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   1 /*                      DirectSound
   2  *
   3  * Copyright 1998 Marcus Meissner
   4  * Copyright 1998 Rob Riggs
   5  * Copyright 2000-2002 TransGaming Technologies, Inc.
   6  *
   7  * This library is free software; you can redistribute it and/or
   8  * modify it under the terms of the GNU Lesser General Public
   9  * License as published by the Free Software Foundation; either
  10  * version 2.1 of the License, or (at your option) any later version.
  11  *
  12  * This library is distributed in the hope that it will be useful,
  13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15  * Lesser General Public License for more details.
  16  *
  17  * You should have received a copy of the GNU Lesser General Public
  18  * License along with this library; if not, write to the Free Software
  19  * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
  20  */
  21
  22 #include <assert.h>
  23 #include <stdarg.h>
  24 #include <math.h>       /* Insomnia - pow() function */
  25
  26 #define NONAMELESSSTRUCT
  27 #define NONAMELESSUNION
  28 #include "windef.h"
  29 #include "winbase.h"
  30 #include "winuser.h"
  31 #include "mmsystem.h"
  32 #include "winreg.h"
  33 #include "winternl.h"
  34 #include "wine/debug.h"
  35 #include "dsound.h"
  36 #include "dsdriver.h"
  37 #include "dsound_private.h"
  38
  39 WINE_DEFAULT_DEBUG_CHANNEL(dsound);
  40
  41 void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan)
  42 {
  43         double temp;
  44         TRACE("(%p)\n",volpan);
  45
  46         TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
  47         /* the AmpFactors are expressed in 16.16 fixed point */
  48         volpan->dwVolAmpFactor = (ULONG) (pow(2.0, volpan->lVolume / 600.0) * 0xffff);
  49         /* FIXME: dwPan{Left|Right}AmpFactor */
  50
  51         /* FIXME: use calculated vol and pan ampfactors */
  52         temp = (double) (volpan->lVolume - (volpan->lPan > 0 ? volpan->lPan : 0));
  53         volpan->dwTotalLeftAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
  54         temp = (double) (volpan->lVolume + (volpan->lPan < 0 ? volpan->lPan : 0));
  55         volpan->dwTotalRightAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
  56
  57         TRACE("left = %x, right = %x\n", volpan->dwTotalLeftAmpFactor, volpan->dwTotalRightAmpFactor);
  58 }
  59
  60 void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan)
  61 {
  62     double left,right;
  63     TRACE("(%p)\n",volpan);
  64
  65     TRACE("left=%x, right=%x\n",volpan->dwTotalLeftAmpFactor,volpan->dwTotalRightAmpFactor);
  66     if (volpan->dwTotalLeftAmpFactor==0)
  67         left=-10000;
  68     else
  69         left=600 * log(((double)volpan->dwTotalLeftAmpFactor) / 0xffff) / log(2);
  70     if (volpan->dwTotalRightAmpFactor==0)
  71         right=-10000;
  72     else
  73         right=600 * log(((double)volpan->dwTotalRightAmpFactor) / 0xffff) / log(2);
  74     if (left<right)
  75     {
  76         volpan->lVolume=right;
  77         volpan->dwVolAmpFactor=volpan->dwTotalRightAmpFactor;
  78     }
  79     else
  80     {
  81         volpan->lVolume=left;
  82         volpan->dwVolAmpFactor=volpan->dwTotalLeftAmpFactor;
  83     }
  84     if (volpan->lVolume < -10000)
  85         volpan->lVolume=-10000;
  86     volpan->lPan=right-left;
  87     if (volpan->lPan < -10000)
  88         volpan->lPan=-10000;
  89
  90     TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
  91 }
  92
  93 void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
  94 {
  95         TRACE("(%p)\n",dsb);
  96
  97         /* calculate the 10ms write lead */
  98         dsb->writelead = (dsb->freq / 100) * dsb->pwfx->nBlockAlign;
  99 }
 100
 101 /**
 102  * Check for application callback requests for when the play position
 103  * reaches certain points.
 104  *
 105  * The offsets that will be triggered will be those between the recorded
 106  * "last played" position for the buffer (i.e. dsb->playpos) and "len" bytes
 107  * beyond that position.
 108  */
 109 void DSOUND_CheckEvent(IDirectSoundBufferImpl *dsb, int len)
 110 {
 111         int                     i;
 112         DWORD                   offset;
 113         LPDSBPOSITIONNOTIFY     event;
 114         TRACE("(%p,%d)\n",dsb,len);
 115
 116         if (dsb->nrofnotifies == 0)
 117                 return;
 118
 119         TRACE("(%p) buflen = %d, playpos = %d, len = %d\n",
 120                 dsb, dsb->buflen, dsb->playpos, len);
 121         for (i = 0; i < dsb->nrofnotifies ; i++) {
 122                 event = dsb->notifies + i;
 123                 offset = event->dwOffset;
 124                 TRACE("checking %d, position %d, event = %p\n",
 125                         i, offset, event->hEventNotify);
 126                 /* DSBPN_OFFSETSTOP has to be the last element. So this is */
 127                 /* OK. [Inside DirectX, p274] */
 128                 /*  */
 129                 /* This also means we can't sort the entries by offset, */
 130                 /* because DSBPN_OFFSETSTOP == -1 */
 131                 if (offset == DSBPN_OFFSETSTOP) {
 132                         if (dsb->state == STATE_STOPPED) {
 133                                 SetEvent(event->hEventNotify);
 134                                 TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
 135                                 return;
 136                         } else
 137                                 return;
 138                 }
 139                 if ((dsb->playpos + len) >= dsb->buflen) {
 140                         if ((offset < ((dsb->playpos + len) % dsb->buflen)) ||
 141                             (offset >= dsb->playpos)) {
 142                                 TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
 143                                 SetEvent(event->hEventNotify);
 144                         }
 145                 } else {
 146                         if ((offset >= dsb->playpos) && (offset < (dsb->playpos + len))) {
 147                                 TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
 148                                 SetEvent(event->hEventNotify);
 149                         }
 150                 }
 151         }
 152 }
 153
 154 /* WAV format info can be found at:
 155  *
 156  *    http://www.cwi.nl/ftp/audio/AudioFormats.part2
 157  *    ftp://ftp.cwi.nl/pub/audio/RIFF-format
 158  *
 159  * Import points to remember:
 160  *    8-bit WAV is unsigned
 161  *    16-bit WAV is signed
 162  */
 163  /* Use the same formulas as pcmconverter.c */
 164 static inline INT16 cvtU8toS16(BYTE b)
 165 {
 166     return (short)((b+(b << 8))-32768);
 167 }
 168
 169 static inline BYTE cvtS16toU8(INT16 s)
 170 {
 171     return (s >> 8) ^ (unsigned char)0x80;
 172 }
 173
 174 /**
 175  * Copy a single frame from the given input buffer to the given output buffer.
 176  * Translate 8 <-> 16 bits and mono <-> stereo
 177  */
 178 static inline void cp_fields(const IDirectSoundBufferImpl *dsb, BYTE *ibuf, BYTE *obuf )
 179 {
 180         DirectSoundDevice * device = dsb->device;
 181         INT fl,fr;
 182
 183         if (dsb->pwfx->wBitsPerSample == 8)  {
 184                 if (device->pwfx->wBitsPerSample == 8 &&
 185                     device->pwfx->nChannels == dsb->pwfx->nChannels) {
 186                         /* avoid needless 8->16->8 conversion */
 187                         *obuf=*ibuf;
 188                         if (dsb->pwfx->nChannels==2)
 189                                 *(obuf+1)=*(ibuf+1);
 190                         return;
 191                 }
 192                 fl = cvtU8toS16(*ibuf);
 193                 fr = (dsb->pwfx->nChannels==2 ? cvtU8toS16(*(ibuf + 1)) : fl);
 194         } else {
 195                 fl = *((INT16 *)ibuf);
 196                 fr = (dsb->pwfx->nChannels==2 ? *(((INT16 *)ibuf) + 1)  : fl);
 197         }
 198
 199         if (device->pwfx->nChannels == 2) {
 200                 if (device->pwfx->wBitsPerSample == 8) {
 201                         *obuf = cvtS16toU8(fl);
 202                         *(obuf + 1) = cvtS16toU8(fr);
 203                         return;
 204                 }
 205                 if (device->pwfx->wBitsPerSample == 16) {
 206                         *((INT16 *)obuf) = fl;
 207                         *(((INT16 *)obuf) + 1) = fr;
 208                         return;
 209                 }
 210         }
 211         if (device->pwfx->nChannels == 1) {
 212                 fl = (fl + fr) >> 1;
 213                 if (device->pwfx->wBitsPerSample == 8) {
 214                         *obuf = cvtS16toU8(fl);
 215                         return;
 216                 }
 217                 if (device->pwfx->wBitsPerSample == 16) {
 218                         *((INT16 *)obuf) = fl;
 219                         return;
 220                 }
 221         }
 222 }
 223
 224 /**
 225  * Mix at most the given amount of data into the given device buffer from the
 226  * given secondary buffer, starting from the dsb's first currently unmixed
 227  * frame (buf_mixpos), translating frequency (pitch), stereo/mono and
 228  * bits-per-sample. The secondary buffer sample is looped if it is not
 229  * long enough and it is a looping buffer.
 230  * (Doesn't perform any mixing - this is a straight copy operation).
 231  *
 232  * Now with PerfectPitch (tm) technology
 233  *
 234  * dsb = the secondary buffer
 235  * buf = the device buffer
 236  * len = number of bytes to store in the device buffer
 237  *
 238  * Returns: the number of bytes read from the secondary buffer
 239  *   (ie. len, adjusted for frequency, number of channels and sample size,
 240  *    and limited by buffer length for non-looping buffers)
 241  */
 242 static INT DSOUND_MixerNorm(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len)
 243 {
 244         INT     i, size, ipos, ilen;
 245         BYTE    *ibp, *obp;
 246         INT     iAdvance = dsb->pwfx->nBlockAlign;
 247         INT     oAdvance = dsb->device->pwfx->nBlockAlign;
 248
 249         ibp = dsb->buffer->memory + dsb->buf_mixpos;
 250         obp = buf;
 251
 252         TRACE("(%p, %p, %p), buf_mixpos=%d\n", dsb, ibp, obp, dsb->buf_mixpos);
 253         /* Check for the best case */
 254         if ((dsb->freq == dsb->device->pwfx->nSamplesPerSec) &&
 255             (dsb->pwfx->wBitsPerSample == dsb->device->pwfx->wBitsPerSample) &&
 256             (dsb->pwfx->nChannels == dsb->device->pwfx->nChannels)) {
 257                 INT bytesleft = dsb->buflen - dsb->buf_mixpos;
 258                 TRACE("(%p) Best case\n", dsb);
 259                 if (len <= bytesleft )
 260                         CopyMemory(obp, ibp, len);
 261                 else { /* wrap */
 262                         CopyMemory(obp, ibp, bytesleft);
 263                         CopyMemory(obp + bytesleft, dsb->buffer->memory, len - bytesleft);
 264                 }
 265                 return len;
 266         }
 267
 268         /* Check for same sample rate */
 269         if (dsb->freq == dsb->device->pwfx->nSamplesPerSec) {
 270                 TRACE("(%p) Same sample rate %d = primary %d\n", dsb,
 271                         dsb->freq, dsb->device->pwfx->nSamplesPerSec);
 272                 ilen = 0;
 273                 for (i = 0; i < len; i += oAdvance) {
 274                         cp_fields(dsb, ibp, obp );
 275                         ibp += iAdvance;
 276                         ilen += iAdvance;
 277                         obp += oAdvance;
 278                         if (ibp >= (BYTE *)(dsb->buffer->memory + dsb->buflen))
 279                                 ibp = dsb->buffer->memory;      /* wrap */
 280                 }
 281                 return (ilen);
 282         }
 283
 284         /* Mix in different sample rates */
 285         /* */
 286         /* New PerfectPitch(tm) Technology (c) 1998 Rob Riggs */
 287         /* Patent Pending :-] */
 288
 289         /* Patent enhancements (c) 2000 Ove Kåven,
 290          * TransGaming Technologies Inc. */
 291
 292         /* FIXME("(%p) Adjusting frequency: %ld -> %ld (need optimization)\n",
 293            dsb, dsb->freq, dsb->device->pwfx->nSamplesPerSec); */
 294
 295         size = len / oAdvance;
 296         ilen = 0;
 297         ipos = dsb->buf_mixpos;
 298         for (i = 0; i < size; i++) {
 299                 cp_fields(dsb, (dsb->buffer->memory + ipos), obp);
 300                 obp += oAdvance;
 301                 dsb->freqAcc += dsb->freqAdjust;
 302                 if (dsb->freqAcc >= (1<<DSOUND_FREQSHIFT)) {
 303                         ULONG adv = (dsb->freqAcc>>DSOUND_FREQSHIFT) * iAdvance;
 304                         dsb->freqAcc &= (1<<DSOUND_FREQSHIFT)-1;
 305                         ipos += adv; ilen += adv;
 306                         ipos %= dsb->buflen;
 307                 }
 308         }
 309         return ilen;
 310 }
 311
 312 static void DSOUND_MixerVol(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len)
 313 {
 314         INT     i;
 315         BYTE    *bpc = buf;
 316         INT16   *bps = (INT16 *) buf;
 317
 318         TRACE("(%p,%p,%d)\n",dsb,buf,len);
 319         TRACE("left = %x, right = %x\n", dsb->cvolpan.dwTotalLeftAmpFactor,
 320                 dsb->cvolpan.dwTotalRightAmpFactor);
 321
 322         if ((!(dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) || (dsb->cvolpan.lPan == 0)) &&
 323             (!(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) || (dsb->cvolpan.lVolume == 0)) &&
 324             !(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
 325                 return;         /* Nothing to do */
 326
 327         /* If we end up with some bozo coder using panning or 3D sound */
 328         /* with a mono primary buffer, it could sound very weird using */
 329         /* this method. Oh well, tough patooties. */
 330
 331         switch (dsb->device->pwfx->wBitsPerSample) {
 332         case 8:
 333                 /* 8-bit WAV is unsigned, but we need to operate */
 334                 /* on signed data for this to work properly */
 335                 switch (dsb->device->pwfx->nChannels) {
 336                 case 1:
 337                         for (i = 0; i < len; i++) {
 338                                 INT val = *bpc - 128;
 339                                 val = (val * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
 340                                 *bpc = val + 128;
 341                                 bpc++;
 342                         }
 343                         break;
 344                 case 2:
 345                         for (i = 0; i < len; i+=2) {
 346                                 INT val = *bpc - 128;
 347                                 val = (val * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
 348                                 *bpc++ = val + 128;
 349                                 val = *bpc - 128;
 350                                 val = (val * dsb->cvolpan.dwTotalRightAmpFactor) >> 16;
 351                                 *bpc = val + 128;
 352                                 bpc++;
 353                         }
 354                         break;
 355                 default:
 356                         FIXME("doesn't support %d channels\n", dsb->device->pwfx->nChannels);
 357                         break;
 358                 }
 359                 break;
 360         case 16:
 361                 /* 16-bit WAV is signed -- much better */
 362                 switch (dsb->device->pwfx->nChannels) {
 363                 case 1:
 364                         for (i = 0; i < len; i += 2) {
 365                                 *bps = (*bps * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
 366                                 bps++;
 367                         }
 368                         break;
 369                 case 2:
 370                         for (i = 0; i < len; i += 4) {
 371                                 *bps = (*bps * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
 372                                 bps++;
 373                                 *bps = (*bps * dsb->cvolpan.dwTotalRightAmpFactor) >> 16;
 374                                 bps++;
 375                         }
 376                         break;
 377                 default:
 378                         FIXME("doesn't support %d channels\n", dsb->device->pwfx->nChannels);
 379                         break;
 380                 }
 381                 break;
 382         default:
 383                 FIXME("doesn't support %d bit samples\n", dsb->device->pwfx->wBitsPerSample);
 384                 break;
 385         }
 386 }
 387
 388 /**
 389  * Make sure the device's tmp_buffer is at least the given size. Return a
 390  * pointer to it.
 391  */
 392 static LPBYTE DSOUND_tmpbuffer(DirectSoundDevice *device, DWORD len)
 393 {
 394     TRACE("(%p,%d)\n", device, len);
 395
 396     if (len > device->tmp_buffer_len) {
 397         if (device->tmp_buffer)
 398             device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, device->tmp_buffer, len);
 399         else
 400             device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, len);
 401
 402         device->tmp_buffer_len = len;
 403     }
 404
 405     return device->tmp_buffer;
 406 }
 407
 408 /**
 409  * Mix (at most) the given number of bytes into the given position of the
 410  * device buffer, from the secondary buffer "dsb" (starting at the current
 411  * mix position for that buffer).
 412  *
 413  * Returns the number of bytes actually mixed into the device buffer. This
 414  * will match fraglen unless the end of the secondary buffer is reached
 415  * (and it is not looping).
 416  *
 417  * dsb  = the secondary buffer to mix from
 418  * writepos = position (offset) in device buffer to write at
 419  * fraglen = number of bytes to mix
 420  */
 421 static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen)
 422 {
 423         INT     i, len, ilen, field, todo;
 424         BYTE    *buf, *ibuf;
 425
 426         TRACE("(%p,%d,%d)\n",dsb,writepos,fraglen);
 427
 428         len = fraglen;
 429         if (!(dsb->playflags & DSBPLAY_LOOPING)) {
 430                 /* This buffer is not looping, so make sure the requested
 431                  * length will not take us past the end of the buffer */
 432                 int secondary_remainder = dsb->buflen - dsb->buf_mixpos;
 433                 int adjusted_remainder = MulDiv(dsb->device->pwfx->nAvgBytesPerSec, secondary_remainder, dsb->nAvgBytesPerSec);
 434                 assert(adjusted_remainder >= 0);
 435                         /* The adjusted remainder must be at least one sample,
 436                          * otherwise we will never reach the end of the
 437                          * secondary buffer, as there will perpetually be a
 438                          * fractional remainder */
 439                 TRACE("secondary_remainder = %d, adjusted_remainder = %d, len = %d\n", secondary_remainder, adjusted_remainder, len);
 440                 if (adjusted_remainder < len) {
 441                         TRACE("clipping len to remainder of secondary buffer\n");
 442                         len = adjusted_remainder;
 443                 }
 444                 if (len == 0)
 445                         return 0;
 446         }
 447
 448         if (len % dsb->device->pwfx->nBlockAlign) {
 449                 INT nBlockAlign = dsb->device->pwfx->nBlockAlign;
 450                 ERR("length not a multiple of block size, len = %d, block size = %d\n", len, nBlockAlign);
 451                 len = (len / nBlockAlign) * nBlockAlign;        /* data alignment */
 452         }
 453
 454         if ((buf = ibuf = DSOUND_tmpbuffer(dsb->device, len)) == NULL)
 455                 return 0;
 456
 457         TRACE("MixInBuffer (%p) len = %d, dest = %d\n", dsb, len, writepos);
 458
 459         /* first, copy the data from the DirectSoundBuffer into the temporary
 460            buffer, translating frequency/bits-per-sample/number-of-channels
 461            to match the device settings */
 462         ilen = DSOUND_MixerNorm(dsb, ibuf, len);
 463         if ((dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) ||
 464             (dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) ||
 465             (dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
 466                 DSOUND_MixerVol(dsb, ibuf, len);
 467
 468         /* Now mix the temporary buffer into the devices main buffer */
 469         if (dsb->device->pwfx->wBitsPerSample == 8) {
 470                 BYTE    *obuf = dsb->device->buffer + writepos;
 471
 472                 if ((writepos + len) <= dsb->device->buflen)
 473                         todo = len;
 474                 else
 475                         todo = dsb->device->buflen - writepos;
 476
 477                 for (i = 0; i < todo; i++) {
 478                         /* 8-bit WAV is unsigned */
 479                         field = (*ibuf++ - 128);
 480                         field += (*obuf - 128);
 481                         if (field > 127) field = 127;
 482                         else if (field < -128) field = -128;
 483                         *obuf++ = field + 128;
 484                 }
 485
 486                 if (todo < len) {
 487                         todo = len - todo;
 488                         obuf = dsb->device->buffer;
 489
 490                         for (i = 0; i < todo; i++) {
 491                                 /* 8-bit WAV is unsigned */
 492                                 field = (*ibuf++ - 128);
 493                                 field += (*obuf - 128);
 494                                 if (field > 127) field = 127;
 495                                 else if (field < -128) field = -128;
 496                                 *obuf++ = field + 128;
 497                         }
 498                 }
 499         } else {
 500                 INT16   *ibufs, *obufs;
 501
 502                 ibufs = (INT16 *) ibuf;
 503                 obufs = (INT16 *)(dsb->device->buffer + writepos);
 504
 505                 if ((writepos + len) <= dsb->device->buflen)
 506                         todo = len / 2;
 507                 else
 508                         todo = (dsb->device->buflen - writepos) / 2;
 509
 510                 for (i = 0; i < todo; i++) {
 511                         /* 16-bit WAV is signed */
 512                         field = *ibufs++;
 513                         field += *obufs;
 514                         if (field > 32767) field = 32767;
 515                         else if (field < -32768) field = -32768;
 516                         *obufs++ = field;
 517                 }
 518
 519                 if (todo < (len / 2)) {
 520                         todo = (len / 2) - todo;
 521                         obufs = (INT16 *)dsb->device->buffer;
 522
 523                         for (i = 0; i < todo; i++) {
 524                                 /* 16-bit WAV is signed */
 525                                 field = *ibufs++;
 526                                 field += *obufs;
 527                                 if (field > 32767) field = 32767;
 528                                 else if (field < -32768) field = -32768;
 529                                 *obufs++ = field;
 530                         }
 531                 }
 532         }
 533
 534         if (dsb->leadin && (dsb->startpos > dsb->buf_mixpos) && (dsb->startpos <= dsb->buf_mixpos + ilen)) {
 535                 /* HACK... leadin should be reset when the PLAY position reaches the startpos,
 536                  * not the MIX position... but if the sound buffer is bigger than our prebuffering
 537                  * (which must be the case for the streaming buffers that need this hack anyway)
 538                  * plus DS_HEL_MARGIN or equivalent, then this ought to work anyway. */
 539                 dsb->leadin = FALSE;
 540         }
 541
 542         dsb->buf_mixpos += ilen;
 543
 544         if (dsb->buf_mixpos >= dsb->buflen) {
 545                 if (dsb->playflags & DSBPLAY_LOOPING) {
 546                         /* wrap */
 547                         dsb->buf_mixpos %= dsb->buflen;
 548                         if (dsb->leadin && (dsb->startpos <= dsb->buf_mixpos))
 549                                 dsb->leadin = FALSE; /* HACK: see above */
 550                 } else if (dsb->buf_mixpos > dsb->buflen) {
 551                         ERR("Mixpos (%u) past buflen (%u), capping...\n", dsb->buf_mixpos, dsb->buflen);
 552                         dsb->buf_mixpos = dsb->buflen;
 553                 }
 554         }
 555
 556         return len;
 557 }
 558
 559 static void DSOUND_PhaseCancel(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD len)
 560 {
 561         INT     ilen, field;
 562         UINT    i, todo;
 563         BYTE    *buf, *ibuf;
 564
 565         TRACE("(%p,%d,%d)\n",dsb,writepos,len);
 566
 567         if (len % dsb->device->pwfx->nBlockAlign) {
 568                 INT nBlockAlign = dsb->device->pwfx->nBlockAlign;
 569                 ERR("length not a multiple of block size, len = %d, block size = %d\n", len, nBlockAlign);
 570                 len = (len / nBlockAlign) * nBlockAlign;        /* data alignment */
 571         }
 572
 573         if ((buf = ibuf = DSOUND_tmpbuffer(dsb->device, len)) == NULL)
 574                 return;
 575
 576         TRACE("PhaseCancel (%p) len = %d, dest = %d\n", dsb, len, writepos);
 577
 578         ilen = DSOUND_MixerNorm(dsb, ibuf, len);
 579         if ((dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) ||
 580             (dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) ||
 581             (dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
 582                 DSOUND_MixerVol(dsb, ibuf, len);
 583
 584         /* subtract instead of add, to phase out premixed data */
 585         if (dsb->device->pwfx->wBitsPerSample == 8) {
 586                 BYTE    *obuf = dsb->device->buffer + writepos;
 587
 588                 if ((writepos + len) <= dsb->device->buflen)
 589                         todo = len;
 590                 else
 591                         todo = dsb->device->buflen - writepos;
 592
 593                 for (i = 0; i < todo; i++) {
 594                         /* 8-bit WAV is unsigned */
 595                         field = (*obuf - 128);
 596                         field -= (*ibuf++ - 128);
 597                         if (field > 127) field = 127;
 598                         else if (field < -128) field = -128;
 599                         *obuf++ = field + 128;
 600                 }
 601
 602                 if (todo < len) {
 603                         todo = len - todo;
 604                         obuf = dsb->device->buffer;
 605
 606                         for (i = 0; i < todo; i++) {
 607                                 /* 8-bit WAV is unsigned */
 608                                 field = (*obuf - 128);
 609                                 field -= (*ibuf++ - 128);
 610                                 if (field > 127) field = 127;
 611                                 else if (field < -128) field = -128;
 612                                 *obuf++ = field + 128;
 613                         }
 614                 }
 615         } else {
 616                 INT16   *ibufs, *obufs;
 617
 618                 ibufs = (INT16 *) ibuf;
 619                 obufs = (INT16 *)(dsb->device->buffer + writepos);
 620
 621                 if ((writepos + len) <= dsb->device->buflen)
 622                         todo = len / 2;
 623                 else
 624                         todo = (dsb->device->buflen - writepos) / 2;
 625
 626                 for (i = 0; i < todo; i++) {
 627                         /* 16-bit WAV is signed */
 628                         field = *obufs;
 629                         field -= *ibufs++;
 630                         if (field > 32767) field = 32767;
 631                         else if (field < -32768) field = -32768;
 632                         *obufs++ = field;
 633                 }
 634
 635                 if (todo < (len / 2)) {
 636                         todo = (len / 2) - todo;
 637                         obufs = (INT16 *)dsb->device->buffer;
 638
 639                         for (i = 0; i < todo; i++) {
 640                                 /* 16-bit WAV is signed */
 641                                 field = *obufs;
 642                                 field -= *ibufs++;
 643                                 if (field > 32767) field = 32767;
 644                                 else if (field < -32768) field = -32768;
 645                                 *obufs++ = field;
 646                         }
 647                 }
 648         }
 649 }
 650
 651 static void DSOUND_MixCancel(IDirectSoundBufferImpl *dsb, DWORD writepos, BOOL cancel)
 652 {
 653         DWORD   size, flen, len, npos, nlen;
 654         INT     iAdvance = dsb->pwfx->nBlockAlign;
 655         INT     oAdvance = dsb->device->pwfx->nBlockAlign;
 656         /* determine amount of premixed data to cancel */
 657         DWORD primary_done =
 658                 ((dsb->primary_mixpos < writepos) ? dsb->device->buflen : 0) +
 659                 dsb->primary_mixpos - writepos;
 660
 661         TRACE("(%p, %d), buf_mixpos=%d\n", dsb, writepos, dsb->buf_mixpos);
 662
 663         /* backtrack the mix position */
 664         size = primary_done / oAdvance;
 665         flen = size * dsb->freqAdjust;
 666         len = (flen >> DSOUND_FREQSHIFT) * iAdvance;
 667         flen &= (1<<DSOUND_FREQSHIFT)-1;
 668         while (dsb->freqAcc < flen) {
 669                 len += iAdvance;
 670                 dsb->freqAcc += 1<<DSOUND_FREQSHIFT;
 671         }
 672         len %= dsb->buflen;
 673         npos = ((dsb->buf_mixpos < len) ? dsb->buflen : 0) +
 674                 dsb->buf_mixpos - len;
 675         if (dsb->leadin && (dsb->startpos > npos) && (dsb->startpos <= npos + len)) {
 676                 /* stop backtracking at startpos */
 677                 npos = dsb->startpos;
 678                 len = ((dsb->buf_mixpos < npos) ? dsb->buflen : 0) +
 679                         dsb->buf_mixpos - npos;
 680                 flen = dsb->freqAcc;
 681                 nlen = len / dsb->pwfx->nBlockAlign;
 682                 nlen = ((nlen << DSOUND_FREQSHIFT) + flen) / dsb->freqAdjust;
 683                 nlen *= dsb->device->pwfx->nBlockAlign;
 684                 writepos =
 685                         ((dsb->primary_mixpos < nlen) ? dsb->device->buflen : 0) +
 686                         dsb->primary_mixpos - nlen;
 687         }
 688
 689         dsb->freqAcc -= flen;
 690         dsb->buf_mixpos = npos;
 691         dsb->primary_mixpos = writepos;
 692
 693         TRACE("new buf_mixpos=%d, primary_mixpos=%d (len=%d)\n",
 694               dsb->buf_mixpos, dsb->primary_mixpos, len);
 695
 696         if (cancel) DSOUND_PhaseCancel(dsb, writepos, len);
 697 }
 698
 699 void DSOUND_MixCancelAt(IDirectSoundBufferImpl *dsb, DWORD buf_writepos)
 700 {
 701 #if 0
 702         DWORD   i, size, flen, len, npos, nlen;
 703         INT     iAdvance = dsb->pwfx->nBlockAlign;
 704         INT     oAdvance = dsb->device->pwfx->nBlockAlign;
 705         /* determine amount of premixed data to cancel */
 706         DWORD buf_done =
 707                 ((dsb->buf_mixpos < buf_writepos) ? dsb->buflen : 0) +
 708                 dsb->buf_mixpos - buf_writepos;
 709 #endif
 710
 711         WARN("(%p, %d), buf_mixpos=%d\n", dsb, buf_writepos, dsb->buf_mixpos);
 712         /* since this is not implemented yet, just cancel *ALL* prebuffering for now
 713          * (which is faster anyway when there's only a single secondary buffer) */
 714         dsb->device->need_remix = TRUE;
 715 }
 716
 717 void DSOUND_ForceRemix(IDirectSoundBufferImpl *dsb)
 718 {
 719         TRACE("(%p)\n",dsb);
 720         EnterCriticalSection(&dsb->lock);
 721         if (dsb->state == STATE_PLAYING)
 722                 dsb->device->need_remix = TRUE;
 723         LeaveCriticalSection(&dsb->lock);
 724 }
 725
 726 /**
 727  * Calculate the distance between two buffer offsets, taking wraparound
 728  * into account.
 729  */
 730 static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
 731 {
 732         if (ptr1 >= ptr2) {
 733                 return ptr1 - ptr2;
 734         } else {
 735                 return buflen + ptr1 - ptr2;
 736         }
 737 }
 738
 739 /**
 740  * Mix some frames from the given secondary buffer "dsb" into the device
 741  * primary buffer.
 742  *
 743  * dsb = the secondary buffer
 744  * playpos = the current play position in the device buffer (primary buffer)
 745  * writepos = the current safe-to-write position in the device buffer
 746  * mixlen = the maximum number of bytes in the primary buffer to mix, from the
 747  *          current writepos.
 748  *
 749  * Returns: the number of bytes beyond the writepos that were mixed.
 750  */
 751 static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, DWORD playpos, DWORD writepos, DWORD mixlen)
 752 {
 753         /* The buffer's primary_mixpos may be before or after the the device
 754          * buffer's mixpos, but both must be ahead of writepos. */
 755
 756         DWORD len, slen;
 757         /* determine this buffer's write position */
 758         DWORD buf_writepos = DSOUND_CalcPlayPosition(dsb, writepos, writepos);
 759         /* determine how much already-mixed data exists */
 760         DWORD buf_done = DSOUND_BufPtrDiff(dsb->buflen, dsb->buf_mixpos, buf_writepos);
 761         DWORD primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);
 762         DWORD adv_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->device->mixpos, writepos);
 763         DWORD played = DSOUND_BufPtrDiff(dsb->buflen, buf_writepos, dsb->playpos);
 764         DWORD buf_left = dsb->buflen - buf_writepos;
 765         int still_behind;
 766
 767         TRACE("(%p,%d,%d,%d)\n",dsb,playpos,writepos,mixlen);
 768         TRACE("buf_writepos=%d, primary_writepos=%d\n", buf_writepos, writepos);
 769         TRACE("buf_done=%d, primary_done=%d\n", buf_done, primary_done);
 770         TRACE("buf_mixpos=%d, primary_mixpos=%d, mixlen=%d\n", dsb->buf_mixpos, dsb->primary_mixpos,
 771               mixlen);
 772         TRACE("looping=%d, startpos=%d, leadin=%d\n", dsb->playflags, dsb->startpos, dsb->leadin);
 773
 774         /* check for notification positions */
 775         if (dsb->dsbd.dwFlags & DSBCAPS_CTRLPOSITIONNOTIFY &&
 776             dsb->state != STATE_STARTING) {
 777                 DSOUND_CheckEvent(dsb, played);
 778         }
 779
 780         /* save write position for non-GETCURRENTPOSITION2... */
 781         dsb->playpos = buf_writepos;
 782
 783         /* check whether CalcPlayPosition detected a mixing underrun */
 784         if ((buf_done == 0) && (dsb->primary_mixpos != writepos)) {
 785                 /* it did, but did we have more to play? */
 786                 if ((dsb->playflags & DSBPLAY_LOOPING) ||
 787                     (dsb->buf_mixpos < dsb->buflen)) {
 788                         /* yes, have to recover */
 789                         ERR("underrun on sound buffer %p\n", dsb);
 790                         TRACE("recovering from underrun: primary_mixpos=%d\n", writepos);
 791                 }
 792                 dsb->primary_mixpos = writepos;
 793                 primary_done = 0;
 794         }
 795         /* determine how far ahead we should mix */
 796         if (((dsb->playflags & DSBPLAY_LOOPING) ||
 797              (dsb->leadin && (dsb->probably_valid_to != 0))) &&
 798             !(dsb->dsbd.dwFlags & DSBCAPS_STATIC)) {
 799                 /* if this is a streaming buffer, it typically means that
 800                  * we should defer mixing past probably_valid_to as long
 801                  * as we can, to avoid unnecessary remixing */
 802                 /* the heavy-looking calculations shouldn't be that bad,
 803                  * as any game isn't likely to be have more than 1 or 2
 804                  * streaming buffers in use at any time anyway... */
 805                 DWORD probably_valid_left =
 806                         (dsb->probably_valid_to == (DWORD)-1) ? dsb->buflen :
 807                         ((dsb->probably_valid_to < buf_writepos) ? dsb->buflen : 0) +
 808                         dsb->probably_valid_to - buf_writepos;
 809                 /* check for leadin condition */
 810                 if ((probably_valid_left == 0) &&
 811                     (dsb->probably_valid_to == dsb->startpos) &&
 812                     dsb->leadin)
 813                         probably_valid_left = dsb->buflen;
 814                 TRACE("streaming buffer probably_valid_to=%d, probably_valid_left=%d\n",
 815                       dsb->probably_valid_to, probably_valid_left);
 816                 /* check whether the app's time is already up */
 817                 if (probably_valid_left < dsb->writelead) {
 818                         WARN("probably_valid_to now within writelead, possible streaming underrun\n");
 819                         /* once we pass the point of no return,
 820                          * no reason to hold back anymore */
 821                         dsb->probably_valid_to = (DWORD)-1;
 822                         /* we just have to go ahead and mix what we have,
 823                          * there's no telling what the app is thinking anyway */
 824                 } else {
 825                         /* adjust for our frequency and our sample size */
 826                         probably_valid_left = MulDiv(probably_valid_left,
 827                                                      1 << DSOUND_FREQSHIFT,
 828                                                      dsb->pwfx->nBlockAlign * dsb->freqAdjust) *
 829                                               dsb->device->pwfx->nBlockAlign;
 830                         /* check whether to clip mix_len */
 831                         if (probably_valid_left < mixlen) {
 832                                 TRACE("clipping to probably_valid_left=%d\n", probably_valid_left);
 833                                 mixlen = probably_valid_left;
 834                         }
 835                 }
 836         }
 837         /* cut mixlen with what's already been mixed */
 838         if (mixlen < primary_done) {
 839                 /* huh? and still CalcPlayPosition didn't
 840                  * detect an underrun? */
 841                 FIXME("problem with underrun detection (mixlen=%d < primary_done=%d)\n", mixlen, primary_done);
 842                 return 0;
 843         }
 844         len = mixlen - primary_done;
 845         TRACE("remaining mixlen=%d\n", len);
 846
 847         if (len < dsb->device->fraglen) {
 848                 /* smaller than a fragment, wait until it gets larger
 849                  * before we take the mixing overhead */
 850                 TRACE("mixlen not worth it, deferring mixing\n");
 851                 still_behind = 1;
 852                 goto post_mix;
 853         }
 854
 855         /* ok, we know how much to mix, let's go */
 856         still_behind = (adv_done > primary_done);
 857         while (len) {
 858                 slen = dsb->device->buflen - dsb->primary_mixpos;
 859                 if (slen > len) slen = len;
 860                 slen = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, slen);
 861
 862                 if ((dsb->primary_mixpos < dsb->device->mixpos) &&
 863                     (dsb->primary_mixpos + slen >= dsb->device->mixpos))
 864                         still_behind = FALSE;
 865
 866                 dsb->primary_mixpos += slen; len -= slen;
 867                 dsb->primary_mixpos %= dsb->device->buflen;
 868
 869                 if ((dsb->state == STATE_STOPPED) || !slen) break;
 870         }
 871         TRACE("new primary_mixpos=%d, primary_advbase=%d\n", dsb->primary_mixpos, dsb->device->mixpos);
 872         TRACE("mixed data len=%d, still_behind=%d\n", mixlen-len, still_behind);
 873
 874 post_mix:
 875         /* check if buffer should be considered complete */
 876         if (buf_left < dsb->writelead &&
 877             !(dsb->playflags & DSBPLAY_LOOPING)) {
 878                 dsb->state = STATE_STOPPED;
 879                 dsb->playpos = 0;
 880                 dsb->last_playpos = 0;
 881                 dsb->buf_mixpos = 0;
 882                 dsb->leadin = FALSE;
 883                 dsb->need_remix = FALSE;
 884                 DSOUND_CheckEvent(dsb, buf_left);
 885         }
 886
 887         /* return how far we think the primary buffer can
 888          * advance its underrun detector...*/
 889         if (still_behind) return 0;
 890         if ((mixlen - len) < primary_done) return 0;
 891         slen = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, dsb->device->mixpos);
 892         if (slen > mixlen) {
 893                 /* the primary_done and still_behind checks above should have worked */
 894                 FIXME("problem with advancement calculation (advlen=%d > mixlen=%d)\n", slen, mixlen);
 895                 slen = 0;
 896         }
 897         return slen;
 898 }
 899
 900 /**
 901  * For a DirectSoundDevice, go through all the currently playing buffers and
 902  * mix them in to the device buffer.
 903  *
 904  * playpos = the current play position in the primary buffer
 905  * writepos = the current safe-to-write position in the primary buffer
 906  * mixlen = the maximum amount to mix into the primary buffer
 907  *          (beyond the current writepos)
 908  * recover = true if the sound device may have been reset and the write
 909  *           position in the device buffer changed
 910  *
 911  * Returns:  the length beyond the writepos that was mixed to.
 912  */
 913 static DWORD DSOUND_MixToPrimary(DirectSoundDevice *device, DWORD playpos, DWORD writepos, DWORD mixlen, BOOL recover)
 914 {
 915         INT                     i, len, maxlen = 0;
 916         IDirectSoundBufferImpl  *dsb;
 917
 918         TRACE("(%d,%d,%d,%d)\n", playpos, writepos, mixlen, recover);
 919         for (i = 0; i < device->nrofbuffers; i++) {
 920                 dsb = device->buffers[i];
 921
 922                 if (dsb->buflen && dsb->state && !dsb->hwbuf) {
 923                         TRACE("Checking %p, mixlen=%d\n", dsb, mixlen);
 924                         EnterCriticalSection(&(dsb->lock));
 925                         if (dsb->state == STATE_STOPPING) {
 926                                 DSOUND_MixCancel(dsb, writepos, TRUE);
 927                                 dsb->state = STATE_STOPPED;
 928                                 DSOUND_CheckEvent(dsb, 0);
 929                         } else {
 930                                 if ((dsb->state == STATE_STARTING) || recover) {
 931                                         dsb->primary_mixpos = writepos;
 932                                         dsb->cvolpan = dsb->volpan;
 933                                         dsb->need_remix = FALSE;
 934                                 }
 935                                 else if (dsb->need_remix) {
 936                                         DSOUND_MixCancel(dsb, writepos, TRUE);
 937                                         dsb->cvolpan = dsb->volpan;
 938                                         dsb->need_remix = FALSE;
 939                                 }
 940                                 len = DSOUND_MixOne(dsb, playpos, writepos, mixlen);
 941                                 if (dsb->state == STATE_STARTING)
 942                                         dsb->state = STATE_PLAYING;
 943                                 maxlen = (len > maxlen) ? len : maxlen;
 944                         }
 945                         LeaveCriticalSection(&(dsb->lock));
 946                 }
 947         }
 948
 949         return maxlen;
 950 }
 951
 952 static void DSOUND_MixReset(DirectSoundDevice *device, DWORD writepos)
 953 {
 954         INT                     i;
 955         IDirectSoundBufferImpl  *dsb;
 956         int nfiller;
 957
 958         TRACE("(%p,%d)\n", device, writepos);
 959
 960         /* the sound of silence */
 961         nfiller = device->pwfx->wBitsPerSample == 8 ? 128 : 0;
 962
 963         /* reset all buffer mix positions */
 964         for (i = 0; i < device->nrofbuffers; i++) {
 965                 dsb = device->buffers[i];
 966
 967                 if (dsb->buflen && dsb->state && !dsb->hwbuf) {
 968                         TRACE("Resetting %p\n", dsb);
 969                         EnterCriticalSection(&(dsb->lock));
 970                         if (dsb->state == STATE_STOPPING) {
 971                                 dsb->state = STATE_STOPPED;
 972                         }
 973                         else if (dsb->state == STATE_STARTING) {
 974                                 /* nothing */
 975                         } else {
 976                                 DSOUND_MixCancel(dsb, writepos, FALSE);
 977                                 dsb->cvolpan = dsb->volpan;
 978                                 dsb->need_remix = FALSE;
 979                         }
 980                         LeaveCriticalSection(&(dsb->lock));
 981                 }
 982         }
 983
 984         /* wipe out premixed data */
 985         if (device->mixpos < writepos) {
 986                 FillMemory(device->buffer + writepos, device->buflen - writepos, nfiller);
 987                 FillMemory(device->buffer, device->mixpos, nfiller);
 988         } else {
 989                 FillMemory(device->buffer + writepos, device->mixpos - writepos, nfiller);
 990         }
 991
 992         /* reset primary mix position */
 993         device->mixpos = writepos;
 994 }
 995
 996 static void DSOUND_CheckReset(DirectSoundDevice *device, DWORD writepos)
 997 {
 998         TRACE("(%p,%d)\n",device,writepos);
 999         if (device->need_remix) {
1000                 DSOUND_MixReset(device, writepos);
1001                 device->need_remix = FALSE;
1002                 /* maximize Half-Life performance */
1003                 device->prebuf = ds_snd_queue_min;
1004                 device->precount = 0;
1005         } else {
1006                 device->precount++;
1007                 if (device->precount >= 4) {
1008                         if (device->prebuf < ds_snd_queue_max)
1009                                 device->prebuf++;
1010                         device->precount = 0;
1011                 }
1012         }
1013         TRACE("premix adjust: %d\n", device->prebuf);
1014 }
1015
1016 void DSOUND_WaveQueue(DirectSoundDevice *device, DWORD mixq)
1017 {
1018         TRACE("(%p,%d)\n", device, mixq);
1019         if (mixq + device->pwqueue > ds_hel_queue) mixq = ds_hel_queue - device->pwqueue;
1020         TRACE("queueing %d buffers, starting at %d\n", mixq, device->pwwrite);
1021         for (; mixq; mixq--) {
1022                 waveOutWrite(device->hwo, device->pwave[device->pwwrite], sizeof(WAVEHDR));
1023                 device->pwwrite++;
1024                 if (device->pwwrite >= DS_HEL_FRAGS) device->pwwrite = 0;
1025                 device->pwqueue++;
1026         }
1027 }
1028
1029 /* #define SYNC_CALLBACK */
1030
1031 /**
1032  * Perform mixing for a Direct Sound device. That is, go through all the
1033  * secondary buffers (the sound bites currently playing) and mix them in
1034  * to the primary buffer (the device buffer).
1035  */
1036 static void DSOUND_PerformMix(DirectSoundDevice *device)
1037 {
1038         int nfiller;
1039         BOOL forced;
1040         HRESULT hres;
1041
1042         TRACE("(%p)\n", device);
1043
1044         /* the sound of silence */
1045         nfiller = device->pwfx->wBitsPerSample == 8 ? 128 : 0;
1046
1047         /* whether the primary is forced to play even without secondary buffers */
1048         forced = ((device->state == STATE_PLAYING) || (device->state == STATE_STARTING));
1049
1050         if (device->priolevel != DSSCL_WRITEPRIMARY) {
1051                 BOOL paused = ((device->state == STATE_STOPPED) || (device->state == STATE_STARTING));
1052                 /* FIXME: document variables */
1053                 DWORD playpos, writepos, inq, maxq, frag;
1054                 if (device->hwbuf) {
1055                         hres = IDsDriverBuffer_GetPosition(device->hwbuf, &playpos, &writepos);
1056                         if (hres) {
1057                             WARN("IDsDriverBuffer_GetPosition failed\n");
1058                             return;
1059                         }
1060                         /* Well, we *could* do Just-In-Time mixing using the writepos,
1061                          * but that's a little bit ambitious and unnecessary... */
1062                         /* rather add our safety margin to the writepos, if we're playing */
1063                         if (!paused) {
1064                                 writepos += device->writelead;
1065                                 writepos %= device->buflen;
1066                         } else writepos = playpos;
1067                 } else {
1068                         playpos = device->pwplay * device->fraglen;
1069                         writepos = playpos;
1070                         if (!paused) {
1071                                 writepos += ds_hel_margin * device->fraglen;
1072                                 writepos %= device->buflen;
1073                         }
1074                 }
1075                 TRACE("primary playpos=%d, writepos=%d, clrpos=%d, mixpos=%d, buflen=%d\n",
1076                       playpos,writepos,device->playpos,device->mixpos,device->buflen);
1077                 assert(device->playpos < device->buflen);
1078                 /* wipe out just-played sound data */
1079                 if (playpos < device->playpos) {
1080                         FillMemory(device->buffer + device->playpos, device->buflen - device->playpos, nfiller);
1081                         FillMemory(device->buffer, playpos, nfiller);
1082                 } else {
1083                         FillMemory(device->buffer + device->playpos, playpos - device->playpos, nfiller);
1084                 }
1085                 device->playpos = playpos;
1086
1087                 EnterCriticalSection(&(device->mixlock));
1088
1089                 /* reset mixing if necessary */
1090                 DSOUND_CheckReset(device, writepos);
1091
1092                 /* check how much prebuffering is left */
1093                 inq = DSOUND_BufPtrDiff(device->buflen, device->mixpos, writepos);
1094
1095                 /* find the maximum we can prebuffer */
1096                 if (!paused)
1097                         maxq = DSOUND_BufPtrDiff(device->buflen, playpos, writepos);
1098                 /* If we get the whole buffer, difference is 0, so we need to set whole buffer then */
1099                 if (paused || !maxq)
1100                         maxq = device->buflen;
1101
1102                 /* clip maxq to device->prebuf */
1103                 frag = device->prebuf * device->fraglen;
1104                 if (maxq > frag)
1105                         maxq = frag;
1106
1107                 /* check for consistency */
1108                 if (inq > maxq) {
1109                         /* the playback position must have passed our last
1110                          * mixed position, i.e. it's an underrun, or we have
1111                          * nothing more to play */
1112                         TRACE("reached end of mixed data (inq=%d, maxq=%d)\n", inq, maxq);
1113                         inq = 0;
1114                         /* stop the playback now, to allow buffers to refill */
1115                         if (device->state == STATE_PLAYING) {
1116                                 device->state = STATE_STARTING;
1117                         }
1118                         else if (device->state == STATE_STOPPING) {
1119                                 device->state = STATE_STOPPED;
1120                         }
1121                         else {
1122                                 /* how can we have an underrun if we aren't playing? */
1123                                 WARN("unexpected primary state (%d)\n", device->state);
1124                         }
1125 #ifdef SYNC_CALLBACK
1126                         /* DSOUND_callback may need this lock */
1127                         LeaveCriticalSection(&(device->mixlock));
1128 #endif
1129                         if (DSOUND_PrimaryStop(device) != DS_OK)
1130                                 WARN("DSOUND_PrimaryStop failed\n");
1131 #ifdef SYNC_CALLBACK
1132                         EnterCriticalSection(&(device->mixlock));
1133 #endif
1134                         if (device->hwbuf) {
1135                                 /* the Stop is supposed to reset play position to beginning of buffer */
1136                                 /* unfortunately, OSS is not able to do so, so get current pointer */
1137                                 hres = IDsDriverBuffer_GetPosition(device->hwbuf, &playpos, NULL);
1138                                 if (hres) {
1139                                         LeaveCriticalSection(&(device->mixlock));
1140                                         WARN("IDsDriverBuffer_GetPosition failed\n");
1141                                         return;
1142                                 }
1143                         } else {
1144                                 playpos = device->pwplay * device->fraglen;
1145                         }
1146                         writepos = playpos;
1147                         device->playpos = playpos;
1148                         device->mixpos = writepos;
1149                         inq = 0;
1150                         maxq = device->buflen;
1151                         if (maxq > frag) maxq = frag;
1152                         FillMemory(device->buffer, device->buflen, nfiller);
1153                         paused = TRUE;
1154                 }
1155
1156                 /* do the mixing */
1157                 frag = DSOUND_MixToPrimary(device, playpos, writepos, maxq, paused);
1158                 if (forced) frag = maxq - inq;
1159                 device->mixpos += frag;
1160                 device->mixpos %= device->buflen;
1161
1162                 if (frag) {
1163                         /* buffers have been filled, restart playback */
1164                         if (device->state == STATE_STARTING) {
1165                                 device->state = STATE_PLAYING;
1166                         }
1167                         else if (device->state == STATE_STOPPED) {
1168                                 /* the dsound is supposed to play if there's something to play
1169                                  * even if it is reported as stopped, so don't let this confuse you */
1170                                 device->state = STATE_STOPPING;
1171                         }
1172                         LeaveCriticalSection(&(device->mixlock));
1173                         if (paused) {
1174                                 if (DSOUND_PrimaryPlay(device) != DS_OK)
1175                                         WARN("DSOUND_PrimaryPlay failed\n");
1176                                 else
1177                                         TRACE("starting playback\n");
1178                         }
1179                 }
1180                 else
1181                         LeaveCriticalSection(&(device->mixlock));
1182         } else {
1183                 /* in the DSSCL_WRITEPRIMARY mode, the app is totally in charge... */
1184                 if (device->state == STATE_STARTING) {
1185                         if (DSOUND_PrimaryPlay(device) != DS_OK)
1186                                 WARN("DSOUND_PrimaryPlay failed\n");
1187                         else
1188                                 device->state = STATE_PLAYING;
1189                 }
1190                 else if (device->state == STATE_STOPPING) {
1191                         if (DSOUND_PrimaryStop(device) != DS_OK)
1192                                 WARN("DSOUND_PrimaryStop failed\n");
1193                         else
1194                                 device->state = STATE_STOPPED;
1195                 }
1196         }
1197 }
1198
1199 void CALLBACK DSOUND_timer(UINT timerID, UINT msg, DWORD_PTR dwUser,
1200                            DWORD_PTR dw1, DWORD_PTR dw2)
1201 {
1202         DirectSoundDevice * device = (DirectSoundDevice*)dwUser;
1203         DWORD start_time =  GetTickCount();
1204         DWORD end_time;
1205         TRACE("(%d,%d,0x%lx,0x%lx,0x%lx)\n",timerID,msg,dwUser,dw1,dw2);
1206         TRACE("entering at %d\n", start_time);
1207
1208         if (DSOUND_renderer[device->drvdesc.dnDevNode] != device) {
1209                 ERR("dsound died without killing us?\n");
1210                 timeKillEvent(timerID);
1211                 timeEndPeriod(DS_TIME_RES);
1212                 return;
1213         }
1214
1215         RtlAcquireResourceShared(&(device->buffer_list_lock), TRUE);
1216
1217         if (device->ref)
1218                 DSOUND_PerformMix(device);
1219
1220         RtlReleaseResource(&(device->buffer_list_lock));
1221
1222         end_time = GetTickCount();
1223         TRACE("completed processing at %d, duration = %d\n", end_time, end_time - start_time);
1224 }
1225
1226 void CALLBACK DSOUND_callback(HWAVEOUT hwo, UINT msg, DWORD dwUser, DWORD dw1, DWORD dw2)
1227 {
1228         DirectSoundDevice * device = (DirectSoundDevice*)dwUser;
1229         TRACE("(%p,%x,%x,%x,%x)\n",hwo,msg,dwUser,dw1,dw2);
1230         TRACE("entering at %d, msg=%08x(%s)\n", GetTickCount(), msg,
1231                 msg==MM_WOM_DONE ? "MM_WOM_DONE" : msg==MM_WOM_CLOSE ? "MM_WOM_CLOSE" :
1232                 msg==MM_WOM_OPEN ? "MM_WOM_OPEN" : "UNKNOWN");
1233         if (msg == MM_WOM_DONE) {
1234                 DWORD inq, mixq, fraglen, buflen, pwplay, playpos, mixpos;
1235                 if (device->pwqueue == (DWORD)-1) {
1236                         TRACE("completed due to reset\n");
1237                         return;
1238                 }
1239 /* it could be a bad idea to enter critical section here... if there's lock contention,
1240  * the resulting scheduling delays might obstruct the winmm player thread */
1241 #ifdef SYNC_CALLBACK
1242                 EnterCriticalSection(&(device->mixlock));
1243 #endif
1244                 /* retrieve current values */
1245                 fraglen = device->fraglen;
1246                 buflen = device->buflen;
1247                 pwplay = device->pwplay;
1248                 playpos = pwplay * fraglen;
1249                 mixpos = device->mixpos;
1250                 /* check remaining mixed data */
1251                 inq = DSOUND_BufPtrDiff(buflen, mixpos, playpos);
1252                 mixq = inq / fraglen;
1253                 if ((inq - (mixq * fraglen)) > 0) mixq++;
1254                 /* complete the playing buffer */
1255                 TRACE("done playing primary pos=%d\n", playpos);
1256                 pwplay++;
1257                 if (pwplay >= DS_HEL_FRAGS) pwplay = 0;
1258                 /* write new values */
1259                 device->pwplay = pwplay;
1260                 device->pwqueue--;
1261                 /* queue new buffer if we have data for it */
1262                 if (inq>1) DSOUND_WaveQueue(device, inq-1);
1263 #ifdef SYNC_CALLBACK
1264                 LeaveCriticalSection(&(device->mixlock));
1265 #endif
1266         }
1267         TRACE("completed\n");
1268 }