merge RageSoundResampler into RageSoundReader_Resample_Fast;
they're just different interfaces for the same thing
This commit is contained in:
@@ -4,7 +4,18 @@
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RageSoundReader_Resample_Fast::RageSoundReader_Resample_Fast()
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{
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m_pSource = NULL;
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m_iOutputSampleRate = -1;
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m_iInputRate = -1;
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m_iOutputRate = -1;
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m_iChannels = 2;
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Reset();
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}
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void RageSoundReader_Resample_Fast::Reset()
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{
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m_bAtEof = false;
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memset( m_iPrevSample, 0, sizeof(m_iPrevSample) );
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m_iPos = 0;
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m_OutBuf.clear();
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}
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void RageSoundReader_Resample_Fast::Open( SoundReader *pSource )
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@@ -12,9 +23,9 @@ void RageSoundReader_Resample_Fast::Open( SoundReader *pSource )
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m_pSource = pSource;
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ASSERT(m_pSource);
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m_iOutputSampleRate = m_pSource->GetSampleRate();
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m_Resamp.SetInputSampleRate( m_iOutputSampleRate );
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m_Resamp.SetChannels( m_pSource->GetNumChannels() );
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m_iInputRate = m_iOutputRate = m_pSource->GetSampleRate();
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ASSERT( m_pSource->GetNumChannels() < MAX_CHANNELS );
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m_iChannels = m_pSource->GetNumChannels();
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}
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@@ -23,33 +34,30 @@ RageSoundReader_Resample_Fast::~RageSoundReader_Resample_Fast()
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delete m_pSource;
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}
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void RageSoundReader_Resample_Fast::SetSampleRate( int hz )
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void RageSoundReader_Resample_Fast::SetSampleRate( int iRate )
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{
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m_iOutputSampleRate = hz;
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m_Resamp.SetOutputSampleRate( m_iOutputSampleRate );
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m_iOutputRate = iRate;
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}
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int RageSoundReader_Resample_Fast::GetLength() const
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{
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m_Resamp.reset();
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return m_pSource->GetLength();
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}
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int RageSoundReader_Resample_Fast::GetLength_Fast() const
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{
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m_Resamp.reset();
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return m_pSource->GetLength_Fast();
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}
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int RageSoundReader_Resample_Fast::SetPosition_Accurate( int ms )
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{
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m_Resamp.reset();
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Reset();
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return m_pSource->SetPosition_Accurate(ms);
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}
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int RageSoundReader_Resample_Fast::SetPosition_Fast( int ms )
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{
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m_Resamp.reset();
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Reset();
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return m_pSource->SetPosition_Fast(ms);
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}
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static const int BUFSIZE = 1024*16;
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@@ -60,11 +68,17 @@ int RageSoundReader_Resample_Fast::Read( char *pBuf, unsigned iSize )
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while( iSize )
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{
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{
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int iGot = m_Resamp.read( pBuf, iSize );
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ASSERT( (iSize % sizeof(int16_t)) == 0 );
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if( iGot == -1 )
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/* If no data is available, and we're m_bAtEof, stop. */
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if( m_OutBuf.size() == 0 && m_bAtEof )
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break;
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/* Fill as much as we have. */
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int iGot = min( m_OutBuf.size()*sizeof(int16_t), iSize );
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memcpy( pBuf, &m_OutBuf[0], iGot );
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m_OutBuf.erase( m_OutBuf.begin(), m_OutBuf.begin() + iGot/sizeof(int16_t) );
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pBuf += iGot;
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iSize -= iGot;
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iBytesRead += iGot;
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@@ -75,7 +89,7 @@ int RageSoundReader_Resample_Fast::Read( char *pBuf, unsigned iSize )
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{
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static char buf[BUFSIZE];
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int iGot = m_pSource->Read(buf, sizeof(buf));
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int iGot = m_pSource->Read( buf, sizeof(buf) );
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if( iGot == -1 )
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{
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@@ -83,9 +97,29 @@ int RageSoundReader_Resample_Fast::Read( char *pBuf, unsigned iSize )
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return -1;
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}
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if( iGot == 0 )
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m_Resamp.eof();
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{
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ASSERT( !m_bAtEof );
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/* Write some silence to flush out the real data. If we don't have any sound,
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* don't do this, so seeking past end of file doesn't write silence. */
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bool bNeedsFlush = false;
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for( int c = 0; c < m_iChannels; ++c )
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if( m_iPrevSample[c] != 0 )
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bNeedsFlush = true;
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if( bNeedsFlush )
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{
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const int iSize = m_iChannels*16;
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int16_t *pData = new int16_t[iSize];
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memset( pData, 0, iSize * sizeof(int16_t) );
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WriteSamples( pData, iSize * sizeof(int16_t) );
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delete[] pData;
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}
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m_bAtEof = true;
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}
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else
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m_Resamp.write( buf, iGot );
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WriteSamples( buf, iGot );
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}
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}
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@@ -97,12 +131,77 @@ SoundReader *RageSoundReader_Resample_Fast::Copy() const
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SoundReader *pNewSource = m_pSource->Copy();
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RageSoundReader_Resample_Fast *pRet = new RageSoundReader_Resample_Fast;
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pRet->Open( pNewSource );
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pRet->SetSampleRate( m_iOutputSampleRate );
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pRet->SetSampleRate( m_iOutputRate );
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return pRet;
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}
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/* Write data to be converted. */
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void RageSoundReader_Resample_Fast::WriteSamples(const void *data_, int bytes)
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{
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ASSERT(!m_bAtEof);
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const int16_t *data = (const int16_t *) data_;
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const unsigned samples = bytes / sizeof(int16_t);
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const unsigned frames = samples / m_iChannels;
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if( m_iInputRate == m_iOutputRate )
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{
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/* Optimization: */
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m_OutBuf.insert(m_OutBuf.end(), data, data+samples);
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return;
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}
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/* Lerp. */
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const int FIXED_SHIFT = 14;
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const int FIXED_ONE = 1<<FIXED_SHIFT;
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int iInputSamplesPerOutputSample =
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(m_iInputRate<<FIXED_SHIFT) / m_iOutputRate;
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m_OutBuf.resize( (frames*m_iChannels*m_iOutputRate)/m_iInputRate + 10 );
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int iSize = 0;
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for( int c = 0; c < m_iChannels; ++c )
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{
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int iPos = m_iPos;
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const int16_t *pInBuf = &data[c];
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int16_t *pOutBuf = &m_OutBuf[c];
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int iSamplesInput = 0;
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int iSamplesOutput = 0;
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int16_t iPrevSample = m_iPrevSample[c];
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for( unsigned f = 0; f < frames; ++f )
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{
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while( iPos < FIXED_ONE )
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{
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int iSamp = iPrevSample * (FIXED_ONE-iPos) +
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pInBuf[iSamplesInput] * iPos;
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pOutBuf[iSamplesOutput] = int16_t(iSamp >> FIXED_SHIFT);
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iSamplesOutput += m_iChannels;
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iPos += iInputSamplesPerOutputSample;
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}
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iPos -= FIXED_ONE;
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iPrevSample = pInBuf[iSamplesInput];
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iSamplesInput += m_iChannels;
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}
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m_iPrevSample[c] = iPrevSample;
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if( c == m_iChannels-1 )
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{
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iSize = iSamplesOutput;
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m_iPos = iPos;
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}
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}
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m_OutBuf.erase( m_OutBuf.begin()+iSize, m_OutBuf.end() );
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}
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/*
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* Copyright (c) 2003-2005 Glenn Maynard
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* Copyright (c) 2003-2006 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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