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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 "winternl.h"
  33 #include "wine/debug.h"
  34 #include "dsound.h"
  35 #include "dsdriver.h"
  36 #include "dsound_private.h"
  37
  38 WINE_DEFAULT_DEBUG_CHANNEL(dsound);
  39
  40 void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan)
  41 {
  42         double temp;
  43         TRACE("(%p)\n",volpan);
  44
  45         TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
  46         /* the AmpFactors are expressed in 16.16 fixed point */
  47         volpan->dwVolAmpFactor = (ULONG) (pow(2.0, volpan->lVolume / 600.0) * 0xffff);
  48         /* FIXME: dwPan{Left|Right}AmpFactor */
  49
  50         /* FIXME: use calculated vol and pan ampfactors */
  51         temp = (double) (volpan->lVolume - (volpan->lPan > 0 ? volpan->lPan : 0));
  52         volpan->dwTotalLeftAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
  53         temp = (double) (volpan->lVolume + (volpan->lPan < 0 ? volpan->lPan : 0));
  54         volpan->dwTotalRightAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
  55
  56         TRACE("left = %x, right = %x\n", volpan->dwTotalLeftAmpFactor, volpan->dwTotalRightAmpFactor);
  57 }
  58
  59 void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan)
  60 {
  61     double left,right;
  62     TRACE("(%p)\n",volpan);
  63
  64     TRACE("left=%x, right=%x\n",volpan->dwTotalLeftAmpFactor,volpan->dwTotalRightAmpFactor);
  65     if (volpan->dwTotalLeftAmpFactor==0)
  66         left=-10000;
  67     else
  68         left=600 * log(((double)volpan->dwTotalLeftAmpFactor) / 0xffff) / log(2);
  69     if (volpan->dwTotalRightAmpFactor==0)
  70         right=-10000;
  71     else
  72         right=600 * log(((double)volpan->dwTotalRightAmpFactor) / 0xffff) / log(2);
  73     if (left<right)
  74     {
  75         volpan->lVolume=right;
  76         volpan->dwVolAmpFactor=volpan->dwTotalRightAmpFactor;
  77     }
  78     else
  79     {
  80         volpan->lVolume=left;
  81         volpan->dwVolAmpFactor=volpan->dwTotalLeftAmpFactor;
  82     }
  83     if (volpan->lVolume < -10000)
  84         volpan->lVolume=-10000;
  85     volpan->lPan=right-left;
  86     if (volpan->lPan < -10000)
  87         volpan->lPan=-10000;
  88
  89     TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
  90 }
  91
  92 void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
  93 {
  94         TRACE("(%p)\n",dsb);
  95
  96         /* calculate the 10ms write lead */
  97         dsb->writelead = (dsb->freq / 100) * dsb->pwfx->nBlockAlign;
  98 }
  99
 100 /**
 101  * Check for application callback requests for when the play position
 102  * reaches certain points.
 103  *
 104  * The offsets that will be triggered will be those between the recorded
 105  * "last played" position for the buffer (i.e. dsb->playpos) and "len" bytes
 106  * beyond that position.
 107  */
 108 void DSOUND_CheckEvent(IDirectSoundBufferImpl *dsb, int len)
 109 {
 110         int                     i;
 111         DWORD                   offset;
 112         LPDSBPOSITIONNOTIFY     event;
 113         TRACE("(%p,%d)\n",dsb,len);
 114
 115         if (dsb->nrofnotifies == 0)
 116                 return;
 117
 118         TRACE("(%p) buflen = %d, playpos = %d, len = %d\n",
 119                 dsb, dsb->buflen, dsb->playpos, len);
 120         for (i = 0; i < dsb->nrofnotifies ; i++) {
 121                 event = dsb->notifies + i;
 122                 offset = event->dwOffset;
 123                 TRACE("checking %d, position %d, event = %p\n",
 124                         i, offset, event->hEventNotify);
 125                 /* DSBPN_OFFSETSTOP has to be the last element. So this is */
 126                 /* OK. [Inside DirectX, p274] */
 127                 /*  */
 128                 /* This also means we can't sort the entries by offset, */
 129                 /* because DSBPN_OFFSETSTOP == -1 */
 130                 if (offset == DSBPN_OFFSETSTOP) {
 131                         if (dsb->state == STATE_STOPPED) {
 132                                 SetEvent(event->hEventNotify);
 133                                 TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
 134                                 return;
 135                         } else
 136                                 return;
 137                 }
 138                 if ((dsb->playpos + len) >= dsb->buflen) {
 139                         if ((offset < ((dsb->playpos + len) % dsb->buflen)) ||
 140                             (offset >= dsb->playpos)) {
 141                                 TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
 142                                 SetEvent(event->hEventNotify);
 143                         }
 144                 } else {
 145                         if ((offset >= dsb->playpos) && (offset < (dsb->playpos + len))) {
 146                                 TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
 147                                 SetEvent(event->hEventNotify);
 148                         }
 149                 }
 150         }
 151 }
 152
 153 /* WAV format info can be found at:
 154  *
 155  *    http://www.cwi.nl/ftp/audio/AudioFormats.part2
 156  *    ftp://ftp.cwi.nl/pub/audio/RIFF-format
 157  *
 158  * Import points to remember:
 159  *    8-bit WAV is unsigned
 160  *    16-bit WAV is signed
 161  */
 162  /* Use the same formulas as pcmconverter.c */
 163 static inline INT16 cvtU8toS16(BYTE b)
 164 {
 165     return (short)((b+(b << 8))-32768);
 166 }
 167
 168 static inline BYTE cvtS16toU8(INT16 s)
 169 {
 170     return (s >> 8) ^ (unsigned char)0x80;
 171 }
 172
 173 /**
 174  * Copy a single frame from the given input buffer to the given output buffer.
 175  * Translate 8 <-> 16 bits and mono <-> stereo
 176  */
 177 static inline void cp_fields(const IDirectSoundBufferImpl *dsb, const BYTE *ibuf, BYTE *obuf )
 178 {
 179         DirectSoundDevice * device = dsb->device;
 180         INT fl,fr;
 181
 182         if (dsb->pwfx->wBitsPerSample == 8)  {
 183                 if (device->pwfx->wBitsPerSample == 8 &&
 184                     device->pwfx->nChannels == dsb->pwfx->nChannels) {
 185                         /* avoid needless 8->16->8 conversion */
 186                         *obuf=*ibuf;
 187                         if (dsb->pwfx->nChannels==2)
 188                                 *(obuf+1)=*(ibuf+1);
 189                         return;
 190                 }
 191                 fl = cvtU8toS16(*ibuf);
 192                 fr = (dsb->pwfx->nChannels==2 ? cvtU8toS16(*(ibuf + 1)) : fl);
 193         } else {
 194                 fl = *((const INT16 *)ibuf);
 195                 fr = (dsb->pwfx->nChannels==2 ? *(((const INT16 *)ibuf) + 1)  : fl);
 196         }
 197
 198         if (device->pwfx->nChannels == 2) {
 199                 if (device->pwfx->wBitsPerSample == 8) {
 200                         *obuf = cvtS16toU8(fl);
 201                         *(obuf + 1) = cvtS16toU8(fr);
 202                         return;
 203                 }
 204                 if (device->pwfx->wBitsPerSample == 16) {
 205                         *((INT16 *)obuf) = fl;
 206                         *(((INT16 *)obuf) + 1) = fr;
 207                         return;
 208                 }
 209         }
 210         if (device->pwfx->nChannels == 1) {
 211                 fl = (fl + fr) >> 1;
 212                 if (device->pwfx->wBitsPerSample == 8) {
 213                         *obuf = cvtS16toU8(fl);
 214                         return;
 215                 }
 216                 if (device->pwfx->wBitsPerSample == 16) {
 217                         *((INT16 *)obuf) = fl;
 218                         return;
 219                 }
 220         }
 221 }
 222
 223 /**
 224  * Mix at most the given amount of data into the given device buffer from the
 225  * given secondary buffer, starting from the dsb's first currently unmixed
 226  * frame (buf_mixpos), translating frequency (pitch), stereo/mono and
 227  * bits-per-sample. The secondary buffer sample is looped if it is not
 228  * long enough and it is a looping buffer.
 229  * (Doesn't perform any mixing - this is a straight copy operation).
 230  *
 231  * Now with PerfectPitch (tm) technology
 232  *
 233  * dsb = the secondary buffer
 234  * buf = the device buffer
 235  * len = number of bytes to store in the device buffer
 236  *
 237  * Returns: the number of bytes read from the secondary buffer
 238  *   (ie. len, adjusted for frequency, number of channels and sample size,
 239  *    and limited by buffer length for non-looping buffers)
 240  */
 241 static INT DSOUND_MixerNorm(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len)
 242 {
 243         INT     i, size, ipos, ilen;
 244         BYTE    *ibp, *obp;
 245         INT     iAdvance = dsb->pwfx->nBlockAlign;
 246         INT     oAdvance = dsb->device->pwfx->nBlockAlign;
 247
 248         ibp = dsb->buffer->memory + dsb->buf_mixpos;
 249         obp = buf;
 250
 251         TRACE("(%p, %p, %p), buf_mixpos=%d\n", dsb, ibp, obp, dsb->buf_mixpos);
 252         /* Check for the best case */
 253         if ((dsb->freq == dsb->device->pwfx->nSamplesPerSec) &&
 254             (dsb->pwfx->wBitsPerSample == dsb->device->pwfx->wBitsPerSample) &&
 255             (dsb->pwfx->nChannels == dsb->device->pwfx->nChannels)) {
 256                 INT bytesleft = dsb->buflen - dsb->buf_mixpos;
 257                 TRACE("(%p) Best case\n", dsb);
 258                 if (len <= bytesleft )
 259                         CopyMemory(obp, ibp, len);
 260                 else { /* wrap */
 261                         CopyMemory(obp, ibp, bytesleft);
 262                         CopyMemory(obp + bytesleft, dsb->buffer->memory, len - bytesleft);
 263                 }
 264                 return len;
 265         }
 266
 267         /* Check for same sample rate */
 268         if (dsb->freq == dsb->device->pwfx->nSamplesPerSec) {
 269                 TRACE("(%p) Same sample rate %d = primary %d\n", dsb,
 270                         dsb->freq, dsb->device->pwfx->nSamplesPerSec);
 271                 ilen = 0;
 272                 for (i = 0; i < len; i += oAdvance) {
 273                         cp_fields(dsb, ibp, obp );
 274                         ibp += iAdvance;
 275                         ilen += iAdvance;
 276                         obp += oAdvance;
 277                         if (ibp >= (BYTE *)(dsb->buffer->memory + dsb->buflen))
 278                                 ibp = dsb->buffer->memory;      /* wrap */
 279                 }
 280                 return (ilen);
 281         }
 282
 283         /* Mix in different sample rates */
 284         /* */
 285         /* New PerfectPitch(tm) Technology (c) 1998 Rob Riggs */
 286         /* Patent Pending :-] */
 287
 288         /* Patent enhancements (c) 2000 Ove Kåven,
 289          * TransGaming Technologies Inc. */
 290
 291         /* FIXME("(%p) Adjusting frequency: %ld -> %ld (need optimization)\n",
 292            dsb, dsb->freq, dsb->device->pwfx->nSamplesPerSec); */
 293
 294         size = len / oAdvance;
 295         ilen = 0;
 296         ipos = dsb->buf_mixpos;
 297         for (i = 0; i < size; i++) {
 298                 cp_fields(dsb, (dsb->buffer->memory + ipos), obp);
 299                 obp += oAdvance;
 300                 dsb->freqAcc += dsb->freqAdjust;
 301                 if (dsb->freqAcc >= (1<<DSOUND_FREQSHIFT)) {
 302                         ULONG adv = (dsb->freqAcc>>DSOUND_FREQSHIFT) * iAdvance;
 303                         dsb->freqAcc &= (1<<DSOUND_FREQSHIFT)-1;
 304                         ipos += adv; ilen += adv;
 305                         ipos %= dsb->buflen;
 306                 }
 307         }
 308         return ilen;
 309 }
 310
 311 static void DSOUND_MixerVol(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len)
 312 {
 313         INT     i;
 314         BYTE    *bpc = buf;
 315         INT16   *bps = (INT16 *) buf;
 316
 317         TRACE("(%p,%p,%d)\n",dsb,buf,len);
 318         TRACE("left = %x, right = %x\n", dsb->cvolpan.dwTotalLeftAmpFactor,
 319                 dsb->cvolpan.dwTotalRightAmpFactor);
 320
 321         if ((!(dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) || (dsb->cvolpan.lPan == 0)) &&
 322             (!(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) || (dsb->cvolpan.lVolume == 0)) &&
 323             !(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
 324                 return;         /* Nothing to do */
 325
 326         /* If we end up with some bozo coder using panning or 3D sound */
 327         /* with a mono primary buffer, it could sound very weird using */
 328         /* this method. Oh well, tough patooties. */
 329
 330         switch (dsb->device->pwfx->wBitsPerSample) {
 331         case 8:
 332                 /* 8-bit WAV is unsigned, but we need to operate */
 333                 /* on signed data for this to work properly */
 334                 switch (dsb->device->pwfx->nChannels) {
 335                 case 1:
 336                         for (i = 0; i < len; i++) {
 337                                 INT val = *bpc - 128;
 338                                 val = (val * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
 339                                 *bpc = val + 128;
 340                                 bpc++;
 341                         }
 342                         break;
 343                 case 2:
 344                         for (i = 0; i < len; i+=2) {
 345                                 INT val = *bpc - 128;
 346                                 val = (val * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
 347                                 *bpc++ = val + 128;
 348                                 val = *bpc - 128;
 349                                 val = (val * dsb->cvolpan.dwTotalRightAmpFactor) >> 16;
 350                                 *bpc = val + 128;
 351                                 bpc++;
 352                         }
 353                         break;
 354                 default:
 355                         FIXME("doesn't support %d channels\n", dsb->device->pwfx->nChannels);
 356                         break;
 357                 }
 358                 break;
 359         case 16:
 360                 /* 16-bit WAV is signed -- much better */
 361                 switch (dsb->device->pwfx->nChannels) {
 362                 case 1:
 363                         for (i = 0; i < len; i += 2) {
 364                                 *bps = (*bps * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
 365                                 bps++;
 366                         }
 367                         break;
 368                 case 2:
 369                         for (i = 0; i < len; i += 4) {
 370                                 *bps = (*bps * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
 371                                 bps++;
 372                                 *bps = (*bps * dsb->cvolpan.dwTotalRightAmpFactor) >> 16;
 373                                 bps++;
 374                         }
 375                         break;
 376                 default:
 377                         FIXME("doesn't support %d channels\n", dsb->device->pwfx->nChannels);
 378                         break;
 379                 }
 380                 break;
 381         default:
 382                 FIXME("doesn't support %d bit samples\n", dsb->device->pwfx->wBitsPerSample);
 383                 break;
 384         }
 385 }
 386
 387 /**
 388  * Make sure the device's tmp_buffer is at least the given size. Return a
 389  * pointer to it.
 390  */
 391 static LPBYTE DSOUND_tmpbuffer(DirectSoundDevice *device, DWORD len)
 392 {
 393     TRACE("(%p,%d)\n", device, len);
 394
 395     if (len > device->tmp_buffer_len) {
 396         if (device->tmp_buffer)
 397             device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, device->tmp_buffer, len);
 398         else
 399             device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, len);
 400
 401         device->tmp_buffer_len = len;
 402     }
 403
 404     return device->tmp_buffer;
 405 }
 406
 407 /**
 408  * Mix (at most) the given number of bytes into the given position of the
 409  * device buffer, from the secondary buffer "dsb" (starting at the current
 410  * mix position for that buffer).
 411  *
 412  * Returns the number of bytes actually mixed into the device buffer. This
 413  * will match fraglen unless the end of the secondary buffer is reached
 414  * (and it is not looping).
 415  *
 416  * dsb  = the secondary buffer to mix from
 417  * writepos = position (offset) in device buffer to write at
 418  * fraglen = number of bytes to mix
 419  */
 420 static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen)
 421 {
 422         INT     i, len, ilen, field, todo;
 423         BYTE    *buf, *ibuf;
 424
 425         TRACE("(%p,%d,%d)\n",dsb,writepos,fraglen);
 426
 427         len = fraglen;
 428         if (!(dsb->playflags & DSBPLAY_LOOPING)) {
 429                 /* This buffer is not looping, so make sure the requested
 430                  * length will not take us past the end of the buffer */
 431                 int secondary_remainder = dsb->buflen - dsb->buf_mixpos;
 432                 int adjusted_remainder = MulDiv(dsb->device->pwfx->nAvgBytesPerSec, secondary_remainder, dsb->nAvgBytesPerSec);
 433                 assert(adjusted_remainder >= 0);
 434                         /* The adjusted remainder must be at least one sample,
 435                          * otherwise we will never reach the end of the
 436                          * secondary buffer, as there will perpetually be a
 437                          * fractional remainder */
 438                 TRACE("secondary_remainder = %d, adjusted_remainder = %d, len = %d\n", secondary_remainder, adjusted_remainder, len);
 439                 if (adjusted_remainder < len) {
 440                         TRACE("clipping len to remainder of secondary buffer\n");
 441                         len = adjusted_remainder;
 442                 }
 443                 if (len == 0)
 444                         return 0;
 445         }
 446
 447         if (len % dsb->device->pwfx->nBlockAlign) {
 448                 INT nBlockAlign = dsb->device->pwfx->nBlockAlign;
 449                 ERR("length not a multiple of block size, len = %d, block size = %d\n", len, nBlockAlign);
 450                 len = (len / nBlockAlign) * nBlockAlign;        /* data alignment */
 451         }
 452
 453         if ((buf = ibuf = DSOUND_tmpbuffer(dsb->device, len)) == NULL)
 454                 return 0;
 455
 456         TRACE("MixInBuffer (%p) len = %d, dest = %d\n", dsb, len, writepos);
 457
 458         /* first, copy the data from the DirectSoundBuffer into the temporary
 459            buffer, translating frequency/bits-per-sample/number-of-channels
 460            to match the device settings */
 461         ilen = DSOUND_MixerNorm(dsb, ibuf, len);
 462         if ((dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) ||
 463             (dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) ||
 464             (dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
 465                 DSOUND_MixerVol(dsb, ibuf, len);
 466
 467         /* Now mix the temporary buffer into the devices main buffer */
 468         if (dsb->device->pwfx->wBitsPerSample == 8) {
 469                 BYTE    *obuf = dsb->device->buffer + writepos;
 470
 471                 if ((writepos + len) <= dsb->device->buflen)
 472                         todo = len;
 473                 else
 474                         todo = dsb->device->buflen - writepos;
 475
 476                 for (i = 0; i < todo; i++) {
 477                         /* 8-bit WAV is unsigned */
 478                         field = (*ibuf++ - 128);
 479                         field += (*obuf - 128);
 480                         if (field > 127) field = 127;
 481                         else if (field < -128) field = -128;
 482                         *obuf++ = field + 128;
 483                 }
 484
 485                 if (todo < len) {
 486                         todo = len - todo;
 487                         obuf = dsb->device->buffer;
 488
 489                         for (i = 0; i < todo; i++) {
 490                                 /* 8-bit WAV is unsigned */
 491                                 field = (*ibuf++ - 128);
 492                                 field += (*obuf - 128);
 493                                 if (field > 127) field = 127;
 494                                 else if (field < -128) field = -128;
 495                                 *obuf++ = field + 128;
 496                         }
 497                 }
 498         } else {
 499                 INT16   *ibufs, *obufs;
 500
 501                 ibufs = (INT16 *) ibuf;
 502                 obufs = (INT16 *)(dsb->device->buffer + writepos);
 503
 504                 if ((writepos + len) <= dsb->device->buflen)
 505                         todo = len / 2;
 506                 else
 507                         todo = (dsb->device->buflen - writepos) / 2;
 508
 509                 for (i = 0; i < todo; i++) {
 510                         /* 16-bit WAV is signed */
 511                         field = *ibufs++;
 512                         field += *obufs;
 513                         if (field > 32767) field = 32767;
 514                         else if (field < -32768) field = -32768;
 515                         *obufs++ = field;
 516                 }
 517
 518                 if (todo < (len / 2)) {
 519                         todo = (len / 2) - todo;
 520                         obufs = (INT16 *)dsb->device->buffer;
 521
 522                         for (i = 0; i < todo; i++) {
 523                                 /* 16-bit WAV is signed */
 524                                 field = *ibufs++;
 525                                 field += *obufs;
 526                                 if (field > 32767) field = 32767;
 527                                 else if (field < -32768) field = -32768;
 528                                 *obufs++ = field;
 529                         }
 530                 }
 531         }
 532
 533         if (dsb->leadin && (dsb->startpos > dsb->buf_mixpos) && (dsb->startpos <= dsb->buf_mixpos + ilen)) {
 534                 /* HACK... leadin should be reset when the PLAY position reaches the startpos,
 535                  * not the MIX position... but if the sound buffer is bigger than our prebuffering
 536                  * (which must be the case for the streaming buffers that need this hack anyway)
 537                  * plus DS_HEL_MARGIN or equivalent, then this ought to work anyway. */
 538                 dsb->leadin = FALSE;
 539         }
 540
 541         dsb->buf_mixpos += ilen;
 542
 543         if (dsb->buf_mixpos >= dsb->buflen) {
 544                 if (dsb->playflags & DSBPLAY_LOOPING) {
 545                         /* wrap */
 546                         dsb->buf_mixpos %= dsb->buflen;
 547                         if (dsb->leadin && (dsb->startpos <= dsb->buf_mixpos))
 548                                 dsb->leadin = FALSE; /* HACK: see above */
 549                 } else if (dsb->buf_mixpos > dsb->buflen) {
 550                         ERR("Mixpos (%u) past buflen (%u), capping...\n", dsb->buf_mixpos, dsb->buflen);
 551                         dsb->buf_mixpos = dsb->buflen;
 552                 }
 553         }
 554
 555         return len;
 556 }
 557
 558 static void DSOUND_PhaseCancel(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD len)
 559 {
 560         INT     ilen, field;
 561         UINT    i, todo;
 562         BYTE    *buf, *ibuf;
 563
 564         TRACE("(%p,%d,%d)\n",dsb,writepos,len);
 565
 566         if (len % dsb->device->pwfx->nBlockAlign) {
 567                 INT nBlockAlign = dsb->device->pwfx->nBlockAlign;
 568                 ERR("length not a multiple of block size, len = %d, block size = %d\n", len, nBlockAlign);
 569                 len = (len / nBlockAlign) * nBlockAlign;        /* data alignment */
 570         }
 571
 572         if ((buf = ibuf = DSOUND_tmpbuffer(dsb->device, len)) == NULL)
 573                 return;
 574
 575         TRACE("PhaseCancel (%p) len = %d, dest = %d\n", dsb, len, writepos);
 576
 577         ilen = DSOUND_MixerNorm(dsb, ibuf, len);
 578         if ((dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) ||
 579             (dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) ||
 580             (dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
 581                 DSOUND_MixerVol(dsb, ibuf, len);
 582
 583         /* subtract instead of add, to phase out premixed data */
 584         if (dsb->device->pwfx->wBitsPerSample == 8) {
 585                 BYTE    *obuf = dsb->device->buffer + writepos;
 586
 587                 if ((writepos + len) <= dsb->device->buflen)
 588                         todo = len;
 589                 else
 590                         todo = dsb->device->buflen - writepos;
 591
 592                 for (i = 0; i < todo; i++) {
 593                         /* 8-bit WAV is unsigned */
 594                         field = (*obuf - 128);
 595                         field -= (*ibuf++ - 128);
 596                         if (field > 127) field = 127;
 597                         else if (field < -128) field = -128;
 598                         *obuf++ = field + 128;
 599                 }
 600
 601                 if (todo < len) {
 602                         todo = len - todo;
 603                         obuf = dsb->device->buffer;
 604
 605                         for (i = 0; i < todo; i++) {
 606                                 /* 8-bit WAV is unsigned */
 607                                 field = (*obuf - 128);
 608                                 field -= (*ibuf++ - 128);
 609                                 if (field > 127) field = 127;
 610                                 else if (field < -128) field = -128;
 611                                 *obuf++ = field + 128;
 612                         }
 613                 }
 614         } else {
 615                 INT16   *ibufs, *obufs;
 616
 617                 ibufs = (INT16 *) ibuf;
 618                 obufs = (INT16 *)(dsb->device->buffer + writepos);
 619
 620                 if ((writepos + len) <= dsb->device->buflen)
 621                         todo = len / 2;
 622                 else
 623                         todo = (dsb->device->buflen - writepos) / 2;
 624
 625                 for (i = 0; i < todo; i++) {
 626                         /* 16-bit WAV is signed */
 627                         field = *obufs;
 628                         field -= *ibufs++;
 629                         if (field > 32767) field = 32767;
 630                         else if (field < -32768) field = -32768;
 631                         *obufs++ = field;
 632                 }
 633
 634                 if (todo < (len / 2)) {
 635                         todo = (len / 2) - todo;
 636                         obufs = (INT16 *)dsb->device->buffer;
 637
 638                         for (i = 0; i < todo; i++) {
 639                                 /* 16-bit WAV is signed */
 640                                 field = *obufs;
 641                                 field -= *ibufs++;
 642                                 if (field > 32767) field = 32767;
 643                                 else if (field < -32768) field = -32768;
 644                                 *obufs++ = field;
 645                         }
 646                 }
 647         }
 648 }
 649
 650 static void DSOUND_MixCancel(IDirectSoundBufferImpl *dsb, DWORD writepos, BOOL cancel)
 651 {
 652         DWORD   size, flen, len, npos, nlen;
 653         INT     iAdvance = dsb->pwfx->nBlockAlign;
 654         INT     oAdvance = dsb->device->pwfx->nBlockAlign;
 655         /* determine amount of premixed data to cancel */
 656         DWORD primary_done =
 657                 ((dsb->primary_mixpos < writepos) ? dsb->device->buflen : 0) +
 658                 dsb->primary_mixpos - writepos;
 659
 660         TRACE("(%p, %d), buf_mixpos=%d\n", dsb, writepos, dsb->buf_mixpos);
 661
 662         /* backtrack the mix position */
 663         size = primary_done / oAdvance;
 664         flen = size * dsb->freqAdjust;
 665         len = (flen >> DSOUND_FREQSHIFT) * iAdvance;
 666         flen &= (1<<DSOUND_FREQSHIFT)-1;
 667         while (dsb->freqAcc < flen) {
 668                 len += iAdvance;
 669                 dsb->freqAcc += 1<<DSOUND_FREQSHIFT;
 670         }
 671         len %= dsb->buflen;
 672         npos = ((dsb->buf_mixpos < len) ? dsb->buflen : 0) +
 673                 dsb->buf_mixpos - len;
 674         if (dsb->leadin && (dsb->startpos > npos) && (dsb->startpos <= npos + len)) {
 675                 /* stop backtracking at startpos */
 676                 npos = dsb->startpos;
 677                 len = ((dsb->buf_mixpos < npos) ? dsb->buflen : 0) +
 678                         dsb->buf_mixpos - npos;
 679                 flen = dsb->freqAcc;
 680                 nlen = len / dsb->pwfx->nBlockAlign;
 681                 nlen = ((nlen << DSOUND_FREQSHIFT) + flen) / dsb->freqAdjust;
 682                 nlen *= dsb->device->pwfx->nBlockAlign;
 683                 writepos =
 684                         ((dsb->primary_mixpos < nlen) ? dsb->device->buflen : 0) +
 685                         dsb->primary_mixpos - nlen;
 686         }
 687
 688         dsb->freqAcc -= flen;
 689         dsb->buf_mixpos = npos;
 690         dsb->primary_mixpos = writepos;
 691
 692         TRACE("new buf_mixpos=%d, primary_mixpos=%d (len=%d)\n",
 693               dsb->buf_mixpos, dsb->primary_mixpos, len);
 694
 695         if (cancel) DSOUND_PhaseCancel(dsb, writepos, len);
 696 }
 697
 698 void DSOUND_MixCancelAt(IDirectSoundBufferImpl *dsb, DWORD buf_writepos)
 699 {
 700 #if 0
 701         DWORD   i, size, flen, len, npos, nlen;
 702         INT     iAdvance = dsb->pwfx->nBlockAlign;
 703         INT     oAdvance = dsb->device->pwfx->nBlockAlign;
 704         /* determine amount of premixed data to cancel */
 705         DWORD buf_done =
 706                 ((dsb->buf_mixpos < buf_writepos) ? dsb->buflen : 0) +
 707                 dsb->buf_mixpos - buf_writepos;
 708 #endif
 709
 710         WARN("(%p, %d), buf_mixpos=%d\n", dsb, buf_writepos, dsb->buf_mixpos);
 711         /* since this is not implemented yet, just cancel *ALL* prebuffering for now
 712          * (which is faster anyway when there's only a single secondary buffer) */
 713         dsb->device->need_remix = TRUE;
 714 }
 715
 716 void DSOUND_ForceRemix(IDirectSoundBufferImpl *dsb)
 717 {
 718         TRACE("(%p)\n",dsb);
 719         EnterCriticalSection(&dsb->lock);
 720         if (dsb->state == STATE_PLAYING)
 721                 dsb->device->need_remix = TRUE;
 722         LeaveCriticalSection(&dsb->lock);
 723 }
 724
 725 /**
 726  * Calculate the distance between two buffer offsets, taking wraparound
 727  * into account.
 728  */
 729 static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
 730 {
 731         if (ptr1 >= ptr2) {
 732                 return ptr1 - ptr2;
 733         } else {
 734                 return buflen + ptr1 - ptr2;
 735         }
 736 }
 737
 738 /**
 739  * Mix some frames from the given secondary buffer "dsb" into the device
 740  * primary buffer.
 741  *
 742  * dsb = the secondary buffer
 743  * playpos = the current play position in the device buffer (primary buffer)
 744  * writepos = the current safe-to-write position in the device buffer
 745  * mixlen = the maximum number of bytes in the primary buffer to mix, from the
 746  *          current writepos.
 747  *
 748  * Returns: the number of bytes beyond the writepos that were mixed.
 749  */
 750 static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, DWORD playpos, DWORD writepos, DWORD mixlen)
 751 {
 752         /* The buffer's primary_mixpos may be before or after the the device
 753          * buffer's mixpos, but both must be ahead of writepos. */
 754
 755         DWORD len, slen;
 756         /* determine this buffer's write position */
 757         DWORD buf_writepos = DSOUND_CalcPlayPosition(dsb, writepos, writepos);
 758         /* determine how much already-mixed data exists */
 759         DWORD buf_done = DSOUND_BufPtrDiff(dsb->buflen, dsb->buf_mixpos, buf_writepos);
 760         DWORD primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);
 761         DWORD adv_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->device->mixpos, writepos);
 762         DWORD played = DSOUND_BufPtrDiff(dsb->buflen, buf_writepos, dsb->playpos);
 763         DWORD buf_left = dsb->buflen - buf_writepos;
 764         int still_behind;
 765
 766         TRACE("(%p,%d,%d,%d)\n",dsb,playpos,writepos,mixlen);
 767         TRACE("buf_writepos=%d, primary_writepos=%d\n", buf_writepos, writepos);
 768         TRACE("buf_done=%d, primary_done=%d\n", buf_done, primary_done);
 769         TRACE("buf_mixpos=%d, primary_mixpos=%d, mixlen=%d\n", dsb->buf_mixpos, dsb->primary_mixpos,
 770               mixlen);
 771         TRACE("looping=%d, startpos=%d, leadin=%d\n", dsb->playflags, dsb->startpos, dsb->leadin);
 772
 773         /* check for notification positions */
 774         if (dsb->dsbd.dwFlags & DSBCAPS_CTRLPOSITIONNOTIFY &&
 775             dsb->state != STATE_STARTING) {
 776                 DSOUND_CheckEvent(dsb, played);
 777         }
 778
 779         /* save write position for non-GETCURRENTPOSITION2... */
 780         dsb->playpos = buf_writepos;
 781
 782         /* check whether CalcPlayPosition detected a mixing underrun */
 783         if ((buf_done == 0) && (dsb->primary_mixpos != writepos)) {
 784                 /* it did, but did we have more to play? */
 785                 if ((dsb->playflags & DSBPLAY_LOOPING) ||
 786                     (dsb->buf_mixpos < dsb->buflen)) {
 787                         /* yes, have to recover */
 788                         ERR("underrun on sound buffer %p\n", dsb);
 789                         TRACE("recovering from underrun: primary_mixpos=%d\n", writepos);
 790                 }
 791                 dsb->primary_mixpos = writepos;
 792                 primary_done = 0;
 793         }
 794         /* determine how far ahead we should mix */
 795         if (((dsb->playflags & DSBPLAY_LOOPING) ||
 796              (dsb->leadin && (dsb->probably_valid_to != 0))) &&
 797             !(dsb->dsbd.dwFlags & DSBCAPS_STATIC)) {
 798                 /* if this is a streaming buffer, it typically means that
 799                  * we should defer mixing past probably_valid_to as long
 800                  * as we can, to avoid unnecessary remixing */
 801                 /* the heavy-looking calculations shouldn't be that bad,
 802                  * as any game isn't likely to be have more than 1 or 2
 803                  * streaming buffers in use at any time anyway... */
 804                 DWORD probably_valid_left =
 805                         (dsb->probably_valid_to == (DWORD)-1) ? dsb->buflen :
 806                         ((dsb->probably_valid_to < buf_writepos) ? dsb->buflen : 0) +
 807                         dsb->probably_valid_to - buf_writepos;
 808                 /* check for leadin condition */
 809                 if ((probably_valid_left == 0) &&
 810                     (dsb->probably_valid_to == dsb->startpos) &&
 811                     dsb->leadin)
 812                         probably_valid_left = dsb->buflen;
 813                 TRACE("streaming buffer probably_valid_to=%d, probably_valid_left=%d\n",
 814                       dsb->probably_valid_to, probably_valid_left);
 815                 /* check whether the app's time is already up */
 816                 if (probably_valid_left < dsb->writelead) {
 817                         WARN("probably_valid_to now within writelead, possible streaming underrun\n");
 818                         /* once we pass the point of no return,
 819                          * no reason to hold back anymore */
 820                         dsb->probably_valid_to = (DWORD)-1;
 821                         /* we just have to go ahead and mix what we have,
 822                          * there's no telling what the app is thinking anyway */
 823                 } else {
 824                         /* adjust for our frequency and our sample size */
 825                         probably_valid_left = MulDiv(probably_valid_left,
 826                                                      1 << DSOUND_FREQSHIFT,
 827                                                      dsb->pwfx->nBlockAlign * dsb->freqAdjust) *
 828                                               dsb->device->pwfx->nBlockAlign;
 829                         /* check whether to clip mix_len */
 830                         if (probably_valid_left < mixlen) {
 831                                 TRACE("clipping to probably_valid_left=%d\n", probably_valid_left);
 832                                 mixlen = probably_valid_left;
 833                         }
 834                 }
 835         }
 836         /* cut mixlen with what's already been mixed */
 837         if (mixlen < primary_done) {
 838                 /* huh? and still CalcPlayPosition didn't
 839                  * detect an underrun? */
 840                 FIXME("problem with underrun detection (mixlen=%d < primary_done=%d)\n", mixlen, primary_done);
 841                 return 0;
 842         }
 843         len = mixlen - primary_done;
 844         TRACE("remaining mixlen=%d\n", len);
 845
 846         if (len < dsb->device->fraglen) {
 847                 /* smaller than a fragment, wait until it gets larger
 848                  * before we take the mixing overhead */
 849                 TRACE("mixlen not worth it, deferring mixing\n");
 850                 still_behind = 1;
 851                 goto post_mix;
 852         }
 853
 854         /* ok, we know how much to mix, let's go */
 855         still_behind = (adv_done > primary_done);
 856         while (len) {
 857                 slen = dsb->device->buflen - dsb->primary_mixpos;
 858                 if (slen > len) slen = len;
 859                 slen = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, slen);
 860
 861                 if ((dsb->primary_mixpos < dsb->device->mixpos) &&
 862                     (dsb->primary_mixpos + slen >= dsb->device->mixpos))
 863                         still_behind = FALSE;
 864
 865                 dsb->primary_mixpos += slen; len -= slen;
 866                 dsb->primary_mixpos %= dsb->device->buflen;
 867
 868                 if ((dsb->state == STATE_STOPPED) || !slen) break;
 869         }
 870         TRACE("new primary_mixpos=%d, primary_advbase=%d\n", dsb->primary_mixpos, dsb->device->mixpos);
 871         TRACE("mixed data len=%d, still_behind=%d\n", mixlen-len, still_behind);
 872
 873 post_mix:
 874         /* check if buffer should be considered complete */
 875         if (buf_left < dsb->writelead &&
 876             !(dsb->playflags & DSBPLAY_LOOPING)) {
 877                 dsb->state = STATE_STOPPED;
 878                 dsb->playpos = 0;
 879                 dsb->last_playpos = 0;
 880                 dsb->buf_mixpos = 0;
 881                 dsb->leadin = FALSE;
 882                 dsb->need_remix = FALSE;
 883                 DSOUND_CheckEvent(dsb, buf_left);
 884         }
 885
 886         /* return how far we think the primary buffer can
 887          * advance its underrun detector...*/
 888         if (still_behind) return 0;
 889         if ((mixlen - len) < primary_done) return 0;
 890         slen = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, dsb->device->mixpos);
 891         if (slen > mixlen) {
 892                 /* the primary_done and still_behind checks above should have worked */
 893                 FIXME("problem with advancement calculation (advlen=%d > mixlen=%d)\n", slen, mixlen);
 894                 slen = 0;
 895         }
 896         return slen;
 897 }
 898
 899 /**
 900  * For a DirectSoundDevice, go through all the currently playing buffers and
 901  * mix them in to the device buffer.
 902  *
 903  * playpos = the current play position in the primary buffer
 904  * writepos = the current safe-to-write position in the primary buffer
 905  * mixlen = the maximum amount to mix into the primary buffer
 906  *          (beyond the current writepos)
 907  * recover = true if the sound device may have been reset and the write
 908  *           position in the device buffer changed
 909  *
 910  * Returns:  the length beyond the writepos that was mixed to.
 911  */
 912 static DWORD DSOUND_MixToPrimary(const DirectSoundDevice *device, DWORD playpos, DWORD writepos,
 913                                  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 }