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https://github.com/Q3Rally-Team/q3rally.git
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6bc3b33eab
Fix axis returned by IQM's LerpTag Calculate bounds for unanimated IQM models Fix loading favorites as initial source in server browser Improve finding obelisk entitynum for bot AI Fix SDL audio playback with surround sound Fix predicting entity origin on rotating mover Allow binds to use hex values for all key codes Disable pulseaudio capture regardless of SDL version Fix SDL audio playback with 16-bit stereo sound Make s_info command display channels instead of stereo Fix cross-compiling using mingw-w64 on Ubuntu 18.04
627 lines
15 KiB
C
627 lines
15 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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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.fullsamples;
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snd_out = (short *) pbuf + (lpos<<1); // lpos * dma.channels
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snd_linear_count = dma.fullsamples - 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; // snd_linear_count *= dma.channels
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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); // snd_linear_count / dma.channels
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if( CL_VideoRecording( ) )
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CL_WriteAVIAudioFrame( (byte *)snd_out, snd_linear_count << 1 ); // snd_linear_count * (dma.samplebits/8)
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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 *p;
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int step;
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int val;
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int i;
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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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// 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_idx = (s_paintedtime * dma.channels) % dma.samples;
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step = 3 - MIN(dma.channels, 2);
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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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for (i=0 ; i<count ; i++)
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{
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if ((i % dma.channels) >= 2)
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{
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val = 0;
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}
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else
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{
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val = *p >> 8;
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p+= step;
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}
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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) % dma.samples;
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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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for (i=0 ; i<count ; i++)
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{
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if ((i % dma.channels) >= 2)
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{
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val = 0;
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}
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else
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{
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val = *p >> 8;
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p+= step;
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}
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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) % dma.samples;
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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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for (i=0 ; i<count ; i++)
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{
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if ((i % dma.channels) >= 2)
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{
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val = 0;
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}
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else
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{
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val = *p >> 8;
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p+= step;
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}
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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) % dma.samples;
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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_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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}
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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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static void S_PaintChannelFrom16( channel_t *ch, const sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
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#if idppc_altivec
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if (com_altivec->integer) {
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// must be in a separate translation unit or G3 systems will crash.
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S_PaintChannelFrom16_altivec( paintbuffer, snd_vol, ch, sc, count, sampleOffset, bufferOffset );
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return;
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}
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#endif
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S_PaintChannelFrom16_scalar( ch, sc, count, sampleOffset, bufferOffset );
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}
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void S_PaintChannelFromWavelet( channel_t *ch, sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
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int data;
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int leftvol, rightvol;
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int i;
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portable_samplepair_t *samp;
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sndBuffer *chunk;
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short *samples;
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leftvol = ch->leftvol*snd_vol;
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rightvol = ch->rightvol*snd_vol;
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i = 0;
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samp = &paintbuffer[ bufferOffset ];
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chunk = sc->soundData;
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while (sampleOffset>=(SND_CHUNK_SIZE_FLOAT*4)) {
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chunk = chunk->next;
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sampleOffset -= (SND_CHUNK_SIZE_FLOAT*4);
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i++;
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}
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if (i!=sfxScratchIndex || sfxScratchPointer != sc) {
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S_AdpcmGetSamples( chunk, sfxScratchBuffer );
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sfxScratchIndex = i;
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sfxScratchPointer = sc;
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}
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samples = sfxScratchBuffer;
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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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samp[i].right += (data * rightvol)>>8;
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if (sampleOffset == SND_CHUNK_SIZE*2) {
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chunk = chunk->next;
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decodeWavelet(chunk, sfxScratchBuffer);
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sfxScratchIndex++;
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sampleOffset = 0;
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}
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}
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}
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void S_PaintChannelFromADPCM( channel_t *ch, sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
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int data;
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int leftvol, rightvol;
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int i;
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portable_samplepair_t *samp;
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sndBuffer *chunk;
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short *samples;
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leftvol = ch->leftvol*snd_vol;
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rightvol = ch->rightvol*snd_vol;
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i = 0;
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samp = &paintbuffer[ bufferOffset ];
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chunk = sc->soundData;
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if (ch->doppler) {
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sampleOffset = sampleOffset*ch->oldDopplerScale;
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}
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while (sampleOffset>=(SND_CHUNK_SIZE*4)) {
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chunk = chunk->next;
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sampleOffset -= (SND_CHUNK_SIZE*4);
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i++;
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}
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if (i!=sfxScratchIndex || sfxScratchPointer != sc) {
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S_AdpcmGetSamples( chunk, sfxScratchBuffer );
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sfxScratchIndex = i;
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sfxScratchPointer = sc;
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}
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samples = sfxScratchBuffer;
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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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samp[i].right += (data * rightvol)>>8;
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if (sampleOffset == SND_CHUNK_SIZE*4) {
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chunk = chunk->next;
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S_AdpcmGetSamples( chunk, sfxScratchBuffer);
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sampleOffset = 0;
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sfxScratchIndex++;
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}
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}
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}
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void S_PaintChannelFromMuLaw( channel_t *ch, sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
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int data;
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int leftvol, rightvol;
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int i;
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portable_samplepair_t *samp;
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sndBuffer *chunk;
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byte *samples;
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float ooff;
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leftvol = ch->leftvol*snd_vol;
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rightvol = ch->rightvol*snd_vol;
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samp = &paintbuffer[ bufferOffset ];
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chunk = sc->soundData;
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while (sampleOffset>=(SND_CHUNK_SIZE*2)) {
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chunk = chunk->next;
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sampleOffset -= (SND_CHUNK_SIZE*2);
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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) {
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samples = (byte *)chunk->sndChunk + sampleOffset;
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for ( i=0 ; i<count ; i++ ) {
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data = mulawToShort[*samples];
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samp[i].left += (data * leftvol)>>8;
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samp[i].right += (data * rightvol)>>8;
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samples++;
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if (chunk != NULL && samples == (byte *)chunk->sndChunk+(SND_CHUNK_SIZE*2)) {
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chunk = chunk->next;
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samples = (byte *)chunk->sndChunk;
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}
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}
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} else {
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ooff = sampleOffset;
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samples = (byte *)chunk->sndChunk;
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for ( i=0 ; i<count ; i++ ) {
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data = mulawToShort[samples[(int)(ooff)]];
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ooff = ooff + ch->dopplerScale;
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samp[i].left += (data * leftvol)>>8;
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samp[i].right += (data * rightvol)>>8;
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if (ooff >= SND_CHUNK_SIZE*2) {
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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 = (byte *)chunk->sndChunk;
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ooff = 0.0;
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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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S_PaintChannels
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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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int stream;
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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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if(s_muted->integer)
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snd_vol = 0;
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else
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snd_vol = s_volume->value*255;
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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 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;
|
|
}
|
|
}
|