mirror of
https://github.com/id-Software/DOOM-3-BFG.git
synced 2024-12-15 15:01:19 +00:00
763 lines
19 KiB
C++
763 lines
19 KiB
C++
/*
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===========================================================================
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Doom 3 BFG Edition GPL Source Code
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Copyright (C) 1993-2012 id Software LLC, a ZeniMax Media company.
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Copyright (C) 2013 Robert Beckebans
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This file is part of the Doom 3 BFG Edition GPL Source Code ("Doom 3 BFG Edition Source Code").
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Doom 3 BFG Edition Source Code is free software: you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation, either version 3 of the License, or
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(at your option) any later version.
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Doom 3 BFG Edition Source Code is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with Doom 3 BFG Edition Source Code. If not, see <http://www.gnu.org/licenses/>.
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In addition, the Doom 3 BFG Edition 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 BFG Edition Source Code. If not, please request a copy in writing from id Software at the address below.
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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.
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===========================================================================
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*/
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#pragma hdrstop
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#include "precompiled.h"
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#include "../snd_local.h"
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idCVar s_skipHardwareSets( "s_skipHardwareSets", "0", CVAR_BOOL, "Do all calculation, but skip XA2 calls" );
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idCVar s_debugHardware( "s_debugHardware", "0", CVAR_BOOL, "Print a message any time a hardware voice changes" );
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// The whole system runs at this sample rate
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static int SYSTEM_SAMPLE_RATE = 44100;
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static float ONE_OVER_SYSTEM_SAMPLE_RATE = 1.0f / SYSTEM_SAMPLE_RATE;
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/*
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========================
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idSoundVoice_OpenAL::idSoundVoice_OpenAL
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========================
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*/
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idSoundVoice_OpenAL::idSoundVoice_OpenAL()
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:
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triggered( false ),
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openalSource( 0 ),
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leadinSample( NULL ),
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loopingSample( NULL ),
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formatTag( 0 ),
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numChannels( 0 ),
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sampleRate( 0 ),
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paused( true ),
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hasVUMeter( false )
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{
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}
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/*
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========================
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idSoundVoice_OpenAL::~idSoundVoice_OpenAL
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========================
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*/
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idSoundVoice_OpenAL::~idSoundVoice_OpenAL()
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{
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DestroyInternal();
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}
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/*
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========================
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idSoundVoice_OpenAL::CompatibleFormat
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========================
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*/
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bool idSoundVoice_OpenAL::CompatibleFormat( idSoundSample_OpenAL* s )
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{
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if( alIsSource( openalSource ) )
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{
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// If this voice has never been allocated, then it's compatible with everything
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return true;
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}
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return false;
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}
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/*
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========================
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idSoundVoice_OpenAL::Create
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========================
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*/
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void idSoundVoice_OpenAL::Create( const idSoundSample* leadinSample_, const idSoundSample* loopingSample_ )
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{
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if( IsPlaying() )
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{
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// This should never hit
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Stop();
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return;
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}
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triggered = true;
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leadinSample = ( idSoundSample_OpenAL* )leadinSample_;
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loopingSample = ( idSoundSample_OpenAL* )loopingSample_;
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if( alIsSource( openalSource ) && CompatibleFormat( leadinSample ) )
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{
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sampleRate = leadinSample->format.basic.samplesPerSec;
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}
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else
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{
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DestroyInternal();
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formatTag = leadinSample->format.basic.formatTag;
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numChannels = leadinSample->format.basic.numChannels;
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sampleRate = leadinSample->format.basic.samplesPerSec;
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//soundSystemLocal.hardware.pXAudio2->CreateSourceVoice( &pSourceVoice, ( const WAVEFORMATEX* )&leadinSample->format, XAUDIO2_VOICE_USEFILTER, 4.0f, &streamContext );
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CheckALErrors();
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alGenSources( 1, &openalSource );
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if( CheckALErrors() != AL_NO_ERROR )
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//if( pSourceVoice == NULL )
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{
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// If this hits, then we are most likely passing an invalid sample format, which should have been caught by the loader (and the sample defaulted)
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return;
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}
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alSourcef( openalSource, AL_ROLLOFF_FACTOR, 0.0f );
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//if( ( loopingSample == NULL && leadinSample->openalBuffer != 0 ) || ( loopingSample != NULL && soundShader->entries[0]->hardwareBuffer ) )
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if( leadinSample->openalBuffer != 0 )
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{
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alSourcei( openalSource, AL_BUFFER, 0 );
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// handle uncompressed (non streaming) single shot and looping sounds
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/*
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if( triggered )
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{
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alSourcei( openalSource, AL_BUFFER, looping ? chan->soundShader->entries[0]->openalBuffer : leadinSample->openalBuffer );
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}
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*/
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}
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else
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{
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//if( triggered )
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// handle streaming sounds (decode on the fly) both single shot AND looping
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alSourcei( openalSource, AL_BUFFER, 0 );
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alDeleteBuffers( 3, &lastopenalStreamingBuffer[0] );
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lastopenalStreamingBuffer[0] = openalStreamingBuffer[0];
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lastopenalStreamingBuffer[1] = openalStreamingBuffer[1];
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lastopenalStreamingBuffer[2] = openalStreamingBuffer[2];
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alGenBuffers( 3, &openalStreamingBuffer[0] );
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/*
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if( soundSystemLocal.alEAXSetBufferMode )
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{
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soundSystemLocal.alEAXSetBufferMode( 3, &chan->openalStreamingBuffer[0], alGetEnumValue( ID_ALCHAR "AL_STORAGE_ACCESSIBLE" ) );
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}
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*/
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openalStreamingBuffer[0];
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openalStreamingBuffer[1];
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openalStreamingBuffer[2];
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}
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if( s_debugHardware.GetBool() )
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{
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if( loopingSample == NULL || loopingSample == leadinSample )
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{
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idLib::Printf( "%dms: %i created for %s\n", Sys_Milliseconds(), openalSource, leadinSample ? leadinSample->GetName() : "<null>" );
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}
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else
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{
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idLib::Printf( "%dms: %i created for %s and %s\n", Sys_Milliseconds(), openalSource, leadinSample ? leadinSample->GetName() : "<null>", loopingSample ? loopingSample->GetName() : "<null>" );
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}
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}
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}
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sourceVoiceRate = sampleRate;
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//pSourceVoice->SetSourceSampleRate( sampleRate );
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//pSourceVoice->SetVolume( 0.0f );
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alSourcei( openalSource, AL_SOURCE_RELATIVE, AL_TRUE );
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alSource3f( openalSource, AL_POSITION, 0.0f, 0.0f, 0.0f );
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// RB: FIXME 0.0f ?
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alSourcef( openalSource, AL_GAIN, 1.0f );
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//OnBufferStart( leadinSample, 0 );
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}
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/*
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========================
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idSoundVoice_OpenAL::DestroyInternal
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========================
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*/
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void idSoundVoice_OpenAL::DestroyInternal()
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{
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if( alIsSource( openalSource ) )
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{
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if( s_debugHardware.GetBool() )
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{
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idLib::Printf( "%dms: %i destroyed\n", Sys_Milliseconds(), openalSource );
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}
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alDeleteSources( 1, &openalSource );
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openalSource = 0;
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alSourcei( openalSource, AL_BUFFER, 0 );
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if( openalStreamingBuffer[0] && openalStreamingBuffer[1] && openalStreamingBuffer[2] )
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{
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CheckALErrors();
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alDeleteBuffers( 3, &openalStreamingBuffer[0] );
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if( CheckALErrors() == AL_NO_ERROR )
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{
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openalStreamingBuffer[0] = openalStreamingBuffer[1] = openalStreamingBuffer[2] = 0;
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}
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}
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if( lastopenalStreamingBuffer[0] && lastopenalStreamingBuffer[1] && lastopenalStreamingBuffer[2] )
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{
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CheckALErrors();
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alDeleteBuffers( 3, &lastopenalStreamingBuffer[0] );
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if( CheckALErrors() == AL_NO_ERROR )
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{
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lastopenalStreamingBuffer[0] = lastopenalStreamingBuffer[1] = lastopenalStreamingBuffer[2] = 0;
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}
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}
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openalStreamingOffset = 0;
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hasVUMeter = false;
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}
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}
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/*
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========================
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idSoundVoice_OpenAL::Start
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========================
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*/
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void idSoundVoice_OpenAL::Start( int offsetMS, int ssFlags )
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{
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if( s_debugHardware.GetBool() )
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{
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idLib::Printf( "%dms: %i starting %s @ %dms\n", Sys_Milliseconds(), openalSource, leadinSample ? leadinSample->GetName() : "<null>", offsetMS );
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}
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if( !leadinSample )
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{
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return;
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}
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if( !alIsSource( openalSource ) )
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{
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return;
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}
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if( leadinSample->IsDefault() )
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{
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idLib::Warning( "Starting defaulted sound sample %s", leadinSample->GetName() );
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}
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bool flicker = ( ssFlags & SSF_NO_FLICKER ) == 0;
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if( flicker != hasVUMeter )
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{
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hasVUMeter = flicker;
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/*
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if( flicker )
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{
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IUnknown* vuMeter = NULL;
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if( XAudio2CreateVolumeMeter( &vuMeter, 0 ) == S_OK )
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{
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XAUDIO2_EFFECT_DESCRIPTOR descriptor;
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descriptor.InitialState = true;
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descriptor.OutputChannels = leadinSample->NumChannels();
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descriptor.pEffect = vuMeter;
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XAUDIO2_EFFECT_CHAIN chain;
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chain.EffectCount = 1;
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chain.pEffectDescriptors = &descriptor;
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pSourceVoice->SetEffectChain( &chain );
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vuMeter->Release();
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}
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}
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else
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{
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pSourceVoice->SetEffectChain( NULL );
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}
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*/
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}
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assert( offsetMS >= 0 );
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int offsetSamples = MsecToSamples( offsetMS, leadinSample->SampleRate() );
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if( loopingSample == NULL && offsetSamples >= leadinSample->playLength )
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{
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return;
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}
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RestartAt( offsetSamples );
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Update();
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UnPause();
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}
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/*
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========================
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idSoundVoice_OpenAL::RestartAt
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========================
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*/
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int idSoundVoice_OpenAL::RestartAt( int offsetSamples )
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{
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offsetSamples &= ~127;
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idSoundSample_OpenAL* sample = leadinSample;
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if( offsetSamples >= leadinSample->playLength )
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{
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if( loopingSample != NULL )
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{
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offsetSamples %= loopingSample->playLength;
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sample = loopingSample;
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}
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else
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{
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return 0;
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}
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}
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int previousNumSamples = 0;
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for( int i = 0; i < sample->buffers.Num(); i++ )
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{
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if( sample->buffers[i].numSamples > sample->playBegin + offsetSamples )
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{
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return SubmitBuffer( sample, i, sample->playBegin + offsetSamples - previousNumSamples );
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}
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previousNumSamples = sample->buffers[i].numSamples;
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}
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return 0;
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}
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/*
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========================
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idSoundVoice_OpenAL::SubmitBuffer
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========================
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*/
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int idSoundVoice_OpenAL::SubmitBuffer( idSoundSample_OpenAL* sample, int bufferNumber, int offset )
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{
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if( sample == NULL || ( bufferNumber < 0 ) || ( bufferNumber >= sample->buffers.Num() ) )
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{
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return 0;
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}
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#if 0
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idSoundSystemLocal::bufferContext_t* bufferContext = soundSystemLocal.ObtainStreamBufferContext();
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if( bufferContext == NULL )
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{
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idLib::Warning( "No free buffer contexts!" );
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return 0;
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}
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bufferContext->voice = this;
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bufferContext->sample = sample;
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bufferContext->bufferNumber = bufferNumber;
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#endif
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if( sample->openalBuffer != 0 )
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{
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alSourcei( openalSource, AL_BUFFER, sample->openalBuffer );
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alSourcei( openalSource, AL_LOOPING, ( sample == loopingSample && loopingSample != NULL ? AL_TRUE : AL_FALSE ) );
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return sample->totalBufferSize;
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}
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else
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{
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ALint finishedbuffers;
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if( !triggered )
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{
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alGetSourcei( openalSource, AL_BUFFERS_PROCESSED, &finishedbuffers );
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alSourceUnqueueBuffers( openalSource, finishedbuffers, &openalStreamingBuffer[0] );
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if( finishedbuffers == 3 )
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{
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triggered = true;
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}
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}
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else
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{
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finishedbuffers = 3;
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}
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ALenum format;
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if( sample->format.basic.formatTag == idWaveFile::FORMAT_PCM )
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{
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format = sample->NumChannels() == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
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}
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else if( sample->format.basic.formatTag == idWaveFile::FORMAT_ADPCM )
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{
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format = sample->NumChannels() == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
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}
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else if( sample->format.basic.formatTag == idWaveFile::FORMAT_XMA2 )
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{
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format = sample->NumChannels() == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
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}
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else
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{
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format = sample->NumChannels() == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
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}
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int rate = sample->SampleRate(); /*44100*/
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for( int j = 0; j < finishedbuffers && j < 1; j++ )
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{
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/*
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chan->GatherChannelSamples( chan->openalStreamingOffset * sample->objectInfo.nChannels, MIXBUFFER_SAMPLES * sample->objectInfo.nChannels, alignedInputSamples );
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for( int i = 0; i < ( MIXBUFFER_SAMPLES * sample->objectInfo.nChannels ); i++ )
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{
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if( alignedInputSamples[i] < -32768.0f )
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( ( short* )alignedInputSamples )[i] = -32768;
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else if( alignedInputSamples[i] > 32767.0f )
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( ( short* )alignedInputSamples )[i] = 32767;
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else
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( ( short* )alignedInputSamples )[i] = idMath::FtoiFast( alignedInputSamples[i] );
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}
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*/
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//alBufferData( buffers[0], sample->NumChannels() == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16, sample->buffers[bufferNumber].buffer, sample->buffers[bufferNumber].bufferSize, sample->SampleRate() /*44100*/ );
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alBufferData( openalStreamingBuffer[j], format, sample->buffers[bufferNumber].buffer, sample->buffers[bufferNumber].bufferSize, rate );
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//openalStreamingOffset += MIXBUFFER_SAMPLES;
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}
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if( finishedbuffers > 0 )
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{
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//alSourceQueueBuffers( openalSource, finishedbuffers, &buffers[0] );
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alSourceQueueBuffers( openalSource, 1, &openalStreamingBuffer[0] );
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if( bufferNumber == 0 )
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{
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//alSourcePlay( openalSource );
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triggered = false;
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}
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return sample->buffers[bufferNumber].bufferSize;
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}
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}
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// should never happen
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return 0;
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/*
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XAUDIO2_BUFFER buffer = { 0 };
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if( offset > 0 )
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{
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int previousNumSamples = 0;
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if( bufferNumber > 0 )
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{
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previousNumSamples = sample->buffers[bufferNumber - 1].numSamples;
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}
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buffer.PlayBegin = offset;
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buffer.PlayLength = sample->buffers[bufferNumber].numSamples - previousNumSamples - offset;
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}
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buffer.AudioBytes = sample->buffers[bufferNumber].bufferSize;
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buffer.pAudioData = ( BYTE* )sample->buffers[bufferNumber].buffer;
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buffer.pContext = bufferContext;
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if( ( loopingSample == NULL ) && ( bufferNumber == sample->buffers.Num() - 1 ) )
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{
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buffer.Flags = XAUDIO2_END_OF_STREAM;
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}
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pSourceVoice->SubmitSourceBuffer( &buffer );
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return buffer.AudioBytes;
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*/
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}
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/*
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========================
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idSoundVoice_OpenAL::Update
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========================
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*/
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bool idSoundVoice_OpenAL::Update()
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{
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/*
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if( pSourceVoice == NULL || leadinSample == NULL )
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{
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return false;
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}
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XAUDIO2_VOICE_STATE state;
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pSourceVoice->GetState( &state );
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const int srcChannels = leadinSample->NumChannels();
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float pLevelMatrix[ MAX_CHANNELS_PER_VOICE * MAX_CHANNELS_PER_VOICE ] = { 0 };
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CalculateSurround( srcChannels, pLevelMatrix, 1.0f );
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if( s_skipHardwareSets.GetBool() )
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{
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return true;
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}
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pSourceVoice->SetOutputMatrix( soundSystemLocal.hardware.pMasterVoice, srcChannels, dstChannels, pLevelMatrix, OPERATION_SET );
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assert( idMath::Fabs( gain ) <= XAUDIO2_MAX_VOLUME_LEVEL );
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pSourceVoice->SetVolume( gain, OPERATION_SET );
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SetSampleRate( sampleRate, OPERATION_SET );
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// we don't do this any longer because we pause and unpause explicitly when the soundworld is paused or unpaused
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// UnPause();
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*/
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return true;
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}
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/*
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========================
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idSoundVoice_OpenAL::IsPlaying
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========================
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*/
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bool idSoundVoice_OpenAL::IsPlaying()
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{
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if( !alIsSource( openalSource ) )
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{
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return false;
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}
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ALint state = AL_INITIAL;
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alGetSourcei( openalSource, AL_SOURCE_STATE, &state );
|
|
|
|
return ( state == AL_PLAYING );
|
|
|
|
//XAUDIO2_VOICE_STATE state;
|
|
//pSourceVoice->GetState( &state );
|
|
|
|
//return ( state.BuffersQueued != 0 );
|
|
}
|
|
|
|
/*
|
|
========================
|
|
idSoundVoice_OpenAL::FlushSourceBuffers
|
|
========================
|
|
*/
|
|
void idSoundVoice_OpenAL::FlushSourceBuffers()
|
|
{
|
|
if( alIsSource( openalSource ) )
|
|
{
|
|
//pSourceVoice->FlushSourceBuffers();
|
|
}
|
|
}
|
|
|
|
/*
|
|
========================
|
|
idSoundVoice_OpenAL::Pause
|
|
========================
|
|
*/
|
|
void idSoundVoice_OpenAL::Pause()
|
|
{
|
|
if( !alIsSource( openalSource ) || paused )
|
|
{
|
|
return;
|
|
}
|
|
|
|
if( s_debugHardware.GetBool() )
|
|
{
|
|
idLib::Printf( "%dms: %i pausing %s\n", Sys_Milliseconds(), openalSource, leadinSample ? leadinSample->GetName() : "<null>" );
|
|
}
|
|
|
|
alSourcePause( openalSource );
|
|
//pSourceVoice->Stop( 0, OPERATION_SET );
|
|
paused = true;
|
|
}
|
|
|
|
/*
|
|
========================
|
|
idSoundVoice_OpenAL::UnPause
|
|
========================
|
|
*/
|
|
void idSoundVoice_OpenAL::UnPause()
|
|
{
|
|
if( !alIsSource( openalSource ) || !paused )
|
|
{
|
|
return;
|
|
}
|
|
|
|
if( s_debugHardware.GetBool() )
|
|
{
|
|
idLib::Printf( "%dms: %i unpausing %s\n", Sys_Milliseconds(), openalSource, leadinSample ? leadinSample->GetName() : "<null>" );
|
|
}
|
|
|
|
alSourcePlay( openalSource );
|
|
//pSourceVoice->Start( 0, OPERATION_SET );
|
|
paused = false;
|
|
}
|
|
|
|
/*
|
|
========================
|
|
idSoundVoice_OpenAL::Stop
|
|
========================
|
|
*/
|
|
void idSoundVoice_OpenAL::Stop()
|
|
{
|
|
if( !alIsSource( openalSource ) )
|
|
{
|
|
return;
|
|
}
|
|
|
|
if( !paused )
|
|
{
|
|
if( s_debugHardware.GetBool() )
|
|
{
|
|
idLib::Printf( "%dms: %i stopping %s\n", Sys_Milliseconds(), openalSource, leadinSample ? leadinSample->GetName() : "<null>" );
|
|
}
|
|
|
|
alSourceStop( openalSource );
|
|
alSourcei( openalSource, AL_BUFFER, 0 );
|
|
|
|
//pSourceVoice->Stop( 0, OPERATION_SET );
|
|
paused = true;
|
|
}
|
|
}
|
|
|
|
/*
|
|
========================
|
|
idSoundVoice_OpenAL::GetAmplitude
|
|
========================
|
|
*/
|
|
float idSoundVoice_OpenAL::GetAmplitude()
|
|
{
|
|
// TODO
|
|
return 1.0f;
|
|
|
|
/*
|
|
if( !hasVUMeter )
|
|
{
|
|
return 1.0f;
|
|
}
|
|
|
|
float peakLevels[ MAX_CHANNELS_PER_VOICE ];
|
|
float rmsLevels[ MAX_CHANNELS_PER_VOICE ];
|
|
|
|
XAUDIO2FX_VOLUMEMETER_LEVELS levels;
|
|
levels.ChannelCount = leadinSample->NumChannels();
|
|
levels.pPeakLevels = peakLevels;
|
|
levels.pRMSLevels = rmsLevels;
|
|
|
|
if( levels.ChannelCount > MAX_CHANNELS_PER_VOICE )
|
|
{
|
|
levels.ChannelCount = MAX_CHANNELS_PER_VOICE;
|
|
}
|
|
|
|
if( pSourceVoice->GetEffectParameters( 0, &levels, sizeof( levels ) ) != S_OK )
|
|
{
|
|
return 0.0f;
|
|
}
|
|
|
|
if( levels.ChannelCount == 1 )
|
|
{
|
|
return rmsLevels[0];
|
|
}
|
|
|
|
float rms = 0.0f;
|
|
for( uint32 i = 0; i < levels.ChannelCount; i++ )
|
|
{
|
|
rms += rmsLevels[i];
|
|
}
|
|
|
|
return rms / ( float )levels.ChannelCount;
|
|
*/
|
|
}
|
|
|
|
/*
|
|
========================
|
|
idSoundVoice_OpenAL::ResetSampleRate
|
|
========================
|
|
*/
|
|
void idSoundVoice_OpenAL::SetSampleRate( uint32 newSampleRate, uint32 operationSet )
|
|
{
|
|
/*
|
|
if( pSourceVoice == NULL || leadinSample == NULL )
|
|
{
|
|
return;
|
|
}
|
|
|
|
sampleRate = newSampleRate;
|
|
|
|
XAUDIO2_FILTER_PARAMETERS filter;
|
|
filter.Type = LowPassFilter;
|
|
filter.OneOverQ = 1.0f; // [0.0f, XAUDIO2_MAX_FILTER_ONEOVERQ]
|
|
float cutoffFrequency = 1000.0f / Max( 0.01f, occlusion );
|
|
if( cutoffFrequency * 6.0f >= ( float )sampleRate )
|
|
{
|
|
filter.Frequency = XAUDIO2_MAX_FILTER_FREQUENCY;
|
|
}
|
|
else
|
|
{
|
|
filter.Frequency = 2.0f * idMath::Sin( idMath::PI * cutoffFrequency / ( float )sampleRate );
|
|
}
|
|
assert( filter.Frequency >= 0.0f && filter.Frequency <= XAUDIO2_MAX_FILTER_FREQUENCY );
|
|
filter.Frequency = idMath::ClampFloat( 0.0f, XAUDIO2_MAX_FILTER_FREQUENCY, filter.Frequency );
|
|
|
|
pSourceVoice->SetFilterParameters( &filter, operationSet );
|
|
|
|
float freqRatio = pitch * ( float )sampleRate / ( float )sourceVoiceRate;
|
|
assert( freqRatio >= XAUDIO2_MIN_FREQ_RATIO && freqRatio <= XAUDIO2_MAX_FREQ_RATIO );
|
|
freqRatio = idMath::ClampFloat( XAUDIO2_MIN_FREQ_RATIO, XAUDIO2_MAX_FREQ_RATIO, freqRatio );
|
|
|
|
// if the value specified for maxFreqRatio is too high for the specified format, the call to CreateSourceVoice will fail
|
|
if( numChannels == 1 )
|
|
{
|
|
assert( freqRatio * ( float )SYSTEM_SAMPLE_RATE <= XAUDIO2_MAX_RATIO_TIMES_RATE_XMA_MONO );
|
|
}
|
|
else
|
|
{
|
|
assert( freqRatio * ( float )SYSTEM_SAMPLE_RATE <= XAUDIO2_MAX_RATIO_TIMES_RATE_XMA_MULTICHANNEL );
|
|
}
|
|
|
|
pSourceVoice->SetFrequencyRatio( freqRatio, operationSet );
|
|
*/
|
|
}
|
|
|
|
/*
|
|
========================
|
|
idSoundVoice_OpenAL::OnBufferStart
|
|
========================
|
|
*/
|
|
void idSoundVoice_OpenAL::OnBufferStart( idSoundSample_OpenAL* sample, int bufferNumber )
|
|
{
|
|
//SetSampleRate( sample->SampleRate(), XAUDIO2_COMMIT_NOW );
|
|
|
|
idSoundSample_OpenAL* nextSample = sample;
|
|
int nextBuffer = bufferNumber + 1;
|
|
if( nextBuffer == sample->buffers.Num() )
|
|
{
|
|
if( sample == leadinSample )
|
|
{
|
|
if( loopingSample == NULL )
|
|
{
|
|
return;
|
|
}
|
|
nextSample = loopingSample;
|
|
}
|
|
nextBuffer = 0;
|
|
}
|
|
|
|
SubmitBuffer( nextSample, nextBuffer, 0 );
|
|
}
|