mirror of
https://github.com/DrBeef/JKXR.git
synced 2024-12-12 21:52:08 +00:00
4597b03873
Opens in Android Studio but haven't even tried to build it yet (it won't.. I know that much!)
392 lines
8.9 KiB
C++
392 lines
8.9 KiB
C++
/*
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===========================================================================
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Copyright (C) 1999 - 2005, Id Software, Inc.
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Copyright (C) 2000 - 2013, Raven Software, Inc.
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Copyright (C) 2001 - 2013, Activision, Inc.
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Copyright (C) 2013 - 2015, OpenJK contributors
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This file is part of the OpenJK source code.
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OpenJK is free software; you can redistribute it and/or modify it
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under the terms of the GNU General Public License version 2 as
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published by the Free Software Foundation.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, see <http://www.gnu.org/licenses/>.
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===========================================================================
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*/
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// snd_mix.c -- portable code to mix sounds for snd_dma.c
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#include "../server/exe_headers.h"
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#include "snd_local.h"
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portable_samplepair_t paintbuffer[PAINTBUFFER_SIZE];
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int *snd_p, snd_linear_count, snd_vol;
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short *snd_out;
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void S_WriteLinearBlastStereo16 (void)
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{
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int i;
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int val;
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for (i=0 ; i<snd_linear_count ; i+=2)
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{
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val = snd_p[i]>>8;
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if (val > 0x7fff)
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snd_out[i] = 0x7fff;
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else if (val < (short)0x8000)
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snd_out[i] = (short)0x8000;
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else
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snd_out[i] = val;
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val = snd_p[i+1]>>8;
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if (val > 0x7fff)
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snd_out[i+1] = 0x7fff;
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else if (val < (short)0x8000)
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snd_out[i+1] = (short)0x8000;
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else
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snd_out[i+1] = val;
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}
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}
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void S_TransferStereo16 (unsigned long *pbuf, int endtime)
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{
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int lpos;
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int ls_paintedtime;
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snd_p = (int *) paintbuffer;
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ls_paintedtime = s_paintedtime;
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while (ls_paintedtime < endtime)
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{
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// handle recirculating buffer issues
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lpos = ls_paintedtime & ((dma.samples>>1)-1);
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snd_out = (short *) pbuf + (lpos<<1);
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snd_linear_count = (dma.samples>>1) - lpos;
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if (ls_paintedtime + snd_linear_count > endtime)
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snd_linear_count = endtime - ls_paintedtime;
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snd_linear_count <<= 1;
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// write a linear blast of samples
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S_WriteLinearBlastStereo16 ();
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snd_p += snd_linear_count;
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ls_paintedtime += (snd_linear_count>>1);
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}
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}
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/*
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===================
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S_TransferPaintBuffer
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===================
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*/
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void S_TransferPaintBuffer(int endtime)
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{
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int out_idx;
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int count;
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int out_mask;
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int *p;
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int step;
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int val;
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unsigned long *pbuf;
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pbuf = (unsigned long *)dma.buffer;
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if ( s_testsound->integer ) {
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int i;
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int count;
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// write a fixed sine wave
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count = (endtime - s_paintedtime);
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for (i=0 ; i<count ; i++)
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paintbuffer[i].left = paintbuffer[i].right = (int)(sin((s_paintedtime+i)*0.1)*20000*256);
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}
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if (dma.samplebits == 16 && dma.channels == 2)
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{ // optimized case
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S_TransferStereo16 (pbuf, endtime);
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}
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else
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{ // general case
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p = (int *) paintbuffer;
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count = (endtime - s_paintedtime) * dma.channels;
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out_mask = dma.samples - 1;
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out_idx = s_paintedtime * dma.channels & out_mask;
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step = 3 - dma.channels;
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if (dma.samplebits == 16)
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{
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short *out = (short *) pbuf;
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while (count--)
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{
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val = *p >> 8;
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p+= step;
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if (val > 0x7fff)
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val = 0x7fff;
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else if (val < (short)0x8000)
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val = (short)0x8000;
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out[out_idx] = (short)val;
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out_idx = (out_idx + 1) & out_mask;
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}
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}
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else if (dma.samplebits == 8)
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{
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unsigned char *out = (unsigned char *) pbuf;
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while (count--)
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{
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val = *p >> 8;
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p+= step;
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if (val > 0x7fff)
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val = 0x7fff;
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else if (val < (short)0x8000)
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val = (short)0x8000;
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out[out_idx] = (short)((val>>8) + 128);
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out_idx = (out_idx + 1) & out_mask;
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}
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}
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}
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}
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/*
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===============================================================================
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CHANNEL MIXING
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===============================================================================
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*/
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static void S_PaintChannelFrom16( channel_t *ch, const sfx_t *sfx, int count, int sampleOffset, int bufferOffset )
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{
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portable_samplepair_t *pSamplesDest;
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int iData;
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int iLeftVol = ch->leftvol * snd_vol;
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int iRightVol = ch->rightvol * snd_vol;
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pSamplesDest = &paintbuffer[ bufferOffset ];
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for ( int i=0 ; i<count ; i++ )
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{
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iData = sfx->pSoundData[ sampleOffset++ ];
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pSamplesDest[i].left += (iData * iLeftVol )>>8;
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pSamplesDest[i].right += (iData * iRightVol)>>8;
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}
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}
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void S_PaintChannelFromMP3( channel_t *ch, const sfx_t *sc, int count, int sampleOffset, int bufferOffset )
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{
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int data;
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int leftvol, rightvol;
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signed short *sfx;
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int i;
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portable_samplepair_t *samp;
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static short tempMP3Buffer[PAINTBUFFER_SIZE];
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MP3Stream_GetSamples( ch, sampleOffset, count, tempMP3Buffer, qfalse ); // qfalse = not stereo
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leftvol = ch->leftvol*snd_vol;
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rightvol = ch->rightvol*snd_vol;
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sfx = tempMP3Buffer;
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samp = &paintbuffer[ bufferOffset ];
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while ( count & 3 ) {
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data = *sfx;
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samp->left += (data * leftvol)>>8;
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samp->right += (data * rightvol)>>8;
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sfx++;
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samp++;
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count--;
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}
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for ( i=0 ; i<count ; i += 4 ) {
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data = sfx[i];
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samp[i].left += (data * leftvol)>>8;
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samp[i].right += (data * rightvol)>>8;
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data = sfx[i+1];
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samp[i+1].left += (data * leftvol)>>8;
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samp[i+1].right += (data * rightvol)>>8;
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data = sfx[i+2];
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samp[i+2].left += (data * leftvol)>>8;
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samp[i+2].right += (data * rightvol)>>8;
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data = sfx[i+3];
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samp[i+3].left += (data * leftvol)>>8;
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samp[i+3].right += (data * rightvol)>>8;
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}
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}
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// subroutinised to save code dup (called twice) -ste
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//
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void ChannelPaint(channel_t *ch, sfx_t *sc, int count, int sampleOffset, int bufferOffset)
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{
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switch (sc->eSoundCompressionMethod)
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{
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case ct_16:
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S_PaintChannelFrom16 (ch, sc, count, sampleOffset, bufferOffset);
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break;
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case ct_MP3:
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S_PaintChannelFromMP3 (ch, sc, count, sampleOffset, bufferOffset);
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break;
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default:
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assert(0); // debug aid, ignored in release. FIXME: Should we ERR_DROP here for badness-catch?
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break;
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}
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}
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void S_PaintChannels( int endtime ) {
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int i;
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int end;
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channel_t *ch;
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sfx_t *sc;
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int ltime, count;
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int sampleOffset;
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int normal_vol,voice_vol;
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snd_vol = normal_vol = s_volume->value*256.0f;
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voice_vol = (s_volumeVoice->value*256.0f);
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//Com_Printf ("%i to %i\n", s_paintedtime, endtime);
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while ( s_paintedtime < endtime ) {
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// if paintbuffer is smaller than DMA buffer
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// we may need to fill it multiple times
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end = endtime;
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if ( endtime - s_paintedtime > PAINTBUFFER_SIZE ) {
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end = s_paintedtime + PAINTBUFFER_SIZE;
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}
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// clear the paint buffer to either music or zeros
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if ( s_rawend < s_paintedtime ) {
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if ( s_rawend ) {
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//Com_DPrintf ("background sound underrun\n");
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}
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memset(paintbuffer, 0, (end - s_paintedtime) * sizeof(portable_samplepair_t));
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} else {
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// copy from the streaming sound source
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int s;
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int stop;
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stop = (end < s_rawend) ? end : s_rawend;
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for ( i = s_paintedtime ; i < stop ; i++ ) {
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s = i&(MAX_RAW_SAMPLES-1);
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paintbuffer[i-s_paintedtime] = s_rawsamples[s];
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}
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// if (i != end)
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// Com_Printf ("partial stream\n");
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// else
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// Com_Printf ("full stream\n");
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for ( ; i < end ; i++ ) {
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paintbuffer[i-s_paintedtime].left =
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paintbuffer[i-s_paintedtime].right = 0;
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}
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}
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// paint in the channels.
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ch = s_channels;
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for ( i = 0; i < MAX_CHANNELS ; i++, ch++ ) {
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if ( !ch->thesfx || (ch->leftvol<0.25 && ch->rightvol<0.25 )) {
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continue;
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}
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if ( ch->entchannel == CHAN_VOICE || ch->entchannel == CHAN_VOICE_ATTEN || ch->entchannel == CHAN_VOICE_GLOBAL )
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snd_vol = voice_vol;
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else
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snd_vol = normal_vol;
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ltime = s_paintedtime;
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sc = ch->thesfx;
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// we might have to make 2 passes if it is
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// a looping sound effect and the end of
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// the sameple is hit...
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//
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do
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{
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if (ch->loopSound) {
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sampleOffset = ltime % sc->iSoundLengthInSamples;
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} else {
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sampleOffset = ltime - ch->startSample;
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}
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count = end - ltime;
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if ( sampleOffset + count > sc->iSoundLengthInSamples ) {
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count = sc->iSoundLengthInSamples - sampleOffset;
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}
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if ( count > 0 ) {
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ChannelPaint(ch, sc, count, sampleOffset, ltime - s_paintedtime);
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ltime += count;
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}
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} while ( ltime < end && ch->loopSound );
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}
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/* temprem
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// paint in the looped channels.
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ch = loop_channels;
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for ( i = 0; i < numLoopChannels ; i++, ch++ ) {
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if ( !ch->thesfx || (!ch->leftvol && !ch->rightvol )) {
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continue;
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}
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{
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ltime = s_paintedtime;
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sc = ch->thesfx;
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if (sc->soundData==NULL || sc->soundLength==0) {
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continue;
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}
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// we might have to make two passes if it
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// is a looping sound effect and the end of
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// the sample is hit
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do {
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sampleOffset = (ltime % sc->soundLength);
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count = end - ltime;
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if ( sampleOffset + count > sc->soundLength ) {
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count = sc->soundLength - sampleOffset;
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}
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if ( count > 0 )
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{
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ChannelPaint(ch, sc, count, sampleOffset, ltime - s_paintedtime);
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ltime += count;
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}
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} while ( ltime < end);
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}
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}
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*/
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// transfer out according to DMA format
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S_TransferPaintBuffer( end );
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s_paintedtime = end;
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}
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}
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