Xash3DArchive/snd_dx/s_mix.c

/*
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 "sound.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;

void S_WriteLinearBlastStereo16( void )
{
	int	i, 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;
	}
}

void S_TransferStereo16( dword *pbuf, int endtime )
{
	int	lpos, 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);
	}
}

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

===================
*/
void S_TransferPaintBuffer( int endtime )
{
	int 	out_idx, count;
	int 	step, val;
	int 	*p, out_mask;
	dword	*pbuf;

	pbuf = (dword *)dma.buffer;

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

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


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

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

CHANNEL MIXING

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

	if( ch->leftvol > 255 ) ch->leftvol = 255;
	if( ch->rightvol > 255 ) ch->rightvol = 255;
		
	lscale = snd_scaletable[ch->leftvol>>3];
	rscale = snd_scaletable[ch->rightvol>>3];
	sfx = (signed char *)sc->data + ch->pos;

	samp = &paintbuffer[offset];

	for( i = 0; i < count; i++, samp++ )
	{
		data = sfx[i];
		samp->left += lscale[data];
		samp->right += rscale[data];
	}
	ch->pos += count;
}

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

	leftvol = ch->leftvol * snd_vol;
	rightvol = ch->rightvol * snd_vol;
	sfx = (signed short *)sc->data + ch->pos;

	samp = &paintbuffer[offset];

	for( i = 0; i < count; i++, samp++ )
	{
		data = sfx[i];
		left = ( data * leftvol ) >> 8;
		right = (data * rightvol) >> 8;
		samp->left += left;
		samp->right += right;
	}
	ch->pos += count;
}

void S_MixAllChannels( int endtime, int end )
{
	channel_t 	*ch;
	sfxcache_t	*sc;
	int		i, count = 0;	
	int		ltime;

	// paint in the channels.
	for( i = 0, ch = channels; i < total_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 )
					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 && ch->use_loop )
				{	// autolooping sounds always go back to start
					ch->pos = 0;
					ch->end = ltime + sc->length;
				}
				else if( sc->loopstart >= 0 && ch->use_loop )
				{
					ch->pos = sc->loopstart;
					ch->end = ltime + sc->length - ch->pos;
				}
				else ch->sfx = NULL; // channel just stopped
			}
		}
	}
}

void S_PaintChannels( int endtime )
{
	playsound_t	*ps;
	int		i, end;

	snd_vol = s_volume->value * 256;

	while( paintedtime < endtime )
	{
		// if paintbuffer is smaller than DMA buffer
		end = endtime;
		if( endtime - 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
		if( s_rawend < paintedtime )
		{
			Mem_Set( paintbuffer, 0, (end - paintedtime) * sizeof( portable_samplepair_t ));
		}
		else
		{	
			int	stop;

			// copy from the streaming sound source
			stop = (end < s_rawend) ? end : s_rawend;

			for( i = paintedtime; i < stop; i++ )
				paintbuffer[i - paintedtime] = s_rawsamples[i & (MAX_RAW_SAMPLES - 1)];
			
			for( ; i < end; i++ )
				paintbuffer[i-paintedtime].left = paintbuffer[i-paintedtime].right = 0;
		}

		S_MixAllChannels( endtime, end );

		SX_RoomFX( endtime, true, true );

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

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

	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;
	}
	s_volume->modified = false;
}