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
35 #include "wine/debug.h"
38 #include "dsound_private.h"
40 WINE_DEFAULT_DEBUG_CHANNEL(dsound
);
42 void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan
)
45 TRACE("(%p)\n",volpan
);
47 TRACE("Vol=%d Pan=%d\n", volpan
->lVolume
, volpan
->lPan
);
48 /* the AmpFactors are expressed in 16.16 fixed point */
49 volpan
->dwVolAmpFactor
= (ULONG
) (pow(2.0, volpan
->lVolume
/ 600.0) * 0xffff);
50 /* FIXME: dwPan{Left|Right}AmpFactor */
52 /* FIXME: use calculated vol and pan ampfactors */
53 temp
= (double) (volpan
->lVolume
- (volpan
->lPan
> 0 ? volpan
->lPan
: 0));
54 volpan
->dwTotalLeftAmpFactor
= (ULONG
) (pow(2.0, temp
/ 600.0) * 0xffff);
55 temp
= (double) (volpan
->lVolume
+ (volpan
->lPan
< 0 ? volpan
->lPan
: 0));
56 volpan
->dwTotalRightAmpFactor
= (ULONG
) (pow(2.0, temp
/ 600.0) * 0xffff);
58 TRACE("left = %x, right = %x\n", volpan
->dwTotalLeftAmpFactor
, volpan
->dwTotalRightAmpFactor
);
61 void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan
)
64 TRACE("(%p)\n",volpan
);
66 TRACE("left=%x, right=%x\n",volpan
->dwTotalLeftAmpFactor
,volpan
->dwTotalRightAmpFactor
);
67 if (volpan
->dwTotalLeftAmpFactor
==0)
70 left
=600 * log(((double)volpan
->dwTotalLeftAmpFactor
) / 0xffff) / log(2);
71 if (volpan
->dwTotalRightAmpFactor
==0)
74 right
=600 * log(((double)volpan
->dwTotalRightAmpFactor
) / 0xffff) / log(2);
77 volpan
->lVolume
=right
;
78 volpan
->dwVolAmpFactor
=volpan
->dwTotalRightAmpFactor
;
83 volpan
->dwVolAmpFactor
=volpan
->dwTotalLeftAmpFactor
;
85 if (volpan
->lVolume
< -10000)
86 volpan
->lVolume
=-10000;
87 volpan
->lPan
=right
-left
;
88 if (volpan
->lPan
< -10000)
91 TRACE("Vol=%d Pan=%d\n", volpan
->lVolume
, volpan
->lPan
);
94 /* NOTE: Not all secpos have to always be mapped to a bufpos, other way around is always the case
95 * DWORD64 is used here because a single DWORD wouldn't be big enough to fit the freqAcc for big buffers
97 /** This function converts a 'native' sample pointer to a resampled pointer that fits for primary
98 * secmixpos is used to decide which freqAcc is needed
99 * overshot tells what the 'actual' secpos is now (optional)
101 DWORD
DSOUND_secpos_to_bufpos(const IDirectSoundBufferImpl
*dsb
, DWORD secpos
, DWORD secmixpos
, DWORD
* overshot
)
103 DWORD64 framelen
= secpos
/ dsb
->pwfx
->nBlockAlign
;
104 DWORD64 freqAdjust
= dsb
->freqAdjust
;
105 DWORD64 acc
, freqAcc
;
107 if (secpos
< secmixpos
)
108 freqAcc
= dsb
->freqAccNext
;
109 else freqAcc
= dsb
->freqAcc
;
110 acc
= (framelen
<< DSOUND_FREQSHIFT
) + (freqAdjust
- 1 - freqAcc
);
114 DWORD64 oshot
= acc
* freqAdjust
+ freqAcc
;
115 assert(oshot
>= framelen
<< DSOUND_FREQSHIFT
);
116 oshot
-= framelen
<< DSOUND_FREQSHIFT
;
117 *overshot
= (DWORD
)oshot
;
118 assert(*overshot
< dsb
->freqAdjust
);
120 return (DWORD
)acc
* dsb
->device
->pwfx
->nBlockAlign
;
123 /** Convert a resampled pointer that fits for primary to a 'native' sample pointer
124 * freqAccNext is used here rather then freqAcc: In case the app wants to fill up to
125 * the play position it won't overwrite it
127 static DWORD
DSOUND_bufpos_to_secpos(const IDirectSoundBufferImpl
*dsb
, DWORD bufpos
)
129 DWORD oAdv
= dsb
->device
->pwfx
->nBlockAlign
, iAdv
= dsb
->pwfx
->nBlockAlign
, pos
;
133 framelen
= bufpos
/oAdv
;
134 acc
= framelen
* (DWORD64
)dsb
->freqAdjust
+ (DWORD64
)dsb
->freqAccNext
;
135 acc
= acc
>> DSOUND_FREQSHIFT
;
136 pos
= (DWORD
)acc
* iAdv
;
137 if (pos
>= dsb
->buflen
)
138 /* Because of differences between freqAcc and freqAccNext, this might happen */
139 pos
= dsb
->buflen
- iAdv
;
140 TRACE("Converted %d/%d to %d/%d\n", bufpos
, dsb
->tmp_buffer_len
, pos
, dsb
->buflen
);
145 * Move freqAccNext to freqAcc, and find new values for buffer length and freqAccNext
147 static void DSOUND_RecalcFreqAcc(IDirectSoundBufferImpl
*dsb
)
149 if (!dsb
->freqneeded
) return;
150 dsb
->freqAcc
= dsb
->freqAccNext
;
151 dsb
->tmp_buffer_len
= DSOUND_secpos_to_bufpos(dsb
, dsb
->buflen
, 0, &dsb
->freqAccNext
);
152 TRACE("New freqadjust: %04x, new buflen: %d\n", dsb
->freqAccNext
, dsb
->tmp_buffer_len
);
156 * Recalculate the size for temporary buffer, and new writelead
157 * Should be called when one of the following things occur:
158 * - Primary buffer format is changed
159 * - This buffer format (frequency) is changed
161 * After this, DSOUND_MixToTemporary(dsb, 0, dsb->buflen) should
162 * be called to refill the temporary buffer with data.
164 void DSOUND_RecalcFormat(IDirectSoundBufferImpl
*dsb
)
166 BOOL needremix
= TRUE
, needresample
= (dsb
->freq
!= dsb
->device
->pwfx
->nSamplesPerSec
);
167 DWORD bAlign
= dsb
->pwfx
->nBlockAlign
, pAlign
= dsb
->device
->pwfx
->nBlockAlign
;
171 /* calculate the 10ms write lead */
172 dsb
->writelead
= (dsb
->freq
/ 100) * dsb
->pwfx
->nBlockAlign
;
174 if ((dsb
->pwfx
->wBitsPerSample
== dsb
->device
->pwfx
->wBitsPerSample
) &&
175 (dsb
->pwfx
->nChannels
== dsb
->device
->pwfx
->nChannels
) && !needresample
)
177 HeapFree(GetProcessHeap(), 0, dsb
->tmp_buffer
);
178 dsb
->tmp_buffer
= NULL
;
179 dsb
->max_buffer_len
= dsb
->freqAcc
= dsb
->freqAccNext
= 0;
180 dsb
->freqneeded
= needresample
;
185 DSOUND_RecalcFreqAcc(dsb
);
187 dsb
->tmp_buffer_len
= dsb
->buflen
/ bAlign
* pAlign
;
188 dsb
->max_buffer_len
= dsb
->tmp_buffer_len
;
189 dsb
->tmp_buffer
= HeapAlloc(GetProcessHeap(), 0, dsb
->max_buffer_len
);
190 FillMemory(dsb
->tmp_buffer
, dsb
->tmp_buffer_len
, dsb
->device
->pwfx
->wBitsPerSample
== 8 ? 128 : 0);
192 else dsb
->max_buffer_len
= dsb
->tmp_buffer_len
= dsb
->buflen
;
193 dsb
->buf_mixpos
= DSOUND_secpos_to_bufpos(dsb
, dsb
->sec_mixpos
, 0, NULL
);
197 * Check for application callback requests for when the play position
198 * reaches certain points.
200 * The offsets that will be triggered will be those between the recorded
201 * "last played" position for the buffer (i.e. dsb->playpos) and "len" bytes
202 * beyond that position.
204 void DSOUND_CheckEvent(const IDirectSoundBufferImpl
*dsb
, DWORD playpos
, int len
)
208 LPDSBPOSITIONNOTIFY event
;
209 TRACE("(%p,%d)\n",dsb
,len
);
211 if (dsb
->nrofnotifies
== 0)
214 TRACE("(%p) buflen = %d, playpos = %d, len = %d\n",
215 dsb
, dsb
->buflen
, playpos
, len
);
216 for (i
= 0; i
< dsb
->nrofnotifies
; i
++) {
217 event
= dsb
->notifies
+ i
;
218 offset
= event
->dwOffset
;
219 TRACE("checking %d, position %d, event = %p\n",
220 i
, offset
, event
->hEventNotify
);
221 /* DSBPN_OFFSETSTOP has to be the last element. So this is */
222 /* OK. [Inside DirectX, p274] */
224 /* This also means we can't sort the entries by offset, */
225 /* because DSBPN_OFFSETSTOP == -1 */
226 if (offset
== DSBPN_OFFSETSTOP
) {
227 if (dsb
->state
== STATE_STOPPED
) {
228 SetEvent(event
->hEventNotify
);
229 TRACE("signalled event %p (%d)\n", event
->hEventNotify
, i
);
234 if ((playpos
+ len
) >= dsb
->buflen
) {
235 if ((offset
< ((playpos
+ len
) % dsb
->buflen
)) ||
236 (offset
>= playpos
)) {
237 TRACE("signalled event %p (%d)\n", event
->hEventNotify
, i
);
238 SetEvent(event
->hEventNotify
);
241 if ((offset
>= playpos
) && (offset
< (playpos
+ len
))) {
242 TRACE("signalled event %p (%d)\n", event
->hEventNotify
, i
);
243 SetEvent(event
->hEventNotify
);
249 /* WAV format info can be found at:
251 * http://www.cwi.nl/ftp/audio/AudioFormats.part2
252 * ftp://ftp.cwi.nl/pub/audio/RIFF-format
254 * Import points to remember:
255 * 8-bit WAV is unsigned
256 * 16-bit WAV is signed
258 /* Use the same formulas as pcmconverter.c */
259 static inline INT16
cvtU8toS16(BYTE b
)
261 return (short)((b
+(b
<< 8))-32768);
264 static inline BYTE
cvtS16toU8(INT16 s
)
266 return (s
>> 8) ^ (unsigned char)0x80;
270 * Copy a single frame from the given input buffer to the given output buffer.
271 * Translate 8 <-> 16 bits and mono <-> stereo
273 static inline void cp_fields(const IDirectSoundBufferImpl
*dsb
, const BYTE
*ibuf
, BYTE
*obuf
)
275 DirectSoundDevice
* device
= dsb
->device
;
278 if (dsb
->pwfx
->wBitsPerSample
== 8) {
279 if (device
->pwfx
->wBitsPerSample
== 8 &&
280 device
->pwfx
->nChannels
== dsb
->pwfx
->nChannels
) {
281 /* avoid needless 8->16->8 conversion */
283 if (dsb
->pwfx
->nChannels
==2)
287 fl
= cvtU8toS16(*ibuf
);
288 fr
= (dsb
->pwfx
->nChannels
==2 ? cvtU8toS16(*(ibuf
+ 1)) : fl
);
290 fl
= *((const INT16
*)ibuf
);
291 fr
= (dsb
->pwfx
->nChannels
==2 ? *(((const INT16
*)ibuf
) + 1) : fl
);
294 if (device
->pwfx
->nChannels
== 2) {
295 if (device
->pwfx
->wBitsPerSample
== 8) {
296 *obuf
= cvtS16toU8(fl
);
297 *(obuf
+ 1) = cvtS16toU8(fr
);
300 if (device
->pwfx
->wBitsPerSample
== 16) {
301 *((INT16
*)obuf
) = fl
;
302 *(((INT16
*)obuf
) + 1) = fr
;
306 if (device
->pwfx
->nChannels
== 1) {
308 if (device
->pwfx
->wBitsPerSample
== 8) {
309 *obuf
= cvtS16toU8(fl
);
312 if (device
->pwfx
->wBitsPerSample
== 16) {
313 *((INT16
*)obuf
) = fl
;
320 * Calculate the distance between two buffer offsets, taking wraparound
323 static inline DWORD
DSOUND_BufPtrDiff(DWORD buflen
, DWORD ptr1
, DWORD ptr2
)
325 /* If these asserts fail, the problem is not here, but in the underlying code */
326 assert(ptr1
< buflen
);
327 assert(ptr2
< buflen
);
331 return buflen
+ ptr1
- ptr2
;
335 * Mix at most the given amount of data into the allocated temporary buffer
336 * of the given secondary buffer, starting from the dsb's first currently
337 * unsampled frame (writepos), translating frequency (pitch), stereo/mono
338 * and bits-per-sample so that it is ideal for the primary buffer.
339 * Doesn't perform any mixing - this is a straight copy/convert operation.
341 * dsb = the secondary buffer
342 * writepos = Starting position of changed buffer
343 * len = number of bytes to resample from writepos
345 * NOTE: writepos + len <= buflen, This function doesn't loop!
347 void DSOUND_MixToTemporary(const IDirectSoundBufferImpl
*dsb
, DWORD writepos
, DWORD len
)
350 BYTE
*ibp
, *obp
, *ibp_begin
, *obp_begin
;
351 INT iAdvance
= dsb
->pwfx
->nBlockAlign
;
352 INT oAdvance
= dsb
->device
->pwfx
->nBlockAlign
;
353 DWORD freqAcc
, target_writepos
, overshot
;
355 if (!dsb
->tmp_buffer
)
356 /* Nothing to do, already ideal format */
359 ibp
= dsb
->buffer
->memory
+ writepos
;
360 ibp_begin
= dsb
->buffer
->memory
;
361 obp_begin
= dsb
->tmp_buffer
;
363 TRACE("(%p, %p)\n", dsb
, ibp
);
364 /* Check for the best case */
365 if ((dsb
->freq
== dsb
->device
->pwfx
->nSamplesPerSec
) &&
366 (dsb
->pwfx
->wBitsPerSample
== dsb
->device
->pwfx
->wBitsPerSample
) &&
367 (dsb
->pwfx
->nChannels
== dsb
->device
->pwfx
->nChannels
)) {
368 obp
= dsb
->tmp_buffer
+ writepos
;
369 /* Why would we need a temporary buffer if we do best case? */
370 FIXME("(%p) Why do we resample for best case??? Bad!!\n", dsb
);
371 CopyMemory(obp
, ibp
, len
);
375 /* Check for same sample rate */
376 if (dsb
->freq
== dsb
->device
->pwfx
->nSamplesPerSec
) {
377 TRACE("(%p) Same sample rate %d = primary %d\n", dsb
,
378 dsb
->freq
, dsb
->device
->pwfx
->nSamplesPerSec
);
379 obp
= dsb
->tmp_buffer
+ writepos
/iAdvance
*oAdvance
;
380 for (i
= 0; i
< len
; i
+= iAdvance
) {
381 cp_fields(dsb
, ibp
, obp
);
388 /* Mix in different sample rates */
389 TRACE("(%p) Adjusting frequency: %d -> %d\n", dsb
, dsb
->freq
, dsb
->device
->pwfx
->nSamplesPerSec
);
390 size
= len
/ iAdvance
;
392 target_writepos
= DSOUND_secpos_to_bufpos(dsb
, writepos
, dsb
->sec_mixpos
, &freqAcc
);
393 overshot
= freqAcc
>> DSOUND_FREQSHIFT
;
396 if (overshot
>= size
)
399 writepos
+= overshot
* iAdvance
;
400 if (writepos
>= dsb
->buflen
)
402 ibp
= dsb
->buffer
->memory
+ writepos
;
403 freqAcc
&= (1 << DSOUND_FREQSHIFT
) - 1;
404 TRACE("Overshot: %d, freqAcc: %04x\n", overshot
, freqAcc
);
407 obp
= dsb
->tmp_buffer
+ target_writepos
;
408 /* FIXME: Small problem here when we're overwriting buf_mixpos, it then STILL uses old freqAcc, not sure if it matters or not */
410 cp_fields(dsb
, ibp
, obp
);
412 freqAcc
+= dsb
->freqAdjust
;
413 if (freqAcc
>= (1<<DSOUND_FREQSHIFT
)) {
414 ULONG adv
= (freqAcc
>>DSOUND_FREQSHIFT
);
415 freqAcc
&= (1<<DSOUND_FREQSHIFT
)-1;
416 ibp
+= adv
* iAdvance
;
422 /** Apply volume to the given soundbuffer from (primary) position writepos and length len
423 * Returns: NULL if no volume needs to be applied
424 * or else a memory handle that holds 'len' volume adjusted buffer */
425 static LPBYTE
DSOUND_MixerVol(const IDirectSoundBufferImpl
*dsb
, DWORD writepos
, INT len
)
431 INT nChannels
= dsb
->device
->pwfx
->nChannels
;
432 LPBYTE mem
= (dsb
->tmp_buffer
? dsb
->tmp_buffer
: dsb
->buffer
->memory
)+writepos
;
434 TRACE("(%p,%d)\n",dsb
,len
);
435 TRACE("left = %x, right = %x\n", dsb
->volpan
.dwTotalLeftAmpFactor
,
436 dsb
->volpan
.dwTotalRightAmpFactor
);
438 if (nChannels
!= 1 && nChannels
!= 2)
440 FIXME("There is no support for %d channels\n", nChannels
);
444 if (dsb
->device
->pwfx
->wBitsPerSample
!= 8 && dsb
->device
->pwfx
->wBitsPerSample
!= 16)
446 FIXME("There is no support for %d bpp\n", dsb
->device
->pwfx
->wBitsPerSample
);
450 if ((!(dsb
->dsbd
.dwFlags
& DSBCAPS_CTRLPAN
) || (dsb
->volpan
.lPan
== 0)) &&
451 (!(dsb
->dsbd
.dwFlags
& DSBCAPS_CTRLVOLUME
) || (dsb
->volpan
.lVolume
== 0)) &&
452 !(dsb
->dsbd
.dwFlags
& DSBCAPS_CTRL3D
))
453 return NULL
; /* Nothing to do */
455 if (dsb
->device
->tmp_buffer_len
< len
|| !dsb
->device
->tmp_buffer
)
457 dsb
->device
->tmp_buffer_len
= len
;
458 if (dsb
->device
->tmp_buffer
)
459 dsb
->device
->tmp_buffer
= HeapReAlloc(GetProcessHeap(), 0, dsb
->device
->tmp_buffer
, len
);
461 dsb
->device
->tmp_buffer
= HeapAlloc(GetProcessHeap(), 0, len
);
463 bpc
= dsb
->device
->tmp_buffer
;
466 vLeft
= dsb
->volpan
.dwTotalLeftAmpFactor
;
468 vRight
= dsb
->volpan
.dwTotalRightAmpFactor
;
472 switch (dsb
->device
->pwfx
->wBitsPerSample
) {
474 /* 8-bit WAV is unsigned, but we need to operate */
475 /* on signed data for this to work properly */
476 for (i
= 0; i
< len
; i
+=2) {
477 *(bpc
++) = (((INT
)(*(mem
++) - 128) * vLeft
) >> 16) + 128;
478 *(bpc
++) = (((INT
)(*(mem
++) - 128) * vRight
) >> 16) + 128;
480 if (len
% 2 == 1 && nChannels
== 1)
481 *(bpc
++) = (((INT
)(*(mem
++) - 128) * vLeft
) >> 16) + 128;
484 /* 16-bit WAV is signed -- much better */
485 for (i
= 0; i
< len
; i
+= 4) {
486 *(bps
++) = (*(mems
++) * vLeft
) >> 16;
487 *(bps
++) = (*(mems
++) * vRight
) >> 16;
489 if (len
% 4 == 2 && nChannels
== 1)
490 *(bps
++) = ((INT
)*(mems
++) * vLeft
) >> 16;
493 return dsb
->device
->tmp_buffer
;
497 * Mix (at most) the given number of bytes into the given position of the
498 * device buffer, from the secondary buffer "dsb" (starting at the current
499 * mix position for that buffer).
501 * Returns the number of bytes actually mixed into the device buffer. This
502 * will match fraglen unless the end of the secondary buffer is reached
503 * (and it is not looping).
505 * dsb = the secondary buffer to mix from
506 * writepos = position (offset) in device buffer to write at
507 * fraglen = number of bytes to mix
509 static DWORD
DSOUND_MixInBuffer(IDirectSoundBufferImpl
*dsb
, DWORD writepos
, DWORD fraglen
)
511 INT i
, len
= fraglen
, field
, todo
, ilen
;
512 BYTE
*ibuf
= (dsb
->tmp_buffer
? dsb
->tmp_buffer
: dsb
->buffer
->memory
) + dsb
->buf_mixpos
, *volbuf
;
515 TRACE("buf_mixpos=%d/%d sec_mixpos=%d/%d\n", dsb
->buf_mixpos
, dsb
->tmp_buffer_len
, dsb
->sec_mixpos
, dsb
->buflen
);
516 TRACE("(%p,%d,%d)\n",dsb
,writepos
,fraglen
);
518 assert(dsb
->buf_mixpos
+ len
<= dsb
->tmp_buffer_len
);
520 if (len
% dsb
->device
->pwfx
->nBlockAlign
) {
521 INT nBlockAlign
= dsb
->device
->pwfx
->nBlockAlign
;
522 ERR("length not a multiple of block size, len = %d, block size = %d\n", len
, nBlockAlign
);
523 len
-= len
% nBlockAlign
; /* data alignment */
526 /* Apply volume if needed */
527 volbuf
= DSOUND_MixerVol(dsb
, dsb
->buf_mixpos
, len
);
531 /* Now mix the temporary buffer into the devices main buffer */
532 if (dsb
->device
->pwfx
->wBitsPerSample
== 8) {
533 BYTE
*obuf
= dsb
->device
->buffer
+ writepos
;
535 if ((writepos
+ len
) <= dsb
->device
->buflen
)
538 todo
= dsb
->device
->buflen
- writepos
;
540 for (i
= 0; i
< todo
; i
++) {
541 /* 8-bit WAV is unsigned */
542 field
= (*ibuf
++ - 128);
543 field
+= (*obuf
- 128);
544 if (field
> 127) field
= 127;
545 else if (field
< -128) field
= -128;
546 *obuf
++ = field
+ 128;
551 obuf
= dsb
->device
->buffer
;
553 for (i
= 0; i
< todo
; i
++) {
554 /* 8-bit WAV is unsigned */
555 field
= (*ibuf
++ - 128);
556 field
+= (*obuf
- 128);
557 if (field
> 127) field
= 127;
558 else if (field
< -128) field
= -128;
559 *obuf
++ = field
+ 128;
563 INT16
*ibufs
, *obufs
;
565 ibufs
= (INT16
*) ibuf
;
566 obufs
= (INT16
*)(dsb
->device
->buffer
+ writepos
);
568 if ((writepos
+ len
) <= dsb
->device
->buflen
)
571 todo
= (dsb
->device
->buflen
- writepos
) / 2;
573 for (i
= 0; i
< todo
; i
++) {
574 /* 16-bit WAV is signed */
578 if (field
> 32767) field
= 32767;
579 else if (field
< -32768) field
= -32768;
583 if (todo
< (len
/ 2)) {
584 todo
= (len
/ 2) - todo
;
585 obufs
= (INT16
*)dsb
->device
->buffer
;
587 for (i
= 0; i
< todo
; i
++) {
588 /* 16-bit WAV is signed */
591 if (field
> 32767) field
= 32767;
592 else if (field
< -32768) field
= -32768;
598 oldpos
= dsb
->sec_mixpos
;
599 dsb
->buf_mixpos
+= len
;
601 if (dsb
->buf_mixpos
>= dsb
->tmp_buffer_len
) {
602 if (dsb
->playflags
& DSBPLAY_LOOPING
) {
603 dsb
->buf_mixpos
-= dsb
->tmp_buffer_len
;
604 } else if (dsb
->buf_mixpos
>= dsb
->tmp_buffer_len
) {
605 if (dsb
->buf_mixpos
> dsb
->tmp_buffer_len
)
606 ERR("Mixpos (%u) past buflen (%u), capping...\n", dsb
->buf_mixpos
, dsb
->tmp_buffer_len
);
607 dsb
->buf_mixpos
= dsb
->sec_mixpos
= 0;
608 dsb
->state
= STATE_STOPPED
;
610 DSOUND_RecalcFreqAcc(dsb
);
613 dsb
->sec_mixpos
= DSOUND_bufpos_to_secpos(dsb
, dsb
->buf_mixpos
);
614 ilen
= DSOUND_BufPtrDiff(dsb
->buflen
, dsb
->sec_mixpos
, oldpos
);
615 /* check for notification positions */
616 if (dsb
->dsbd
.dwFlags
& DSBCAPS_CTRLPOSITIONNOTIFY
&&
617 dsb
->state
!= STATE_STARTING
) {
618 DSOUND_CheckEvent(dsb
, oldpos
, ilen
);
621 /* increase mix position */
622 dsb
->primary_mixpos
+= len
;
623 if (dsb
->primary_mixpos
>= dsb
->device
->buflen
)
624 dsb
->primary_mixpos
-= dsb
->device
->buflen
;
629 * Mix some frames from the given secondary buffer "dsb" into the device
632 * dsb = the secondary buffer
633 * playpos = the current play position in the device buffer (primary buffer)
634 * writepos = the current safe-to-write position in the device buffer
635 * mixlen = the maximum number of bytes in the primary buffer to mix, from the
638 * Returns: the number of bytes beyond the writepos that were mixed.
640 static DWORD
DSOUND_MixOne(IDirectSoundBufferImpl
*dsb
, DWORD writepos
, DWORD mixlen
)
642 /* The buffer's primary_mixpos may be before or after the the device
643 * buffer's mixpos, but both must be ahead of writepos. */
646 TRACE("(%p,%d,%d)\n",dsb
,writepos
,mixlen
);
647 TRACE("writepos=%d, buf_mixpos=%d, primary_mixpos=%d, mixlen=%d\n", writepos
, dsb
->buf_mixpos
, dsb
->primary_mixpos
, mixlen
);
648 TRACE("looping=%d, leadin=%d, buflen=%d\n", dsb
->playflags
, dsb
->leadin
, dsb
->tmp_buffer_len
);
650 /* If leading in, only mix about 20 ms, and 'skip' mixing the rest, for more fluid pointer advancement */
651 if (dsb
->leadin
&& dsb
->state
== STATE_STARTING
)
653 if (mixlen
> 2 * dsb
->device
->fraglen
)
655 dsb
->primary_mixpos
+= mixlen
- 2 * dsb
->device
->fraglen
;
656 dsb
->primary_mixpos
%= dsb
->device
->buflen
;
661 /* calculate how much pre-buffering has already been done for this buffer */
662 primary_done
= DSOUND_BufPtrDiff(dsb
->device
->buflen
, dsb
->primary_mixpos
, writepos
);
665 if(mixlen
< primary_done
)
667 /* Should *NEVER* happen */
668 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
);
672 /* take into acount already mixed data */
673 mixlen
-= primary_done
;
675 TRACE("primary_done=%d, mixlen (primary) = %i\n", primary_done
, mixlen
);
680 /* First try to mix to the end of the buffer if possible
681 * Theoretically it would allow for better optimization
683 if (mixlen
+ dsb
->buf_mixpos
>= dsb
->tmp_buffer_len
)
685 DWORD newmixed
, mixfirst
= dsb
->tmp_buffer_len
- dsb
->buf_mixpos
;
686 newmixed
= DSOUND_MixInBuffer(dsb
, dsb
->primary_mixpos
, mixfirst
);
689 if (dsb
->playflags
& DSBPLAY_LOOPING
)
690 while (newmixed
&& mixlen
)
692 mixfirst
= (dsb
->tmp_buffer_len
< mixlen
? dsb
->tmp_buffer_len
: mixlen
);
693 newmixed
= DSOUND_MixInBuffer(dsb
, dsb
->primary_mixpos
, mixfirst
);
697 else DSOUND_MixInBuffer(dsb
, dsb
->primary_mixpos
, mixlen
);
699 /* re-calculate the primary done */
700 primary_done
= DSOUND_BufPtrDiff(dsb
->device
->buflen
, dsb
->primary_mixpos
, writepos
);
702 TRACE("new primary_mixpos=%d, total mixed data=%d\n", dsb
->primary_mixpos
, primary_done
);
704 /* Report back the total prebuffered amount for this buffer */
709 * For a DirectSoundDevice, go through all the currently playing buffers and
710 * mix them in to the device buffer.
712 * writepos = the current safe-to-write position in the primary buffer
713 * mixlen = the maximum amount to mix into the primary buffer
714 * (beyond the current writepos)
715 * mustlock = Do we have to fight for lock because we otherwise risk an underrun?
716 * recover = true if the sound device may have been reset and the write
717 * position in the device buffer changed
718 * all_stopped = reports back if all buffers have stopped
720 * Returns: the length beyond the writepos that was mixed to.
723 static DWORD
DSOUND_MixToPrimary(const DirectSoundDevice
*device
, DWORD writepos
, DWORD mixlen
, BOOL mustlock
, BOOL recover
, BOOL
*all_stopped
)
727 IDirectSoundBufferImpl
*dsb
;
730 /* unless we find a running buffer, all have stopped */
733 TRACE("(%d,%d,%d)\n", writepos
, mixlen
, recover
);
734 for (i
= 0; i
< device
->nrofbuffers
; i
++) {
735 dsb
= device
->buffers
[i
];
737 TRACE("MixToPrimary for %p, state=%d\n", dsb
, dsb
->state
);
739 if (dsb
->buflen
&& dsb
->state
&& !dsb
->hwbuf
) {
740 TRACE("Checking %p, mixlen=%d\n", dsb
, mixlen
);
741 if (!RtlAcquireResourceShared(&dsb
->lock
, mustlock
))
746 /* if buffer is stopping it is stopped now */
747 if (dsb
->state
== STATE_STOPPING
) {
748 dsb
->state
= STATE_STOPPED
;
749 DSOUND_CheckEvent(dsb
, 0, 0);
750 } else if (dsb
->state
!= STATE_STOPPED
) {
752 /* if recovering, reset the mix position */
753 if ((dsb
->state
== STATE_STARTING
) || recover
) {
754 dsb
->primary_mixpos
= writepos
;
757 /* mix next buffer into the main buffer */
758 len
= DSOUND_MixOne(dsb
, writepos
, mixlen
);
760 /* if the buffer was starting, it must be playing now */
761 if (dsb
->state
== STATE_STARTING
)
762 dsb
->state
= STATE_PLAYING
;
764 if (!minlen
) minlen
= len
;
766 /* record the minimum length mixed from all buffers */
767 /* we only want to return the length which *all* buffers have mixed */
768 else if (len
) minlen
= (len
< minlen
) ? len
: minlen
;
770 *all_stopped
= FALSE
;
772 RtlReleaseResource(&dsb
->lock
);
776 TRACE("Mixed at least %d from all buffers\n", minlen
);
777 if (!gotall
) return 0;
782 * Add buffers to the emulated wave device system.
784 * device = The current dsound playback device
785 * force = If TRUE, the function will buffer up as many frags as possible,
786 * even though and will ignore the actual state of the primary buffer.
791 static void DSOUND_WaveQueue(DirectSoundDevice
*device
, BOOL force
)
793 DWORD prebuf_frags
, wave_writepos
, wave_fragpos
, i
;
794 TRACE("(%p)\n", device
);
796 /* calculate the current wave frag position */
797 wave_fragpos
= (device
->pwplay
+ device
->pwqueue
) % device
->helfrags
;
799 /* calculte the current wave write position */
800 wave_writepos
= wave_fragpos
* device
->fraglen
;
802 TRACE("wave_fragpos = %i, wave_writepos = %i, pwqueue = %i, prebuf = %i\n",
803 wave_fragpos
, wave_writepos
, device
->pwqueue
, device
->prebuf
);
806 /* check remaining prebuffered frags */
807 prebuf_frags
= DSOUND_BufPtrDiff(device
->buflen
, device
->mixpos
, wave_writepos
);
808 prebuf_frags
= prebuf_frags
/ device
->fraglen
;
811 /* buffer the maximum amount of frags */
812 prebuf_frags
= device
->prebuf
;
815 /* limit to the queue we have left */
816 if((prebuf_frags
+ device
->pwqueue
) > device
->prebuf
)
817 prebuf_frags
= device
->prebuf
- device
->pwqueue
;
819 TRACE("prebuf_frags = %i\n", prebuf_frags
);
822 device
->pwqueue
+= prebuf_frags
;
824 /* get out of CS when calling the wave system */
825 LeaveCriticalSection(&(device
->mixlock
));
828 /* queue up the new buffers */
829 for(i
=0; i
<prebuf_frags
; i
++){
830 TRACE("queueing wave buffer %i\n", wave_fragpos
);
831 waveOutWrite(device
->hwo
, &device
->pwave
[wave_fragpos
], sizeof(WAVEHDR
));
833 wave_fragpos
%= device
->helfrags
;
837 EnterCriticalSection(&(device
->mixlock
));
839 TRACE("queue now = %i\n", device
->pwqueue
);
843 * Perform mixing for a Direct Sound device. That is, go through all the
844 * secondary buffers (the sound bites currently playing) and mix them in
845 * to the primary buffer (the device buffer).
847 static void DSOUND_PerformMix(DirectSoundDevice
*device
)
850 TRACE("(%p)\n", device
);
853 EnterCriticalSection(&(device
->mixlock
));
855 if (device
->priolevel
!= DSSCL_WRITEPRIMARY
) {
856 BOOL recover
= FALSE
, all_stopped
= FALSE
;
857 DWORD playpos
, writepos
, writelead
, maxq
, frag
, prebuff_max
, prebuff_left
, size1
, size2
;
859 BOOL lock
= (device
->hwbuf
&& !(device
->drvdesc
.dwFlags
& DSDDESC_DONTNEEDPRIMARYLOCK
));
860 BOOL mustlock
= FALSE
;
863 /* the sound of silence */
864 nfiller
= device
->pwfx
->wBitsPerSample
== 8 ? 128 : 0;
866 /* get the position in the primary buffer */
867 if (DSOUND_PrimaryGetPosition(device
, &playpos
, &writepos
) != 0){
868 LeaveCriticalSection(&(device
->mixlock
));
872 TRACE("primary playpos=%d, writepos=%d, clrpos=%d, mixpos=%d, buflen=%d\n",
873 playpos
,writepos
,device
->playpos
,device
->mixpos
,device
->buflen
);
874 assert(device
->playpos
< device
->buflen
);
876 /* wipe out just-played sound data */
877 if (playpos
< device
->playpos
) {
878 buf1
= device
->buffer
+ device
->playpos
;
879 buf2
= device
->buffer
;
880 size1
= device
->buflen
- device
->playpos
;
883 IDsDriverBuffer_Lock(device
->hwbuf
, &buf1
, &size1
, &buf2
, &size2
, device
->playpos
, size1
+size2
, 0);
884 FillMemory(buf1
, size1
, nfiller
);
885 if (playpos
&& (!buf2
|| !size2
))
886 FIXME("%d: (%d, %d)=>(%d, %d) There should be an additional buffer here!!\n", __LINE__
, device
->playpos
, device
->mixpos
, playpos
, writepos
);
887 FillMemory(buf2
, size2
, nfiller
);
889 IDsDriverBuffer_Unlock(device
->hwbuf
, buf1
, size1
, buf2
, size2
);
891 buf1
= device
->buffer
+ device
->playpos
;
893 size1
= playpos
- device
->playpos
;
896 IDsDriverBuffer_Lock(device
->hwbuf
, &buf1
, &size1
, &buf2
, &size2
, device
->playpos
, size1
+size2
, 0);
897 FillMemory(buf1
, size1
, nfiller
);
900 FIXME("%d: There should be no additional buffer here!!\n", __LINE__
);
901 FillMemory(buf2
, size2
, nfiller
);
904 IDsDriverBuffer_Unlock(device
->hwbuf
, buf1
, size1
, buf2
, size2
);
906 device
->playpos
= playpos
;
908 /* calc maximum prebuff */
909 prebuff_max
= (device
->prebuf
* device
->fraglen
);
911 /* check how close we are to an underrun. It occurs when the writepos overtakes the mixpos */
912 prebuff_left
= DSOUND_BufPtrDiff(device
->buflen
, device
->mixpos
, playpos
);
913 writelead
= DSOUND_BufPtrDiff(device
->buflen
, writepos
, playpos
);
915 /* find the maximum we can prebuffer from current write position */
916 maxq
= (writelead
< prebuff_max
) ? (prebuff_max
- writelead
) : 0;
918 TRACE("prebuff_left = %d, prebuff_max = %dx%d=%d, writelead=%d\n",
919 prebuff_left
, device
->prebuf
, device
->fraglen
, prebuff_max
, writelead
);
921 /* check for underrun. underrun occurs when the write position passes the mix position */
922 if((prebuff_left
> prebuff_max
) || (device
->state
== STATE_STOPPED
) || (device
->state
== STATE_STARTING
)){
923 if (device
->state
== STATE_STOPPING
|| device
->state
== STATE_PLAYING
)
924 WARN("Probable buffer underrun\n");
925 else TRACE("Buffer starting or buffer underrun\n");
927 /* recover mixing for all buffers */
930 /* reset mix position to write position */
931 device
->mixpos
= writepos
;
934 /* Do we risk an 'underrun' if we don't advance pointer? */
935 if (writelead
/device
->fraglen
<= ds_snd_queue_min
|| recover
)
939 IDsDriverBuffer_Lock(device
->hwbuf
, &buf1
, &size1
, &buf2
, &size2
, writepos
, maxq
, 0);
942 frag
= DSOUND_MixToPrimary(device
, writepos
, maxq
, mustlock
, recover
, &all_stopped
);
944 /* update the mix position, taking wrap-around into acount */
945 device
->mixpos
= writepos
+ frag
;
946 device
->mixpos
%= device
->buflen
;
950 DWORD frag2
= (frag
> size1
? frag
- size1
: 0);
954 FIXME("Buffering too much! (%d, %d, %d, %d)\n", maxq
, frag
, size2
, frag2
- size2
);
957 IDsDriverBuffer_Unlock(device
->hwbuf
, buf1
, frag
, buf2
, frag2
);
960 /* update prebuff left */
961 prebuff_left
= DSOUND_BufPtrDiff(device
->buflen
, device
->mixpos
, playpos
);
963 /* check if have a whole fragment */
964 if (prebuff_left
>= device
->fraglen
){
966 /* update the wave queue if using wave system */
967 if(device
->hwbuf
== NULL
){
968 DSOUND_WaveQueue(device
,TRUE
);
971 /* buffers are full. start playing if applicable */
972 if(device
->state
== STATE_STARTING
){
973 TRACE("started primary buffer\n");
974 if(DSOUND_PrimaryPlay(device
) != DS_OK
){
975 WARN("DSOUND_PrimaryPlay failed\n");
978 /* we are playing now */
979 device
->state
= STATE_PLAYING
;
983 /* buffers are full. start stopping if applicable */
984 if(device
->state
== STATE_STOPPED
){
985 TRACE("restarting primary buffer\n");
986 if(DSOUND_PrimaryPlay(device
) != DS_OK
){
987 WARN("DSOUND_PrimaryPlay failed\n");
990 /* start stopping again. as soon as there is no more data, it will stop */
991 device
->state
= STATE_STOPPING
;
996 /* if device was stopping, its for sure stopped when all buffers have stopped */
997 else if((all_stopped
== TRUE
) && (device
->state
== STATE_STOPPING
)){
998 TRACE("All buffers have stopped. Stopping primary buffer\n");
999 device
->state
= STATE_STOPPED
;
1001 /* stop the primary buffer now */
1002 DSOUND_PrimaryStop(device
);
1007 /* update the wave queue if using wave system */
1008 if(device
->hwbuf
== NULL
)
1009 DSOUND_WaveQueue(device
, TRUE
);
1011 /* Keep alsa happy, which needs GetPosition called once every 10 ms */
1012 IDsDriverBuffer_GetPosition(device
->hwbuf
, NULL
, NULL
);
1014 /* in the DSSCL_WRITEPRIMARY mode, the app is totally in charge... */
1015 if (device
->state
== STATE_STARTING
) {
1016 if (DSOUND_PrimaryPlay(device
) != DS_OK
)
1017 WARN("DSOUND_PrimaryPlay failed\n");
1019 device
->state
= STATE_PLAYING
;
1021 else if (device
->state
== STATE_STOPPING
) {
1022 if (DSOUND_PrimaryStop(device
) != DS_OK
)
1023 WARN("DSOUND_PrimaryStop failed\n");
1025 device
->state
= STATE_STOPPED
;
1029 LeaveCriticalSection(&(device
->mixlock
));
1033 void CALLBACK
DSOUND_timer(UINT timerID
, UINT msg
, DWORD_PTR dwUser
,
1034 DWORD_PTR dw1
, DWORD_PTR dw2
)
1036 DirectSoundDevice
* device
= (DirectSoundDevice
*)dwUser
;
1037 DWORD start_time
= GetTickCount();
1039 TRACE("(%d,%d,0x%lx,0x%lx,0x%lx)\n",timerID
,msg
,dwUser
,dw1
,dw2
);
1040 TRACE("entering at %d\n", start_time
);
1042 if (DSOUND_renderer
[device
->drvdesc
.dnDevNode
] != device
) {
1043 ERR("dsound died without killing us?\n");
1044 timeKillEvent(timerID
);
1045 timeEndPeriod(DS_TIME_RES
);
1049 RtlAcquireResourceShared(&(device
->buffer_list_lock
), TRUE
);
1052 DSOUND_PerformMix(device
);
1054 RtlReleaseResource(&(device
->buffer_list_lock
));
1056 end_time
= GetTickCount();
1057 TRACE("completed processing at %d, duration = %d\n", end_time
, end_time
- start_time
);
1060 void CALLBACK
DSOUND_callback(HWAVEOUT hwo
, UINT msg
, DWORD dwUser
, DWORD dw1
, DWORD dw2
)
1062 DirectSoundDevice
* device
= (DirectSoundDevice
*)dwUser
;
1063 TRACE("(%p,%x,%x,%x,%x)\n",hwo
,msg
,dwUser
,dw1
,dw2
);
1064 TRACE("entering at %d, msg=%08x(%s)\n", GetTickCount(), msg
,
1065 msg
==MM_WOM_DONE
? "MM_WOM_DONE" : msg
==MM_WOM_CLOSE
? "MM_WOM_CLOSE" :
1066 msg
==MM_WOM_OPEN
? "MM_WOM_OPEN" : "UNKNOWN");
1068 /* check if packet completed from wave driver */
1069 if (msg
== MM_WOM_DONE
) {
1072 EnterCriticalSection(&(device
->mixlock
));
1074 TRACE("done playing primary pos=%d\n", device
->pwplay
* device
->fraglen
);
1076 /* update playpos */
1078 device
->pwplay
%= device
->helfrags
;
1081 if(device
->pwqueue
== 0){
1082 ERR("Wave queue corrupted!\n");
1088 LeaveCriticalSection(&(device
->mixlock
));
1091 TRACE("completed\n");