL;
437 if (dsb->device->tmp_buffer_len < len || !dsb->device->tmp_buffer)
439 /* If we just resampled in DSOUND_MixToTemporary, we shouldn't need to resize here */
440 assert(!dsb->resampleinmixer);
441 dsb->device->tmp_buffer_len = len;
442 if (dsb->device->tmp_buffer)
443 dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, len);
444 else
445 dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, len);
448 bpc = dsb->device->tmp_buffer;
449 bps = (INT16 *)bpc;
450 mems = (INT16 *)mem;
451 vLeft = dsb->volpan.dwTotalLeftAmpFactor;
452 if (nChannels > 1)
453 vRight = dsb->volpan.dwTotalRightAmpFactor;
454 else
455 vRight = vLeft;
457 switch (dsb->device->pwfx->wBitsPerSample) {
458 case 8:
459 /* 8-bit WAV is unsigned, but we need to operate */
460 /* on signed data for this to work properly */
461 for (i = 0; i < len-1; i+=2) {
462 *(bpc++) = (((*(mem++) - 128) * vLeft) >> 16) + 128;
463 *(bpc++) = (((*(mem++) - 128) * vRight) >> 16) + 128;
465 if (len % 2 == 1 && nChannels == 1)
466 *(bpc++) = (((*(mem++) - 128) * vLeft) >> 16) + 128;
467 break;
468 case 16:
469 /* 16-bit WAV is signed -- much better */
470 for (i = 0; i < len-3; i += 4) {
471 *(bps++) = (*(mems++) * vLeft) >> 16;
472 *(bps++) = (*(mems++) * vRight) >> 16;
474 if (len % 4 == 2 && nChannels == 1)
475 *(bps++) = ((INT)*(mems++) * vLeft) >> 16;
476 break;
478 return dsb->device->tmp_buffer;
482 * Mix (at most) the given number of bytes into the given position of the
483 * device buffer, from the secondary buffer "dsb" (starting at the current
484 * mix position for that buffer).
486 * Returns the number of bytes actually mixed into the device buffer. This
487 * will match fraglen unless the end of the secondary buffer is reached
488 * (and it is not looping).
490 * dsb = the secondary buffer to mix from
491 * writepos = position (offset) in device buffer to write at
492 * fraglen = number of bytes to mix
494 static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen)
496 INT len = fraglen, ilen;
497 BYTE *ibuf = (dsb->tmp_buffer ? dsb->tmp_buffer : dsb->buffer->memory) + dsb->buf_mixpos, *volbuf;
498 DWORD oldpos, mixbufpos;
500 TRACE("buf_mixpos=%d/%d sec_mixpos=%d/%d\n", dsb->buf_mixpos, dsb->tmp_buffer_len, dsb->sec_mixpos, dsb->buflen);
501 TRACE("(%p,%d,%d)\n",dsb,writepos,fraglen);
503 assert(dsb->buf_mixpos + len <= dsb->tmp_buffer_len);
505 if (len % dsb->device->pwfx->nBlockAlign) {
506 INT nBlockAlign = dsb->device->pwfx->nBlockAlign;
507 ERR("length not a multiple of block size, len = %d, block size = %d\n", len, nBlockAlign);
508 len -= len % nBlockAlign; /* data alignment */
511 /* Resample buffer to temporary buffer specifically allocated for this purpose, if needed */
512 DSOUND_MixToTemporary(dsb, dsb->sec_mixpos, DSOUND_bufpos_to_secpos(dsb, dsb->buf_mixpos+len) - dsb->sec_mixpos, TRUE);
513 if (dsb->resampleinmixer)
514 ibuf = dsb->device->tmp_buffer;
516 /* Apply volume if needed */
517 volbuf = DSOUND_MixerVol(dsb, len);
518 if (volbuf)
519 ibuf = volbuf;
521 mixbufpos = DSOUND_bufpos_to_mixpos(dsb->device, writepos);
522 /* Now mix the temporary buffer into the devices main buffer */
523 if ((writepos + len) <= dsb->device->buflen)
524 dsb->device->mixfunction(ibuf, dsb->device->mix_buffer + mixbufpos, len);
525 else
527 DWORD todo = dsb->device->buflen - writepos;
528 dsb->device->mixfunction(ibuf, dsb->device->mix_buffer + mixbufpos, todo);
529 dsb->device->mixfunction(ibuf + todo, dsb->device->mix_buffer, len - todo);
532 oldpos = dsb->sec_mixpos;
533 dsb->buf_mixpos += len;
535 if (dsb->buf_mixpos >= dsb->tmp_buffer_len) {
536 if (dsb->buf_mixpos > dsb->tmp_buffer_len)
537 ERR("Mixpos (%u) past buflen (%u), capping...\n", dsb->buf_mixpos, dsb->tmp_buffer_len);
538 if (dsb->playflags & DSBPLAY_LOOPING) {
539 dsb->buf_mixpos -= dsb->tmp_buffer_len;
540 } else if (dsb->buf_mixpos >= dsb->tmp_buffer_len) {
541 dsb->buf_mixpos = dsb->sec_mixpos = 0;
542 dsb->state = STATE_STOPPED;
544 DSOUND_RecalcFreqAcc(dsb);
547 dsb->sec_mixpos = DSOUND_bufpos_to_secpos(dsb, dsb->buf_mixpos);
548 ilen = DSOUND_BufPtrDiff(dsb->buflen, dsb->sec_mixpos, oldpos);
549 /* check for notification positions */
550 if (dsb->dsbd.dwFlags & DSBCAPS_CTRLPOSITIONNOTIFY &&
551 dsb->state != STATE_STARTING) {
552 DSOUND_CheckEvent(dsb, oldpos, ilen);
555 /* increase mix position */
556 dsb->primary_mixpos += len;
557 if (dsb->primary_mixpos >= dsb->device->buflen)
558 dsb->primary_mixpos -= dsb->device->buflen;
559 return len;
563 * Mix some frames from the given secondary buffer "dsb" into the device
564 * primary buffer.
566 * dsb = the secondary buffer
567 * playpos = the current play position in the device buffer (primary buffer)
568 * writepos = the current safe-to-write position in the device buffer
569 * mixlen = the maximum number of bytes in the primary buffer to mix, from the
570 * current writepos.
572 * Returns: the number of bytes beyond the writepos that were mixed.
574 static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD mixlen)
576 /* The buffer's primary_mixpos may be before or after the device
577 * buffer's mixpos, but both must be ahead of writepos. */
578 DWORD primary_done;
580 TRACE("(%p,%d,%d)\n",dsb,writepos,mixlen);
581 TRACE("writepos=%d, buf_mixpos=%d, primary_mixpos=%d, mixlen=%d\n", writepos, dsb->buf_mixpos, dsb->primary_mixpos, mixlen);
582 TRACE("looping=%d, leadin=%d, buflen=%d\n", dsb->playflags, dsb->leadin, dsb->tmp_buffer_len);
584 /* If leading in, only mix about 20 ms, and 'skip' mixing the rest, for more fluid pointer advancement */
585 if (dsb->leadin && dsb->state == STATE_STARTING)
587 if (mixlen > 2 * dsb->device->fraglen)
589 dsb->primary_mixpos += mixlen - 2 * dsb->device->fraglen;
590 dsb->primary_mixpos %= dsb->device->buflen;
593 dsb->leadin = FALSE;
595 /* calculate how much pre-buffering has already been done for this buffer */
596 primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);
598 /* sanity */
599 if(mixlen < primary_done)
601 /* Should *NEVER* happen */
602 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);
603 dsb->primary_mixpos = writepos + mixlen;
604 dsb->primary_mixpos %= dsb->device->buflen;
605 return mixlen;
608 /* take into account already mixed data */
609 mixlen -= primary_done;
611 TRACE("primary_done=%d, mixlen (primary) = %i\n", primary_done, mixlen);
613 if (!mixlen)
614 return primary_done;
616 /* First try to mix to the end of the buffer if possible
617 * Theoretically it would allow for better optimization
619 if (mixlen + dsb->buf_mixpos >= dsb->tmp_buffer_len)
621 DWORD newmixed, mixfirst = dsb->tmp_buffer_len - dsb->buf_mixpos;
622 newmixed = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixfirst);
623 mixlen -= newmixed;
625 if (dsb->playflags & DSBPLAY_LOOPING)
626 while (newmixed && mixlen)
628 mixfirst = (dsb->tmp_buffer_len < mixlen ? dsb->tmp_buffer_len : mixlen);
629 newmixed = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixfirst);
630 mixlen -= newmixed;
633 else DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixlen);
635 /* re-calculate the primary done */
636 primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);
638 TRACE("new primary_mixpos=%d, total mixed data=%d\n", dsb->primary_mixpos, primary_done);
640 /* Report back the total prebuffered amount for this buffer */
641 return primary_done;
645 * For a DirectSoundDevice, go through all the currently playing buffers and
646 * mix them in to the device buffer.
648 * writepos = the current safe-to-write position in the primary buffer
649 * mixlen = the maximum amount to mix into the primary buffer
650 * (beyond the current writepos)
651 * mustlock = Do we have to fight for lock because we otherwise risk an underrun?
652 * recover = true if the sound device may have been reset and the write
653 * position in the device buffer changed
654 * all_stopped = reports back if all buffers have stopped
656 * Returns: the length beyond the writepos that was mixed to.
659 static DWORD DSOUND_MixToPrimary(const DirectSoundDevice *device, DWORD writepos, DWORD mixlen, BOOL mustlock, BOOL recover, BOOL *all_stopped)
661 INT i, len;
662 DWORD minlen = 0;
663 IDirectSoundBufferImpl *dsb;
664 BOOL gotall = TRUE;
666 /* unless we find a running buffer, all have stopped */
667 *all_stopped = TRUE;
669 TRACE("(%d,%d,%d)\n", writepos, mixlen, recover);
670 for (i = 0; i < device->nrofbuffers; i++) {
671 dsb = device->buffers[i];
673 TRACE("MixToPrimary for %p, state=%d\n", dsb, dsb->state);
675 if (dsb->buflen && dsb->state && !dsb->hwbuf) {
676 TRACE("Checking %p, mixlen=%d\n", dsb, mixlen);
677 if (!RtlAcquireResourceShared(&dsb->lock, mustlock))
679 gotall = FALSE;
680 continue;
682 /* if buffer is stopping it is stopped now */
683 if (dsb->state == STATE_STOPPING) {
684 dsb->state = STATE_STOPPED;
685 DSOUND_CheckEvent(dsb, 0, 0);
686 } else if (dsb->state != STATE_STOPPED) {
688 /* if recovering, reset the mix position */
689 if ((dsb->state == STATE_STARTING) || recover) {
690 dsb->primary_mixpos = writepos;
693 /* if the buffer was starting, it must be playing now */
694 if (dsb->state == STATE_STARTING)
695 dsb->state = STATE_PLAYING;
697 /* mix next buffer into the main buffer */
698 len = DSOUND_MixOne(dsb, writepos, mixlen);
700 if (!minlen) minlen = len;
702 /* record the minimum length mixed from all buffers */
703 /* we only want to return the length which *all* buffers have mixed */
704 else if (len) minlen = (len < minlen) ? len : minlen;
706 *all_stopped = FALSE;
708 RtlReleaseResource(&dsb->lock);
712 TRACE("Mixed at least %d from all buffers\n", minlen);
713 if (!gotall) return 0;
714 return minlen;
718 * Add buffers to the emulated wave device system.
720 * device = The current dsound playback device
721 * force = If TRUE, the function will buffer up as many frags as possible,
722 * even though and will ignore the actual state of the primary buffer.
724 * Returns: None
727 static void DSOUND_WaveQueue(DirectSoundDevice *device, BOOL force)
729 DWORD prebuf_frags, wave_writepos, wave_fragpos, i;
730 TRACE("(%p)\n", device);
732 /* calculate the current wave frag position */
733 wave_fragpos = (device->pwplay + device->pwqueue) % device->helfrags;
735 /* calculate the current wave write position */
736 wave_writepos = wave_fragpos * device->fraglen;
738 TRACE("wave_fragpos = %i, wave_writepos = %i, pwqueue = %i, prebuf = %i\n",
739 wave_fragpos, wave_writepos, device->pwqueue, device->prebuf);
741 if (!force)
743 /* check remaining prebuffered frags */
744 prebuf_frags = device->mixpos / device->fraglen;
745 if (prebuf_frags == device->helfrags)
746 --prebuf_frags;
747 TRACE("wave_fragpos = %d, mixpos_frags = %d\n", wave_fragpos, prebuf_frags);
748 if (prebuf_frags < wave_fragpos)
749 prebuf_frags += device->helfrags;
750 prebuf_frags -= wave_fragpos;
751 TRACE("wanted prebuf_frags = %d\n", prebuf_frags);
753 else
754 /* buffer the maximum amount of frags */
755 prebuf_frags = device->prebuf;
757 /* limit to the queue we have left */
758 if ((prebuf_frags + device->pwqueue) > device->prebuf)
759 prebuf_frags = device->prebuf - device->pwqueue;
761 TRACE("prebuf_frags = %i\n", prebuf_frags);
763 /* adjust queue */
764 device->pwqueue += prebuf_frags;
766 /* get out of CS when calling the wave system */
767 LeaveCriticalSection(&(device->mixlock));
768 /* **** */
770 /* queue up the new buffers */
771 for(i=0; i<prebuf_frags; i++){
772 TRACE("queueing wave buffer %i\n", wave_fragpos);
773 waveOutWrite(device->hwo, &device->pwave[wave_fragpos], sizeof(WAVEHDR));
774 wave_fragpos++;
775 wave_fragpos %= device->helfrags;
778 /* **** */
779 EnterCriticalSection(&(device->mixlock));
781 TRACE("queue now = %i\n", device->pwqueue);
785 * Perform mixing for a Direct Sound device. That is, go through all the
786 * secondary buffers (the sound bites currently playing) and mix them in
787 * to the primary buffer (the device buffer).
789 static void DSOUND_PerformMix(DirectSoundDevice *device)
791 TRACE("(%p)\n", device);
793 /* **** */
794 EnterCriticalSection(&(device->mixlock));
796 if (device->priolevel != DSSCL_WRITEPRIMARY) {
797 BOOL recover = FALSE, all_stopped = FALSE;
798 DWORD playpos, writepos, writelead, maxq, frag, prebuff_max, prebuff_left, size1, size2, mixplaypos, mixplaypos2;
799 LPVOID buf1, buf2;
800 BOOL lock = (device->hwbuf && !(device->drvdesc.dwFlags & DSDDESC_DONTNEEDPRIMARYLOCK));
801 BOOL mustlock = FALSE;
802 int nfiller;
804 /* the sound of silence */
805 nfiller = device->pwfx->wBitsPerSample == 8 ? 128 : 0;
807 /* get the position in the primary buffer */
808 if (DSOUND_PrimaryGetPosition(device, &playpos, &writepos) != 0){
809 LeaveCriticalSection(&(device->mixlock));
810 return;
813 TRACE("primary playpos=%d, writepos=%d, clrpos=%d, mixpos=%d, buflen=%d\n",
814 playpos,writepos,device->playpos,device->mixpos,device->buflen);
815 assert(device->playpos < device->buflen);
817 mixplaypos = DSOUND_bufpos_to_mixpos(device, device->playpos);
818 mixplaypos2 = DSOUND_bufpos_to_mixpos(device, playpos);
820 /* calc maximum prebuff */
821 prebuff_max = (device->prebuf * device->fraglen);
822 if (!device->hwbuf && playpos + prebuff_max >= device->helfrags * device->fraglen)
823 prebuff_max += device->buflen - device->helfrags * device->fraglen;
825 /* check how close we are to an underrun. It occurs when the writepos overtakes the mixpos */
826 prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);
827 writelead = DSOUND_BufPtrDiff(device->buflen, writepos, playpos);
829 /* check for underrun. underrun occurs when the write position passes the mix position
830 * also wipe out just-played sound data */
831 if((prebuff_left > prebuff_max) || (device->state == STATE_STOPPED) || (device->state == STATE_STARTING)){
832 if (device->state == STATE_STOPPING || device->state == STATE_PLAYING)
833 WARN("Probable buffer underrun\n");
834 else TRACE("Buffer starting or buffer underrun\n");
836 /* recover mixing for all buffers */
837 recover = TRUE;
839 /* reset mix position to write position */
840 device->mixpos = writepos;
842 ZeroMemory(device->mix_buffer, device->mix_buffer_len);
843 ZeroMemory(device->buffer, device->buflen);
844 } else if (playpos < device->playpos) {
845 buf1 = device->buffer + device->playpos;
846 buf2 = device->buffer;
847 size1 = device->buflen - device->playpos;
848 size2 = playpos;
849 FillMemory(device->mix_buffer + mixplaypos, device->mix_buffer_len - mixplaypos, 0);
850 FillMemory(device->mix_buffer, mixplaypos2, 0);
851 if (lock)
852 IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, device->playpos, size1+size2, 0);
853 FillMemory(buf1, size1, nfiller);
854 if (playpos && (!buf2 || !size2))
855 FIXME("%d: (%d, %d)=>(%d, %d) There should be an additional buffer here!!\n", __LINE__, device->playpos, device->mixpos, playpos, writepos);
856 FillMemory(buf2, size2, nfiller);
857 if (lock)
858 IDsDriverBuffer_Unlock(device->hwbuf, buf1, size1, buf2, size2);
859 } else {
860 buf1 = device->buffer + device->playpos;
861 buf2 = NULL;
862 size1 = playpos - device->playpos;
863 size2 = 0;
864 FillMemory(device->mix_buffer + mixplaypos, mixplaypos2 - mixplaypos, 0);
865 if (lock)
866 IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, device->playpos, size1+size2, 0);
867 FillMemory(buf1, size1, nfiller);
868 if (buf2 && size2)
870 FIXME("%d: There should be no additional buffer here!!\n", __LINE__);
871 FillMemory(buf2, size2, nfiller);
873 if (lock)
874 IDsDriverBuffer_Unlock(device->hwbuf, buf1, size1, buf2, size2);
876 device->playpos = playpos;
878 /* find the maximum we can prebuffer from current write position */
879 maxq = (writelead < prebuff_max) ? (prebuff_max - writelead) : 0;
881 TRACE("prebuff_left = %d, prebuff_max = %dx%d=%d, writelead=%d\n",
882 prebuff_left, device->prebuf, device->fraglen, prebuff_max, writelead);
884 /* Do we risk an 'underrun' if we don't advance pointer? */
885 if (writelead/device->fraglen <= ds_snd_queue_min || recover)
886 mustlock = TRUE;
888 if (lock)
889 IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, writepos, maxq, 0);
891 /* do the mixing */
892 frag = DSOUND_MixToPrimary(device, writepos, maxq, mustlock, recover, &all_stopped);
894 if (frag + writepos > device->buflen)
896 DWORD todo = device->buflen - writepos;
897 device->normfunction(device->mix_buffer + DSOUND_bufpos_to_mixpos(device, writepos), device->buffer + writepos, todo);
898 device->normfunction(device->mix_buffer, device->buffer, frag - todo);
900 else
901 device->normfunction(device->mix_buffer + DSOUND_bufpos_to_mixpos(device, writepos), device->buffer + writepos, frag);
903 /* update the mix position, taking wrap-around into account */
904 device->mixpos = writepos + frag;
905 device->mixpos %= device->buflen;
907 if (lock)
909 DWORD frag2 = (frag > size1 ? frag - size1 : 0);
910 frag -= frag2;
911 if (frag2 > size2)
913 FIXME("Buffering too much! (%d, %d, %d, %d)\n", maxq, frag, size2, frag2 - size2);
914 frag2 = size2;
916 IDsDriverBuffer_Unlock(device->hwbuf, buf1, frag, buf2, frag2);
919 /* update prebuff left */
920 prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);
922 /* check if have a whole fragment */
923 if (prebuff_left >= device->fraglen){
925 /* update the wave queue if using wave system */
926 if (!device->hwbuf)
927 DSOUND_WaveQueue(device, FALSE);
929 /* buffers are full. start playing if applicable */
930 if(device->state == STATE_STARTING){
931 TRACE("started primary buffer\n");
932 if(DSOUND_PrimaryPlay(device) != DS_OK){
933 WARN("DSOUND_PrimaryPlay failed\n");
935 else{
936 /* we are playing now */
937 device->state = STATE_PLAYING;
941 /* buffers are full. start stopping if applicable */
942 if(device->state == STATE_STOPPED){
943 TRACE("restarting primary buffer\n");
944 if(DSOUND_PrimaryPlay(device) != DS_OK){
945 WARN("DSOUND_PrimaryPlay failed\n");
947 else{
948 /* start stopping again. as soon as there is no more data, it will stop */
949 device->state = STATE_STOPPING;
954 /* if device was stopping, its for sure stopped when all buffers have stopped */
955 else if((all_stopped == TRUE) && (device->state == STATE_STOPPING)){
956 TRACE("All buffers have stopped. Stopping primary buffer\n");
957 device->state = STATE_STOPPED;
959 /* stop the primary buffer now */
960 DSOUND_PrimaryStop(device);
963 } else {
965 /* update the wave queue if using wave system */
966 if (!device->hwbuf)
967 DSOUND_WaveQueue(device, TRUE);
968 else
969 /* Keep alsa happy, which needs GetPosition called once every 10 ms */
970 IDsDriverBuffer_GetPosition(device->hwbuf, NULL, NULL);
972 /* in the DSSCL_WRITEPRIMARY mode, the app is totally in charge... */
973 if (device->state == STATE_STARTING) {
974 if (DSOUND_PrimaryPlay(device) != DS_OK)
975 WARN("DSOUND_PrimaryPlay failed\n");
976 else
977 device->state = STATE_PLAYING;
979 else if (device->state == STATE_STOPPING) {
980 if (DSOUND_PrimaryStop(device) != DS_OK)
981 WARN("DSOUND_PrimaryStop failed\n");
982 else
983 device->state = STATE_STOPPED;
987 LeaveCriticalSection(&(device->mixlock));
988 /* **** */
991 void CALLBACK DSOUND_timer(UINT timerID, UINT msg, DWORD_PTR dwUser,
992 DWORD_PTR dw1, DWORD_PTR dw2)
994 DirectSoundDevice * device = (DirectSoundDevice*)dwUser;
995 DWORD start_time = GetTickCount();
996 DWORD end_time;
997 TRACE("(%d,%d,0x%lx,0x%lx,0x%lx)\n",timerID,msg,dwUser,dw1,dw2);
998 TRACE("entering at %d\n", start_time);
1000 if (DSOUND_renderer[device->drvdesc.dnDevNode] != device) {
1001 ERR("dsound died without killing us?\n");
1002 timeKillEvent(timerID);
1003 timeEndPeriod(DS_TIME_RES);
1004 return;
1007 RtlAcquireResourceShared(&(device->buffer_list_lock), TRUE);
1009 if (device->ref)
1010 DSOUND_PerformMix(device);
1012 RtlReleaseResource(&(device->buffer_list_lock));
1014 end_time = GetTickCount();
1015 TRACE("completed processing at %d, duration = %d\n", end_time, end_time - start_time);
1018 void CALLBACK DSOUND_callback(HWAVEOUT hwo, UINT msg, DWORD_PTR dwUser, DWORD_PTR dw1, DWORD_PTR dw2)
1020 DirectSoundDevice * device = (DirectSoundDevice*)dwUser;
1021 TRACE("(%p,%x,%lx,%lx,%lx)\n",hwo,msg,dwUser,dw1,dw2);
1022 TRACE("entering at %d, msg=%08x(%s)\n", GetTickCount(), msg,
1023 msg==MM_WOM_DONE ? "MM_WOM_DONE" : msg==MM_WOM_CLOSE ? "MM_WOM_CLOSE" :
1024 msg==MM_WOM_OPEN ? "MM_WOM_OPEN" : "UNKNOWN");
1026 /* check if packet completed from wave driver */
1027 if (msg == MM_WOM_DONE) {
1029 /* **** */
1030 EnterCriticalSection(&(device->mixlock));
1032 TRACE("done playing primary pos=%d\n", device->pwplay * device->fraglen);
1034 /* update playpos */
1035 device->pwplay++;
1036 device->pwplay %= device->helfrags;
1038 /* sanity */
1039 if(device->pwqueue == 0){
1040 ERR("Wave queue corrupted!\n");
1043 /* update queue */
1044 device->pwqueue--;
1046 LeaveCriticalSection(&(device->mixlock));
1047 /* **** */
1049 TRACE("completed\n");