1bd5513698
This allows us to change the volume of sounds in filters ("volume",
"fade" properties) without quantizing the audio prematurely. Previously,
we had a hack to apply volume changes as a mixing-time step. This
can be done at any point in the filter chain now.
This also means we only need to worry about clipping at the very
last mixing step, instead of at every place we change sound.
Dithering for 16-bit audio and more intelligent gain control
is possible, as we have full-resolution, unclipped sound at the
mixing phase.
We support high resolution source sounds; both Vorbis and MAD
supply more than 16 bits.
Several filters, such as resampling, are easier to implement as
floats; these no longer need to convert back and forth.
Negatives:
- More memory use. The main case of this is in RageSoundReader_Preload,
which will be fixed: we can preload in 16-bit without losing most of the
above.
- Some extra overhead for accessing more memory.
442 lines
13 KiB
C++
442 lines
13 KiB
C++
#include "global.h"
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#include "RageSoundReader_Chain.h"
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#include "RageSoundReader_FileReader.h"
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#include "RageSoundReader_Resample_Good.h"
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#include "RageSoundReader_Preload.h"
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#include "RageSoundReader_Pan.h"
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#include "RageLog.h"
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#include "RageUtil.h"
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#include "RageSoundMixBuffer.h"
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#include "RageSoundUtil.h"
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#include "Foreach.h"
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/*
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* Keyed sounds should pass this object to SoundReader_Preload, to preprocess it.
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* Streaming more than two or three sounds is too expensive (keyed games can play
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* two dozen), and reading from disk is too latent.
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*
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* This can also be used for chained background music, which should always stream,
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* so we don't do the preloading in here.
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*/
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RageSoundReader_Chain::RageSoundReader_Chain()
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{
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m_iPreferredSampleRate = 44100;
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m_iActualSampleRate = -1;
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m_iChannels = 0;
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m_iCurrentFrame = 0;
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m_iNextSound = 0;
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}
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RageSoundReader_Chain::~RageSoundReader_Chain()
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{
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/* Clear m_apActiveSounds. */
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while( !m_apActiveSounds.empty() )
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ReleaseSound( m_apActiveSounds.front() );
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map<RString, RageSoundReader *>::iterator it;
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FOREACH( RageSoundReader *, m_apLoadedSounds, it )
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delete *it;
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}
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RageSoundReader_Chain *RageSoundReader_Chain::Copy() const
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{
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// XXX
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FAIL_M("unimplemented");
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}
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/* The same sound may be used several times, and by several different chains. Avoid
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* loading the same sound multiple times. We need to make a Copy() if we need to
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* read it more than once at a time. */
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void RageSoundReader_Chain::AddSound( int iIndex, float fOffsetSecs, float fPan )
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{
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if( iIndex == -1 )
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return;
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Sound s;
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s.iIndex = iIndex;
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s.iOffsetMS = lrintf( fOffsetSecs * 1000 );
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s.fPan = fPan;
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s.pSound = NULL;
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m_aSounds.push_back( s );
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}
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int RageSoundReader_Chain::LoadSound( RString sPath )
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{
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sPath.MakeLower();
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map<RString, RageSoundReader *>::const_iterator it = m_apNamedSounds.find( sPath );
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if( it != m_apNamedSounds.end() )
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{
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const RageSoundReader *pReader = it->second;
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for( int i = 0; i < (int) m_apLoadedSounds.size(); ++i )
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if( m_apLoadedSounds[i] == pReader )
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return i;
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FAIL_M( sPath );
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}
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RString sError;
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RageSoundReader *pReader = RageSoundReader_FileReader::OpenFile( sPath, sError );
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if( pReader == NULL )
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{
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LOG->Warn( "RageSoundReader_Chain: error opening sound \"%s\": %s",
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sPath.c_str(), sError.c_str() );
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return -1;
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}
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m_apNamedSounds[sPath] = pReader;
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m_apLoadedSounds.push_back( m_apNamedSounds[sPath] );
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return m_apLoadedSounds.size()-1;
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}
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int RageSoundReader_Chain::LoadSound( RageSoundReader *pSound )
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{
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m_apLoadedSounds.push_back( pSound );
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return m_apLoadedSounds.size()-1;
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}
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/* If every sound has the same sample rate, return it. Otherwise, return -1. */
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int RageSoundReader_Chain::GetSampleRateInternal() const
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{
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if( m_apLoadedSounds.empty() )
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return m_iPreferredSampleRate;
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map<RString, RageSoundReader *>::const_iterator it;
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int iRate = -1;
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FOREACH_CONST( RageSoundReader *, m_apLoadedSounds, it )
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{
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if( iRate == -1 )
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iRate = (*it)->GetSampleRate();
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else if( iRate != (*it)->GetSampleRate() )
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return -1;
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}
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return iRate;
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}
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void RageSoundReader_Chain::Finish()
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{
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/* Figure out how many channels we have. All sounds must either have 1 or 2 channels,
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* which will be converted as needed, or have the same number of channels. */
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m_iChannels = 1;
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map<RString, RageSoundReader *>::iterator it;
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FOREACH( RageSoundReader *, m_apLoadedSounds, it )
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m_iChannels = max( m_iChannels, (*it)->GetNumChannels() );
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if( m_iChannels > 2 )
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{
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FOREACH( RageSoundReader *, m_apLoadedSounds, it )
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{
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if( (*it)->GetNumChannels() != m_iChannels )
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{
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LOG->Warn( "Discarded sound with %i channels, not %i",
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(*it)->GetNumChannels(), m_iChannels );
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delete (*it);
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(*it) = NULL;
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}
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}
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}
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/* Remove any sounds that don't have corresponding RageSoundReaders. */
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for( unsigned i = 0; i < m_aSounds.size(); )
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{
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Sound &sound = m_aSounds[i];
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if( m_apLoadedSounds[sound.iIndex] == NULL )
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{
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m_aSounds.erase( m_aSounds.begin()+i );
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continue;
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}
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++i;
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}
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/*
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* We might get different sample rates from our sources. If they're all the same
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* sample rate, just leave it alone, so the whole sound can be resampled as a group.
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* If not, resample eveything to the preferred rate. (Using the preferred rate
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* should avoid redundant resampling later.)
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*/
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m_iActualSampleRate = GetSampleRateInternal();
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if( m_iActualSampleRate == -1 )
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{
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FOREACH( RageSoundReader *, m_apLoadedSounds, it )
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{
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RageSoundReader *&pSound = (*it);
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RageSoundReader_Resample_Good *pResample = new RageSoundReader_Resample_Good( pSound, m_iPreferredSampleRate );
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pSound = pResample;
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}
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m_iActualSampleRate = m_iPreferredSampleRate;
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}
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/* Attempt to preload all sounds. */
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FOREACH( RageSoundReader *, m_apLoadedSounds, it )
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{
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RageSoundReader *&pSound = (*it);
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RageSoundReader_Preload::PreloadSound( pSound );
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}
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/* Sort the sounds by start time. */
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sort( m_aSounds.begin(), m_aSounds.end() );
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}
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int RageSoundReader_Chain::SetPosition( int iFrame )
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{
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/* Clear m_apActiveSounds. */
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while( !m_apActiveSounds.empty() )
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ReleaseSound( m_apActiveSounds.front() );
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m_iCurrentFrame = iFrame;
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/* Run through all sounds in the chain, and activate all sounds which have data
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* at iFrame. */
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for( m_iNextSound = 0; m_iNextSound < m_aSounds.size(); ++m_iNextSound )
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{
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Sound *pSound = &m_aSounds[m_iNextSound];
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int iOffsetFrame = pSound->GetOffsetFrame( GetSampleRate() );
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/* If this sound is in the future, skip it. */
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if( iOffsetFrame > iFrame )
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break;
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/* Find the RageSoundReader. */
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ActivateSound( pSound );
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RageSoundReader *pReader = pSound->pSound;
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int iOffsetFrames = iFrame - iOffsetFrame;
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if( pReader->SetPosition(iOffsetFrames) == 0 )
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{
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/* We're past the end of this sound. */
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ReleaseSound( pSound );
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continue;
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}
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}
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/* If no sounds were started, and we have no sounds ahead of us, we've seeked
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* past EOF. */
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if( m_apActiveSounds.empty() && m_iNextSound == m_aSounds.size() )
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return 0;
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return 1;
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}
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void RageSoundReader_Chain::ActivateSound( Sound *s )
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{
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RageSoundReader *pSound = m_apLoadedSounds[s->iIndex];
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s->pSound = pSound->Copy();
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/* Add a balance filter. If this source has the same number of channels
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* as this sound, and does not need to be panned, we can omit this. */
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if( s->fPan != 0.0f || s->pSound->GetNumChannels() != this->GetNumChannels() )
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{
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s->pSound = new RageSoundReader_Pan( s->pSound );
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s->pSound->SetProperty( "Pan", s->fPan );
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}
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m_apActiveSounds.push_back( s );
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}
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void RageSoundReader_Chain::ReleaseSound( Sound *s )
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{
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vector<Sound *>::iterator it = find( m_apActiveSounds.begin(), m_apActiveSounds.end(), s );
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ASSERT( it != m_apActiveSounds.end() );
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RageSoundReader *&pSound = s->pSound;
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delete pSound;
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pSound = NULL;
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m_apActiveSounds.erase( it );
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}
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bool RageSoundReader_Chain::SetProperty( const RString &sProperty, float fValue )
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{
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bool bRet = false;
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for( unsigned i = 0; i < m_apActiveSounds.size(); ++i )
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{
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if( m_apActiveSounds[i]->pSound->SetProperty(sProperty, fValue) )
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bRet = true;
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}
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return bRet;
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}
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int RageSoundReader_Chain::GetNextSourceFrame() const
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{
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return m_iCurrentFrame;
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/* if( m_apActiveSounds.empty() )
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return m_iCurrentFrame;
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int iPosition = m_apActiveSounds[0]->pSound->GetNextSourceFrame();
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iPosition += m_apActiveSounds[0]->GetOffsetFrame( GetSampleRate() );
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for( unsigned i = 1; i < m_apActiveSounds.size(); ++i )
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{
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int iThisPosition = m_apActiveSounds[i]->pSound->GetNextSourceFrame();
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iThisPosition += m_apActiveSounds[i]->GetOffsetFrame( GetSampleRate() );
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if( iThisPosition != iPosition )
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LOG->Warn( "RageSoundReader_Chain: sound positions moving at different rates" );
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}
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return iPosition;
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*/
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}
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float RageSoundReader_Chain::GetStreamToSourceRatio() const
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{
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if( m_apActiveSounds.empty() )
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return 1.0f;
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float iRate = m_apActiveSounds[0]->pSound->GetStreamToSourceRatio();
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for( unsigned i = 1; i < m_apActiveSounds.size(); ++i )
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{
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if( m_apActiveSounds[i]->pSound->GetStreamToSourceRatio() != iRate )
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LOG->Warn( "RageSoundReader_Chain: sound rates changing differently" );
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}
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return iRate;
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}
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/* As we iterate through the sound tree, we'll find that we need data from different
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* sounds; a sound may be needed by more than one other sound. */
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int RageSoundReader_Chain::Read( float *pBuffer, int iFrames )
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{
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while( m_iNextSound < m_aSounds.size() && m_iCurrentFrame == m_aSounds[m_iNextSound].GetOffsetFrame(m_iActualSampleRate) )
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{
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Sound *pSound = &m_aSounds[m_iNextSound];
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ActivateSound( pSound );
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++m_iNextSound;
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}
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if( m_iNextSound == m_aSounds.size() && m_apActiveSounds.empty() )
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return END_OF_FILE;
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/* Clamp iFrames to the beginning of the next sound we need to start. */
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if( m_iNextSound < m_aSounds.size() )
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{
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int iOffsetFrame = m_aSounds[m_iNextSound].GetOffsetFrame( m_iActualSampleRate );
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ASSERT_M( iOffsetFrame >= m_iCurrentFrame, ssprintf("%i %i", iOffsetFrame, m_iCurrentFrame) );
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int iFramesToRead = iOffsetFrame - m_iCurrentFrame;
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iFrames = min( iFramesToRead, iFrames );
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}
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if( m_apActiveSounds.size() == 1 &&
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m_apActiveSounds.front()->pSound->GetNumChannels() == m_iChannels &&
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m_apActiveSounds.front()->pSound->GetSampleRate() == m_iActualSampleRate )
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{
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/* We have only one source, and it matches our target. Don't mix; read
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* directly from the source into the destination. This is to optimize
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* the common case of having one BGM track and no autoplay sounds. */
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iFrames = m_apActiveSounds.front()->pSound->Read( pBuffer, iFrames );
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if( iFrames < 0 )
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ReleaseSound( m_apActiveSounds.front() );
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if( iFrames > 0 )
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m_iCurrentFrame += iFrames;
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return iFrames;
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}
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if( m_apActiveSounds.empty() )
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{
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/* If we have more sounds ahead of us, pretend we read the entire block, since
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* there's silence in between. Otherwise, we're at EOF. */
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memset( pBuffer, 0, iFrames * m_iChannels * sizeof(float) );
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m_iCurrentFrame += iFrames;
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return iFrames;
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}
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RageSoundMixBuffer mix;
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/* Read iFrames from each sound. */
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float Buffer[2048];
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iFrames = min( iFrames, (int) (ARRAYLEN(Buffer) / m_iChannels) );
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for( unsigned i = 0; i < m_apActiveSounds.size(); )
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{
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RageSoundReader *pSound = m_apActiveSounds[i]->pSound;
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ASSERT( pSound->GetNumChannels() == m_iChannels ); // guaranteed by ActivateSound and Finish
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/* If we receive less than we were asked for, keep asking for more data. Most
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* filters would simply return what they have in this situation, but we want
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* to deal transparently with separate sounds returning differently-sized partial
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* blocks. */
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int iFramesRead = 0;
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while( iFramesRead < iFrames )
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{
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int iGotFrames = pSound->RetriedRead( Buffer, iFrames - iFramesRead );
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if( iGotFrames < 0 )
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{
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iFramesRead = iGotFrames;
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break;
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}
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mix.SetWriteOffset( iFramesRead * pSound->GetNumChannels() );
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mix.write( Buffer, iFramesRead * pSound->GetNumChannels() );
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iFramesRead += iGotFrames;
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}
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if( iFramesRead < 0 )
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ReleaseSound( m_apActiveSounds[i] );
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else
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++i;
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}
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/* Read mixed frames into the output buffer. */
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int iMaxFramesRead = mix.size() / m_iChannels;
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mix.read( pBuffer );
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m_iCurrentFrame += iMaxFramesRead;
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return iMaxFramesRead;
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}
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int RageSoundReader_Chain::GetLength() const
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{
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int iLength = 0;
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for( unsigned i = 0; i < m_aSounds.size(); ++i )
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{
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const Sound &sound = m_aSounds[i];
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const RageSoundReader *pSound = m_apLoadedSounds[sound.iIndex];
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int iThisLength = pSound->GetLength();
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if( iThisLength )
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iLength = max( iLength, iThisLength + sound.iOffsetMS );
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}
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return iLength;
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}
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int RageSoundReader_Chain::GetLength_Fast() const
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{
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int iLength = 0;
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for( unsigned i = 0; i < m_aSounds.size(); ++i )
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{
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const Sound &sound = m_aSounds[i];
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const RageSoundReader *pSound = m_apLoadedSounds[sound.iIndex];
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int iThisLength = pSound->GetLength_Fast();
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if( iThisLength )
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iLength = max( iLength, iThisLength + sound.iOffsetMS );
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}
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return iLength;
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}
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/*
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* Copyright (c) 2004-2006 Glenn Maynard
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* All rights reserved.
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*
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* Permission is hereby granted, free of charge, to any person obtaining a
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* copy of this software and associated documentation files (the
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* "Software"), to deal in the Software without restriction, including
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* without limitation the rights to use, copy, modify, merge, publish,
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* distribute, and/or sell copies of the Software, and to permit persons to
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* whom the Software is furnished to do so, provided that the above
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* copyright notice(s) and this permission notice appear in all copies of
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* the Software and that both the above copyright notice(s) and this
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* permission notice appear in supporting documentation.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
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* OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
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* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OF
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* THIRD PARTY RIGHTS. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR HOLDERS
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* INCLUDED IN THIS NOTICE BE LIABLE FOR ANY CLAIM, OR ANY SPECIAL INDIRECT
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* OR CONSEQUENTIAL DAMAGES, OR ANY DAMAGES WHATSOEVER RESULTING FROM LOSS
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* OF USE, DATA OR PROFITS, WHETHER IN AN ACTION OF CONTRACT, NEGLIGENCE OR
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* OTHER TORTIOUS ACTION, ARISING OUT OF OR IN CONNECTION WITH THE USE OR
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* PERFORMANCE OF THIS SOFTWARE.
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*/
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