mirror of
https://github.com/nzp-team/fteqw.git
synced 2024-11-21 19:41:14 +00:00
8dadfb4878
Cmake: Add FTE_WERROR option, defaults to true in debug builds and off in release builds (in case future compilers have issues).
Cmake: Pull in libXscreensaver so we don't get interrupted by screensavers when playing demos.
Make: Added `make webcl-rel` for a web build without server bloat (eg for sites focused on demo playback. Yes, this means you XantoM).
fteqcc: Include the decompiler in fteqcc (non-gui) builds ('-d' arg).
fteqcc: Decompiler can now mostly handle hexen2 mods without any unknown opcodes.
Allow ezHud and OpenSSL to be compiled as in-engine plugins, potentially for web and windows ports respectively.
Web: Fix support for ogg vorbis. Add support for voip.
Web: Added basic support for WebXR.
QTV: Don't try seeking on unseekable qtv streams. Don't spam when developer 1 is set.
QTV: add support for some eztv extensions.
MVD: added hack to use ktx's vweps in mvd where mvdsv doesn't bother to record the info.
qwfwd: hack around a hack in qwfwd, allowing it to work again.
recording: favour qwd in single player, instead of mvd.
Protocol: reduce client memory used for precache names. Bump maximum precache counts - some people are just abusive, yes you Orl.
hexen2: add enough clientside protocol compat to play the demo included with h2mp. lacks effects.
in_xflip: restored this setting.
fs_hidesyspaths: new cvar, defaults to enabled so you won't find your username or whatever turning up in screenshots or the like. change it to 0 before debuging stuff eg via 'path'.
gl_overbright_models: Added cvar to match QS.
netchan: Added MTU determination, we'll no longer fail to connect when routers stupidly drop icmp packets.
Win: try a few other versions of xinput too.
CSQC: Added a CSQC_GenerateMaterial function, to give the csqc a chance to generate custom materials.
MenuQC: Added support for the skeletal objects API.
2037 lines
69 KiB
C
2037 lines
69 KiB
C
#include "quakedef.h"
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/*
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This is based on Jogi's OpenAL support.
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Much of it is stripped, to try and get it clean/compliant.
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Emscripten/WebAudio is buggy or limited.
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This means we force distance models and use hacks to avoid bugs in browsers.
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We also have no doppler with WebAudio.
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*/
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/*Bug list:
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underwater cacaphoy
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openal bug with reverb. either disable reverb or disable openal.
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"build/openal-soft-1.19.1/Alc/filters/filter.c:25: BiquadFilter_setParams: Assertion `gain > 0.00001f' failed." + SIGABRT
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bug started with 1.19.1. Not fte's bug. either disable reverb or disable openal.
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(happens when reverb properties are changed too fast)
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AL_OUT_OF_MEMORY
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shitty openal implementation with too-low limits on number of sources.
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AL_INVALID_VALUE
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shitty (apple) openal implementation with too-low limits on number of sources.
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*/
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#ifdef AVAIL_OPENAL
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#ifdef FTE_TARGET_WEB
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#include <emscripten/emscripten.h>
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//emscripten provides an openal -> webaudio wrapper. its not the best, but does get the job done.
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#define OPENAL_STATIC //our javascript port doesn't support dynamic linking (bss+data segments get too messy).
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#define SDRVNAME "WebAudio" //IE doesn't support webaudio, resulting in noticable error messages about no openal, which is technically incorrect. So lets be clear about this.
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#define SDRVNAMEDESC "WebAudio:"
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#define QMIX_SDRVNAME "qmix" //IE doesn't support webaudio, resulting in noticable error messages about no openal, which is technically incorrect. So lets be clear about this.
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#define QMIX_SDRVNAMEDESC "QMIX:"
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#else
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#define SDRVNAME "OpenAL"
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#define SDRVNAMEDESC "OAL:"
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#define QMIX_SDRVNAME "qmix"
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#define QMIX_SDRVNAMEDESC "OALQ:"
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#define USEEFX
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#endif
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#ifndef HAVE_MIXER
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#undef SDRVNAMEDESC
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#define SDRVNAMEDESC "" //remove the prefixes in user-visible desciptions when there's (probably) no other devices anyway
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#endif
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#ifdef OPENAL_STATIC
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#include <AL/al.h> //output
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#include <AL/alc.h> //context+input
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#ifdef USEEFX
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#include <AL/efx.h>
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#endif
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#ifndef AL_API
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#define AL_API
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#endif
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#ifndef AL_APIENTRY
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#define AL_APIENTRY
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#endif
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#define palGetError alGetError
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#define palSourcef alSourcef
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#define palSourcei alSourcei
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#define palSourcePlayv alSourcePlayv
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#define palSourceStopv alSourceStopv
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#define palSourcePlay alSourcePlay
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#define palSourceStop alSourceStop
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#define palDopplerFactor alDopplerFactor
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#define palGenBuffers alGenBuffers
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#define palIsBuffer alIsBuffer
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#define palBufferData alBufferData
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#define palBufferiv alBufferiv
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#define palDeleteBuffers alDeleteBuffers
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#define palListenerfv alListenerfv
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#define palSourcefv alSourcefv
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#define palGetString alGetString
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#define palGenSources alGenSources
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#define palIsSource alIsSource
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#define palListenerf alListenerf
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#define palDeleteBuffers alDeleteBuffers
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#define palDeleteSources alDeleteSources
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#define palSpeedOfSound alSpeedOfSound
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#define palDistanceModel alDistanceModel
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#define palGetSourcei alGetSourcei
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#define palSourceQueueBuffers alSourceQueueBuffers
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#define palSourceUnqueueBuffers alSourceUnqueueBuffers
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#define palIsExtensionPresent alIsExtensionPresent
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#define palcOpenDevice alcOpenDevice
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#define palcCloseDevice alcCloseDevice
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#define palcCreateContext alcCreateContext
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#define palcDestroyContext alcDestroyContext
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#define palcMakeContextCurrent alcMakeContextCurrent
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#define palcProcessContext alcProcessContext
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#define palcGetString alcGetString
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#define palcGetIntegerv alcGetIntegerv
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#define palcIsExtensionPresent alcIsExtensionPresent
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#define palcGetProcAddress alcGetProcAddress
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#define palGetProcAddress alGetProcAddress
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//voip stuff
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#define palcCaptureOpenDevice alcCaptureOpenDevice
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#define palcCaptureCloseDevice alcCaptureCloseDevice
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#define palcCaptureStart alcCaptureStart
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#define palcCaptureStop alcCaptureStop
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#define palcCaptureSamples alcCaptureSamples
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#ifdef FTE_TARGET_WEB //emscripten sucks.
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AL_API void (AL_APIENTRY alSpeedOfSound)( ALfloat value ) {}
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#endif
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#else
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#if defined(_WIN32)
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#define AL_APIENTRY __cdecl
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#else
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#define AL_APIENTRY
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#endif
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#define AL_API
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#undef AL_ALEXT_PROTOTYPES
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typedef int ALint;
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typedef unsigned int ALuint;
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typedef float ALfloat;
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typedef int ALenum;
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typedef char ALchar;
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typedef char ALboolean;
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typedef int ALsizei;
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typedef void ALvoid;
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typedef unsigned char ALubyte;
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static dllhandle_t *openallib;
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static qboolean openallib_tried;
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static AL_API ALenum (AL_APIENTRY *palGetError)( void );
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static AL_API void (AL_APIENTRY *palSourcef)( ALuint sid, ALenum param, ALfloat value );
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static AL_API void (AL_APIENTRY *palSourcei)( ALuint sid, ALenum param, ALint value );
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static AL_API void (AL_APIENTRY *palSource3i)(ALuint source, ALenum param, ALint value1, ALint value2, ALint value3);
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static AL_API void (AL_APIENTRY *palSourcePlayv)( ALsizei ns, const ALuint *sids );
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static AL_API void (AL_APIENTRY *palSourceStopv)( ALsizei ns, const ALuint *sids );
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static AL_API void (AL_APIENTRY *palSourcePlay)( ALuint sid );
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static AL_API void (AL_APIENTRY *palSourceStop)( ALuint sid );
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static AL_API void (AL_APIENTRY *palDopplerFactor)( ALfloat value );
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static AL_API void (AL_APIENTRY *palGenBuffers)( ALsizei n, ALuint* buffers );
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static AL_API ALboolean (AL_APIENTRY *palIsBuffer)( ALuint bid );
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static AL_API void (AL_APIENTRY *palBufferData)( ALuint bid, ALenum format, const ALvoid* data, ALsizei size, ALsizei freq );
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static AL_API void (AL_APIENTRY *palBufferiv)(ALuint buffer, ALenum param, const ALint *values);
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static AL_API void (AL_APIENTRY *palDeleteBuffers)( ALsizei n, const ALuint* buffers );
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static AL_API void (AL_APIENTRY *palListenerfv)( ALenum param, const ALfloat* values );
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static AL_API void (AL_APIENTRY *palSourcefv)( ALuint sid, ALenum param, const ALfloat* values );
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static AL_API const ALchar* (AL_APIENTRY *palGetString)( ALenum param );
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static AL_API void (AL_APIENTRY *palGenSources)( ALsizei n, ALuint* sources );
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static AL_API ALboolean (AL_APIENTRY *palIsSource)( ALuint sourceName );
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static AL_API void (AL_APIENTRY *palListenerf)( ALenum param, ALfloat value );
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static AL_API void (AL_APIENTRY *palDeleteBuffers)( ALsizei n, const ALuint* buffers );
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static AL_API void (AL_APIENTRY *palDeleteSources)( ALsizei n, const ALuint* sources );
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static AL_API void (AL_APIENTRY *palSpeedOfSound)( ALfloat value );
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static AL_API void (AL_APIENTRY *palDistanceModel)( ALenum distanceModel );
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//needed for streaming
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static AL_API void (AL_APIENTRY *palGetSourcei)(ALuint source, ALenum pname, ALint *value);
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static AL_API void (AL_APIENTRY *palSourceQueueBuffers)(ALuint source, ALsizei n, ALuint* buffers);
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static AL_API void (AL_APIENTRY *palSourceUnqueueBuffers)(ALuint source, ALsizei n, ALuint* buffers);
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//for extensions like efx
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static AL_API ALboolean (AL_APIENTRY *palIsExtensionPresent)(const ALchar *fextame);
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static AL_API void*(AL_APIENTRY *palGetProcAddress)(const ALchar *fname);
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#define AL_NONE 0
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#define AL_FALSE 0
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#define AL_TRUE 1
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#define AL_SOURCE_RELATIVE 0x202
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#define AL_PITCH 0x1003
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#define AL_POSITION 0x1004
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#define AL_VELOCITY 0x1006
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#define AL_LOOPING 0x1007
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#define AL_BUFFER 0x1009
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#define AL_GAIN 0x100A
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#define AL_ORIENTATION 0x100F
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#define AL_SOURCE_STATE 0x1010
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#define AL_PLAYING 0x1012
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#define AL_BUFFERS_QUEUED 0x1015
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#define AL_BUFFERS_PROCESSED 0x1016
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#define AL_REFERENCE_DISTANCE 0x1020
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#define AL_ROLLOFF_FACTOR 0x1021
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#define AL_MAX_DISTANCE 0x1023
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#define AL_SAMPLE_OFFSET 0x1025
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#define AL_SOURCE_TYPE 0x1027
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#define AL_STREAMING 0x1029
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#define AL_FORMAT_MONO8 0x1100
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#define AL_FORMAT_MONO16 0x1101
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#define AL_FORMAT_STEREO8 0x1102
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#define AL_FORMAT_STEREO16 0x1103
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#define AL_INVALID_NAME 0xA001
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#define AL_INVALID_ENUM 0xA002
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#define AL_INVALID_VALUE 0xA003
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#define AL_INVALID_OPERATION 0xA004
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#define AL_OUT_OF_MEMORY 0xA005
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#define AL_VENDOR 0xB001
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#define AL_VERSION 0xB002
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#define AL_RENDERER 0xB003
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#define AL_EXTENSIONS 0xB004
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#define AL_DISTANCE_MODEL 0xD000
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#define AL_INVERSE_DISTANCE 0xD001
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#define AL_INVERSE_DISTANCE_CLAMPED 0xD002
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#define AL_LINEAR_DISTANCE 0xD003
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#define AL_LINEAR_DISTANCE_CLAMPED 0xD004
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#define AL_EXPONENT_DISTANCE 0xD005
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#define AL_EXPONENT_DISTANCE_CLAMPED 0xD006
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#if defined(_WIN32)
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#define ALC_APIENTRY __cdecl
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#else
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#define ALC_APIENTRY
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#endif
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#define ALC_API
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typedef char ALCboolean;
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typedef char ALCchar;
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typedef int ALCint;
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typedef unsigned int ALCuint;
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typedef int ALCenum;
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typedef size_t ALCsizei;
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typedef struct ALCdevice_struct ALCdevice;
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typedef struct ALCcontext_struct ALCcontext;
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typedef void ALCvoid;
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static ALC_API ALCdevice * (ALC_APIENTRY *palcOpenDevice)( const ALCchar *devicename );
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static ALC_API ALCboolean (ALC_APIENTRY *palcCloseDevice)( ALCdevice *device );
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static ALC_API ALCcontext * (ALC_APIENTRY *palcCreateContext)( ALCdevice *device, const ALCint* attrlist );
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static ALC_API void (ALC_APIENTRY *palcDestroyContext)( ALCcontext *context );
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static ALC_API ALCboolean (ALC_APIENTRY *palcMakeContextCurrent)( ALCcontext *context );
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static ALC_API void (ALC_APIENTRY *palcProcessContext)( ALCcontext *context );
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static ALC_API const ALCchar * (ALC_APIENTRY *palcGetString)( ALCdevice *device, ALCenum param );
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static ALC_API ALCboolean (ALC_APIENTRY *palcIsExtensionPresent)( ALCdevice *device, const ALCchar *extname );
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static ALC_API void* (ALC_APIENTRY *palcGetProcAddress)(ALCdevice *device, const ALCchar *funcname);
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#define ALC_DEFAULT_DEVICE_SPECIFIER 0x1004
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#define ALC_DEVICE_SPECIFIER 0x1005
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#define ALC_EXTENSIONS 0x1006
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#define ALC_ALL_DEVICES_SPECIFIER 0x1013 //ALC_ENUMERATE_ALL_EXT
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//#include "AL/alut.h"
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//#include "AL/al.h"
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//#include "AL/alext.h"
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#if defined(VOICECHAT)
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//capture-specific stuff
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static void (ALC_APIENTRY *palcGetIntegerv)( ALCdevice *device, ALCenum param, ALCsizei size, ALCint *data );
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static ALCdevice * (ALC_APIENTRY *palcCaptureOpenDevice)( const ALCchar *devicename, ALCuint frequency, ALCenum format, ALCsizei buffersize );
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static ALCboolean (ALC_APIENTRY *palcCaptureCloseDevice)( ALCdevice *device );
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static void (ALC_APIENTRY *palcCaptureStart)( ALCdevice *device );
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static void (ALC_APIENTRY *palcCaptureStop)( ALCdevice *device );
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static void (ALC_APIENTRY *palcCaptureSamples)( ALCdevice *device, ALCvoid *buffer, ALCsizei samples );
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#define ALC_CAPTURE_DEVICE_SPECIFIER 0x310
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#define ALC_CAPTURE_DEFAULT_DEVICE_SPECIFIER 0x311
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#define ALC_CAPTURE_SAMPLES 0x312
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#endif
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//efx
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#ifdef USEEFX
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#define AL_AUXILIARY_SEND_FILTER 0x20006
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#define AL_FILTER_NULL 0x0000
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#define AL_EFFECTSLOT_EFFECT 0x0001
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#define AL_EFFECT_TYPE 0x8001
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#define AL_EFFECT_REVERB 0x0001
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#define AL_EFFECT_EAXREVERB 0x8000
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#define AL_REVERB_DENSITY 0x0001
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#define AL_REVERB_DIFFUSION 0x0002
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#define AL_REVERB_GAIN 0x0003
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#define AL_REVERB_GAINHF 0x0004
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#define AL_REVERB_DECAY_TIME 0x0005
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#define AL_REVERB_DECAY_HFRATIO 0x0006
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#define AL_REVERB_REFLECTIONS_GAIN 0x0007
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#define AL_REVERB_REFLECTIONS_DELAY 0x0008
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#define AL_REVERB_LATE_REVERB_GAIN 0x0009
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#define AL_REVERB_LATE_REVERB_DELAY 0x000A
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#define AL_REVERB_AIR_ABSORPTION_GAINHF 0x000B
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#define AL_REVERB_ROOM_ROLLOFF_FACTOR 0x000C
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#define AL_REVERB_DECAY_HFLIMIT 0x000D
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/* EAX Reverb effect parameters */
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#define AL_EAXREVERB_DENSITY 0x0001
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#define AL_EAXREVERB_DIFFUSION 0x0002
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#define AL_EAXREVERB_GAIN 0x0003
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#define AL_EAXREVERB_GAINHF 0x0004
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#define AL_EAXREVERB_GAINLF 0x0005
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#define AL_EAXREVERB_DECAY_TIME 0x0006
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#define AL_EAXREVERB_DECAY_HFRATIO 0x0007
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#define AL_EAXREVERB_DECAY_LFRATIO 0x0008
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#define AL_EAXREVERB_REFLECTIONS_GAIN 0x0009
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#define AL_EAXREVERB_REFLECTIONS_DELAY 0x000A
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#define AL_EAXREVERB_REFLECTIONS_PAN 0x000B
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#define AL_EAXREVERB_LATE_REVERB_GAIN 0x000C
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#define AL_EAXREVERB_LATE_REVERB_DELAY 0x000D
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#define AL_EAXREVERB_LATE_REVERB_PAN 0x000E
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#define AL_EAXREVERB_ECHO_TIME 0x000F
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#define AL_EAXREVERB_ECHO_DEPTH 0x0010
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#define AL_EAXREVERB_MODULATION_TIME 0x0011
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#define AL_EAXREVERB_MODULATION_DEPTH 0x0012
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#define AL_EAXREVERB_AIR_ABSORPTION_GAINHF 0x0013
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#define AL_EAXREVERB_HFREFERENCE 0x0014
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#define AL_EAXREVERB_LFREFERENCE 0x0015
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#define AL_EAXREVERB_ROOM_ROLLOFF_FACTOR 0x0016
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#define AL_EAXREVERB_DECAY_HFLIMIT 0x0017
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#endif
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#endif
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#ifdef USEEFX
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#if defined(AL_ALEXT_PROTOTYPES) && defined(OPENAL_STATIC)
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#define palAuxiliaryEffectSloti alAuxiliaryEffectSloti
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#define palGenAuxiliaryEffectSlots alGenAuxiliaryEffectSlots
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#define palDeleteAuxiliaryEffectSlots alDeleteAuxiliaryEffectSlots
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#define palDeleteEffects alDeleteEffects
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#define palGenEffects alGenEffects
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#define palEffecti alEffecti
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// #define palEffectiv alEffectiv
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#define palEffectf alEffectf
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#define palEffectfv alEffectfv
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#else
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static void (AL_APIENTRY *palAuxiliaryEffectSloti)(ALuint effectslot, ALenum param, ALint iValue);
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static ALvoid (AL_APIENTRY *palGenAuxiliaryEffectSlots)(ALsizei n, ALuint *effectslots);
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static ALvoid (AL_APIENTRY *palDeleteAuxiliaryEffectSlots)(ALsizei n, const ALuint *effectslots);
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static ALvoid (AL_APIENTRY *palDeleteEffects)(ALsizei n, const ALuint *effects);
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static ALvoid (AL_APIENTRY *palGenEffects)(ALsizei n, ALuint *effects);
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static ALvoid (AL_APIENTRY *palEffecti)(ALuint effect, ALenum param, ALint iValue);
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// static ALvoid (AL_APIENTRY *palEffectiv)(ALuint effect, ALenum param, const ALint *piValues);
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static ALvoid (AL_APIENTRY *palEffectf)(ALuint effect, ALenum param, ALfloat flValue);
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static ALvoid (AL_APIENTRY *palEffectfv)(ALuint effect, ALenum param, const ALfloat *pflValues);
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#endif
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#endif
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//AL_EXT_float32
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#define AL_FORMAT_MONO_FLOAT32 0x10010
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#define AL_FORMAT_STEREO_FLOAT32 0x10011
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//AL_SOFT_source_spatialize
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#define AL_SOURCE_SPATIALIZE_SOFT 0x1214
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//AL_SOFT_loop_points
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#define AL_LOOP_POINTS_SOFT 0x2015
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//ALC_SOFT_HRTF
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#define ALC_HRTF_SOFT 0x1992
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#define ALC_DONT_CARE_SOFT 0x0002
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#define ALC_HRTF_STATUS_SOFT 0x1993
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#define ALC_HRTF_DISABLED_SOFT 0x0000
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#define ALC_HRTF_ENABLED_SOFT 0x0001
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#define ALC_HRTF_DENIED_SOFT 0x0002
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#define ALC_HRTF_REQUIRED_SOFT 0x0003
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#define ALC_HRTF_HEADPHONES_DETECTED_SOFT 0x0004
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#define ALC_HRTF_UNSUPPORTED_FORMAT_SOFT 0x0005
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#define ALC_NUM_HRTF_SPECIFIERS_SOFT 0x1994
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#define ALC_HRTF_SPECIFIER_SOFT 0x1995
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#define ALC_HRTF_ID_SOFT 0x1996
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static const ALCchar *(*palcGetStringiSOFT)(ALCdevice *device, ALCenum paramName, ALCsizei index);
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#define SOUNDVARS SDRVNAME" variables"
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extern sfx_t *known_sfx; //sfxindex = (sfx-known_sfx);
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|
|
|
#ifdef USEEFX
|
|
static ALuint OpenAL_LoadEffect(const struct reverbproperties_s *reverb);
|
|
#endif
|
|
static void OpenAL_Shutdown (soundcardinfo_t *sc);
|
|
static void QDECL OnChangeALSettings (cvar_t *var, char *value);
|
|
/*
|
|
static void S_Init_f(void);
|
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static void S_Info(void);
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|
|
static void S_Shutdown_f(void);
|
|
*/
|
|
#ifdef FTE_TARGET_WEB
|
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static cvar_t s_al_disable = CVARD("s_al_disable", "1", "0: OpenAL works (generally as the highest priority).\n1: OpenAL will be used only when a specific device is selected.\n2: Don't allow ANY use of OpenAl.\nWith OpenAL disabled, audio ouput will fall back to platform-specific output, avoiding miscilaneous third-party openal limitation bugs.");
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#else
|
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static cvar_t s_al_disable = CVARD("s_al_disable", "0", "0: OpenAL works (generally as the highest priority).\n1: OpenAL will be used only when a specific device is selected.\n2: Don't allow ANY use of OpenAl.\nWith OpenAL disabled, audio ouput will fall back to platform-specific output, avoiding miscilaneous third-party openal limitation bugs.");
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#endif
|
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static cvar_t s_al_debug = CVARD("s_al_debug", "0", "Enables periodic checks for OpenAL errors.");
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static cvar_t s_al_hrtf = CVARD("s_al_hrtf", "", "Enables use of HRTF, and which HRTF table to use.\nempty: auto, depending on openal config to enable it.\n\0: force off.\n1: Use the default HRTF.");
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static cvar_t s_al_use_reverb = CVARD("s_al_use_reverb", "1", "Controls whether reverb effects will be used. Set to 0 to block them. Reverb requires gamecode to configure the reverb properties, other than underwater.");
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//static cvar_t s_al_max_distance = CVARFC("s_al_max_distance", "1000",0,OnChangeALSettings);
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static cvar_t s_al_speedofsound = CVARFCD("s_al_speedofsound", "343.3",0,OnChangeALSettings, "Configures the speed of sound, in game units per second. This affects doppler.");
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static cvar_t s_al_dopplerfactor = CVARFCD("s_al_dopplerfactor", "1.0",0,OnChangeALSettings, "Multiplies the strength of doppler effects.");
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static cvar_t s_al_distancemodel = CVARFCD("s_al_distancemodel", legacyval("2","0"),0,OnChangeALSettings, "Controls how sounds fade with distance.\n0: Inverse (most realistic)\n1: Inverse Clamped\n2: Linear (Quake-like)\n3: Linear Clamped\n4: Exponential\n5: Exponential Clamped\n6: None");
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//static cvar_t s_al_rolloff_factor = CVAR("s_al_rolloff_factor", "1");
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static cvar_t s_al_reference_distance = CVARD("s_al_reference_distance", "120", "This is the distance at which the sound is audiable with standard volume in the inverse distance models. Nearer sounds will be louder than the original sample.");
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static cvar_t s_al_velocityscale = CVARD("s_al_velocityscale", "1", "Rescales velocity values, before doppler can be calculated.");
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static cvar_t s_al_static_listener = CVAR("s_al_static_listener", "0"); //cheat
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|
extern cvar_t snd_doppler;
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|
|
|
enum distancemodel_e
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{
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|
DM_INVERSE = 0,
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DM_INVERSE_CLAMPED = 1,
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DM_LINEAR = 2,
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DM_LINEAR_CLAMPED = 3,
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DM_EXPONENT = 4,
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DM_EXPONENT_CLAMPED = 5,
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DM_NONE = 6
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};
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|
|
|
typedef struct
|
|
{
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|
struct
|
|
{
|
|
ALuint handle;
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|
unsigned int queuesize;
|
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ALuint queue[64];
|
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} qmix;
|
|
struct
|
|
{
|
|
ALuint handle;
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qbyte allocated; //there is no guarenteed-unused handle (and I don't want to have to keep spamming alIsSource).
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qbyte queuesize;
|
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ALuint queue[3];
|
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} *source;
|
|
size_t max_sources;
|
|
|
|
struct
|
|
{
|
|
ALuint buffer;
|
|
qbyte allocated; //again no guarentee.
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} *sounds;
|
|
size_t max_sounds;
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|
|
|
ALCdevice *OpenAL_Device;
|
|
ALCcontext *OpenAL_Context;
|
|
qboolean can_source_spatialise;
|
|
qboolean can_looppoints;
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|
|
|
int ListenEnt; //listener's entity number, so we don't get weird sound displacements
|
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ALfloat ListenPos[3]; //their origin.
|
|
ALfloat ListenVel[3]; // Velocity of the listener.
|
|
ALfloat ListenOri[6]; // Orientation of the listener. (first 3 elements are "at", second 3 are "up")
|
|
unsigned int listenerdirty;
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|
|
|
#ifdef MIXER_F32
|
|
qboolean canfloataudio;
|
|
#endif
|
|
|
|
int cureffect;
|
|
ALuint effectslot; //the global reverb slot
|
|
size_t numeffecttypes;
|
|
struct
|
|
{
|
|
ALuint effect;
|
|
unsigned int modificationcount; //so we know if reverb state needs to get rebuilt
|
|
} *effecttype;
|
|
} oalinfo_t;
|
|
static void PrintALError(char *string)
|
|
{
|
|
ALenum err;
|
|
char *text = NULL;
|
|
if (!s_al_debug.value)
|
|
return;
|
|
err = palGetError();
|
|
switch(err)
|
|
{
|
|
case 0:
|
|
return;
|
|
case AL_INVALID_NAME:
|
|
text = "invalid name";
|
|
break;
|
|
case AL_INVALID_ENUM:
|
|
text = "invalid enum";
|
|
break;
|
|
case AL_INVALID_VALUE:
|
|
text = "invalid value";
|
|
break;
|
|
case AL_INVALID_OPERATION:
|
|
text = "invalid operation";
|
|
break;
|
|
case AL_OUT_OF_MEMORY:
|
|
text = "out of memory";
|
|
break;
|
|
default:
|
|
text = "unknown";
|
|
break;
|
|
}
|
|
Con_Printf("OpenAL - %s: %x: %s\n",string,err,text);
|
|
}
|
|
|
|
static qboolean OpenAL_LoadCache(oalinfo_t *oali, unsigned int *bufptr, sfxcache_t *sc, float volume, int loopstart)
|
|
{
|
|
unsigned int fmt;
|
|
unsigned int size;
|
|
switch(sc->format)
|
|
{
|
|
#ifdef FTE_TARGET_WEB
|
|
case QAF_BLOB:
|
|
palGenBuffers(1, bufptr);
|
|
emscriptenfte_al_loadaudiofile(*bufptr, sc->data, sc->length);
|
|
//alIsBuffer will report false until success or failure...
|
|
return true; //but we do have a 'proper' reference to the buffer.
|
|
#endif
|
|
case QAF_S8:
|
|
if (sc->numchannels == 2)
|
|
{
|
|
fmt = AL_FORMAT_STEREO8;
|
|
size = sc->length*2;
|
|
}
|
|
else
|
|
{
|
|
fmt = AL_FORMAT_MONO8;
|
|
size = sc->length*1;
|
|
}
|
|
break;
|
|
case QAF_S16:
|
|
if (sc->numchannels == 2)
|
|
{
|
|
fmt = AL_FORMAT_STEREO16;
|
|
size = sc->length*4;
|
|
}
|
|
else
|
|
{
|
|
fmt = AL_FORMAT_MONO16;
|
|
size = sc->length*2;
|
|
}
|
|
break;
|
|
#ifdef MIXER_F32
|
|
case QAF_F32:
|
|
if (!oali->canfloataudio)
|
|
return false;
|
|
if (sc->numchannels == 2)
|
|
{
|
|
fmt = AL_FORMAT_STEREO_FLOAT32;
|
|
size = sc->length*8;
|
|
}
|
|
else
|
|
{
|
|
fmt = AL_FORMAT_MONO_FLOAT32;
|
|
size = sc->length*4;
|
|
}
|
|
break;
|
|
#endif
|
|
default:
|
|
return false;
|
|
}
|
|
PrintALError("pre Buffer Data");
|
|
palGenBuffers(1, bufptr);
|
|
/*openal is inconsistant and supports only 8bit unsigned or 16bit signed*/
|
|
if (!sc->data)
|
|
{
|
|
//buffer some silence.
|
|
short *tmp = malloc(size);
|
|
memset(tmp, 0, size);
|
|
palBufferData(*bufptr, fmt, tmp, size, sc->speed);
|
|
free(tmp);
|
|
}
|
|
else if (volume != 1)
|
|
{
|
|
switch(sc->format)
|
|
{
|
|
#ifdef FTE_TARGET_WEB
|
|
case QAF_BLOB:
|
|
break; //unreachable
|
|
#endif
|
|
case QAF_S8:
|
|
{
|
|
unsigned char *tmp = malloc(size);
|
|
char *src = sc->data;
|
|
int i;
|
|
for (i = 0; i < size; i++)
|
|
tmp[i] = src[i]*volume+128; //signed->unsigned
|
|
palBufferData(*bufptr, fmt, tmp, size, sc->speed);
|
|
free(tmp);
|
|
}
|
|
break;
|
|
case QAF_S16:
|
|
{
|
|
short *tmp = malloc(size);
|
|
short *src = (short*)sc->data;
|
|
int i;
|
|
for (i = 0; i < (size>>1); i++)
|
|
tmp[i] = bound(-32767, src[i]*volume, 32767); //signed.
|
|
palBufferData(*bufptr, fmt, tmp, size, sc->speed);
|
|
free(tmp);
|
|
}
|
|
break;
|
|
#ifdef MIXER_F32
|
|
case QAF_F32:
|
|
{
|
|
float *tmp = malloc(size);
|
|
float *src = (float*)sc->data;
|
|
int i;
|
|
for (i = 0; i < (size>>2); i++)
|
|
tmp[i] = src[i]*volume; //signed. oversaturation isn't my problem
|
|
palBufferData(*bufptr, fmt, tmp, size, sc->speed);
|
|
free(tmp);
|
|
}
|
|
break;
|
|
#endif
|
|
}
|
|
}
|
|
else
|
|
{
|
|
switch(sc->format)
|
|
{
|
|
#ifdef FTE_TARGET_WEB
|
|
case QAF_BLOB:
|
|
break; //unreachable
|
|
#endif
|
|
case QAF_S8:
|
|
{
|
|
unsigned char *tmp = malloc(size);
|
|
char *src = sc->data;
|
|
int i;
|
|
for (i = 0; i < size; i++)
|
|
{
|
|
tmp[i] = src[i]+128;
|
|
}
|
|
palBufferData(*bufptr, fmt, tmp, size, sc->speed);
|
|
free(tmp);
|
|
}
|
|
break;
|
|
//case QAF_U8:
|
|
case QAF_S16:
|
|
//case QAF_S32:
|
|
#ifdef MIXER_F32
|
|
case QAF_F32:
|
|
#endif
|
|
palBufferData(*bufptr, fmt, sc->data, size, sc->speed);
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (oali->can_looppoints && loopstart>0)
|
|
{
|
|
ALint points[2] = {loopstart, sc->length};
|
|
palBufferiv(*bufptr, AL_LOOP_POINTS_SOFT, points);
|
|
}
|
|
|
|
//FIXME: we need to handle oal-oom error codes
|
|
|
|
PrintALError("Buffer Data");
|
|
return true;
|
|
}
|
|
|
|
static void QDECL OpenAL_CvarInit(void)
|
|
{
|
|
Cvar_Register(&s_al_disable, SOUNDVARS);
|
|
Cvar_Register(&s_al_debug, SOUNDVARS);
|
|
Cvar_Register(&s_al_hrtf, SOUNDVARS);
|
|
Cvar_Register(&s_al_use_reverb, SOUNDVARS);
|
|
// Cvar_Register(&s_al_max_distance, SOUNDVARS);
|
|
Cvar_Register(&s_al_dopplerfactor, SOUNDVARS);
|
|
Cvar_Register(&s_al_distancemodel, SOUNDVARS);
|
|
Cvar_Register(&s_al_reference_distance, SOUNDVARS);
|
|
// Cvar_Register(&s_al_rolloff_factor, SOUNDVARS);
|
|
Cvar_Register(&s_al_velocityscale, SOUNDVARS);
|
|
Cvar_Register(&s_al_static_listener, SOUNDVARS);
|
|
Cvar_Register(&s_al_speedofsound, SOUNDVARS);
|
|
}
|
|
|
|
static void OpenAL_ListenerUpdate(soundcardinfo_t *sc, int entnum, vec3_t origin, vec3_t forward, vec3_t right, vec3_t up, vec3_t velocity)
|
|
{
|
|
oalinfo_t *oali = sc->handle;
|
|
vec3_t vel;
|
|
|
|
if (snd_doppler.modified)
|
|
{
|
|
snd_doppler.modified = false;
|
|
OnChangeALSettings(NULL,NULL);
|
|
}
|
|
|
|
if (!VectorCompare(origin, oali->ListenPos))
|
|
{
|
|
VectorCopy(origin, oali->ListenPos);
|
|
oali->listenerdirty |= 1;
|
|
}
|
|
|
|
VectorScale(velocity, s_al_velocityscale.value/35.0, vel);
|
|
if (!VectorCompare(vel, oali->ListenVel))
|
|
{
|
|
VectorCopy(vel, oali->ListenVel);
|
|
oali->listenerdirty |= 2;
|
|
}
|
|
|
|
oali->ListenEnt = entnum;
|
|
if (!VectorCompare(forward, oali->ListenOri) || !VectorCompare(up, oali->ListenOri+3))
|
|
{
|
|
oali->ListenOri[0] = forward[0];
|
|
oali->ListenOri[1] = forward[1];
|
|
oali->ListenOri[2] = forward[2];
|
|
oali->ListenOri[3] = up[0];
|
|
oali->ListenOri[4] = up[1];
|
|
oali->ListenOri[5] = up[2];
|
|
oali->listenerdirty |= 4;
|
|
}
|
|
|
|
|
|
if (!s_al_static_listener.value)
|
|
{
|
|
//I'm using listenerdirty flags because emscripten's openal stuff seems to be wasting massive amounts of time on these.
|
|
// palListenerf(AL_GAIN, 1);
|
|
if (oali->listenerdirty & 1)
|
|
palListenerfv(AL_POSITION, oali->ListenPos);
|
|
#ifndef FTE_TARGET_WEB //webaudio sucks.
|
|
if (oali->listenerdirty & 2)
|
|
palListenerfv(AL_VELOCITY, oali->ListenVel);
|
|
#endif
|
|
if (oali->listenerdirty & 4)
|
|
palListenerfv(AL_ORIENTATION, oali->ListenOri);
|
|
|
|
oali->listenerdirty = 0;
|
|
}
|
|
}
|
|
|
|
static qboolean OpenAL_ReclaimASource(soundcardinfo_t *sc)
|
|
{
|
|
oalinfo_t *oali = sc->handle;
|
|
ALuint src;
|
|
ALuint buf;
|
|
int i;
|
|
int success = 0;
|
|
for (i = 0; i < sc->total_chans; i++)
|
|
{
|
|
// channel_t *chan = &sc->channel[i];
|
|
src = oali->source[i].handle;
|
|
if (oali->source[i].allocated)
|
|
{
|
|
palGetSourcei(src, AL_SOURCE_STATE, &buf);
|
|
if (buf != AL_PLAYING)
|
|
{
|
|
palDeleteSources(1, &src);
|
|
if (oali->source[i].queuesize)
|
|
palDeleteBuffers(oali->source[i].queuesize, oali->source[i].queue);
|
|
oali->source[i].queuesize = 0;
|
|
oali->source[i].handle = 0;
|
|
oali->source[i].allocated = false;
|
|
success++;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (!success)
|
|
{
|
|
int furthest=-1;
|
|
float dist, bdist=-1;
|
|
vec3_t d;
|
|
for (i = DYNAMIC_FIRST; i < sc->total_chans; i++)
|
|
{
|
|
if (oali->source[i].allocated)
|
|
{
|
|
VectorSubtract(sc->channel[i].origin, oali->ListenPos, d);
|
|
dist = DotProduct(d,d);
|
|
if (dist > bdist)
|
|
{
|
|
bdist = dist;
|
|
furthest = i;
|
|
}
|
|
}
|
|
}
|
|
if (furthest >= 0)
|
|
{
|
|
i = furthest;
|
|
palDeleteSources(1, &oali->source[i].handle);
|
|
if (oali->source[i].queuesize)
|
|
palDeleteBuffers(oali->source[i].queuesize, oali->source[i].queue);
|
|
oali->source[i].queuesize = 0;
|
|
oali->source[i].handle = 0;
|
|
oali->source[i].allocated = false;
|
|
success++;
|
|
}
|
|
}
|
|
|
|
return success;
|
|
}
|
|
|
|
//for querying sound offsets (for various hacks).
|
|
static ssamplepos_t OpenAL_GetChannelPos(soundcardinfo_t *sc, channel_t *chan)
|
|
{
|
|
ALint spos = 0;
|
|
oalinfo_t *oali = sc->handle;
|
|
int chnum = chan - sc->channel;
|
|
ALuint src;
|
|
src = oali->source[chnum].handle;
|
|
if (!oali->source[chnum].allocated)
|
|
return (ssamplepos_t)(~(usamplepos_t)0)>>1; //not actually playing...
|
|
|
|
//alcMakeContextCurrent
|
|
|
|
palGetSourcei(src, AL_SAMPLE_OFFSET, &spos);
|
|
return spos; //FIXME: result is probably going to be wrong when streaming
|
|
}
|
|
|
|
//schanged says the sample has changed, otherwise its merely moved around a little, maybe changed in volume, but nothing that will restart it.
|
|
static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdatereason_t schanged)
|
|
{
|
|
oalinfo_t *oali = sc->handle;
|
|
ALuint src;
|
|
sfx_t *sfx = chan->sfx;
|
|
float pitch, cvolume;
|
|
int chnum = chan - sc->channel;
|
|
ALuint buf;
|
|
qboolean stream;
|
|
qboolean srcrel;
|
|
ALuint processed;
|
|
|
|
if (chnum >= oali->max_sources)
|
|
Z_ReallocElements((void**)&oali->source, &oali->max_sources, chnum+1+64, sizeof(*oali->source));
|
|
|
|
//alcMakeContextCurrent
|
|
|
|
if (!oali->source[chnum].allocated)
|
|
{
|
|
//not currently playing. be prepared to create one
|
|
if (!sfx || chan->master_vol == 0)
|
|
return;
|
|
palGetError(); //gah this is so shite
|
|
palGenSources(1, &src);
|
|
if (palGetError() || !palIsSource(src))
|
|
{ //can't just test for invalid, and failure leaving src unchanged could refer to a different sound.
|
|
//try to kill some pther sound
|
|
if (OpenAL_ReclaimASource(sc))
|
|
{ //okay, we killed one. hopefully we can start a new one now.
|
|
palGenSources(1, &src);
|
|
if (palGetError() || !palIsSource(src))
|
|
{
|
|
PrintALError("alGenSources");
|
|
return;
|
|
}
|
|
}
|
|
else return;
|
|
}
|
|
oali->source[chnum].handle = src;
|
|
oali->source[chnum].allocated = true;
|
|
oali->source[chnum].queuesize = 0;
|
|
schanged |= CUR_EVERYTHING; //should normally be true anyway, but hey
|
|
}
|
|
else
|
|
src = oali->source[chnum].handle;
|
|
|
|
PrintALError("pre start sound");
|
|
|
|
if (schanged&CUR_SOUNDCHANGE)
|
|
{
|
|
palSourceStop(src);
|
|
palSourcei(src, AL_BUFFER, 0);
|
|
if (oali->source[chnum].queuesize)
|
|
palDeleteBuffers(oali->source[chnum].queuesize, oali->source[chnum].queue);
|
|
oali->source[chnum].queuesize = 0;
|
|
|
|
}
|
|
else if (oali->source[chnum].queuesize)
|
|
{
|
|
//reclaim any queued buffers
|
|
palGetSourcei(src, AL_BUFFERS_PROCESSED, &processed);
|
|
if (processed)
|
|
{
|
|
palSourceUnqueueBuffers(src, processed, oali->source[chnum].queue);
|
|
palDeleteBuffers(processed, oali->source[chnum].queue);
|
|
oali->source[chnum].queuesize -= processed;
|
|
memmove(oali->source[chnum].queue, oali->source[chnum].queue+processed, oali->source[chnum].queuesize*sizeof(*oali->source[chnum].queue));
|
|
}
|
|
}
|
|
|
|
if (!schanged && sfx) //if we don't figure out when they've finished, they'll not get replaced properly.
|
|
{
|
|
palGetSourcei(src, AL_SOURCE_STATE, &buf);
|
|
if (buf != AL_PLAYING)
|
|
{
|
|
schanged |= CUR_EVERYTHING;
|
|
if(sfx->loopstart != -1)
|
|
chan->pos = sfx->loopstart<<PITCHSHIFT;
|
|
else if (chan->flags & CF_FORCELOOP)
|
|
chan->pos = 0;
|
|
else
|
|
sfx = chan->sfx = NULL;
|
|
}
|
|
}
|
|
|
|
/*just wanted to stop it?*/
|
|
if (!sfx)
|
|
{
|
|
#ifdef FTE_TARGET_WEB
|
|
//emscripten's webaudio wrapper spams error messages after alDeleteSources has been called, if the context isn't also killed.
|
|
if (!schanged)
|
|
palSourceStop(src);
|
|
#else
|
|
palDeleteSources(1, &src);
|
|
if (oali->source[chnum].queuesize)
|
|
palDeleteBuffers(oali->source[chnum].queuesize, oali->source[chnum].queue);
|
|
oali->source[chnum].queuesize = 0;
|
|
oali->source[chnum].handle = 0;
|
|
oali->source[chnum].allocated = false;
|
|
#endif
|
|
return;
|
|
}
|
|
|
|
cvolume = chan->master_vol/255.0f;
|
|
if (!(chan->flags & CF_CL_ABSVOLUME))
|
|
cvolume *= volume.value*voicevolumemod;
|
|
else
|
|
cvolume *= mastervolume.value;
|
|
|
|
//openal doesn't support loopstart (entire sample loops or not at all), so if we're meant to skip the first half then we need to stream it.
|
|
//FIXME: AL_SOFT_loop_points
|
|
stream = sfx->decoder.decodedata || (sfx->loopstart > 0 && !oali->can_looppoints);
|
|
srcrel = (chan->flags & CF_NOSPACIALISE) || (chan->entnum && chan->entnum == oali->ListenEnt) || !chan->dist_mult;
|
|
|
|
if ((schanged&CUR_SOUNDCHANGE) || stream)
|
|
{
|
|
int sndnum = sfx-known_sfx;
|
|
int buf;
|
|
if (sndnum >= oali->max_sounds)
|
|
Z_ReallocElements((void**)&oali->sounds, &oali->max_sounds, sndnum+1+64, sizeof(*oali->sounds));
|
|
buf = oali->sounds[sndnum].buffer;
|
|
if (!oali->sounds[sndnum].allocated || stream)
|
|
{
|
|
if (!S_LoadSound(sfx, false))
|
|
return; //can't load it
|
|
if (sfx->loadstate != SLS_LOADED)
|
|
{
|
|
if (sfx->loadstate == SLS_LOADING)
|
|
{ //kill the source so that it gets regenerated again soonish
|
|
palDeleteSources(1, &src);
|
|
oali->source[chnum].handle = 0;
|
|
oali->source[chnum].allocated = false;
|
|
}
|
|
return; //not available yet
|
|
}
|
|
|
|
if (stream)
|
|
{
|
|
int offset;
|
|
sfxcache_t sbuf, *sc;
|
|
while (oali->source[chnum].queuesize < countof(oali->source[chnum].queue))
|
|
{ //decode periodically instead of all at the start.
|
|
int tryduration = snd_speed*0.5;
|
|
ssamplepos_t pos = chan->pos>>PITCHSHIFT;
|
|
|
|
if (sfx->decoder.decodedata)
|
|
sc = sfx->decoder.decodedata(sfx, &sbuf, pos, tryduration);
|
|
else
|
|
{
|
|
sc = sfx->decoder.buf;
|
|
if (pos >= sc->length)
|
|
sc = NULL;
|
|
}
|
|
if (sc)
|
|
{
|
|
memcpy(&sbuf, sc, sizeof(sbuf));
|
|
|
|
//hack up the sound to offset it correctly
|
|
if (pos < sbuf.soundoffset || pos > sbuf.soundoffset+sbuf.length)
|
|
sbuf.length = 0; //didn't contain the requested samples... the decoder is struggling.
|
|
else
|
|
{
|
|
offset = pos - sbuf.soundoffset;
|
|
sbuf.data += offset * QAF_BYTES(sc->format)*sc->numchannels;
|
|
sbuf.length -= offset;
|
|
}
|
|
sbuf.soundoffset = 0;
|
|
|
|
if (sbuf.length > tryduration)
|
|
sbuf.length = tryduration; //don't bother queuing more than 3*0.5 secs
|
|
|
|
if (sbuf.length)
|
|
{
|
|
//build a buffer with it and queue it up.
|
|
//buffer will be purged later on when its unqueued
|
|
if (OpenAL_LoadCache(oali, &buf, &sbuf, max(1,cvolume), 0))
|
|
{
|
|
palSourceQueueBuffers(src, 1, &buf);
|
|
oali->source[chnum].queue[oali->source[chnum].queuesize++] = buf;
|
|
}
|
|
}
|
|
else
|
|
{ //decoder isn't ready yet, but didn't signal an error/eof. queue a little silence, because that's better than constant micro stutters
|
|
sfxcache_t silence;
|
|
silence.speed = snd_speed;
|
|
silence.format = QAF_S16;
|
|
silence.numchannels = 1;
|
|
silence.data = NULL;
|
|
silence.length = 0.1 * silence.speed;
|
|
silence.soundoffset = 0;
|
|
if (OpenAL_LoadCache(oali, &buf, &silence, 1, 0))
|
|
{
|
|
palSourceQueueBuffers(src, 1, &buf);
|
|
oali->source[chnum].queue[oali->source[chnum].queuesize++] = buf;
|
|
}
|
|
}
|
|
|
|
//yay
|
|
chan->pos += sbuf.length<<PITCHSHIFT;
|
|
|
|
palGetSourcei(src, AL_SOURCE_STATE, &buf);
|
|
if (buf != AL_PLAYING)
|
|
schanged = CUR_EVERYTHING;
|
|
}
|
|
else
|
|
{
|
|
if(sfx->loopstart != -1)
|
|
chan->pos = sfx->loopstart<<PITCHSHIFT;
|
|
else if (chan->flags & CF_FORCELOOP)
|
|
chan->pos = 0;
|
|
else //we don't want to play anything more.
|
|
break;
|
|
if (!oali->source[chnum].queuesize)
|
|
{ //queue 0.1 secs if we're starting/resetting a new stream this is to try to cover up discontinuities caused by packetloss or whatever
|
|
sfxcache_t silence;
|
|
silence.speed = snd_speed;
|
|
silence.format = QAF_S16;
|
|
silence.numchannels = 1;
|
|
silence.data = NULL;
|
|
silence.length = 0.1 * silence.speed;
|
|
silence.soundoffset = 0;
|
|
if (OpenAL_LoadCache(oali, &buf, &silence, 1, 0))
|
|
{
|
|
palSourceQueueBuffers(src, 1, &buf);
|
|
oali->source[chnum].queue[oali->source[chnum].queuesize++] = buf;
|
|
if (oali->can_source_spatialise) //force spacialisation as desired, if supported (this solves browsers forcing stereo on mono files which should mean static audio is full volume...)
|
|
palSourcei(src, AL_SOURCE_SPATIALIZE_SOFT, !srcrel);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
if (!oali->source[chnum].queuesize)
|
|
{
|
|
palGetSourcei(src, AL_SOURCE_STATE, &buf);
|
|
if (buf != AL_PLAYING)
|
|
{
|
|
chan->sfx = NULL;
|
|
if (sfx->decoder.ended)
|
|
{
|
|
if (!S_IsPlayingSomewhere(sfx))
|
|
sfx->decoder.ended(sfx);
|
|
}
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
else
|
|
{ //unstreamed
|
|
if (!sfx->decoder.buf)
|
|
return;
|
|
oali->sounds[sndnum].allocated = OpenAL_LoadCache(oali, &buf, sfx->decoder.buf, 1, sfx->loopstart);
|
|
if (!oali->sounds[sndnum].allocated)
|
|
return;
|
|
oali->sounds[sndnum].buffer = buf;
|
|
}
|
|
}
|
|
if (!stream)
|
|
{
|
|
#ifdef FTE_TARGET_WEB
|
|
//loading an ogg is async, so we must wait until its valid.
|
|
//our javascript will hack the buffer so that its not valid until the browser has decoded it for us.
|
|
if (!palIsBuffer(buf))
|
|
{ //same as the SLS_LOADING case above
|
|
palDeleteSources(1, &src);
|
|
oali->source[chnum].handle = 0;
|
|
oali->source[chnum].allocated = false;
|
|
return;
|
|
}
|
|
#endif
|
|
palSourcei(src, AL_BUFFER, buf);
|
|
if (oali->can_source_spatialise) //force spacialisation as desired, if supported (this solves browsers forcing stereo on mono files which should mean static audio is full volume...)
|
|
palSourcei(src, AL_SOURCE_SPATIALIZE_SOFT, !srcrel);
|
|
}
|
|
}
|
|
palSourcef(src, AL_GAIN, min(cvolume, 1)); //openal only supports a max volume of 1. anything above is an error and will be clamped.
|
|
if (srcrel)
|
|
{
|
|
palSourcefv(src, AL_POSITION, vec3_origin);
|
|
#ifndef FTE_TARGET_WEB //webaudio sucks.
|
|
palSourcefv(src, AL_VELOCITY, vec3_origin);
|
|
#endif
|
|
}
|
|
else
|
|
{
|
|
palSourcefv(src, AL_POSITION, chan->origin);
|
|
#ifndef FTE_TARGET_WEB //webaudio sucks.
|
|
palSourcefv(src, AL_VELOCITY, chan->velocity);
|
|
#endif
|
|
}
|
|
|
|
if (schanged)
|
|
{
|
|
if (schanged & CUR_OFFSET && chan->pos)
|
|
{ //complex update, but not restart. pos contains an offset, rather than an absolute time.
|
|
palSourcei(src, AL_SAMPLE_OFFSET, (chan->pos>>PITCHSHIFT));
|
|
}
|
|
|
|
pitch = (float)chan->rate/(1<<PITCHSHIFT);
|
|
pitch = max(0.01, pitch); // OpenAL will clamp inside the implementation if need be, only min is important
|
|
palSourcef(src, AL_PITCH, pitch);
|
|
|
|
#ifdef USEEFX
|
|
if (chan->flags & CF_NOREVERB) //don't do the underwater thing on static sounds. it sounds like arse with all those sources.
|
|
palSource3i(src, AL_AUXILIARY_SEND_FILTER, 0, 0, AL_FILTER_NULL);
|
|
else
|
|
palSource3i(src, AL_AUXILIARY_SEND_FILTER, oali->effectslot, 0, AL_FILTER_NULL);
|
|
#endif
|
|
|
|
palSourcei(src, AL_LOOPING, (!stream && ((chan->flags & CF_FORCELOOP)||(sfx->loopstart>=0&&!stream)))?AL_TRUE:AL_FALSE);
|
|
if (srcrel)
|
|
{
|
|
palSourcei(src, AL_SOURCE_RELATIVE, AL_TRUE);
|
|
// palSourcef(src, AL_ROLLOFF_FACTOR, 0.0f);
|
|
}
|
|
else
|
|
{
|
|
palSourcei(src, AL_SOURCE_RELATIVE, AL_FALSE);
|
|
// palSourcef(src, AL_ROLLOFF_FACTOR, s_al_rolloff_factor.value*chan->dist_mult);
|
|
}
|
|
|
|
//this is disgustingly shit.
|
|
//logically we want to set the distance divisor to 1 and the rolloff factor to dist_mult.
|
|
//but openal clamps in an annoying order (probably to keep things signed in hardware) and webaudio refuses infinity, so we need to special case no attenuation to get around the issue
|
|
if (srcrel)
|
|
{
|
|
#if 0//def FTE_TARGET_WEB
|
|
switch(DM_INVERSE) //emscripten omits it, and this is webaudio's default too.
|
|
#else
|
|
switch((enum distancemodel_e)s_al_distancemodel.ival)
|
|
#endif
|
|
{
|
|
default:
|
|
case DM_INVERSE:
|
|
case DM_INVERSE_CLAMPED:
|
|
palSourcef(src, AL_ROLLOFF_FACTOR, 0);
|
|
palSourcef(src, AL_REFERENCE_DISTANCE, 1); //0 would be silent, or a division by 0
|
|
palSourcef(src, AL_MAX_DISTANCE, 1); //only used for clamped mode
|
|
break;
|
|
case DM_LINEAR:
|
|
case DM_LINEAR_CLAMPED:
|
|
palSourcef(src, AL_ROLLOFF_FACTOR, 0);
|
|
palSourcef(src, AL_REFERENCE_DISTANCE, 0); //doesn't matter when rolloff is 0
|
|
palSourcef(src, AL_MAX_DISTANCE, 1); //doesn't matter, so long as its not a nan
|
|
break;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
#if 0//def FTE_TARGET_WEB
|
|
switch(DM_LINEAR) //emscripten hardcodes it in a buggy kind of way.
|
|
#else
|
|
switch((enum distancemodel_e)s_al_distancemodel.ival)
|
|
#endif
|
|
{
|
|
default:
|
|
case DM_INVERSE:
|
|
case DM_INVERSE_CLAMPED:
|
|
palSourcef(src, AL_ROLLOFF_FACTOR, s_al_reference_distance.value);
|
|
palSourcef(src, AL_REFERENCE_DISTANCE, 1);
|
|
palSourcef(src, AL_MAX_DISTANCE, 1/chan->dist_mult); //clamp to the maximum distance you'd normally be allowed to hear... this is probably going to be annoying.
|
|
break;
|
|
case DM_LINEAR: //linear, mimic quake.
|
|
case DM_LINEAR_CLAMPED: //linear clamped to further than ref distance
|
|
palSourcef(src, AL_ROLLOFF_FACTOR, 1);
|
|
#if 0//def FTE_TARGET_WEB
|
|
//chrome complains about 0.
|
|
//with the expontential model, 0 results in division by zero, but we're not using that model and the maths for the linear model is fine with it.
|
|
//the web audio spec says 'The default value is 1. A RangeError exception must be thrown if this is set to a non-negative value.'
|
|
//which of course means that the PannerNode's constructor must throw an exception, which kinda prevents you ever creating one.
|
|
//it also says elsewhere 'If dref = 0, the value of the [exponential|inverse] model is taken to be 0, ...'
|
|
//which shows that the spec should read 'negative values' for rangeerrors (rather than non-positive). so chrome is being shit.
|
|
//unfortutely due to the nature of javascript and exceptions, this is fucking everything else up. thanks chrome!
|
|
palSourcef(src, AL_REFERENCE_DISTANCE, 0.01);
|
|
#else
|
|
palSourcef(src, AL_REFERENCE_DISTANCE, 0);
|
|
#endif
|
|
palSourcef(src, AL_MAX_DISTANCE, 1/chan->dist_mult);
|
|
break;
|
|
}
|
|
}
|
|
|
|
/*and start it up again*/
|
|
if (schanged != CUR_UPDATE)
|
|
palSourcePlay(src);
|
|
}
|
|
|
|
PrintALError(sfx?sfx->name:"post start sound");
|
|
}
|
|
|
|
/*
|
|
static void S_Info (void)
|
|
{
|
|
if (OpenAL_Device == NULL)
|
|
return;
|
|
|
|
Con_Printf("OpenAL Version : %s\n",palGetString(AL_VERSION));
|
|
Con_Printf("OpenAL Vendor : %s\n",palGetString(AL_VENDOR));
|
|
Con_Printf("OpenAL Renderer : %s\n",palGetString(AL_RENDERER));
|
|
if(palcIsExtensionPresent(NULL, (const ALCchar*)"ALC_ENUMERATION_EXT")==AL_TRUE)
|
|
{
|
|
Con_Printf("OpenAL Device : %s\n",palcGetString(OpenAL_Device,ALC_DEVICE_SPECIFIER));
|
|
}
|
|
Con_Printf("OpenAL Default Device : %s\n",palcGetString(OpenAL_Device,ALC_DEFAULT_DEVICE_SPECIFIER));
|
|
Con_Printf("OpenAL AL Extension : %s\n",palGetString(AL_EXTENSIONS));
|
|
Con_Printf("OpenAL ALC Extension : %s\n",palcGetString(NULL,ALC_EXTENSIONS));
|
|
}
|
|
*/
|
|
|
|
static qboolean OpenAL_InitLibrary(void)
|
|
{
|
|
#ifdef OPENAL_STATIC
|
|
if (s_al_disable.ival > 1)
|
|
return false;
|
|
return true;
|
|
#else
|
|
static dllfunction_t openalfuncs[] =
|
|
{
|
|
{(void*)&palGetError, "alGetError"},
|
|
{(void*)&palSourcef, "alSourcef"},
|
|
{(void*)&palSourcei, "alSourcei"},
|
|
{(void*)&palSource3i, "alSource3i"},
|
|
{(void*)&palSourcePlayv, "alSourcePlayv"},
|
|
{(void*)&palSourceStopv, "alSourceStopv"},
|
|
{(void*)&palSourcePlay, "alSourcePlay"},
|
|
{(void*)&palSourceStop, "alSourceStop"},
|
|
{(void*)&palDopplerFactor, "alDopplerFactor"},
|
|
{(void*)&palGenBuffers, "alGenBuffers"},
|
|
{(void*)&palIsBuffer, "alIsBuffer"},
|
|
{(void*)&palBufferData, "alBufferData"},
|
|
{(void*)&palBufferiv, "alBufferiv"},
|
|
{(void*)&palDeleteBuffers, "alDeleteBuffers"},
|
|
{(void*)&palListenerfv, "alListenerfv"},
|
|
{(void*)&palSourcefv, "alSourcefv"},
|
|
{(void*)&palGetString, "alGetString"},
|
|
{(void*)&palGenSources, "alGenSources"},
|
|
{(void*)&palIsSource, "alIsSource"},
|
|
{(void*)&palListenerf, "alListenerf"},
|
|
{(void*)&palDeleteSources, "alDeleteSources"},
|
|
{(void*)&palSpeedOfSound, "alSpeedOfSound"},
|
|
{(void*)&palDistanceModel, "alDistanceModel"},
|
|
|
|
{(void*)&palIsExtensionPresent, "alIsExtensionPresent"},
|
|
{(void*)&palGetProcAddress, "alGetProcAddress"},
|
|
{(void*)&palGetSourcei, "alGetSourcei"},
|
|
{(void*)&palSourceQueueBuffers, "alSourceQueueBuffers"},
|
|
{(void*)&palSourceUnqueueBuffers, "alSourceUnqueueBuffers"},
|
|
|
|
{(void*)&palcOpenDevice, "alcOpenDevice"},
|
|
{(void*)&palcCloseDevice, "alcCloseDevice"},
|
|
{(void*)&palcCreateContext, "alcCreateContext"},
|
|
{(void*)&palcDestroyContext, "alcDestroyContext"},
|
|
{(void*)&palcMakeContextCurrent, "alcMakeContextCurrent"},
|
|
{(void*)&palcProcessContext, "alcProcessContext"},
|
|
{(void*)&palcGetString, "alcGetString"},
|
|
{(void*)&palcGetIntegerv, "alcGetIntegerv"},
|
|
{(void*)&palcIsExtensionPresent, "alcIsExtensionPresent"},
|
|
{(void*)&palcGetProcAddress, "alcGetProcAddress"},
|
|
{NULL}
|
|
};
|
|
|
|
if (s_al_disable.ival > 1)
|
|
return false;
|
|
if (COM_CheckParm("-noopenal"))
|
|
return false;
|
|
|
|
if (!openallib_tried)
|
|
{
|
|
openallib_tried = true;
|
|
#ifdef _WIN32
|
|
openallib = Sys_LoadLibrary("OpenAL32", openalfuncs);
|
|
if (!openallib)
|
|
openallib = Sys_LoadLibrary("soft_oal", openalfuncs);
|
|
#else
|
|
openallib = Sys_LoadLibrary("libopenal.so.1", openalfuncs);
|
|
if (!openallib)
|
|
openallib = Sys_LoadLibrary("libopenal", openalfuncs);
|
|
#endif
|
|
}
|
|
return !!openallib;
|
|
#endif
|
|
}
|
|
|
|
static qboolean OpenAL_Init(soundcardinfo_t *sc, const char *devname, qboolean qmix)
|
|
{
|
|
oalinfo_t *oali;
|
|
|
|
if (!OpenAL_InitLibrary())
|
|
{
|
|
if (!s_al_disable.ival)
|
|
{
|
|
if (devname)
|
|
Con_Printf(SDRVNAME" library is not installed\n");
|
|
else
|
|
Con_DPrintf(SDRVNAME" library is not installed\n");
|
|
}
|
|
return false;
|
|
}
|
|
|
|
if (!devname || !*devname)
|
|
{
|
|
if (s_al_disable.ival && !qmix)
|
|
return false; //no default device
|
|
devname = palcGetString(NULL, ALC_DEFAULT_DEVICE_SPECIFIER);
|
|
}
|
|
Q_snprintfz(sc->name, sizeof(sc->name), "%s", devname);
|
|
if (qmix)
|
|
Con_TPrintf("Initiating "QMIX_SDRVNAME": %s.\n", devname);
|
|
else
|
|
Con_TPrintf("Initiating "SDRVNAME": %s.\n", devname);
|
|
|
|
oali = Z_Malloc(sizeof(oalinfo_t));
|
|
sc->handle = oali;
|
|
|
|
oali->OpenAL_Device = palcOpenDevice(devname);
|
|
if (oali->OpenAL_Device == NULL)
|
|
PrintALError("Could not init a sound device\n");
|
|
else
|
|
{
|
|
size_t i = 0;
|
|
ALCint attrs[5];
|
|
|
|
palcGetStringiSOFT = (!qmix&&palcIsExtensionPresent(oali->OpenAL_Device, "ALC_SOFT_HRTF"))?palcGetProcAddress(oali->OpenAL_Device, "alcGetStringiSOFT"):NULL;
|
|
if (palcGetStringiSOFT)
|
|
{
|
|
if (!*s_al_hrtf.string)
|
|
{
|
|
attrs[i++] = ALC_HRTF_SOFT;
|
|
attrs[i++] = ALC_DONT_CARE_SOFT;
|
|
}
|
|
else if (!strcmp(s_al_hrtf.string, "0") || !strcmp(s_al_hrtf.string, "1"))
|
|
{ //explicitly switch it off or on(default)
|
|
attrs[i++] = ALC_HRTF_SOFT;
|
|
attrs[i++] = !strcmp(s_al_hrtf.string, "1");
|
|
}
|
|
else
|
|
{ //we want an explicit hrtf
|
|
ALCint hrtf_count = 0;
|
|
ALCint idx;
|
|
const ALCchar *hrtfname;
|
|
attrs[i++] = ALC_HRTF_SOFT;
|
|
attrs[i++] = true;
|
|
|
|
palcGetIntegerv(oali->OpenAL_Device, ALC_NUM_HRTF_SPECIFIERS_SOFT, 1, &hrtf_count);
|
|
for (idx = 0; idx < hrtf_count; idx++)
|
|
{
|
|
hrtfname = palcGetStringiSOFT(oali->OpenAL_Device, ALC_HRTF_SPECIFIER_SOFT, idx);
|
|
if (hrtfname && !strcmp(hrtfname, s_al_hrtf.string))
|
|
break;
|
|
}
|
|
|
|
if (idx < hrtf_count)
|
|
{
|
|
attrs[i++] = ALC_HRTF_ID_SOFT;
|
|
attrs[i++] = idx;
|
|
}
|
|
else if (hrtf_count)
|
|
{
|
|
Con_Printf("HRTF \"%s\" not known, available options are:\n", s_al_hrtf.string);
|
|
for (idx = 0; idx < hrtf_count; idx++)
|
|
{
|
|
hrtfname = palcGetStringiSOFT(oali->OpenAL_Device, ALC_HRTF_SPECIFIER_SOFT, idx);
|
|
if (hrtfname)
|
|
Con_Printf("\t\"%s\"\n", hrtfname);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
attrs[i] = 0; //EOL
|
|
|
|
oali->OpenAL_Context = palcCreateContext(oali->OpenAL_Device, attrs);
|
|
if (!oali->OpenAL_Context)
|
|
PrintALError("Could not init a sound context\n");
|
|
else
|
|
{
|
|
palcMakeContextCurrent(oali->OpenAL_Context);
|
|
// palcProcessContext(oali->OpenAL_Context);
|
|
|
|
//S_Info();
|
|
|
|
//fixme...
|
|
memset(oali->source, 0, sizeof(*oali->source)*oali->max_sources);
|
|
PrintALError("alGensources for normal sources");
|
|
|
|
palListenerf(AL_GAIN, 1);
|
|
palListenerfv(AL_POSITION, oali->ListenPos);
|
|
#ifndef FTE_TARGET_WEB //webaudio sucks.
|
|
palListenerfv(AL_VELOCITY, oali->ListenVel);
|
|
#endif
|
|
palListenerfv(AL_ORIENTATION, oali->ListenOri);
|
|
|
|
oali->can_source_spatialise = palIsExtensionPresent("AL_SOFT_source_spatialize");
|
|
oali->can_looppoints = palIsExtensionPresent("AL_SOFT_loop_points");
|
|
|
|
if (palcGetStringiSOFT)
|
|
{
|
|
ALCint stat;
|
|
palcGetIntegerv(oali->OpenAL_Device, ALC_HRTF_STATUS_SOFT, 1, &stat);
|
|
safeswitch(stat)
|
|
{
|
|
case ALC_HRTF_DISABLED_SOFT: Con_Printf("AL_HRTF_STATUS: DISABLED.\n"); break;
|
|
case ALC_HRTF_ENABLED_SOFT: Con_Printf("AL_HRTF_STATUS: ENABLED.\n"); break;
|
|
case ALC_HRTF_DENIED_SOFT: Con_Printf("AL_HRTF_STATUS: DENIED.\n"); break;
|
|
case ALC_HRTF_REQUIRED_SOFT: Con_Printf("AL_HRTF_STATUS: REQUIRED.\n"); break;
|
|
case ALC_HRTF_HEADPHONES_DETECTED_SOFT: Con_Printf("AL_HRTF_STATUS: HEADPHONES_DETECTED.\n"); break;
|
|
case ALC_HRTF_UNSUPPORTED_FORMAT_SOFT: Con_Printf("AL_HRTF_STATUS: UNSUPPORTED_FORMAT.\n"); break;
|
|
safedefault: Con_Printf("AL_HRTF_STATUS: %#x.\n", stat); break;
|
|
}
|
|
}
|
|
return true;
|
|
}
|
|
palcCloseDevice(oali->OpenAL_Device);
|
|
}
|
|
Z_Free(oali);
|
|
return false;
|
|
}
|
|
|
|
//called when some al-specific cvar has changed that is linked to openal state.
|
|
static void QDECL OnChangeALSettings (cvar_t *var, char *value)
|
|
{
|
|
soundcardinfo_t *sc;
|
|
for (sc = sndcardinfo; sc; sc = sc->next)
|
|
{
|
|
//we only want openal devices.
|
|
if (sc->Shutdown != OpenAL_Shutdown)
|
|
continue;
|
|
//alcMakeContextCurrent
|
|
|
|
if (palSpeedOfSound)
|
|
palSpeedOfSound(s_al_speedofsound.value);
|
|
|
|
if (palDopplerFactor)
|
|
palDopplerFactor(s_al_dopplerfactor.value * snd_doppler.value);
|
|
|
|
if (palDistanceModel)
|
|
{
|
|
switch ((enum distancemodel_e)s_al_distancemodel.ival)
|
|
{
|
|
case DM_INVERSE:
|
|
//gain = AL_REFERENCE_DISTANCE / (AL_REFERENCE_DISTANCE + AL_ROLLOFF_FACTOR * (distance - AL_REFERENCE_DISTANCE) )
|
|
palDistanceModel(AL_INVERSE_DISTANCE);
|
|
break;
|
|
case DM_INVERSE_CLAMPED: //openal's default mode
|
|
//istance = max(distance,AL_REFERENCE_DISTANCE);
|
|
//distance = min(distance,AL_MAX_DISTANCE);
|
|
//gain = AL_REFERENCE_DISTANCE / (AL_REFERENCE_DISTANCE + AL_ROLLOFF_FACTOR * (distance - AL_REFERENCE_DISTANCE) )
|
|
palDistanceModel(AL_INVERSE_DISTANCE_CLAMPED);
|
|
break;
|
|
case DM_LINEAR: //most quake-like. linear
|
|
//distance = min(distance, AL_MAX_DISTANCE) // avoid negative gain
|
|
//gain = ( 1 - AL_ROLLOFF_FACTOR * (distance - AL_REFERENCE_DISTANCE) / (AL_MAX_DISTANCE - AL_REFERENCE_DISTANCE) )
|
|
palDistanceModel(AL_LINEAR_DISTANCE);
|
|
break;
|
|
case DM_LINEAR_CLAMPED: //linear, with near stuff clamped to further away
|
|
//distance = max(distance, AL_REFERENCE_DISTANCE)
|
|
//distance = min(distance, AL_MAX_DISTANCE)
|
|
//gain = ( 1 - AL_ROLLOFF_FACTOR * (distance - AL_REFERENCE_DISTANCE) / (AL_MAX_DISTANCE - AL_REFERENCE_DISTANCE) )
|
|
palDistanceModel(AL_LINEAR_DISTANCE_CLAMPED);
|
|
break;
|
|
case DM_EXPONENT:
|
|
//gain = (distance / AL_REFERENCE_DISTANCE) ^ (- AL_ROLLOFF_FACTOR)
|
|
palDistanceModel(AL_EXPONENT_DISTANCE);
|
|
break;
|
|
case DM_EXPONENT_CLAMPED:
|
|
//distance = max(distance, AL_REFERENCE_DISTANCE)
|
|
//distance = min(distance, AL_MAX_DISTANCE)
|
|
//gain = (distance / AL_REFERENCE_DISTANCE) ^ (- AL_ROLLOFF_FACTOR)
|
|
palDistanceModel(AL_EXPONENT_DISTANCE_CLAMPED);
|
|
break;
|
|
case DM_NONE:
|
|
//gain = 1
|
|
palDistanceModel(AL_NONE);
|
|
break;
|
|
default:
|
|
Cvar_ForceSet(&s_al_distancemodel, "2");
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/*stub should not be called*/
|
|
static void *OpenAL_LockBuffer (soundcardinfo_t *sc, unsigned int *sampidx)
|
|
{
|
|
//Con_Printf("OpenAL: LockBuffer\n");
|
|
return NULL;
|
|
}
|
|
|
|
/*stub should not be called*/
|
|
static void OpenAL_UnlockBuffer (soundcardinfo_t *sc, void *buffer)
|
|
{
|
|
//Con_Printf("OpenAL: UnlockBuffer\n");
|
|
}
|
|
|
|
/*stub should not be called*/
|
|
static void OpenAL_Submit (soundcardinfo_t *sc, int start, int end)
|
|
{
|
|
//Con_Printf("OpenAL: Submit\n");
|
|
}
|
|
|
|
/*stub should not be called*/
|
|
static unsigned int OpenAL_GetDMAPos (soundcardinfo_t *sc)
|
|
{
|
|
//Con_Printf("OpenAL: GetDMAPos\n");
|
|
return 0;
|
|
}
|
|
|
|
#ifdef USEEFX
|
|
static void OpenAL_SetReverb (soundcardinfo_t *sc, size_t reverbeffect)
|
|
{
|
|
#ifdef USEEFX
|
|
oalinfo_t *oali = sc->handle;
|
|
|
|
if (!oali->effectslot)
|
|
return;
|
|
|
|
if (reverbeffect >= numreverbproperties)
|
|
return; //err... you're doing it wrong.
|
|
|
|
//alcMakeContextCurrent
|
|
|
|
if (reverbeffect >= oali->numeffecttypes)
|
|
{
|
|
void *n = BZ_Realloc(oali->effecttype, sizeof(*oali->effecttype)*numreverbproperties);
|
|
if (!n)
|
|
return; //erk?
|
|
oali->effecttype = n;
|
|
memset(oali->effecttype + oali->numeffecttypes, 0, sizeof(*oali->effecttype)*(numreverbproperties-oali->numeffecttypes));
|
|
oali->numeffecttypes = numreverbproperties;
|
|
}
|
|
if (oali->effecttype[reverbeffect].modificationcount != reverbproperties[reverbeffect].modificationcount)
|
|
{ //the desired effect was modified
|
|
oali->cureffect = ~0;
|
|
oali->effecttype[reverbeffect].modificationcount = reverbproperties[reverbeffect].modificationcount;
|
|
|
|
palDeleteEffects(1, &oali->effecttype[reverbeffect].effect);
|
|
oali->effecttype[reverbeffect].effect = OpenAL_LoadEffect(&reverbproperties[reverbeffect].props);
|
|
}
|
|
else
|
|
{
|
|
//don't spam it
|
|
if (oali->cureffect == reverbeffect)
|
|
return;
|
|
}
|
|
oali->cureffect = reverbeffect;
|
|
PrintALError("preunderwater");
|
|
palAuxiliaryEffectSloti(oali->effectslot, AL_EFFECTSLOT_EFFECT, oali->effecttype[oali->cureffect].effect);
|
|
PrintALError("postunderwater");
|
|
//Con_Printf("OpenAL: SetUnderWater %i\n", underwater);
|
|
#endif
|
|
}
|
|
#endif
|
|
|
|
static void OpenAL_Shutdown (soundcardinfo_t *sc)
|
|
{
|
|
oalinfo_t *oali = sc->handle;
|
|
int i;
|
|
|
|
//alcMakeContextCurrent
|
|
|
|
for (i=0;i<oali->max_sources;i++)
|
|
{
|
|
if (oali->source[i].allocated)
|
|
{
|
|
palDeleteSources(1, &oali->source[i].handle);
|
|
oali->source[i].handle = 0;
|
|
oali->source[i].allocated = false;
|
|
}
|
|
}
|
|
|
|
/*make sure the buffers are cleared from the sound effects*/
|
|
for (i=0;i<oali->max_sounds;i++)
|
|
{
|
|
if (oali->sounds[i].allocated)
|
|
{
|
|
palDeleteBuffers(1,&oali->sounds[i].buffer);
|
|
oali->sounds[i].allocated = false;
|
|
}
|
|
}
|
|
|
|
#ifdef USEEFX
|
|
if (palDeleteAuxiliaryEffectSlots)
|
|
{
|
|
palDeleteAuxiliaryEffectSlots(1, &oali->effectslot);
|
|
for (i = 0; i < oali->numeffecttypes; i++)
|
|
{
|
|
if (oali->effecttype[i].effect)
|
|
palDeleteEffects(1, &oali->effecttype[i].effect);
|
|
}
|
|
}
|
|
Z_Free(oali->effecttype);
|
|
#endif
|
|
|
|
palcMakeContextCurrent(NULL);
|
|
palcDestroyContext(oali->OpenAL_Context);
|
|
palcCloseDevice(oali->OpenAL_Device);
|
|
Z_Free(oali->sounds);
|
|
Z_Free(oali->source);
|
|
Z_Free(oali);
|
|
}
|
|
|
|
#ifdef USEEFX
|
|
static ALuint OpenAL_LoadEffect(const struct reverbproperties_s *reverb)
|
|
{
|
|
ALuint effect = 0;
|
|
#ifdef AL_EFFECT_EAXREVERB
|
|
palGetError();
|
|
palGenEffects(1, &effect);
|
|
|
|
//try eax reverb for more settings
|
|
palEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_EAXREVERB);
|
|
if (!palGetError())
|
|
{
|
|
/* EAX Reverb is available. Set the EAX effect type then load the
|
|
* reverb properties. */
|
|
palEffectf(effect, AL_EAXREVERB_DENSITY, reverb->flDensity);
|
|
palEffectf(effect, AL_EAXREVERB_DIFFUSION, reverb->flDiffusion);
|
|
palEffectf(effect, AL_EAXREVERB_GAIN, reverb->flGain);
|
|
palEffectf(effect, AL_EAXREVERB_GAINHF, reverb->flGainHF);
|
|
palEffectf(effect, AL_EAXREVERB_GAINLF, reverb->flGainLF);
|
|
palEffectf(effect, AL_EAXREVERB_DECAY_TIME, reverb->flDecayTime);
|
|
palEffectf(effect, AL_EAXREVERB_DECAY_HFRATIO, reverb->flDecayHFRatio);
|
|
palEffectf(effect, AL_EAXREVERB_DECAY_LFRATIO, reverb->flDecayLFRatio);
|
|
palEffectf(effect, AL_EAXREVERB_REFLECTIONS_GAIN, reverb->flReflectionsGain);
|
|
palEffectf(effect, AL_EAXREVERB_REFLECTIONS_DELAY, reverb->flReflectionsDelay);
|
|
palEffectfv(effect, AL_EAXREVERB_REFLECTIONS_PAN, reverb->flReflectionsPan);
|
|
palEffectf(effect, AL_EAXREVERB_LATE_REVERB_GAIN, reverb->flLateReverbGain);
|
|
palEffectf(effect, AL_EAXREVERB_LATE_REVERB_DELAY, reverb->flLateReverbDelay);
|
|
palEffectfv(effect, AL_EAXREVERB_LATE_REVERB_PAN, reverb->flLateReverbPan);
|
|
palEffectf(effect, AL_EAXREVERB_ECHO_TIME, reverb->flEchoTime);
|
|
palEffectf(effect, AL_EAXREVERB_ECHO_DEPTH, reverb->flEchoDepth);
|
|
palEffectf(effect, AL_EAXREVERB_MODULATION_TIME, reverb->flModulationTime);
|
|
palEffectf(effect, AL_EAXREVERB_MODULATION_DEPTH, reverb->flModulationDepth);
|
|
palEffectf(effect, AL_EAXREVERB_AIR_ABSORPTION_GAINHF, reverb->flAirAbsorptionGainHF);
|
|
palEffectf(effect, AL_EAXREVERB_HFREFERENCE, reverb->flHFReference);
|
|
palEffectf(effect, AL_EAXREVERB_LFREFERENCE, reverb->flLFReference);
|
|
palEffectf(effect, AL_EAXREVERB_ROOM_ROLLOFF_FACTOR, reverb->flRoomRolloffFactor);
|
|
palEffecti(effect, AL_EAXREVERB_DECAY_HFLIMIT, reverb->iDecayHFLimit);
|
|
}
|
|
else
|
|
#endif
|
|
{
|
|
#ifdef AL_EFFECT_REVERB
|
|
/* No EAX Reverb. Set the standard reverb effect type then load the
|
|
* available reverb properties. */
|
|
palEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_REVERB);
|
|
|
|
palEffectf(effect, AL_REVERB_DENSITY, reverb->flDensity);
|
|
palEffectf(effect, AL_REVERB_DIFFUSION, reverb->flDiffusion);
|
|
palEffectf(effect, AL_REVERB_GAIN, reverb->flGain);
|
|
palEffectf(effect, AL_REVERB_GAINHF, reverb->flGainHF);
|
|
palEffectf(effect, AL_REVERB_DECAY_TIME, reverb->flDecayTime);
|
|
palEffectf(effect, AL_REVERB_DECAY_HFRATIO, reverb->flDecayHFRatio);
|
|
palEffectf(effect, AL_REVERB_REFLECTIONS_GAIN, reverb->flReflectionsGain);
|
|
palEffectf(effect, AL_REVERB_REFLECTIONS_DELAY, reverb->flReflectionsDelay);
|
|
palEffectf(effect, AL_REVERB_LATE_REVERB_GAIN, reverb->flLateReverbGain);
|
|
palEffectf(effect, AL_REVERB_LATE_REVERB_DELAY, reverb->flLateReverbDelay);
|
|
palEffectf(effect, AL_REVERB_AIR_ABSORPTION_GAINHF, reverb->flAirAbsorptionGainHF);
|
|
palEffectf(effect, AL_REVERB_ROOM_ROLLOFF_FACTOR, reverb->flRoomRolloffFactor);
|
|
palEffecti(effect, AL_REVERB_DECAY_HFLIMIT, reverb->iDecayHFLimit);
|
|
#endif
|
|
}
|
|
return effect;
|
|
}
|
|
#endif
|
|
|
|
#ifdef HAVE_MIXER
|
|
#define CHUNKSAMPLES 1024
|
|
static void *OAQM_LockBuffer (soundcardinfo_t *sc, unsigned int *sampidx)
|
|
{
|
|
oalinfo_t *oali = sc->handle;
|
|
if (oali->qmix.queuesize==countof(oali->qmix.queue))
|
|
return NULL; //not available yet.
|
|
return sc->sn.buffer;
|
|
}
|
|
static void OAQM_UnlockBuffer (soundcardinfo_t *sc, void *buffer)
|
|
{
|
|
}
|
|
static void OAQM_Submit (soundcardinfo_t *sc, int start, int end)
|
|
{
|
|
oalinfo_t *oali = sc->handle;
|
|
ALint buf;
|
|
int framesize = sc->sn.samplebytes*sc->sn.numchannels;
|
|
if (end==start)
|
|
return;
|
|
if (oali->qmix.queuesize == countof(oali->qmix.queue))
|
|
return;
|
|
palGenBuffers(1, &buf);
|
|
switch(sc->sn.sampleformat)
|
|
{
|
|
case QSF_F32:
|
|
palBufferData(buf, (sc->sn.numchannels>1)?AL_FORMAT_STEREO_FLOAT32:AL_FORMAT_MONO_FLOAT32, sc->sn.buffer, (end-start)*framesize, sc->sn.speed);
|
|
break;
|
|
case QSF_S16:
|
|
palBufferData(buf, (sc->sn.numchannels>1)?AL_FORMAT_STEREO16:AL_FORMAT_MONO16, sc->sn.buffer, (end-start)*framesize, sc->sn.speed);
|
|
break;
|
|
case QSF_U8:
|
|
palBufferData(buf, (sc->sn.numchannels>1)?AL_FORMAT_STEREO8:AL_FORMAT_MONO8, sc->sn.buffer, (end-start)*framesize, sc->sn.speed);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
palSourceQueueBuffers(oali->qmix.handle, 1, &buf);
|
|
oali->qmix.queue[oali->qmix.queuesize++] = buf;
|
|
sc->snd_completed += (end-start);
|
|
|
|
palGetSourcei(oali->qmix.handle, AL_SOURCE_STATE, &buf);
|
|
if (buf != AL_PLAYING)
|
|
palSourcePlay(oali->qmix.handle);
|
|
}
|
|
|
|
/*stub should not be called*/
|
|
static unsigned int OAQM_GetDMAPos (soundcardinfo_t *sc)
|
|
{
|
|
extern cvar_t _snd_mixahead;
|
|
oalinfo_t *oali = sc->handle;
|
|
ALint src = oali->qmix.handle;
|
|
ALint processed = 0;
|
|
unsigned int avail;
|
|
palGetSourcei(src, AL_BUFFERS_PROCESSED, &processed);
|
|
if (processed)
|
|
{
|
|
palSourceUnqueueBuffers(src, processed, oali->qmix.queue);
|
|
palDeleteBuffers(processed, oali->qmix.queue);
|
|
oali->qmix.queuesize -= processed;
|
|
memmove(oali->qmix.queue, oali->qmix.queue+processed, oali->qmix.queuesize*sizeof(*oali->qmix.queue));
|
|
}
|
|
|
|
avail = ((_snd_mixahead.value*sc->sn.speed)+CHUNKSAMPLES/2)/CHUNKSAMPLES; //how many buffers we want to try using.
|
|
avail = bound(2, avail, countof(oali->qmix.queue));
|
|
if (oali->qmix.queuesize > avail)
|
|
avail = 0;
|
|
else
|
|
avail = avail-oali->qmix.queuesize;
|
|
avail *= CHUNKSAMPLES;
|
|
|
|
sc->sn.samplepos = (sc->snd_completed+avail);
|
|
sc->sn.samplepos *= sc->sn.numchannels;
|
|
return sc->sn.samplepos;
|
|
}
|
|
|
|
static qboolean QDECL OpenAL_Enumerate_QMix(void (QDECL *callback)(const char *driver, const char *devicecode, const char *readabledevice))
|
|
{
|
|
const char *devnames;
|
|
if (!OpenAL_InitLibrary())
|
|
return true; //enumerate nothing if al is disabled
|
|
|
|
devnames = palcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER);
|
|
if (!devnames)
|
|
devnames = palcGetString(NULL, ALC_DEVICE_SPECIFIER);
|
|
while(*devnames)
|
|
{
|
|
callback(QMIX_SDRVNAME, devnames, va(QMIX_SDRVNAMEDESC"%s", devnames));
|
|
devnames += strlen(devnames)+1;
|
|
}
|
|
return true;
|
|
}
|
|
#endif
|
|
|
|
static qboolean QDECL OpenAL_Enumerate(void (QDECL *callback)(const char *driver, const char *devicecode, const char *readabledevice))
|
|
{
|
|
const char *devnames;
|
|
if (!OpenAL_InitLibrary())
|
|
return true; //enumerate nothing if al is disabled
|
|
|
|
devnames = palcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER);
|
|
if (!devnames)
|
|
devnames = palcGetString(NULL, ALC_DEVICE_SPECIFIER);
|
|
while(*devnames)
|
|
{
|
|
callback(SDRVNAME, devnames, va(SDRVNAMEDESC"%s", devnames));
|
|
devnames += strlen(devnames)+1;
|
|
}
|
|
return true;
|
|
}
|
|
|
|
|
|
static qboolean QDECL OpenAL_InitCard2(soundcardinfo_t *sc, const char *devname, qboolean qmix)
|
|
{
|
|
oalinfo_t *oali;
|
|
|
|
soundcardinfo_t *old;
|
|
// extern soundcardinfo_t *sndcardinfo;
|
|
|
|
//
|
|
for (old = sndcardinfo; old; old = old->next)
|
|
{
|
|
if (old->Shutdown == OpenAL_Shutdown)
|
|
{
|
|
//in theory, we could relax this by using alcMakeContextCurrent lots, but we'd also need to do something about the per-sound audio buffer handle hack
|
|
Con_Printf(CON_ERROR SDRVNAME ": only a single device may be active at once\n");
|
|
return false;
|
|
}
|
|
}
|
|
|
|
|
|
if (OpenAL_Init(sc, devname, qmix) == false)
|
|
return false;
|
|
oali = sc->handle;
|
|
|
|
Con_DPrintf( "AL_VERSION: %s\n",palGetString(AL_VERSION));
|
|
Con_DPrintf( "AL_RENDERER: %s\n",palGetString(AL_RENDERER));
|
|
Con_DPrintf( "AL_VENDOR: %s\n",palGetString(AL_VENDOR));
|
|
Con_DPrintf("AL_EXTENSIONS: %s\n",palGetString(AL_EXTENSIONS));
|
|
Con_DPrintf("ALC_EXTENSIONS: %s\n",palcGetString(oali->OpenAL_Device,ALC_EXTENSIONS));
|
|
|
|
#ifdef MIXER_F32
|
|
oali->canfloataudio = palIsExtensionPresent("AL_EXT_float32");
|
|
#endif
|
|
|
|
sc->inactive_sound = true;
|
|
sc->Shutdown = OpenAL_Shutdown;
|
|
#ifdef HAVE_MIXER
|
|
if (qmix)
|
|
{
|
|
sc->Lock = OAQM_LockBuffer;
|
|
sc->Unlock = OAQM_UnlockBuffer;
|
|
sc->GetDMAPos = OAQM_GetDMAPos;
|
|
sc->Submit = OAQM_Submit;
|
|
|
|
sc->sn.numchannels = bound(1, sc->sn.numchannels, 2);
|
|
sc->sn.samples = CHUNKSAMPLES*sc->sn.numchannels;
|
|
#ifdef MIXER_F32
|
|
if (sc->sn.samplebytes == 4 && oali->canfloataudio)
|
|
{
|
|
sc->sn.sampleformat = QSF_F32;
|
|
sc->sn.samplebytes = 4;
|
|
}
|
|
else
|
|
#endif
|
|
if (sc->sn.samplebytes > 1)
|
|
{
|
|
sc->sn.sampleformat = QSF_S16;
|
|
sc->sn.samplebytes = 2;
|
|
}
|
|
else
|
|
{
|
|
sc->sn.sampleformat = QSF_U8;
|
|
sc->sn.samplebytes = 1;
|
|
}
|
|
// sc->sn.speed = 11025;
|
|
sc->sn.buffer = malloc(sc->sn.samples * sc->sn.samplebytes);
|
|
sc->samplequeue = -1;
|
|
|
|
oali->qmix.handle = 0;
|
|
oali->qmix.queuesize = 0;
|
|
palGenSources(1, &oali->qmix.handle);
|
|
palSourcef(oali->qmix.handle, AL_GAIN, 1);
|
|
palSourcei(oali->qmix.handle, AL_SOURCE_RELATIVE, AL_TRUE);
|
|
//palSourcePlay(oali->qmix.handle);
|
|
}
|
|
else
|
|
#endif
|
|
{
|
|
#ifdef USEEFX
|
|
sc->SetEnvironmentReverb = OpenAL_SetReverb;
|
|
#endif
|
|
sc->ChannelUpdate = OpenAL_ChannelUpdate;
|
|
sc->ListenerUpdate = OpenAL_ListenerUpdate;
|
|
sc->GetChannelPos = OpenAL_GetChannelPos;
|
|
//these are stubs for our software mixer, and are not used with hardware mixing.
|
|
sc->Lock = OpenAL_LockBuffer;
|
|
sc->Unlock = OpenAL_UnlockBuffer;
|
|
sc->Submit = OpenAL_Submit;
|
|
sc->GetDMAPos = OpenAL_GetDMAPos;
|
|
|
|
sc->selfpainting = true;
|
|
sc->sn.sampleformat = QSF_EXTERNALMIXER;
|
|
|
|
OnChangeALSettings(NULL, NULL);
|
|
|
|
#ifdef USEEFX
|
|
PrintALError("preeffects");
|
|
#ifndef AL_ALEXT_PROTOTYPES
|
|
palAuxiliaryEffectSloti = palGetProcAddress("alAuxiliaryEffectSloti");
|
|
palGenAuxiliaryEffectSlots = palGetProcAddress("alGenAuxiliaryEffectSlots");
|
|
palDeleteAuxiliaryEffectSlots = palGetProcAddress("alDeleteAuxiliaryEffectSlots");
|
|
palDeleteEffects = palGetProcAddress("alDeleteEffects");
|
|
palGenEffects = palGetProcAddress("alGenEffects");
|
|
palEffecti = palGetProcAddress("alEffecti");
|
|
// palEffectiv = palGetProcAddress("alEffectiv");
|
|
palEffectf = palGetProcAddress("alEffectf");
|
|
palEffectfv = palGetProcAddress("alEffectfv");
|
|
#endif
|
|
|
|
if (palGenAuxiliaryEffectSlots && s_al_use_reverb.ival)
|
|
palGenAuxiliaryEffectSlots(1, &oali->effectslot);
|
|
|
|
oali->cureffect = ~0;
|
|
PrintALError("posteffects");
|
|
#endif
|
|
}
|
|
return true;
|
|
}
|
|
|
|
static qboolean QDECL OpenAL_InitCard(soundcardinfo_t *sc, const char *devname)
|
|
{
|
|
return OpenAL_InitCard2(sc, devname, false);
|
|
}
|
|
|
|
sounddriver_t OPENAL_Output =
|
|
{
|
|
SDRVNAME,
|
|
OpenAL_InitCard,
|
|
OpenAL_Enumerate,
|
|
OpenAL_CvarInit
|
|
};
|
|
#ifdef HAVE_MIXER
|
|
static qboolean QDECL OpenAL_InitCard_QMix(soundcardinfo_t *sc, const char *devname)
|
|
{
|
|
return OpenAL_InitCard2(sc, devname, true);
|
|
}
|
|
sounddriver_t OPENAL_Output_Lame =
|
|
{
|
|
QMIX_SDRVNAME,
|
|
OpenAL_InitCard_QMix,
|
|
OpenAL_Enumerate_QMix,
|
|
NULL
|
|
};
|
|
#endif
|
|
|
|
|
|
#if defined(VOICECHAT)
|
|
|
|
static qboolean OpenAL_InitCapture(void)
|
|
{
|
|
if (!OpenAL_InitLibrary())
|
|
return false;
|
|
|
|
//is there really much point checking for the name when the functions should exist or not regardless?
|
|
//if its not really supported, I would trust the open+enumerate operations to reliably fail. the functions are exported as actual symbols after all, not some hidden driver feature.
|
|
//it doesn't really matter if the default driver supports it, so long as one does, I guess.
|
|
//if (!palcIsExtensionPresent(NULL, "ALC_EXT_capture"))
|
|
// return false;
|
|
|
|
#ifdef OPENAL_STATIC
|
|
return true;
|
|
#else
|
|
if(!palcCaptureOpenDevice)
|
|
{
|
|
palcCaptureOpenDevice = Sys_GetAddressForName(openallib, "alcCaptureOpenDevice");
|
|
palcCaptureStart = Sys_GetAddressForName(openallib, "alcCaptureStart");
|
|
palcCaptureSamples = Sys_GetAddressForName(openallib, "alcCaptureSamples");
|
|
palcCaptureStop = Sys_GetAddressForName(openallib, "alcCaptureStop");
|
|
palcCaptureCloseDevice = Sys_GetAddressForName(openallib, "alcCaptureCloseDevice");
|
|
}
|
|
|
|
return palcGetIntegerv&&palcCaptureOpenDevice&&palcCaptureStart&&palcCaptureSamples&&palcCaptureStop&&palcCaptureCloseDevice;
|
|
#endif
|
|
}
|
|
static qboolean QDECL OPENAL_Capture_Enumerate (void (QDECL *callback) (const char *drivername, const char *devicecode, const char *readablename))
|
|
{
|
|
const char *devnames;
|
|
if (!OpenAL_InitCapture())
|
|
return true; //enumerate nothing if al is disabled
|
|
|
|
devnames = palcGetString(NULL, ALC_CAPTURE_DEVICE_SPECIFIER);
|
|
while(*devnames)
|
|
{
|
|
callback(SDRVNAME, devnames, va(SDRVNAMEDESC"%s", devnames));
|
|
devnames += strlen(devnames)+1;
|
|
}
|
|
return true;
|
|
}
|
|
//fte's capture api specifies mono 16.
|
|
static void *QDECL OPENAL_Capture_Init (int samplerate, const char *device)
|
|
{
|
|
#ifndef OPENAL_STATIC
|
|
if (!device) //no default devices please, too buggy for that.
|
|
return NULL;
|
|
#endif
|
|
|
|
if (!OpenAL_InitCapture())
|
|
return NULL; //enumerate nothing if al is disabled
|
|
|
|
if (!device || !*device)
|
|
{
|
|
#if defined(FTE_TARGET_WEB) && (__EMSCRIPTEN_major__>2 || (__EMSCRIPTEN_major__==2&&__EMSCRIPTEN_tiny__>=14))
|
|
//emscripten, and recent enough to actually work. don't check s_al_disable here as we don't have dsound/alsa fallbacks and we do want to actually use it.
|
|
//older versions of emscripten are too buggy to use.
|
|
#else
|
|
if (s_al_disable.ival)
|
|
return NULL; //no default device
|
|
#endif
|
|
device = palcGetString(NULL, ALC_CAPTURE_DEFAULT_DEVICE_SPECIFIER);
|
|
}
|
|
|
|
return palcCaptureOpenDevice(device, samplerate, AL_FORMAT_MONO16, 0.5*samplerate);
|
|
}
|
|
static void QDECL OPENAL_Capture_Start (void *ctx)
|
|
{
|
|
ALCdevice *device = ctx;
|
|
palcCaptureStart(device);
|
|
}
|
|
static unsigned int QDECL OPENAL_Capture_Update (void *ctx, unsigned char *buffer, unsigned int minbytes, unsigned int maxbytes)
|
|
{
|
|
#define samplesize sizeof(short)
|
|
ALCdevice *device = ctx;
|
|
int avail = 0;
|
|
palcGetIntegerv(device, ALC_CAPTURE_SAMPLES, sizeof(ALint), &avail);
|
|
if (avail*samplesize < minbytes)
|
|
return 0; //don't bother grabbing it if its below the threshold.
|
|
palcCaptureSamples(device, (ALCvoid *)buffer, avail);
|
|
return avail * samplesize;
|
|
}
|
|
static void QDECL OPENAL_Capture_Stop (void *ctx)
|
|
{
|
|
ALCdevice *device = ctx;
|
|
palcCaptureStop(device);
|
|
}
|
|
static void QDECL OPENAL_Capture_Shutdown (void *ctx)
|
|
{
|
|
ALCdevice *device = ctx;
|
|
palcCaptureCloseDevice(device);
|
|
}
|
|
|
|
snd_capture_driver_t OPENAL_Capture =
|
|
{
|
|
1,
|
|
SDRVNAME,
|
|
OPENAL_Capture_Enumerate,
|
|
OPENAL_Capture_Init,
|
|
OPENAL_Capture_Start,
|
|
OPENAL_Capture_Update,
|
|
OPENAL_Capture_Stop,
|
|
OPENAL_Capture_Shutdown
|
|
};
|
|
#endif
|
|
|
|
#endif
|