dlls/dsound/mixer.c

   1 /*                      DirectSound
   2  *
   3  * Copyright 1998 Marcus Meissner
   4  * Copyright 1998 Rob Riggs
   5  * Copyright 2000-2002 TransGaming Technologies, Inc.
   6  * Copyright 2007 Peter Dons Tychsen
   7  * Copyright 2007 Maarten Lankhorst
   8  * Copyright 2011 Owen Rudge for CodeWeavers
   9  *
  10  * This library is free software; you can redistribute it and/or
  11  * modify it under the terms of the GNU Lesser General Public
  12  * License as published by the Free Software Foundation; either
  13  * version 2.1 of the License, or (at your option) any later version.
  14  *
  15  * This library is distributed in the hope that it will be useful,
  16  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  17  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  18  * Lesser General Public License for more details.
  19  *
  20  * You should have received a copy of the GNU Lesser General Public
  21  * License along with this library; if not, write to the Free Software
  22  * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
  23  */
  24
  25 #include <assert.h>
  26 #include <stdarg.h>
  27 #include <math.h>       /* Insomnia - pow() function */
  28
  29 #define COBJMACROS
  30
  31 #include "windef.h"
  32 #include "winbase.h"
  33 #include "mmsystem.h"
  34 #include "wingdi.h"
  35 #include "mmreg.h"
  36 #include "winternl.h"
  37 #include "wine/debug.h"
  38 #include "dsound.h"
  39 #include "ks.h"
  40 #include "ksmedia.h"
  41 #include "dsound_private.h"
  42 #include "fir.h"
  43
  44 WINE_DEFAULT_DEBUG_CHANNEL(dsound);
  45
  46 void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan)
  47 {
  48         double temp;
  49         TRACE("(%p)\n",volpan);
  50
  51         TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
  52         /* the AmpFactors are expressed in 16.16 fixed point */
  53
  54         /* FIXME: use calculated vol and pan ampfactors */
  55         temp = (double) (volpan->lVolume - (volpan->lPan > 0 ? volpan->lPan : 0));
  56         volpan->dwTotalAmpFactor[0] = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
  57         temp = (double) (volpan->lVolume + (volpan->lPan < 0 ? volpan->lPan : 0));
  58         volpan->dwTotalAmpFactor[1] = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
  59
  60         TRACE("left = %x, right = %x\n", volpan->dwTotalAmpFactor[0], volpan->dwTotalAmpFactor[1]);
  61 }
  62
  63 void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan)
  64 {
  65     double left,right;
  66     TRACE("(%p)\n",volpan);
  67
  68     TRACE("left=%x, right=%x\n",volpan->dwTotalAmpFactor[0],volpan->dwTotalAmpFactor[1]);
  69     if (volpan->dwTotalAmpFactor[0]==0)
  70         left=-10000;
  71     else
  72         left=600 * log(((double)volpan->dwTotalAmpFactor[0]) / 0xffff) / log(2);
  73     if (volpan->dwTotalAmpFactor[1]==0)
  74         right=-10000;
  75     else
  76         right=600 * log(((double)volpan->dwTotalAmpFactor[1]) / 0xffff) / log(2);
  77     if (left<right)
  78         volpan->lVolume=right;
  79     else
  80         volpan->lVolume=left;
  81     if (volpan->lVolume < -10000)
  82         volpan->lVolume=-10000;
  83     volpan->lPan=right-left;
  84     if (volpan->lPan < -10000)
  85         volpan->lPan=-10000;
  86
  87     TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
  88 }
  89
  90 /**
  91  * Recalculate the size for temporary buffer, and new writelead
  92  * Should be called when one of the following things occur:
  93  * - Primary buffer format is changed
  94  * - This buffer format (frequency) is changed
  95  */
  96 void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
  97 {
  98         DWORD ichannels = dsb->pwfx->nChannels;
  99         DWORD ochannels = dsb->device->pwfx->nChannels;
 100         WAVEFORMATEXTENSIBLE *pwfxe;
 101         BOOL ieee = FALSE;
 102
 103         TRACE("(%p)\n",dsb);
 104
 105         pwfxe = (WAVEFORMATEXTENSIBLE *) dsb->pwfx;
 106         dsb->freqAdjustNum = dsb->freq;
 107         dsb->freqAdjustDen = dsb->device->pwfx->nSamplesPerSec;
 108
 109         if ((pwfxe->Format.wFormatTag == WAVE_FORMAT_IEEE_FLOAT) || ((pwfxe->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE)
 110             && (IsEqualGUID(&pwfxe->SubFormat, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT))))
 111                 ieee = TRUE;
 112
 113         /**
 114          * Recalculate FIR step and gain.
 115          *
 116          * firstep says how many points of the FIR exist per one
 117          * sample in the secondary buffer. firgain specifies what
 118          * to multiply the FIR output by in order to attenuate it correctly.
 119          */
 120         if (dsb->freqAdjustNum / dsb->freqAdjustDen > 0) {
 121                 /**
 122                  * Yes, round it a bit to make sure that the
 123                  * linear interpolation factor never changes.
 124                  */
 125                 dsb->firstep = fir_step * dsb->freqAdjustDen / dsb->freqAdjustNum;
 126         } else {
 127                 dsb->firstep = fir_step;
 128         }
 129         dsb->firgain = (float)dsb->firstep / fir_step;
 130
 131         /* calculate the 10ms write lead */
 132         dsb->writelead = (dsb->freq / 100) * dsb->pwfx->nBlockAlign;
 133
 134         dsb->freqAccNum = 0;
 135
 136         dsb->get_aux = ieee ? getbpp[4] : getbpp[dsb->pwfx->wBitsPerSample/8 - 1];
 137         dsb->put_aux = putieee32;
 138
 139         dsb->get = dsb->get_aux;
 140         dsb->put = dsb->put_aux;
 141
 142         if (ichannels == ochannels)
 143         {
 144                 dsb->mix_channels = ichannels;
 145                 if (ichannels > 32) {
 146                         FIXME("Copying %u channels is unsupported, limiting to first 32\n", ichannels);
 147                         dsb->mix_channels = 32;
 148                 }
 149         }
 150         else if (ichannels == 1)
 151         {
 152                 dsb->mix_channels = 1;
 153
 154                 if (ochannels == 2)
 155                         dsb->put = put_mono2stereo;
 156                 else if (ochannels == 4)
 157                         dsb->put = put_mono2quad;
 158                 else if (ochannels == 6)
 159                         dsb->put = put_mono2surround51;
 160         }
 161         else if (ochannels == 1)
 162         {
 163                 dsb->mix_channels = 1;
 164                 dsb->get = get_mono;
 165         }
 166         else if (ichannels == 2 && ochannels == 4)
 167         {
 168                 dsb->mix_channels = 2;
 169                 dsb->put = put_stereo2quad;
 170         }
 171         else if (ichannels == 2 && ochannels == 6)
 172         {
 173                 dsb->mix_channels = 2;
 174                 dsb->put = put_stereo2surround51;
 175         }
 176         else
 177         {
 178                 if (ichannels > 2)
 179                         FIXME("Conversion from %u to %u channels is not implemented, falling back to stereo\n", ichannels, ochannels);
 180                 dsb->mix_channels = 2;
 181         }
 182 }
 183
 184 /**
 185  * Check for application callback requests for when the play position
 186  * reaches certain points.
 187  *
 188  * The offsets that will be triggered will be those between the recorded
 189  * "last played" position for the buffer (i.e. dsb->playpos) and "len" bytes
 190  * beyond that position.
 191  */
 192 void DSOUND_CheckEvent(const IDirectSoundBufferImpl *dsb, DWORD playpos, int len)
 193 {
 194     int first, left, right, check;
 195
 196     if(dsb->nrofnotifies == 0)
 197         return;
 198
 199     if(dsb->state == STATE_STOPPED){
 200         TRACE("Stopped...\n");
 201         /* DSBPN_OFFSETSTOP notifies are always at the start of the sorted array */
 202         for(left = 0; left < dsb->nrofnotifies; ++left){
 203             if(dsb->notifies[left].dwOffset != DSBPN_OFFSETSTOP)
 204                 break;
 205
 206             TRACE("Signalling %p\n", dsb->notifies[left].hEventNotify);
 207             SetEvent(dsb->notifies[left].hEventNotify);
 208         }
 209         return;
 210     }
 211
 212     for(first = 0; first < dsb->nrofnotifies && dsb->notifies[first].dwOffset == DSBPN_OFFSETSTOP; ++first)
 213         ;
 214
 215     if(first == dsb->nrofnotifies)
 216         return;
 217
 218     check = left = first;
 219     right = dsb->nrofnotifies - 1;
 220
 221     /* find leftmost notify that is greater than playpos */
 222     while(left != right){
 223         check = left + (right - left) / 2;
 224         if(dsb->notifies[check].dwOffset < playpos)
 225             left = check + 1;
 226         else if(dsb->notifies[check].dwOffset > playpos)
 227             right = check;
 228         else{
 229             left = check;
 230             break;
 231         }
 232     }
 233
 234     TRACE("Not stopped: first notify: %u (%u), left notify: %u (%u), range: [%u,%u)\n",
 235             first, dsb->notifies[first].dwOffset,
 236             left, dsb->notifies[left].dwOffset,
 237             playpos, (playpos + len) % dsb->buflen);
 238
 239     /* send notifications in range */
 240     if(dsb->notifies[left].dwOffset >= playpos){
 241         for(check = left; check < dsb->nrofnotifies; ++check){
 242             if(dsb->notifies[check].dwOffset >= playpos + len)
 243                 break;
 244
 245             TRACE("Signalling %p (%u)\n", dsb->notifies[check].hEventNotify, dsb->notifies[check].dwOffset);
 246             SetEvent(dsb->notifies[check].hEventNotify);
 247         }
 248     }
 249
 250     if(playpos + len > dsb->buflen){
 251         for(check = first; check < left; ++check){
 252             if(dsb->notifies[check].dwOffset >= (playpos + len) % dsb->buflen)
 253                 break;
 254
 255             TRACE("Signalling %p (%u)\n", dsb->notifies[check].hEventNotify, dsb->notifies[check].dwOffset);
 256             SetEvent(dsb->notifies[check].hEventNotify);
 257         }
 258     }
 259 }
 260
 261 static inline float get_current_sample(const IDirectSoundBufferImpl *dsb,
 262         DWORD mixpos, DWORD channel)
 263 {
 264     if (mixpos >= dsb->buflen && !(dsb->playflags & DSBPLAY_LOOPING))
 265         return 0.0f;
 266     return dsb->get(dsb, mixpos % dsb->buflen, channel);
 267 }
 268
 269 static UINT cp_fields_noresample(IDirectSoundBufferImpl *dsb, UINT count)
 270 {
 271     UINT istride = dsb->pwfx->nBlockAlign;
 272     UINT ostride = dsb->device->pwfx->nChannels * sizeof(float);
 273     DWORD channel, i;
 274     for (i = 0; i < count; i++)
 275         for (channel = 0; channel < dsb->mix_channels; channel++)
 276             dsb->put(dsb, i * ostride, channel, get_current_sample(dsb,
 277                     dsb->sec_mixpos + i * istride, channel));
 278     return count;
 279 }
 280
 281 static UINT cp_fields_resample(IDirectSoundBufferImpl *dsb, UINT count, LONG64 *freqAccNum)
 282 {
 283     UINT i, channel;
 284     UINT istride = dsb->pwfx->nBlockAlign;
 285     UINT ostride = dsb->device->pwfx->nChannels * sizeof(float);
 286
 287     LONG64 freqAcc_start = *freqAccNum;
 288     LONG64 freqAcc_end = freqAcc_start + count * dsb->freqAdjustNum;
 289     UINT dsbfirstep = dsb->firstep;
 290     UINT channels = dsb->mix_channels;
 291     UINT max_ipos = (freqAcc_start + count * dsb->freqAdjustNum) / dsb->freqAdjustDen;
 292
 293     UINT fir_cachesize = (fir_len + dsbfirstep - 2) / dsbfirstep;
 294     UINT required_input = max_ipos + fir_cachesize;
 295
 296     float* intermediate = HeapAlloc(GetProcessHeap(), 0,
 297             sizeof(float) * required_input * channels);
 298
 299     float* fir_copy = HeapAlloc(GetProcessHeap(), 0,
 300             sizeof(float) * fir_cachesize);
 301
 302     /* Important: this buffer MUST be non-interleaved
 303      * if you want -msse3 to have any effect.
 304      * This is good for CPU cache effects, too.
 305      */
 306     float* itmp = intermediate;
 307     for (channel = 0; channel < channels; channel++)
 308         for (i = 0; i < required_input; i++)
 309             *(itmp++) = get_current_sample(dsb,
 310                     dsb->sec_mixpos + i * istride, channel);
 311
 312     for(i = 0; i < count; ++i) {
 313         UINT int_fir_steps = (freqAcc_start + i * dsb->freqAdjustNum) * dsbfirstep / dsb->freqAdjustDen;
 314         float total_fir_steps = (freqAcc_start + i * dsb->freqAdjustNum) * dsbfirstep / (float)dsb->freqAdjustDen;
 315         UINT ipos = int_fir_steps / dsbfirstep;
 316
 317         UINT idx = (ipos + 1) * dsbfirstep - int_fir_steps - 1;
 318         float rem = int_fir_steps + 1.0 - total_fir_steps;
 319
 320         int fir_used = 0;
 321         while (idx < fir_len - 1) {
 322             fir_copy[fir_used++] = fir[idx] * (1.0 - rem) + fir[idx + 1] * rem;
 323             idx += dsb->firstep;
 324         }
 325
 326         assert(fir_used <= fir_cachesize);
 327         assert(ipos + fir_used <= required_input);
 328
 329         for (channel = 0; channel < dsb->mix_channels; channel++) {
 330             int j;
 331             float sum = 0.0;
 332             float* cache = &intermediate[channel * required_input + ipos];
 333             for (j = 0; j < fir_used; j++)
 334                 sum += fir_copy[j] * cache[j];
 335             dsb->put(dsb, i * ostride, channel, sum * dsb->firgain);
 336         }
 337     }
 338
 339     *freqAccNum = freqAcc_end % dsb->freqAdjustDen;
 340
 341     HeapFree(GetProcessHeap(), 0, fir_copy);
 342     HeapFree(GetProcessHeap(), 0, intermediate);
 343
 344     return max_ipos;
 345 }
 346
 347 static void cp_fields(IDirectSoundBufferImpl *dsb, UINT count, LONG64 *freqAccNum)
 348 {
 349     DWORD ipos, adv;
 350
 351     if (dsb->freqAdjustNum == dsb->freqAdjustDen)
 352         adv = cp_fields_noresample(dsb, count); /* *freqAccNum is unmodified */
 353     else
 354         adv = cp_fields_resample(dsb, count, freqAccNum);
 355
 356     ipos = dsb->sec_mixpos + adv * dsb->pwfx->nBlockAlign;
 357     if (ipos >= dsb->buflen) {
 358         if (dsb->playflags & DSBPLAY_LOOPING)
 359             ipos %= dsb->buflen;
 360         else {
 361             ipos = 0;
 362             dsb->state = STATE_STOPPED;
 363         }
 364     }
 365
 366     dsb->sec_mixpos = ipos;
 367 }
 368
 369 /**
 370  * Calculate the distance between two buffer offsets, taking wraparound
 371  * into account.
 372  */
 373 static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
 374 {
 375 /* If these asserts fail, the problem is not here, but in the underlying code */
 376         assert(ptr1 < buflen);
 377         assert(ptr2 < buflen);
 378         if (ptr1 >= ptr2) {
 379                 return ptr1 - ptr2;
 380         } else {
 381                 return buflen + ptr1 - ptr2;
 382         }
 383 }
 384 /**
 385  * Mix at most the given amount of data into the allocated temporary buffer
 386  * of the given secondary buffer, starting from the dsb's first currently
 387  * unsampled frame (writepos), translating frequency (pitch), stereo/mono
 388  * and bits-per-sample so that it is ideal for the primary buffer.
 389  * Doesn't perform any mixing - this is a straight copy/convert operation.
 390  *
 391  * dsb = the secondary buffer
 392  * writepos = Starting position of changed buffer
 393  * len = number of bytes to resample from writepos
 394  *
 395  * NOTE: writepos + len <= buflen. When called by mixer, MixOne makes sure of this.
 396  */
 397 static void DSOUND_MixToTemporary(IDirectSoundBufferImpl *dsb, DWORD frames)
 398 {
 399         UINT size_bytes = frames * sizeof(float) * dsb->device->pwfx->nChannels;
 400         HRESULT hr;
 401         int i;
 402
 403         if (dsb->device->tmp_buffer_len < size_bytes || !dsb->device->tmp_buffer)
 404         {
 405                 dsb->device->tmp_buffer_len = size_bytes;
 406                 if (dsb->device->tmp_buffer)
 407                         dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, size_bytes);
 408                 else
 409                         dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, size_bytes);
 410         }
 411
 412         cp_fields(dsb, frames, &dsb->freqAccNum);
 413
 414         if (size_bytes > 0) {
 415                 for (i = 0; i < dsb->num_filters; i++) {
 416                         if (dsb->filters[i].inplace) {
 417                                 hr = IMediaObjectInPlace_Process(dsb->filters[i].inplace, size_bytes, (BYTE*)dsb->device->tmp_buffer, 0, DMO_INPLACE_NORMAL);
 418
 419                                 if (FAILED(hr))
 420                                         WARN("IMediaObjectInPlace_Process failed for filter %u\n", i);
 421                         } else
 422                                 WARN("filter %u has no inplace object - unsupported\n", i);
 423                 }
 424         }
 425 }
 426
 427 static void DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, INT frames)
 428 {
 429         INT     i;
 430         float vols[DS_MAX_CHANNELS];
 431         UINT channels = dsb->device->pwfx->nChannels, chan;
 432
 433         TRACE("(%p,%d)\n",dsb,frames);
 434         TRACE("left = %x, right = %x\n", dsb->volpan.dwTotalAmpFactor[0],
 435                 dsb->volpan.dwTotalAmpFactor[1]);
 436
 437         if ((!(dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) || (dsb->volpan.lPan == 0)) &&
 438             (!(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) || (dsb->volpan.lVolume == 0)) &&
 439              !(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
 440                 return; /* Nothing to do */
 441
 442         if (channels > DS_MAX_CHANNELS)
 443         {
 444                 FIXME("There is no support for %u channels\n", channels);
 445                 return;
 446         }
 447
 448         for (i = 0; i < channels; ++i)
 449                 vols[i] = dsb->volpan.dwTotalAmpFactor[i] / ((float)0xFFFF);
 450
 451         for(i = 0; i < frames; ++i){
 452                 for(chan = 0; chan < channels; ++chan){
 453                         dsb->device->tmp_buffer[i * channels + chan] *= vols[chan];
 454                 }
 455         }
 456 }
 457
 458 /**
 459  * Mix (at most) the given number of bytes into the given position of the
 460  * device buffer, from the secondary buffer "dsb" (starting at the current
 461  * mix position for that buffer).
 462  *
 463  * Returns the number of bytes actually mixed into the device buffer. This
 464  * will match fraglen unless the end of the secondary buffer is reached
 465  * (and it is not looping).
 466  *
 467  * dsb  = the secondary buffer to mix from
 468  * writepos = position (offset) in device buffer to write at
 469  * fraglen = number of bytes to mix
 470  */
 471 static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, float *mix_buffer, DWORD writepos, DWORD fraglen)
 472 {
 473         INT len = fraglen;
 474         float *ibuf;
 475         DWORD oldpos;
 476         UINT frames = fraglen / dsb->device->pwfx->nBlockAlign;
 477
 478         TRACE("sec_mixpos=%d/%d\n", dsb->sec_mixpos, dsb->buflen);
 479         TRACE("(%p,%d,%d)\n",dsb,writepos,fraglen);
 480
 481         if (len % dsb->device->pwfx->nBlockAlign) {
 482                 INT nBlockAlign = dsb->device->pwfx->nBlockAlign;
 483                 ERR("length not a multiple of block size, len = %d, block size = %d\n", len, nBlockAlign);
 484                 len -= len % nBlockAlign; /* data alignment */
 485         }
 486
 487         /* Resample buffer to temporary buffer specifically allocated for this purpose, if needed */
 488         oldpos = dsb->sec_mixpos;
 489
 490         DSOUND_MixToTemporary(dsb, frames);
 491         ibuf = dsb->device->tmp_buffer;
 492
 493         /* Apply volume if needed */
 494         DSOUND_MixerVol(dsb, frames);
 495
 496         mixieee32(ibuf, mix_buffer, frames * dsb->device->pwfx->nChannels);
 497
 498         /* check for notification positions */
 499         if (dsb->dsbd.dwFlags & DSBCAPS_CTRLPOSITIONNOTIFY &&
 500             dsb->state != STATE_STARTING) {
 501                 INT ilen = DSOUND_BufPtrDiff(dsb->buflen, dsb->sec_mixpos, oldpos);
 502                 DSOUND_CheckEvent(dsb, oldpos, ilen);
 503         }
 504
 505         return len;
 506 }
 507
 508 /**
 509  * Mix some frames from the given secondary buffer "dsb" into the device
 510  * primary buffer.
 511  *
 512  * dsb = the secondary buffer
 513  * playpos = the current play position in the device buffer (primary buffer)
 514  * writepos = the current safe-to-write position in the device buffer
 515  * mixlen = the maximum number of bytes in the primary buffer to mix, from the
 516  *          current writepos.
 517  *
 518  * Returns: the number of bytes beyond the writepos that were mixed.
 519  */
 520 static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, float *mix_buffer, DWORD writepos, DWORD mixlen)
 521 {
 522         DWORD primary_done = 0;
 523
 524         TRACE("(%p,%d,%d)\n",dsb,writepos,mixlen);
 525         TRACE("writepos=%d, mixlen=%d\n", writepos, mixlen);
 526         TRACE("looping=%d, leadin=%d\n", dsb->playflags, dsb->leadin);
 527
 528         /* If leading in, only mix about 20 ms, and 'skip' mixing the rest, for more fluid pointer advancement */
 529         /* FIXME: Is this needed? */
 530         if (dsb->leadin && dsb->state == STATE_STARTING) {
 531                 if (mixlen > 2 * dsb->device->fraglen) {
 532                         primary_done = mixlen - 2 * dsb->device->fraglen;
 533                         mixlen = 2 * dsb->device->fraglen;
 534                         writepos += primary_done;
 535                         dsb->sec_mixpos += (primary_done / dsb->device->pwfx->nBlockAlign) *
 536                                 dsb->pwfx->nBlockAlign * dsb->freqAdjustNum / dsb->freqAdjustDen;
 537                 }
 538         }
 539
 540         dsb->leadin = FALSE;
 541
 542         TRACE("mixlen (primary) = %i\n", mixlen);
 543
 544         /* First try to mix to the end of the buffer if possible
 545          * Theoretically it would allow for better optimization
 546         */
 547         primary_done += DSOUND_MixInBuffer(dsb, mix_buffer, writepos, mixlen);
 548
 549         TRACE("total mixed data=%d\n", primary_done);
 550
 551         /* Report back the total prebuffered amount for this buffer */
 552         return primary_done;
 553 }
 554
 555 /**
 556  * For a DirectSoundDevice, go through all the currently playing buffers and
 557  * mix them in to the device buffer.
 558  *
 559  * writepos = the current safe-to-write position in the primary buffer
 560  * mixlen = the maximum amount to mix into the primary buffer
 561  *          (beyond the current writepos)
 562  * all_stopped = reports back if all buffers have stopped
 563  *
 564  * Returns:  the length beyond the writepos that was mixed to.
 565  */
 566
 567 static void DSOUND_MixToPrimary(const DirectSoundDevice *device, float *mix_buffer, DWORD writepos, DWORD mixlen, BOOL *all_stopped)
 568 {
 569         INT i;
 570         IDirectSoundBufferImpl  *dsb;
 571
 572         /* unless we find a running buffer, all have stopped */
 573         *all_stopped = TRUE;
 574
 575         TRACE("(%d,%d)\n", writepos, mixlen);
 576         for (i = 0; i < device->nrofbuffers; i++) {
 577                 dsb = device->buffers[i];
 578
 579                 TRACE("MixToPrimary for %p, state=%d\n", dsb, dsb->state);
 580
 581                 if (dsb->buflen && dsb->state) {
 582                         TRACE("Checking %p, mixlen=%d\n", dsb, mixlen);
 583                         RtlAcquireResourceShared(&dsb->lock, TRUE);
 584                         /* if buffer is stopping it is stopped now */
 585                         if (dsb->state == STATE_STOPPING) {
 586                                 dsb->state = STATE_STOPPED;
 587                                 DSOUND_CheckEvent(dsb, 0, 0);
 588                         } else if (dsb->state != STATE_STOPPED) {
 589
 590                                 /* if the buffer was starting, it must be playing now */
 591                                 if (dsb->state == STATE_STARTING)
 592                                         dsb->state = STATE_PLAYING;
 593
 594                                 /* mix next buffer into the main buffer */
 595                                 DSOUND_MixOne(dsb, mix_buffer, writepos, mixlen);
 596
 597                                 *all_stopped = FALSE;
 598                         }
 599                         RtlReleaseResource(&dsb->lock);
 600                 }
 601         }
 602 }
 603
 604 /**
 605  * Add buffers to the emulated wave device system.
 606  *
 607  * device = The current dsound playback device
 608  * force = If TRUE, the function will buffer up as many frags as possible,
 609  *         even though and will ignore the actual state of the primary buffer.
 610  *
 611  * Returns:  None
 612  */
 613
 614 static void DSOUND_WaveQueue(DirectSoundDevice *device, LPBYTE pos, DWORD bytes)
 615 {
 616         BYTE *buffer;
 617         HRESULT hr;
 618
 619         TRACE("(%p)\n", device);
 620
 621         hr = IAudioRenderClient_GetBuffer(device->render, bytes / device->pwfx->nBlockAlign, &buffer);
 622         if(FAILED(hr)){
 623                 WARN("GetBuffer failed: %08x\n", hr);
 624                 return;
 625         }
 626
 627         memcpy(buffer, pos, bytes);
 628
 629         hr = IAudioRenderClient_ReleaseBuffer(device->render, bytes / device->pwfx->nBlockAlign, 0);
 630         if(FAILED(hr)) {
 631                 ERR("ReleaseBuffer failed: %08x\n", hr);
 632                 IAudioRenderClient_ReleaseBuffer(device->render, 0, 0);
 633                 return;
 634         }
 635
 636         device->pad += bytes;
 637 }
 638
 639 /**
 640  * Perform mixing for a Direct Sound device. That is, go through all the
 641  * secondary buffers (the sound bites currently playing) and mix them in
 642  * to the primary buffer (the device buffer).
 643  *
 644  * The mixing procedure goes:
 645  *
 646  * secondary->buffer (secondary format)
 647  *   =[Resample]=> device->tmp_buffer (float format)
 648  *   =[Volume]=> device->tmp_buffer (float format)
 649  *   =[Reformat]=> device->buffer (device format, skipped on float)
 650  */
 651 static void DSOUND_PerformMix(DirectSoundDevice *device)
 652 {
 653         UINT32 pad, maxq, writepos;
 654         DWORD block;
 655         HRESULT hr;
 656
 657         TRACE("(%p)\n", device);
 658
 659         /* **** */
 660         EnterCriticalSection(&device->mixlock);
 661
 662         hr = IAudioClient_GetCurrentPadding(device->client, &pad);
 663         if(FAILED(hr)){
 664                 WARN("GetCurrentPadding failed: %08x\n", hr);
 665                 LeaveCriticalSection(&device->mixlock);
 666                 return;
 667         }
 668         block = device->pwfx->nBlockAlign;
 669         pad *= block;
 670         device->playpos += device->pad - pad;
 671         device->playpos %= device->buflen;
 672         device->pad = pad;
 673
 674         maxq = device->aclen - pad;
 675         if(!maxq){
 676                 /* nothing to do! */
 677                 LeaveCriticalSection(&device->mixlock);
 678                 return;
 679         }
 680         if (maxq > device->fraglen * 3)
 681                 maxq = device->fraglen * 3;
 682
 683         writepos = (device->playpos + pad) % device->buflen;
 684
 685         if (device->priolevel != DSSCL_WRITEPRIMARY) {
 686                 BOOL all_stopped = FALSE;
 687                 int nfiller;
 688                 void *buffer = NULL;
 689
 690                 /* the sound of silence */
 691                 nfiller = device->pwfx->wBitsPerSample == 8 ? 128 : 0;
 692
 693                 /* check for underrun. underrun occurs when the write position passes the mix position
 694                  * also wipe out just-played sound data */
 695                 if (!pad)
 696                         WARN("Probable buffer underrun\n");
 697
 698                 hr = IAudioRenderClient_GetBuffer(device->render, maxq / block, (void*)&buffer);
 699                 if(FAILED(hr)){
 700                         WARN("GetBuffer failed: %08x\n", hr);
 701                         LeaveCriticalSection(&device->mixlock);
 702                         return;
 703                 }
 704
 705                 memset(buffer, nfiller, maxq);
 706
 707                 if (!device->normfunction)
 708                         DSOUND_MixToPrimary(device, buffer, writepos, maxq, &all_stopped);
 709                 else {
 710
 711                         /* do the mixing */
 712                         DSOUND_MixToPrimary(device, (float*)device->buffer, writepos, maxq, &all_stopped);
 713
 714                         device->normfunction(device->buffer, buffer, maxq);
 715                 }
 716
 717                 hr = IAudioRenderClient_ReleaseBuffer(device->render, maxq / block, 0);
 718                 if(FAILED(hr))
 719                         ERR("ReleaseBuffer failed: %08x\n", hr);
 720
 721                 device->pad += maxq;
 722         } else if (!device->stopped) {
 723                 if (maxq > device->buflen)
 724                         maxq = device->buflen;
 725                 if (writepos + maxq > device->buflen) {
 726                         DSOUND_WaveQueue(device, device->buffer + writepos, device->buflen - writepos);
 727                         DSOUND_WaveQueue(device, device->buffer, writepos + maxq - device->buflen);
 728                 } else
 729                         DSOUND_WaveQueue(device, device->buffer + writepos, maxq);
 730         }
 731
 732         LeaveCriticalSection(&(device->mixlock));
 733         /* **** */
 734 }
 735
 736 DWORD CALLBACK DSOUND_mixthread(void *p)
 737 {
 738         DirectSoundDevice *dev = p;
 739         TRACE("(%p)\n", dev);
 740
 741         while (dev->ref) {
 742                 DWORD ret;
 743
 744                 /*
 745                  * Some audio drivers are retarded and won't fire after being
 746                  * stopped, add a timeout to handle this.
 747                  */
 748                 ret = WaitForSingleObject(dev->sleepev, dev->sleeptime);
 749                 if (ret == WAIT_FAILED)
 750                         WARN("wait returned error %u %08x!\n", GetLastError(), GetLastError());
 751                 else if (ret != WAIT_OBJECT_0)
 752                         WARN("wait returned %08x!\n", ret);
 753                 if (!dev->ref)
 754                         break;
 755
 756                 RtlAcquireResourceShared(&(dev->buffer_list_lock), TRUE);
 757                 DSOUND_PerformMix(dev);
 758                 RtlReleaseResource(&(dev->buffer_list_lock));
 759         }
 760         SetEvent(dev->thread_finished);
 761         return 0;
 762 }