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
8 * Copyright 2011 Owen Rudge for CodeWeavers
10 * This library is free software; you can redistribute it and/or
11 * modify it under the terms of the GNU Lesser General Public
12 * License as published by the Free Software Foundation; either
13 * version 2.1 of the License, or (at your option) any later version.
15 * This library is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18 * Lesser General Public License for more details.
20 * You should have received a copy of the GNU Lesser General Public
21 * License along with this library; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
27 #include <math.h> /* Insomnia - pow() function */
30 #define NONAMELESSSTRUCT
31 #define NONAMELESSUNION
38 #include "wine/debug.h"
42 #include "dsound_private.h"
45 WINE_DEFAULT_DEBUG_CHANNEL(dsound);
47 void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan)
50 TRACE("(%p)\n",volpan);
52 TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
53 /* the AmpFactors are expressed in 16.16 fixed point */
54 volpan->dwVolAmpFactor = (ULONG) (pow(2.0, volpan->lVolume / 600.0) * 0xffff);
55 /* FIXME: dwPan{Left|Right}AmpFactor */
57 /* FIXME: use calculated vol and pan ampfactors */
58 temp = (double) (volpan->lVolume - (volpan->lPan > 0 ? volpan->lPan : 0));
59 volpan->dwTotalLeftAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
60 temp = (double) (volpan->lVolume + (volpan->lPan < 0 ? volpan->lPan : 0));
61 volpan->dwTotalRightAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
63 TRACE("left = %x, right = %x\n", volpan->dwTotalLeftAmpFactor, volpan->dwTotalRightAmpFactor);
66 void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan)
69 TRACE("(%p)\n",volpan);
71 TRACE("left=%x, right=%x\n",volpan->dwTotalLeftAmpFactor,volpan->dwTotalRightAmpFactor);
72 if (volpan->dwTotalLeftAmpFactor==0)
75 left=600 * log(((double)volpan->dwTotalLeftAmpFactor) / 0xffff) / log(2);
76 if (volpan->dwTotalRightAmpFactor==0)
79 right=600 * log(((double)volpan->dwTotalRightAmpFactor) / 0xffff) / log(2);
82 volpan->lVolume=right;
83 volpan->dwVolAmpFactor=volpan->dwTotalRightAmpFactor;
88 volpan->dwVolAmpFactor=volpan->dwTotalLeftAmpFactor;
90 if (volpan->lVolume < -10000)
91 volpan->lVolume=-10000;
92 volpan->lPan=right-left;
93 if (volpan->lPan < -10000)
96 TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
100 * Recalculate the size for temporary buffer, and new writelead
101 * Should be called when one of the following things occur:
102 * - Primary buffer format is changed
103 * - This buffer format (frequency) is changed
105 void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
107 DWORD ichannels = dsb->pwfx->nChannels;
108 DWORD ochannels = dsb->device->pwfx->nChannels;
109 WAVEFORMATEXTENSIBLE *pwfxe;
114 pwfxe = (WAVEFORMATEXTENSIBLE *) dsb->pwfx;
115 dsb->freqAdjust = (float)dsb->freq / dsb->device->pwfx->nSamplesPerSec;
117 if ((pwfxe->Format.wFormatTag == WAVE_FORMAT_IEEE_FLOAT) || ((pwfxe->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE)
118 && (IsEqualGUID(&pwfxe->SubFormat, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT))))
122 * Recalculate FIR step and gain.
124 * firstep says how many points of the FIR exist per one
125 * sample in the secondary buffer. firgain specifies what
126 * to multiply the FIR output by in order to attenuate it correctly.
128 if (dsb->freqAdjust > 1.0f) {
130 * Yes, round it a bit to make sure that the
131 * linear interpolation factor never changes.
133 dsb->firstep = ceil(fir_step / dsb->freqAdjust);
135 dsb->firstep = fir_step;
137 dsb->firgain = (float)dsb->firstep / fir_step;
139 /* calculate the 10ms write lead */
140 dsb->writelead = (dsb->freq / 100) * dsb->pwfx->nBlockAlign;
144 dsb->get_aux = ieee ? getbpp[4] : getbpp[dsb->pwfx->wBitsPerSample/8 - 1];
145 dsb->put_aux = putieee32;
147 dsb->get = dsb->get_aux;
148 dsb->put = dsb->put_aux;
150 if (ichannels == ochannels)
152 dsb->mix_channels = ichannels;
153 if (ichannels > 32) {
154 FIXME("Copying %u channels is unsupported, limiting to first 32\n", ichannels);
155 dsb->mix_channels = 32;
158 else if (ichannels == 1)
160 dsb->mix_channels = 1;
161 dsb->put = put_mono2stereo;
163 else if (ochannels == 1)
165 dsb->mix_channels = 1;
171 FIXME("Conversion from %u to %u channels is not implemented, falling back to stereo\n", ichannels, ochannels);
172 dsb->mix_channels = 2;
177 * Check for application callback requests for when the play position
178 * reaches certain points.
180 * The offsets that will be triggered will be those between the recorded
181 * "last played" position for the buffer (i.e. dsb->playpos) and "len" bytes
182 * beyond that position.
184 void DSOUND_CheckEvent(const IDirectSoundBufferImpl *dsb, DWORD playpos, int len)
188 LPDSBPOSITIONNOTIFY event;
189 TRACE("(%p,%d)\n",dsb,len);
191 if (dsb->nrofnotifies == 0)
194 TRACE("(%p) buflen = %d, playpos = %d, len = %d\n",
195 dsb, dsb->buflen, playpos, len);
196 for (i = 0; i < dsb->nrofnotifies ; i++) {
197 event = dsb->notifies + i;
198 offset = event->dwOffset;
199 TRACE("checking %d, position %d, event = %p\n",
200 i, offset, event->hEventNotify);
201 /* DSBPN_OFFSETSTOP has to be the last element. So this is */
202 /* OK. [Inside DirectX, p274] */
203 /* Windows does not seem to enforce this, and some apps rely */
204 /* on that, so we can't stop there. */
206 /* This also means we can't sort the entries by offset, */
207 /* because DSBPN_OFFSETSTOP == -1 */
208 if (offset == DSBPN_OFFSETSTOP) {
209 if (dsb->state == STATE_STOPPED) {
210 SetEvent(event->hEventNotify);
211 TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
215 if ((playpos + len) >= dsb->buflen) {
216 if ((offset < ((playpos + len) % dsb->buflen)) ||
217 (offset >= playpos)) {
218 TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
219 SetEvent(event->hEventNotify);
222 if ((offset >= playpos) && (offset < (playpos + len))) {
223 TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
224 SetEvent(event->hEventNotify);
230 static inline float get_current_sample(const IDirectSoundBufferImpl *dsb,
231 DWORD mixpos, DWORD channel)
233 if (mixpos >= dsb->buflen && !(dsb->playflags & DSBPLAY_LOOPING))
235 return dsb->get(dsb, mixpos % dsb->buflen, channel);
238 static UINT cp_fields_noresample(IDirectSoundBufferImpl *dsb, UINT count)
240 UINT istride = dsb->pwfx->nBlockAlign;
241 UINT ostride = dsb->device->pwfx->nChannels * sizeof(float);
243 for (i = 0; i < count; i++)
244 for (channel = 0; channel < dsb->mix_channels; channel++)
245 dsb->put(dsb, i * ostride, channel, get_current_sample(dsb,
246 dsb->sec_mixpos + i * istride, channel));
250 static UINT cp_fields_resample(IDirectSoundBufferImpl *dsb, UINT count, float *freqAcc)
253 UINT istride = dsb->pwfx->nBlockAlign;
254 UINT ostride = dsb->device->pwfx->nChannels * sizeof(float);
256 float freqAdjust = dsb->freqAdjust;
257 float freqAcc_start = *freqAcc;
258 float freqAcc_end = freqAcc_start + count * freqAdjust;
259 UINT dsbfirstep = dsb->firstep;
260 UINT channels = dsb->mix_channels;
261 UINT max_ipos = freqAcc_start + count * freqAdjust;
263 UINT fir_cachesize = (fir_len + dsbfirstep - 2) / dsbfirstep;
264 UINT required_input = max_ipos + fir_cachesize;
266 float* intermediate = HeapAlloc(GetProcessHeap(), 0,
267 sizeof(float) * required_input * channels);
269 float* fir_copy = HeapAlloc(GetProcessHeap(), 0,
270 sizeof(float) * fir_cachesize);
272 /* Important: this buffer MUST be non-interleaved
273 * if you want -msse3 to have any effect.
274 * This is good for CPU cache effects, too.
276 float* itmp = intermediate;
277 for (channel = 0; channel < channels; channel++)
278 for (i = 0; i < required_input; i++)
279 *(itmp++) = get_current_sample(dsb,
280 dsb->sec_mixpos + i * istride, channel);
282 for(i = 0; i < count; ++i) {
283 float total_fir_steps = (freqAcc_start + i * freqAdjust) * dsbfirstep;
284 UINT int_fir_steps = total_fir_steps;
285 UINT ipos = int_fir_steps / dsbfirstep;
287 UINT idx = (ipos + 1) * dsbfirstep - int_fir_steps - 1;
288 float rem = int_fir_steps + 1.0 - total_fir_steps;
291 while (idx < fir_len - 1) {
292 fir_copy[fir_used++] = fir[idx] * (1.0 - rem) + fir[idx + 1] * rem;
296 assert(fir_used <= fir_cachesize);
297 assert(ipos + fir_used <= required_input);
299 for (channel = 0; channel < dsb->mix_channels; channel++) {
302 float* cache = &intermediate[channel * required_input + ipos];
303 for (j = 0; j < fir_used; j++)
304 sum += fir_copy[j] * cache[j];
305 dsb->put(dsb, i * ostride, channel, sum * dsb->firgain);
309 freqAcc_end -= (int)freqAcc_end;
310 *freqAcc = freqAcc_end;
312 HeapFree(GetProcessHeap(), 0, fir_copy);
313 HeapFree(GetProcessHeap(), 0, intermediate);
318 static void cp_fields(IDirectSoundBufferImpl *dsb, UINT count, float *freqAcc)
322 if (dsb->freqAdjust == 1.0)
323 adv = cp_fields_noresample(dsb, count); /* *freqAcc is unmodified */
325 adv = cp_fields_resample(dsb, count, freqAcc);
327 ipos = dsb->sec_mixpos + adv * dsb->pwfx->nBlockAlign;
328 if (ipos >= dsb->buflen) {
329 if (dsb->playflags & DSBPLAY_LOOPING)
333 dsb->state = STATE_STOPPED;
337 dsb->sec_mixpos = ipos;
341 * Calculate the distance between two buffer offsets, taking wraparound
344 static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
346 /* If these asserts fail, the problem is not here, but in the underlying code */
347 assert(ptr1 < buflen);
348 assert(ptr2 < buflen);
352 return buflen + ptr1 - ptr2;
356 * Mix at most the given amount of data into the allocated temporary buffer
357 * of the given secondary buffer, starting from the dsb's first currently
358 * unsampled frame (writepos), translating frequency (pitch), stereo/mono
359 * and bits-per-sample so that it is ideal for the primary buffer.
360 * Doesn't perform any mixing - this is a straight copy/convert operation.
362 * dsb = the secondary buffer
363 * writepos = Starting position of changed buffer
364 * len = number of bytes to resample from writepos
366 * NOTE: writepos + len <= buflen. When called by mixer, MixOne makes sure of this.
368 static void DSOUND_MixToTemporary(IDirectSoundBufferImpl *dsb, DWORD frames)
370 UINT size_bytes = frames * sizeof(float) * dsb->device->pwfx->nChannels;
372 if (dsb->device->tmp_buffer_len < size_bytes || !dsb->device->tmp_buffer)
374 dsb->device->tmp_buffer_len = size_bytes;
375 if (dsb->device->tmp_buffer)
376 dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, size_bytes);
378 dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, size_bytes);
381 cp_fields(dsb, frames, &dsb->freqAcc);
384 static void DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, INT frames)
388 UINT channels = dsb->device->pwfx->nChannels, chan;
390 TRACE("(%p,%d)\n",dsb,frames);
391 TRACE("left = %x, right = %x\n", dsb->volpan.dwTotalLeftAmpFactor,
392 dsb->volpan.dwTotalRightAmpFactor);
394 if ((!(dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) || (dsb->volpan.lPan == 0)) &&
395 (!(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) || (dsb->volpan.lVolume == 0)) &&
396 !(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
397 return; /* Nothing to do */
399 if (channels != 1 && channels != 2)
401 FIXME("There is no support for %u channels\n", channels);
405 vLeft = dsb->volpan.dwTotalLeftAmpFactor / ((float)0xFFFF);
406 vRight = dsb->volpan.dwTotalRightAmpFactor / ((float)0xFFFF);
407 for(i = 0; i < frames; ++i){
408 for(chan = 0; chan < channels; ++chan){
410 dsb->device->tmp_buffer[i * channels + chan] *= vLeft;
412 dsb->device->tmp_buffer[i * channels + chan] *= vRight;
418 * Mix (at most) the given number of bytes into the given position of the
419 * device buffer, from the secondary buffer "dsb" (starting at the current
420 * mix position for that buffer).
422 * Returns the number of bytes actually mixed into the device buffer. This
423 * will match fraglen unless the end of the secondary buffer is reached
424 * (and it is not looping).
426 * dsb = the secondary buffer to mix from
427 * writepos = position (offset) in device buffer to write at
428 * fraglen = number of bytes to mix
430 static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen)
435 UINT frames = fraglen / dsb->device->pwfx->nBlockAlign;
437 TRACE("sec_mixpos=%d/%d\n", dsb->sec_mixpos, dsb->buflen);
438 TRACE("(%p,%d,%d)\n",dsb,writepos,fraglen);
440 if (len % dsb->device->pwfx->nBlockAlign) {
441 INT nBlockAlign = dsb->device->pwfx->nBlockAlign;
442 ERR("length not a multiple of block size, len = %d, block size = %d\n", len, nBlockAlign);
443 len -= len % nBlockAlign; /* data alignment */
446 /* Resample buffer to temporary buffer specifically allocated for this purpose, if needed */
447 oldpos = dsb->sec_mixpos;
449 DSOUND_MixToTemporary(dsb, frames);
450 ibuf = dsb->device->tmp_buffer;
452 /* Apply volume if needed */
453 DSOUND_MixerVol(dsb, frames);
455 mixieee32(ibuf, dsb->device->mix_buffer, frames * dsb->device->pwfx->nChannels);
457 /* check for notification positions */
458 if (dsb->dsbd.dwFlags & DSBCAPS_CTRLPOSITIONNOTIFY &&
459 dsb->state != STATE_STARTING) {
460 INT ilen = DSOUND_BufPtrDiff(dsb->buflen, dsb->sec_mixpos, oldpos);
461 DSOUND_CheckEvent(dsb, oldpos, ilen);
468 * Mix some frames from the given secondary buffer "dsb" into the device
471 * dsb = the secondary buffer
472 * playpos = the current play position in the device buffer (primary buffer)
473 * writepos = the current safe-to-write position in the device buffer
474 * mixlen = the maximum number of bytes in the primary buffer to mix, from the
477 * Returns: the number of bytes beyond the writepos that were mixed.
479 static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD mixlen)
481 DWORD primary_done = 0;
483 TRACE("(%p,%d,%d)\n",dsb,writepos,mixlen);
484 TRACE("writepos=%d, mixlen=%d\n", writepos, mixlen);
485 TRACE("looping=%d, leadin=%d\n", dsb->playflags, dsb->leadin);
487 /* If leading in, only mix about 20 ms, and 'skip' mixing the rest, for more fluid pointer advancement */
488 /* FIXME: Is this needed? */
489 if (dsb->leadin && dsb->state == STATE_STARTING) {
490 if (mixlen > 2 * dsb->device->fraglen) {
491 primary_done = mixlen - 2 * dsb->device->fraglen;
492 mixlen = 2 * dsb->device->fraglen;
493 writepos += primary_done;
494 dsb->sec_mixpos += (primary_done / dsb->device->pwfx->nBlockAlign) *
495 dsb->pwfx->nBlockAlign * dsb->freqAdjust;
501 TRACE("mixlen (primary) = %i\n", mixlen);
503 /* First try to mix to the end of the buffer if possible
504 * Theoretically it would allow for better optimization
506 primary_done += DSOUND_MixInBuffer(dsb, writepos, mixlen);
508 TRACE("total mixed data=%d\n", primary_done);
510 /* Report back the total prebuffered amount for this buffer */
515 * For a DirectSoundDevice, go through all the currently playing buffers and
516 * mix them in to the device buffer.
518 * writepos = the current safe-to-write position in the primary buffer
519 * mixlen = the maximum amount to mix into the primary buffer
520 * (beyond the current writepos)
521 * recover = true if the sound device may have been reset and the write
522 * position in the device buffer changed
523 * all_stopped = reports back if all buffers have stopped
525 * Returns: the length beyond the writepos that was mixed to.
528 static void DSOUND_MixToPrimary(const DirectSoundDevice *device, DWORD writepos, DWORD mixlen, BOOL recover, BOOL *all_stopped)
531 IDirectSoundBufferImpl *dsb;
533 /* unless we find a running buffer, all have stopped */
536 TRACE("(%d,%d,%d)\n", writepos, mixlen, recover);
537 for (i = 0; i < device->nrofbuffers; i++) {
538 dsb = device->buffers[i];
540 TRACE("MixToPrimary for %p, state=%d\n", dsb, dsb->state);
542 if (dsb->buflen && dsb->state) {
543 TRACE("Checking %p, mixlen=%d\n", dsb, mixlen);
544 RtlAcquireResourceShared(&dsb->lock, TRUE);
545 /* if buffer is stopping it is stopped now */
546 if (dsb->state == STATE_STOPPING) {
547 dsb->state = STATE_STOPPED;
548 DSOUND_CheckEvent(dsb, 0, 0);
549 } else if (dsb->state != STATE_STOPPED) {
551 /* if the buffer was starting, it must be playing now */
552 if (dsb->state == STATE_STARTING)
553 dsb->state = STATE_PLAYING;
555 /* mix next buffer into the main buffer */
556 DSOUND_MixOne(dsb, writepos, mixlen);
558 *all_stopped = FALSE;
560 RtlReleaseResource(&dsb->lock);
566 * Add buffers to the emulated wave device system.
568 * device = The current dsound playback device
569 * force = If TRUE, the function will buffer up as many frags as possible,
570 * even though and will ignore the actual state of the primary buffer.
575 static void DSOUND_WaveQueue(DirectSoundDevice *device, BOOL force)
577 DWORD prebuf_frames, prebuf_bytes, read_offs_bytes;
581 TRACE("(%p)\n", device);
583 read_offs_bytes = (device->playing_offs_bytes + device->in_mmdev_bytes) % device->buflen;
585 TRACE("read_offs_bytes = %u, playing_offs_bytes = %u, in_mmdev_bytes: %u, prebuf = %u\n",
586 read_offs_bytes, device->playing_offs_bytes, device->in_mmdev_bytes, device->prebuf);
590 if(device->mixpos < device->playing_offs_bytes)
591 prebuf_bytes = device->mixpos + device->buflen - device->playing_offs_bytes;
593 prebuf_bytes = device->mixpos - device->playing_offs_bytes;
596 /* buffer the maximum amount of frags */
597 prebuf_bytes = device->prebuf * device->fraglen;
599 /* limit to the queue we have left */
600 if(device->in_mmdev_bytes + prebuf_bytes > device->prebuf * device->fraglen)
601 prebuf_bytes = device->prebuf * device->fraglen - device->in_mmdev_bytes;
603 TRACE("prebuf_bytes = %u\n", prebuf_bytes);
608 device->in_mmdev_bytes += prebuf_bytes;
610 if(prebuf_bytes + read_offs_bytes > device->buflen){
611 DWORD chunk_bytes = device->buflen - read_offs_bytes;
612 prebuf_frames = chunk_bytes / device->pwfx->nBlockAlign;
613 prebuf_bytes -= chunk_bytes;
615 prebuf_frames = prebuf_bytes / device->pwfx->nBlockAlign;
619 hr = IAudioRenderClient_GetBuffer(device->render, prebuf_frames, &buffer);
621 WARN("GetBuffer failed: %08x\n", hr);
625 memcpy(buffer, device->buffer + read_offs_bytes,
626 prebuf_frames * device->pwfx->nBlockAlign);
628 hr = IAudioRenderClient_ReleaseBuffer(device->render, prebuf_frames, 0);
630 WARN("ReleaseBuffer failed: %08x\n", hr);
634 /* check if anything wrapped */
635 if(prebuf_bytes > 0){
636 prebuf_frames = prebuf_bytes / device->pwfx->nBlockAlign;
638 hr = IAudioRenderClient_GetBuffer(device->render, prebuf_frames, &buffer);
640 WARN("GetBuffer failed: %08x\n", hr);
644 memcpy(buffer, device->buffer, prebuf_frames * device->pwfx->nBlockAlign);
646 hr = IAudioRenderClient_ReleaseBuffer(device->render, prebuf_frames, 0);
648 WARN("ReleaseBuffer failed: %08x\n", hr);
653 TRACE("in_mmdev_bytes now = %i\n", device->in_mmdev_bytes);
657 * Perform mixing for a Direct Sound device. That is, go through all the
658 * secondary buffers (the sound bites currently playing) and mix them in
659 * to the primary buffer (the device buffer).
661 * The mixing procedure goes:
663 * secondary->buffer (secondary format)
664 * =[Resample]=> device->tmp_buffer (float format)
665 * =[Volume]=> device->tmp_buffer (float format)
666 * =[Mix]=> device->mix_buffer (float format)
667 * =[Reformat]=> device->buffer (device format)
669 static void DSOUND_PerformMix(DirectSoundDevice *device)
671 UINT32 pad, to_mix_frags, to_mix_bytes;
674 TRACE("(%p)\n", device);
677 EnterCriticalSection(&device->mixlock);
679 hr = IAudioClient_GetCurrentPadding(device->client, &pad);
681 WARN("GetCurrentPadding failed: %08x\n", hr);
682 LeaveCriticalSection(&device->mixlock);
686 to_mix_frags = device->prebuf - (pad * device->pwfx->nBlockAlign + device->fraglen - 1) / device->fraglen;
688 to_mix_bytes = to_mix_frags * device->fraglen;
690 if(device->in_mmdev_bytes > 0){
691 DWORD delta_bytes = min(to_mix_bytes, device->in_mmdev_bytes);
692 device->in_mmdev_bytes -= delta_bytes;
693 device->playing_offs_bytes += delta_bytes;
694 device->playing_offs_bytes %= device->buflen;
697 if (device->priolevel != DSSCL_WRITEPRIMARY) {
698 BOOL recover = FALSE, all_stopped = FALSE;
699 DWORD playpos, writepos, writelead, maxq, prebuff_max, prebuff_left, size1, size2;
703 /* the sound of silence */
704 nfiller = device->pwfx->wBitsPerSample == 8 ? 128 : 0;
706 /* get the position in the primary buffer */
707 if (DSOUND_PrimaryGetPosition(device, &playpos, &writepos) != 0){
708 LeaveCriticalSection(&(device->mixlock));
712 TRACE("primary playpos=%d, writepos=%d, clrpos=%d, mixpos=%d, buflen=%d\n",
713 playpos,writepos,device->playpos,device->mixpos,device->buflen);
714 assert(device->playpos < device->buflen);
716 /* calc maximum prebuff */
717 prebuff_max = (device->prebuf * device->fraglen);
719 /* check how close we are to an underrun. It occurs when the writepos overtakes the mixpos */
720 prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);
721 writelead = DSOUND_BufPtrDiff(device->buflen, writepos, playpos);
723 /* check for underrun. underrun occurs when the write position passes the mix position
724 * also wipe out just-played sound data */
725 if((prebuff_left > prebuff_max) || (device->state == STATE_STOPPED) || (device->state == STATE_STARTING)){
726 if (device->state == STATE_STOPPING || device->state == STATE_PLAYING)
727 WARN("Probable buffer underrun\n");
728 else TRACE("Buffer starting or buffer underrun\n");
730 /* recover mixing for all buffers */
733 /* reset mix position to write position */
734 device->mixpos = writepos;
736 ZeroMemory(device->buffer, device->buflen);
737 } else if (playpos < device->playpos) {
738 buf1 = device->buffer + device->playpos;
739 buf2 = device->buffer;
740 size1 = device->buflen - device->playpos;
742 FillMemory(buf1, size1, nfiller);
743 if (playpos && (!buf2 || !size2))
744 FIXME("%d: (%d, %d)=>(%d, %d) There should be an additional buffer here!!\n", __LINE__, device->playpos, device->mixpos, playpos, writepos);
745 FillMemory(buf2, size2, nfiller);
747 buf1 = device->buffer + device->playpos;
749 size1 = playpos - device->playpos;
751 FillMemory(buf1, size1, nfiller);
753 device->playpos = playpos;
755 /* find the maximum we can prebuffer from current write position */
756 maxq = (writelead < prebuff_max) ? (prebuff_max - writelead) : 0;
758 TRACE("prebuff_left = %d, prebuff_max = %dx%d=%d, writelead=%d\n",
759 prebuff_left, device->prebuf, device->fraglen, prebuff_max, writelead);
761 ZeroMemory(device->mix_buffer, device->mix_buffer_len);
764 DSOUND_MixToPrimary(device, writepos, maxq, recover, &all_stopped);
766 if (maxq + writepos > device->buflen)
768 DWORD todo = device->buflen - writepos;
769 DWORD offs_float = (todo / device->pwfx->nBlockAlign) * device->pwfx->nChannels;
770 device->normfunction(device->mix_buffer, device->buffer + writepos, todo);
771 device->normfunction(device->mix_buffer + offs_float, device->buffer, maxq - todo);
774 device->normfunction(device->mix_buffer, device->buffer + writepos, maxq);
776 /* update the mix position, taking wrap-around into account */
777 device->mixpos = writepos + maxq;
778 device->mixpos %= device->buflen;
780 /* update prebuff left */
781 prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);
783 /* check if have a whole fragment */
784 if (prebuff_left >= device->fraglen){
786 /* update the wave queue */
787 DSOUND_WaveQueue(device, FALSE);
789 /* buffers are full. start playing if applicable */
790 if(device->state == STATE_STARTING){
791 TRACE("started primary buffer\n");
792 if(DSOUND_PrimaryPlay(device) != DS_OK){
793 WARN("DSOUND_PrimaryPlay failed\n");
796 /* we are playing now */
797 device->state = STATE_PLAYING;
801 /* buffers are full. start stopping if applicable */
802 if(device->state == STATE_STOPPED){
803 TRACE("restarting primary buffer\n");
804 if(DSOUND_PrimaryPlay(device) != DS_OK){
805 WARN("DSOUND_PrimaryPlay failed\n");
808 /* start stopping again. as soon as there is no more data, it will stop */
809 device->state = STATE_STOPPING;
814 /* if device was stopping, its for sure stopped when all buffers have stopped */
815 else if((all_stopped == TRUE) && (device->state == STATE_STOPPING)){
816 TRACE("All buffers have stopped. Stopping primary buffer\n");
817 device->state = STATE_STOPPED;
819 /* stop the primary buffer now */
820 DSOUND_PrimaryStop(device);
823 } else if (device->state != STATE_STOPPED) {
825 DSOUND_WaveQueue(device, TRUE);
827 /* in the DSSCL_WRITEPRIMARY mode, the app is totally in charge... */
828 if (device->state == STATE_STARTING) {
829 if (DSOUND_PrimaryPlay(device) != DS_OK)
830 WARN("DSOUND_PrimaryPlay failed\n");
832 device->state = STATE_PLAYING;
834 else if (device->state == STATE_STOPPING) {
835 if (DSOUND_PrimaryStop(device) != DS_OK)
836 WARN("DSOUND_PrimaryStop failed\n");
838 device->state = STATE_STOPPED;
842 LeaveCriticalSection(&(device->mixlock));
846 DWORD CALLBACK DSOUND_mixthread(void *p)
848 DirectSoundDevice *dev = p;
849 TRACE("(%p)\n", dev);
855 * Some audio drivers are retarded and won't fire after being
856 * stopped, add a timeout to handle this.
858 ret = WaitForSingleObject(dev->sleepev, dev->sleeptime);
859 if (ret == WAIT_FAILED)
860 WARN("wait returned error %u %08x!\n", GetLastError(), GetLastError());
861 else if (ret != WAIT_OBJECT_0)
862 WARN("wait returned %08x!\n", ret);
866 RtlAcquireResourceShared(&(dev->buffer_list_lock), TRUE);
867 DSOUND_PerformMix(dev);
868 RtlReleaseResource(&(dev->buffer_list_lock));