dhewm3/neo/sound/snd_emitter.cpp

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/*
===========================================================================
Doom 3 GPL Source Code
Copyright (C) 1999-2011 id Software LLC, a ZeniMax Media company.
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2011-12-06 16:14:59 +00:00
This file is part of the Doom 3 GPL Source Code ("Doom 3 Source Code").
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Doom 3 Source Code is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.
Doom 3 Source Code is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with Doom 3 Source Code. If not, see <http://www.gnu.org/licenses/>.
In addition, the Doom 3 Source Code is also subject to certain additional terms. You should have received a copy of these additional terms immediately following the terms and conditions of the GNU General Public License which accompanied the Doom 3 Source Code. If not, please request a copy in writing from id Software at the address below.
If you have questions concerning this license or the applicable additional terms, you may contact in writing id Software LLC, c/o ZeniMax Media Inc., Suite 120, Rockville, Maryland 20850 USA.
===========================================================================
*/
#include "sys/platform.h"
#include "framework/Session.h"
#include "renderer/RenderWorld.h"
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#include "sound/snd_local.h"
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/*
===================
idSoundFade::Clear
===================
*/
void idSoundFade::Clear() {
fadeStart44kHz = 0;
fadeEnd44kHz = 0;
fadeStartVolume = 0;
fadeEndVolume = 0;
}
/*
===================
idSoundFade::FadeDbAt44kHz
===================
*/
float idSoundFade::FadeDbAt44kHz( int current44kHz ) {
float fadeDb;
if ( current44kHz >= fadeEnd44kHz ) {
fadeDb = fadeEndVolume;
} else if ( current44kHz > fadeStart44kHz ) {
float fraction = ( fadeEnd44kHz - fadeStart44kHz );
float over = ( current44kHz - fadeStart44kHz );
fadeDb = fadeStartVolume + ( fadeEndVolume - fadeStartVolume ) * over / fraction;
} else {
fadeDb = fadeStartVolume;
}
return fadeDb;
}
//========================================================================
/*
=======================
GeneratePermutedList
Fills in elements[0] .. elements[numElements-1] with a permutation of
0 .. numElements-1 based on the permute parameter
numElements == 3
maxPermute = 6
permute 0 = 012
permute 1 = 021
permute 2 = 102
permute 3 = 120
permute 4 = 201
permute 5 = 210
=======================
*/
void PermuteList_r( int *list, int listLength, int permute, int maxPermute ) {
if ( listLength < 2 ) {
return;
}
permute %= maxPermute;
int swap = permute * listLength / maxPermute;
int old = list[swap];
list[swap] = list[0];
list[0] = old;
maxPermute /= listLength;
PermuteList_r( list + 1, listLength - 1, permute, maxPermute );
}
int Factorial( int val ) {
int fact = val;
while ( val > 1 ) {
val--;
fact *= val;
}
return fact;
}
void GeneratePermutedList( int *list, int listLength, int permute ) {
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for ( int i = 0 ; i < listLength ; i++ ) {
list[i] = i;
}
// we can't calculate > 12 factorial, so we can't easily build a permuted list
if ( listLength > 12 ) {
return;
}
// calculate listLength factorial
int maxPermute = Factorial( listLength );
// recursively permute
PermuteList_r( list, listLength, permute, maxPermute );
}
void TestPermutations( void ) {
int list[SOUND_MAX_LIST_WAVS];
for ( int len = 1 ; len < 5 ; len++ ) {
common->Printf( "list length: %i\n", len );
int max = Factorial( len );
for ( int j = 0 ; j < max * 2 ; j++ ) {
GeneratePermutedList( list, len, j );
common->Printf( "%4i : ", j );
for ( int k = 0 ; k < len ; k++ ) {
common->Printf( "%i", list[k] );
}
common->Printf( "\n" );
}
}
}
//=====================================================================================
/*
===================
idSoundChannel::idSoundChannel
===================
*/
idSoundChannel::idSoundChannel( void ) {
decoder = NULL;
Clear();
}
/*
===================
idSoundChannel::~idSoundChannel
===================
*/
idSoundChannel::~idSoundChannel( void ) {
Clear();
}
/*
===================
idSoundChannel::Clear
===================
*/
void idSoundChannel::Clear( void ) {
int j;
Stop();
soundShader = NULL;
lastVolume = 0.0f;
triggerChannel = SCHANNEL_ANY;
channelFade.Clear();
diversity = 0.0f;
leadinSample = NULL;
trigger44kHzTime = 0;
stopped = false;
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for( j = 0; j < 6; j++ ) {
lastV[j] = 0.0f;
}
memset( &parms, 0, sizeof(parms) );
triggered = false;
paused = false;
openalSource = 0;
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openalStreamingOffset = 0;
openalStreamingBuffer[0] = openalStreamingBuffer[1] = openalStreamingBuffer[2] = 0;
lastopenalStreamingBuffer[0] = lastopenalStreamingBuffer[1] = lastopenalStreamingBuffer[2] = 0;
}
/*
===================
idSoundChannel::Start
===================
*/
void idSoundChannel::Start( void ) {
triggerState = true;
if ( decoder == NULL ) {
decoder = idSampleDecoder::Alloc();
}
}
/*
===================
idSoundChannel::Stop
===================
*/
void idSoundChannel::Stop( void ) {
triggerState = false;
stopped = true;
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if ( decoder != NULL ) {
idSampleDecoder::Free( decoder );
decoder = NULL;
}
}
/*
===================
idSoundChannel::ALStop
===================
*/
void idSoundChannel::ALStop( void ) {
if ( alIsSource( openalSource ) ) {
alSourceStop( openalSource );
alSourcei( openalSource, AL_BUFFER, 0 );
// unassociate effect slot from source, so the effect slot can be deleted on shutdown
// even though the source itself is deleted later (in idSoundSystemLocal::Shutdown())
alSource3i( openalSource, AL_AUXILIARY_SEND_FILTER, AL_EFFECTSLOT_NULL, 0, AL_FILTER_NULL );
soundSystemLocal.FreeOpenALSource( openalSource );
}
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if ( openalStreamingBuffer[0] && openalStreamingBuffer[1] && openalStreamingBuffer[2] ) {
alGetError();
alDeleteBuffers( 3, &openalStreamingBuffer[0] );
if ( alGetError() == AL_NO_ERROR ) {
openalStreamingBuffer[0] = openalStreamingBuffer[1] = openalStreamingBuffer[2] = 0;
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}
}
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if ( lastopenalStreamingBuffer[0] && lastopenalStreamingBuffer[1] && lastopenalStreamingBuffer[2] ) {
alGetError();
alDeleteBuffers( 3, &lastopenalStreamingBuffer[0] );
if ( alGetError() == AL_NO_ERROR ) {
lastopenalStreamingBuffer[0] = lastopenalStreamingBuffer[1] = lastopenalStreamingBuffer[2] = 0;
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}
}
}
/*
===================
idSoundChannel::GatherChannelSamples
Will always return 44kHz samples for the given range, even if it deeply looped or
out of the range of the unlooped samples. Handles looping between multiple different
samples and leadins
===================
*/
void idSoundChannel::GatherChannelSamples( int sampleOffset44k, int sampleCount44k, float *dest ) const {
float *dest_p = dest;
int len;
//Sys_DebugPrintf( "msec:%i sample:%i : %i : %i\n", Sys_Milliseconds(), soundSystemLocal.GetCurrent44kHzTime(), sampleOffset44k, sampleCount44k ); //!@#
// negative offset times will just zero fill
if ( sampleOffset44k < 0 ) {
len = -sampleOffset44k;
if ( len > sampleCount44k ) {
len = sampleCount44k;
}
memset( dest_p, 0, len * sizeof( dest_p[0] ) );
dest_p += len;
sampleCount44k -= len;
sampleOffset44k += len;
}
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// grab part of the leadin sample
idSoundSample *leadin = leadinSample;
if ( !leadin || sampleOffset44k < 0 || sampleCount44k <= 0 ) {
memset( dest_p, 0, sampleCount44k * sizeof( dest_p[0] ) );
return;
}
if ( sampleOffset44k < leadin->LengthIn44kHzSamples() ) {
len = leadin->LengthIn44kHzSamples() - sampleOffset44k;
if ( len > sampleCount44k ) {
len = sampleCount44k;
}
// decode the sample
decoder->Decode( leadin, sampleOffset44k, len, dest_p );
dest_p += len;
sampleCount44k -= len;
sampleOffset44k += len;
}
// if not looping, zero fill any remaining spots
if ( !soundShader || !( parms.soundShaderFlags & SSF_LOOPING ) ) {
memset( dest_p, 0, sampleCount44k * sizeof( dest_p[0] ) );
return;
}
// fill the remainder with looped samples
idSoundSample *loop = soundShader->entries[0];
if ( !loop ) {
memset( dest_p, 0, sampleCount44k * sizeof( dest_p[0] ) );
return;
}
sampleOffset44k -= leadin->LengthIn44kHzSamples();
while( sampleCount44k > 0 ) {
int totalLen = loop->LengthIn44kHzSamples();
sampleOffset44k %= totalLen;
len = totalLen - sampleOffset44k;
if ( len > sampleCount44k ) {
len = sampleCount44k;
}
// decode the sample
decoder->Decode( loop, sampleOffset44k, len, dest_p );
dest_p += len;
sampleCount44k -= len;
sampleOffset44k += len;
}
}
//=====================================================================================
/*
===============
idSoundEmitterLocal::idSoundEmitterLocal
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===============
*/
idSoundEmitterLocal::idSoundEmitterLocal( void ) {
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soundWorld = NULL;
Clear();
}
/*
===============
idSoundEmitterLocal::~idSoundEmitterLocal
===============
*/
idSoundEmitterLocal::~idSoundEmitterLocal( void ) {
Clear();
}
/*
===============
idSoundEmitterLocal::Clear
===============
*/
void idSoundEmitterLocal::Clear( void ) {
int i;
for( i = 0; i < SOUND_MAX_CHANNELS; i++ ) {
channels[i].ALStop();
channels[i].Clear();
}
removeStatus = REMOVE_STATUS_SAMPLEFINISHED;
distance = 0.0f;
lastValidPortalArea = -1;
playing = false;
hasShakes = false;
ampTime = 0; // last time someone queried
amplitude = 0;
maxDistance = 10.0f; // meters
spatializedOrigin.Zero();
memset( &parms, 0, sizeof( parms ) );
}
/*
==================
idSoundEmitterLocal::OverrideParms
==================
*/
void idSoundEmitterLocal::OverrideParms( const soundShaderParms_t *base,
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const soundShaderParms_t *over, soundShaderParms_t *out ) {
if ( !over ) {
*out = *base;
return;
}
if ( over->minDistance ) {
out->minDistance = over->minDistance;
} else {
out->minDistance = base->minDistance;
}
if ( over->maxDistance ) {
out->maxDistance = over->maxDistance;
} else {
out->maxDistance = base->maxDistance;
}
if ( over->shakes ) {
out->shakes = over->shakes;
} else {
out->shakes = base->shakes;
}
if ( over->volume ) {
out->volume = over->volume;
} else {
out->volume = base->volume;
}
if ( over->soundClass ) {
out->soundClass = over->soundClass;
} else {
out->soundClass = base->soundClass;
}
out->soundShaderFlags = base->soundShaderFlags | over->soundShaderFlags;
}
/*
==================
idSoundEmitterLocal::CheckForCompletion
Checks to see if all the channels have completed, clearing the playing flag if necessary.
Sets the playing and shakes bools.
==================
*/
void idSoundEmitterLocal::CheckForCompletion( int current44kHzTime ) {
bool hasActive;
int i;
hasActive = false;
hasShakes = false;
if ( playing ) {
for ( i = 0; i < SOUND_MAX_CHANNELS; i++ ) {
idSoundChannel *chan = &channels[i];
if ( !chan->triggerState ) {
continue;
}
const idSoundShader *shader = chan->soundShader;
if ( !shader ) {
continue;
}
// see if this channel has completed
if ( !( chan->parms.soundShaderFlags & SSF_LOOPING ) ) {
ALint state = AL_PLAYING;
if ( alIsSource( chan->openalSource ) ) {
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alGetSourcei( chan->openalSource, AL_SOURCE_STATE, &state );
}
idSlowChannel slow = GetSlowChannel( chan );
if ( soundWorld->slowmoActive && slow.IsActive() ) {
if ( slow.GetCurrentPosition().time >= chan->leadinSample->LengthIn44kHzSamples() / 2 ) {
chan->Stop();
// if this was an onDemand sound, purge the sample now
if ( chan->leadinSample->onDemand ) {
chan->leadinSample->PurgeSoundSample();
}
}
} else if ( ( chan->trigger44kHzTime + chan->leadinSample->LengthIn44kHzSamples() < current44kHzTime ) || ( chan->stopped ) ) {
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chan->Stop();
// free hardware resources
chan->ALStop();
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// if this was an onDemand sound, purge the sample now
if ( chan->leadinSample->onDemand ) {
chan->leadinSample->PurgeSoundSample();
}
}
}
// free decoder memory if no sound was decoded for a while
if ( chan->decoder != NULL && chan->decoder->GetLastDecodeTime() < current44kHzTime - SOUND_DECODER_FREE_DELAY ) {
chan->decoder->ClearDecoder();
}
hasActive = true;
if ( chan->parms.shakes > 0.0f ) {
hasShakes = true;
}
}
}
// mark the entire sound emitter as non-playing if there aren't any active channels
if ( !hasActive ) {
playing = false;
if ( removeStatus == REMOVE_STATUS_WAITSAMPLEFINISHED ) {
// this can now be reused by the next request for a new soundEmitter
removeStatus = REMOVE_STATUS_SAMPLEFINISHED;
}
}
}
/*
===================
idSoundEmitterLocal::Spatialize
Called once each sound frame by the main thread from idSoundWorldLocal::PlaceOrigin
===================
*/
void idSoundEmitterLocal::Spatialize( idVec3 listenerPos, int listenerArea, idRenderWorld *rw ) {
int i;
//
// work out the maximum distance of all the playing channels
//
maxDistance = 0;
for ( i = 0; i < SOUND_MAX_CHANNELS; i++ ) {
idSoundChannel *chan = &channels[i];
if ( !chan->triggerState ) {
continue;
}
if ( chan->parms.maxDistance > maxDistance ) {
maxDistance = chan->parms.maxDistance;
}
}
//
// work out where the sound comes from
//
idVec3 realOrigin = origin * DOOM_TO_METERS;
idVec3 len = listenerPos - realOrigin;
realDistance = len.LengthFast();
if ( realDistance >= maxDistance ) {
// no way to possibly hear it
distance = realDistance;
return;
}
//
// work out virtual origin and distance, which may be from a portal instead of the actual origin
//
distance = maxDistance * METERS_TO_DOOM;
if ( listenerArea == -1 ) { // listener is outside the world
return;
}
if ( rw ) {
// we have a valid renderWorld
int soundInArea = rw->PointInArea( origin );
if ( soundInArea == -1 ) {
if ( lastValidPortalArea == -1 ) { // sound is outside the world
distance = realDistance;
spatializedOrigin = origin; // sound is in our area
return;
}
soundInArea = lastValidPortalArea;
}
lastValidPortalArea = soundInArea;
if ( soundInArea == listenerArea ) {
distance = realDistance;
spatializedOrigin = origin; // sound is in our area
return;
}
soundWorld->ResolveOrigin( 0, NULL, soundInArea, 0.0f, origin, this );
distance /= METERS_TO_DOOM;
} else {
// no portals available
distance = realDistance;
spatializedOrigin = origin; // sound is in our area
}
}
/*
===========================================================================================
PUBLIC FUNCTIONS
===========================================================================================
*/
/*
=====================
idSoundEmitterLocal::UpdateEmitter
=====================
*/
void idSoundEmitterLocal::UpdateEmitter( const idVec3 &origin, int listenerId, const soundShaderParms_t *parms ) {
if ( !parms ) {
common->Error( "idSoundEmitterLocal::UpdateEmitter: NULL parms" );
}
if ( soundWorld && soundWorld->writeDemo ) {
soundWorld->writeDemo->WriteInt( DS_SOUND );
soundWorld->writeDemo->WriteInt( SCMD_UPDATE );
soundWorld->writeDemo->WriteInt( index );
soundWorld->writeDemo->WriteVec3( origin );
soundWorld->writeDemo->WriteInt( listenerId );
soundWorld->writeDemo->WriteFloat( parms->minDistance );
soundWorld->writeDemo->WriteFloat( parms->maxDistance );
soundWorld->writeDemo->WriteFloat( parms->volume );
soundWorld->writeDemo->WriteFloat( parms->shakes );
soundWorld->writeDemo->WriteInt( parms->soundShaderFlags );
soundWorld->writeDemo->WriteInt( parms->soundClass );
}
this->origin = origin;
this->listenerId = listenerId;
this->parms = *parms;
// FIXME: change values on all channels?
}
/*
=====================
idSoundEmitterLocal::Free
They are never truly freed, just marked so they can be reused by the soundWorld
=====================
*/
void idSoundEmitterLocal::Free( bool immediate ) {
if ( removeStatus != REMOVE_STATUS_ALIVE ) {
return;
}
if ( idSoundSystemLocal::s_showStartSound.GetInteger() ) {
common->Printf( "FreeSound (%i,%i)\n", index, (int)immediate );
}
if ( soundWorld && soundWorld->writeDemo ) {
soundWorld->writeDemo->WriteInt( DS_SOUND );
soundWorld->writeDemo->WriteInt( SCMD_FREE );
soundWorld->writeDemo->WriteInt( index );
soundWorld->writeDemo->WriteInt( immediate );
}
if ( !immediate ) {
removeStatus = REMOVE_STATUS_WAITSAMPLEFINISHED;
} else {
Clear();
}
}
/*
=====================
idSoundEmitterLocal::StartSound
returns the length of the started sound in msec
=====================
*/
int idSoundEmitterLocal::StartSound( const idSoundShader *shader, const s_channelType channel, float diversity, int soundShaderFlags, bool allowSlow ) {
int i;
if ( !shader ) {
return 0;
}
if ( idSoundSystemLocal::s_showStartSound.GetInteger() ) {
common->Printf( "StartSound %ims (%i,%i,%s) = ", soundWorld->gameMsec, index, (int)channel, shader->GetName() );
}
if ( soundWorld && soundWorld->writeDemo ) {
soundWorld->writeDemo->WriteInt( DS_SOUND );
soundWorld->writeDemo->WriteInt( SCMD_START );
soundWorld->writeDemo->WriteInt( index );
soundWorld->writeDemo->WriteHashString( shader->GetName() );
soundWorld->writeDemo->WriteInt( channel );
soundWorld->writeDemo->WriteFloat( diversity );
soundWorld->writeDemo->WriteInt( soundShaderFlags );
}
// build the channel parameters by taking the shader parms and optionally overriding
soundShaderParms_t chanParms;
chanParms = shader->parms;
OverrideParms( &chanParms, &this->parms, &chanParms );
chanParms.soundShaderFlags |= soundShaderFlags;
if ( chanParms.shakes > 0.0f ) {
shader->CheckShakesAndOgg();
}
// this is the sample time it will be first mixed
int start44kHz;
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if ( soundWorld->fpa[0] ) {
// if we are recording an AVI demo, don't use hardware time
start44kHz = soundWorld->lastAVI44kHz + MIXBUFFER_SAMPLES;
} else {
start44kHz = soundSystemLocal.GetCurrent44kHzTime() + MIXBUFFER_SAMPLES;
}
//
// pick which sound to play from the shader
//
if ( !shader->numEntries ) {
if ( idSoundSystemLocal::s_showStartSound.GetInteger() ) {
common->Printf( "no samples in sound shader\n" );
}
return 0; // no sounds
}
int choice;
// pick a sound from the list based on the passed diversity
choice = (int)(diversity * shader->numEntries);
if ( choice < 0 || choice >= shader->numEntries ) {
choice = 0;
}
// bump the choice if the exact sound was just played and we are NO_DUPS
if ( chanParms.soundShaderFlags & SSF_NO_DUPS ) {
idSoundSample *sample;
if ( shader->leadins[ choice ] ) {
sample = shader->leadins[ choice ];
} else {
sample = shader->entries[ choice ];
}
for( i = 0; i < SOUND_MAX_CHANNELS; i++ ) {
idSoundChannel *chan = &channels[i];
if ( chan->leadinSample == sample ) {
choice = ( choice + 1 ) % shader->numEntries;
break;
}
}
}
// PLAY_ONCE sounds will never be restarted while they are running
if ( chanParms.soundShaderFlags & SSF_PLAY_ONCE ) {
for( i = 0; i < SOUND_MAX_CHANNELS; i++ ) {
idSoundChannel *chan = &channels[i];
if ( chan->triggerState && chan->soundShader == shader ) {
if ( idSoundSystemLocal::s_showStartSound.GetInteger() ) {
common->Printf( "PLAY_ONCE not restarting\n" );
}
return 0;
}
}
}
// never play the same sound twice with the same starting time, even
// if they are on different channels
for( i = 0; i < SOUND_MAX_CHANNELS; i++ ) {
idSoundChannel *chan = &channels[i];
if ( chan->triggerState && chan->soundShader == shader && chan->trigger44kHzTime == start44kHz ) {
if ( idSoundSystemLocal::s_showStartSound.GetInteger() ) {
common->Printf( "already started this frame\n" );
}
return 0;
}
}
Sys_EnterCriticalSection();
// kill any sound that is currently playing on this channel
if ( channel != SCHANNEL_ANY ) {
for( i = 0; i < SOUND_MAX_CHANNELS; i++ ) {
idSoundChannel *chan = &channels[i];
if ( chan->triggerState && chan->soundShader && chan->triggerChannel == channel ) {
if ( idSoundSystemLocal::s_showStartSound.GetInteger() ) {
common->Printf( "(override %s)", chan->soundShader->base->GetName() );
}
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chan->Stop();
// if this was an onDemand sound, purge the sample now
if ( chan->leadinSample->onDemand ) {
chan->ALStop();
chan->leadinSample->PurgeSoundSample();
}
break;
}
}
}
// find a free channel to play the sound on
idSoundChannel *chan;
for( i = 0; i < SOUND_MAX_CHANNELS; i++ ) {
chan = &channels[i];
if ( !chan->triggerState ) {
break;
}
}
if ( i == SOUND_MAX_CHANNELS ) {
// we couldn't find a channel for it
Sys_LeaveCriticalSection();
if ( idSoundSystemLocal::s_showStartSound.GetInteger() ) {
common->Printf( "no channels available\n" );
}
return 0;
}
chan = &channels[i];
if ( shader->leadins[ choice ] ) {
chan->leadinSample = shader->leadins[ choice ];
} else {
chan->leadinSample = shader->entries[ choice ];
}
// if the sample is onDemand (voice mails, etc), load it now
if ( chan->leadinSample->purged ) {
int start = Sys_Milliseconds();
chan->leadinSample->Load();
int end = Sys_Milliseconds();
session->TimeHitch( end - start );
// recalculate start44kHz, because loading may have taken a fair amount of time
if ( !soundWorld->fpa[0] ) {
start44kHz = soundSystemLocal.GetCurrent44kHzTime() + MIXBUFFER_SAMPLES;
}
}
if ( idSoundSystemLocal::s_showStartSound.GetInteger() ) {
common->Printf( "'%s'\n", chan->leadinSample->name.c_str() );
}
if ( idSoundSystemLocal::s_skipHelltimeFX.GetBool() ) {
chan->disallowSlow = true;
} else {
chan->disallowSlow = !allowSlow;
}
ResetSlowChannel( chan );
// the sound will start mixing in the next async mix block
chan->triggered = true;
chan->openalStreamingOffset = 0;
chan->trigger44kHzTime = start44kHz;
chan->parms = chanParms;
chan->triggerGame44kHzTime = soundWorld->game44kHz;
chan->soundShader = shader;
chan->triggerChannel = channel;
chan->stopped = false;
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chan->Start();
// we need to start updating the def and mixing it in
playing = true;
// spatialize it immediately, so it will start the next mix block
// even if that happens before the next PlaceOrigin()
Spatialize( soundWorld->listenerPos, soundWorld->listenerArea, soundWorld->rw );
// return length of sound in milliseconds
int length = chan->leadinSample->LengthIn44kHzSamples();
if ( chan->leadinSample->objectInfo.nChannels == 2 ) {
length /= 2; // stereo samples
}
// adjust the start time based on diversity for looping sounds, so they don't all start
// at the same point
if ( chan->parms.soundShaderFlags & SSF_LOOPING && !chan->leadinSample->LengthIn44kHzSamples() ) {
chan->trigger44kHzTime -= diversity * length;
chan->trigger44kHzTime &= ~7; // so we don't have to worry about the 22kHz and 11kHz expansions
// starting in fractional samples
chan->triggerGame44kHzTime -= diversity * length;
chan->triggerGame44kHzTime &= ~7;
}
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length *= 1000 / (float)PRIMARYFREQ;
Sys_LeaveCriticalSection();
return length;
}
/*
===================
idSoundEmitterLocal::ModifySound
===================
*/
void idSoundEmitterLocal::ModifySound( const s_channelType channel, const soundShaderParms_t *parms ) {
if ( !parms ) {
common->Error( "idSoundEmitterLocal::ModifySound: NULL parms" );
}
if ( idSoundSystemLocal::s_showStartSound.GetInteger() ) {
common->Printf( "ModifySound(%i,%i)\n", index, channel );
}
if ( soundWorld && soundWorld->writeDemo ) {
soundWorld->writeDemo->WriteInt( DS_SOUND );
soundWorld->writeDemo->WriteInt( SCMD_MODIFY );
soundWorld->writeDemo->WriteInt( index );
soundWorld->writeDemo->WriteInt( channel );
soundWorld->writeDemo->WriteFloat( parms->minDistance );
soundWorld->writeDemo->WriteFloat( parms->maxDistance );
soundWorld->writeDemo->WriteFloat( parms->volume );
soundWorld->writeDemo->WriteFloat( parms->shakes );
soundWorld->writeDemo->WriteInt( parms->soundShaderFlags );
soundWorld->writeDemo->WriteInt( parms->soundClass );
}
for ( int i = 0; i < SOUND_MAX_CHANNELS; i++ ) {
idSoundChannel *chan = &channels[i];
if ( !chan->triggerState ) {
continue;
}
if ( channel != SCHANNEL_ANY && chan->triggerChannel != channel ) {
continue;
}
OverrideParms( &chan->parms, parms, &chan->parms );
if ( chan->parms.shakes > 0.0f && chan->soundShader != NULL ) {
chan->soundShader->CheckShakesAndOgg();
}
}
}
/*
===================
idSoundEmitterLocal::StopSound
can pass SCHANNEL_ANY
===================
*/
void idSoundEmitterLocal::StopSound( const s_channelType channel ) {
int i;
if ( idSoundSystemLocal::s_showStartSound.GetInteger() ) {
common->Printf( "StopSound(%i,%i)\n", index, channel );
}
if ( soundWorld && soundWorld->writeDemo ) {
soundWorld->writeDemo->WriteInt( DS_SOUND );
soundWorld->writeDemo->WriteInt( SCMD_STOP );
soundWorld->writeDemo->WriteInt( index );
soundWorld->writeDemo->WriteInt( channel );
}
Sys_EnterCriticalSection();
for( i = 0; i < SOUND_MAX_CHANNELS; i++ ) {
idSoundChannel *chan = &channels[i];
if ( !chan->triggerState ) {
continue;
}
if ( channel != SCHANNEL_ANY && chan->triggerChannel != channel ) {
continue;
}
// stop it
chan->Stop();
// free hardware resources
chan->ALStop();
// if this was an onDemand sound, purge the sample now
if ( chan->leadinSample->onDemand ) {
chan->leadinSample->PurgeSoundSample();
}
chan->leadinSample = NULL;
chan->soundShader = NULL;
}
Sys_LeaveCriticalSection();
}
// DG: to pause active OpenAL sources when entering menu etc
void idSoundEmitterLocal::PauseAll( void ) {
Sys_EnterCriticalSection();
for( int i = 0; i < SOUND_MAX_CHANNELS; i++ ) {
idSoundChannel *chan = &channels[i];
if ( !chan->triggerState ) {
continue;
}
if ( alIsSource( chan->openalSource ) ) {
alSourcePause( chan->openalSource );
chan->paused = true;
}
}
Sys_LeaveCriticalSection();
}
// DG: to resume active OpenAL sources when leaving menu etc
void idSoundEmitterLocal::UnPauseAll( void ) {
Sys_EnterCriticalSection();
for( int i = 0; i < SOUND_MAX_CHANNELS; i++ ) {
idSoundChannel *chan = &channels[i];
if ( !chan->triggerState ) {
continue;
}
if ( alIsSource( chan->openalSource ) && chan->paused ) {
alSourcePlay( chan->openalSource );
chan->paused = false;
}
}
Sys_LeaveCriticalSection();
}
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/*
===================
idSoundEmitterLocal::FadeSound
to is in Db (sigh), over is in seconds
===================
*/
void idSoundEmitterLocal::FadeSound( const s_channelType channel, float to, float over ) {
if ( idSoundSystemLocal::s_showStartSound.GetInteger() ) {
common->Printf( "FadeSound(%i,%i,%f,%f )\n", index, channel, to, over );
}
if ( !soundWorld ) {
return;
}
if ( soundWorld->writeDemo ) {
soundWorld->writeDemo->WriteInt( DS_SOUND );
soundWorld->writeDemo->WriteInt( SCMD_FADE );
soundWorld->writeDemo->WriteInt( index );
soundWorld->writeDemo->WriteInt( channel );
soundWorld->writeDemo->WriteFloat( to );
soundWorld->writeDemo->WriteFloat( over );
}
int start44kHz;
if ( soundWorld->fpa[0] ) {
// if we are recording an AVI demo, don't use hardware time
start44kHz = soundWorld->lastAVI44kHz + MIXBUFFER_SAMPLES;
} else {
start44kHz = soundSystemLocal.GetCurrent44kHzTime() + MIXBUFFER_SAMPLES;
}
int length44kHz = soundSystemLocal.MillisecondsToSamples( over * 1000 );
for( int i = 0; i < SOUND_MAX_CHANNELS ; i++ ) {
idSoundChannel *chan = &channels[i];
if ( !chan->triggerState ) {
continue;
}
if ( channel != SCHANNEL_ANY && chan->triggerChannel != channel ) {
continue;
}
// if it is already fading to this volume at this rate, don't change it
if ( chan->channelFade.fadeEndVolume == to &&
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chan->channelFade.fadeEnd44kHz - chan->channelFade.fadeStart44kHz == length44kHz ) {
continue;
}
// fade it
chan->channelFade.fadeStartVolume = chan->channelFade.FadeDbAt44kHz( start44kHz );
chan->channelFade.fadeStart44kHz = start44kHz;
chan->channelFade.fadeEnd44kHz = start44kHz + length44kHz;
chan->channelFade.fadeEndVolume = to;
}
}
/*
===================
idSoundEmitterLocal::CurrentlyPlaying
===================
*/
bool idSoundEmitterLocal::CurrentlyPlaying( void ) const {
return playing;
}
/*
===================
idSoundEmitterLocal::Index
===================
*/
int idSoundEmitterLocal::Index( void ) const {
return index;
}
/*
===================
idSoundEmitterLocal::CurrentAmplitude
this is called from the main thread by the material shader system
to allow lights and surface flares to vary with the sound amplitude
===================
*/
float idSoundEmitterLocal::CurrentAmplitude( void ) {
if ( idSoundSystemLocal::s_constantAmplitude.GetFloat() >= 0.0f ) {
return idSoundSystemLocal::s_constantAmplitude.GetFloat();
}
if ( removeStatus > REMOVE_STATUS_WAITSAMPLEFINISHED ) {
return 0.0;
}
int localTime = soundSystemLocal.GetCurrent44kHzTime();
// see if we can use our cached value
if ( ampTime == localTime ) {
return amplitude;
}
// calculate a new value
ampTime = localTime;
amplitude = soundWorld->FindAmplitude( this, localTime, NULL, SCHANNEL_ANY, false );
return amplitude;
}
/*
===================
idSoundEmitterLocal::GetSlowChannel
===================
*/
idSlowChannel idSoundEmitterLocal::GetSlowChannel( const idSoundChannel *chan ) {
return slowChannels[chan - channels];
}
/*
===================
idSoundEmitterLocal::SetSlowChannel
===================
*/
void idSoundEmitterLocal::SetSlowChannel( const idSoundChannel *chan, idSlowChannel slow ) {
slowChannels[chan - channels] = slow;
}
/*
===================
idSoundEmitterLocal::ResetSlowChannel
===================
*/
void idSoundEmitterLocal::ResetSlowChannel( const idSoundChannel *chan ) {
int index = chan - channels;
slowChannels[index].Reset();
}
/*
===================
idSlowChannel::Reset
===================
*/
void idSlowChannel::Reset() {
memset( this, 0, sizeof( *this ) );
this->chan = chan;
curPosition.Set( 0 );
newPosition.Set( 0 );
curSampleOffset = -10000;
newSampleOffset = -10000;
triggerOffset = 0;
}
/*
===================
idSlowChannel::AttachSoundChannel
===================
*/
void idSlowChannel::AttachSoundChannel( const idSoundChannel *chan ) {
this->chan = chan;
}
/*
===================
idSlowChannel::GetSlowmoSpeed
===================
*/
float idSlowChannel::GetSlowmoSpeed() {
idSoundWorldLocal *sw = static_cast<idSoundWorldLocal*>( soundSystemLocal.GetPlayingSoundWorld() );
if ( sw ) {
return sw->slowmoSpeed;
} else {
return 0;
}
}
/*
===================
idSlowChannel::GenerateSlowChannel
===================
*/
void idSlowChannel::GenerateSlowChannel( FracTime& playPos, int sampleCount44k, float* finalBuffer ) {
idSoundWorldLocal *sw = static_cast<idSoundWorldLocal*>( soundSystemLocal.GetPlayingSoundWorld() );
float in[MIXBUFFER_SAMPLES+3], out[MIXBUFFER_SAMPLES+3], *src, *spline, slowmoSpeed;
int i, neededSamples, zeroedPos, count = 0;
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src = in + 2;
spline = out + 2;
if ( sw ) {
slowmoSpeed = sw->slowmoSpeed;
}
else {
slowmoSpeed = 1;
}
neededSamples = sampleCount44k * slowmoSpeed + 4;
// get the channel's samples
chan->GatherChannelSamples( playPos.time * 2, neededSamples, src );
for ( i = 0; i < neededSamples >> 1; i++ ) {
spline[i] = src[i*2];
}
// interpolate channel
zeroedPos = playPos.time;
playPos.time = 0;
for ( i = 0; i < sampleCount44k >> 1; i++, count += 2 ) {
float val;
val = spline[playPos.time];
src[i] = val;
playPos.Increment( slowmoSpeed );
}
// lowpass filter
float *in_p = in + 2, *out_p = out + 2;
int numSamples = sampleCount44k >> 1;
lowpass.GetContinuitySamples( in_p[-1], in_p[-2], out_p[-1], out_p[-2] );
lowpass.SetParms( slowmoSpeed * 15000, 1.2f );
for ( int i = 0, count = 0; i < numSamples; i++, count += 2 ) {
lowpass.ProcessSample( in_p + i, out_p + i );
finalBuffer[count] = finalBuffer[count+1] = out[i];
}
lowpass.SetContinuitySamples( in_p[numSamples-2], in_p[numSamples-3], out_p[numSamples-2], out_p[numSamples-3] );
playPos.time += zeroedPos;
}
/*
===================
idSlowChannel::GatherChannelSamples
===================
*/
void idSlowChannel::GatherChannelSamples( int sampleOffset44k, int sampleCount44k, float *dest ) {
int state = 0;
// setup chan
active = true;
newSampleOffset = sampleOffset44k >> 1;
// set state
if ( newSampleOffset < curSampleOffset ) {
state = PLAYBACK_RESET;
} else if ( newSampleOffset > curSampleOffset ) {
state = PLAYBACK_ADVANCING;
}
if ( state == PLAYBACK_RESET ) {
curPosition.Set( newSampleOffset );
}
// set current vars
curSampleOffset = newSampleOffset;
newPosition = curPosition;
// do the slow processing
GenerateSlowChannel( newPosition, sampleCount44k, dest );
// finish off
if ( state == PLAYBACK_ADVANCING )
curPosition = newPosition;
}