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Add support for AL_SOFT_loop_points. Implement quake-mixer-over-openal as a workaround for weird performance issues in Chromium.

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git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@6086 fc73d0e0-1445-4013-8a0c-d673dee63da5
This commit is contained in:
Spoike 2021-10-22 22:27:21 +00:00
parent 5018d3e650
commit 15bd67c8f0
5 changed files with 312 additions and 99 deletions
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399
engine/client/snd_al.c
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@ -35,9 +35,14 @@ We also have no doppler with WebAudio.
#define OPENAL_STATIC //our javascript port doesn't support dynamic linking (bss+data segments get too messy). #define OPENAL_STATIC //our javascript port doesn't support dynamic linking (bss+data segments get too messy).
#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. #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.
#define SDRVNAMEDESC "WebAudio:" #define SDRVNAMEDESC "WebAudio:"
#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.
#define QMIX_SDRVNAMEDESC "QMIX:"
#else #else
#define SDRVNAME "OpenAL" #define SDRVNAME "OpenAL"
#define SDRVNAMEDESC "OAL:" #define SDRVNAMEDESC "OAL:"
#define QMIX_SDRVNAME "qmix"
#define QMIX_SDRVNAMEDESC "OALQ:"
#define USEEFX #define USEEFX
#endif #endif
#ifndef HAVE_MIXER #ifndef HAVE_MIXER
@ -67,6 +72,7 @@ We also have no doppler with WebAudio.
#define palGenBuffers alGenBuffers #define palGenBuffers alGenBuffers
#define palIsBuffer alIsBuffer #define palIsBuffer alIsBuffer
#define palBufferData alBufferData #define palBufferData alBufferData
#define palBufferiv alBufferiv
#define palDeleteBuffers alDeleteBuffers #define palDeleteBuffers alDeleteBuffers
#define palListenerfv alListenerfv #define palListenerfv alListenerfv
#define palSourcefv alSourcefv #define palSourcefv alSourcefv
@ -136,6 +142,7 @@ static AL_API void (AL_APIENTRY *palDopplerFactor)( ALfloat value );
static AL_API void (AL_APIENTRY *palGenBuffers)( ALsizei n, ALuint* buffers ); static AL_API void (AL_APIENTRY *palGenBuffers)( ALsizei n, ALuint* buffers );
static AL_API ALboolean (AL_APIENTRY *palIsBuffer)( ALuint bid ); static AL_API ALboolean (AL_APIENTRY *palIsBuffer)( ALuint bid );
static AL_API void (AL_APIENTRY *palBufferData)( ALuint bid, ALenum format, const ALvoid* data, ALsizei size, ALsizei freq ); static AL_API void (AL_APIENTRY *palBufferData)( ALuint bid, ALenum format, const ALvoid* data, ALsizei size, ALsizei freq );
static AL_API void (AL_APIENTRY *palBufferiv)(ALuint buffer, ALenum param, const ALint *values);
static AL_API void (AL_APIENTRY *palDeleteBuffers)( ALsizei n, const ALuint* buffers ); static AL_API void (AL_APIENTRY *palDeleteBuffers)( ALsizei n, const ALuint* buffers );
static AL_API void (AL_APIENTRY *palListenerfv)( ALenum param, const ALfloat* values ); static AL_API void (AL_APIENTRY *palListenerfv)( ALenum param, const ALfloat* values );
@ -326,6 +333,9 @@ static AL_API ALvoid (AL_APIENTRY *palEffectfv)(ALuint effect, ALenum param, con
//AL_SOFT_source_spatialize //AL_SOFT_source_spatialize
#define AL_SOURCE_SPATIALIZE_SOFT 0x1214 #define AL_SOURCE_SPATIALIZE_SOFT 0x1214
//AL_SOFT_loop_points
#define AL_LOOP_POINTS_SOFT 0x2015
//ALC_SOFT_HRTF //ALC_SOFT_HRTF
#define ALC_HRTF_SOFT 0x1992 #define ALC_HRTF_SOFT 0x1992
#define ALC_DONT_CARE_SOFT 0x0002 #define ALC_DONT_CARE_SOFT 0x0002
@ -357,7 +367,11 @@ static void S_Info(void);
static void S_Shutdown_f(void); static void S_Shutdown_f(void);
*/ */
#ifdef FTE_TARGET_WEB
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.");
#else
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."); 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.");
#endif
static cvar_t s_al_debug = CVARD("s_al_debug", "0", "Enables periodic checks for OpenAL errors."); static cvar_t s_al_debug = CVARD("s_al_debug", "0", "Enables periodic checks for OpenAL errors.");
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."); 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.");
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."); 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.");
@ -384,6 +398,12 @@ enum distancemodel_e
typedef struct typedef struct
{ {
struct
{
ALuint handle;
unsigned int queuesize;
ALuint queue[64];
} qmix;
struct struct
{ {
ALuint handle; ALuint handle;
@ -403,11 +423,13 @@ typedef struct
ALCdevice *OpenAL_Device; ALCdevice *OpenAL_Device;
ALCcontext *OpenAL_Context; ALCcontext *OpenAL_Context;
qboolean can_source_spatialise; qboolean can_source_spatialise;
qboolean can_looppoints;
int ListenEnt; //listener's entity number, so we don't get weird sound displacements int ListenEnt; //listener's entity number, so we don't get weird sound displacements
ALfloat ListenPos[3]; //their origin. ALfloat ListenPos[3]; //their origin.
ALfloat ListenVel[3]; // Velocity of the listener. ALfloat ListenVel[3]; // Velocity of the listener.
ALfloat ListenOri[6]; // Orientation of the listener. (first 3 elements are "at", second 3 are "up") ALfloat ListenOri[6]; // Orientation of the listener. (first 3 elements are "at", second 3 are "up")
unsigned int listenerdirty;
#ifdef MIXER_F32 #ifdef MIXER_F32
qboolean canfloataudio; qboolean canfloataudio;
@ -455,7 +477,7 @@ static void PrintALError(char *string)
Con_Printf("OpenAL - %s: %x: %s\n",string,err,text); 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) static qboolean OpenAL_LoadCache(oalinfo_t *oali, unsigned int *bufptr, sfxcache_t *sc, float volume, int loopstart)
{ {
unsigned int fmt; unsigned int fmt;
unsigned int size; unsigned int size;
@ -599,6 +621,12 @@ static qboolean OpenAL_LoadCache(oalinfo_t *oali, unsigned int *bufptr, sfxcache
} }
} }
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 //FIXME: we need to handle oal-oom error codes
PrintALError("Buffer Data"); PrintALError("Buffer Data");
@ -624,6 +652,7 @@ static void QDECL OpenAL_CvarInit(void)
static void OpenAL_ListenerUpdate(soundcardinfo_t *sc, int entnum, vec3_t origin, vec3_t forward, vec3_t right, vec3_t up, vec3_t velocity) 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; oalinfo_t *oali = sc->handle;
vec3_t vel;
if (snd_doppler.modified) if (snd_doppler.modified)
{ {
@ -631,26 +660,46 @@ static void OpenAL_ListenerUpdate(soundcardinfo_t *sc, int entnum, vec3_t origin
OnChangeALSettings(NULL,NULL); OnChangeALSettings(NULL,NULL);
} }
VectorScale(velocity, s_al_velocityscale.value/35.0, oali->ListenVel); if (!VectorCompare(origin, oali->ListenPos))
VectorCopy(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; oali->ListenEnt = entnum;
oali->ListenOri[0] = forward[0]; if (!VectorCompare(forward, oali->ListenOri) || !VectorCompare(up, oali->ListenOri+3))
oali->ListenOri[1] = forward[1]; {
oali->ListenOri[2] = forward[2]; oali->ListenOri[0] = forward[0];
oali->ListenOri[3] = up[0]; oali->ListenOri[1] = forward[1];
oali->ListenOri[4] = up[1]; oali->ListenOri[2] = forward[2];
oali->ListenOri[5] = up[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) if (!s_al_static_listener.value)
{ {
palListenerf(AL_GAIN, 1); //I'm using listenerdirty flags because emscripten's openal stuff seems to be wasting massive amounts of time on these.
palListenerfv(AL_POSITION, oali->ListenPos); // palListenerf(AL_GAIN, 1);
if (oali->listenerdirty & 1)
palListenerfv(AL_POSITION, oali->ListenPos);
#ifndef FTE_TARGET_WEB //webaudio sucks. #ifndef FTE_TARGET_WEB //webaudio sucks.
palListenerfv(AL_VELOCITY, oali->ListenVel); if (oali->listenerdirty & 2)
palListenerfv(AL_VELOCITY, oali->ListenVel);
#endif #endif
palListenerfv(AL_ORIENTATION, oali->ListenOri); if (oali->listenerdirty & 4)
palListenerfv(AL_ORIENTATION, oali->ListenOri);
oali->listenerdirty = 0;
} }
} }
@ -760,6 +809,7 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
} }
oali->source[chnum].handle = src; oali->source[chnum].handle = src;
oali->source[chnum].allocated = true; oali->source[chnum].allocated = true;
oali->source[chnum].queuesize = 0;
schanged |= CUR_EVERYTHING; //should normally be true anyway, but hey schanged |= CUR_EVERYTHING; //should normally be true anyway, but hey
} }
else else
@ -829,7 +879,9 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
cvolume *= mastervolume.value; 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. //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.
stream = sfx->decoder.decodedata || sfx->loopstart > 0; //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) if ((schanged&CUR_SOUNDCHANGE) || stream)
{ {
@ -892,7 +944,7 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
{ {
//build a buffer with it and queue it up. //build a buffer with it and queue it up.
//buffer will be purged later on when its unqueued //buffer will be purged later on when its unqueued
if (OpenAL_LoadCache(oali, &buf, &sbuf, max(1,cvolume))) if (OpenAL_LoadCache(oali, &buf, &sbuf, max(1,cvolume), 0))
{ {
palSourceQueueBuffers(src, 1, &buf); palSourceQueueBuffers(src, 1, &buf);
oali->source[chnum].queue[oali->source[chnum].queuesize++] = buf; oali->source[chnum].queue[oali->source[chnum].queuesize++] = buf;
@ -907,7 +959,7 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
silence.data = NULL; silence.data = NULL;
silence.length = 0.1 * silence.speed; silence.length = 0.1 * silence.speed;
silence.soundoffset = 0; silence.soundoffset = 0;
if (OpenAL_LoadCache(oali, &buf, &silence, 1)) if (OpenAL_LoadCache(oali, &buf, &silence, 1, 0))
{ {
palSourceQueueBuffers(src, 1, &buf); palSourceQueueBuffers(src, 1, &buf);
oali->source[chnum].queue[oali->source[chnum].queuesize++] = buf; oali->source[chnum].queue[oali->source[chnum].queuesize++] = buf;
@ -938,10 +990,12 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
silence.data = NULL; silence.data = NULL;
silence.length = 0.1 * silence.speed; silence.length = 0.1 * silence.speed;
silence.soundoffset = 0; silence.soundoffset = 0;
if (OpenAL_LoadCache(oali, &buf, &silence, 1)) if (OpenAL_LoadCache(oali, &buf, &silence, 1, 0))
{ {
palSourceQueueBuffers(src, 1, &buf); palSourceQueueBuffers(src, 1, &buf);
oali->source[chnum].queue[oali->source[chnum].queuesize++] = 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);
} }
} }
} }
@ -965,7 +1019,7 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
{ //unstreamed { //unstreamed
if (!sfx->decoder.buf) if (!sfx->decoder.buf)
return; return;
oali->sounds[sndnum].allocated = OpenAL_LoadCache(oali, &buf, sfx->decoder.buf, 1); oali->sounds[sndnum].allocated = OpenAL_LoadCache(oali, &buf, sfx->decoder.buf, 1, sfx->loopstart);
if (!oali->sounds[sndnum].allocated) if (!oali->sounds[sndnum].allocated)
return; return;
oali->sounds[sndnum].buffer = buf; oali->sounds[sndnum].buffer = buf;
@ -985,10 +1039,11 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
} }
#endif #endif
palSourcei(src, AL_BUFFER, buf); 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. palSourcef(src, AL_GAIN, min(cvolume, 1)); //openal only supports a max volume of 1. anything above is an error and will be clamped.
srcrel = (chan->flags & CF_NOSPACIALISE) || (chan->entnum && chan->entnum == oali->ListenEnt) || !chan->dist_mult;
if (srcrel) if (srcrel)
{ {
palSourcefv(src, AL_POSITION, vec3_origin); palSourcefv(src, AL_POSITION, vec3_origin);
@ -1004,9 +1059,6 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
#endif #endif
} }
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 (schanged) if (schanged)
{ {
if (schanged == CUR_UPDATE && chan->pos) if (schanged == CUR_UPDATE && chan->pos)
@ -1028,7 +1080,7 @@ static void OpenAL_ChannelUpdate(soundcardinfo_t *sc, channel_t *chan, chanupdat
} }
#endif #endif
palSourcei(src, AL_LOOPING, (!stream && ((chan->flags & CF_FORCELOOP)||sfx->loopstart==0))?AL_TRUE:AL_FALSE); palSourcei(src, AL_LOOPING, (!stream && ((chan->flags & CF_FORCELOOP)||(sfx->loopstart>=0&&!stream)))?AL_TRUE:AL_FALSE);
if (srcrel) if (srcrel)
{ {
palSourcei(src, AL_SOURCE_RELATIVE, AL_TRUE); palSourcei(src, AL_SOURCE_RELATIVE, AL_TRUE);
@ -1148,6 +1200,7 @@ static qboolean OpenAL_InitLibrary(void)
{(void*)&palGenBuffers, "alGenBuffers"}, {(void*)&palGenBuffers, "alGenBuffers"},
{(void*)&palIsBuffer, "alIsBuffer"}, {(void*)&palIsBuffer, "alIsBuffer"},
{(void*)&palBufferData, "alBufferData"}, {(void*)&palBufferData, "alBufferData"},
{(void*)&palBufferiv, "alBufferiv"},
{(void*)&palDeleteBuffers, "alDeleteBuffers"}, {(void*)&palDeleteBuffers, "alDeleteBuffers"},
{(void*)&palListenerfv, "alListenerfv"}, {(void*)&palListenerfv, "alListenerfv"},
{(void*)&palSourcefv, "alSourcefv"}, {(void*)&palSourcefv, "alSourcefv"},
@ -1199,7 +1252,7 @@ static qboolean OpenAL_InitLibrary(void)
#endif #endif
} }
static qboolean OpenAL_Init(soundcardinfo_t *sc, const char *devname) static qboolean OpenAL_Init(soundcardinfo_t *sc, const char *devname, qboolean qmix)
{ {
oalinfo_t *oali; oalinfo_t *oali;
@ -1217,12 +1270,15 @@ static qboolean OpenAL_Init(soundcardinfo_t *sc, const char *devname)
if (!devname || !*devname) if (!devname || !*devname)
{ {
if (s_al_disable.ival) if (s_al_disable.ival && !qmix)
return false; //no default device return false; //no default device
devname = palcGetString(NULL, ALC_DEFAULT_DEVICE_SPECIFIER); devname = palcGetString(NULL, ALC_DEFAULT_DEVICE_SPECIFIER);
} }
Q_snprintfz(sc->name, sizeof(sc->name), "%s", devname); Q_snprintfz(sc->name, sizeof(sc->name), "%s", devname);
Con_TPrintf("Initiating "SDRVNAME": %s.\n", 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)); oali = Z_Malloc(sizeof(oalinfo_t));
sc->handle = oali; sc->handle = oali;
@ -1235,7 +1291,7 @@ static qboolean OpenAL_Init(soundcardinfo_t *sc, const char *devname)
size_t i = 0; size_t i = 0;
ALCint attrs[5]; ALCint attrs[5];
palcGetStringiSOFT = (palcIsExtensionPresent(oali->OpenAL_Device, "ALC_SOFT_HRTF"))?palcGetProcAddress(oali->OpenAL_Device, "alcGetStringiSOFT"):NULL; palcGetStringiSOFT = (!qmix&&palcIsExtensionPresent(oali->OpenAL_Device, "ALC_SOFT_HRTF"))?palcGetProcAddress(oali->OpenAL_Device, "alcGetStringiSOFT"):NULL;
if (palcGetStringiSOFT) if (palcGetStringiSOFT)
{ {
if (!*s_al_hrtf.string) if (!*s_al_hrtf.string)
@ -1297,6 +1353,7 @@ static qboolean OpenAL_Init(soundcardinfo_t *sc, const char *devname)
memset(oali->source, 0, sizeof(*oali->source)*oali->max_sources); memset(oali->source, 0, sizeof(*oali->source)*oali->max_sources);
PrintALError("alGensources for normal sources"); PrintALError("alGensources for normal sources");
palListenerf(AL_GAIN, 1);
palListenerfv(AL_POSITION, oali->ListenPos); palListenerfv(AL_POSITION, oali->ListenPos);
#ifndef FTE_TARGET_WEB //webaudio sucks. #ifndef FTE_TARGET_WEB //webaudio sucks.
palListenerfv(AL_VELOCITY, oali->ListenVel); palListenerfv(AL_VELOCITY, oali->ListenVel);
@ -1304,6 +1361,7 @@ static qboolean OpenAL_Init(soundcardinfo_t *sc, const char *devname)
palListenerfv(AL_ORIENTATION, oali->ListenOri); palListenerfv(AL_ORIENTATION, oali->ListenOri);
oali->can_source_spatialise = palIsExtensionPresent("AL_SOFT_source_spatialize"); oali->can_source_spatialise = palIsExtensionPresent("AL_SOFT_source_spatialize");
oali->can_looppoints = palIsExtensionPresent("AL_SOFT_loop_points");
if (palcGetStringiSOFT) if (palcGetStringiSOFT)
{ {
@ -1576,85 +1634,98 @@ static ALuint OpenAL_LoadEffect(const struct reverbproperties_s *reverb)
} }
#endif #endif
static qboolean QDECL OpenAL_InitCard(soundcardinfo_t *sc, const char *devname) #ifdef HAVE_MIXER
#define CHUNKSAMPLES 1024
static void *OAQM_LockBuffer (soundcardinfo_t *sc, unsigned int *sampidx)
{ {
oalinfo_t *oali; oalinfo_t *oali = sc->handle;
if (oali->qmix.queuesize==countof(oali->qmix.queue))
soundcardinfo_t *old; return NULL; //not available yet.
// extern soundcardinfo_t *sndcardinfo; return sc->sn.buffer;
}
// static void OAQM_UnlockBuffer (soundcardinfo_t *sc, void *buffer)
for (old = sndcardinfo; old; old = old->next) {
}
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)
{ {
if (old->Shutdown == OpenAL_Shutdown) 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);
//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 break;
Con_Printf(CON_ERROR SDRVNAME ": only a single device may be active at once\n"); case QSF_S16:
return false; 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) sc->sn.samplepos = (sc->snd_completed+avail);
return false; sc->sn.samplepos *= sc->sn.numchannels;
oali = sc->handle; return sc->sn.samplepos;
}
#ifdef FTE_TARGET_WEB static qboolean QDECL OpenAL_Enumerate_QMix(void (QDECL *callback)(const char *driver, const char *devicecode, const char *readabledevice))
Con_DPrintf( "AL_VERSION: %s\n",palGetString(AL_VERSION)); {
Con_DPrintf( "AL_RENDERER: %s\n",palGetString(AL_RENDERER)); const char *devnames;
Con_DPrintf( "AL_VENDOR: %s\n",palGetString(AL_VENDOR)); if (!OpenAL_InitLibrary())
#else return true; //enumerate nothing if al is disabled
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));
sc->Shutdown = OpenAL_Shutdown; devnames = palcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER);
#ifdef USEEFX if (!devnames)
sc->SetEnvironmentReverb = OpenAL_SetReverb; devnames = palcGetString(NULL, ALC_DEVICE_SPECIFIER);
#endif while(*devnames)
sc->ChannelUpdate = OpenAL_ChannelUpdate; {
sc->ListenerUpdate = OpenAL_ListenerUpdate; callback(QMIX_SDRVNAME, devnames, va(QMIX_SDRVNAMEDESC"%s", devnames));
sc->GetChannelPos = OpenAL_GetChannelPos; devnames += strlen(devnames)+1;
//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
return true; return true;
} }
#endif
static qboolean QDECL OpenAL_Enumerate(void (QDECL *callback)(const char *driver, const char *devicecode, const char *readabledevice)) 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; 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 = sounddriver_t OPENAL_Output =
{ {
SDRVNAME, SDRVNAME,
@ -1680,6 +1872,19 @@ sounddriver_t OPENAL_Output =
OpenAL_Enumerate, OpenAL_Enumerate,
OpenAL_CvarInit 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) #if defined(VOICECHAT)

4
engine/client/snd_dma.c
View file

@ -1843,6 +1843,7 @@ extern sounddriver_t Pulse_Output;
sounddriver_t fte_weakstruct OSS_Output; sounddriver_t fte_weakstruct OSS_Output;
#ifdef AVAIL_OPENAL #ifdef AVAIL_OPENAL
extern sounddriver_t OPENAL_Output; extern sounddriver_t OPENAL_Output;
extern sounddriver_t OPENAL_Output_Lame;
#endif #endif
#ifdef __DJGPP__ #ifdef __DJGPP__
extern sounddriver_t SBLASTER_Output; extern sounddriver_t SBLASTER_Output;
@ -1915,6 +1916,9 @@ static sounddriver_t *outputdrivers[] =
#endif #endif
&SNDIO_AudioOutput, //prefered on OpenBSD &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 #endif
NULL NULL
}; };

4
engine/client/snd_mix.c
View file

@ -393,6 +393,10 @@ void S_PaintChannels(soundcardinfo_t *sc, int endtime)
else else
SND_PaintChannel32F_O8I1(ch, scache, count, rate); SND_PaintChannel32F_O8I1(ch, scache, count, rate);
break; break;
#endif
#ifdef FTE_TARGET_WEB
case QAF_BLOB:
break;
#endif #endif
} }
ltime += count; ltime += count;

2
engine/client/sound.h
View file

@ -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 (*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 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) 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 (*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 (*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. 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.

2
engine/common/bothdefs.h
View file

@ -302,7 +302,7 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
#undef MAP_PROC //meh #undef MAP_PROC //meh
#define HALFLIFEMODELS //blurgh #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 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 //extra features stripped to try to reduce memory footprints
#undef RUNTIMELIGHTING //too slow anyway #undef RUNTIMELIGHTING //too slow anyway