Add support for AL_SOFT_loop_points. Implement quake-mixer-over-openal as a workaround for weird performance issues in Chromium.
git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@6086 fc73d0e0-1445-4013-8a0c-d673dee63da5
This commit is contained in:
Spoike
parent
5018d3e650
commit
15bd67c8f0
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@ -35,9 +35,14 @@ We also have no doppler with WebAudio.
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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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@ -67,6 +72,7 @@ We also have no doppler with WebAudio.
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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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@ -136,6 +142,7 @@ 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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@ -326,6 +333,9 @@ static AL_API ALvoid (AL_APIENTRY *palEffectfv)(ALuint effect, ALenum param, con
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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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@ -357,7 +367,11 @@ static void S_Info(void);
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static void S_Shutdown_f(void);
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*/
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#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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@ -384,6 +398,12 @@ enum distancemodel_e
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typedef struct
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{
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struct
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{
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ALuint handle;
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unsigned int queuesize;
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ALuint queue[64];
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} qmix;
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struct
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{
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ALuint handle;
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@ -403,11 +423,13 @@ typedef struct
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ALCdevice *OpenAL_Device;
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ALCcontext *OpenAL_Context;
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qboolean can_source_spatialise;
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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.
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ALfloat ListenVel[3]; // Velocity of the listener.
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ALfloat ListenOri[6]; // Orientation of the listener. (first 3 elements are "at", second 3 are "up")
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unsigned int listenerdirty;
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#ifdef MIXER_F32
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qboolean canfloataudio;
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@ -455,7 +477,7 @@ static void PrintALError(char *string)
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Con_Printf("OpenAL - %s: %x: %s\n",string,err,text);
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}
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static qboolean OpenAL_LoadCache(oalinfo_t *oali, unsigned int *bufptr, sfxcache_t *sc, float volume)
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static qboolean OpenAL_LoadCache(oalinfo_t *oali, unsigned int *bufptr, sfxcache_t *sc, float volume, int loopstart)
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{
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unsigned int fmt;
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unsigned int size;
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@ -599,6 +621,12 @@ static qboolean OpenAL_LoadCache(oalinfo_t *oali, unsigned int *bufptr, sfxcache
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}
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}
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if (oali->can_looppoints && loopstart>0)
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{
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ALint points[2] = {loopstart, sc->length};
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palBufferiv(*bufptr, AL_LOOP_POINTS_SOFT, points);
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}
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//FIXME: we need to handle oal-oom error codes
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PrintALError("Buffer Data");
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@ -624,6 +652,7 @@ static void QDECL OpenAL_CvarInit(void)
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static void OpenAL_ListenerUpdate(soundcardinfo_t *sc, int entnum, vec3_t origin, vec3_t forward, vec3_t right, vec3_t up, vec3_t velocity)
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{
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oalinfo_t *oali = sc->handle;
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vec3_t vel;
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if (snd_doppler.modified)
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{
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@ -631,26 +660,46 @@ static void OpenAL_ListenerUpdate(soundcardinfo_t *sc, int entnum, vec3_t origin
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OnChangeALSettings(NULL,NULL);
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}
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VectorScale(velocity, s_al_velocityscale.value/35.0, oali->ListenVel);
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VectorCopy(origin, oali->ListenPos);
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if (!VectorCompare(origin, oali->ListenPos))
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{
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VectorCopy(origin, oali->ListenPos);
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oali->listenerdirty |= 1;
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}
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VectorScale(velocity, s_al_velocityscale.value/35.0, vel);
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if (!VectorCompare(vel, oali->ListenVel))
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{
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VectorCopy(vel, oali->ListenVel);
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oali->listenerdirty |= 2;
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}
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oali->ListenEnt = entnum;
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oali->ListenOri[0] = forward[0];
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oali->ListenOri[1] = forward[1];
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oali->ListenOri[2] = forward[2];
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oali->ListenOri[3] = up[0];
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oali->ListenOri[4] = up[1];
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oali->ListenOri[5] = up[2];
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if (!VectorCompare(forward, oali->ListenOri) || !VectorCompare(up, oali->ListenOri+3))
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{
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oali->ListenOri[0] = forward[0];
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oali->ListenOri[1] = forward[1];
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oali->ListenOri[2] = forward[2];
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oali->ListenOri[3] = up[0];
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oali->ListenOri[4] = up[1];
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oali->ListenOri[5] = up[2];
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oali->listenerdirty |= 4;
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}
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if (!s_al_static_listener.value)
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{
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palListenerf(AL_GAIN, 1);
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palListenerfv(AL_POSITION, oali->ListenPos);
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//I'm using listenerdirty flags because emscripten's openal stuff seems to be wasting massive amounts of time on these.
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// palListenerf(AL_GAIN, 1);
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if (oali->listenerdirty & 1)
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palListenerfv(AL_POSITION, oali->ListenPos);
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#ifndef FTE_TARGET_WEB //webaudio sucks.
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palListenerfv(AL_VELOCITY, oali->ListenVel);
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if (oali->listenerdirty & 2)
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palListenerfv(AL_VELOCITY, oali->ListenVel);
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#endif
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palListenerfv(AL_ORIENTATION, oali->ListenOri);
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if (oali->listenerdirty & 4)
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palListenerfv(AL_ORIENTATION, oali->ListenOri);
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oali->listenerdirty = 0;
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}
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}
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@ -760,6 +809,7 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
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}
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oali->source[chnum].handle = src;
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oali->source[chnum].allocated = true;
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oali->source[chnum].queuesize = 0;
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schanged |= CUR_EVERYTHING; //should normally be true anyway, but hey
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}
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else
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@ -829,7 +879,9 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
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cvolume *= mastervolume.value;
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//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.
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stream = sfx->decoder.decodedata || sfx->loopstart > 0;
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//FIXME: AL_SOFT_loop_points
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stream = sfx->decoder.decodedata || (sfx->loopstart > 0 && !oali->can_looppoints);
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srcrel = (chan->flags & CF_NOSPACIALISE) || (chan->entnum && chan->entnum == oali->ListenEnt) || !chan->dist_mult;
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if ((schanged&CUR_SOUNDCHANGE) || stream)
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{
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@ -892,7 +944,7 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
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{
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//build a buffer with it and queue it up.
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//buffer will be purged later on when its unqueued
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if (OpenAL_LoadCache(oali, &buf, &sbuf, max(1,cvolume)))
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if (OpenAL_LoadCache(oali, &buf, &sbuf, max(1,cvolume), 0))
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{
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palSourceQueueBuffers(src, 1, &buf);
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oali->source[chnum].queue[oali->source[chnum].queuesize++] = buf;
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@ -907,7 +959,7 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
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silence.data = NULL;
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silence.length = 0.1 * silence.speed;
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silence.soundoffset = 0;
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if (OpenAL_LoadCache(oali, &buf, &silence, 1))
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if (OpenAL_LoadCache(oali, &buf, &silence, 1, 0))
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{
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palSourceQueueBuffers(src, 1, &buf);
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oali->source[chnum].queue[oali->source[chnum].queuesize++] = buf;
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@ -938,10 +990,12 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
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silence.data = NULL;
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silence.length = 0.1 * silence.speed;
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silence.soundoffset = 0;
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if (OpenAL_LoadCache(oali, &buf, &silence, 1))
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if (OpenAL_LoadCache(oali, &buf, &silence, 1, 0))
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{
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palSourceQueueBuffers(src, 1, &buf);
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oali->source[chnum].queue[oali->source[chnum].queuesize++] = buf;
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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...)
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palSourcei(src, AL_SOURCE_SPATIALIZE_SOFT, !srcrel);
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}
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}
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}
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@ -965,7 +1019,7 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
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{ //unstreamed
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if (!sfx->decoder.buf)
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return;
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oali->sounds[sndnum].allocated = OpenAL_LoadCache(oali, &buf, sfx->decoder.buf, 1);
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oali->sounds[sndnum].allocated = OpenAL_LoadCache(oali, &buf, sfx->decoder.buf, 1, sfx->loopstart);
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if (!oali->sounds[sndnum].allocated)
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return;
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oali->sounds[sndnum].buffer = buf;
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@ -985,10 +1039,11 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
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}
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#endif
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palSourcei(src, AL_BUFFER, buf);
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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...)
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palSourcei(src, AL_SOURCE_SPATIALIZE_SOFT, !srcrel);
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}
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}
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palSourcef(src, AL_GAIN, min(cvolume, 1)); //openal only supports a max volume of 1. anything above is an error and will be clamped.
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srcrel = (chan->flags & CF_NOSPACIALISE) || (chan->entnum && chan->entnum == oali->ListenEnt) || !chan->dist_mult;
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if (srcrel)
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{
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palSourcefv(src, AL_POSITION, vec3_origin);
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@ -1004,9 +1059,6 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
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#endif
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}
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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...)
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palSourcei(src, AL_SOURCE_SPATIALIZE_SOFT, !srcrel);
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if (schanged)
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{
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if (schanged == CUR_UPDATE && chan->pos)
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@ -1028,7 +1080,7 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
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}
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#endif
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palSourcei(src, AL_LOOPING, (!stream && ((chan->flags & CF_FORCELOOP)||sfx->loopstart==0))?AL_TRUE:AL_FALSE);
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palSourcei(src, AL_LOOPING, (!stream && ((chan->flags & CF_FORCELOOP)||(sfx->loopstart>=0&&!stream)))?AL_TRUE:AL_FALSE);
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if (srcrel)
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{
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palSourcei(src, AL_SOURCE_RELATIVE, AL_TRUE);
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@ -1148,6 +1200,7 @@ static qboolean OpenAL_InitLibrary(void)
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{(void*)&palGenBuffers, "alGenBuffers"},
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{(void*)&palIsBuffer, "alIsBuffer"},
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{(void*)&palBufferData, "alBufferData"},
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{(void*)&palBufferiv, "alBufferiv"},
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{(void*)&palDeleteBuffers, "alDeleteBuffers"},
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{(void*)&palListenerfv, "alListenerfv"},
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{(void*)&palSourcefv, "alSourcefv"},
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@ -1199,7 +1252,7 @@ static qboolean OpenAL_InitLibrary(void)
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#endif
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}
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static qboolean OpenAL_Init(soundcardinfo_t *sc, const char *devname)
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static qboolean OpenAL_Init(soundcardinfo_t *sc, const char *devname, qboolean qmix)
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{
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oalinfo_t *oali;
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@ -1217,12 +1270,15 @@ static qboolean OpenAL_Init(soundcardinfo_t *sc, const char *devname)
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if (!devname || !*devname)
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{
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if (s_al_disable.ival)
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if (s_al_disable.ival && !qmix)
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return false; //no default device
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devname = palcGetString(NULL, ALC_DEFAULT_DEVICE_SPECIFIER);
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}
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Q_snprintfz(sc->name, sizeof(sc->name), "%s", devname);
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Con_TPrintf("Initiating "SDRVNAME": %s.\n", devname);
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if (qmix)
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Con_TPrintf("Initiating "QMIX_SDRVNAME": %s.\n", devname);
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else
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Con_TPrintf("Initiating "SDRVNAME": %s.\n", devname);
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oali = Z_Malloc(sizeof(oalinfo_t));
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sc->handle = oali;
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@ -1235,7 +1291,7 @@ static qboolean OpenAL_Init(soundcardinfo_t *sc, const char *devname)
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size_t i = 0;
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ALCint attrs[5];
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palcGetStringiSOFT = (palcIsExtensionPresent(oali->OpenAL_Device, "ALC_SOFT_HRTF"))?palcGetProcAddress(oali->OpenAL_Device, "alcGetStringiSOFT"):NULL;
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palcGetStringiSOFT = (!qmix&&palcIsExtensionPresent(oali->OpenAL_Device, "ALC_SOFT_HRTF"))?palcGetProcAddress(oali->OpenAL_Device, "alcGetStringiSOFT"):NULL;
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if (palcGetStringiSOFT)
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{
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if (!*s_al_hrtf.string)
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@ -1297,6 +1353,7 @@ static qboolean OpenAL_Init(soundcardinfo_t *sc, const char *devname)
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memset(oali->source, 0, sizeof(*oali->source)*oali->max_sources);
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PrintALError("alGensources for normal sources");
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palListenerf(AL_GAIN, 1);
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palListenerfv(AL_POSITION, oali->ListenPos);
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#ifndef FTE_TARGET_WEB //webaudio sucks.
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palListenerfv(AL_VELOCITY, oali->ListenVel);
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@ -1304,6 +1361,7 @@ static qboolean OpenAL_Init(soundcardinfo_t *sc, const char *devname)
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palListenerfv(AL_ORIENTATION, oali->ListenOri);
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oali->can_source_spatialise = palIsExtensionPresent("AL_SOFT_source_spatialize");
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oali->can_looppoints = palIsExtensionPresent("AL_SOFT_loop_points");
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if (palcGetStringiSOFT)
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{
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@ -1576,85 +1634,98 @@ static ALuint OpenAL_LoadEffect(const struct reverbproperties_s *reverb)
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}
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#endif
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static qboolean QDECL OpenAL_InitCard(soundcardinfo_t *sc, const char *devname)
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#ifdef HAVE_MIXER
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#define CHUNKSAMPLES 1024
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static void *OAQM_LockBuffer (soundcardinfo_t *sc, unsigned int *sampidx)
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{
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oalinfo_t *oali;
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soundcardinfo_t *old;
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// extern soundcardinfo_t *sndcardinfo;
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//
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for (old = sndcardinfo; old; old = old->next)
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oalinfo_t *oali = sc->handle;
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if (oali->qmix.queuesize==countof(oali->qmix.queue))
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return NULL; //not available yet.
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return sc->sn.buffer;
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}
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static void OAQM_UnlockBuffer (soundcardinfo_t *sc, void *buffer)
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{
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}
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static void OAQM_Submit (soundcardinfo_t *sc, int start, int end)
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{
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oalinfo_t *oali = sc->handle;
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ALint buf;
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int framesize = sc->sn.samplebytes*sc->sn.numchannels;
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if (end==start)
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return;
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if (oali->qmix.queuesize == countof(oali->qmix.queue))
|
||||
return;
|
||||
palGenBuffers(1, &buf);
|
||||
switch(sc->sn.sampleformat)
|
||||
{
|
||||
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;
|
||||
}
|
||||
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;
|
||||
|
||||
if (OpenAL_Init(sc, devname) == false)
|
||||
return false;
|
||||
oali = sc->handle;
|
||||
sc->sn.samplepos = (sc->snd_completed+avail);
|
||||
sc->sn.samplepos *= sc->sn.numchannels;
|
||||
return sc->sn.samplepos;
|
||||
}
|
||||
|
||||
#ifdef FTE_TARGET_WEB
|
||||
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));
|
||||
#else
|
||||
Con_Printf( "AL_VERSION: %s\n",palGetString(AL_VERSION));
|
||||
Con_Printf( "AL_RENDERER: %s\n",palGetString(AL_RENDERER));
|
||||
Con_Printf( "AL_VENDOR: %s\n",palGetString(AL_VENDOR));
|
||||
#endif
|
||||
Con_DPrintf("AL_EXTENSIONS: %s\n",palGetString(AL_EXTENSIONS));
|
||||
Con_DPrintf("ALC_EXTENSIONS: %s\n",palcGetString(oali->OpenAL_Device,ALC_EXTENSIONS));
|
||||
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
|
||||
|
||||
sc->Shutdown = OpenAL_Shutdown;
|
||||
#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;
|
||||
|
||||
#ifdef MIXER_F32
|
||||
oali->canfloataudio = palIsExtensionPresent("AL_EXT_float32");
|
||||
#endif
|
||||
|
||||
sc->inactive_sound = true;
|
||||
sc->selfpainting = true;
|
||||
sc->sn.sampleformat = QSF_EXTERNALMIXER;
|
||||
|
||||
OnChangeALSettings(NULL, NULL);
|
||||
|
||||
#ifdef USEEFX
|
||||
PrintALError("preeffects");
|
||||
palSource3i = palGetProcAddress("alSource3i");
|
||||
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");
|
||||
|
||||
if (palGenAuxiliaryEffectSlots && s_al_use_reverb.ival)
|
||||
palGenAuxiliaryEffectSlots(1, &oali->effectslot);
|
||||
|
||||
oali->cureffect = ~0;
|
||||
PrintALError("posteffects");
|
||||
#endif
|
||||
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))
|
||||
{
|
||||
|
|
@ -1673,6 +1744,127 @@ static qboolean QDECL OpenAL_Enumerate(void (QDECL *callback)(const char *driver
|
|||
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;
|
||||
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
|
||||
{
|
||||
#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");
|
||||
palSource3i = palGetProcAddress("alSource3i");
|
||||
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");
|
||||
|
||||
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,
|
||||
|
|
@ -1680,6 +1872,19 @@ sounddriver_t OPENAL_Output =
|
|||
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)
|
||||
|
|
|
|||
|
|
@ -1843,6 +1843,7 @@ extern sounddriver_t Pulse_Output;
|
|||
sounddriver_t fte_weakstruct OSS_Output;
|
||||
#ifdef AVAIL_OPENAL
|
||||
extern sounddriver_t OPENAL_Output;
|
||||
extern sounddriver_t OPENAL_Output_Lame;
|
||||
#endif
|
||||
#ifdef __DJGPP__
|
||||
extern sounddriver_t SBLASTER_Output;
|
||||
|
|
@ -1915,6 +1916,9 @@ static sounddriver_t *outputdrivers[] =
|
|||
#endif
|
||||
&SNDIO_AudioOutput, //prefered on OpenBSD
|
||||
|
||||
#ifdef AVAIL_OPENAL
|
||||
&OPENAL_Output_Lame,//streaming quake's audio via openal instead of using openal properly. used in our browser port to work around issues with webaudio (at least in chromium).
|
||||
#endif
|
||||
#endif
|
||||
NULL
|
||||
};
|
||||
|
|
|
|||
|
|
@ -393,6 +393,10 @@ void S_PaintChannels(soundcardinfo_t *sc, int endtime)
|
|||
else
|
||||
SND_PaintChannel32F_O8I1(ch, scache, count, rate);
|
||||
break;
|
||||
#endif
|
||||
#ifdef FTE_TARGET_WEB
|
||||
case QAF_BLOB:
|
||||
break;
|
||||
#endif
|
||||
}
|
||||
ltime += count;
|
||||
|
|
|
|||
|
|
@ -390,7 +390,7 @@ struct soundcardinfo_s { //windows has one defined AFTER directsound
|
|||
void (*Submit) (soundcardinfo_t *sc, int start, int end); //if the ringbuffer is emulated, this is where you should push it to the device.
|
||||
void (*Shutdown) (soundcardinfo_t *sc); //kill the device
|
||||
unsigned int (*GetDMAPos) (soundcardinfo_t *sc); //get the current point that the hardware is reading from (the return value should not wrap, at least not very often)
|
||||
void (*SetEnvironmentReverb) (soundcardinfo_t *sc, size_t reverb); //if you have eax enabled, change the environment. fixme. generally this is a stub. optional.
|
||||
void (*SetEnvironmentReverb) (soundcardinfo_t *sc, size_t reverb); //if you have eax enabled, change the environment. generally this is a stub. optional.
|
||||
void (*Restore) (soundcardinfo_t *sc); //called before lock/unlock/lock/unlock/submit. optional
|
||||
void (*ChannelUpdate) (soundcardinfo_t *sc, channel_t *channel, chanupdatereason_t schanged); //properties of a sound effect changed. this is to notify hardware mixers. optional.
|
||||
void (*ListenerUpdate) (soundcardinfo_t *sc, int entnum, vec3_t origin, vec3_t forward, vec3_t right, vec3_t up, vec3_t velocity); //player moved or something. this is to notify hardware mixers. optional.
|
||||
|
|
|
|||
|
|
@ -302,7 +302,7 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
|
|||
#undef MAP_PROC //meh
|
||||
#define HALFLIFEMODELS //blurgh
|
||||
#undef SUPPORT_ICE //requires udp, so not usable. webrtc could be used instead, but that logic is out of our hands.
|
||||
#undef HAVE_MIXER //depend upon openal instead.
|
||||
// #undef HAVE_MIXER //depend upon openal instead.
|
||||
|
||||
//extra features stripped to try to reduce memory footprints
|
||||
#undef RUNTIMELIGHTING //too slow anyway
|
||||
|
|
|
|||
Loading…
Reference in New Issue