mirror of
https://github.com/nzp-team/fteqw.git
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c6c3d3f4e0
Quick hack preliminary d3d11 renderer. I'm not likely to touch this again for quite some time. Sys_Error works properly on android, with an error message shown. Should be less of a stab in the dark if you get errors. Seg faults are still instantly fatal. Not much I can reliably do about those - most of the juicy ones will likely occur within the gl drivers (even if its my fault) and java will still call in to it. Stereoscopic (quad-buffered) rendering is in the gl renderer. You'll likely need a quatro to use it despite it being a gl 1.0 feature. No idea about ati. See r_stereo_method for non-quad-buffered alternatives. Tweaked networking to not overflow so much. Needs testing against other qw clients. Fixed an issue with surface numbers > 32k on limit-breaking maps. Fixed a preparse issue resulting in QW clients dying with the ne_ruins map's progs. Support for the DP-variant of BSP2. The depricated RMQ variant is still supported. QTV proxy now uses ipv6 hybrid sockets where possible. Preliminary pext support. git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@4105 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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typedef struct
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{
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sfx_t *sfx; // sfx number
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int vol[MAXSOUNDCHANNELS]; // 0-255 volume
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// int delay[MAXSOUNDCHANNELS];
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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 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 (void);
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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);
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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_LoadCache(sfx_t *s, sfxcache_t *sc);
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void OpenAL_StartSound(int entnum, int entchannel, sfx_t * sfx, vec3_t origin, float fvol, float attenuation, float pitchscale);
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void OpenAL_Update_Listener(vec3_t origin, vec3_t forward, vec3_t right, vec3_t up, vec3_t velocity);
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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_usemultipledevices;
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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 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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//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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// no clue how else to handle this yet!
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#ifdef AVAIL_OPENAL
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int openal;
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#endif
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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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