397 lines
16 KiB
C
397 lines
16 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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#define MAXSOUNDCHANNELS 8 //on a per device basis
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//pitch/rate changes require that we track stuff with subsample precision.
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//this can result in some awkward overflows.
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#define ssamplepos_t qintptr_t
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#define usamplepos_t quintptr_t
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#define PITCHSHIFT 6 /*max audio file length = ((1<<32)>>PITCHSHIFT)/KHZ*/
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struct sfx_s;
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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 *(QDECL *decodedata) (struct sfx_s *sfx, struct sfxcache_s *buf, ssamplepos_t start, int length); //return true when done.
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float (QDECL *querydata) (struct sfx_s *sfx, struct sfxcache_s *buf, char *title, size_t titlesize); //reports length + original format info without actually decoding anything.
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void (QDECL *ended) (struct sfx_s *sfx); //sound stopped playing and is now silent (allow rewinding or something).
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void (QDECL *purge) (struct sfx_s *sfx); //sound is being purged from memory. destroy everything.
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void *buf;
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} sfxdecode_t;
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enum
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{
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SLS_NOTLOADED, //not tried to load it
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SLS_LOADING, //loading it on a worker thread.
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SLS_LOADED, //currently in memory and usable.
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SLS_FAILED //already tried to load it. it won't work. not found, invalid format, etc
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};
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typedef struct sfx_s
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{
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char name[MAX_OSPATH];
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sfxdecode_t decoder;
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int loadstate; //no more super-spammy
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qboolean touched:1; //if the sound is still relevent
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qboolean syspath:1; //if the sound is still relevent
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int loopstart; //-1 or sample index to begin looping at once the sample ends
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#ifdef AVAIL_OPENAL
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unsigned int openal_buffer;
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#endif
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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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usamplepos_t length; //sample count
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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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usamplepos_t 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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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 samplepos; // in mono samples
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int samplebytes; // per channel (NOT per frame)
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enum
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{
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QSF_INVALID, //not selected yet...
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QSF_EXTERNALMIXER, //this sample format is totally irrelevant as this device uses some sort of external mixer.
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QSF_U8, //FIXME: more unsigned formats need changes to S_ClearBuffer
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QSF_S8, //signed 8bit format is actually quite rare.
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QSF_S16, //normal format
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// QSF_X8_S24, //upper 8 bits unused. hopefully we don't need any packed thing
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// QSF_S32, //lower 8 bits probably unused. this makes overflow detection messy.
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QSF_F32, //modern mixers can use SSE/SIMD stuff, and we can skip clamping so this can be quite nippy.
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} sampleformat;
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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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//client and server
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#define CF_SV_RELIABLE 1 // send reliably
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#define CF_NET_SENTVELOCITY CF_SV_RELIABLE
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#define CF_FORCELOOP 2 // forces looping. set on static sounds.
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#define CF_NOSPACIALISE 4 // these sounds are played at a fixed volume in both speakers, but still gets quieter with distance.
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//#define CF_PAUSED 8 // rate = 0. or something.
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#define CF_CL_ABSVOLUME 16 // ignores volume cvar. this is ignored if received from the server because there's no practical way for the server to respect the client's preferences.
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//#define CF_SV_RESERVED CF_CL_ABSVOLUME
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#define CF_NOREVERB 32 // disables reverb on this channel, if possible.
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#define CF_FOLLOW 64 // follows the owning entity (stops moving if we lose track)
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//#define CF_RESERVEDN 128 // reserved for things that should be networked.
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#define CF_SV_UNICAST 256 // serverside only. the sound is sent to msg_entity only.
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#define CF_SV_SENDVELOCITY 512 // serverside hint that velocity is important
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#define CF_CLI_AUTOSOUND 1024 // generated from q2 entities, which avoids breaking regular sounds, using it outside the sound system will probably break things.
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#define CF_CLI_INACTIVE 2048 // try to play even when inactive
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#define CF_NETWORKED (CF_NOSPACIALISE|CF_NOREVERB|CF_FORCELOOP|CF_FOLLOW/*|CF_RESERVEDN*/)
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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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ssamplepos_t pos; // sample position in sfx, <0 means delay sound start (shifted up by PITCHSHIFT)
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int rate; // fixed point rate scaling
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int flags; // cf_ flags
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int entnum; // to allow overriding a specific sound
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int entchannel; // to avoid overriding a specific sound too easily
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vec3_t origin; // origin of sound effect
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vec3_t velocity; // velocity 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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struct soundcardinfo_s;
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typedef struct soundcardinfo_s soundcardinfo_t;
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extern struct sndreverbproperties_s
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{
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int modificationcount;
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struct reverbproperties_s
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{ //note: this struct originally comes from openal's eaxreverb
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//it is shared with gamecode
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float flDensity;
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float flDiffusion;
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float flGain;
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float flGainHF;
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float flGainLF;
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float flDecayTime;
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float flDecayHFRatio;
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float flDecayLFRatio;
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float flReflectionsGain;
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float flReflectionsDelay;
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float flReflectionsPan[3];
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float flLateReverbGain;
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float flLateReverbDelay;
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float flLateReverbPan[3];
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float flEchoTime;
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float flEchoDepth;
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float flModulationTime;
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float flModulationDepth;
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float flAirAbsorptionGainHF;
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float flHFReference;
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float flLFReference;
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float flRoomRolloffFactor;
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int iDecayHFLimit;
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} props;
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} *reverbproperties;
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extern size_t numreverbproperties;
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//reverbproperties_s presets, from efx-presets.h
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//mostly for testing
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#define REVERB_PRESET_PSYCHOTIC \
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{ 0.0625f, 0.5000f, 0.3162f, 0.8404f, 1.0000f, 7.5600f, 0.9100f, 1.0000f, 0.4864f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 2.4378f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 4.0000f, 1.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
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//default reverb 1
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#define REVERB_PRESET_UNDERWATER \
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{ 0.3645f, 1.0000f, 0.3162f, 0.0100f, 1.0000f, 1.4900f, 0.1000f, 1.0000f, 0.5963f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 7.0795f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 1.1800f, 0.3480f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
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void S_Init (void);
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void S_Startup (void);
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void S_EnumerateDevices(void);
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void S_Shutdown (qboolean final);
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float S_GetSoundTime(int entnum, int entchannel);
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void S_StartSound (int entnum, int entchannel, sfx_t *sfx, vec3_t origin, vec3_t velocity, float fvol, float attenuation, float timeofs, float pitchadj, unsigned int flags);
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float S_UpdateSound(int entnum, int entchannel, sfx_t *sfx, vec3_t origin, vec3_t velocity, float fvol, float attenuation, float timeofs, float pitchadj, unsigned int flags);
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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(int seat, int entnum, vec3_t origin, vec3_t forward, vec3_t right, vec3_t up, size_t reverbtype, vec3_t velocity);
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qboolean S_UpdateReverb(size_t reverbtype, void *reverb, size_t reverbsize);
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void S_GetListenerInfo(int seat, 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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void S_LockMixer(void);
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void S_UnlockMixer(void);
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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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qboolean S_GetMusicInfo(int musicchannel, float *time, float *duration, char *title, size_t titlesize);
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qboolean S_Music_Playing(int musicchannel);
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float Media_CrossFade(int musicchanel, float vol, float time); //queries the volume we're meant to be playing (checks for fade out). -1 for no more, otherwise returns vol.
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sfx_t *Media_NextTrack(int musicchanel, float *time); //queries the track we're meant to be playing now.
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sfx_t *S_FindName (const char *name, qboolean create, qboolean syspath);
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sfx_t *S_PrecacheSound2 (const char *sample, qboolean syspath);
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#define S_PrecacheSound(s) S_PrecacheSound2(s,false)
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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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soundcardinfo_t *S_SetupDeviceSeat(char *driver, char *device, int seat);
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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 snd_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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int S_Voip_ClientLoudness(unsigned int plno);
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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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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 (qboolean onlyifneeded);
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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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// ====================================================================
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// User-setable variables
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// ====================================================================
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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 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 (const char *s);
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void S_LocalSound2 (const char *sound, int channel, float volume);
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qboolean S_LoadSound (sfx_t *s, qboolean forcedecode);
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typedef qboolean (QDECL *S_LoadSound_t) (sfx_t *s, qbyte *data, size_t datalen, int sndspeed, qboolean forcedecode);
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qboolean S_RegisterSoundInputPlugin(S_LoadSound_t loadfnc); //called to register additional sound input plugins
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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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void (QDECL *RegisterCvars) (void);
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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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*/
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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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char guid[256]; //device name as detected (so input code can create sound devices without bugging out too much)
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struct soundcardinfo_s *next;
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int seat;
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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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//FIXME: use a linked list
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channel_t *channel;
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size_t total_chans;
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size_t max_chans;
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float ambientlevels[NUM_AMBIENTS]; //we use a float instead of the channel's int volume value to avoid framerate dependancies with slow transitions.
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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 (*SetEnvironmentReverb) (soundcardinfo_t *sc, size_t reverb); //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, int entnum, 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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int apiver;
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char *drivername;
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qboolean (QDECL *Enumerate) (void (QDECL *callback) (const char *drivername, const char *devicecode, const char *readablename));
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void *(QDECL *Init) (int samplerate, const char *device); /*create a new context*/
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void (QDECL *Start) (void *ctx); /*begin grabbing new data, old data is potentially flushed*/
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unsigned int (QDECL *Update) (void *ctx, unsigned char *buffer, unsigned int minbytes, unsigned int maxbytes); /*grab the data into a different buffer*/
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void (QDECL *Stop) (void *ctx); /*stop grabbing new data, old data may remain*/
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void (QDECL *Shutdown) (void *ctx); /*destroy everything*/
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} snd_capture_driver_t;
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#endif
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