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];
196 dsb->resampleinmixer = FALSE;
201 DSOUND_RecalcFreqAcc(dsb);
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);
208 FillMemory(dsb->tmp_buffer, dsb->tmp_buffer_len, dsb->device->pwfx->wBitsPerSample == 8 ? 128 : 0);
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)
228 LPDSBPOSITIONNOTIFY event;
229 TRACE("(%p,%d)\n",dsb,len);
231 if (dsb->nrofnotifies == 0)
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. */
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);
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);
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 dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
282 if (device->pwfx->nChannels == 2)
283 dsb->convert(ibuf + istep, obuf + ostep, istride, ostride, count, freqAcc, adj);
286 if (device->pwfx->nChannels == 1 && dsb->pwfx->nChannels == 2)
288 dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
291 if (device->pwfx->nChannels == 2 && dsb->pwfx->nChannels == 1)
293 dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
294 dsb->convert(ibuf, obuf + ostep, istride, ostride, count, freqAcc, adj);
299 * Calculate the distance between two buffer offsets, taking wraparound
302 static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
304 /* If these asserts fail, the problem is not here, but in the underlying code */
305 assert(ptr1 < buflen);
306 assert(ptr2 < buflen);
310 return buflen + ptr1 - ptr2;
314 * Mix at most the given amount of data into the allocated temporary buffer
315 * of the given secondary buffer, starting from the dsb's first currently
316 * unsampled frame (writepos), translating frequency (pitch), stereo/mono
317 * and bits-per-sample so that it is ideal for the primary buffer.
318 * Doesn't perform any mixing - this is a straight copy/convert operation.
320 * dsb = the secondary buffer
321 * writepos = Starting position of changed buffer
322 * len = number of bytes to resample from writepos
324 * NOTE: writepos + len <= buflen. When called by mixer, MixOne makes sure of this.
326 void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD len, BOOL inmixer)
329 BYTE *ibp, *obp, *obp_begin;
330 INT iAdvance = dsb->pwfx->nBlockAlign;
331 INT oAdvance = dsb->device->pwfx->nBlockAlign;
332 DWORD freqAcc, target_writepos = 0, overshot, maxlen;
334 /* We resample only when needed */
335 if ((dsb->tmp_buffer && inmixer) || (!dsb->tmp_buffer && !inmixer) || dsb->resampleinmixer != inmixer)
338 assert(writepos + len <= dsb->buflen);
339 if (inmixer && writepos + len < dsb->buflen)
340 len += dsb->pwfx->nBlockAlign;
342 maxlen = DSOUND_secpos_to_bufpos(dsb, len, 0, NULL);
344 ibp = dsb->buffer->memory + writepos;
346 obp_begin = dsb->tmp_buffer;
347 else if (dsb->device->tmp_buffer_len < maxlen || !dsb->device->tmp_buffer)
349 dsb->device->tmp_buffer_len = maxlen;
350 if (dsb->device->tmp_buffer)
351 dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, maxlen);
353 dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, maxlen);
354 obp_begin = dsb->device->tmp_buffer;
357 obp_begin = dsb->device->tmp_buffer;
359 TRACE("(%p, %p)\n", dsb, ibp);
360 size = len / iAdvance;
362 /* Check for same sample rate */
363 if (dsb->freq == dsb->device->pwfx->nSamplesPerSec) {
364 TRACE("(%p) Same sample rate %d = primary %d\n", dsb,
365 dsb->freq, dsb->device->pwfx->nSamplesPerSec);
368 obp += writepos/iAdvance*oAdvance;
370 cp_fields(dsb, ibp, obp, iAdvance, oAdvance, size, 0, 1 << DSOUND_FREQSHIFT);
374 /* Mix in different sample rates */
375 TRACE("(%p) Adjusting frequency: %d -> %d\n", dsb, dsb->freq, dsb->device->pwfx->nSamplesPerSec);
377 target_writepos = DSOUND_secpos_to_bufpos(dsb, writepos, dsb->sec_mixpos, &freqAcc);
378 overshot = freqAcc >> DSOUND_FREQSHIFT;
381 if (overshot >= size)
384 writepos += overshot * iAdvance;
385 if (writepos >= dsb->buflen)
387 ibp = dsb->buffer->memory + writepos;
388 freqAcc &= (1 << DSOUND_FREQSHIFT) - 1;
389 TRACE("Overshot: %d, freqAcc: %04x\n", overshot, freqAcc);
393 obp = obp_begin + target_writepos;
394 else obp = obp_begin;
396 /* FIXME: Small problem here when we're overwriting buf_mixpos, it then STILL uses old freqAcc, not sure if it matters or not */
397 cp_fields(dsb, ibp, obp, iAdvance, oAdvance, size, freqAcc, dsb->freqAdjust);
400 /** Apply volume to the given soundbuffer from (primary) position writepos and length len
401 * Returns: NULL if no volume needs to be applied
402 * or else a memory handle that holds 'len' volume adjusted buffer */
403 static LPBYTE DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, INT len)
409 INT nChannels = dsb->device->pwfx->nChannels;
410 LPBYTE mem = (dsb->tmp_buffer ? dsb->tmp_buffer : dsb->buffer->memory) + dsb->buf_mixpos;
412 if (dsb->resampleinmixer)
413 mem = dsb->device->tmp_buffer;
415 TRACE("(%p,%d)\n",dsb,len);
416 TRACE("left = %x, right = %x\n", dsb->volpan.dwTotalLeftAmpFactor,
417 dsb->volpan.dwTotalRightAmpFactor);
419 if ((!(dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) || (dsb->volpan.lPan == 0)) &&
420 (!(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) || (dsb->volpan.lVolume == 0)) &&
421 !(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
422 return NULL; /* Nothing to do */
424 if (nChannels != 1 && nChannels != 2)
426 FIXME("There is no support for %d channels\n", nChannels);
430 if (dsb->device->pwfx->wBitsPerSample != 8 && dsb->device->pwfx->wBitsPerSample != 16)
432 FIXME("There is no support for %d bpp\n", dsb->device->pwfx->wBitsPerSample);
436 if (dsb->device->tmp_buffer_len < len || !dsb->device->tmp_buffer)
438 /* If we just resampled in DSOUND_MixToTemporary, we shouldn't need to resize here */
439 assert(!dsb->resampleinmixer);
440 dsb->device->tmp_buffer_len = len;
441 if (dsb->device->tmp_buffer)
442 dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, len);
444 dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, len);
447 bpc = dsb->device->tmp_buffer;
450 vLeft = dsb->volpan.dwTotalLeftAmpFactor;
452 vRight = dsb->volpan.dwTotalRightAmpFactor;
456 switch (dsb->device->pwfx->wBitsPerSample) {
458 /* 8-bit WAV is unsigned, but we need to operate */
459 /* on signed data for this to work properly */
460 for (i = 0; i < len-1; i+=2) {
461 *(bpc++) = (((*(mem++) - 128) * vLeft) >> 16) + 128;
462 *(bpc++) = (((*(mem++) - 128) * vRight) >> 16) + 128;
464 if (len % 2 == 1 && nChannels == 1)
465 *(bpc++) = (((*(mem++) - 128) * vLeft) >> 16) + 128;
468 /* 16-bit WAV is signed -- much better */
469 for (i = 0; i < len-3; i += 4) {
470 *(bps++) = (*(mems++) * vLeft) >> 16;
471 *(bps++) = (*(mems++) * vRight) >> 16;
473 if (len % 4 == 2 && nChannels == 1)
474 *(bps++) = ((INT)*(mems++) * vLeft) >> 16;
477 return dsb->device->tmp_buffer;
481 * Mix (at most) the given number of bytes into the given position of the
482 * device buffer, from the secondary buffer "dsb" (starting at the current
483 * mix position for that buffer).
485 * Returns the number of bytes actually mixed into the device buffer. This
486 * will match fraglen unless the end of the secondary buffer is reached
487 * (and it is not looping).
489 * dsb = the secondary buffer to mix from
490 * writepos = position (offset) in device buffer to write at
491 * fraglen = number of bytes to mix
493 static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen)
495 INT len = fraglen, ilen;
496 BYTE *ibuf = (dsb->tmp_buffer ? dsb->tmp_buffer : dsb->buffer->memory) + dsb->buf_mixpos, *volbuf;
497 DWORD oldpos, mixbufpos;
499 TRACE("buf_mixpos=%d/%d sec_mixpos=%d/%d\n", dsb->buf_mixpos, dsb->tmp_buffer_len, dsb->sec_mixpos, dsb->buflen);
500 TRACE("(%p,%d,%d)\n",dsb,writepos,fraglen);
502 assert(dsb->buf_mixpos + len <= dsb->tmp_buffer_len);
504 if (len % dsb->device->pwfx->nBlockAlign) {
505 INT nBlockAlign = dsb->device->pwfx->nBlockAlign;
506 ERR("length not a multiple of block size, len = %d, block size = %d\n", len, nBlockAlign);
507 len -= len % nBlockAlign; /* data alignment */
510 /* Resample buffer to temporary buffer specifically allocated for this purpose, if needed */
511 DSOUND_MixToTemporary(dsb, dsb->sec_mixpos, DSOUND_bufpos_to_secpos(dsb, dsb->buf_mixpos+len) - dsb->sec_mixpos, TRUE);
512 if (dsb->resampleinmixer)
513 ibuf = dsb->device->tmp_buffer;
515 /* Apply volume if needed */
516 volbuf = DSOUND_MixerVol(dsb, len);
520 mixbufpos = DSOUND_bufpos_to_mixpos(dsb->device, writepos);
521 /* Now mix the temporary buffer into the devices main buffer */
522 if ((writepos + len) <= dsb->device->buflen)
523 dsb->device->mixfunction(ibuf, dsb->device->mix_buffer + mixbufpos, len);
526 DWORD todo = dsb->device->buflen - writepos;
527 dsb->device->mixfunction(ibuf, dsb->device->mix_buffer + mixbufpos, todo);
528 dsb->device->mixfunction(ibuf + todo, dsb->device->mix_buffer, len - todo);
531 oldpos = dsb->sec_mixpos;
532 dsb->buf_mixpos += len;
534 if (dsb->buf_mixpos >= dsb->tmp_buffer_len) {
535 if (dsb->buf_mixpos > dsb->tmp_buffer_len)
536 ERR("Mixpos (%u) past buflen (%u), capping...\n", dsb->buf_mixpos, dsb->tmp_buffer_len);
537 if (dsb->playflags & DSBPLAY_LOOPING) {
538 dsb->buf_mixpos -= dsb->tmp_buffer_len;
539 } else if (dsb->buf_mixpos >= dsb->tmp_buffer_len) {
540 dsb->buf_mixpos = dsb->sec_mixpos = 0;
541 dsb->state = STATE_STOPPED;
543 DSOUND_RecalcFreqAcc(dsb);
546 dsb->sec_mixpos = DSOUND_bufpos_to_secpos(dsb, dsb->buf_mixpos);
547 ilen = DSOUND_BufPtrDiff(dsb->buflen, dsb->sec_mixpos, oldpos);
548 /* check for notification positions */
549 if (dsb->dsbd.dwFlags & DSBCAPS_CTRLPOSITIONNOTIFY &&
550 dsb->state != STATE_STARTING) {
551 DSOUND_CheckEvent(dsb, oldpos, ilen);
554 /* increase mix position */
555 dsb->primary_mixpos += len;
556 if (dsb->primary_mixpos >= dsb->device->buflen)
557 dsb->primary_mixpos -= dsb->device->buflen;
562 * Mix some frames from the given secondary buffer "dsb" into the device
565 * dsb = the secondary buffer
566 * playpos = the current play position in the device buffer (primary buffer)
567 * writepos = the current safe-to-write position in the device buffer
568 * mixlen = the maximum number of bytes in the primary buffer to mix, from the
571 * Returns: the number of bytes beyond the writepos that were mixed.
573 static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD mixlen)
575 /* The buffer's primary_mixpos may be before or after the device
576 * buffer's mixpos, but both must be ahead of writepos. */
579 TRACE("(%p,%d,%d)\n",dsb,writepos,mixlen);
580 TRACE("writepos=%d, buf_mixpos=%d, primary_mixpos=%d, mixlen=%d\n", writepos, dsb->buf_mixpos, dsb->primary_mixpos, mixlen);
581 TRACE("looping=%d, leadin=%d, buflen=%d\n", dsb->playflags, dsb->leadin, dsb->tmp_buffer_len);
583 /* If leading in, only mix about 20 ms, and 'skip' mixing the rest, for more fluid pointer advancement */
584 if (dsb->leadin && dsb->state == STATE_STARTING)
586 if (mixlen > 2 * dsb->device->fraglen)
588 dsb->primary_mixpos += mixlen - 2 * dsb->device->fraglen;
589 dsb->primary_mixpos %= dsb->device->buflen;
594 /* calculate how much pre-buffering has already been done for this buffer */
595 primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);
598 if(mixlen < primary_done)
600 /* Should *NEVER* happen */
601 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);
605 /* take into account already mixed data */
606 mixlen -= primary_done;
608 TRACE("primary_done=%d, mixlen (primary) = %i\n", primary_done, mixlen);
613 /* First try to mix to the end of the buffer if possible
614 * Theoretically it would allow for better optimization
616 if (mixlen + dsb->buf_mixpos >= dsb->tmp_buffer_len)
618 DWORD newmixed, mixfirst = dsb->tmp_buffer_len - dsb->buf_mixpos;
619 newmixed = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixfirst);
622 if (dsb->playflags & DSBPLAY_LOOPING)
623 while (newmixed && mixlen)
625 mixfirst = (dsb->tmp_buffer_len < mixlen ? dsb->tmp_buffer_len : mixlen);
626 newmixed = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixfirst);
630 else DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixlen);
632 /* re-calculate the primary done */
633 primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);
635 TRACE("new primary_mixpos=%d, total mixed data=%d\n", dsb->primary_mixpos, primary_done);
637 /* Report back the total prebuffered amount for this buffer */
642 * For a DirectSoundDevice, go through all the currently playing buffers and
643 * mix them in to the device buffer.
645 * writepos = the current safe-to-write position in the primary buffer
646 * mixlen = the maximum amount to mix into the primary buffer
647 * (beyond the current writepos)
648 * mustlock = Do we have to fight for lock because we otherwise risk an underrun?
649 * recover = true if the sound device may have been reset and the write
650 * position in the device buffer changed
651 * all_stopped = reports back if all buffers have stopped
653 * Returns: the length beyond the writepos that was mixed to.
656 static DWORD DSOUND_MixToPrimary(const DirectSoundDevice *device, DWORD writepos, DWORD mixlen, BOOL mustlock, BOOL recover, BOOL *all_stopped)
660 IDirectSoundBufferImpl *dsb;
663 /* unless we find a running buffer, all have stopped */
666 TRACE("(%d,%d,%d)\n", writepos, mixlen, recover);
667 for (i = 0; i < device->nrofbuffers; i++) {
668 dsb = device->buffers[i];
670 TRACE("MixToPrimary for %p, state=%d\n", dsb, dsb->state);
672 if (dsb->buflen && dsb->state && !dsb->hwbuf) {
673 TRACE("Checking %p, mixlen=%d\n", dsb, mixlen);
674 if (!RtlAcquireResourceShared(&dsb->lock, mustlock))
679 /* if buffer is stopping it is stopped now */
680 if (dsb->state == STATE_STOPPING) {
681 dsb->state = STATE_STOPPED;
682 DSOUND_CheckEvent(dsb, 0, 0);
683 } else if (dsb->state != STATE_STOPPED) {
685 /* if recovering, reset the mix position */
686 if ((dsb->state == STATE_STARTING) || recover) {
687 dsb->primary_mixpos = writepos;
690 /* if the buffer was starting, it must be playing now */
691 if (dsb->state == STATE_STARTING)
692 dsb->state = STATE_PLAYING;
694 /* mix next buffer into the main buffer */
695 len = DSOUND_MixOne(dsb, writepos, mixlen);
697 if (!minlen) minlen = len;
699 /* record the minimum length mixed from all buffers */
700 /* we only want to return the length which *all* buffers have mixed */
701 else if (len) minlen = (len < minlen) ? len : minlen;
703 *all_stopped = FALSE;
705 RtlReleaseResource(&dsb->lock);
709 TRACE("Mixed at least %d from all buffers\n", minlen);
710 if (!gotall) return 0;
715 * Add buffers to the emulated wave device system.
717 * device = The current dsound playback device
718 * force = If TRUE, the function will buffer up as many frags as possible,
719 * even though and will ignore the actual state of the primary buffer.
724 static void DSOUND_WaveQueue(DirectSoundDevice *device, BOOL force)
726 DWORD prebuf_frags, wave_writepos, wave_fragpos, i;
727 TRACE("(%p)\n", device);
729 /* calculate the current wave frag position */
730 wave_fragpos = (device->pwplay + device->pwqueue) % device->helfrags;
732 /* calculate the current wave write position */
733 wave_writepos = wave_fragpos * device->fraglen;
735 TRACE("wave_fragpos = %i, wave_writepos = %i, pwqueue = %i, prebuf = %i\n",
736 wave_fragpos, wave_writepos, device->pwqueue, device->prebuf);
740 /* check remaining prebuffered frags */
741 prebuf_frags = device->mixpos / device->fraglen;
742 if (prebuf_frags == device->helfrags)
744 TRACE("wave_fragpos = %d, mixpos_frags = %d\n", wave_fragpos, prebuf_frags);
745 if (prebuf_frags < wave_fragpos)
746 prebuf_frags += device->helfrags;
747 prebuf_frags -= wave_fragpos;
748 TRACE("wanted prebuf_frags = %d\n", prebuf_frags);
751 /* buffer the maximum amount of frags */
752 prebuf_frags = device->prebuf;
754 /* limit to the queue we have left */
755 if ((prebuf_frags + device->pwqueue) > device->prebuf)
756 prebuf_frags = device->prebuf - device->pwqueue;
758 TRACE("prebuf_frags = %i\n", prebuf_frags);
761 device->pwqueue += prebuf_frags;
763 /* get out of CS when calling the wave system */
764 LeaveCriticalSection(&(device->mixlock));
767 /* queue up the new buffers */
768 for(i=0; i<prebuf_frags; i++){
769 TRACE("queueing wave buffer %i\n", wave_fragpos);
770 waveOutWrite(device->hwo, &device->pwave[wave_fragpos], sizeof(WAVEHDR));
772 wave_fragpos %= device->helfrags;
776 EnterCriticalSection(&(device->mixlock));
778 TRACE("queue now = %i\n", device->pwqueue);
782 * Perform mixing for a Direct Sound device. That is, go through all the
783 * secondary buffers (the sound bites currently playing) and mix them in
784 * to the primary buffer (the device buffer).
786 static void DSOUND_PerformMix(DirectSoundDevice *device)
788 TRACE("(%p)\n", device);
791 EnterCriticalSection(&(device->mixlock));
793 if (device->priolevel != DSSCL_WRITEPRIMARY) {
794 BOOL recover = FALSE, all_stopped = FALSE;
795 DWORD playpos, writepos, writelead, maxq, frag, prebuff_max, prebuff_left, size1, size2, mixplaypos, mixplaypos2;
797 BOOL lock = (device->hwbuf && !(device->drvdesc.dwFlags & DSDDESC_DONTNEEDPRIMARYLOCK));
798 BOOL mustlock = FALSE;
801 /* the sound of silence */
802 nfiller = device->pwfx->wBitsPerSample == 8 ? 128 : 0;
804 /* get the position in the primary buffer */
805 if (DSOUND_PrimaryGetPosition(device, &playpos, &writepos) != 0){
806 LeaveCriticalSection(&(device->mixlock));
810 TRACE("primary playpos=%d, writepos=%d, clrpos=%d, mixpos=%d, buflen=%d\n",
811 playpos,writepos,device->playpos,device->mixpos,device->buflen);
812 assert(device->playpos < device->buflen);
814 mixplaypos = DSOUND_bufpos_to_mixpos(device, device->playpos);
815 mixplaypos2 = DSOUND_bufpos_to_mixpos(device, playpos);
817 /* calc maximum prebuff */
818 prebuff_max = (device->prebuf * device->fraglen);
819 if (!device->hwbuf && playpos + prebuff_max >= device->helfrags * device->fraglen)
820 prebuff_max += device->buflen - device->helfrags * device->fraglen;
822 /* check how close we are to an underrun. It occurs when the writepos overtakes the mixpos */
823 prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);
824 writelead = DSOUND_BufPtrDiff(device->buflen, writepos, playpos);
826 /* check for underrun. underrun occurs when the write position passes the mix position
827 * also wipe out just-played sound data */
828 if((prebuff_left > prebuff_max) || (device->state == STATE_STOPPED) || (device->state == STATE_STARTING)){
829 if (device->state == STATE_STOPPING || device->state == STATE_PLAYING)
830 WARN("Probable buffer underrun\n");
831 else TRACE("Buffer starting or buffer underrun\n");
833 /* recover mixing for all buffers */
836 /* reset mix position to write position */
837 device->mixpos = writepos;
839 ZeroMemory(device->mix_buffer, device->mix_buffer_len);
840 ZeroMemory(device->buffer, device->buflen);
841 } else if (playpos < device->playpos) {
842 buf1 = device->buffer + device->playpos;
843 buf2 = device->buffer;
844 size1 = device->buflen - device->playpos;
846 FillMemory(device->mix_buffer + mixplaypos, device->mix_buffer_len - mixplaypos, 0);
847 FillMemory(device->mix_buffer, mixplaypos2, 0);
849 IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, device->playpos, size1+size2, 0);
850 FillMemory(buf1, size1, nfiller);
851 if (playpos && (!buf2 || !size2))
852 FIXME("%d: (%d, %d)=>(%d, %d) There should be an additional buffer here!!\n", __LINE__, device->playpos, device->mixpos, playpos, writepos);
853 FillMemory(buf2, size2, nfiller);
855 IDsDriverBuffer_Unlock(device->hwbuf, buf1, size1, buf2, size2);
857 buf1 = device->buffer + device->playpos;
859 size1 = playpos - device->playpos;
861 FillMemory(device->mix_buffer + mixplaypos, mixplaypos2 - mixplaypos, 0);
863 IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, device->playpos, size1+size2, 0);
864 FillMemory(buf1, size1, nfiller);
867 FIXME("%d: There should be no additional buffer here!!\n", __LINE__);
868 FillMemory(buf2, size2, nfiller);
871 IDsDriverBuffer_Unlock(device->hwbuf, buf1, size1, buf2, size2);
873 device->playpos = playpos;
875 /* find the maximum we can prebuffer from current write position */
876 maxq = (writelead < prebuff_max) ? (prebuff_max - writelead) : 0;
878 TRACE("prebuff_left = %d, prebuff_max = %dx%d=%d, writelead=%d\n",
879 prebuff_left, device->prebuf, device->fraglen, prebuff_max, writelead);
881 /* Do we risk an 'underrun' if we don't advance pointer? */
882 if (writelead/device->fraglen <= ds_snd_queue_min || recover)
886 IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, writepos, maxq, 0);
889 frag = DSOUND_MixToPrimary(device, writepos, maxq, mustlock, recover, &all_stopped);
891 if (frag + writepos > device->buflen)
893 DWORD todo = device->buflen - writepos;
894 device->normfunction(device->mix_buffer + DSOUND_bufpos_to_mixpos(device, writepos), device->buffer + writepos, todo);
895 device->normfunction(device->mix_buffer, device->buffer, frag - todo);
898 device->normfunction(device->mix_buffer + DSOUND_bufpos_to_mixpos(device, writepos), device->buffer + writepos, frag);
900 /* update the mix position, taking wrap-around into account */
901 device->mixpos = writepos + frag;
902 device->mixpos %= device->buflen;
906 DWORD frag2 = (frag > size1 ? frag - size1 : 0);
910 FIXME("Buffering too much! (%d, %d, %d, %d)\n", maxq, frag, size2, frag2 - size2);
913 IDsDriverBuffer_Unlock(device->hwbuf, buf1, frag, buf2, frag2);
916 /* update prebuff left */
917 prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);
919 /* check if have a whole fragment */
920 if (prebuff_left >= device->fraglen){
922 /* update the wave queue if using wave system */
924 DSOUND_WaveQueue(device, FALSE);
926 /* buffers are full. start playing if applicable */
927 if(device->state == STATE_STARTING){
928 TRACE("started primary buffer\n");
929 if(DSOUND_PrimaryPlay(device) != DS_OK){
930 WARN("DSOUND_PrimaryPlay failed\n");
933 /* we are playing now */
934 device->state = STATE_PLAYING;
938 /* buffers are full. start stopping if applicable */
939 if(device->state == STATE_STOPPED){
940 TRACE("restarting primary buffer\n");
941 if(DSOUND_PrimaryPlay(device) != DS_OK){
942 WARN("DSOUND_PrimaryPlay failed\n");
945 /* start stopping again. as soon as there is no more data, it will stop */
946 device->state = STATE_STOPPING;
951 /* if device was stopping, its for sure stopped when all buffers have stopped */
952 else if((all_stopped == TRUE) && (device->state == STATE_STOPPING)){
953 TRACE("All buffers have stopped. Stopping primary buffer\n");
954 device->state = STATE_STOPPED;
956 /* stop the primary buffer now */
957 DSOUND_PrimaryStop(device);
962 /* update the wave queue if using wave system */
964 DSOUND_WaveQueue(device, TRUE);
966 /* Keep alsa happy, which needs GetPosition called once every 10 ms */
967 IDsDriverBuffer_GetPosition(device->hwbuf, NULL, NULL);
969 /* in the DSSCL_WRITEPRIMARY mode, the app is totally in charge... */
970 if (device->state == STATE_STARTING) {
971 if (DSOUND_PrimaryPlay(device) != DS_OK)
972 WARN("DSOUND_PrimaryPlay failed\n");
974 device->state = STATE_PLAYING;
976 else if (device->state == STATE_STOPPING) {
977 if (DSOUND_PrimaryStop(device) != DS_OK)
978 WARN("DSOUND_PrimaryStop failed\n");
980 device->state = STATE_STOPPED;
984 LeaveCriticalSection(&(device->mixlock));
988 void CALLBACK DSOUND_timer(UINT timerID, UINT msg, DWORD_PTR dwUser,
989 DWORD_PTR dw1, DWORD_PTR dw2)
991 DirectSoundDevice * device = (DirectSoundDevice*)dwUser;
992 DWORD start_time = GetTickCount();
994 TRACE("(%d,%d,0x%lx,0x%lx,0x%lx)\n",timerID,msg,dwUser,dw1,dw2);
995 TRACE("entering at %d\n", start_time);
997 if (DSOUND_renderer[device->drvdesc.dnDevNode] != device) {
998 ERR("dsound died without killing us?\n");
999 timeKillEvent(timerID);
1000 timeEndPeriod(DS_TIME_RES);
1004 RtlAcquireResourceShared(&(device->buffer_list_lock), TRUE);
1007 DSOUND_PerformMix(device);
1009 RtlReleaseResource(&(device->buffer_list_lock));
1011 end_time = GetTickCount();
1012 TRACE("completed processing at %d, duration = %d\n", end_time, end_time - start_time);
1015 void CALLBACK DSOUND_callback(HWAVEOUT hwo, UINT msg, DWORD_PTR dwUser, DWORD_PTR dw1, DWORD_PTR dw2)
1017 DirectSoundDevice * device = (DirectSoundDevice*)dwUser;
1018 TRACE("(%p,%x,%lx,%lx,%lx)\n",hwo,msg,dwUser,dw1,dw2);
1019 TRACE("entering at %d, msg=%08x(%s)\n", GetTickCount(), msg,
1020 msg==MM_WOM_DONE ? "MM_WOM_DONE" : msg==MM_WOM_CLOSE ? "MM_WOM_CLOSE" :
1021 msg==MM_WOM_OPEN ? "MM_WOM_OPEN" : "UNKNOWN");
1023 /* check if packet completed from wave driver */
1024 if (msg == MM_WOM_DONE) {
1027 EnterCriticalSection(&(device->mixlock));
1029 TRACE("done playing primary pos=%d\n", device->pwplay * device->fraglen);
1031 /* update playpos */
1033 device->pwplay %= device->helfrags;
1036 if(device->pwqueue == 0){
1037 ERR("Wave queue corrupted!\n");
1043 LeaveCriticalSection(&(device->mixlock));
1046 TRACE("completed\n");