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/*
Copyright (C) 1997-2001 Id Software, Inc.

This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.

This program 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 General Public License for more details.

You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.

*/
// snd_mix.c -- portable code to mix sounds for snd_dma.c

#include "cl_local.h"
#include "snd_local.h"

#define	PAINTBUFFER_SIZE	2048
portable_samplepair_t paintbuffer[PAINTBUFFER_SIZE];
int		snd_scaletable[32][256];
int 	*snd_p, snd_linear_count, snd_vol;
short	*snd_out;

#ifndef USE_ASM
static void S_WriteLinearBlastStereo16 (void) {
	int		i;
	int		val;

	for (i=0 ; i<snd_linear_count ; i+=2) {
		val = snd_p[i]>>8;
		if (val > 0x7fff)
			snd_out[i] = 0x7fff;
		else if (val < (short)0x8000)
			snd_out[i] = (short)0x8000;
		else
			snd_out[i] = val;

		val = snd_p[i+1]>>8;
		if (val > 0x7fff)
			snd_out[i+1] = 0x7fff;
		else if (val < (short)0x8000)
			snd_out[i+1] = (short)0x8000;
		else
			snd_out[i+1] = val;
	}
}
#else
void S_WriteLinearBlastStereo16 (void);
#endif

static void S_TransferStereo16 (unsigned long *pbuf, int endtime) {
	int		lpos;
	int		lpaintedtime;
	
	snd_p = (int *) paintbuffer;
	lpaintedtime = paintedtime;

	while (lpaintedtime < endtime) {
	// handle recirculating buffer issues
		lpos = lpaintedtime & ((dma.samples>>1)-1);

		snd_out = (short *) pbuf + (lpos<<1);

		snd_linear_count = (dma.samples>>1) - lpos;
		if (lpaintedtime + snd_linear_count > endtime)
			snd_linear_count = endtime - lpaintedtime;

		snd_linear_count <<= 1;

	// write a linear blast of samples
		S_WriteLinearBlastStereo16 ();

		snd_p += snd_linear_count;
		lpaintedtime += (snd_linear_count>>1);
	}
}

static void S_TransferStereo( unsigned long *pbuf, int endtime ) {
	int 	out_idx;
	int 	count;
	int 	out_mask;
	int 	*p;
	int 	step;
	int		val;

    p = (int *) paintbuffer;
    count = (endtime - paintedtime) * dma.channels;
    out_mask = dma.samples - 1; 
    out_idx = paintedtime * dma.channels & out_mask;
    step = 3 - dma.channels;

    if (dma.samplebits == 16) {
        short *out = (short *) pbuf;
        while (count--) {
            val = *p >> 8;
            p+= step;
            if (val > 0x7fff)
                val = 0x7fff;
            else if (val < (short)0x8000)
                val = (short)0x8000;
            out[out_idx] = val;
            out_idx = (out_idx + 1) & out_mask;
        }
    } else if (dma.samplebits == 8) {
        unsigned char *out = (unsigned char *) pbuf;
        while (count--) {
            val = *p >> 8;
            p+= step;
            if (val > 0x7fff)
                val = 0x7fff;
            else if (val < (short)0x8000)
                val = (short)0x8000;
            out[out_idx] = (val>>8) + 128;
            out_idx = (out_idx + 1) & out_mask;
        }
    }
}

/*
===================
S_TransferPaintBuffer

===================
*/
static void S_TransferPaintBuffer(int endtime) {
	unsigned long *pbuf = (unsigned long *)dma.buffer;

	if (s_testsound->integer) {
		int		i;
		int		count;

		// write a fixed sine wave
		count = (endtime - paintedtime);
		for (i=0 ; i<count ; i++)
			paintbuffer[i].left = paintbuffer[i].right = sin((paintedtime+i)*0.1)*20000*256;
	}

	if (dma.samplebits == 16 && dma.channels == 2) {
        // optimized case
		S_TransferStereo16 (pbuf, endtime);
	} else {
        // general case
        S_TransferStereo( pbuf, endtime );
	}
}


/*
===============================================================================

CHANNEL MIXING

===============================================================================
*/

static void S_PaintChannelFrom8 (channel_t *ch, sfxcache_t *sc, int count, int offset) {
	int 	data;
	int		*lscale, *rscale;
	byte     *sfx;
	int		i;
	portable_samplepair_t	*samp;

	if (ch->leftvol > 255)
		ch->leftvol = 255;
	if (ch->rightvol > 255)
		ch->rightvol = 255;
		
	//ZOID-- >>11 has been changed to >>3, >>11 didn't make much sense
	//as it would always be zero.
	lscale = snd_scaletable[ ch->leftvol >> 3];
	rscale = snd_scaletable[ ch->rightvol >> 3];
	samp = &paintbuffer[offset];

	sfx = sc->data + ch->pos * sc->channels;

	/*if( sc->channels == 2 ) {
		for( i=0 ; i<count ; i++, samp++ ) {
			samp->left += lscale[*sfx++];
			samp->right += rscale[*sfx++];
		}
	} else */{
		for( i=0 ; i<count ; i++, samp++ ) {
			data = *sfx++;
			samp->left += lscale[data];
			samp->right += rscale[data];
		}
	}
	
	ch->pos += count;
}

static void S_PaintChannelFrom16 (channel_t *ch, sfxcache_t *sc, int count, int offset) {
	int data;
	int left, right;
	int leftvol, rightvol;
	signed short *sfx;
	int	i;
	portable_samplepair_t	*samp;

	leftvol = ch->leftvol*snd_vol;
	rightvol = ch->rightvol*snd_vol;
	samp = &paintbuffer[offset];

	sfx = (signed short *)sc->data + ch->pos * sc->channels;

	/*if( sc->channels == 2 ) {
		for( i=0 ; i<count ; i++, samp++ ) {
			left = ( *sfx++ * leftvol ) >> 8;
			right = ( *sfx++ * rightvol ) >> 8;
			samp->left += left;
			samp->right += right;
		}
	} else*/ {
		for( i=0 ; i<count ; i++, samp++ ) {
			data = *sfx++;
			left = ( data * leftvol ) >> 8;
			right = ( data * rightvol ) >> 8;
			samp->left += left;
			samp->right += right;
		}
	}

	ch->pos += count;
}


void S_PaintChannels(int endtime) {
	int 	i;
	int 	end;
	channel_t *ch;
	sfxcache_t	*sc;
	int		ltime, count;
	playsound_t	*ps;

	snd_vol = s_volume->value*256;

	while (paintedtime < endtime) {
        // if paintbuffer is smaller than DMA buffer
		end = endtime;
		if (end - paintedtime > PAINTBUFFER_SIZE)
			end = paintedtime + PAINTBUFFER_SIZE;

		// start any playsounds
		while (1) {
			ps = s_pendingplays.next;
			if (ps == &s_pendingplays)
				break;	// no more pending sounds
			if (ps->begin <= paintedtime) {
				S_IssuePlaysound (ps);
				continue;
			}

			if (ps->begin < end)
				end = ps->begin;		// stop here
			break;
		}

    	// clear the paint buffer
		memset(paintbuffer, 0, (end - paintedtime) * sizeof(portable_samplepair_t));

        // paint in the channels.
		ch = channels;
		for (i=0; i<MAX_CHANNELS ; i++, ch++) {
			ltime = paintedtime;
		
			while (ltime < end) {
				if (!ch->sfx || (!ch->leftvol && !ch->rightvol) )
					break;

				// max painting is to the end of the buffer
				count = end - ltime;

				// might be stopped by running out of data
				if (ch->end - ltime < count)
					count = ch->end - ltime;
		
				sc = S_LoadSound (ch->sfx);
				if (!sc)
					break;

				if (count > 0 && ch->sfx) {	
					if (sc->width == 1)// FIXME; 8 bit asm is wrong now
						S_PaintChannelFrom8(ch, sc, count,  ltime - paintedtime);
					else
						S_PaintChannelFrom16(ch, sc, count, ltime - paintedtime);
	
					ltime += count;
				}

			    // if at end of loop, restart
				if (ltime >= ch->end) {
					if (ch->autosound) {
                        // autolooping sounds always go back to start
						ch->pos = 0;
						ch->end = ltime + sc->length;
					} else if (sc->loopstart >= 0) {
						ch->pos = sc->loopstart;
						ch->end = ltime + sc->length - ch->pos;
					} else {
                        // channel just stopped
						ch->sfx = NULL;
					}
				}
			}
															  
		}

	    // transfer out according to DMA format
		S_TransferPaintBuffer(end);
		paintedtime = end;
	}
}

void S_InitScaletable (void) {
	int		i, j;
	int		scale;

    Cvar_ClampValue( s_volume, 0, 2 );

	s_volume->modified = qfalse;
	for (i=0 ; i<32 ; i++) {
		scale = i * 8 * 256 * s_volume->value;
		for (j=0 ; j<256 ; j++)
			snd_scaletable[i][j] = ((signed char)j) * scale;
	}
}