Logo Search packages:      
Sourcecode: wine version File versions

mixer.c

/*                DirectSound
 *
 * Copyright 1998 Marcus Meissner
 * Copyright 1998 Rob Riggs
 * Copyright 2000-2002 TransGaming Technologies, Inc.
 * Copyright 2007 Peter Dons Tychsen
 * Copyright 2007 Maarten Lankhorst
 *
 * This library is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * This library is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with this library; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
 */

#include <assert.h>
#include <stdarg.h>
#include <math.h> /* Insomnia - pow() function */

#define NONAMELESSSTRUCT
#define NONAMELESSUNION
#include "windef.h"
#include "winbase.h"
#include "winuser.h"
#include "mmsystem.h"
#include "winternl.h"
#include "wine/debug.h"
#include "dsound.h"
#include "dsdriver.h"
#include "dsound_private.h"

WINE_DEFAULT_DEBUG_CHANNEL(dsound);

void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan)
{
      double temp;
      TRACE("(%p)\n",volpan);

      TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
      /* the AmpFactors are expressed in 16.16 fixed point */
      volpan->dwVolAmpFactor = (ULONG) (pow(2.0, volpan->lVolume / 600.0) * 0xffff);
      /* FIXME: dwPan{Left|Right}AmpFactor */

      /* FIXME: use calculated vol and pan ampfactors */
      temp = (double) (volpan->lVolume - (volpan->lPan > 0 ? volpan->lPan : 0));
      volpan->dwTotalLeftAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
      temp = (double) (volpan->lVolume + (volpan->lPan < 0 ? volpan->lPan : 0));
      volpan->dwTotalRightAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);

      TRACE("left = %x, right = %x\n", volpan->dwTotalLeftAmpFactor, volpan->dwTotalRightAmpFactor);
}

void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan)
{
    double left,right;
    TRACE("(%p)\n",volpan);

    TRACE("left=%x, right=%x\n",volpan->dwTotalLeftAmpFactor,volpan->dwTotalRightAmpFactor);
    if (volpan->dwTotalLeftAmpFactor==0)
        left=-10000;
    else
        left=600 * log(((double)volpan->dwTotalLeftAmpFactor) / 0xffff) / log(2);
    if (volpan->dwTotalRightAmpFactor==0)
        right=-10000;
    else
        right=600 * log(((double)volpan->dwTotalRightAmpFactor) / 0xffff) / log(2);
    if (left<right)
    {
        volpan->lVolume=right;
        volpan->dwVolAmpFactor=volpan->dwTotalRightAmpFactor;
    }
    else
    {
        volpan->lVolume=left;
        volpan->dwVolAmpFactor=volpan->dwTotalLeftAmpFactor;
    }
    if (volpan->lVolume < -10000)
        volpan->lVolume=-10000;
    volpan->lPan=right-left;
    if (volpan->lPan < -10000)
        volpan->lPan=-10000;

    TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
}

/* NOTE: Not all secpos have to always be mapped to a bufpos, other way around is always the case
 * DWORD64 is used here because a single DWORD wouldn't be big enough to fit the freqAcc for big buffers
 */
/** This function converts a 'native' sample pointer to a resampled pointer that fits for primary
 * secmixpos is used to decide which freqAcc is needed
 * overshot tells what the 'actual' secpos is now (optional)
 */
DWORD DSOUND_secpos_to_bufpos(const IDirectSoundBufferImpl *dsb, DWORD secpos, DWORD secmixpos, DWORD* overshot)
{
      DWORD64 framelen = secpos / dsb->pwfx->nBlockAlign;
      DWORD64 freqAdjust = dsb->freqAdjust;
      DWORD64 acc, freqAcc;

      if (secpos < secmixpos)
            freqAcc = dsb->freqAccNext;
      else freqAcc = dsb->freqAcc;
      acc = (framelen << DSOUND_FREQSHIFT) + (freqAdjust - 1 - freqAcc);
      acc /= freqAdjust;
      if (overshot)
      {
            DWORD64 oshot = acc * freqAdjust + freqAcc;
            assert(oshot >= framelen << DSOUND_FREQSHIFT);
            oshot -= framelen << DSOUND_FREQSHIFT;
            *overshot = (DWORD)oshot;
            assert(*overshot < dsb->freqAdjust);
      }
      return (DWORD)acc * dsb->device->pwfx->nBlockAlign;
}

/** Convert a resampled pointer that fits for primary to a 'native' sample pointer
 * freqAccNext is used here rather than freqAcc: In case the app wants to fill up to
 * the play position it won't overwrite it
 */
static DWORD DSOUND_bufpos_to_secpos(const IDirectSoundBufferImpl *dsb, DWORD bufpos)
{
      DWORD oAdv = dsb->device->pwfx->nBlockAlign, iAdv = dsb->pwfx->nBlockAlign, pos;
      DWORD64 framelen;
      DWORD64 acc;

      framelen = bufpos/oAdv;
      acc = framelen * (DWORD64)dsb->freqAdjust + (DWORD64)dsb->freqAccNext;
      acc = acc >> DSOUND_FREQSHIFT;
      pos = (DWORD)acc * iAdv;
      if (pos >= dsb->buflen)
            /* Because of differences between freqAcc and freqAccNext, this might happen */
            pos = dsb->buflen - iAdv;
      TRACE("Converted %d/%d to %d/%d\n", bufpos, dsb->tmp_buffer_len, pos, dsb->buflen);
      return pos;
}

/**
 * Move freqAccNext to freqAcc, and find new values for buffer length and freqAccNext
 */
static void DSOUND_RecalcFreqAcc(IDirectSoundBufferImpl *dsb)
{
      if (!dsb->freqneeded) return;
      dsb->freqAcc = dsb->freqAccNext;
      dsb->tmp_buffer_len = DSOUND_secpos_to_bufpos(dsb, dsb->buflen, 0, &dsb->freqAccNext);
      TRACE("New freqadjust: %04x, new buflen: %d\n", dsb->freqAccNext, dsb->tmp_buffer_len);
}

/**
 * Recalculate the size for temporary buffer, and new writelead
 * Should be called when one of the following things occur:
 * - Primary buffer format is changed
 * - This buffer format (frequency) is changed
 *
 * After this, DSOUND_MixToTemporary(dsb, 0, dsb->buflen) should
 * be called to refill the temporary buffer with data.
 */
void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
{
      BOOL needremix = TRUE, needresample = (dsb->freq != dsb->device->pwfx->nSamplesPerSec);
      DWORD bAlign = dsb->pwfx->nBlockAlign, pAlign = dsb->device->pwfx->nBlockAlign;

      TRACE("(%p)\n",dsb);

      /* calculate the 10ms write lead */
      dsb->writelead = (dsb->freq / 100) * dsb->pwfx->nBlockAlign;

      if ((dsb->pwfx->wBitsPerSample == dsb->device->pwfx->wBitsPerSample) &&
          (dsb->pwfx->nChannels == dsb->device->pwfx->nChannels) && !needresample)
            needremix = FALSE;
      HeapFree(GetProcessHeap(), 0, dsb->tmp_buffer);
      dsb->tmp_buffer = NULL;
      dsb->max_buffer_len = dsb->freqAcc = dsb->freqAccNext = 0;
      dsb->freqneeded = needresample;

      if (needremix)
      {
            if (needresample)
                  DSOUND_RecalcFreqAcc(dsb);
            else
                  dsb->tmp_buffer_len = dsb->buflen / bAlign * pAlign;
            dsb->max_buffer_len = dsb->tmp_buffer_len;
            dsb->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, dsb->max_buffer_len);
            FillMemory(dsb->tmp_buffer, dsb->tmp_buffer_len, dsb->device->pwfx->wBitsPerSample == 8 ? 128 : 0);
      }
      else dsb->max_buffer_len = dsb->tmp_buffer_len = dsb->buflen;
      dsb->buf_mixpos = DSOUND_secpos_to_bufpos(dsb, dsb->sec_mixpos, 0, NULL);
}

/**
 * Check for application callback requests for when the play position
 * reaches certain points.
 *
 * The offsets that will be triggered will be those between the recorded
 * "last played" position for the buffer (i.e. dsb->playpos) and "len" bytes
 * beyond that position.
 */
void DSOUND_CheckEvent(const IDirectSoundBufferImpl *dsb, DWORD playpos, int len)
{
      int               i;
      DWORD             offset;
      LPDSBPOSITIONNOTIFY     event;
      TRACE("(%p,%d)\n",dsb,len);

      if (dsb->nrofnotifies == 0)
            return;

      TRACE("(%p) buflen = %d, playpos = %d, len = %d\n",
            dsb, dsb->buflen, playpos, len);
      for (i = 0; i < dsb->nrofnotifies ; i++) {
            event = dsb->notifies + i;
            offset = event->dwOffset;
            TRACE("checking %d, position %d, event = %p\n",
                  i, offset, event->hEventNotify);
            /* DSBPN_OFFSETSTOP has to be the last element. So this is */
            /* OK. [Inside DirectX, p274] */
            /*  */
            /* This also means we can't sort the entries by offset, */
            /* because DSBPN_OFFSETSTOP == -1 */
            if (offset == DSBPN_OFFSETSTOP) {
                  if (dsb->state == STATE_STOPPED) {
                        SetEvent(event->hEventNotify);
                        TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
                        return;
                  } else
                        return;
            }
            if ((playpos + len) >= dsb->buflen) {
                  if ((offset < ((playpos + len) % dsb->buflen)) ||
                      (offset >= playpos)) {
                        TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
                        SetEvent(event->hEventNotify);
                  }
            } else {
                  if ((offset >= playpos) && (offset < (playpos + len))) {
                        TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
                        SetEvent(event->hEventNotify);
                  }
            }
      }
}

/* WAV format info can be found at:
 *
 *    http://www.cwi.nl/ftp/audio/AudioFormats.part2
 *    ftp://ftp.cwi.nl/pub/audio/RIFF-format
 *
 * Import points to remember:
 *    8-bit WAV is unsigned
 *    16-bit WAV is signed
 */
 /* Use the same formulas as pcmconverter.c */
static inline INT16 cvtU8toS16(BYTE b)
{
    return (short)((b+(b << 8))-32768);
}

static inline BYTE cvtS16toU8(INT16 s)
{
    return (s >> 8) ^ (unsigned char)0x80;
}

/**
 * Copy a single frame from the given input buffer to the given output buffer.
 * Translate 8 <-> 16 bits and mono <-> stereo
 */
static inline void cp_fields(const IDirectSoundBufferImpl *dsb, const BYTE *ibuf, BYTE *obuf )
{
      DirectSoundDevice * device = dsb->device;
        INT fl,fr;

        if (dsb->pwfx->wBitsPerSample == 8)  {
                if (device->pwfx->wBitsPerSample == 8 &&
                    device->pwfx->nChannels == dsb->pwfx->nChannels) {
                        /* avoid needless 8->16->8 conversion */
                        *obuf=*ibuf;
                        if (dsb->pwfx->nChannels==2)
                                *(obuf+1)=*(ibuf+1);
                        return;
                }
                fl = cvtU8toS16(*ibuf);
                fr = (dsb->pwfx->nChannels==2 ? cvtU8toS16(*(ibuf + 1)) : fl);
        } else {
                fl = *((const INT16 *)ibuf);
                fr = (dsb->pwfx->nChannels==2 ? *(((const INT16 *)ibuf) + 1)  : fl);
        }

        if (device->pwfx->nChannels == 2) {
                if (device->pwfx->wBitsPerSample == 8) {
                        *obuf = cvtS16toU8(fl);
                        *(obuf + 1) = cvtS16toU8(fr);
                        return;
                }
                if (device->pwfx->wBitsPerSample == 16) {
                        *((INT16 *)obuf) = fl;
                        *(((INT16 *)obuf) + 1) = fr;
                        return;
                }
        }
        if (device->pwfx->nChannels == 1) {
                fl = (fl + fr) >> 1;
                if (device->pwfx->wBitsPerSample == 8) {
                        *obuf = cvtS16toU8(fl);
                        return;
                }
                if (device->pwfx->wBitsPerSample == 16) {
                        *((INT16 *)obuf) = fl;
                        return;
                }
        }
}

/**
 * Calculate the distance between two buffer offsets, taking wraparound
 * into account.
 */
static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
{
/* If these asserts fail, the problem is not here, but in the underlying code */
      assert(ptr1 < buflen);
      assert(ptr2 < buflen);
      if (ptr1 >= ptr2) {
            return ptr1 - ptr2;
      } else {
            return buflen + ptr1 - ptr2;
      }
}
/**
 * Mix at most the given amount of data into the allocated temporary buffer
 * of the given secondary buffer, starting from the dsb's first currently
 * unsampled frame (writepos), translating frequency (pitch), stereo/mono
 * and bits-per-sample so that it is ideal for the primary buffer.
 * Doesn't perform any mixing - this is a straight copy/convert operation.
 *
 * dsb = the secondary buffer
 * writepos = Starting position of changed buffer
 * len = number of bytes to resample from writepos
 *
 * NOTE: writepos + len <= buflen, This function doesn't loop!
 */
void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD len)
{
      INT   i, size;
      BYTE  *ibp, *obp, *ibp_begin, *obp_begin;
      INT   iAdvance = dsb->pwfx->nBlockAlign;
      INT   oAdvance = dsb->device->pwfx->nBlockAlign;
      DWORD freqAcc, target_writepos, overshot;

      if (!dsb->tmp_buffer)
            /* Nothing to do, already ideal format */
            return;

      ibp = dsb->buffer->memory + writepos;
      ibp_begin = dsb->buffer->memory;
      obp_begin = dsb->tmp_buffer;

      TRACE("(%p, %p)\n", dsb, ibp);
      /* Check for the best case */
      if ((dsb->freq == dsb->device->pwfx->nSamplesPerSec) &&
          (dsb->pwfx->wBitsPerSample == dsb->device->pwfx->wBitsPerSample) &&
          (dsb->pwfx->nChannels == dsb->device->pwfx->nChannels)) {
            obp = dsb->tmp_buffer + writepos;
            /* Why would we need a temporary buffer if we do best case? */
            FIXME("(%p) Why do we resample for best case??? Bad!!\n", dsb);
            CopyMemory(obp, ibp, len);
            return;
      }

      /* Check for same sample rate */
      if (dsb->freq == dsb->device->pwfx->nSamplesPerSec) {
            TRACE("(%p) Same sample rate %d = primary %d\n", dsb,
                  dsb->freq, dsb->device->pwfx->nSamplesPerSec);
            obp = dsb->tmp_buffer + writepos/iAdvance*oAdvance;
            for (i = 0; i < len; i += iAdvance) {
                  cp_fields(dsb, ibp, obp);
                  ibp += iAdvance;
                  obp += oAdvance;
            }
            return;
      }

      /* Mix in different sample rates */
      TRACE("(%p) Adjusting frequency: %d -> %d\n", dsb, dsb->freq, dsb->device->pwfx->nSamplesPerSec);
      size = len / iAdvance;

      target_writepos = DSOUND_secpos_to_bufpos(dsb, writepos, dsb->sec_mixpos, &freqAcc);
      overshot = freqAcc >> DSOUND_FREQSHIFT;
      if (overshot)
      {
            if (overshot >= size)
                  return;
            size -= overshot;
            writepos += overshot * iAdvance;
            if (writepos >= dsb->buflen)
                  return;
            ibp = dsb->buffer->memory + writepos;
            freqAcc &= (1 << DSOUND_FREQSHIFT) - 1;
            TRACE("Overshot: %d, freqAcc: %04x\n", overshot, freqAcc);
      }

      obp = dsb->tmp_buffer + target_writepos;
      /* FIXME: Small problem here when we're overwriting buf_mixpos, it then STILL uses old freqAcc, not sure if it matters or not */
      while (size > 0) {
            cp_fields(dsb, ibp, obp);
            obp += oAdvance;
            freqAcc += dsb->freqAdjust;
            if (freqAcc >= (1<<DSOUND_FREQSHIFT)) {
                  ULONG adv = (freqAcc>>DSOUND_FREQSHIFT);
                  freqAcc &= (1<<DSOUND_FREQSHIFT)-1;
                  ibp += adv * iAdvance;
                  size -= adv;
            }
      }
}

/** Apply volume to the given soundbuffer from (primary) position writepos and length len
 * Returns: NULL if no volume needs to be applied
 * or else a memory handle that holds 'len' volume adjusted buffer */
static LPBYTE DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, DWORD writepos, INT len)
{
      INT   i;
      BYTE  *bpc;
      INT16 *bps, *mems;
      DWORD vLeft, vRight;
      INT nChannels = dsb->device->pwfx->nChannels;
      LPBYTE mem = (dsb->tmp_buffer ? dsb->tmp_buffer : dsb->buffer->memory)+writepos;

      TRACE("(%p,%d)\n",dsb,len);
      TRACE("left = %x, right = %x\n", dsb->volpan.dwTotalLeftAmpFactor,
            dsb->volpan.dwTotalRightAmpFactor);

      if (nChannels != 1 && nChannels != 2)
      {
            FIXME("There is no support for %d channels\n", nChannels);
            return NULL;
      }

      if (dsb->device->pwfx->wBitsPerSample != 8 && dsb->device->pwfx->wBitsPerSample != 16)
      {
            FIXME("There is no support for %d bpp\n", dsb->device->pwfx->wBitsPerSample);
            return NULL;
      }

      if ((!(dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) || (dsb->volpan.lPan == 0)) &&
          (!(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) || (dsb->volpan.lVolume == 0)) &&
           !(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
            return NULL; /* Nothing to do */

      if (dsb->device->tmp_buffer_len < len || !dsb->device->tmp_buffer)
      {
            dsb->device->tmp_buffer_len = len;
            if (dsb->device->tmp_buffer)
                  dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, len);
            else
                  dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, len);
      }
      bpc = dsb->device->tmp_buffer;
      bps = (INT16 *)bpc;
      mems = (INT16 *)mem;
      vLeft = dsb->volpan.dwTotalLeftAmpFactor;
      if (nChannels > 1)
            vRight = dsb->volpan.dwTotalRightAmpFactor;
      else
            vRight = vLeft;

      switch (dsb->device->pwfx->wBitsPerSample) {
      case 8:
            /* 8-bit WAV is unsigned, but we need to operate */
            /* on signed data for this to work properly */
            for (i = 0; i < len; i+=2) {
                  *(bpc++) = (((INT)(*(mem++) - 128) * vLeft) >> 16) + 128;
                  *(bpc++) = (((INT)(*(mem++) - 128) * vRight) >> 16) + 128;
            }
            if (len % 2 == 1 && nChannels == 1)
                  *(bpc++) = (((INT)(*(mem++) - 128) * vLeft) >> 16) + 128;
            break;
      case 16:
            /* 16-bit WAV is signed -- much better */
            for (i = 0; i < len; i += 4) {
                  *(bps++) = (*(mems++) * vLeft) >> 16;
                  *(bps++) = (*(mems++) * vRight) >> 16;
            }
            if (len % 4 == 2 && nChannels == 1)
                  *(bps++) = ((INT)*(mems++) * vLeft) >> 16;
            break;
      }
      return dsb->device->tmp_buffer;
}

/**
 * Mix (at most) the given number of bytes into the given position of the
 * device buffer, from the secondary buffer "dsb" (starting at the current
 * mix position for that buffer).
 *
 * Returns the number of bytes actually mixed into the device buffer. This
 * will match fraglen unless the end of the secondary buffer is reached
 * (and it is not looping).
 *
 * dsb  = the secondary buffer to mix from
 * writepos = position (offset) in device buffer to write at
 * fraglen = number of bytes to mix
 */
static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen)
{
      INT i, len = fraglen, field, todo, ilen;
      BYTE *ibuf = (dsb->tmp_buffer ? dsb->tmp_buffer : dsb->buffer->memory) + dsb->buf_mixpos, *volbuf;
      DWORD oldpos;

      TRACE("buf_mixpos=%d/%d sec_mixpos=%d/%d\n", dsb->buf_mixpos, dsb->tmp_buffer_len, dsb->sec_mixpos, dsb->buflen);
      TRACE("(%p,%d,%d)\n",dsb,writepos,fraglen);

      assert(dsb->buf_mixpos + len <= dsb->tmp_buffer_len);

      if (len % dsb->device->pwfx->nBlockAlign) {
            INT nBlockAlign = dsb->device->pwfx->nBlockAlign;
            ERR("length not a multiple of block size, len = %d, block size = %d\n", len, nBlockAlign);
            len -= len % nBlockAlign; /* data alignment */
      }

      /* Apply volume if needed */
      volbuf = DSOUND_MixerVol(dsb, dsb->buf_mixpos, len);
      if (volbuf)
            ibuf = volbuf;

      /* Now mix the temporary buffer into the devices main buffer */
      if (dsb->device->pwfx->wBitsPerSample == 8) {
            BYTE  *obuf = dsb->device->buffer + writepos;

            if ((writepos + len) <= dsb->device->buflen)
                  todo = len;
            else
                  todo = dsb->device->buflen - writepos;

            for (i = 0; i < todo; i++) {
                  /* 8-bit WAV is unsigned */
                  field = (*ibuf++ - 128);
                  field += (*obuf - 128);
                  if (field > 127) field = 127;
                  else if (field < -128) field = -128;
                  *obuf++ = field + 128;
            }
 
            if (todo < len) {
                  todo = len - todo;
                  obuf = dsb->device->buffer;

                  for (i = 0; i < todo; i++) {
                        /* 8-bit WAV is unsigned */
                        field = (*ibuf++ - 128);
                        field += (*obuf - 128);
                        if (field > 127) field = 127;
                        else if (field < -128) field = -128;
                        *obuf++ = field + 128;
                  }
            }
      } else {
            INT16 *ibufs, *obufs;

            ibufs = (INT16 *) ibuf;
            obufs = (INT16 *)(dsb->device->buffer + writepos);

            if ((writepos + len) <= dsb->device->buflen)
                  todo = len / 2;
            else
                  todo = (dsb->device->buflen - writepos) / 2;

            for (i = 0; i < todo; i++) {
                  /* 16-bit WAV is signed */
                  field = *ibufs++;

                  field += *obufs;
                  if (field > 32767) field = 32767;
                  else if (field < -32768) field = -32768;
                  *obufs++ = field;
            }

            if (todo < (len / 2)) {
                  todo = (len / 2) - todo;
                  obufs = (INT16 *)dsb->device->buffer;

                  for (i = 0; i < todo; i++) {
                        /* 16-bit WAV is signed */
                        field = *ibufs++;
                        field += *obufs;
                        if (field > 32767) field = 32767;
                        else if (field < -32768) field = -32768;
                        *obufs++ = field;
                  }
            }
      }

      oldpos = dsb->sec_mixpos;
      dsb->buf_mixpos += len;

      if (dsb->buf_mixpos >= dsb->tmp_buffer_len) {
            if (dsb->playflags & DSBPLAY_LOOPING) {
                  dsb->buf_mixpos -= dsb->tmp_buffer_len;
            } else if (dsb->buf_mixpos >= dsb->tmp_buffer_len) {
                  if (dsb->buf_mixpos > dsb->tmp_buffer_len)
                        ERR("Mixpos (%u) past buflen (%u), capping...\n", dsb->buf_mixpos, dsb->tmp_buffer_len);
                  dsb->buf_mixpos = dsb->sec_mixpos = 0;
                  dsb->state = STATE_STOPPED;
            }
            DSOUND_RecalcFreqAcc(dsb);
      }

      dsb->sec_mixpos = DSOUND_bufpos_to_secpos(dsb, dsb->buf_mixpos);
      ilen = DSOUND_BufPtrDiff(dsb->buflen, dsb->sec_mixpos, oldpos);
      /* check for notification positions */
      if (dsb->dsbd.dwFlags & DSBCAPS_CTRLPOSITIONNOTIFY &&
          dsb->state != STATE_STARTING) {
            DSOUND_CheckEvent(dsb, oldpos, ilen);
      }

      /* increase mix position */
      dsb->primary_mixpos += len;
      if (dsb->primary_mixpos >= dsb->device->buflen)
            dsb->primary_mixpos -= dsb->device->buflen;
      return len;
}

/**
 * Mix some frames from the given secondary buffer "dsb" into the device
 * primary buffer.
 *
 * dsb = the secondary buffer
 * playpos = the current play position in the device buffer (primary buffer)
 * writepos = the current safe-to-write position in the device buffer
 * mixlen = the maximum number of bytes in the primary buffer to mix, from the
 *          current writepos.
 *
 * Returns: the number of bytes beyond the writepos that were mixed.
 */
static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD mixlen)
{
      /* The buffer's primary_mixpos may be before or after the the device
       * buffer's mixpos, but both must be ahead of writepos. */
      DWORD primary_done;

      TRACE("(%p,%d,%d)\n",dsb,writepos,mixlen);
      TRACE("writepos=%d, buf_mixpos=%d, primary_mixpos=%d, mixlen=%d\n", writepos, dsb->buf_mixpos, dsb->primary_mixpos, mixlen);
      TRACE("looping=%d, leadin=%d, buflen=%d\n", dsb->playflags, dsb->leadin, dsb->tmp_buffer_len);

      /* If leading in, only mix about 20 ms, and 'skip' mixing the rest, for more fluid pointer advancement */
      if (dsb->leadin && dsb->state == STATE_STARTING)
      {
            if (mixlen > 2 * dsb->device->fraglen)
            {
                  dsb->primary_mixpos += mixlen - 2 * dsb->device->fraglen;
                  dsb->primary_mixpos %= dsb->device->buflen;
            }
      }
      dsb->leadin = FALSE;

      /* calculate how much pre-buffering has already been done for this buffer */
      primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);

      /* sanity */
      if(mixlen < primary_done)
      {
            /* Should *NEVER* happen */
            ERR("Fatal error. Under/Overflow? primary_done=%d, mixpos=%d/%d (%d/%d), primary_mixpos=%d, writepos=%d, mixlen=%d\n", primary_done,dsb->buf_mixpos,dsb->tmp_buffer_len,dsb->sec_mixpos, dsb->buflen, dsb->primary_mixpos, writepos, mixlen);
            return 0;
      }

      /* take into acount already mixed data */
      mixlen -= primary_done;

      TRACE("primary_done=%d, mixlen (primary) = %i\n", primary_done, mixlen);

      if (!mixlen)
            return 0;

      /* First try to mix to the end of the buffer if possible
       * Theoretically it would allow for better optimization
      */
      if (mixlen + dsb->buf_mixpos >= dsb->tmp_buffer_len)
      {
            DWORD newmixed, mixfirst = dsb->tmp_buffer_len - dsb->buf_mixpos;
            newmixed = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixfirst);
            mixlen -= newmixed;

            if (dsb->playflags & DSBPLAY_LOOPING)
                  while (newmixed && mixlen)
                  {
                        mixfirst = (dsb->tmp_buffer_len < mixlen ? dsb->tmp_buffer_len : mixlen);
                        newmixed = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixfirst);
                        mixlen -= newmixed;
                  }
      }
      else DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixlen);

      /* re-calculate the primary done */
      primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);

      TRACE("new primary_mixpos=%d, total mixed data=%d\n", dsb->primary_mixpos, primary_done);

      /* Report back the total prebuffered amount for this buffer */
      return primary_done;
}

/**
 * For a DirectSoundDevice, go through all the currently playing buffers and
 * mix them in to the device buffer.
 *
 * writepos = the current safe-to-write position in the primary buffer
 * mixlen = the maximum amount to mix into the primary buffer
 *          (beyond the current writepos)
 * mustlock = Do we have to fight for lock because we otherwise risk an underrun?
 * recover = true if the sound device may have been reset and the write
 *           position in the device buffer changed
 * all_stopped = reports back if all buffers have stopped
 *
 * Returns:  the length beyond the writepos that was mixed to.
 */

static DWORD DSOUND_MixToPrimary(const DirectSoundDevice *device, DWORD writepos, DWORD mixlen, BOOL mustlock, BOOL recover, BOOL *all_stopped)
{
      INT i, len;
      DWORD minlen = 0;
      IDirectSoundBufferImpl  *dsb;
      BOOL gotall = TRUE;

      /* unless we find a running buffer, all have stopped */
      *all_stopped = TRUE;

      TRACE("(%d,%d,%d)\n", writepos, mixlen, recover);
      for (i = 0; i < device->nrofbuffers; i++) {
            dsb = device->buffers[i];

            TRACE("MixToPrimary for %p, state=%d\n", dsb, dsb->state);

            if (dsb->buflen && dsb->state && !dsb->hwbuf) {
                  TRACE("Checking %p, mixlen=%d\n", dsb, mixlen);
                  if (!RtlAcquireResourceShared(&dsb->lock, mustlock))
                  {
                        gotall = FALSE;
                        continue;
                  }
                  /* if buffer is stopping it is stopped now */
                  if (dsb->state == STATE_STOPPING) {
                        dsb->state = STATE_STOPPED;
                        DSOUND_CheckEvent(dsb, 0, 0);
                  } else if (dsb->state != STATE_STOPPED) {

                        /* if recovering, reset the mix position */
                        if ((dsb->state == STATE_STARTING) || recover) {
                              dsb->primary_mixpos = writepos;
                        }

                        /* mix next buffer into the main buffer */
                        len = DSOUND_MixOne(dsb, writepos, mixlen);

                        /* if the buffer was starting, it must be playing now */
                        if (dsb->state == STATE_STARTING)
                              dsb->state = STATE_PLAYING;

                        if (!minlen) minlen = len;

                        /* record the minimum length mixed from all buffers */
                        /* we only want to return the length which *all* buffers have mixed */
                        else if (len) minlen = (len < minlen) ? len : minlen;

                        *all_stopped = FALSE;
                  }
                  RtlReleaseResource(&dsb->lock);
            }
      }

      TRACE("Mixed at least %d from all buffers\n", minlen);
      if (!gotall) return 0;
      return minlen;
}

/**
 * Add buffers to the emulated wave device system.
 *
 * device = The current dsound playback device
 * force = If TRUE, the function will buffer up as many frags as possible,
 *         even though and will ignore the actual state of the primary buffer.
 *
 * Returns:  None
 */

static void DSOUND_WaveQueue(DirectSoundDevice *device, BOOL force)
{
      DWORD prebuf_frags, wave_writepos, wave_fragpos, i;
      TRACE("(%p)\n", device);

      /* calculate the current wave frag position */
      wave_fragpos = (device->pwplay + device->pwqueue) % device->helfrags;

      /* calculte the current wave write position */
      wave_writepos = wave_fragpos * device->fraglen;

      TRACE("wave_fragpos = %i, wave_writepos = %i, pwqueue = %i, prebuf = %i\n",
            wave_fragpos, wave_writepos, device->pwqueue, device->prebuf);

      if (!force)
      {
            /* check remaining prebuffered frags */
            prebuf_frags = device->mixpos / device->fraglen;
            if (prebuf_frags == device->helfrags)
                  --prebuf_frags;
            TRACE("wave_fragpos = %d, mixpos_frags = %d\n", wave_fragpos, prebuf_frags);
            if (prebuf_frags < wave_fragpos)
                  prebuf_frags += device->helfrags;
            prebuf_frags -= wave_fragpos;
            TRACE("wanted prebuf_frags = %d\n", prebuf_frags);
      }
      else
            /* buffer the maximum amount of frags */
            prebuf_frags = device->prebuf;

      /* limit to the queue we have left */
      if ((prebuf_frags + device->pwqueue) > device->prebuf)
            prebuf_frags = device->prebuf - device->pwqueue;

      TRACE("prebuf_frags = %i\n", prebuf_frags);

      /* adjust queue */
      device->pwqueue += prebuf_frags;

      /* get out of CS when calling the wave system */
      LeaveCriticalSection(&(device->mixlock));
      /* **** */

      /* queue up the new buffers */
      for(i=0; i<prebuf_frags; i++){
            TRACE("queueing wave buffer %i\n", wave_fragpos);
            waveOutWrite(device->hwo, &device->pwave[wave_fragpos], sizeof(WAVEHDR));
            wave_fragpos++;
            wave_fragpos %= device->helfrags;
      }

      /* **** */
      EnterCriticalSection(&(device->mixlock));

      TRACE("queue now = %i\n", device->pwqueue);
}

/**
 * Perform mixing for a Direct Sound device. That is, go through all the
 * secondary buffers (the sound bites currently playing) and mix them in
 * to the primary buffer (the device buffer).
 */
static void DSOUND_PerformMix(DirectSoundDevice *device)
{
      TRACE("(%p)\n", device);

      /* **** */
      EnterCriticalSection(&(device->mixlock));

      if (device->priolevel != DSSCL_WRITEPRIMARY) {
            BOOL recover = FALSE, all_stopped = FALSE;
            DWORD playpos, writepos, writelead, maxq, frag, prebuff_max, prebuff_left, size1, size2;
            LPVOID buf1, buf2;
            BOOL lock = (device->hwbuf && !(device->drvdesc.dwFlags & DSDDESC_DONTNEEDPRIMARYLOCK));
            BOOL mustlock = FALSE;
            int nfiller;

            /* the sound of silence */
            nfiller = device->pwfx->wBitsPerSample == 8 ? 128 : 0;

            /* get the position in the primary buffer */
            if (DSOUND_PrimaryGetPosition(device, &playpos, &writepos) != 0){
                  LeaveCriticalSection(&(device->mixlock));
                  return;
            }

            TRACE("primary playpos=%d, writepos=%d, clrpos=%d, mixpos=%d, buflen=%d\n",
                  playpos,writepos,device->playpos,device->mixpos,device->buflen);
            assert(device->playpos < device->buflen);

            /* wipe out just-played sound data */
            if (playpos < device->playpos) {
                  buf1 = device->buffer + device->playpos;
                  buf2 = device->buffer;
                  size1 = device->buflen - device->playpos;
                  size2 = playpos;
                  if (lock)
                        IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, device->playpos, size1+size2, 0);
                  FillMemory(buf1, size1, nfiller);
                  if (playpos && (!buf2 || !size2))
                        FIXME("%d: (%d, %d)=>(%d, %d) There should be an additional buffer here!!\n", __LINE__, device->playpos, device->mixpos, playpos, writepos);
                  FillMemory(buf2, size2, nfiller);
                  if (lock)
                        IDsDriverBuffer_Unlock(device->hwbuf, buf1, size1, buf2, size2);
            } else {
                  buf1 = device->buffer + device->playpos;
                  buf2 = NULL;
                  size1 = playpos - device->playpos;
                  size2 = 0;
                  if (lock)
                        IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, device->playpos, size1+size2, 0);
                  FillMemory(buf1, size1, nfiller);
                  if (buf2 && size2)
                  {
                        FIXME("%d: There should be no additional buffer here!!\n", __LINE__);
                        FillMemory(buf2, size2, nfiller);
                  }
                  if (lock)
                        IDsDriverBuffer_Unlock(device->hwbuf, buf1, size1, buf2, size2);
            }
            device->playpos = playpos;

            /* calc maximum prebuff */
            prebuff_max = (device->prebuf * device->fraglen);
            if (!device->hwbuf && playpos + prebuff_max >= device->helfrags * device->fraglen)
                  prebuff_max += device->buflen - device->helfrags * device->fraglen;

            /* check how close we are to an underrun. It occurs when the writepos overtakes the mixpos */
            prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);
            writelead = DSOUND_BufPtrDiff(device->buflen, writepos, playpos);

            /* find the maximum we can prebuffer from current write position */
            maxq = (writelead < prebuff_max) ? (prebuff_max - writelead) : 0;

            TRACE("prebuff_left = %d, prebuff_max = %dx%d=%d, writelead=%d\n",
                  prebuff_left, device->prebuf, device->fraglen, prebuff_max, writelead);

            /* check for underrun. underrun occurs when the write position passes the mix position */
            if((prebuff_left > prebuff_max) || (device->state == STATE_STOPPED) || (device->state == STATE_STARTING)){
                  if (device->state == STATE_STOPPING || device->state == STATE_PLAYING)
                        WARN("Probable buffer underrun\n");
                  else TRACE("Buffer starting or buffer underrun\n");

                  /* recover mixing for all buffers */
                  recover = TRUE;

                  /* reset mix position to write position */
                  device->mixpos = writepos;
            }

            /* Do we risk an 'underrun' if we don't advance pointer? */
            if (writelead/device->fraglen <= ds_snd_queue_min || recover)
                  mustlock = TRUE;

            if (lock)
                  IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, writepos, maxq, 0);

            /* do the mixing */
            frag = DSOUND_MixToPrimary(device, writepos, maxq, mustlock, recover, &all_stopped);

            /* update the mix position, taking wrap-around into acount */
            device->mixpos = writepos + frag;
            device->mixpos %= device->buflen;

            if (lock)
            {
                  DWORD frag2 = (frag > size1 ? frag - size1 : 0);
                  frag -= frag2;
                  if (frag2 > size2)
                  {
                        FIXME("Buffering too much! (%d, %d, %d, %d)\n", maxq, frag, size2, frag2 - size2);
                        frag2 = size2;
                  }
                  IDsDriverBuffer_Unlock(device->hwbuf, buf1, frag, buf2, frag2);
            }

            /* update prebuff left */
            prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);

            /* check if have a whole fragment */
            if (prebuff_left >= device->fraglen){

                  /* update the wave queue if using wave system */
                  if (!device->hwbuf)
                        DSOUND_WaveQueue(device, FALSE);

                  /* buffers are full. start playing if applicable */
                  if(device->state == STATE_STARTING){
                        TRACE("started primary buffer\n");
                        if(DSOUND_PrimaryPlay(device) != DS_OK){
                              WARN("DSOUND_PrimaryPlay failed\n");
                        }
                        else{
                              /* we are playing now */
                              device->state = STATE_PLAYING;
                        }
                  }

                  /* buffers are full. start stopping if applicable */
                  if(device->state == STATE_STOPPED){
                        TRACE("restarting primary buffer\n");
                        if(DSOUND_PrimaryPlay(device) != DS_OK){
                              WARN("DSOUND_PrimaryPlay failed\n");
                        }
                        else{
                              /* start stopping again. as soon as there is no more data, it will stop */
                              device->state = STATE_STOPPING;
                        }
                  }
            }

            /* if device was stopping, its for sure stopped when all buffers have stopped */
            else if((all_stopped == TRUE) && (device->state == STATE_STOPPING)){
                  TRACE("All buffers have stopped. Stopping primary buffer\n");
                  device->state = STATE_STOPPED;

                  /* stop the primary buffer now */
                  DSOUND_PrimaryStop(device);
            }

      } else {

            /* update the wave queue if using wave system */
            if (!device->hwbuf)
                  DSOUND_WaveQueue(device, TRUE);
            else
                  /* Keep alsa happy, which needs GetPosition called once every 10 ms */
                  IDsDriverBuffer_GetPosition(device->hwbuf, NULL, NULL);

            /* in the DSSCL_WRITEPRIMARY mode, the app is totally in charge... */
            if (device->state == STATE_STARTING) {
                  if (DSOUND_PrimaryPlay(device) != DS_OK)
                        WARN("DSOUND_PrimaryPlay failed\n");
                  else
                        device->state = STATE_PLAYING;
            }
            else if (device->state == STATE_STOPPING) {
                  if (DSOUND_PrimaryStop(device) != DS_OK)
                        WARN("DSOUND_PrimaryStop failed\n");
                  else
                        device->state = STATE_STOPPED;
            }
      }

      LeaveCriticalSection(&(device->mixlock));
      /* **** */
}

void CALLBACK DSOUND_timer(UINT timerID, UINT msg, DWORD_PTR dwUser,
                           DWORD_PTR dw1, DWORD_PTR dw2)
{
      DirectSoundDevice * device = (DirectSoundDevice*)dwUser;
      DWORD start_time =  GetTickCount();
      DWORD end_time;
      TRACE("(%d,%d,0x%lx,0x%lx,0x%lx)\n",timerID,msg,dwUser,dw1,dw2);
      TRACE("entering at %d\n", start_time);

      if (DSOUND_renderer[device->drvdesc.dnDevNode] != device) {
            ERR("dsound died without killing us?\n");
            timeKillEvent(timerID);
            timeEndPeriod(DS_TIME_RES);
            return;
      }

      RtlAcquireResourceShared(&(device->buffer_list_lock), TRUE);

      if (device->ref)
            DSOUND_PerformMix(device);

      RtlReleaseResource(&(device->buffer_list_lock));

      end_time = GetTickCount();
      TRACE("completed processing at %d, duration = %d\n", end_time, end_time - start_time);
}

void CALLBACK DSOUND_callback(HWAVEOUT hwo, UINT msg, DWORD dwUser, DWORD dw1, DWORD dw2)
{
      DirectSoundDevice * device = (DirectSoundDevice*)dwUser;
      TRACE("(%p,%x,%x,%x,%x)\n",hwo,msg,dwUser,dw1,dw2);
      TRACE("entering at %d, msg=%08x(%s)\n", GetTickCount(), msg,
            msg==MM_WOM_DONE ? "MM_WOM_DONE" : msg==MM_WOM_CLOSE ? "MM_WOM_CLOSE" : 
            msg==MM_WOM_OPEN ? "MM_WOM_OPEN" : "UNKNOWN");

      /* check if packet completed from wave driver */
      if (msg == MM_WOM_DONE) {

            /* **** */
            EnterCriticalSection(&(device->mixlock));

            TRACE("done playing primary pos=%d\n", device->pwplay * device->fraglen);

            /* update playpos */
            device->pwplay++;
            device->pwplay %= device->helfrags;

            /* sanity */
            if(device->pwqueue == 0){
                  ERR("Wave queue corrupted!\n");
            }

            /* update queue */
            device->pwqueue--;

            LeaveCriticalSection(&(device->mixlock));
            /* **** */
      }
      TRACE("completed\n");
}

Generated by  Doxygen 1.6.0   Back to index