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"
44 WINE_DEFAULT_DEBUG_CHANNEL(dsound
);
46 void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan
)
49 TRACE("(%p)\n",volpan
);
51 TRACE("Vol=%d Pan=%d\n", volpan
->lVolume
, volpan
->lPan
);
52 /* the AmpFactors are expressed in 16.16 fixed point */
53 volpan
->dwVolAmpFactor
= (ULONG
) (pow(2.0, volpan
->lVolume
/ 600.0) * 0xffff);
54 /* FIXME: dwPan{Left|Right}AmpFactor */
56 /* FIXME: use calculated vol and pan ampfactors */
57 temp
= (double) (volpan
->lVolume
- (volpan
->lPan
> 0 ? volpan
->lPan
: 0));
58 volpan
->dwTotalLeftAmpFactor
= (ULONG
) (pow(2.0, temp
/ 600.0) * 0xffff);
59 temp
= (double) (volpan
->lVolume
+ (volpan
->lPan
< 0 ? volpan
->lPan
: 0));
60 volpan
->dwTotalRightAmpFactor
= (ULONG
) (pow(2.0, temp
/ 600.0) * 0xffff);
62 TRACE("left = %x, right = %x\n", volpan
->dwTotalLeftAmpFactor
, volpan
->dwTotalRightAmpFactor
);
65 void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan
)
68 TRACE("(%p)\n",volpan
);
70 TRACE("left=%x, right=%x\n",volpan
->dwTotalLeftAmpFactor
,volpan
->dwTotalRightAmpFactor
);
71 if (volpan
->dwTotalLeftAmpFactor
==0)
74 left
=600 * log(((double)volpan
->dwTotalLeftAmpFactor
) / 0xffff) / log(2);
75 if (volpan
->dwTotalRightAmpFactor
==0)
78 right
=600 * log(((double)volpan
->dwTotalRightAmpFactor
) / 0xffff) / log(2);
81 volpan
->lVolume
=right
;
82 volpan
->dwVolAmpFactor
=volpan
->dwTotalRightAmpFactor
;
87 volpan
->dwVolAmpFactor
=volpan
->dwTotalLeftAmpFactor
;
89 if (volpan
->lVolume
< -10000)
90 volpan
->lVolume
=-10000;
91 volpan
->lPan
=right
-left
;
92 if (volpan
->lPan
< -10000)
95 TRACE("Vol=%d Pan=%d\n", volpan
->lVolume
, volpan
->lPan
);
98 /** Convert a primary buffer position to a pointer position for device->mix_buffer
99 * device: DirectSoundDevice for which to calculate
100 * pos: Primary buffer position to converts
101 * Returns: Offset for mix_buffer
103 DWORD
DSOUND_bufpos_to_mixpos(const DirectSoundDevice
* device
, DWORD pos
)
105 DWORD ret
= pos
* 32 / device
->pwfx
->wBitsPerSample
;
106 if (device
->pwfx
->wBitsPerSample
== 32)
111 /* NOTE: Not all secpos have to always be mapped to a bufpos, other way around is always the case
112 * DWORD64 is used here because a single DWORD wouldn't be big enough to fit the freqAcc for big buffers
114 /** This function converts a 'native' sample pointer to a resampled pointer that fits for primary
115 * secmixpos is used to decide which freqAcc is needed
116 * overshot tells what the 'actual' secpos is now (optional)
118 DWORD
DSOUND_secpos_to_bufpos(const IDirectSoundBufferImpl
*dsb
, DWORD secpos
, DWORD secmixpos
, DWORD
* overshot
)
120 DWORD64 framelen
= secpos
/ dsb
->pwfx
->nBlockAlign
;
121 DWORD64 freqAdjust
= dsb
->freqAdjust
;
122 DWORD64 acc
, freqAcc
;
124 if (secpos
< secmixpos
)
125 freqAcc
= dsb
->freqAccNext
;
126 else freqAcc
= dsb
->freqAcc
;
127 acc
= (framelen
<< DSOUND_FREQSHIFT
) + (freqAdjust
- 1 - freqAcc
);
131 DWORD64 oshot
= acc
* freqAdjust
+ freqAcc
;
132 assert(oshot
>= framelen
<< DSOUND_FREQSHIFT
);
133 oshot
-= framelen
<< DSOUND_FREQSHIFT
;
134 *overshot
= (DWORD
)oshot
;
135 assert(*overshot
< dsb
->freqAdjust
);
137 return (DWORD
)acc
* dsb
->device
->pwfx
->nBlockAlign
;
140 /** Convert a resampled pointer that fits for primary to a 'native' sample pointer
141 * freqAccNext is used here rather than freqAcc: In case the app wants to fill up to
142 * the play position it won't overwrite it
144 static DWORD
DSOUND_bufpos_to_secpos(const IDirectSoundBufferImpl
*dsb
, DWORD bufpos
)
146 DWORD oAdv
= dsb
->device
->pwfx
->nBlockAlign
, iAdv
= dsb
->pwfx
->nBlockAlign
, pos
;
150 framelen
= bufpos
/oAdv
;
151 acc
= framelen
* (DWORD64
)dsb
->freqAdjust
+ (DWORD64
)dsb
->freqAccNext
;
152 acc
= acc
>> DSOUND_FREQSHIFT
;
153 pos
= (DWORD
)acc
* iAdv
;
154 if (pos
>= dsb
->buflen
)
155 /* Because of differences between freqAcc and freqAccNext, this might happen */
156 pos
= dsb
->buflen
- iAdv
;
157 TRACE("Converted %d/%d to %d/%d\n", bufpos
, dsb
->tmp_buffer_len
, pos
, dsb
->buflen
);
162 * Move freqAccNext to freqAcc, and find new values for buffer length and freqAccNext
164 static void DSOUND_RecalcFreqAcc(IDirectSoundBufferImpl
*dsb
)
166 if (!dsb
->freqneeded
) return;
167 dsb
->freqAcc
= dsb
->freqAccNext
;
168 dsb
->tmp_buffer_len
= DSOUND_secpos_to_bufpos(dsb
, dsb
->buflen
, 0, &dsb
->freqAccNext
);
169 TRACE("New freqadjust: %04x, new buflen: %d\n", dsb
->freqAccNext
, dsb
->tmp_buffer_len
);
173 * Recalculate the size for temporary buffer, and new writelead
174 * Should be called when one of the following things occur:
175 * - Primary buffer format is changed
176 * - This buffer format (frequency) is changed
178 void DSOUND_RecalcFormat(IDirectSoundBufferImpl
*dsb
)
180 BOOL needremix
= TRUE
, needresample
= (dsb
->freq
!= dsb
->device
->pwfx
->nSamplesPerSec
);
181 DWORD bAlign
= dsb
->pwfx
->nBlockAlign
, pAlign
= dsb
->device
->pwfx
->nBlockAlign
;
182 WAVEFORMATEXTENSIBLE
*pwfxe
;
187 pwfxe
= (WAVEFORMATEXTENSIBLE
*) dsb
->pwfx
;
189 if ((pwfxe
->Format
.wFormatTag
== WAVE_FORMAT_IEEE_FLOAT
) || ((pwfxe
->Format
.wFormatTag
== WAVE_FORMAT_EXTENSIBLE
)
190 && (IsEqualGUID(&pwfxe
->SubFormat
, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT
))))
193 /* calculate the 10ms write lead */
194 dsb
->writelead
= (dsb
->freq
/ 100) * dsb
->pwfx
->nBlockAlign
;
196 if ((dsb
->pwfx
->wBitsPerSample
== dsb
->device
->pwfx
->wBitsPerSample
) &&
197 (dsb
->pwfx
->nChannels
== dsb
->device
->pwfx
->nChannels
) && !needresample
&& !ieee
)
199 dsb
->freqAcc
= dsb
->freqAccNext
= 0;
200 dsb
->freqneeded
= needresample
;
203 dsb
->convert
= convertbpp
[4][dsb
->device
->pwfx
->wBitsPerSample
/8 - 1];
205 dsb
->convert
= convertbpp
[dsb
->pwfx
->wBitsPerSample
/8 - 1][dsb
->device
->pwfx
->wBitsPerSample
/8 - 1];
210 DSOUND_RecalcFreqAcc(dsb
);
212 dsb
->tmp_buffer_len
= dsb
->buflen
/ bAlign
* pAlign
;
214 else dsb
->tmp_buffer_len
= dsb
->buflen
;
215 dsb
->buf_mixpos
= DSOUND_secpos_to_bufpos(dsb
, dsb
->sec_mixpos
, 0, NULL
);
219 * Check for application callback requests for when the play position
220 * reaches certain points.
222 * The offsets that will be triggered will be those between the recorded
223 * "last played" position for the buffer (i.e. dsb->playpos) and "len" bytes
224 * beyond that position.
226 void DSOUND_CheckEvent(const IDirectSoundBufferImpl
*dsb
, DWORD playpos
, int len
)
230 LPDSBPOSITIONNOTIFY event
;
231 TRACE("(%p,%d)\n",dsb
,len
);
233 if (dsb
->nrofnotifies
== 0)
236 TRACE("(%p) buflen = %d, playpos = %d, len = %d\n",
237 dsb
, dsb
->buflen
, playpos
, len
);
238 for (i
= 0; i
< dsb
->nrofnotifies
; i
++) {
239 event
= dsb
->notifies
+ i
;
240 offset
= event
->dwOffset
;
241 TRACE("checking %d, position %d, event = %p\n",
242 i
, offset
, event
->hEventNotify
);
243 /* DSBPN_OFFSETSTOP has to be the last element. So this is */
244 /* OK. [Inside DirectX, p274] */
245 /* Windows does not seem to enforce this, and some apps rely */
246 /* on that, so we can't stop there. */
248 /* This also means we can't sort the entries by offset, */
249 /* because DSBPN_OFFSETSTOP == -1 */
250 if (offset
== DSBPN_OFFSETSTOP
) {
251 if (dsb
->state
== STATE_STOPPED
) {
252 SetEvent(event
->hEventNotify
);
253 TRACE("signalled event %p (%d)\n", event
->hEventNotify
, i
);
257 if ((playpos
+ len
) >= dsb
->buflen
) {
258 if ((offset
< ((playpos
+ len
) % dsb
->buflen
)) ||
259 (offset
>= playpos
)) {
260 TRACE("signalled event %p (%d)\n", event
->hEventNotify
, i
);
261 SetEvent(event
->hEventNotify
);
264 if ((offset
>= playpos
) && (offset
< (playpos
+ len
))) {
265 TRACE("signalled event %p (%d)\n", event
->hEventNotify
, i
);
266 SetEvent(event
->hEventNotify
);
273 * Copy a single frame from the given input buffer to the given output buffer.
274 * Translate 8 <-> 16 bits and mono <-> stereo
276 static inline void cp_fields(const IDirectSoundBufferImpl
*dsb
, const BYTE
*ibuf
, BYTE
*obuf
,
277 UINT istride
, UINT ostride
, UINT count
, UINT freqAcc
, UINT adj
)
279 DirectSoundDevice
*device
= dsb
->device
;
280 INT istep
= dsb
->pwfx
->wBitsPerSample
/ 8, ostep
= device
->pwfx
->wBitsPerSample
/ 8;
282 if (device
->pwfx
->nChannels
== dsb
->pwfx
->nChannels
||
283 (device
->pwfx
->nChannels
== 2 && dsb
->pwfx
->nChannels
== 6) ||
284 (device
->pwfx
->nChannels
== 8 && dsb
->pwfx
->nChannels
== 2) ||
285 (device
->pwfx
->nChannels
== 6 && dsb
->pwfx
->nChannels
== 2)) {
286 dsb
->convert(ibuf
, obuf
, istride
, ostride
, count
, freqAcc
, adj
);
287 if (device
->pwfx
->nChannels
== 2 || dsb
->pwfx
->nChannels
== 2)
288 dsb
->convert(ibuf
+ istep
, obuf
+ ostep
, istride
, ostride
, count
, freqAcc
, adj
);
292 if (device
->pwfx
->nChannels
== 1 && dsb
->pwfx
->nChannels
== 2)
294 dsb
->convert(ibuf
, obuf
, istride
, ostride
, count
, freqAcc
, adj
);
298 if (device
->pwfx
->nChannels
== 2 && dsb
->pwfx
->nChannels
== 1)
300 dsb
->convert(ibuf
, obuf
, istride
, ostride
, count
, freqAcc
, adj
);
301 dsb
->convert(ibuf
, obuf
+ ostep
, istride
, ostride
, count
, freqAcc
, adj
);
305 WARN("Unable to remap channels: device=%u, buffer=%u\n", device
->pwfx
->nChannels
,
306 dsb
->pwfx
->nChannels
);
310 * Calculate the distance between two buffer offsets, taking wraparound
313 static inline DWORD
DSOUND_BufPtrDiff(DWORD buflen
, DWORD ptr1
, DWORD ptr2
)
315 /* If these asserts fail, the problem is not here, but in the underlying code */
316 assert(ptr1
< buflen
);
317 assert(ptr2
< buflen
);
321 return buflen
+ ptr1
- ptr2
;
325 * Mix at most the given amount of data into the allocated temporary buffer
326 * of the given secondary buffer, starting from the dsb's first currently
327 * unsampled frame (writepos), translating frequency (pitch), stereo/mono
328 * and bits-per-sample so that it is ideal for the primary buffer.
329 * Doesn't perform any mixing - this is a straight copy/convert operation.
331 * dsb = the secondary buffer
332 * writepos = Starting position of changed buffer
333 * len = number of bytes to resample from writepos
335 * NOTE: writepos + len <= buflen. When called by mixer, MixOne makes sure of this.
337 static void DSOUND_MixToTemporary(const IDirectSoundBufferImpl
*dsb
, DWORD writepos
, DWORD len
)
340 BYTE
*ibp
, *obp
, *obp_begin
;
341 INT iAdvance
= dsb
->pwfx
->nBlockAlign
;
342 INT oAdvance
= dsb
->device
->pwfx
->nBlockAlign
;
343 DWORD freqAcc
, overshot
, maxlen
;
345 assert(writepos
+ len
<= dsb
->buflen
);
346 if (writepos
+ len
< dsb
->buflen
)
347 len
+= dsb
->pwfx
->nBlockAlign
;
349 maxlen
= DSOUND_secpos_to_bufpos(dsb
, len
, 0, NULL
);
351 ibp
= dsb
->buffer
->memory
+ writepos
;
352 if (dsb
->device
->tmp_buffer_len
< maxlen
|| !dsb
->device
->tmp_buffer
)
354 dsb
->device
->tmp_buffer_len
= maxlen
;
355 if (dsb
->device
->tmp_buffer
)
356 dsb
->device
->tmp_buffer
= HeapReAlloc(GetProcessHeap(), 0, dsb
->device
->tmp_buffer
, maxlen
);
358 dsb
->device
->tmp_buffer
= HeapAlloc(GetProcessHeap(), 0, maxlen
);
359 obp_begin
= dsb
->device
->tmp_buffer
;
362 obp_begin
= dsb
->device
->tmp_buffer
;
364 TRACE("(%p, %p)\n", dsb
, ibp
);
365 size
= len
/ iAdvance
;
367 /* Check for same sample rate */
368 if (dsb
->freq
== dsb
->device
->pwfx
->nSamplesPerSec
) {
369 TRACE("(%p) Same sample rate %d = primary %d\n", dsb
,
370 dsb
->freq
, dsb
->device
->pwfx
->nSamplesPerSec
);
373 cp_fields(dsb
, ibp
, obp
, iAdvance
, oAdvance
, size
, 0, 1 << DSOUND_FREQSHIFT
);
377 /* Mix in different sample rates */
378 TRACE("(%p) Adjusting frequency: %d -> %d\n", dsb
, dsb
->freq
, dsb
->device
->pwfx
->nSamplesPerSec
);
380 DSOUND_secpos_to_bufpos(dsb
, writepos
, dsb
->sec_mixpos
, &freqAcc
);
381 overshot
= freqAcc
>> DSOUND_FREQSHIFT
;
384 if (overshot
>= size
)
387 writepos
+= overshot
* iAdvance
;
388 if (writepos
>= dsb
->buflen
)
390 ibp
= dsb
->buffer
->memory
+ writepos
;
391 freqAcc
&= (1 << DSOUND_FREQSHIFT
) - 1;
392 TRACE("Overshot: %d, freqAcc: %04x\n", overshot
, freqAcc
);
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
)
410 INT nChannels
= dsb
->device
->pwfx
->nChannels
;
411 LPBYTE mem
= dsb
->device
->tmp_buffer
;
413 TRACE("(%p,%d)\n",dsb
,len
);
414 TRACE("left = %x, right = %x\n", dsb
->volpan
.dwTotalLeftAmpFactor
,
415 dsb
->volpan
.dwTotalRightAmpFactor
);
417 if ((!(dsb
->dsbd
.dwFlags
& DSBCAPS_CTRLPAN
) || (dsb
->volpan
.lPan
== 0)) &&
418 (!(dsb
->dsbd
.dwFlags
& DSBCAPS_CTRLVOLUME
) || (dsb
->volpan
.lVolume
== 0)) &&
419 !(dsb
->dsbd
.dwFlags
& DSBCAPS_CTRL3D
))
420 return NULL
; /* Nothing to do */
422 if (nChannels
!= 1 && nChannels
!= 2)
424 FIXME("There is no support for %d channels\n", nChannels
);
428 if (dsb
->device
->pwfx
->wBitsPerSample
!= 8 && dsb
->device
->pwfx
->wBitsPerSample
!= 16)
430 FIXME("There is no support for %d bpp\n", dsb
->device
->pwfx
->wBitsPerSample
);
434 assert(dsb
->device
->tmp_buffer_len
>= len
&& dsb
->device
->tmp_buffer
);
436 bpc
= dsb
->device
->tmp_buffer
;
439 vLeft
= dsb
->volpan
.dwTotalLeftAmpFactor
;
441 vRight
= dsb
->volpan
.dwTotalRightAmpFactor
;
445 switch (dsb
->device
->pwfx
->wBitsPerSample
) {
447 /* 8-bit WAV is unsigned, but we need to operate */
448 /* on signed data for this to work properly */
449 for (i
= 0; i
< len
-1; i
+=2) {
450 *(bpc
++) = (((*(mem
++) - 128) * vLeft
) >> 16) + 128;
451 *(bpc
++) = (((*(mem
++) - 128) * vRight
) >> 16) + 128;
453 if (len
% 2 == 1 && nChannels
== 1)
454 *(bpc
++) = (((*(mem
++) - 128) * vLeft
) >> 16) + 128;
457 /* 16-bit WAV is signed -- much better */
458 for (i
= 0; i
< len
-3; i
+= 4) {
459 *(bps
++) = (*(mems
++) * vLeft
) >> 16;
460 *(bps
++) = (*(mems
++) * vRight
) >> 16;
462 if (len
% 4 == 2 && nChannels
== 1)
463 *(bps
++) = ((INT
)*(mems
++) * vLeft
) >> 16;
466 return dsb
->device
->tmp_buffer
;
470 * Mix (at most) the given number of bytes into the given position of the
471 * device buffer, from the secondary buffer "dsb" (starting at the current
472 * mix position for that buffer).
474 * Returns the number of bytes actually mixed into the device buffer. This
475 * will match fraglen unless the end of the secondary buffer is reached
476 * (and it is not looping).
478 * dsb = the secondary buffer to mix from
479 * writepos = position (offset) in device buffer to write at
480 * fraglen = number of bytes to mix
482 static DWORD
DSOUND_MixInBuffer(IDirectSoundBufferImpl
*dsb
, DWORD writepos
, DWORD fraglen
)
484 INT len
= fraglen
, ilen
;
486 DWORD oldpos
, mixbufpos
;
488 TRACE("buf_mixpos=%d/%d sec_mixpos=%d/%d\n", dsb
->buf_mixpos
, dsb
->tmp_buffer_len
, dsb
->sec_mixpos
, dsb
->buflen
);
489 TRACE("(%p,%d,%d)\n",dsb
,writepos
,fraglen
);
491 assert(dsb
->buf_mixpos
+ len
<= dsb
->tmp_buffer_len
);
493 if (len
% dsb
->device
->pwfx
->nBlockAlign
) {
494 INT nBlockAlign
= dsb
->device
->pwfx
->nBlockAlign
;
495 ERR("length not a multiple of block size, len = %d, block size = %d\n", len
, nBlockAlign
);
496 len
-= len
% nBlockAlign
; /* data alignment */
499 /* Resample buffer to temporary buffer specifically allocated for this purpose, if needed */
500 DSOUND_MixToTemporary(dsb
, dsb
->sec_mixpos
, DSOUND_bufpos_to_secpos(dsb
, dsb
->buf_mixpos
+len
) - dsb
->sec_mixpos
);
501 ibuf
= dsb
->device
->tmp_buffer
;
503 /* Apply volume if needed */
504 volbuf
= DSOUND_MixerVol(dsb
, len
);
508 mixbufpos
= DSOUND_bufpos_to_mixpos(dsb
->device
, writepos
);
509 /* Now mix the temporary buffer into the devices main buffer */
510 if ((writepos
+ len
) <= dsb
->device
->buflen
)
511 dsb
->device
->mixfunction(ibuf
, dsb
->device
->mix_buffer
+ mixbufpos
, len
);
514 DWORD todo
= dsb
->device
->buflen
- writepos
;
515 dsb
->device
->mixfunction(ibuf
, dsb
->device
->mix_buffer
+ mixbufpos
, todo
);
516 dsb
->device
->mixfunction(ibuf
+ todo
, dsb
->device
->mix_buffer
, len
- todo
);
519 oldpos
= dsb
->sec_mixpos
;
520 dsb
->buf_mixpos
+= len
;
522 if (dsb
->buf_mixpos
>= dsb
->tmp_buffer_len
) {
523 if (dsb
->playflags
& DSBPLAY_LOOPING
) {
524 dsb
->buf_mixpos
-= dsb
->tmp_buffer_len
;
526 dsb
->buf_mixpos
= dsb
->sec_mixpos
= 0;
527 dsb
->state
= STATE_STOPPED
;
529 DSOUND_RecalcFreqAcc(dsb
);
532 dsb
->sec_mixpos
= DSOUND_bufpos_to_secpos(dsb
, dsb
->buf_mixpos
);
533 ilen
= DSOUND_BufPtrDiff(dsb
->buflen
, dsb
->sec_mixpos
, oldpos
);
534 /* check for notification positions */
535 if (dsb
->dsbd
.dwFlags
& DSBCAPS_CTRLPOSITIONNOTIFY
&&
536 dsb
->state
!= STATE_STARTING
) {
537 DSOUND_CheckEvent(dsb
, oldpos
, ilen
);
540 /* increase mix position */
541 dsb
->primary_mixpos
+= len
;
542 if (dsb
->primary_mixpos
>= dsb
->device
->buflen
)
543 dsb
->primary_mixpos
-= dsb
->device
->buflen
;
548 * Mix some frames from the given secondary buffer "dsb" into the device
551 * dsb = the secondary buffer
552 * playpos = the current play position in the device buffer (primary buffer)
553 * writepos = the current safe-to-write position in the device buffer
554 * mixlen = the maximum number of bytes in the primary buffer to mix, from the
557 * Returns: the number of bytes beyond the writepos that were mixed.
559 static DWORD
DSOUND_MixOne(IDirectSoundBufferImpl
*dsb
, DWORD writepos
, DWORD mixlen
)
561 /* The buffer's primary_mixpos may be before or after the device
562 * buffer's mixpos, but both must be ahead of writepos. */
565 TRACE("(%p,%d,%d)\n",dsb
,writepos
,mixlen
);
566 TRACE("writepos=%d, buf_mixpos=%d, primary_mixpos=%d, mixlen=%d\n", writepos
, dsb
->buf_mixpos
, dsb
->primary_mixpos
, mixlen
);
567 TRACE("looping=%d, leadin=%d, buflen=%d\n", dsb
->playflags
, dsb
->leadin
, dsb
->tmp_buffer_len
);
569 /* If leading in, only mix about 20 ms, and 'skip' mixing the rest, for more fluid pointer advancement */
570 if (dsb
->leadin
&& dsb
->state
== STATE_STARTING
)
572 if (mixlen
> 2 * dsb
->device
->fraglen
)
574 dsb
->primary_mixpos
+= mixlen
- 2 * dsb
->device
->fraglen
;
575 dsb
->primary_mixpos
%= dsb
->device
->buflen
;
580 /* calculate how much pre-buffering has already been done for this buffer */
581 primary_done
= DSOUND_BufPtrDiff(dsb
->device
->buflen
, dsb
->primary_mixpos
, writepos
);
584 if(mixlen
< primary_done
)
586 /* Should *NEVER* happen */
587 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
);
588 dsb
->primary_mixpos
= writepos
+ mixlen
;
589 dsb
->primary_mixpos
%= dsb
->device
->buflen
;
593 /* take into account already mixed data */
594 mixlen
-= primary_done
;
596 TRACE("primary_done=%d, mixlen (primary) = %i\n", primary_done
, mixlen
);
601 /* First try to mix to the end of the buffer if possible
602 * Theoretically it would allow for better optimization
604 if (mixlen
+ dsb
->buf_mixpos
>= dsb
->tmp_buffer_len
)
606 DWORD newmixed
, mixfirst
= dsb
->tmp_buffer_len
- dsb
->buf_mixpos
;
607 newmixed
= DSOUND_MixInBuffer(dsb
, dsb
->primary_mixpos
, mixfirst
);
610 if (dsb
->playflags
& DSBPLAY_LOOPING
)
611 while (newmixed
&& mixlen
)
613 mixfirst
= (dsb
->tmp_buffer_len
< mixlen
? dsb
->tmp_buffer_len
: mixlen
);
614 newmixed
= DSOUND_MixInBuffer(dsb
, dsb
->primary_mixpos
, mixfirst
);
618 else DSOUND_MixInBuffer(dsb
, dsb
->primary_mixpos
, mixlen
);
620 /* re-calculate the primary done */
621 primary_done
= DSOUND_BufPtrDiff(dsb
->device
->buflen
, dsb
->primary_mixpos
, writepos
);
623 TRACE("new primary_mixpos=%d, total mixed data=%d\n", dsb
->primary_mixpos
, primary_done
);
625 /* Report back the total prebuffered amount for this buffer */
630 * For a DirectSoundDevice, go through all the currently playing buffers and
631 * mix them in to the device buffer.
633 * writepos = the current safe-to-write position in the primary buffer
634 * mixlen = the maximum amount to mix into the primary buffer
635 * (beyond the current writepos)
636 * recover = true if the sound device may have been reset and the write
637 * position in the device buffer changed
638 * all_stopped = reports back if all buffers have stopped
640 * Returns: the length beyond the writepos that was mixed to.
643 static DWORD
DSOUND_MixToPrimary(const DirectSoundDevice
*device
, DWORD writepos
, DWORD mixlen
, BOOL recover
, BOOL
*all_stopped
)
647 IDirectSoundBufferImpl
*dsb
;
649 /* unless we find a running buffer, all have stopped */
652 TRACE("(%d,%d,%d)\n", writepos
, mixlen
, recover
);
653 for (i
= 0; i
< device
->nrofbuffers
; i
++) {
654 dsb
= device
->buffers
[i
];
656 TRACE("MixToPrimary for %p, state=%d\n", dsb
, dsb
->state
);
658 if (dsb
->buflen
&& dsb
->state
) {
659 TRACE("Checking %p, mixlen=%d\n", dsb
, mixlen
);
660 RtlAcquireResourceShared(&dsb
->lock
, TRUE
);
661 /* if buffer is stopping it is stopped now */
662 if (dsb
->state
== STATE_STOPPING
) {
663 dsb
->state
= STATE_STOPPED
;
664 DSOUND_CheckEvent(dsb
, 0, 0);
665 } else if (dsb
->state
!= STATE_STOPPED
) {
667 /* if recovering, reset the mix position */
668 if ((dsb
->state
== STATE_STARTING
) || recover
) {
669 dsb
->primary_mixpos
= writepos
;
672 /* if the buffer was starting, it must be playing now */
673 if (dsb
->state
== STATE_STARTING
)
674 dsb
->state
= STATE_PLAYING
;
676 /* mix next buffer into the main buffer */
677 len
= DSOUND_MixOne(dsb
, writepos
, mixlen
);
679 if (!minlen
) minlen
= len
;
681 /* record the minimum length mixed from all buffers */
682 /* we only want to return the length which *all* buffers have mixed */
683 else if (len
) minlen
= (len
< minlen
) ? len
: minlen
;
685 *all_stopped
= FALSE
;
687 RtlReleaseResource(&dsb
->lock
);
691 TRACE("Mixed at least %d from all buffers\n", minlen
);
696 * Add buffers to the emulated wave device system.
698 * device = The current dsound playback device
699 * force = If TRUE, the function will buffer up as many frags as possible,
700 * even though and will ignore the actual state of the primary buffer.
705 static void DSOUND_WaveQueue(DirectSoundDevice
*device
, BOOL force
)
707 DWORD prebuf_frames
, buf_offs_bytes
, wave_fragpos
;
712 TRACE("(%p)\n", device
);
714 /* calculate the current wave frag position */
715 wave_fragpos
= (device
->pwplay
+ device
->pwqueue
) % device
->helfrags
;
717 /* calculate the current wave write position */
718 buf_offs_bytes
= wave_fragpos
* device
->fraglen
;
720 TRACE("wave_fragpos = %i, buf_offs_bytes = %i, pwqueue = %i, prebuf = %i\n",
721 wave_fragpos
, buf_offs_bytes
, device
->pwqueue
, device
->prebuf
);
725 /* check remaining prebuffered frags */
726 prebuf_frags
= device
->mixpos
/ device
->fraglen
;
727 if (prebuf_frags
== device
->helfrags
)
729 TRACE("wave_fragpos = %d, mixpos_frags = %d\n", wave_fragpos
, prebuf_frags
);
730 if (prebuf_frags
< wave_fragpos
)
731 prebuf_frags
+= device
->helfrags
;
732 prebuf_frags
-= wave_fragpos
;
733 TRACE("wanted prebuf_frags = %d\n", prebuf_frags
);
736 /* buffer the maximum amount of frags */
737 prebuf_frags
= device
->prebuf
;
739 /* limit to the queue we have left */
740 if ((prebuf_frags
+ device
->pwqueue
) > device
->prebuf
)
741 prebuf_frags
= device
->prebuf
- device
->pwqueue
;
743 TRACE("prebuf_frags = %i\n", prebuf_frags
);
749 device
->pwqueue
+= prebuf_frags
;
751 prebuf_frames
= ((prebuf_frags
+ wave_fragpos
> device
->helfrags
) ?
752 (device
->helfrags
- wave_fragpos
) :
753 (prebuf_frags
)) * device
->fraglen
/ device
->pwfx
->nBlockAlign
;
755 hr
= IAudioRenderClient_GetBuffer(device
->render
, prebuf_frames
, &buffer
);
757 WARN("GetBuffer failed: %08x\n", hr
);
761 memcpy(buffer
, device
->buffer
+ buf_offs_bytes
,
762 prebuf_frames
* device
->pwfx
->nBlockAlign
);
764 hr
= IAudioRenderClient_ReleaseBuffer(device
->render
, prebuf_frames
, 0);
766 WARN("ReleaseBuffer failed: %08x\n", hr
);
770 /* check if anything wrapped */
771 prebuf_frags
= prebuf_frags
+ wave_fragpos
- device
->helfrags
;
772 if(prebuf_frags
> 0){
773 prebuf_frames
= prebuf_frags
* device
->fraglen
/ device
->pwfx
->nBlockAlign
;
775 hr
= IAudioRenderClient_GetBuffer(device
->render
, prebuf_frames
, &buffer
);
777 WARN("GetBuffer failed: %08x\n", hr
);
781 memcpy(buffer
, device
->buffer
, prebuf_frames
* device
->pwfx
->nBlockAlign
);
783 hr
= IAudioRenderClient_ReleaseBuffer(device
->render
, prebuf_frames
, 0);
785 WARN("ReleaseBuffer failed: %08x\n", hr
);
790 TRACE("queue now = %i\n", device
->pwqueue
);
794 * Perform mixing for a Direct Sound device. That is, go through all the
795 * secondary buffers (the sound bites currently playing) and mix them in
796 * to the primary buffer (the device buffer).
798 static void DSOUND_PerformMix(DirectSoundDevice
*device
)
800 UINT64 clock_pos
, clock_freq
, pos_bytes
;
804 TRACE("(%p)\n", device
);
807 EnterCriticalSection(&device
->mixlock
);
809 hr
= IAudioClock_GetFrequency(device
->clock
, &clock_freq
);
811 WARN("GetFrequency failed: %08x\n", hr
);
812 LeaveCriticalSection(&device
->mixlock
);
816 hr
= IAudioClock_GetPosition(device
->clock
, &clock_pos
, NULL
);
818 WARN("GetCurrentPadding failed: %08x\n", hr
);
819 LeaveCriticalSection(&device
->mixlock
);
823 pos_bytes
= (clock_pos
* device
->pwfx
->nSamplesPerSec
* device
->pwfx
->nBlockAlign
) / clock_freq
;
825 delta_frags
= (pos_bytes
- device
->last_pos_bytes
) / device
->fraglen
;
827 device
->pwplay
+= delta_frags
;
828 device
->pwplay
%= device
->helfrags
;
829 device
->pwqueue
-= delta_frags
;
830 device
->last_pos_bytes
= pos_bytes
- (pos_bytes
% device
->fraglen
);
833 if (device
->priolevel
!= DSSCL_WRITEPRIMARY
) {
834 BOOL recover
= FALSE
, all_stopped
= FALSE
;
835 DWORD playpos
, writepos
, writelead
, maxq
, frag
, prebuff_max
, prebuff_left
, size1
, size2
, mixplaypos
, mixplaypos2
;
839 /* the sound of silence */
840 nfiller
= device
->pwfx
->wBitsPerSample
== 8 ? 128 : 0;
842 /* get the position in the primary buffer */
843 if (DSOUND_PrimaryGetPosition(device
, &playpos
, &writepos
) != 0){
844 LeaveCriticalSection(&(device
->mixlock
));
848 TRACE("primary playpos=%d, writepos=%d, clrpos=%d, mixpos=%d, buflen=%d\n",
849 playpos
,writepos
,device
->playpos
,device
->mixpos
,device
->buflen
);
850 assert(device
->playpos
< device
->buflen
);
852 mixplaypos
= DSOUND_bufpos_to_mixpos(device
, device
->playpos
);
853 mixplaypos2
= DSOUND_bufpos_to_mixpos(device
, playpos
);
855 /* calc maximum prebuff */
856 prebuff_max
= (device
->prebuf
* device
->fraglen
);
857 if (playpos
+ prebuff_max
>= device
->helfrags
* device
->fraglen
)
858 prebuff_max
+= device
->buflen
- device
->helfrags
* device
->fraglen
;
860 /* check how close we are to an underrun. It occurs when the writepos overtakes the mixpos */
861 prebuff_left
= DSOUND_BufPtrDiff(device
->buflen
, device
->mixpos
, playpos
);
862 writelead
= DSOUND_BufPtrDiff(device
->buflen
, writepos
, playpos
);
864 /* check for underrun. underrun occurs when the write position passes the mix position
865 * also wipe out just-played sound data */
866 if((prebuff_left
> prebuff_max
) || (device
->state
== STATE_STOPPED
) || (device
->state
== STATE_STARTING
)){
867 if (device
->state
== STATE_STOPPING
|| device
->state
== STATE_PLAYING
)
868 WARN("Probable buffer underrun\n");
869 else TRACE("Buffer starting or buffer underrun\n");
871 /* recover mixing for all buffers */
874 /* reset mix position to write position */
875 device
->mixpos
= writepos
;
877 ZeroMemory(device
->mix_buffer
, device
->mix_buffer_len
);
878 ZeroMemory(device
->buffer
, device
->buflen
);
879 } else if (playpos
< device
->playpos
) {
880 buf1
= device
->buffer
+ device
->playpos
;
881 buf2
= device
->buffer
;
882 size1
= device
->buflen
- device
->playpos
;
884 FillMemory(device
->mix_buffer
+ mixplaypos
, device
->mix_buffer_len
- mixplaypos
, 0);
885 FillMemory(device
->mix_buffer
, mixplaypos2
, 0);
886 FillMemory(buf1
, size1
, nfiller
);
887 if (playpos
&& (!buf2
|| !size2
))
888 FIXME("%d: (%d, %d)=>(%d, %d) There should be an additional buffer here!!\n", __LINE__
, device
->playpos
, device
->mixpos
, playpos
, writepos
);
889 FillMemory(buf2
, size2
, nfiller
);
891 buf1
= device
->buffer
+ device
->playpos
;
893 size1
= playpos
- device
->playpos
;
895 FillMemory(device
->mix_buffer
+ mixplaypos
, mixplaypos2
- mixplaypos
, 0);
896 FillMemory(buf1
, size1
, nfiller
);
898 device
->playpos
= playpos
;
900 /* find the maximum we can prebuffer from current write position */
901 maxq
= (writelead
< prebuff_max
) ? (prebuff_max
- writelead
) : 0;
903 TRACE("prebuff_left = %d, prebuff_max = %dx%d=%d, writelead=%d\n",
904 prebuff_left
, device
->prebuf
, device
->fraglen
, prebuff_max
, writelead
);
907 frag
= DSOUND_MixToPrimary(device
, writepos
, maxq
, recover
, &all_stopped
);
909 if (frag
+ writepos
> device
->buflen
)
911 DWORD todo
= device
->buflen
- writepos
;
912 device
->normfunction(device
->mix_buffer
+ DSOUND_bufpos_to_mixpos(device
, writepos
), device
->buffer
+ writepos
, todo
);
913 device
->normfunction(device
->mix_buffer
, device
->buffer
, frag
- todo
);
916 device
->normfunction(device
->mix_buffer
+ DSOUND_bufpos_to_mixpos(device
, writepos
), device
->buffer
+ writepos
, frag
);
918 /* update the mix position, taking wrap-around into account */
919 device
->mixpos
= writepos
+ frag
;
920 device
->mixpos
%= device
->buflen
;
922 /* update prebuff left */
923 prebuff_left
= DSOUND_BufPtrDiff(device
->buflen
, device
->mixpos
, playpos
);
925 /* check if have a whole fragment */
926 if (prebuff_left
>= device
->fraglen
){
928 /* update the wave queue */
929 DSOUND_WaveQueue(device
, FALSE
);
931 /* buffers are full. start playing if applicable */
932 if(device
->state
== STATE_STARTING
){
933 TRACE("started primary buffer\n");
934 if(DSOUND_PrimaryPlay(device
) != DS_OK
){
935 WARN("DSOUND_PrimaryPlay failed\n");
938 /* we are playing now */
939 device
->state
= STATE_PLAYING
;
943 /* buffers are full. start stopping if applicable */
944 if(device
->state
== STATE_STOPPED
){
945 TRACE("restarting primary buffer\n");
946 if(DSOUND_PrimaryPlay(device
) != DS_OK
){
947 WARN("DSOUND_PrimaryPlay failed\n");
950 /* start stopping again. as soon as there is no more data, it will stop */
951 device
->state
= STATE_STOPPING
;
956 /* if device was stopping, its for sure stopped when all buffers have stopped */
957 else if((all_stopped
== TRUE
) && (device
->state
== STATE_STOPPING
)){
958 TRACE("All buffers have stopped. Stopping primary buffer\n");
959 device
->state
= STATE_STOPPED
;
961 /* stop the primary buffer now */
962 DSOUND_PrimaryStop(device
);
967 DSOUND_WaveQueue(device
, TRUE
);
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 RtlAcquireResourceShared(&(device
->buffer_list_lock
), TRUE
);
1000 DSOUND_PerformMix(device
);
1002 RtlReleaseResource(&(device
->buffer_list_lock
));
1004 end_time
= GetTickCount();
1005 TRACE("completed processing at %d, duration = %d\n", end_time
, end_time
- start_time
);