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
)
186 int first
, left
, right
, check
;
188 if(dsb
->nrofnotifies
== 0)
191 if(dsb
->state
== STATE_STOPPED
){
192 TRACE("Stopped...\n");
193 /* DSBPN_OFFSETSTOP notifies are always at the start of the sorted array */
194 for(left
= 0; left
< dsb
->nrofnotifies
; ++left
){
195 if(dsb
->notifies
[left
].dwOffset
!= DSBPN_OFFSETSTOP
)
198 TRACE("Signalling %p\n", dsb
->notifies
[left
].hEventNotify
);
199 SetEvent(dsb
->notifies
[left
].hEventNotify
);
204 for(first
= 0; first
< dsb
->nrofnotifies
&& dsb
->notifies
[first
].dwOffset
== DSBPN_OFFSETSTOP
; ++first
)
207 if(first
== dsb
->nrofnotifies
)
210 check
= left
= first
;
211 right
= dsb
->nrofnotifies
- 1;
213 /* find leftmost notify that is greater than playpos */
214 while(left
!= right
){
215 check
= left
+ (right
- left
) / 2;
216 if(dsb
->notifies
[check
].dwOffset
< playpos
)
218 else if(dsb
->notifies
[check
].dwOffset
> playpos
)
226 TRACE("Not stopped: first notify: %u (%u), range: [%u,%u)\n", first
,
227 dsb
->notifies
[check
].dwOffset
, playpos
, (playpos
+ len
) % dsb
->buflen
);
229 /* send notifications in range */
230 for(check
= left
; check
< dsb
->nrofnotifies
; ++check
){
231 if(dsb
->notifies
[check
].dwOffset
>= playpos
+ len
)
234 TRACE("Signalling %p (%u)\n", dsb
->notifies
[check
].hEventNotify
, dsb
->notifies
[check
].dwOffset
);
235 SetEvent(dsb
->notifies
[check
].hEventNotify
);
238 if(playpos
+ len
> dsb
->buflen
){
239 for(check
= first
; check
< left
; ++check
){
240 if(dsb
->notifies
[check
].dwOffset
>= (playpos
+ len
) % dsb
->buflen
)
243 TRACE("Signalling %p (%u)\n", dsb
->notifies
[check
].hEventNotify
, dsb
->notifies
[check
].dwOffset
);
244 SetEvent(dsb
->notifies
[check
].hEventNotify
);
249 static inline float get_current_sample(const IDirectSoundBufferImpl
*dsb
,
250 DWORD mixpos
, DWORD channel
)
252 if (mixpos
>= dsb
->buflen
&& !(dsb
->playflags
& DSBPLAY_LOOPING
))
254 return dsb
->get(dsb
, mixpos
% dsb
->buflen
, channel
);
257 static UINT
cp_fields_noresample(IDirectSoundBufferImpl
*dsb
, UINT count
)
259 UINT istride
= dsb
->pwfx
->nBlockAlign
;
260 UINT ostride
= dsb
->device
->pwfx
->nChannels
* sizeof(float);
262 for (i
= 0; i
< count
; i
++)
263 for (channel
= 0; channel
< dsb
->mix_channels
; channel
++)
264 dsb
->put(dsb
, i
* ostride
, channel
, get_current_sample(dsb
,
265 dsb
->sec_mixpos
+ i
* istride
, channel
));
269 static UINT
cp_fields_resample(IDirectSoundBufferImpl
*dsb
, UINT count
, float *freqAcc
)
272 UINT istride
= dsb
->pwfx
->nBlockAlign
;
273 UINT ostride
= dsb
->device
->pwfx
->nChannels
* sizeof(float);
275 float freqAdjust
= dsb
->freqAdjust
;
276 float freqAcc_start
= *freqAcc
;
277 float freqAcc_end
= freqAcc_start
+ count
* freqAdjust
;
278 UINT dsbfirstep
= dsb
->firstep
;
279 UINT channels
= dsb
->mix_channels
;
280 UINT max_ipos
= freqAcc_start
+ count
* freqAdjust
;
282 UINT fir_cachesize
= (fir_len
+ dsbfirstep
- 2) / dsbfirstep
;
283 UINT required_input
= max_ipos
+ fir_cachesize
;
285 float* intermediate
= HeapAlloc(GetProcessHeap(), 0,
286 sizeof(float) * required_input
* channels
);
288 float* fir_copy
= HeapAlloc(GetProcessHeap(), 0,
289 sizeof(float) * fir_cachesize
);
291 /* Important: this buffer MUST be non-interleaved
292 * if you want -msse3 to have any effect.
293 * This is good for CPU cache effects, too.
295 float* itmp
= intermediate
;
296 for (channel
= 0; channel
< channels
; channel
++)
297 for (i
= 0; i
< required_input
; i
++)
298 *(itmp
++) = get_current_sample(dsb
,
299 dsb
->sec_mixpos
+ i
* istride
, channel
);
301 for(i
= 0; i
< count
; ++i
) {
302 float total_fir_steps
= (freqAcc_start
+ i
* freqAdjust
) * dsbfirstep
;
303 UINT int_fir_steps
= total_fir_steps
;
304 UINT ipos
= int_fir_steps
/ dsbfirstep
;
306 UINT idx
= (ipos
+ 1) * dsbfirstep
- int_fir_steps
- 1;
307 float rem
= int_fir_steps
+ 1.0 - total_fir_steps
;
310 while (idx
< fir_len
- 1) {
311 fir_copy
[fir_used
++] = fir
[idx
] * (1.0 - rem
) + fir
[idx
+ 1] * rem
;
315 assert(fir_used
<= fir_cachesize
);
316 assert(ipos
+ fir_used
<= required_input
);
318 for (channel
= 0; channel
< dsb
->mix_channels
; channel
++) {
321 float* cache
= &intermediate
[channel
* required_input
+ ipos
];
322 for (j
= 0; j
< fir_used
; j
++)
323 sum
+= fir_copy
[j
] * cache
[j
];
324 dsb
->put(dsb
, i
* ostride
, channel
, sum
* dsb
->firgain
);
328 freqAcc_end
-= (int)freqAcc_end
;
329 *freqAcc
= freqAcc_end
;
331 HeapFree(GetProcessHeap(), 0, fir_copy
);
332 HeapFree(GetProcessHeap(), 0, intermediate
);
337 static void cp_fields(IDirectSoundBufferImpl
*dsb
, UINT count
, float *freqAcc
)
341 if (dsb
->freqAdjust
== 1.0)
342 adv
= cp_fields_noresample(dsb
, count
); /* *freqAcc is unmodified */
344 adv
= cp_fields_resample(dsb
, count
, freqAcc
);
346 ipos
= dsb
->sec_mixpos
+ adv
* dsb
->pwfx
->nBlockAlign
;
347 if (ipos
>= dsb
->buflen
) {
348 if (dsb
->playflags
& DSBPLAY_LOOPING
)
352 dsb
->state
= STATE_STOPPED
;
356 dsb
->sec_mixpos
= ipos
;
360 * Calculate the distance between two buffer offsets, taking wraparound
363 static inline DWORD
DSOUND_BufPtrDiff(DWORD buflen
, DWORD ptr1
, DWORD ptr2
)
365 /* If these asserts fail, the problem is not here, but in the underlying code */
366 assert(ptr1
< buflen
);
367 assert(ptr2
< buflen
);
371 return buflen
+ ptr1
- ptr2
;
375 * Mix at most the given amount of data into the allocated temporary buffer
376 * of the given secondary buffer, starting from the dsb's first currently
377 * unsampled frame (writepos), translating frequency (pitch), stereo/mono
378 * and bits-per-sample so that it is ideal for the primary buffer.
379 * Doesn't perform any mixing - this is a straight copy/convert operation.
381 * dsb = the secondary buffer
382 * writepos = Starting position of changed buffer
383 * len = number of bytes to resample from writepos
385 * NOTE: writepos + len <= buflen. When called by mixer, MixOne makes sure of this.
387 static void DSOUND_MixToTemporary(IDirectSoundBufferImpl
*dsb
, DWORD frames
)
389 UINT size_bytes
= frames
* sizeof(float) * dsb
->device
->pwfx
->nChannels
;
391 if (dsb
->device
->tmp_buffer_len
< size_bytes
|| !dsb
->device
->tmp_buffer
)
393 dsb
->device
->tmp_buffer_len
= size_bytes
;
394 if (dsb
->device
->tmp_buffer
)
395 dsb
->device
->tmp_buffer
= HeapReAlloc(GetProcessHeap(), 0, dsb
->device
->tmp_buffer
, size_bytes
);
397 dsb
->device
->tmp_buffer
= HeapAlloc(GetProcessHeap(), 0, size_bytes
);
400 cp_fields(dsb
, frames
, &dsb
->freqAcc
);
403 static void DSOUND_MixerVol(const IDirectSoundBufferImpl
*dsb
, INT frames
)
407 UINT channels
= dsb
->device
->pwfx
->nChannels
, chan
;
409 TRACE("(%p,%d)\n",dsb
,frames
);
410 TRACE("left = %x, right = %x\n", dsb
->volpan
.dwTotalLeftAmpFactor
,
411 dsb
->volpan
.dwTotalRightAmpFactor
);
413 if ((!(dsb
->dsbd
.dwFlags
& DSBCAPS_CTRLPAN
) || (dsb
->volpan
.lPan
== 0)) &&
414 (!(dsb
->dsbd
.dwFlags
& DSBCAPS_CTRLVOLUME
) || (dsb
->volpan
.lVolume
== 0)) &&
415 !(dsb
->dsbd
.dwFlags
& DSBCAPS_CTRL3D
))
416 return; /* Nothing to do */
418 if (channels
!= 1 && channels
!= 2)
420 FIXME("There is no support for %u channels\n", channels
);
424 vLeft
= dsb
->volpan
.dwTotalLeftAmpFactor
/ ((float)0xFFFF);
425 vRight
= dsb
->volpan
.dwTotalRightAmpFactor
/ ((float)0xFFFF);
426 for(i
= 0; i
< frames
; ++i
){
427 for(chan
= 0; chan
< channels
; ++chan
){
429 dsb
->device
->tmp_buffer
[i
* channels
+ chan
] *= vLeft
;
431 dsb
->device
->tmp_buffer
[i
* channels
+ chan
] *= vRight
;
437 * Mix (at most) the given number of bytes into the given position of the
438 * device buffer, from the secondary buffer "dsb" (starting at the current
439 * mix position for that buffer).
441 * Returns the number of bytes actually mixed into the device buffer. This
442 * will match fraglen unless the end of the secondary buffer is reached
443 * (and it is not looping).
445 * dsb = the secondary buffer to mix from
446 * writepos = position (offset) in device buffer to write at
447 * fraglen = number of bytes to mix
449 static DWORD
DSOUND_MixInBuffer(IDirectSoundBufferImpl
*dsb
, DWORD writepos
, DWORD fraglen
)
454 UINT frames
= fraglen
/ dsb
->device
->pwfx
->nBlockAlign
;
456 TRACE("sec_mixpos=%d/%d\n", dsb
->sec_mixpos
, dsb
->buflen
);
457 TRACE("(%p,%d,%d)\n",dsb
,writepos
,fraglen
);
459 if (len
% dsb
->device
->pwfx
->nBlockAlign
) {
460 INT nBlockAlign
= dsb
->device
->pwfx
->nBlockAlign
;
461 ERR("length not a multiple of block size, len = %d, block size = %d\n", len
, nBlockAlign
);
462 len
-= len
% nBlockAlign
; /* data alignment */
465 /* Resample buffer to temporary buffer specifically allocated for this purpose, if needed */
466 oldpos
= dsb
->sec_mixpos
;
468 DSOUND_MixToTemporary(dsb
, frames
);
469 ibuf
= dsb
->device
->tmp_buffer
;
471 /* Apply volume if needed */
472 DSOUND_MixerVol(dsb
, frames
);
474 mixieee32(ibuf
, dsb
->device
->mix_buffer
, frames
* dsb
->device
->pwfx
->nChannels
);
476 /* check for notification positions */
477 if (dsb
->dsbd
.dwFlags
& DSBCAPS_CTRLPOSITIONNOTIFY
&&
478 dsb
->state
!= STATE_STARTING
) {
479 INT ilen
= DSOUND_BufPtrDiff(dsb
->buflen
, dsb
->sec_mixpos
, oldpos
);
480 DSOUND_CheckEvent(dsb
, oldpos
, ilen
);
487 * Mix some frames from the given secondary buffer "dsb" into the device
490 * dsb = the secondary buffer
491 * playpos = the current play position in the device buffer (primary buffer)
492 * writepos = the current safe-to-write position in the device buffer
493 * mixlen = the maximum number of bytes in the primary buffer to mix, from the
496 * Returns: the number of bytes beyond the writepos that were mixed.
498 static DWORD
DSOUND_MixOne(IDirectSoundBufferImpl
*dsb
, DWORD writepos
, DWORD mixlen
)
500 DWORD primary_done
= 0;
502 TRACE("(%p,%d,%d)\n",dsb
,writepos
,mixlen
);
503 TRACE("writepos=%d, mixlen=%d\n", writepos
, mixlen
);
504 TRACE("looping=%d, leadin=%d\n", dsb
->playflags
, dsb
->leadin
);
506 /* If leading in, only mix about 20 ms, and 'skip' mixing the rest, for more fluid pointer advancement */
507 /* FIXME: Is this needed? */
508 if (dsb
->leadin
&& dsb
->state
== STATE_STARTING
) {
509 if (mixlen
> 2 * dsb
->device
->fraglen
) {
510 primary_done
= mixlen
- 2 * dsb
->device
->fraglen
;
511 mixlen
= 2 * dsb
->device
->fraglen
;
512 writepos
+= primary_done
;
513 dsb
->sec_mixpos
+= (primary_done
/ dsb
->device
->pwfx
->nBlockAlign
) *
514 dsb
->pwfx
->nBlockAlign
* dsb
->freqAdjust
;
520 TRACE("mixlen (primary) = %i\n", mixlen
);
522 /* First try to mix to the end of the buffer if possible
523 * Theoretically it would allow for better optimization
525 primary_done
+= DSOUND_MixInBuffer(dsb
, writepos
, mixlen
);
527 TRACE("total mixed data=%d\n", primary_done
);
529 /* Report back the total prebuffered amount for this buffer */
534 * For a DirectSoundDevice, go through all the currently playing buffers and
535 * mix them in to the device buffer.
537 * writepos = the current safe-to-write position in the primary buffer
538 * mixlen = the maximum amount to mix into the primary buffer
539 * (beyond the current writepos)
540 * recover = true if the sound device may have been reset and the write
541 * position in the device buffer changed
542 * all_stopped = reports back if all buffers have stopped
544 * Returns: the length beyond the writepos that was mixed to.
547 static void DSOUND_MixToPrimary(const DirectSoundDevice
*device
, DWORD writepos
, DWORD mixlen
, BOOL recover
, BOOL
*all_stopped
)
550 IDirectSoundBufferImpl
*dsb
;
552 /* unless we find a running buffer, all have stopped */
555 TRACE("(%d,%d,%d)\n", writepos
, mixlen
, recover
);
556 for (i
= 0; i
< device
->nrofbuffers
; i
++) {
557 dsb
= device
->buffers
[i
];
559 TRACE("MixToPrimary for %p, state=%d\n", dsb
, dsb
->state
);
561 if (dsb
->buflen
&& dsb
->state
) {
562 TRACE("Checking %p, mixlen=%d\n", dsb
, mixlen
);
563 RtlAcquireResourceShared(&dsb
->lock
, TRUE
);
564 /* if buffer is stopping it is stopped now */
565 if (dsb
->state
== STATE_STOPPING
) {
566 dsb
->state
= STATE_STOPPED
;
567 DSOUND_CheckEvent(dsb
, 0, 0);
568 } else if (dsb
->state
!= STATE_STOPPED
) {
570 /* if the buffer was starting, it must be playing now */
571 if (dsb
->state
== STATE_STARTING
)
572 dsb
->state
= STATE_PLAYING
;
574 /* mix next buffer into the main buffer */
575 DSOUND_MixOne(dsb
, writepos
, mixlen
);
577 *all_stopped
= FALSE
;
579 RtlReleaseResource(&dsb
->lock
);
585 * Add buffers to the emulated wave device system.
587 * device = The current dsound playback device
588 * force = If TRUE, the function will buffer up as many frags as possible,
589 * even though and will ignore the actual state of the primary buffer.
594 static void DSOUND_WaveQueue(DirectSoundDevice
*device
, BOOL force
)
596 DWORD prebuf_frames
, prebuf_bytes
, read_offs_bytes
;
600 TRACE("(%p)\n", device
);
602 read_offs_bytes
= (device
->playing_offs_bytes
+ device
->in_mmdev_bytes
) % device
->buflen
;
604 TRACE("read_offs_bytes = %u, playing_offs_bytes = %u, in_mmdev_bytes: %u, prebuf = %u\n",
605 read_offs_bytes
, device
->playing_offs_bytes
, device
->in_mmdev_bytes
, device
->prebuf
);
609 if(device
->mixpos
< device
->playing_offs_bytes
)
610 prebuf_bytes
= device
->mixpos
+ device
->buflen
- device
->playing_offs_bytes
;
612 prebuf_bytes
= device
->mixpos
- device
->playing_offs_bytes
;
615 /* buffer the maximum amount of frags */
616 prebuf_bytes
= device
->prebuf
* device
->fraglen
;
618 /* limit to the queue we have left */
619 if(device
->in_mmdev_bytes
+ prebuf_bytes
> device
->prebuf
* device
->fraglen
)
620 prebuf_bytes
= device
->prebuf
* device
->fraglen
- device
->in_mmdev_bytes
;
622 TRACE("prebuf_bytes = %u\n", prebuf_bytes
);
627 if(prebuf_bytes
+ read_offs_bytes
> device
->buflen
){
628 DWORD chunk_bytes
= device
->buflen
- read_offs_bytes
;
629 prebuf_frames
= chunk_bytes
/ device
->pwfx
->nBlockAlign
;
630 prebuf_bytes
-= chunk_bytes
;
632 prebuf_frames
= prebuf_bytes
/ device
->pwfx
->nBlockAlign
;
636 hr
= IAudioRenderClient_GetBuffer(device
->render
, prebuf_frames
, &buffer
);
638 WARN("GetBuffer failed: %08x\n", hr
);
642 memcpy(buffer
, device
->buffer
+ read_offs_bytes
,
643 prebuf_frames
* device
->pwfx
->nBlockAlign
);
645 hr
= IAudioRenderClient_ReleaseBuffer(device
->render
, prebuf_frames
, 0);
647 WARN("ReleaseBuffer failed: %08x\n", hr
);
651 device
->in_mmdev_bytes
+= prebuf_frames
* device
->pwfx
->nBlockAlign
;
653 /* check if anything wrapped */
654 if(prebuf_bytes
> 0){
655 prebuf_frames
= prebuf_bytes
/ device
->pwfx
->nBlockAlign
;
657 hr
= IAudioRenderClient_GetBuffer(device
->render
, prebuf_frames
, &buffer
);
659 WARN("GetBuffer failed: %08x\n", hr
);
663 memcpy(buffer
, device
->buffer
, prebuf_frames
* device
->pwfx
->nBlockAlign
);
665 hr
= IAudioRenderClient_ReleaseBuffer(device
->render
, prebuf_frames
, 0);
667 WARN("ReleaseBuffer failed: %08x\n", hr
);
670 device
->in_mmdev_bytes
+= prebuf_frames
* device
->pwfx
->nBlockAlign
;
673 TRACE("in_mmdev_bytes now = %i\n", device
->in_mmdev_bytes
);
677 * Perform mixing for a Direct Sound device. That is, go through all the
678 * secondary buffers (the sound bites currently playing) and mix them in
679 * to the primary buffer (the device buffer).
681 * The mixing procedure goes:
683 * secondary->buffer (secondary format)
684 * =[Resample]=> device->tmp_buffer (float format)
685 * =[Volume]=> device->tmp_buffer (float format)
686 * =[Mix]=> device->mix_buffer (float format)
687 * =[Reformat]=> device->buffer (device format)
689 static void DSOUND_PerformMix(DirectSoundDevice
*device
)
691 UINT32 pad
, to_mix_frags
, to_mix_bytes
;
694 TRACE("(%p)\n", device
);
697 EnterCriticalSection(&device
->mixlock
);
699 hr
= IAudioClient_GetCurrentPadding(device
->client
, &pad
);
701 WARN("GetCurrentPadding failed: %08x\n", hr
);
702 LeaveCriticalSection(&device
->mixlock
);
706 to_mix_frags
= device
->prebuf
- (pad
* device
->pwfx
->nBlockAlign
+ device
->fraglen
- 1) / device
->fraglen
;
708 to_mix_bytes
= to_mix_frags
* device
->fraglen
;
710 if(device
->in_mmdev_bytes
> 0){
711 DWORD delta_bytes
= min(to_mix_bytes
, device
->in_mmdev_bytes
);
712 device
->in_mmdev_bytes
-= delta_bytes
;
713 device
->playing_offs_bytes
+= delta_bytes
;
714 device
->playing_offs_bytes
%= device
->buflen
;
717 if (device
->priolevel
!= DSSCL_WRITEPRIMARY
) {
718 BOOL recover
= FALSE
, all_stopped
= FALSE
;
719 DWORD playpos
, writepos
, writelead
, maxq
, prebuff_max
, prebuff_left
, size1
, size2
;
723 /* the sound of silence */
724 nfiller
= device
->pwfx
->wBitsPerSample
== 8 ? 128 : 0;
726 /* get the position in the primary buffer */
727 if (DSOUND_PrimaryGetPosition(device
, &playpos
, &writepos
) != 0){
728 LeaveCriticalSection(&(device
->mixlock
));
732 TRACE("primary playpos=%d, writepos=%d, clrpos=%d, mixpos=%d, buflen=%d\n",
733 playpos
,writepos
,device
->playpos
,device
->mixpos
,device
->buflen
);
734 assert(device
->playpos
< device
->buflen
);
736 /* calc maximum prebuff */
737 prebuff_max
= (device
->prebuf
* device
->fraglen
);
739 /* check how close we are to an underrun. It occurs when the writepos overtakes the mixpos */
740 prebuff_left
= DSOUND_BufPtrDiff(device
->buflen
, device
->mixpos
, playpos
);
741 writelead
= DSOUND_BufPtrDiff(device
->buflen
, writepos
, playpos
);
743 /* check for underrun. underrun occurs when the write position passes the mix position
744 * also wipe out just-played sound data */
745 if((prebuff_left
> prebuff_max
) || (device
->state
== STATE_STOPPED
) || (device
->state
== STATE_STARTING
)){
746 if (device
->state
== STATE_STOPPING
|| device
->state
== STATE_PLAYING
)
747 WARN("Probable buffer underrun\n");
748 else TRACE("Buffer starting or buffer underrun\n");
750 /* recover mixing for all buffers */
753 /* reset mix position to write position */
754 device
->mixpos
= writepos
;
756 ZeroMemory(device
->buffer
, device
->buflen
);
757 } else if (playpos
< device
->playpos
) {
758 buf1
= device
->buffer
+ device
->playpos
;
759 buf2
= device
->buffer
;
760 size1
= device
->buflen
- device
->playpos
;
762 FillMemory(buf1
, size1
, nfiller
);
763 if (playpos
&& (!buf2
|| !size2
))
764 FIXME("%d: (%d, %d)=>(%d, %d) There should be an additional buffer here!!\n", __LINE__
, device
->playpos
, device
->mixpos
, playpos
, writepos
);
765 FillMemory(buf2
, size2
, nfiller
);
767 buf1
= device
->buffer
+ device
->playpos
;
769 size1
= playpos
- device
->playpos
;
771 FillMemory(buf1
, size1
, nfiller
);
773 device
->playpos
= playpos
;
775 /* find the maximum we can prebuffer from current write position */
776 maxq
= (writelead
< prebuff_max
) ? (prebuff_max
- writelead
) : 0;
778 TRACE("prebuff_left = %d, prebuff_max = %dx%d=%d, writelead=%d\n",
779 prebuff_left
, device
->prebuf
, device
->fraglen
, prebuff_max
, writelead
);
781 ZeroMemory(device
->mix_buffer
, device
->mix_buffer_len
);
784 DSOUND_MixToPrimary(device
, writepos
, maxq
, recover
, &all_stopped
);
786 if (maxq
+ writepos
> device
->buflen
)
788 DWORD todo
= device
->buflen
- writepos
;
789 DWORD offs_float
= (todo
/ device
->pwfx
->nBlockAlign
) * device
->pwfx
->nChannels
;
790 device
->normfunction(device
->mix_buffer
, device
->buffer
+ writepos
, todo
);
791 device
->normfunction(device
->mix_buffer
+ offs_float
, device
->buffer
, maxq
- todo
);
794 device
->normfunction(device
->mix_buffer
, device
->buffer
+ writepos
, maxq
);
796 /* update the mix position, taking wrap-around into account */
797 device
->mixpos
= writepos
+ maxq
;
798 device
->mixpos
%= device
->buflen
;
800 /* update prebuff left */
801 prebuff_left
= DSOUND_BufPtrDiff(device
->buflen
, device
->mixpos
, playpos
);
803 /* check if have a whole fragment */
804 if (prebuff_left
>= device
->fraglen
){
806 /* update the wave queue */
807 DSOUND_WaveQueue(device
, FALSE
);
809 /* buffers are full. start playing if applicable */
810 if(device
->state
== STATE_STARTING
){
811 TRACE("started primary buffer\n");
812 if(DSOUND_PrimaryPlay(device
) != DS_OK
){
813 WARN("DSOUND_PrimaryPlay failed\n");
816 /* we are playing now */
817 device
->state
= STATE_PLAYING
;
821 /* buffers are full. start stopping if applicable */
822 if(device
->state
== STATE_STOPPED
){
823 TRACE("restarting primary buffer\n");
824 if(DSOUND_PrimaryPlay(device
) != DS_OK
){
825 WARN("DSOUND_PrimaryPlay failed\n");
828 /* start stopping again. as soon as there is no more data, it will stop */
829 device
->state
= STATE_STOPPING
;
834 /* if device was stopping, its for sure stopped when all buffers have stopped */
835 else if((all_stopped
== TRUE
) && (device
->state
== STATE_STOPPING
)){
836 TRACE("All buffers have stopped. Stopping primary buffer\n");
837 device
->state
= STATE_STOPPED
;
839 /* stop the primary buffer now */
840 DSOUND_PrimaryStop(device
);
843 } else if (device
->state
!= STATE_STOPPED
) {
845 DSOUND_WaveQueue(device
, TRUE
);
847 /* in the DSSCL_WRITEPRIMARY mode, the app is totally in charge... */
848 if (device
->state
== STATE_STARTING
) {
849 if (DSOUND_PrimaryPlay(device
) != DS_OK
)
850 WARN("DSOUND_PrimaryPlay failed\n");
852 device
->state
= STATE_PLAYING
;
854 else if (device
->state
== STATE_STOPPING
) {
855 if (DSOUND_PrimaryStop(device
) != DS_OK
)
856 WARN("DSOUND_PrimaryStop failed\n");
858 device
->state
= STATE_STOPPED
;
862 LeaveCriticalSection(&(device
->mixlock
));
866 DWORD CALLBACK
DSOUND_mixthread(void *p
)
868 DirectSoundDevice
*dev
= p
;
869 TRACE("(%p)\n", dev
);
875 * Some audio drivers are retarded and won't fire after being
876 * stopped, add a timeout to handle this.
878 ret
= WaitForSingleObject(dev
->sleepev
, dev
->sleeptime
);
879 if (ret
== WAIT_FAILED
)
880 WARN("wait returned error %u %08x!\n", GetLastError(), GetLastError());
881 else if (ret
!= WAIT_OBJECT_0
)
882 WARN("wait returned %08x!\n", ret
);
886 RtlAcquireResourceShared(&(dev
->buffer_list_lock
), TRUE
);
887 DSOUND_PerformMix(dev
);
888 RtlReleaseResource(&(dev
->buffer_list_lock
));