mixer.c

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/*  			DirectSound
 *
 * Copyright 1998 Marcus Meissner
 * Copyright 1998 Rob Riggs
 * Copyright 2000-2002 TransGaming Technologies, Inc.
 *
 * 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */

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

#define NONAMELESSSTRUCT
#define NONAMELESSUNION
#include "windef.h"
#include "winbase.h"
#include "mmsystem.h"
#include "winreg.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=%ld Pan=%ld\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 = %lx, right = %lx\n", volpan->dwTotalLeftAmpFactor, volpan->dwTotalRightAmpFactor);
}

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

    TRACE("left=%lx, right=%lx\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=%ld Pan=%ld\n", volpan->lVolume, volpan->lPan);
}

void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
{
	TRACE("(%p)\n",dsb);

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

void DSOUND_CheckEvent(IDirectSoundBufferImpl *dsb, int len)
{
	int			i;
	DWORD			offset;
	LPDSBPOSITIONNOTIFY	event;
	TRACE("(%p,%d)\n",dsb,len);

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

	TRACE("(%p) buflen = %ld, playpos = %ld, len = %d\n",
		dsb, dsb->buflen, dsb->playpos, len);
	for (i = 0; i < dsb->nrofnotifies ; i++) {
		event = dsb->notifies + i;
		offset = event->dwOffset;
		TRACE("checking %d, position %ld, 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 ((dsb->playpos + len) >= dsb->buflen) {
			if ((offset < ((dsb->playpos + len) % dsb->buflen)) ||
			    (offset >= dsb->playpos)) {
				TRACE("signalled event %p (%d)\n", event->hEventNotify, i);
				SetEvent(event->hEventNotify);
			}
		} else {
			if ((offset >= dsb->playpos) && (offset < (dsb->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;
}

static inline void cp_fields(const IDirectSoundBufferImpl *dsb, BYTE *ibuf, BYTE *obuf )
{
	DirectSoundDevice * device = dsb->dsound->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 = *((INT16 *)ibuf);
                fr = (dsb->pwfx->nChannels==2 ? *(((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;
                }
        }
}

/* Now with PerfectPitch (tm) technology */
static INT DSOUND_MixerNorm(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len)
{
	INT	i, size, ipos, ilen;
	BYTE	*ibp, *obp;
	INT	iAdvance = dsb->pwfx->nBlockAlign;
	INT	oAdvance = dsb->dsound->device->pwfx->nBlockAlign;

	ibp = dsb->buffer->memory + dsb->buf_mixpos;
	obp = buf;

	TRACE("(%p, %p, %p), buf_mixpos=%ld\n", dsb, ibp, obp, dsb->buf_mixpos);
	/* Check for the best case */
	if ((dsb->freq == dsb->dsound->device->pwfx->nSamplesPerSec) &&
	    (dsb->pwfx->wBitsPerSample == dsb->dsound->device->pwfx->wBitsPerSample) &&
	    (dsb->pwfx->nChannels == dsb->dsound->device->pwfx->nChannels)) {
	        INT bytesleft = dsb->buflen - dsb->buf_mixpos;
		TRACE("(%p) Best case\n", dsb);
	    	if (len <= bytesleft )
			CopyMemory(obp, ibp, len);
		else { /* wrap */
			CopyMemory(obp, ibp, bytesleft);
			CopyMemory(obp + bytesleft, dsb->buffer->memory, len - bytesleft);
		}
		return len;
	}

	/* Check for same sample rate */
	if (dsb->freq == dsb->dsound->device->pwfx->nSamplesPerSec) {
		TRACE("(%p) Same sample rate %ld = primary %ld\n", dsb,
			dsb->freq, dsb->dsound->device->pwfx->nSamplesPerSec);
		ilen = 0;
		for (i = 0; i < len; i += oAdvance) {
			cp_fields(dsb, ibp, obp );
			ibp += iAdvance;
			ilen += iAdvance;
			obp += oAdvance;
			if (ibp >= (BYTE *)(dsb->buffer->memory + dsb->buflen))
				ibp = dsb->buffer->memory;	/* wrap */
		}
		return (ilen);
	}

	/* Mix in different sample rates */
	/* */
	/* New PerfectPitch(tm) Technology (c) 1998 Rob Riggs */
	/* Patent Pending :-] */

	/* Patent enhancements (c) 2000 Ove K鍁en,
	 * TransGaming Technologies Inc. */

	/* FIXME("(%p) Adjusting frequency: %ld -> %ld (need optimization)\n",
	   dsb, dsb->freq, dsb->dsound->device->pwfx->nSamplesPerSec); */

	size = len / oAdvance;
	ilen = 0;
	ipos = dsb->buf_mixpos;
	for (i = 0; i < size; i++) {
                cp_fields(dsb, (dsb->buffer->memory + ipos), obp);
		obp += oAdvance;
		dsb->freqAcc += dsb->freqAdjust;
		if (dsb->freqAcc >= (1<<DSOUND_FREQSHIFT)) {
			ULONG adv = (dsb->freqAcc>>DSOUND_FREQSHIFT) * iAdvance;
			dsb->freqAcc &= (1<<DSOUND_FREQSHIFT)-1;
			ipos += adv; ilen += adv;
			ipos %= dsb->buflen;
		}
	}
	return ilen;
}

static void DSOUND_MixerVol(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len)
{
	INT	i;
	BYTE	*bpc = buf;
	INT16	*bps = (INT16 *) buf;

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

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

	/* If we end up with some bozo coder using panning or 3D sound */
	/* with a mono primary buffer, it could sound very weird using */
	/* this method. Oh well, tough patooties. */

	switch (dsb->dsound->device->pwfx->wBitsPerSample) {
	case 8:
		/* 8-bit WAV is unsigned, but we need to operate */
		/* on signed data for this to work properly */
		switch (dsb->dsound->device->pwfx->nChannels) {
		case 1:
			for (i = 0; i < len; i++) {
				INT val = *bpc - 128;
				val = (val * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
				*bpc = val + 128;
				bpc++;
			}
			break;
		case 2:
			for (i = 0; i < len; i+=2) {
				INT val = *bpc - 128;
				val = (val * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
				*bpc++ = val + 128;
				val = *bpc - 128;
				val = (val * dsb->cvolpan.dwTotalRightAmpFactor) >> 16;
				*bpc = val + 128;
				bpc++;
			}
			break;
		default:
			FIXME("doesn't support %d channels\n", dsb->dsound->device->pwfx->nChannels);
			break;
		}
		break;
	case 16:
		/* 16-bit WAV is signed -- much better */
		switch (dsb->dsound->device->pwfx->nChannels) {
		case 1:
			for (i = 0; i < len; i += 2) {
				*bps = (*bps * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
				bps++;
			}
			break;
		case 2:
			for (i = 0; i < len; i += 4) {
				*bps = (*bps * dsb->cvolpan.dwTotalLeftAmpFactor) >> 16;
				bps++;
				*bps = (*bps * dsb->cvolpan.dwTotalRightAmpFactor) >> 16;
				bps++;
			}
			break;
		default:
			FIXME("doesn't support %d channels\n", dsb->dsound->device->pwfx->nChannels);
			break;
		}
		break;
	default:
		FIXME("doesn't support %d bit samples\n", dsb->dsound->device->pwfx->wBitsPerSample);
		break;
	}
}

static LPBYTE DSOUND_tmpbuffer(DirectSoundDevice *device, DWORD len)
{
    TRACE("(%p,%ld)\n", device, len);

    if (len > device->tmp_buffer_len) {
        if (device->tmp_buffer)
            device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, device->tmp_buffer, len);
        else
            device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, len);

        device->tmp_buffer_len = len;
    }

    return device->tmp_buffer;
}

static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen)
{
	INT	i, len, ilen, field, todo;
	BYTE	*buf, *ibuf;

	TRACE("(%p,%ld,%ld)\n",dsb,writepos,fraglen);

	len = fraglen;
	if (!(dsb->playflags & DSBPLAY_LOOPING)) {
		int secondary_remainder = dsb->buflen - dsb->buf_mixpos;
		int adjusted_remainder = MulDiv(dsb->dsound->device->pwfx->nAvgBytesPerSec, secondary_remainder, dsb->nAvgBytesPerSec);
		assert(adjusted_remainder >= 0);
		TRACE("secondary_remainder = %d, adjusted_remainder = %d, len = %d\n", secondary_remainder, adjusted_remainder, len);
		if (adjusted_remainder < len) {
			TRACE("clipping len to remainder of secondary buffer\n");
			len = adjusted_remainder;
		}
		if (len == 0)
			return 0;
	}