mixdrv.c

这个是延伸mame的在wince平台下的游戏模拟器的代码

/*
 * $Id: mixdrv.c 1.13 1996/09/08 21:14:54 chasan released $
 *               1.14 1998/10/24 18:20:53 chasan released (Mixer API)
 *
 * Software-based waveform synthesizer emulator driver.
 *
 * Copyright (C) 1995-1999 Carlos Hasan
 *
 * This program 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 of the License, or
 * (at your option) any later version.
 */

#define __FILTER__

#ifdef __GNUC__
#include <memory.h>
#define cdecl
#endif
#include <string.h>
#include <malloc.h>
#include "audio.h"
#include "drivers.h"


/*
 * Voice control bit fields
 */
#define VOICE_STOP              0x01
#define VOICE_16BITS            0x02
#define VOICE_LOOP              0x04
#define VOICE_BIDILOOP          0x08
#define VOICE_REVERSE           0x10

/*
 * Voice pitch accurary and mixing buffer size
 */
#define ACCURACY                12
#define BUFFERSIZE              512


/*
 * Waveform synthesizer voice structure
 */
typedef struct {
    LPVOID  lpData;
    LONG    dwAccum;
    LONG    dwFrequency;
    LONG    dwLoopStart;
    LONG    dwLoopEnd;
    BYTE    nVolume;
    BYTE    nPanning;
    BYTE    bControl;
    BYTE    bReserved;
} VOICE, *LPVOICE;

/*
 * Low level voice mixing routine prototype
 */
typedef VOID (cdecl* LPFNMIXAUDIO)(LPLONG, UINT, LPVOICE);

/*
 * Waveform synthesizer state structure
 */
static struct {
    VOICE   aVoices[AUDIO_MAX_VOICES];
    UINT    nVoices;
    UINT    wFormat;
    UINT    nSampleRate;
    LONG    dwTimerRate;
    LONG    dwTimerAccum;
    LPBYTE  lpMemory;
    LPFNAUDIOTIMER lpfnAudioTimer;
    LPFNMIXAUDIO lpfnMixAudioProc[2];
} Synth;

LPLONG lpVolumeTable;
LPBYTE lpFilterTable;

static VOID AIAPI UpdateVoices(LPBYTE lpData, UINT nCount);

/* low level resamplation and quantization routines */
#ifdef __ASM__

VOID cdecl QuantAudioData08(LPVOID lpBuffer, LPLONG lpData, UINT nCount);
VOID cdecl QuantAudioData16(LPVOID lpBuffer, LPLONG lpData, UINT nCount);
VOID cdecl MixAudioData08M(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice);
VOID cdecl MixAudioData08S(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice);
VOID cdecl MixAudioData08MI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice);
VOID cdecl MixAudioData08SI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice);
VOID cdecl MixAudioData16M(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice);
VOID cdecl MixAudioData16S(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice);
VOID cdecl MixAudioData16MI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice);
VOID cdecl MixAudioData16SI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice);

static VOID QuantAudioData(LPVOID lpBuffer, LPLONG lpData, UINT nCount)
{
    if (Synth.wFormat & AUDIO_FORMAT_16BITS)
	QuantAudioData16(lpBuffer, lpData, nCount);
    else
	QuantAudioData08(lpBuffer, lpData, nCount);
}

#else

typedef short SHORT;
typedef SHORT* LPSHORT;

static VOID QuantAudioData(LPVOID lpBuffer, LPLONG lpData, UINT nCount)
{
    LPSHORT lpwBuffer;
    LPBYTE lpbBuffer;
    LONG dwSample;

    if (Synth.wFormat & AUDIO_FORMAT_16BITS) {
	lpwBuffer = (LPSHORT) lpBuffer;
	while (nCount-- > 0) {
	    dwSample = *lpData++;
	    if (dwSample < -32768)
		dwSample = -32768;
	    else if (dwSample > +32767)
		dwSample = +32767;
	    *lpwBuffer++ = (SHORT) dwSample;
	}
    }
    else {
	lpbBuffer = (LPBYTE) lpBuffer;
	while (nCount-- > 0) {
	    dwSample = *lpData++;
	    if (dwSample < -32768)
		dwSample = -32768;
	    else if (dwSample > +32767)
		dwSample = +32767;
	    *lpbBuffer++ = (BYTE) ((dwSample >> 8) + 128);
	}
    }
}

static VOID AIAPI
MixAudioData08M(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
{
    register UINT count;
    register DWORD accum, delta;
    register LPLONG table, buf;
    register LPBYTE data;

    accum = lpVoice->dwAccum;
    delta = lpVoice->dwFrequency;
    table = lpVolumeTable + ((UINT) lpVoice->nVolume << 8);
    data = lpVoice->lpData;
    buf = lpBuffer;
    count = nCount;

    do {
	*buf++ += table[data[accum >> ACCURACY]];
	accum += delta;
    } while (--count != 0);

    lpVoice->dwAccum = accum;
}

#define min(a, b) ((a) < (b) ? (a) : (b))

static VOID AIAPI
MixAudioData08S(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
{
    register UINT a;
    register DWORD accum, delta;
    register LPLONG ltable, rtable, buf;
    register LPBYTE data;

    accum = lpVoice->dwAccum;
    delta = lpVoice->dwFrequency;
    a = ((UINT) lpVoice->nVolume * lpVoice->nPanning) >> 8;
    rtable = lpVolumeTable + (a << 8);
    ltable = lpVolumeTable + ((lpVoice->nVolume - a) << 8);
    data = lpVoice->lpData;
    buf = lpBuffer;

    do {
	a = data[accum >> ACCURACY];
	buf[0] += ltable[a];
	buf[1] += rtable[a];
	accum += delta;
	buf += 2;
    } while (--nCount != 0);

    lpVoice->dwAccum = accum;
}

static VOID AIAPI
MixAudioData08MI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
{
    register INT a, frac;
    register DWORD accum, delta;
    register LPLONG table, buf;
    register LPBYTE data, ftable;

    /* adaptive oversampling interpolation comparison */
    if (lpVoice->dwFrequency < -(1 << ACCURACY) ||
	lpVoice->dwFrequency > +(1 << ACCURACY)) {
	MixAudioData08M(lpBuffer, nCount, lpVoice);
	return;
    }

    accum = lpVoice->dwAccum;
    delta = lpVoice->dwFrequency;
    table = lpVolumeTable + ((UINT) lpVoice->nVolume << 8);
    data = lpVoice->lpData;
    buf = lpBuffer;

#ifdef __FILTER__
    a = (BYTE) lpVoice->bReserved;
    frac = ((long) delta < 0 ? -delta : +delta) >> (ACCURACY - 5);
    ftable = lpFilterTable + (frac << 8);
    do {
	a = (BYTE)(a + ftable[data[accum >> ACCURACY]] - ftable[a]);
	*buf++ += table[a];
	accum += delta;
    } while (--nCount != 0);
    lpVoice->bReserved = (BYTE) a;
#else
    do {
	register INT b;
	a = (signed char) data[accum >> ACCURACY];
	b = (signed char) data[(accum >> ACCURACY) + 1];
	frac = (accum & ((1 << ACCURACY) - 1)) >> (ACCURACY - 5);
	a = (BYTE)(a + ((frac * (b - a)) >> 5));
	*buf++ += table[a];
	accum += delta;
    } while (--nCount != 0);
#endif

    lpVoice->dwAccum = accum;
}

static VOID AIAPI
MixAudioData08SI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
{
    register INT a, frac;
    register DWORD accum, delta;
    register LPLONG buf, ltable, rtable;
    register LPBYTE data, ftable;

    /* adaptive oversampling interpolation comparison */
    if (lpVoice->dwFrequency < -(1 << ACCURACY) ||
	lpVoice->dwFrequency > +(1 << ACCURACY)) {
	MixAudioData08S(lpBuffer, nCount, lpVoice);
	return;
    }

    accum = lpVoice->dwAccum;
    delta = lpVoice->dwFrequency;
    a = ((UINT) lpVoice->nVolume * lpVoice->nPanning) >> 8;
    rtable = lpVolumeTable + (a << 8);
    ltable = lpVolumeTable + ((lpVoice->nVolume - a) << 8);
    data = lpVoice->lpData;
    buf = lpBuffer;

#ifdef __FILTER__
    a = (BYTE) lpVoice->bReserved;
    frac = ((long) delta < 0 ? -delta : +delta) >> (ACCURACY - 5);
    ftable = lpFilterTable + (frac << 8);
    do {
	a = (BYTE)(a + ftable[data[accum >> ACCURACY]] - ftable[a]);
	buf[0] += ltable[a];
	buf[1] += rtable[a];
	accum += delta;
	buf += 2;
    } while (--nCount != 0);
    lpVoice->bReserved = (BYTE) a;
#else
    do {
	register INT b;
	a = (signed char) data[accum >> ACCURACY];
	b = (signed char) data[(accum >> ACCURACY) + 1];
	frac = (accum & ((1 << ACCURACY) - 1)) >> (ACCURACY - 5);
	a = (BYTE)(a + ((frac * (b - a)) >> 5));
	buf[0] += ltable[a];
	buf[1] += rtable[a];
	accum += delta;
	buf += 2;
    } while (--nCount != 0);
#endif

    lpVoice->dwAccum = accum;
}

static VOID AIAPI
MixAudioData16M(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
{
    register UINT a, count;
    register DWORD accum, delta;
    register LPLONG table, buf;
    register LPWORD data;

    accum = lpVoice->dwAccum;
    delta = lpVoice->dwFrequency;
    table = lpVolumeTable + ((UINT) lpVoice->nVolume << 8);
    data = lpVoice->lpData;
    buf = lpBuffer;
    count = nCount;

    do {
	a = data[accum >> ACCURACY];
	*buf++ += table[a >> 8];
	accum += delta;
    } while (--count != 0);

    lpVoice->dwAccum = accum;
}

static VOID AIAPI
MixAudioData16S(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
{
    register UINT a;
    register DWORD accum, delta;
    register LPLONG ltable, rtable, buf;
    register LPWORD data;

    accum = lpVoice->dwAccum;
    delta = lpVoice->dwFrequency;
    a = ((UINT) lpVoice->nVolume * lpVoice->nPanning) >> 8;
    rtable = lpVolumeTable + (a << 8);
    ltable = lpVolumeTable + ((lpVoice->nVolume - a) << 8);
    data = lpVoice->lpData;
    buf = lpBuffer;

    do {
	a = data[accum >> ACCURACY];
	buf[0] += ltable[a >> 8];
	buf[1] += rtable[a >> 8];
	accum += delta;
	buf += 2;
    } while (--nCount != 0);

    lpVoice->dwAccum = accum;
}


static VOID AIAPI
MixAudioData16MI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
{
    register UINT frac, count;
    register DWORD a, b, accum, delta;
    register LPLONG table, buf;
    register LPWORD data;

    /* adaptive oversampling interpolation comparison */
    if (lpVoice->dwFrequency < -(1 << ACCURACY) ||
	lpVoice->dwFrequency > +(1 << ACCURACY)) {
	MixAudioData16M(lpBuffer, nCount, lpVoice);
	return;
    }

    accum = lpVoice->dwAccum;
    delta = lpVoice->dwFrequency;
    table = lpVolumeTable + ((UINT) lpVoice->nVolume << 8);
    data = lpVoice->lpData;
    buf = lpBuffer;
    count = nCount;

    do {
	a = data[accum >> ACCURACY];
	b = data[(accum >> ACCURACY) + 1];
	frac = (accum & ((1 << ACCURACY) - 1)) >> (ACCURACY - 5);
	a = (WORD)((SHORT)a + ((frac * ((SHORT)b - (SHORT)a)) >> 5));
	*buf++ += table[a >> 8];
	accum += delta;
    } while (--count != 0);

    lpVoice->dwAccum = accum;
}

static VOID AIAPI
MixAudioData16SI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
{
    register UINT frac;
    register DWORD a, b, accum, delta;
    register LPLONG ltable, rtable, buf;
    register LPWORD data;

    /* adaptive oversampling interpolation comparison */
    if (lpVoice->dwFrequency < -(1 << ACCURACY) ||
	lpVoice->dwFrequency > +(1 << ACCURACY)) {
	MixAudioData16S(lpBuffer, nCount, lpVoice);
	return;
    }

    accum = lpVoice->dwAccum;
    delta = lpVoice->dwFrequency;
    a = ((UINT) lpVoice->nVolume * lpVoice->nPanning) >> 8;
    rtable = lpVolumeTable + (a << 8);
    ltable = lpVolumeTable + ((lpVoice->nVolume - a) << 8);
    data = lpVoice->lpData;
    buf = lpBuffer;

    do {
	a = data[accum >> ACCURACY];
	b = data[(accum >> ACCURACY) + 1];
	frac = (accum & ((1 << ACCURACY) - 1)) >> (ACCURACY - 5);
	a = (WORD)((SHORT)a + ((frac * ((int)(SHORT)b - (int)(SHORT)a)) >> 5));
	buf[0] += ltable[a >> 8];
	buf[1] += rtable[a >> 8];
	accum += delta;
	buf += 2;
    } while (--nCount != 0);

    lpVoice->dwAccum = accum;
}

#endif

static VOID MixAudioData(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
{
    UINT nSize;

    if (Synth.wFormat & AUDIO_FORMAT_STEREO)
	nCount >>= 1;

    while (nCount > 0 && !(lpVoice->bControl & VOICE_STOP)) {
	/* check boundary conditions */
	if (lpVoice->bControl & VOICE_REVERSE) {
	    if (lpVoice->dwAccum < lpVoice->dwLoopStart) {
		if (lpVoice->bControl & VOICE_BIDILOOP) {
		    lpVoice->dwAccum = lpVoice->dwLoopStart +
			(lpVoice->dwLoopStart - lpVoice->dwAccum);
		    lpVoice->bControl ^= VOICE_REVERSE;
		    continue;
		}
		else if (lpVoice->bControl & VOICE_LOOP) {
		    lpVoice->dwAccum = lpVoice->dwLoopEnd -
			(lpVoice->dwLoopStart - lpVoice->dwAccum);
		    continue;
		}
		else {
		    lpVoice->bControl |= VOICE_STOP;
		    break;
		}
	    }
	}
	else {
	    if (lpVoice->dwAccum > lpVoice->dwLoopEnd) {
		if (lpVoice->bControl & VOICE_BIDILOOP) {
		    lpVoice->dwAccum = lpVoice->dwLoopEnd -
			(lpVoice->dwAccum - lpVoice->dwLoopEnd);
		    lpVoice->bControl ^= VOICE_REVERSE;
		    continue;
		}
		else if (lpVoice->bControl & VOICE_LOOP) {
		    lpVoice->dwAccum = lpVoice->dwLoopStart +
			(lpVoice->dwAccum - lpVoice->dwLoopEnd);
		    continue;