Xash3D FWGS engine.
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/*
sound.h - sndlib main header
Copyright (C) 2009 Uncle Mike
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
*/
#ifndef SOUND_H
#define SOUND_H
extern poolhandle_t sndpool;
#include "xash3d_mathlib.h"
// sound engine rate defines
#define SOUND_DMA_SPEED 44100 // hardware playback rate
#define SOUND_11k 11025 // 11khz sample rate
#define SOUND_16k 16000 // 16khz sample rate
#define SOUND_22k 22050 // 22khz sample rate
#define SOUND_32k 32000 // 32khz sample rate
#define SOUND_44k 44100 // 44khz sample rate
#define SOUND_48k 48000 // 48khz sample rate
#define DMA_MSEC_PER_SAMPLE ((float)(1000.0 / SOUND_DMA_SPEED))
// fixed point stuff for real-time resampling
#define FIX_BITS 28
#define FIX_SCALE (1 << FIX_BITS)
#define FIX_MASK ((1 << FIX_BITS)-1)
#define FIX_FLOAT(a) ((int)((a) * FIX_SCALE))
#define FIX(a) (((int)(a)) << FIX_BITS)
#define FIX_INTPART(a) (((int)(a)) >> FIX_BITS)
#define FIX_FRACTION(a,b) (FIX(a)/(b))
#define FIX_FRACPART(a) ((a) & FIX_MASK)
// NOTE: clipped sound at 32760 to avoid overload
#define CLIP( x ) (( x ) > 32760 ? 32760 : (( x ) < -32760 ? -32760 : ( x )))
#define PAINTBUFFER_SIZE 1024 // 44k: was 512
#define PAINTBUFFER (g_curpaintbuffer)
#define CPAINTBUFFERS 3
// sound mixing buffer
#define CPAINTFILTERMEM 3
#define CPAINTFILTERS 4 // maximum number of consecutive upsample passes per paintbuffer
#define S_RAW_SOUND_IDLE_SEC 10 // time interval for idling raw sound before it's freed
#define S_RAW_SOUND_BACKGROUNDTRACK -2
#define S_RAW_SOUND_SOUNDTRACK -1
#define S_RAW_SAMPLES_PRECISION_BITS 14
#define CIN_FRAMETIME (1.0f / 30.0f)
typedef struct
{
int left;
int right;
} portable_samplepair_t;
typedef struct
{
qboolean factive; // if true, mix to this paintbuffer using flags
portable_samplepair_t *pbuf; // front stereo mix buffer, for 2 or 4 channel mixing
int ifilter; // current filter memory buffer to use for upsampling pass
portable_samplepair_t fltmem[CPAINTFILTERS][CPAINTFILTERMEM];
} paintbuffer_t;
typedef struct sfx_s
{
char name[MAX_QPATH];
wavdata_t *cache;
int servercount;
uint hashValue;
struct sfx_s *hashNext;
} sfx_t;
extern portable_samplepair_t paintbuffer[];
extern portable_samplepair_t roombuffer[];
extern portable_samplepair_t temppaintbuffer[];
extern portable_samplepair_t *g_curpaintbuffer;
extern paintbuffer_t paintbuffers[];
// structure used for fading in and out client sound volume.
typedef struct
{
float initial_percent;
float percent; // how far to adjust client's volume down by.
float starttime; // GetHostTime() when we started adjusting volume
float fadeouttime; // # of seconds to get to faded out state
float holdtime; // # of seconds to hold
float fadeintime; // # of seconds to restore
} soundfade_t;
typedef struct
{
float percent;
} musicfade_t;
typedef struct snd_format_s
{
unsigned int speed;
unsigned int width;
unsigned int channels;
} snd_format_t;
typedef struct
{
snd_format_t format;
int samples; // mono samples in buffer
int samplepos; // in mono samples
byte *buffer;
qboolean initialized; // sound engine is active
const char *backendName;
} dma_t;
#include "vox.h"
typedef struct
{
double sample;
wavdata_t *pData;
double forcedEndSample;
qboolean finished;
} mixer_t;
typedef struct rawchan_s
{
int entnum;
int master_vol;
int leftvol; // 0-255 left volume
int rightvol; // 0-255 right volume
float dist_mult; // distance multiplier (attenuation/clipK)
vec3_t origin; // only use if fixed_origin is set
volatile uint s_rawend;
wavdata_t sound_info; // advance play position
float oldtime; // catch time jumps
size_t max_samples; // buffer length
portable_samplepair_t rawsamples[1]; // variable sized
} rawchan_t;
typedef struct channel_s
{
char name[16]; // keept sentence name
sfx_t *sfx; // sfx number
int leftvol; // 0-255 left volume
int rightvol; // 0-255 right volume
int entnum; // entity soundsource
int entchannel; // sound channel (CHAN_STREAM, CHAN_VOICE, etc.)
vec3_t origin; // only use if fixed_origin is set
float dist_mult; // distance multiplier (attenuation/clipK)
int master_vol; // 0-255 master volume
qboolean isSentence; // bit who indicated sentence
int basePitch; // base pitch percent (100% is normal pitch playback)
float pitch; // real-time pitch after any modulation or shift by dynamic data
qboolean use_loop; // don't loop default and local sounds
qboolean staticsound; // use origin instead of fetching entnum's origin
qboolean localsound; // it's a local menu sound (not looped, not paused)
mixer_t pMixer;
// sentence mixer
int wordIndex;
mixer_t *currentWord; // NULL if sentence is finished
voxword_t words[CVOXWORDMAX];
} channel_t;
typedef struct
{
vec3_t origin; // simorg + view_ofs
vec3_t velocity;
vec3_t forward;
vec3_t right;
vec3_t up;
int entnum;
int waterlevel;
float frametime; // used for sound fade
qboolean active;
qboolean inmenu; // listener in-menu ?
qboolean paused;
qboolean streaming; // playing AVI-file
qboolean stream_paused; // pause only background track
} listener_t;
typedef struct
{
string current; // a currently playing track
string loopName; // may be empty
stream_t *stream;
int source; // may be game, menu, etc
} bg_track_t;
//====================================================================
#define MAX_DYNAMIC_CHANNELS (60 + NUM_AMBIENTS)
#define MAX_CHANNELS (256 + MAX_DYNAMIC_CHANNELS) // Scourge Of Armagon has too many static sounds on hip2m4.bsp
#define MAX_RAW_CHANNELS 48
#define MAX_RAW_SAMPLES 8192
extern sound_t ambient_sfx[NUM_AMBIENTS];
extern qboolean snd_ambient;
extern channel_t channels[MAX_CHANNELS];
extern rawchan_t *raw_channels[MAX_RAW_CHANNELS];
extern int total_channels;
extern int paintedtime;
extern int soundtime;
extern listener_t s_listener;
extern int idsp_room;
extern dma_t dma;
extern convar_t s_musicvolume;
extern convar_t s_lerping;
extern convar_t *dsp_off;
extern convar_t s_test; // cvar to testify new effects
extern convar_t s_samplecount;
extern convar_t snd_mute_losefocus;
extern convar_t s_warn_late_precache;
void S_InitScaletable( void );
wavdata_t *S_LoadSound( sfx_t *sfx );
float S_GetMasterVolume( void );
float S_GetMusicVolume( void );
//
// s_main.c
//
void S_FreeChannel( channel_t *ch );
//
// s_mix.c
//
int S_MixDataToDevice( channel_t *pChannel, int sampleCount, int outputRate, int outputOffset, int timeCompress );
void MIX_ClearAllPaintBuffers( int SampleCount, qboolean clearFilters );
void MIX_InitAllPaintbuffers( void );
void MIX_FreeAllPaintbuffers( void );
void MIX_PaintChannels( int endtime );
// s_load.c
qboolean S_TestSoundChar( const char *pch, char c );
char *S_SkipSoundChar( const char *pch );
sfx_t *S_FindName( const char *name, int *pfInCache );
sound_t S_RegisterSound( const char *name );
void S_FreeSound( sfx_t *sfx );
void S_InitSounds( void );
// s_dsp.c
void SX_Init( void );
void SX_Free( void );
void CheckNewDspPresets( void );
void DSP_Process( int idsp, portable_samplepair_t *pbfront, int sampleCount );
float DSP_GetGain( int idsp );
void DSP_ClearState( void );
qboolean S_Init( void );
void S_Shutdown( void );
void S_SoundList_f( void );
void S_SoundInfo_f( void );
struct ref_viewpass_s;
channel_t *SND_PickDynamicChannel( int entnum, int channel, sfx_t *sfx, qboolean *ignore );
channel_t *SND_PickStaticChannel( const vec3_t pos, sfx_t *sfx );
int S_GetCurrentStaticSounds( soundlist_t *pout, int size );
int S_GetCurrentDynamicSounds( soundlist_t *pout, int size );
sfx_t *S_GetSfxByHandle( sound_t handle );
rawchan_t *S_FindRawChannel( int entnum, qboolean create );
void S_RawEntSamples( int entnum, uint samples, uint rate, word width, word channels, const byte *data, int snd_vol );
void S_RawSamples( uint samples, uint rate, word width, word channels, const byte *data, int entnum );
void S_StopSound( int entnum, int channel, const char *soundname );
void S_UpdateFrame( struct ref_viewpass_s *rvp );
uint S_GetRawSamplesLength( int entnum );
void S_ClearRawChannel( int entnum );
void S_StopAllSounds( qboolean ambient );
void S_FreeSounds( void );
//
// s_mouth.c
//
void SND_InitMouth( int entnum, int entchannel );
void SND_ForceInitMouth( int entnum );
void SND_MoveMouth8( channel_t *ch, wavdata_t *pSource, int count );
void SND_MoveMouth16( channel_t *ch, wavdata_t *pSource, int count );
void SND_MoveMouthRaw( rawchan_t *ch, portable_samplepair_t *pData, int count );
void SND_CloseMouth( channel_t *ch );
void SND_ForceCloseMouth( int entnum );
//
// s_stream.c
//
void S_StreamSoundTrack( void );
void S_StreamBackgroundTrack( void );
qboolean S_StreamGetCurrentState( char *currentTrack, char *loopTrack, int *position );
void S_PrintBackgroundTrackState( void );
void S_FadeMusicVolume( float fadePercent );
//
// s_utils.c
//
int S_ZeroCrossingAfter( wavdata_t *pWaveData, int sample );
int S_ZeroCrossingBefore( wavdata_t *pWaveData, int sample );
int S_GetOutputData( wavdata_t *pSource, void **pData, int samplePosition, int sampleCount, qboolean use_loop );
void S_SetSampleStart( channel_t *pChan, wavdata_t *pSource, int newPosition );
void S_SetSampleEnd( channel_t *pChan, wavdata_t *pSource, int newEndPosition );
//
// s_vox.c
//
void VOX_Init( void );
void VOX_Shutdown( void );
void VOX_SetChanVol( channel_t *ch );
void VOX_LoadSound( channel_t *pchan, const char *psz );
float VOX_ModifyPitch( channel_t *ch, float pitch );
int VOX_MixDataToDevice( channel_t *pChannel, int sampleCount, int outputRate, int outputOffset );
#endif//SOUND_H