mirror of
https://github.com/UberGames/lilium-voyager.git
synced 2024-12-13 21:51:09 +00:00
4f8c7c2f2f
Fixes missing audio when playing on Windows with SDL 2.0.7, which started using WASAPI, which demands floating point audio.
803 lines
20 KiB
C
803 lines
20 KiB
C
/*
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===========================================================================
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Copyright (C) 1999-2005 Id Software, Inc.
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This file is part of Quake III Arena source code.
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Quake III Arena source code is free software; you can redistribute it
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and/or modify it under the terms of the GNU General Public License as
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published by the Free Software Foundation; either version 2 of the License,
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or (at your option) any later version.
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Quake III Arena source code is distributed in the hope that it will be
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useful, 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 Quake III Arena source code; if not, write to the Free Software
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Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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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 "client.h"
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#include "snd_local.h"
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#if idppc_altivec && !defined(__APPLE__)
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#include <altivec.h>
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#endif
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static portable_samplepair_t paintbuffer[PAINTBUFFER_SIZE];
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static int snd_vol;
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int* snd_p;
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int snd_linear_count;
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short* snd_out;
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#if !id386 // if configured not to use asm
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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 < -32768)
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snd_out[i] = -32768;
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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 < -32768)
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snd_out[i+1] = -32768;
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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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#elif defined(__GNUC__)
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// uses snd_mixa.s
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void S_WriteLinearBlastStereo16 (void);
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#else
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__declspec( naked ) void S_WriteLinearBlastStereo16 (void)
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{
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__asm {
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push edi
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push ebx
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mov ecx,ds:dword ptr[snd_linear_count]
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mov ebx,ds:dword ptr[snd_p]
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mov edi,ds:dword ptr[snd_out]
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LWLBLoopTop:
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mov eax,ds:dword ptr[-8+ebx+ecx*4]
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sar eax,8
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cmp eax,07FFFh
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jg LClampHigh
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cmp eax,0FFFF8000h
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jnl LClampDone
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mov eax,0FFFF8000h
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jmp LClampDone
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LClampHigh:
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mov eax,07FFFh
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LClampDone:
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mov edx,ds:dword ptr[-4+ebx+ecx*4]
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sar edx,8
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cmp edx,07FFFh
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jg LClampHigh2
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cmp edx,0FFFF8000h
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jnl LClampDone2
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mov edx,0FFFF8000h
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jmp LClampDone2
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LClampHigh2:
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mov edx,07FFFh
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LClampDone2:
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shl edx,16
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and eax,0FFFFh
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or edx,eax
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mov ds:dword ptr[-4+edi+ecx*2],edx
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sub ecx,2
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jnz LWLBLoopTop
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pop ebx
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pop edi
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ret
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}
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}
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#endif
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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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if( CL_VideoRecording( ) )
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CL_WriteAVIAudioFrame( (byte *)snd_out, 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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// 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 = 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.isfloat) && (dma.samplebits == 32))
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{
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float *out = (float *) 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 < -32767) /* clamp to one less than max to make division max out at -1.0f. */
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val = -32767;
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out[out_idx] = ((float) val) / 32767.0f;
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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 == 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 < -32768)
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val = -32768;
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out[out_idx] = 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 < -32768)
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val = -32768;
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out[out_idx] = (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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#if idppc_altivec
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static void S_PaintChannelFrom16_altivec( channel_t *ch, const sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
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int data, aoff, boff;
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int leftvol, rightvol;
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int i, j;
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portable_samplepair_t *samp;
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sndBuffer *chunk;
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short *samples;
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float ooff, fdata[2], fdiv, fleftvol, frightvol;
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if (sc->soundChannels <= 0) {
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return;
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}
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samp = &paintbuffer[ bufferOffset ];
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if (ch->doppler) {
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sampleOffset = sampleOffset*ch->oldDopplerScale;
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}
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if ( sc->soundChannels == 2 ) {
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sampleOffset *= sc->soundChannels;
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if ( sampleOffset & 1 ) {
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sampleOffset &= ~1;
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}
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}
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chunk = sc->soundData;
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while (sampleOffset>=SND_CHUNK_SIZE) {
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chunk = chunk->next;
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sampleOffset -= SND_CHUNK_SIZE;
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if (!chunk) {
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chunk = sc->soundData;
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}
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}
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if (!ch->doppler || ch->dopplerScale==1.0f) {
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vector signed short volume_vec;
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vector unsigned int volume_shift;
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int vectorCount, samplesLeft, chunkSamplesLeft;
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leftvol = ch->leftvol*snd_vol;
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rightvol = ch->rightvol*snd_vol;
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samples = chunk->sndChunk;
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((short *)&volume_vec)[0] = leftvol;
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((short *)&volume_vec)[1] = leftvol;
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((short *)&volume_vec)[4] = leftvol;
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((short *)&volume_vec)[5] = leftvol;
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((short *)&volume_vec)[2] = rightvol;
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((short *)&volume_vec)[3] = rightvol;
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((short *)&volume_vec)[6] = rightvol;
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((short *)&volume_vec)[7] = rightvol;
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volume_shift = vec_splat_u32(8);
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i = 0;
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while(i < count) {
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/* Try to align destination to 16-byte boundary */
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while(i < count && (((unsigned long)&samp[i] & 0x1f) || ((count-i) < 8) || ((SND_CHUNK_SIZE - sampleOffset) < 8))) {
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data = samples[sampleOffset++];
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samp[i].left += (data * leftvol)>>8;
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if ( sc->soundChannels == 2 ) {
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data = samples[sampleOffset++];
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}
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samp[i].right += (data * rightvol)>>8;
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if (sampleOffset == SND_CHUNK_SIZE) {
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chunk = chunk->next;
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samples = chunk->sndChunk;
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sampleOffset = 0;
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}
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i++;
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}
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/* Destination is now aligned. Process as many 8-sample
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chunks as we can before we run out of room from the current
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sound chunk. We do 8 per loop to avoid extra source data reads. */
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samplesLeft = count - i;
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chunkSamplesLeft = SND_CHUNK_SIZE - sampleOffset;
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if(samplesLeft > chunkSamplesLeft)
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samplesLeft = chunkSamplesLeft;
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vectorCount = samplesLeft / 8;
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if(vectorCount)
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{
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vector unsigned char tmp;
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vector short s0, s1, sampleData0, sampleData1;
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vector signed int merge0, merge1;
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vector signed int d0, d1, d2, d3;
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vector unsigned char samplePermute0 =
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VECCONST_UINT8(0, 1, 4, 5, 0, 1, 4, 5, 2, 3, 6, 7, 2, 3, 6, 7);
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vector unsigned char samplePermute1 =
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VECCONST_UINT8(8, 9, 12, 13, 8, 9, 12, 13, 10, 11, 14, 15, 10, 11, 14, 15);
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vector unsigned char loadPermute0, loadPermute1;
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// Rather than permute the vectors after we load them to do the sample
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// replication and rearrangement, we permute the alignment vector so
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// we do everything in one step below and avoid data shuffling.
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tmp = vec_lvsl(0,&samples[sampleOffset]);
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loadPermute0 = vec_perm(tmp,tmp,samplePermute0);
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loadPermute1 = vec_perm(tmp,tmp,samplePermute1);
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s0 = *(vector short *)&samples[sampleOffset];
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while(vectorCount)
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{
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/* Load up source (16-bit) sample data */
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s1 = *(vector short *)&samples[sampleOffset+7];
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/* Load up destination sample data */
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d0 = *(vector signed int *)&samp[i];
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d1 = *(vector signed int *)&samp[i+2];
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d2 = *(vector signed int *)&samp[i+4];
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d3 = *(vector signed int *)&samp[i+6];
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sampleData0 = vec_perm(s0,s1,loadPermute0);
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sampleData1 = vec_perm(s0,s1,loadPermute1);
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merge0 = vec_mule(sampleData0,volume_vec);
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merge0 = vec_sra(merge0,volume_shift); /* Shift down to proper range */
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merge1 = vec_mulo(sampleData0,volume_vec);
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merge1 = vec_sra(merge1,volume_shift);
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d0 = vec_add(merge0,d0);
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d1 = vec_add(merge1,d1);
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merge0 = vec_mule(sampleData1,volume_vec);
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merge0 = vec_sra(merge0,volume_shift); /* Shift down to proper range */
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merge1 = vec_mulo(sampleData1,volume_vec);
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merge1 = vec_sra(merge1,volume_shift);
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d2 = vec_add(merge0,d2);
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d3 = vec_add(merge1,d3);
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/* Store destination sample data */
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*(vector signed int *)&samp[i] = d0;
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*(vector signed int *)&samp[i+2] = d1;
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*(vector signed int *)&samp[i+4] = d2;
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*(vector signed int *)&samp[i+6] = d3;
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i += 8;
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vectorCount--;
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s0 = s1;
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sampleOffset += 8;
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}
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if (sampleOffset == SND_CHUNK_SIZE) {
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chunk = chunk->next;
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samples = chunk->sndChunk;
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sampleOffset = 0;
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}
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}
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}
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} else {
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fleftvol = ch->leftvol*snd_vol;
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frightvol = ch->rightvol*snd_vol;
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ooff = sampleOffset;
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samples = chunk->sndChunk;
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for ( i=0 ; i<count ; i++ ) {
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aoff = ooff;
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ooff = ooff + ch->dopplerScale * sc->soundChannels;
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boff = ooff;
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fdata[0] = fdata[1] = 0;
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for (j=aoff; j<boff; j += sc->soundChannels) {
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if (j == SND_CHUNK_SIZE) {
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chunk = chunk->next;
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if (!chunk) {
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chunk = sc->soundData;
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}
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samples = chunk->sndChunk;
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ooff -= SND_CHUNK_SIZE;
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}
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if ( sc->soundChannels == 2 ) {
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fdata[0] += samples[j&(SND_CHUNK_SIZE-1)];
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fdata[1] += samples[(j+1)&(SND_CHUNK_SIZE-1)];
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} else {
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fdata[0] += samples[j&(SND_CHUNK_SIZE-1)];
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fdata[1] += samples[j&(SND_CHUNK_SIZE-1)];
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}
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}
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fdiv = 256 * (boff-aoff) / sc->soundChannels;
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samp[i].left += (fdata[0] * fleftvol)/fdiv;
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samp[i].right += (fdata[1] * frightvol)/fdiv;
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}
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}
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}
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#endif
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static void S_PaintChannelFrom16_scalar( channel_t *ch, const sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
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int data, aoff, boff;
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int leftvol, rightvol;
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int i, j;
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portable_samplepair_t *samp;
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sndBuffer *chunk;
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short *samples;
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float ooff, fdata[2], fdiv, fleftvol, frightvol;
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if (sc->soundChannels <= 0) {
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return;
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}
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samp = &paintbuffer[ bufferOffset ];
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if (ch->doppler) {
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sampleOffset = sampleOffset*ch->oldDopplerScale;
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}
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if ( sc->soundChannels == 2 ) {
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sampleOffset *= sc->soundChannels;
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if ( sampleOffset & 1 ) {
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sampleOffset &= ~1;
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}
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}
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chunk = sc->soundData;
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while (sampleOffset>=SND_CHUNK_SIZE) {
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chunk = chunk->next;
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sampleOffset -= SND_CHUNK_SIZE;
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if (!chunk) {
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chunk = sc->soundData;
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}
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}
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if (!ch->doppler || ch->dopplerScale==1.0f) {
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leftvol = ch->leftvol*snd_vol;
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rightvol = ch->rightvol*snd_vol;
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samples = chunk->sndChunk;
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for ( i=0 ; i<count ; i++ ) {
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data = samples[sampleOffset++];
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samp[i].left += (data * leftvol)>>8;
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if ( sc->soundChannels == 2 ) {
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data = samples[sampleOffset++];
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}
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samp[i].right += (data * rightvol)>>8;
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if (sampleOffset == SND_CHUNK_SIZE) {
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chunk = chunk->next;
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samples = chunk->sndChunk;
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sampleOffset = 0;
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}
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}
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} else {
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fleftvol = ch->leftvol*snd_vol;
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frightvol = ch->rightvol*snd_vol;
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ooff = sampleOffset;
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samples = chunk->sndChunk;
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for ( i=0 ; i<count ; i++ ) {
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aoff = ooff;
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ooff = ooff + ch->dopplerScale * sc->soundChannels;
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boff = ooff;
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fdata[0] = fdata[1] = 0;
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for (j=aoff; j<boff; j += sc->soundChannels) {
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if (j == SND_CHUNK_SIZE) {
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chunk = chunk->next;
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if (!chunk) {
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chunk = sc->soundData;
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}
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samples = chunk->sndChunk;
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ooff -= SND_CHUNK_SIZE;
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}
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if ( sc->soundChannels == 2 ) {
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fdata[0] += samples[j&(SND_CHUNK_SIZE-1)];
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fdata[1] += samples[(j+1)&(SND_CHUNK_SIZE-1)];
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} else {
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fdata[0] += samples[j&(SND_CHUNK_SIZE-1)];
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fdata[1] += samples[j&(SND_CHUNK_SIZE-1)];
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}
|
|
}
|
|
fdiv = 256 * (boff-aoff) / sc->soundChannels;
|
|
samp[i].left += (fdata[0] * fleftvol)/fdiv;
|
|
samp[i].right += (fdata[1] * frightvol)/fdiv;
|
|
}
|
|
}
|
|
}
|
|
|
|
static void S_PaintChannelFrom16( channel_t *ch, const sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
|
|
#if idppc_altivec
|
|
if (com_altivec->integer) {
|
|
// must be in a separate function or G3 systems will crash.
|
|
S_PaintChannelFrom16_altivec( ch, sc, count, sampleOffset, bufferOffset );
|
|
return;
|
|
}
|
|
#endif
|
|
S_PaintChannelFrom16_scalar( ch, sc, count, sampleOffset, bufferOffset );
|
|
}
|
|
|
|
void S_PaintChannelFromWavelet( channel_t *ch, sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
|
|
int data;
|
|
int leftvol, rightvol;
|
|
int i;
|
|
portable_samplepair_t *samp;
|
|
sndBuffer *chunk;
|
|
short *samples;
|
|
|
|
leftvol = ch->leftvol*snd_vol;
|
|
rightvol = ch->rightvol*snd_vol;
|
|
|
|
i = 0;
|
|
samp = &paintbuffer[ bufferOffset ];
|
|
chunk = sc->soundData;
|
|
while (sampleOffset>=(SND_CHUNK_SIZE_FLOAT*4)) {
|
|
chunk = chunk->next;
|
|
sampleOffset -= (SND_CHUNK_SIZE_FLOAT*4);
|
|
i++;
|
|
}
|
|
|
|
if (i!=sfxScratchIndex || sfxScratchPointer != sc) {
|
|
S_AdpcmGetSamples( chunk, sfxScratchBuffer );
|
|
sfxScratchIndex = i;
|
|
sfxScratchPointer = sc;
|
|
}
|
|
|
|
samples = sfxScratchBuffer;
|
|
|
|
for ( i=0 ; i<count ; i++ ) {
|
|
data = samples[sampleOffset++];
|
|
samp[i].left += (data * leftvol)>>8;
|
|
samp[i].right += (data * rightvol)>>8;
|
|
|
|
if (sampleOffset == SND_CHUNK_SIZE*2) {
|
|
chunk = chunk->next;
|
|
decodeWavelet(chunk, sfxScratchBuffer);
|
|
sfxScratchIndex++;
|
|
sampleOffset = 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
void S_PaintChannelFromADPCM( channel_t *ch, sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
|
|
int data;
|
|
int leftvol, rightvol;
|
|
int i;
|
|
portable_samplepair_t *samp;
|
|
sndBuffer *chunk;
|
|
short *samples;
|
|
|
|
leftvol = ch->leftvol*snd_vol;
|
|
rightvol = ch->rightvol*snd_vol;
|
|
|
|
i = 0;
|
|
samp = &paintbuffer[ bufferOffset ];
|
|
chunk = sc->soundData;
|
|
|
|
if (ch->doppler) {
|
|
sampleOffset = sampleOffset*ch->oldDopplerScale;
|
|
}
|
|
|
|
while (sampleOffset>=(SND_CHUNK_SIZE*4)) {
|
|
chunk = chunk->next;
|
|
sampleOffset -= (SND_CHUNK_SIZE*4);
|
|
i++;
|
|
}
|
|
|
|
if (i!=sfxScratchIndex || sfxScratchPointer != sc) {
|
|
S_AdpcmGetSamples( chunk, sfxScratchBuffer );
|
|
sfxScratchIndex = i;
|
|
sfxScratchPointer = sc;
|
|
}
|
|
|
|
samples = sfxScratchBuffer;
|
|
|
|
for ( i=0 ; i<count ; i++ ) {
|
|
data = samples[sampleOffset++];
|
|
samp[i].left += (data * leftvol)>>8;
|
|
samp[i].right += (data * rightvol)>>8;
|
|
|
|
if (sampleOffset == SND_CHUNK_SIZE*4) {
|
|
chunk = chunk->next;
|
|
S_AdpcmGetSamples( chunk, sfxScratchBuffer);
|
|
sampleOffset = 0;
|
|
sfxScratchIndex++;
|
|
}
|
|
}
|
|
}
|
|
|
|
void S_PaintChannelFromMuLaw( channel_t *ch, sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
|
|
int data;
|
|
int leftvol, rightvol;
|
|
int i;
|
|
portable_samplepair_t *samp;
|
|
sndBuffer *chunk;
|
|
byte *samples;
|
|
float ooff;
|
|
|
|
leftvol = ch->leftvol*snd_vol;
|
|
rightvol = ch->rightvol*snd_vol;
|
|
|
|
samp = &paintbuffer[ bufferOffset ];
|
|
chunk = sc->soundData;
|
|
while (sampleOffset>=(SND_CHUNK_SIZE*2)) {
|
|
chunk = chunk->next;
|
|
sampleOffset -= (SND_CHUNK_SIZE*2);
|
|
if (!chunk) {
|
|
chunk = sc->soundData;
|
|
}
|
|
}
|
|
|
|
if (!ch->doppler) {
|
|
samples = (byte *)chunk->sndChunk + sampleOffset;
|
|
for ( i=0 ; i<count ; i++ ) {
|
|
data = mulawToShort[*samples];
|
|
samp[i].left += (data * leftvol)>>8;
|
|
samp[i].right += (data * rightvol)>>8;
|
|
samples++;
|
|
if (chunk != NULL && samples == (byte *)chunk->sndChunk+(SND_CHUNK_SIZE*2)) {
|
|
chunk = chunk->next;
|
|
samples = (byte *)chunk->sndChunk;
|
|
}
|
|
}
|
|
} else {
|
|
ooff = sampleOffset;
|
|
samples = (byte *)chunk->sndChunk;
|
|
for ( i=0 ; i<count ; i++ ) {
|
|
data = mulawToShort[samples[(int)(ooff)]];
|
|
ooff = ooff + ch->dopplerScale;
|
|
samp[i].left += (data * leftvol)>>8;
|
|
samp[i].right += (data * rightvol)>>8;
|
|
if (ooff >= SND_CHUNK_SIZE*2) {
|
|
chunk = chunk->next;
|
|
if (!chunk) {
|
|
chunk = sc->soundData;
|
|
}
|
|
samples = (byte *)chunk->sndChunk;
|
|
ooff = 0.0;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/*
|
|
===================
|
|
S_PaintChannels
|
|
===================
|
|
*/
|
|
void S_PaintChannels( int endtime ) {
|
|
int i;
|
|
int end;
|
|
int stream;
|
|
channel_t *ch;
|
|
sfx_t *sc;
|
|
int ltime, count;
|
|
int sampleOffset;
|
|
|
|
if(s_muted->integer)
|
|
snd_vol = 0;
|
|
else
|
|
snd_vol = s_volume->value*255;
|
|
|
|
//Com_Printf ("%i to %i\n", s_paintedtime, endtime);
|
|
while ( s_paintedtime < endtime ) {
|
|
// if paintbuffer is smaller than DMA buffer
|
|
// we may need to fill it multiple times
|
|
end = endtime;
|
|
if ( endtime - s_paintedtime > PAINTBUFFER_SIZE ) {
|
|
end = s_paintedtime + PAINTBUFFER_SIZE;
|
|
}
|
|
|
|
// clear the paint buffer and mix any raw samples...
|
|
Com_Memset(paintbuffer, 0, sizeof (paintbuffer));
|
|
for (stream = 0; stream < MAX_RAW_STREAMS; stream++) {
|
|
if ( s_rawend[stream] >= s_paintedtime ) {
|
|
// copy from the streaming sound source
|
|
const portable_samplepair_t *rawsamples = s_rawsamples[stream];
|
|
const int stop = (end < s_rawend[stream]) ? end : s_rawend[stream];
|
|
for ( i = s_paintedtime ; i < stop ; i++ ) {
|
|
const int s = i&(MAX_RAW_SAMPLES-1);
|
|
paintbuffer[i-s_paintedtime].left += rawsamples[s].left;
|
|
paintbuffer[i-s_paintedtime].right += rawsamples[s].right;
|
|
}
|
|
}
|
|
}
|
|
|
|
// paint in the channels.
|
|
ch = s_channels;
|
|
for ( i = 0; i < MAX_CHANNELS ; i++, ch++ ) {
|
|
if ( !ch->thesfx || (ch->leftvol<0.25 && ch->rightvol<0.25 )) {
|
|
continue;
|
|
}
|
|
|
|
ltime = s_paintedtime;
|
|
sc = ch->thesfx;
|
|
|
|
if (sc->soundData==NULL || sc->soundLength==0) {
|
|
continue;
|
|
}
|
|
|
|
sampleOffset = ltime - ch->startSample;
|
|
count = end - ltime;
|
|
if ( sampleOffset + count > sc->soundLength ) {
|
|
count = sc->soundLength - sampleOffset;
|
|
}
|
|
|
|
if ( count > 0 ) {
|
|
if( sc->soundCompressionMethod == 1) {
|
|
S_PaintChannelFromADPCM (ch, sc, count, sampleOffset, ltime - s_paintedtime);
|
|
} else if( sc->soundCompressionMethod == 2) {
|
|
S_PaintChannelFromWavelet (ch, sc, count, sampleOffset, ltime - s_paintedtime);
|
|
} else if( sc->soundCompressionMethod == 3) {
|
|
S_PaintChannelFromMuLaw (ch, sc, count, sampleOffset, ltime - s_paintedtime);
|
|
} else {
|
|
S_PaintChannelFrom16 (ch, sc, count, sampleOffset, ltime - s_paintedtime);
|
|
}
|
|
}
|
|
}
|
|
|
|
// paint in the looped channels.
|
|
ch = loop_channels;
|
|
for ( i = 0; i < numLoopChannels ; i++, ch++ ) {
|
|
if ( !ch->thesfx || (!ch->leftvol && !ch->rightvol )) {
|
|
continue;
|
|
}
|
|
|
|
ltime = s_paintedtime;
|
|
sc = ch->thesfx;
|
|
|
|
if (sc->soundData==NULL || sc->soundLength==0) {
|
|
continue;
|
|
}
|
|
// we might have to make two passes if it
|
|
// is a looping sound effect and the end of
|
|
// the sample is hit
|
|
do {
|
|
sampleOffset = (ltime % sc->soundLength);
|
|
|
|
count = end - ltime;
|
|
if ( sampleOffset + count > sc->soundLength ) {
|
|
count = sc->soundLength - sampleOffset;
|
|
}
|
|
|
|
if ( count > 0 ) {
|
|
if( sc->soundCompressionMethod == 1) {
|
|
S_PaintChannelFromADPCM (ch, sc, count, sampleOffset, ltime - s_paintedtime);
|
|
} else if( sc->soundCompressionMethod == 2) {
|
|
S_PaintChannelFromWavelet (ch, sc, count, sampleOffset, ltime - s_paintedtime);
|
|
} else if( sc->soundCompressionMethod == 3) {
|
|
S_PaintChannelFromMuLaw (ch, sc, count, sampleOffset, ltime - s_paintedtime);
|
|
} else {
|
|
S_PaintChannelFrom16 (ch, sc, count, sampleOffset, ltime - s_paintedtime);
|
|
}
|
|
ltime += count;
|
|
}
|
|
} while ( ltime < end);
|
|
}
|
|
|
|
// transfer out according to DMA format
|
|
S_TransferPaintBuffer( end );
|
|
s_paintedtime = end;
|
|
}
|
|
}
|