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https://github.com/w23/xash3d-fwgs
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361 lines
12 KiB
C
361 lines
12 KiB
C
/*
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sound.h - sndlib main header
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Copyright (C) 2009 Uncle Mike
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This program is free software: you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation, either version 3 of the License, or
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(at your option) any later version.
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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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*/
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#ifndef SOUND_H
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#define SOUND_H
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extern byte *sndpool;
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#include "mathlib.h"
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// sound engine rate defines
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#define SOUND_DMA_SPEED 44100 // hardware playback rate
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#define SOUND_11k 11025 // 11khz sample rate
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#define SOUND_16k 16000 // 16khz sample rate
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#define SOUND_22k 22050 // 22khz sample rate
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#define SOUND_32k 32000 // 32khz sample rate
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#define SOUND_44k 44100 // 44khz sample rate
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#define DMA_MSEC_PER_SAMPLE ((float)(1000.0 / SOUND_DMA_SPEED))
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#define SND_TRACE_UPDATE_MAX 2 // max of N channels may be checked for obscured source per frame
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#define SND_RADIUS_MAX 240.0f // max sound source radius
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#define SND_RADIUS_MIN 24.0f // min sound source radius
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#define SND_OBSCURED_LOSS_DB -2.70f // dB loss due to obscured sound source
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// calculate gain based on atmospheric attenuation.
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// as gain excedes threshold, round off (compress) towards 1.0 using spline
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#define SND_GAIN_COMP_EXP_MAX 2.5f // Increasing SND_GAIN_COMP_EXP_MAX fits compression curve
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// more closely to original gain curve as it approaches 1.0.
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#define SND_GAIN_FADE_TIME 0.25f // xfade seconds between obscuring gain changes
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#define SND_GAIN_COMP_EXP_MIN 0.8f
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#define SND_GAIN_COMP_THRESH 0.5f // gain value above which gain curve is rounded to approach 1.0
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#define SND_DB_MAX 140.0f // max db of any sound source
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#define SND_DB_MED 90.0f // db at which compression curve changes
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#define SND_DB_MIN 60.0f // min db of any sound source
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#define SND_GAIN_PLAYER_WEAPON_DB 2.0f // increase player weapon gain by N dB
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// fixed point stuff for real-time resampling
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#define FIX_BITS 28
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#define FIX_SCALE (1 << FIX_BITS)
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#define FIX_MASK ((1 << FIX_BITS)-1)
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#define FIX_FLOAT(a) ((int)((a) * FIX_SCALE))
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#define FIX(a) (((int)(a)) << FIX_BITS)
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#define FIX_INTPART(a) (((int)(a)) >> FIX_BITS)
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#define FIX_FRACTION(a,b) (FIX(a)/(b))
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#define FIX_FRACPART(a) ((a) & FIX_MASK)
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#define SNDLVL_TO_DIST_MULT( sndlvl ) \
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( sndlvl ? ((pow( 10, s_refdb->value / 20 ) / pow( 10, (float)sndlvl / 20 )) / s_refdist->value ) : 0 )
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#define DIST_MULT_TO_SNDLVL( dist_mult ) \
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(int)( dist_mult ? ( 20 * log10( pow( 10, s_refdb->value / 20 ) / (dist_mult * s_refdist->value ))) : 0 )
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// NOTE: clipped sound at 32760 to avoid overload
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#define CLIP( x ) (( x ) > 32760 ? 32760 : (( x ) < -32760 ? -32760 : ( x )))
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#define SWAP( a, b, t ) {(t) = (a); (a) = (b); (b) = (t);}
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#define AVG( a, b ) (((a) + (b)) >> 1 )
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#define AVG4( a, b, c, d ) (((a) + (b) + (c) + (d)) >> 2 )
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#define PAINTBUFFER_SIZE 1024 // 44k: was 512
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#define PAINTBUFFER (g_curpaintbuffer)
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#define CPAINTBUFFERS 3
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// sound mixing buffer
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#define CPAINTFILTERMEM 3
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#define CPAINTFILTERS 4 // maximum number of consecutive upsample passes per paintbuffer
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#define S_RAW_SOUND_IDLE_SEC 10 // time interval for idling raw sound before it's freed
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#define S_RAW_SOUND_BACKGROUNDTRACK -2
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#define S_RAW_SOUND_SOUNDTRACK -1
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#define S_RAW_SAMPLES_PRECISION_BITS 14
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#define CIN_FRAMETIME (1.0f / 30.0f)
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typedef struct
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{
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int left;
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int right;
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} portable_samplepair_t;
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typedef struct
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{
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qboolean factive; // if true, mix to this paintbuffer using flags
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portable_samplepair_t *pbuf; // front stereo mix buffer, for 2 or 4 channel mixing
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int ifilter; // current filter memory buffer to use for upsampling pass
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portable_samplepair_t fltmem[CPAINTFILTERS][CPAINTFILTERMEM];
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} paintbuffer_t;
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typedef struct sfx_s
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{
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char name[MAX_QPATH];
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wavdata_t *cache;
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int servercount;
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uint hashValue;
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struct sfx_s *hashNext;
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} sfx_t;
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extern portable_samplepair_t paintbuffer[];
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extern portable_samplepair_t roombuffer[];
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extern portable_samplepair_t temppaintbuffer[];
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extern portable_samplepair_t *g_curpaintbuffer;
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extern paintbuffer_t paintbuffers[];
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// structure used for fading in and out client sound volume.
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typedef struct
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{
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float initial_percent;
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float percent; // how far to adjust client's volume down by.
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float starttime; // GetHostTime() when we started adjusting volume
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float fadeouttime; // # of seconds to get to faded out state
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float holdtime; // # of seconds to hold
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float fadeintime; // # of seconds to restore
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} soundfade_t;
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typedef struct
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{
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float percent;
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} musicfade_t;
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typedef struct snd_format_s
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{
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unsigned int speed;
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unsigned int width;
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unsigned int channels;
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} snd_format_t;
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typedef struct
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{
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snd_format_t format;
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int samples; // mono samples in buffer
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int samplepos; // in mono samples
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byte *buffer;
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qboolean initialized; // sound engine is active
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} dma_t;
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#include "vox.h"
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typedef struct
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{
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double sample;
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wavdata_t *pData;
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double forcedEndSample;
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qboolean finished;
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} mixer_t;
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typedef struct rawchan_s
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{
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int entnum;
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int master_vol;
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int leftvol; // 0-255 left volume
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int rightvol; // 0-255 right volume
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float dist_mult; // distance multiplier (attenuation/clipK)
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vec3_t origin; // only use if fixed_origin is set
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float radius; // radius of this sound effect
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volatile uint s_rawend;
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wavdata_t sound_info; // advance play position
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float oldtime; // catch time jumps
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size_t max_samples; // buffer length
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portable_samplepair_t rawsamples[1]; // variable sized
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} rawchan_t;
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typedef struct channel_s
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{
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char name[16]; // keept sentence name
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sfx_t *sfx; // sfx number
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int leftvol; // 0-255 left volume
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int rightvol; // 0-255 right volume
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int entnum; // entity soundsource
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int entchannel; // sound channel (CHAN_STREAM, CHAN_VOICE, etc.)
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vec3_t origin; // only use if fixed_origin is set
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float dist_mult; // distance multiplier (attenuation/clipK)
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int master_vol; // 0-255 master volume
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qboolean isSentence; // bit who indicated sentence
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int basePitch; // base pitch percent (100% is normal pitch playback)
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float pitch; // real-time pitch after any modulation or shift by dynamic data
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qboolean use_loop; // don't loop default and local sounds
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qboolean staticsound; // use origin instead of fetching entnum's origin
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qboolean localsound; // it's a local menu sound (not looped, not paused)
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mixer_t pMixer;
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// sound culling
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qboolean bfirstpass; // true if this is first time sound is spatialized
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float ob_gain; // gain drop if sound source obscured from listener
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float ob_gain_target; // target gain while crossfading between ob_gain & ob_gain_target
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float ob_gain_inc; // crossfade increment
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qboolean bTraced; // true if channel was already checked this frame for obscuring
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float radius; // radius of this sound effect
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vec3_t absmin, absmax; // filled in CL_GetEntitySpatialization
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int movetype; // to determine point entities
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// sentence mixer
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int wordIndex;
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mixer_t *currentWord; // NULL if sentence is finished
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voxword_t words[CVOXWORDMAX];
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} channel_t;
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typedef struct
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{
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vec3_t origin; // simorg + view_ofs
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vec3_t velocity;
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vec3_t forward;
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vec3_t right;
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vec3_t up;
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int entnum;
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int waterlevel;
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float frametime; // used for sound fade
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qboolean active;
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qboolean inmenu; // listener in-menu ?
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qboolean paused;
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qboolean streaming; // playing AVI-file
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qboolean stream_paused; // pause only background track
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byte pasbytes[(MAX_MAP_LEAFS+7)/8];// actual PHS for current frame
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} listener_t;
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typedef struct
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{
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string current; // a currently playing track
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string loopName; // may be empty
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stream_t *stream;
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int source; // may be game, menu, etc
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} bg_track_t;
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//====================================================================
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#define MAX_DYNAMIC_CHANNELS (60 + NUM_AMBIENTS)
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#define MAX_CHANNELS (256 + MAX_DYNAMIC_CHANNELS) // Scourge Of Armagon has too many static sounds on hip2m4.bsp
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#define MAX_RAW_CHANNELS 16
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#define MAX_RAW_SAMPLES 8192
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extern sound_t ambient_sfx[NUM_AMBIENTS];
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extern qboolean snd_ambient;
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extern channel_t channels[MAX_CHANNELS];
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extern rawchan_t *raw_channels[MAX_RAW_CHANNELS];
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extern int total_channels;
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extern int paintedtime;
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extern int soundtime;
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extern listener_t s_listener;
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extern int idsp_room;
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extern dma_t dma;
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extern convar_t *s_volume;
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extern convar_t *s_musicvolume;
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extern convar_t *s_show;
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extern convar_t *s_mixahead;
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extern convar_t *s_lerping;
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extern convar_t *dsp_off;
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extern convar_t *s_test; // cvar to testify new effects
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extern convar_t *s_samplecount;
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extern convar_t *snd_mute_losefocus;
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void S_InitScaletable( void );
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wavdata_t *S_LoadSound( sfx_t *sfx );
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float S_GetMasterVolume( void );
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float S_GetMusicVolume( void );
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//
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// s_main.c
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//
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void S_FreeChannel( channel_t *ch );
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//
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// s_mix.c
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//
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int S_MixDataToDevice( channel_t *pChannel, int sampleCount, int outputRate, int outputOffset, int timeCompress );
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void MIX_ClearAllPaintBuffers( int SampleCount, qboolean clearFilters );
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void MIX_InitAllPaintbuffers( void );
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void MIX_FreeAllPaintbuffers( void );
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void MIX_PaintChannels( int endtime );
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// s_load.c
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qboolean S_TestSoundChar( const char *pch, char c );
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char *S_SkipSoundChar( const char *pch );
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sfx_t *S_FindName( const char *name, int *pfInCache );
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sound_t S_RegisterSound( const char *name );
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void S_FreeSound( sfx_t *sfx );
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void S_InitSounds( void );
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// s_dsp.c
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void SX_Init( void );
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void SX_Free( void );
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void CheckNewDspPresets( void );
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void DSP_Process( int idsp, portable_samplepair_t *pbfront, int sampleCount );
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float DSP_GetGain( int idsp );
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void DSP_ClearState( void );
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qboolean S_Init( void );
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void S_Shutdown( void );
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void S_Activate( qboolean active );
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void S_SoundList_f( void );
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void S_SoundInfo_f( void );
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struct ref_viewpass_s;
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channel_t *SND_PickDynamicChannel( int entnum, int channel, sfx_t *sfx, qboolean *ignore );
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channel_t *SND_PickStaticChannel( const vec3_t pos, sfx_t *sfx );
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int S_GetCurrentStaticSounds( soundlist_t *pout, int size );
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int S_GetCurrentDynamicSounds( soundlist_t *pout, int size );
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sfx_t *S_GetSfxByHandle( sound_t handle );
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rawchan_t *S_FindRawChannel( int entnum, qboolean create );
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void S_RawSamples( uint samples, uint rate, word width, word channels, const byte *data, int entnum );
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void S_StopSound( int entnum, int channel, const char *soundname );
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void S_UpdateFrame( struct ref_viewpass_s *rvp );
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uint S_GetRawSamplesLength( int entnum );
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void S_ClearRawChannel( int entnum );
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void S_StopAllSounds( qboolean ambient );
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void S_FreeSounds( void );
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//
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// s_mouth.c
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//
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void SND_InitMouth( int entnum, int entchannel );
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void SND_MoveMouth8( channel_t *ch, wavdata_t *pSource, int count );
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void SND_MoveMouth16( channel_t *ch, wavdata_t *pSource, int count );
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void SND_CloseMouth( channel_t *ch );
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//
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// s_stream.c
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//
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void S_StreamSoundTrack( void );
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void S_StreamBackgroundTrack( void );
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qboolean S_StreamGetCurrentState( char *currentTrack, char *loopTrack, int *position );
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void S_PrintBackgroundTrackState( void );
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void S_FadeMusicVolume( float fadePercent );
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//
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// s_utils.c
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//
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int S_ZeroCrossingAfter( wavdata_t *pWaveData, int sample );
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int S_ZeroCrossingBefore( wavdata_t *pWaveData, int sample );
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int S_GetOutputData( wavdata_t *pSource, void **pData, int samplePosition, int sampleCount, qboolean use_loop );
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void S_SetSampleStart( channel_t *pChan, wavdata_t *pSource, int newPosition );
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void S_SetSampleEnd( channel_t *pChan, wavdata_t *pSource, int newEndPosition );
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//
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// s_vox.c
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//
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void VOX_Init( void );
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void VOX_Shutdown( void );
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void VOX_SetChanVol( channel_t *ch );
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void VOX_LoadSound( channel_t *pchan, const char *psz );
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float VOX_ModifyPitch( channel_t *ch, float pitch );
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int VOX_MixDataToDevice( channel_t *pChannel, int sampleCount, int outputRate, int outputOffset );
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#endif//SOUND_H
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