fteqw/engine/client/sound.h
Spoike 0c22b8c344 Some q3 tweaks.
git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@5416 fc73d0e0-1445-4013-8a0c-d673dee63da5
2019-02-24 08:32:45 +00:00

402 lines
16 KiB
C

/*
Copyright (C) 1996-1997 Id Software, Inc.
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 2
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.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
// sound.h -- client sound i/o functions
#ifndef __SOUND__
#define __SOUND__
#define MAXSOUNDCHANNELS 8 //on a per device basis
//pitch/rate changes require that we track stuff with subsample precision.
//this can result in some awkward overflows.
#define ssamplepos_t qintptr_t
#define usamplepos_t quintptr_t
#define PITCHSHIFT 6 /*max audio file length = ((1<<32)>>PITCHSHIFT)/KHZ*/
struct sfx_s;
typedef struct
{
int s[MAXSOUNDCHANNELS];
} portable_samplegroup_t;
typedef struct {
struct sfxcache_s *(QDECL *decodedata) (struct sfx_s *sfx, struct sfxcache_s *buf, ssamplepos_t start, int length); //return true when done.
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.
void (QDECL *ended) (struct sfx_s *sfx); //sound stopped playing and is now silent (allow rewinding or something).
void (QDECL *purge) (struct sfx_s *sfx); //sound is being purged from memory. destroy everything.
void *buf;
} sfxdecode_t;
enum
{
SLS_NOTLOADED, //not tried to load it
SLS_LOADING, //loading it on a worker thread.
SLS_LOADED, //currently in memory and usable.
SLS_FAILED //already tried to load it. it won't work. not found, invalid format, etc
};
typedef struct sfx_s
{
char name[MAX_OSPATH];
sfxdecode_t decoder;
int loadstate; //no more super-spammy
qboolean touched:1; //if the sound is still relevent
qboolean syspath:1; //if the sound is still relevent
int loopstart; //-1 or sample index to begin looping at once the sample ends
#ifdef AVAIL_OPENAL
unsigned int openal_buffer;
#endif
} sfx_t;
// !!! if this is changed, it much be changed in asm_i386.h too !!!
typedef struct sfxcache_s
{
usamplepos_t length; //sample count
unsigned int speed;
unsigned int width;
unsigned int numchannels;
usamplepos_t soundoffset; //byte index into the sound
qbyte *data; // variable sized
} sfxcache_t;
typedef struct
{
int numchannels; // this many samples per frame
int samples; // mono samples in buffer (individual, non grouped)
int samplepos; // in mono samples
int samplebytes; // per channel (NOT per frame)
enum
{
QSF_INVALID, //not selected yet...
QSF_EXTERNALMIXER, //this sample format is totally irrelevant as this device uses some sort of external mixer.
QSF_U8, //FIXME: more unsigned formats need changes to S_ClearBuffer
QSF_S8, //signed 8bit format is actually quite rare.
QSF_S16, //normal format
// QSF_X8_S24, //upper 8 bits unused. hopefully we don't need any packed thing
// QSF_S32, //lower 8 bits probably unused. this makes overflow detection messy.
QSF_F32, //modern mixers can use SSE/SIMD stuff, and we can skip clamping so this can be quite nippy.
} sampleformat;
int speed; // this many frames per second
unsigned char *buffer; // pointer to mixed pcm buffer (not directly used by mixer)
} dma_t;
//client and server
#define CF_SV_RELIABLE 1 // send reliably
#define CF_NET_SENTVELOCITY CF_SV_RELIABLE
#define CF_FORCELOOP 2 // forces looping. set on static sounds.
#define CF_NOSPACIALISE 4 // these sounds are played at a fixed volume in both speakers, but still gets quieter with distance.
//#define CF_PAUSED 8 // rate = 0. or something.
#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.
//#define CF_SV_RESERVED CF_CL_ABSVOLUME
#define CF_NOREVERB 32 // disables reverb on this channel, if possible.
#define CF_FOLLOW 64 // follows the owning entity (stops moving if we lose track)
//#define CF_RESERVEDN 128 // reserved for things that should be networked.
#define CF_SV_UNICAST 256 // serverside only. the sound is sent to msg_entity only.
#define CF_SV_SENDVELOCITY 512 // serverside hint that velocity is important
#define CF_CLI_AUTOSOUND 1024 // generated from q2 entities, which avoids breaking regular sounds, using it outside the sound system will probably break things.
#define CF_CLI_INACTIVE 2048 // try to play even when inactive
#ifdef Q3CLIENT
#define CF_CLI_NODUPES 4096 // block multiple identical sounds being started on the same entity within rapid succession. required by quake3.
#endif
#define CF_NETWORKED (CF_NOSPACIALISE|CF_NOREVERB|CF_FORCELOOP|CF_FOLLOW/*|CF_RESERVEDN*/)
typedef struct
{
sfx_t *sfx; // sfx number
int vol[MAXSOUNDCHANNELS]; // volume, .8 fixed point.
ssamplepos_t pos; // sample position in sfx, <0 means delay sound start (shifted up by PITCHSHIFT)
int rate; // fixed point rate scaling
int flags; // cf_ flags
int entnum; // to allow overriding a specific sound
int entchannel; // to avoid overriding a specific sound too easily
vec3_t origin; // origin of sound effect
vec3_t velocity; // velocity of sound effect
vec_t dist_mult; // distance multiplier (attenuation/clipK)
int master_vol; // 0-255 master volume
#ifdef Q3CLIENT
unsigned int starttime; // start time, to replicate q3's 50ms embargo on duped sounds.
#endif
} channel_t;
struct soundcardinfo_s;
typedef struct soundcardinfo_s soundcardinfo_t;
extern struct sndreverbproperties_s
{
int modificationcount;
struct reverbproperties_s
{ //note: this struct originally comes from openal's eaxreverb
//it is shared with gamecode
float flDensity;
float flDiffusion;
float flGain;
float flGainHF;
float flGainLF;
float flDecayTime;
float flDecayHFRatio;
float flDecayLFRatio;
float flReflectionsGain;
float flReflectionsDelay;
float flReflectionsPan[3];
float flLateReverbGain;
float flLateReverbDelay;
float flLateReverbPan[3];
float flEchoTime;
float flEchoDepth;
float flModulationTime;
float flModulationDepth;
float flAirAbsorptionGainHF;
float flHFReference;
float flLFReference;
float flRoomRolloffFactor;
int iDecayHFLimit;
} props;
} *reverbproperties;
extern size_t numreverbproperties;
//reverbproperties_s presets, from efx-presets.h
//mostly for testing
#define REVERB_PRESET_PSYCHOTIC \
{ 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 }
//default reverb 1
#define REVERB_PRESET_UNDERWATER \
{ 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 }
void S_Init (void);
void S_Startup (void);
void S_EnumerateDevices(void);
void S_Shutdown (qboolean final);
float S_GetSoundTime(int entnum, int entchannel);
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);
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);
void S_StaticSound (sfx_t *sfx, vec3_t origin, float vol, float attenuation);
void S_StopSound (int entnum, int entchannel);
void S_StopAllSounds(qboolean clear);
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);
qboolean S_UpdateReverb(size_t reverbtype, void *reverb, size_t reverbsize);
void S_GetListenerInfo(int seat, float *origin, float *forward, float *right, float *up);
void S_Update (void);
void S_ExtraUpdate (void);
void S_MixerThread(soundcardinfo_t *sc);
void S_Purge(qboolean retaintouched);
void S_LockMixer(void);
void S_UnlockMixer(void);
qboolean S_HaveOutput(void);
void S_Music_Clear(sfx_t *onlyifsample);
void S_Music_Seek(float time);
qboolean S_GetMusicInfo(int musicchannel, float *time, float *duration, char *title, size_t titlesize);
qboolean S_Music_Playing(int musicchannel);
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.
sfx_t *Media_NextTrack(int musicchanel, float *time); //queries the track we're meant to be playing now.
sfx_t *S_FindName (const char *name, qboolean create, qboolean syspath);
sfx_t *S_PrecacheSound2 (const char *sample, qboolean syspath);
#define S_PrecacheSound(s) S_PrecacheSound2(s,false)
void S_TouchSound (char *sample);
void S_UntouchAll(void);
void S_ClearPrecache (void);
void S_BeginPrecaching (void);
void S_EndPrecaching (void);
void S_PaintChannels(soundcardinfo_t *sc, int endtime);
void S_InitPaintChannels (soundcardinfo_t *sc);
soundcardinfo_t *S_SetupDeviceSeat(char *driver, char *device, int seat);
void S_ShutdownCard (soundcardinfo_t *sc);
void S_DefaultSpeakerConfiguration(soundcardinfo_t *sc);
void S_ResetFailedLoad(void);
#ifdef PEXT2_VOICECHAT
void S_Voip_Parse(void);
#endif
#ifdef VOICECHAT
extern cvar_t snd_voip_showmeter;
void S_Voip_Transmit(unsigned char clc, sizebuf_t *buf);
void S_Voip_MapChange(void);
int S_Voip_Loudness(qboolean ignorevad); //-1 for not capturing, otherwise between 0 and 100
int S_Voip_ClientLoudness(unsigned int plno);
qboolean S_Voip_Speaking(unsigned int plno);
void S_Voip_Ignore(unsigned int plno, qboolean ignore);
#else
#define S_Voip_Loudness() -1
#define S_Voip_Speaking(p) false
#define S_Voip_Ignore(p,s)
#endif
qboolean S_IsPlayingSomewhere(sfx_t *s);
//qboolean ResampleSfx (sfx_t *sfx, int inrate, int inchannels, int inwidth, int insamps, int inloopstart, qbyte *data);
// picks a channel based on priorities, empty slots, number of channels
channel_t *SND_PickChannel(soundcardinfo_t *sc, int entnum, int entchannel);
void SND_ResampleStream (void *in, int inrate, int inwidth, int inchannels, int insamps, void *out, int outrate, int outwidth, int outchannels, int resampstyle);
// restart entire sound subsystem (doesn't flush old sounds, so make sure that happens)
void S_DoRestart (qboolean onlyifneeded);
void S_Restart_f (void);
//plays streaming audio
void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels, int width, float volume);
void CLVC_Poll (void);
void SNDVC_MicInput(qbyte *buffer, int samples, int freq, int width);
// ====================================================================
// User-setable variables
// ====================================================================
#define MAX_DYNAMIC_CHANNELS 64 /*playing sounds (identical ones merge)*/
#define NUM_MUSICS 1
#define AMBIENT_FIRST 0
#define AMBIENT_STOP NUM_AMBIENTS
#define MUSIC_FIRST AMBIENT_STOP
#define MUSIC_STOP (MUSIC_FIRST + NUM_MUSICS)
#define DYNAMIC_FIRST MUSIC_STOP
#define DYNAMIC_STOP (DYNAMIC_FIRST + MAX_DYNAMIC_CHANNELS)
//
// Fake dma is a synchronous faking of the DMA progress used for
// isolating performance in the renderer. The fakedma_updates is
// number of times S_Update() is called per second.
//
extern int snd_speed;
extern vec_t sound_nominal_clip_dist;
extern cvar_t loadas8bit;
extern cvar_t bgmvolume;
extern cvar_t volume;
extern cvar_t snd_capture;
extern float voicevolumemod;
extern qboolean snd_initialized;
extern cvar_t snd_mixerthread;
extern int snd_blocked;
void S_LocalSound (const char *s);
void S_LocalSound2 (const char *sound, int channel, float volume);
qboolean S_LoadSound (sfx_t *s, qboolean forcedecode);
typedef qboolean (QDECL *S_LoadSound_t) (sfx_t *s, qbyte *data, size_t datalen, int sndspeed, qboolean forcedecode);
qboolean S_RegisterSoundInputPlugin(S_LoadSound_t loadfnc); //called to register additional sound input plugins
void S_AmbientOff (void);
void S_AmbientOn (void);
//inititalisation functions.
typedef struct
{
const char *name; //must be a single token, with no :
qboolean (QDECL *InitCard) (soundcardinfo_t *sc, const char *cardname); //NULL for default device.
qboolean (QDECL *Enumerate) (void (QDECL *callback) (const char *drivername, const char *devicecode, const char *readablename));
void (QDECL *RegisterCvars) (void);
} sounddriver_t;
/*typedef int (*sounddriver) (soundcardinfo_t *sc, int cardnum);
extern sounddriver pOPENAL_InitCard;
extern sounddriver pDSOUND_InitCard;
extern sounddriver pALSA_InitCard;
extern sounddriver pSNDIO_InitCard;
extern sounddriver pOSS_InitCard;
extern sounddriver pSDL_InitCard;
extern sounddriver pWAV_InitCard;
extern sounddriver pAHI_InitCard;
*/
struct soundcardinfo_s { //windows has one defined AFTER directsound
char name[256]; //a description of the card.
char guid[256]; //device name as detected (so input code can create sound devices without bugging out too much)
struct soundcardinfo_s *next;
int seat;
//speaker orientations for spacialisation.
float dist[MAXSOUNDCHANNELS];
vec3_t speakerdir[MAXSOUNDCHANNELS];
//info on which sound effects are playing
//FIXME: use a linked list
channel_t *channel;
size_t total_chans;
size_t max_chans;
float ambientlevels[NUM_AMBIENTS]; //we use a float instead of the channel's int volume value to avoid framerate dependancies with slow transitions.
//mixer
volatile dma_t sn; //why is this volatile?
qboolean inactive_sound; //continue mixing for this card even when the window isn't active.
qboolean selfpainting; //allow the sound code to call the right functions when it feels the need (not properly supported).
int paintedtime; //used in the mixer as last-written pos (in frames)
int oldsamplepos; //this is used to track buffer wraps
int buffers; //used to keep track of how many buffer wraps for consistant sound
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).
//callbacks
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.
void (*Unlock) (soundcardinfo_t *sc, void *buffer); //release the hardware ringbuffer memory
void (*Submit) (soundcardinfo_t *sc, int start, int end); //if the ringbuffer is emulated, this is where you should push it to the device.
void (*Shutdown) (soundcardinfo_t *sc); //kill the device
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)
void (*SetEnvironmentReverb) (soundcardinfo_t *sc, size_t reverb); //if you have eax enabled, change the environment. fixme. generally this is a stub. optional.
void (*Restore) (soundcardinfo_t *sc); //called before lock/unlock/lock/unlock/submit. optional
void (*ChannelUpdate) (soundcardinfo_t *sc, channel_t *channel, unsigned int schanged); //properties of a sound effect changed. this is to notify hardware mixers. optional.
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.
//driver-specific - if you need more stuff, you should just shove it in the handle pointer
void *thread;
void *handle;
int snd_sent;
int snd_completed;
int audio_fd;
};
extern soundcardinfo_t *sndcardinfo;
typedef struct
{
int apiver;
char *drivername;
qboolean (QDECL *Enumerate) (void (QDECL *callback) (const char *drivername, const char *devicecode, const char *readablename));
void *(QDECL *Init) (int samplerate, const char *device); /*create a new context*/
void (QDECL *Start) (void *ctx); /*begin grabbing new data, old data is potentially flushed*/
unsigned int (QDECL *Update) (void *ctx, unsigned char *buffer, unsigned int minbytes, unsigned int maxbytes); /*grab the data into a different buffer*/
void (QDECL *Stop) (void *ctx); /*stop grabbing new data, old data may remain*/
void (QDECL *Shutdown) (void *ctx); /*destroy everything*/
} snd_capture_driver_t;
#endif