e0aab1c134
fix issue where ftenq protocol positions were sticking. fixed manifest file windowless infinite wait. Added support for ^[\img\foo^] images on the console if they are at the start of the line. Console objects may now be drawn by csqc. Key input is blocked for now, lest csqc get access to things its not allowed to access. Will add provisions later. fixed sound menu crash. added ultrawide speex codec. clamp micamp levels to try to boost audio quality/audiability. rtp control plugins are now allowed to use opus, speex support is correctly queried instead of assumed. WriteFloat now works properly, and not just for ents. no longer sending prespawn/precaches/signon commands to clients that do not expect them. fix sql column header queries. more verbose about active nq protocol verions. -condebug overrides config. git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@4461 fc73d0e0-1445-4013-8a0c-d673dee63da5
322 lines
11 KiB
C
322 lines
11 KiB
C
/*
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Copyright (C) 1996-1997 Id Software, Inc.
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This program is free software; you can redistribute it and/or
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modify it under the terms of the GNU General Public License
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as published by the Free Software Foundation; either version 2
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of the License, or (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.
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See the GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, write to the Free Software
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Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*/
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// sound.h -- client sound i/o functions
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#ifndef __SOUND__
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#define __SOUND__
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// !!! if this is changed, it much be changed in asm_i386.h too !!!
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#define MAXSOUNDCHANNELS 8 //on a per device basis
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// !!! if this is changed, it much be changed in asm_i386.h too !!!
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struct sfx_s;
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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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*/
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typedef struct
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{
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int s[MAXSOUNDCHANNELS];
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} portable_samplegroup_t;
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typedef struct {
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struct sfxcache_s *(*decodedata) (struct sfx_s *sfx, struct sfxcache_s *buf, int start, int length); //retrurn true when done.
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void (*abort) (struct sfx_s *sfx); //it's not playing elsewhere. free entirly
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void *buf;
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} sfxdecode_t;
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typedef struct sfx_s
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{
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char name[MAX_OSPATH];
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#ifdef AVAIL_OPENAL
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unsigned int openal_buffer;
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#endif
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qboolean failedload:1; //no more super-spammy
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qboolean touched:1; //if the sound is still relevent
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sfxdecode_t decoder;
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} sfx_t;
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// !!! if this is changed, it much be changed in asm_i386.h too !!!
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typedef struct sfxcache_s
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{
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unsigned int length; //sample count
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unsigned int loopstart; //-1 or sample index to begin looping at once the sample ends
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unsigned int speed;
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unsigned int width;
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unsigned int numchannels;
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unsigned int soundoffset; //byte index into the sound
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qbyte *data; // variable sized
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} sfxcache_t;
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typedef struct
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{
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// qboolean gamealive;
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// qboolean soundalive;
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// qboolean splitbuffer;
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int numchannels; // this many samples per frame
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int samples; // mono samples in buffer (individual, non grouped)
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// int submission_chunk; // don't mix less than this #
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int samplepos; // in mono samples
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int samplebits;
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int speed; // this many frames per second
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unsigned char *buffer; // pointer to mixed pcm buffer (not directly used by mixer)
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} dma_t;
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#define PITCHSHIFT 6 /*max audio file length = (1<<32>>PITCHSHIFT)/KHZ*/
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#define CF_ABSVOLUME 1 // ignores volume cvar.
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typedef struct
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{
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sfx_t *sfx; // sfx number
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int vol[MAXSOUNDCHANNELS]; // volume, .8 fixed point.
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int start; // start time in global paintsamples
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int pos; // sample position in sfx, <0 means delay sound start (shifted up by 8)
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int rate; // 24.8 fixed point rate scaling
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int flags; // cf_ flags
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int looping; // where to loop, -1 = no looping
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int entnum; // to allow overriding a specific sound
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int entchannel; //int audio_fd
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vec3_t origin; // origin of sound effect
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vec_t dist_mult; // distance multiplier (attenuation/clipK)
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int master_vol; // 0-255 master volume
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} channel_t;
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typedef struct
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{
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int rate;
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int width;
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int numchannels;
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int loopstart;
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int samples;
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int dataofs; // chunk starts this many bytes from file start
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} wavinfo_t;
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struct soundcardinfo_s;
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typedef struct soundcardinfo_s soundcardinfo_t;
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void S_Init (void);
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void S_Startup (void);
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void S_Shutdown (qboolean final);
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void S_StartSound (int entnum, int entchannel, sfx_t *sfx, vec3_t origin, float fvol, float attenuation, float timeofs, float pitchadj);
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void S_StaticSound (sfx_t *sfx, vec3_t origin, float vol, float attenuation);
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void S_StopSound (int entnum, int entchannel);
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void S_StopAllSounds(qboolean clear);
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void S_UpdateListener(vec3_t origin, vec3_t forward, vec3_t right, vec3_t up);
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void S_GetListenerInfo(float *origin, float *forward, float *right, float *up);
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void S_Update (void);
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void S_ExtraUpdate (void);
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void S_MixerThread(soundcardinfo_t *sc);
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void S_Purge(qboolean retaintouched);
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qboolean S_HaveOutput(void);
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void S_Music_Clear(sfx_t *onlyifsample);
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void S_Music_Seek(float time);
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sfx_t *S_PrecacheSound (char *sample);
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void S_TouchSound (char *sample);
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void S_UntouchAll(void);
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void S_ClearPrecache (void);
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void S_BeginPrecaching (void);
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void S_EndPrecaching (void);
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void S_PaintChannels(soundcardinfo_t *sc, int endtime);
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void S_InitPaintChannels (soundcardinfo_t *sc);
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void S_ShutdownCard (soundcardinfo_t *sc);
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void S_DefaultSpeakerConfiguration(soundcardinfo_t *sc);
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void S_ResetFailedLoad(void);
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#ifdef PEXT2_VOICECHAT
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void S_Voip_Parse(void);
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#endif
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#ifdef VOICECHAT
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extern cvar_t cl_voip_showmeter;
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void S_Voip_Transmit(unsigned char clc, sizebuf_t *buf);
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void S_Voip_MapChange(void);
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int S_Voip_Loudness(qboolean ignorevad); //-1 for not capturing, otherwise between 0 and 100
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qboolean S_Voip_Speaking(unsigned int plno);
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void S_Voip_Ignore(unsigned int plno, qboolean ignore);
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#else
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#define S_Voip_Loudness() -1
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#define S_Voip_Speaking(p) false
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#define S_Voip_Ignore(p,s)
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#endif
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qboolean S_IsPlayingSomewhere(sfx_t *s);
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qboolean ResampleSfx (sfx_t *sfx, int inrate, int inchannels, int inwidth, int insamps, int inloopstart, qbyte *data);
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// picks a channel based on priorities, empty slots, number of channels
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channel_t *SND_PickChannel(soundcardinfo_t *sc, int entnum, int entchannel);
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// spatializes a channel
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void SND_Spatialize(soundcardinfo_t *sc, channel_t *ch);
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void SND_ResampleStream (void *in, int inrate, int inwidth, int inchannels, int insamps, void *out, int outrate, int outwidth, int outchannels, int resampstyle);
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// restart entire sound subsystem (doesn't flush old sounds, so make sure that happens)
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void S_DoRestart (void);
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void S_SetUnderWater(qboolean underwater);
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void S_Restart_f (void);
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//plays streaming audio
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void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels, int width, float volume);
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void CLVC_Poll (void);
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void SNDVC_MicInput(qbyte *buffer, int samples, int freq, int width);
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#ifdef AVAIL_OPENAL
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void OpenAL_CvarInit(void);
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#endif
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// ====================================================================
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// User-setable variables
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// ====================================================================
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#define MAX_CHANNELS 1024/*tracked sounds (including statics)*/
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#define MAX_DYNAMIC_CHANNELS 64 /*playing sounds (identical ones merge)*/
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#define NUM_MUSICS 1
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#define AMBIENT_FIRST 0
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#define AMBIENT_STOP NUM_AMBIENTS
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#define MUSIC_FIRST AMBIENT_STOP
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#define MUSIC_STOP (MUSIC_FIRST + NUM_MUSICS)
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#define DYNAMIC_FIRST MUSIC_STOP
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#define DYNAMIC_STOP (DYNAMIC_FIRST + MAX_DYNAMIC_CHANNELS)
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//
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// Fake dma is a synchronous faking of the DMA progress used for
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// isolating performance in the renderer. The fakedma_updates is
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// number of times S_Update() is called per second.
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//
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extern int snd_speed;
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extern vec3_t listener_origin;
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extern vec3_t listener_forward;
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extern vec3_t listener_right;
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extern vec3_t listener_up;
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extern vec_t sound_nominal_clip_dist;
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extern cvar_t loadas8bit;
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extern cvar_t bgmvolume;
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extern cvar_t volume;
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extern cvar_t snd_capture;
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extern float voicevolumemod;
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extern qboolean snd_initialized;
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extern cvar_t snd_mixerthread;
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extern int snd_blocked;
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void S_LocalSound (char *s);
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qboolean S_LoadSound (sfx_t *s);
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typedef qboolean (*S_LoadSound_t) (sfx_t *s, qbyte *data, int datalen, int sndspeed);
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qboolean S_RegisterSoundInputPlugin(S_LoadSound_t loadfnc); //called to register additional sound input plugins
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wavinfo_t GetWavinfo (char *name, qbyte *wav, int wavlength);
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void S_AmbientOff (void);
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void S_AmbientOn (void);
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//inititalisation functions.
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typedef struct
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{
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const char *name; //must be a single token, with no :
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qboolean (QDECL *InitCard) (soundcardinfo_t *sc, const char *cardname); //NULL for default device.
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qboolean (QDECL *Enumerate) (void (QDECL *callback) (const char *drivername, const char *devicecode, const char *readablename));
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} sounddriver_t;
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typedef int (*sounddriver) (soundcardinfo_t *sc, int cardnum);
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extern sounddriver pOPENAL_InitCard;
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extern sounddriver pDSOUND_InitCard;
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extern sounddriver pALSA_InitCard;
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extern sounddriver pSNDIO_InitCard;
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extern sounddriver pOSS_InitCard;
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extern sounddriver pSDL_InitCard;
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extern sounddriver pWAV_InitCard;
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extern sounddriver pAHI_InitCard;
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struct soundcardinfo_s { //windows has one defined AFTER directsound
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char name[256]; //a description of the card.
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struct soundcardinfo_s *next;
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//speaker orientations for spacialisation.
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float dist[MAXSOUNDCHANNELS];
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vec3_t speakerdir[MAXSOUNDCHANNELS];
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//info on which sound effects are playing
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channel_t channel[MAX_CHANNELS];
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int total_chans;
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//mixer
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volatile dma_t sn; //why is this volatile?
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qboolean inactive_sound; //continue mixing for this card even when the window isn't active.
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qboolean selfpainting; //allow the sound code to call the right functions when it feels the need (not properly supported).
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int paintedtime; //used in the mixer as last-written pos (in frames)
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int oldsamplepos; //this is used to track buffer wraps
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int buffers; //used to keep track of how many buffer wraps for consistant sound
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int samplequeue; //this is the number of samples the device can enqueue. if set, DMAPos returns the write point (rather than hardware read point) (in samplepairs).
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//callbacks
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void *(*Lock) (soundcardinfo_t *sc, unsigned int *startoffset); //grab a pointer to the hardware ringbuffer or whatever. startoffset is the starting offset. you can set it to 0 and bump the start offset if you need.
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void (*Unlock) (soundcardinfo_t *sc, void *buffer); //release the hardware ringbuffer memory
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void (*Submit) (soundcardinfo_t *sc, int start, int end); //if the ringbuffer is emulated, this is where you should push it to the device.
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void (*Shutdown) (soundcardinfo_t *sc); //kill the device
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unsigned int (*GetDMAPos) (soundcardinfo_t *sc); //get the current point that the hardware is reading from (the return value should not wrap, at least not very often)
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void (*SetWaterDistortion) (soundcardinfo_t *sc, qboolean underwater); //if you have eax enabled, change the environment. fixme. generally this is a stub. optional.
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void (*Restore) (soundcardinfo_t *sc); //called before lock/unlock/lock/unlock/submit. optional
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void (*ChannelUpdate) (soundcardinfo_t *sc, channel_t *channel, unsigned int schanged); //properties of a sound effect changed. this is to notify hardware mixers. optional.
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void (*ListenerUpdate) (soundcardinfo_t *sc, vec3_t origin, vec3_t forward, vec3_t right, vec3_t up, vec3_t velocity); //player moved or something. this is to notify hardware mixers. optional.
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//driver-specific - if you need more stuff, you should just shove it in the handle pointer
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void *thread;
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void *handle;
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int snd_sent;
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int snd_completed;
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int audio_fd;
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};
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extern soundcardinfo_t *sndcardinfo;
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typedef struct
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{
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void *(*Init) (int samplerate); /*create a new context*/
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void (*Start) (void *ctx); /*begin grabbing new data, old data is potentially flushed*/
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unsigned int (*Update) (void *ctx, unsigned char *buffer, unsigned int minbytes, unsigned int maxbytes); /*grab the data into a different buffer*/
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void (*Stop) (void *ctx); /*stop grabbing new data, old data may remain*/
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void (*Shutdown) (void *ctx); /*destroy everything*/
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} snd_capture_driver_t;
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#endif
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