Glenn Maynard
5d64dff234
This eliminates the (optional) libresample dependency, which will clean up building and eliminate the goofy 5k DLL. It's also faster. This doesn't implement "HighQuality". That would involve changing L from 8 to 16. I might implement that in the future, but it sounds decent to me. (It's a little tricky, since PolyphaseFilter::RunPolyphaseFilter needs L to be a constant for speed.)
658 lines
17 KiB
C++
658 lines
17 KiB
C++
/*
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* This implements audio resampling, using the method described at:
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* http://www.dspguru.com/info/faqs/mrfaq.htm
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*
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* Each conversion ratio uses some memory, but the resulting table is
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* shared, so the memory overhead per stream is negligible.
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*/
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#include "global.h"
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#include "RageSoundReader_Resample_Good.h"
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#include "RageLog.h"
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#include "RageUtil.h"
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#include "RageMath.h"
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#include "RageThreads.h"
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#include <numeric>
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/* Filter length. This must be a power of 2. */
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#define L 8
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namespace
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{
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float sincf( float f )
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{
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if( f == 0 )
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return 1;
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return sinf(f)/f;
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}
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/* Modified Bessel function I0. From Abramowitz and Stegun "Handbook of Mathematical
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* Functions", "Modified Bessel Functions I and K". */
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float BesselI0( float fX )
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{
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float fAbsX = fabsf( fX );
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if( fAbsX < 3.75f )
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{
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float y = fX / 3.75f;
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y *= y;
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float fRet = 1.0f+y*(+3.5156229f+y*(+3.0899424f+y*(+1.2067492f+y*(+0.2659732f+y*(+0.0360768f+y*+0.0045813f)))));
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return fRet;
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}
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else
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{
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float y = 3.75f/fAbsX;
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float fRet = (exp(fAbsX)/sqrt(fAbsX)) *
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(+0.39894228f+y*(+0.01328592f+y*(+0.00225319f+y*(-0.00157565f+y*(0.00916281f+
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y*(-0.02057706f+y*(+0.02635537f+y*(-0.01647633f+y*+0.00392377f))))))));
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return fRet;
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}
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}
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/*
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* Kaiser window:
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*
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* K(n) = I0( B*sqrt(1-(n/p)^2) )
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* -----------------------
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* I0(B)
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*
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* where B is the beta parameter, p is len/2, and n is in [-len/2,+len/2].
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*/
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void ApplyKaiserWindow( float *pBuf, int iLen, float fBeta )
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{
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const float fDenom = BesselI0(fBeta);
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float p = (iLen-1)/2.0f;
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for( int n = 0; n < iLen; ++n )
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{
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float fN1 = fabsf((n-p)/p);
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float fNum = fBeta * sqrtf( max(1-fN1*fN1, 0) );
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fNum = BesselI0( fNum );
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float fVal = fNum/fDenom;
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pBuf[n] *= fVal;
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}
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}
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void MultiplyVector( float *pStart, float *pEnd, float f )
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{
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for( ; pStart != pEnd; ++pStart )
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*pStart *= f;
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}
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void GenerateSincLowPassFilter( float *pFIR, int iWinSize, float fCutoff )
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{
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float p = (iWinSize-1)/2.0f;
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for( int n = 0; n < iWinSize; ++n )
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{
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float fN1 = (n-p);
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float fVal = sincf(2*PI*fCutoff * fN1)*(2*fCutoff);
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// printf( "n %i, %f, %f -> %f\n", n, p, fN1, fVal );
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pFIR[n] = fVal;
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}
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#if 0
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float *pFIRp = pFIR+iWinSize/2;
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for(int i=-iWinSize/2;i<=iWinSize/2;i++)
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{
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float ff = sinc(2*M_PI*fCutoff * (i + 0.0))*(2*fCutoff);
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printf( "%i: %f\n", i, ff );
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pFIRp[i]=ff;
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}
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for( int i=0; i < iWinSize; i++ )
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printf( "sinc: %i: %f\n", i, pFIR[i] );
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#endif
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}
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void NormalizeVector( float *pBuf, int iSize )
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{
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float fTotal = accumulate( &pBuf[0], &pBuf[iSize], 0.0f );
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MultiplyVector( &pBuf[0], &pBuf[iSize], 1/fTotal );
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}
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int GCD( int i1, int i2 )
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{
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while(1)
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{
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unsigned iRem = i2 % i1;
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if( iRem == 0 )
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return i1;
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i2 = i1;
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i1 = iRem;
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}
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return i1;
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}
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}
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#if 0
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void RunFIRFilter( float *pIn, float *pOut, int iInputValues, float *pFIR, int iWinSize )
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{
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for( int i = 0; i < iInputValues; ++i )
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{
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float fSum = 0;
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const float *pInData = &pIn[i];
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for( int j = 0; j < iWinSize; ++j )
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{
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float in = pInData[j];
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fSum += in*pFIR[j];
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printf( "%i: in %f * %f, += %f\n", j, pInData[j], pFIR[j], in*pFIR[j] );
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}
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pOut[i] = fSum;
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}
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}
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#endif
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template<typename T>
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class AlignedBuffer
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{
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public:
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AlignedBuffer( int iSize )
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{
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m_iSize = iSize;
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m_pBuf = new T[m_iSize];
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}
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AlignedBuffer( const AlignedBuffer &cpy )
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{
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m_iSize = cpy.m_iSize;
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m_pBuf = new T[m_iSize];
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memcpy( m_pBuf, cpy.m_pBuf, sizeof(T)*m_iSize );
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}
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~AlignedBuffer()
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{
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delete [] m_pBuf;
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}
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operator T*() { return m_pBuf; }
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operator const T*() const { return m_pBuf; }
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private:
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T& operator=( T &rhs );
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int m_iSize;
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T *m_pBuf;
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};
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struct PolyphaseFilter
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{
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struct State
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{
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State( const PolyphaseFilter &Target ):
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m_fBuf( L * 2 )
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{
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m_iPolyIndex = Target.m_iUpFactor-1;
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m_iFilled = 0;
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m_iBufNext = 0;
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}
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int m_iPolyIndex;
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int m_iFilled;
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/* This buffer is duplicated. If the circular buffer is size L, the actual buffer
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* is size L*2, and data at buf[N] is also at buf[N+L]. That way, we can access
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* up to buf[N*2-1] without having to wrap. */
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AlignedBuffer<float> m_fBuf;
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int m_iBufNext;
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};
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PolyphaseFilter( int iUpFactor ):
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m_pPolyphase( L*iUpFactor )
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{
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m_iUpFactor = iUpFactor;
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}
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void Generate( const float *pFIR );
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int RunPolyphaseFilter( State &State, const float *pIn, int iSamplesIn, int iDownFactor,
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float *pOut, int iSamplesOut ) const;
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int GetLatency() const { return L/2; }
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int NumInputsForOutputSamples( const State &State, int iOut, int iDownFactor ) const;
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private:
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AlignedBuffer<float> m_pPolyphase;
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int m_iUpFactor;
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};
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/*
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* Convert an FIR filter to a polyphase filter.
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*
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* pFIR is the input FIR filter, which has iL*iUpFactor values.
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* iL is the number of real samples each output sample looks at.
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* iUpFactor is the actual upsampling factor; the amount of zero-stuffing between each real sample.
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* pOutput is the 2D output polyphase filter, with iL*iL values.
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*
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* With an upsampling factor (iUpFactor) of 3, and a sinc filter length of 12 (iL*iUpFactor),
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*
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* input first output sample (before decimation)
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* sample second output sample
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* third output sample
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*
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* 0 0
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* 0 1 0
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* 1592 2 1 0
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* 0 3 2 1
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* 0 4 3 2
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* 1623 5 4 3
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* 0 6 5 4
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* 0 7 6 5
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* 1682 8 7 6
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* 0 9 8 7
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* 0 10 9 8
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* 1730 11 10 9
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* 0 11 10
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* 0 11
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*
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* first row: 2, 5, 8, 11
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* second: 1, 4, 7, 10
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* third: 0, 3, 6, 9
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* Read a new sample after passing the last line.
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*/
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void PolyphaseFilter::Generate( const float *pFIR )
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{
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float *pOutput=m_pPolyphase;
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int iInputSize = L*m_iUpFactor;
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for( int iRow = 0; iRow < m_iUpFactor; ++iRow )
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{
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int iInputOffset = (m_iUpFactor-iRow-1) % m_iUpFactor;
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for( int iCol = 0; iCol < L; ++iCol )
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{
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*pOutput = pFIR[iInputOffset];
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++pOutput;
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iInputOffset += m_iUpFactor;
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iInputOffset %= iInputSize;
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}
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}
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}
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/*
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* We only want one boundary check when running the filter; either on the
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* number of inputs used, or the number of outputs produced. Otherwise, we'll
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* have to maintain two counters, and check two values per iteration.
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*
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* First, call NumInputsForOutputSamples(out), to find out how many inputs to supply to get
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* the desired number of outputs. Then, pass the data, the input count
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* and the output count to RunPolyphaseFilter.
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*
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* - When downsampling, we use the number of inputs as the boundary. For example,
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* if the ratio is 1:3 (downsample x3), and the user gives us 10 samples, then we
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* process until we've consumed all of the input. (This will result in exactly
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* the number of samples the user asked for with NumInputsForOutputSamples.)
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*
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* - When upsampling, we use the number of outputs as the boundary. For example,
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* if the ratio is 3:1 (upsample x3), and the user wants 8 samples to be output,
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* we'll have been given 3 samples as input. Process until we've produced 8
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* samples.
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*
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* In both cases, we have overlap. In the first, it's possible that we could
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* have consumed an additional input without producing an output. In the second,
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* it's possible that we could have produced an additional output without
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* consuming an input.
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*/
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int PolyphaseFilter::RunPolyphaseFilter(
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State &State,
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const float *pIn, int iSamplesIn, int iDownFactor,
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float *pOut, int iSamplesOut ) const
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{
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ASSERT( iSamplesIn >= 0 );
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float *pOutOrig = pOut;
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const float *pInEnd = pIn + iSamplesIn;
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const float *pOutEnd = pOut + iSamplesOut;
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int iFilled = State.m_iFilled;
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int iPolyIndex = State.m_iPolyIndex;
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while( pOut != pOutEnd )
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{
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if( iFilled < L )
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{
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if( pIn == pInEnd )
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break;
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State.m_fBuf[State.m_iBufNext] = *pIn;
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State.m_fBuf[State.m_iBufNext + L] = *pIn;
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++State.m_iBufNext;
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State.m_iBufNext &= L-1;
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++pIn;
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++iFilled;
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continue;
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}
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while( pOut != pOutEnd )
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{
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const float *pCurPoly = &m_pPolyphase[iPolyIndex*L];
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const float *pInData = &State.m_fBuf[State.m_iBufNext];
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float fTot = 0;
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for( int j = 0; j < L; ++j )
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fTot += pInData[j]*pCurPoly[j];
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*pOut = fTot;
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++pOut;
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iPolyIndex += iDownFactor;
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if( iPolyIndex >= m_iUpFactor )
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break;
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}
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iFilled -= iPolyIndex/m_iUpFactor;
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iPolyIndex %= m_iUpFactor;
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}
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State.m_iFilled = iFilled;
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State.m_iPolyIndex = iPolyIndex;
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return pOut - pOutOrig;
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}
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/*
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* Return the number of input samples needed to produce the given number of output
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* samples. This is dependent on the number of bytes in the buffer and the current
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* position of the stream.
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*/
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int PolyphaseFilter::NumInputsForOutputSamples( const State &State, int iOut, int iDownFactor ) const
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{
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int iIn = 0;
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int iFilled = State.m_iFilled;
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int iPolyIndex = State.m_iPolyIndex;
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#if 0
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while( iOut > 0 )
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{
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if( iFilled < L )
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{
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int iToFill = L-iFilled;
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iIn += iToFill;
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iFilled += iToFill;
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}
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while( iFilled == L && iOut )
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{
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--iOut;
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iPolyIndex += iDownFactor;
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if( iPolyIndex >= m_iUpFactor )
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break;
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}
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iFilled -= iPolyIndex/m_iUpFactor;
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iPolyIndex %= m_iUpFactor;
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}
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#endif
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if( iOut > 0 )
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{
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if( iFilled < L )
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{
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int iToFill = L-iFilled;
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iIn += iToFill;
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}
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// The -1 here is because we don't refill m_fBuf after writing the last output.
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iPolyIndex += iDownFactor*(iOut-1);
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iIn += iPolyIndex/m_iUpFactor;
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}
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return iIn;
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}
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/*
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* Interface to PolyphaseFilter, providing a simple resampling interface. This handles
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* reuse of PolyphaseFilters. This does not handle delay, flushing, or multiple channels.
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*/
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class RageSoundResampler_Polyphase
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{
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public:
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RageSoundResampler_Polyphase( int iSourceRate, int iDestRate )
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{
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int iUpFactor = iDestRate;
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m_iDownFactor = iSourceRate;
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{
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int iGCD = GCD( iUpFactor, m_iDownFactor );
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iUpFactor /= iGCD;
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m_iDownFactor /= iGCD;
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}
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float fCutoffFrequency;
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{
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/*
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* If we're upsampling, we want the low-pass filter to cut off at the
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* nyquist frequency of the original sample.
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*
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* If we're downsampling, we want the low-pass filter to cut off at the
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* nyquist frequency of the new sample.
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*/
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fCutoffFrequency = 1.0f / (2*iUpFactor);
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fCutoffFrequency = min( fCutoffFrequency, 1.0f / (2*m_iDownFactor) );
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LOG->Trace( "cutoff frequency %f -> %f, %f", fCutoffFrequency, 1.0f / (2*iUpFactor), 1.0f / (2*m_iDownFactor) );
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}
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/* Cache filter data, and reuse it without copying. All operations after creation
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* are const, so this doesn't cause thread-safety problems. */
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typedef map<pair<int,float>, PolyphaseFilter *> FilterMap;
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static RageMutex PolyphaseFiltersLock("PolyphaseFiltersLock");
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static FilterMap g_mapPolyphaseFilters;
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PolyphaseFiltersLock.Lock();
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pair<int,float> params( make_pair(iUpFactor, fCutoffFrequency) );
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FilterMap::const_iterator it = g_mapPolyphaseFilters.find(params);
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if( it == g_mapPolyphaseFilters.end() )
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{
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int iWinSize = L*iUpFactor;
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float *pFIR = new float[iWinSize];
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GenerateSincLowPassFilter( pFIR, iWinSize, fCutoffFrequency );
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ApplyKaiserWindow( pFIR, iWinSize, 8 );
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NormalizeVector( pFIR, iWinSize );
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MultiplyVector( &pFIR[0], &pFIR[iWinSize], (float) iUpFactor );
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PolyphaseFilter *pPolyphase = new PolyphaseFilter( iUpFactor );
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pPolyphase->Generate( pFIR );
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delete [] pFIR;
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g_mapPolyphaseFilters[params] = pPolyphase;
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m_pPolyphase = pPolyphase;
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}
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else
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{
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/* We already have a filter for this upsampling factor and cutoff; use it. */
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m_pPolyphase = it->second;
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}
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PolyphaseFiltersLock.Unlock();
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m_pState = new PolyphaseFilter::State( *m_pPolyphase );
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}
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~RageSoundResampler_Polyphase()
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{
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delete m_pState;
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}
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int Run( const float *pIn, int iSamplesIn, float *pOut, int iSamplesOut ) const
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{
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return m_pPolyphase->RunPolyphaseFilter( *m_pState, pIn, iSamplesIn, m_iDownFactor, pOut, iSamplesOut );
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}
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void Reset()
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{
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delete m_pState;
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m_pState = new PolyphaseFilter::State( *m_pPolyphase );
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}
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int NumInputsForOutputSamples( int iOut ) const { return m_pPolyphase->NumInputsForOutputSamples(*m_pState, iOut, m_iDownFactor); }
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int GetLatency() const { return m_pPolyphase->GetLatency(); }
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RageSoundResampler_Polyphase( const RageSoundResampler_Polyphase &cpy )
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{
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m_pPolyphase = new PolyphaseFilter(*cpy.m_pPolyphase);
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m_pState = new PolyphaseFilter::State(*cpy.m_pState);
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m_iDownFactor = cpy.m_iDownFactor;
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}
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private:
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const PolyphaseFilter *m_pPolyphase;
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PolyphaseFilter::State *m_pState;
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int m_iDownFactor;
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};
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RageSoundReader_Resample_Good::RageSoundReader_Resample_Good()
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{
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m_pSource = NULL;
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m_iSampleRate = -1;
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}
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/* Call this if the input position is changed or reset. */
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void RageSoundReader_Resample_Good::Reset()
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{
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for( size_t iChannel = 0; iChannel < m_pSource->GetNumChannels(); ++iChannel )
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resamplers[iChannel]->Reset();
|
|
}
|
|
|
|
|
|
/* Call this if the sample factor changes. */
|
|
void RageSoundReader_Resample_Good::ReopenResampler()
|
|
{
|
|
for( size_t iChannel = 0; iChannel < resamplers.size(); ++iChannel )
|
|
delete resamplers[iChannel];
|
|
resamplers.clear();
|
|
for( size_t iChannel = 0; iChannel < m_pSource->GetNumChannels(); ++iChannel )
|
|
{
|
|
RageSoundResampler_Polyphase *p = new RageSoundResampler_Polyphase( m_pSource->GetSampleRate(), m_iSampleRate );
|
|
resamplers.push_back( p );
|
|
}
|
|
}
|
|
|
|
void RageSoundReader_Resample_Good::Open( SoundReader *pSource )
|
|
{
|
|
ASSERT(pSource);
|
|
m_pSource = pSource;
|
|
}
|
|
|
|
|
|
RageSoundReader_Resample_Good::~RageSoundReader_Resample_Good()
|
|
{
|
|
for( size_t iChannel = 0; iChannel < resamplers.size(); ++iChannel )
|
|
delete resamplers[iChannel];
|
|
delete m_pSource;
|
|
}
|
|
|
|
void RageSoundReader_Resample_Good::SetSampleRate( int iHZ )
|
|
{
|
|
m_iSampleRate = iHZ;
|
|
ReopenResampler();
|
|
}
|
|
|
|
int RageSoundReader_Resample_Good::GetLength() const
|
|
{
|
|
return m_pSource->GetLength();
|
|
}
|
|
|
|
int RageSoundReader_Resample_Good::GetLength_Fast() const
|
|
{
|
|
return m_pSource->GetLength_Fast();
|
|
}
|
|
|
|
int RageSoundReader_Resample_Good::SetPosition_Accurate(int ms)
|
|
{
|
|
Reset();
|
|
return m_pSource->SetPosition_Accurate(ms);
|
|
}
|
|
|
|
int RageSoundReader_Resample_Good::SetPosition_Fast(int ms)
|
|
{
|
|
Reset();
|
|
return m_pSource->SetPosition_Fast(ms);
|
|
}
|
|
|
|
int RageSoundReader_Resample_Good::Read( char *bufp, unsigned len )
|
|
{
|
|
int iChannels = resamplers.size();
|
|
int iBytesPerFrame = sizeof(int16_t) * iChannels;
|
|
|
|
int iFrames = len / iBytesPerFrame; /* bytes -> frames */
|
|
int16_t *pBuf = (int16_t *) bufp;
|
|
|
|
int iFramesRead = 0;
|
|
|
|
{
|
|
int iFramesNeeded = resamplers[0]->NumInputsForOutputSamples(iFrames);
|
|
int iBytesNeeded = iFramesNeeded * sizeof(int16_t) * iChannels;
|
|
int16_t *pTmpBuf = (int16_t *) alloca( iBytesNeeded );
|
|
ASSERT( pTmpBuf );
|
|
int iBytesIn = m_pSource->Read( (char *) pTmpBuf, iBytesNeeded );
|
|
|
|
if( iBytesIn == -1 )
|
|
{
|
|
SetError( m_pSource->GetError() );
|
|
return -1;
|
|
}
|
|
|
|
iBytesNeeded -= iBytesIn;
|
|
iFramesNeeded -= iBytesIn / (sizeof(int16_t) * iChannels);
|
|
|
|
const int iSamplesIn = iBytesIn / sizeof(int16_t);
|
|
const int iFramesIn = iSamplesIn / iChannels;
|
|
|
|
float *pFloatBuf = (float *) alloca( iSamplesIn * sizeof(float) );
|
|
float *pFloatOut = (float *) alloca( iFrames * sizeof(float) );
|
|
for( int iChannel = 0; iChannel < iChannels; ++iChannel )
|
|
{
|
|
{
|
|
int16_t *pBufIn = pTmpBuf + iChannel;
|
|
float *pBufOut = pFloatBuf;
|
|
for( int i = 0; i < iSamplesIn; i += iChannels )
|
|
*(pBufOut++) = (float) pBufIn[i];
|
|
}
|
|
|
|
int iGotFrames = resamplers[iChannel]->Run( pFloatBuf, iFramesIn, pFloatOut, iFrames );
|
|
ASSERT( iGotFrames <= iFrames );
|
|
|
|
int16_t *pBufOut = pBuf + iChannel;
|
|
for( int i = 0; i < iGotFrames; ++i )
|
|
{
|
|
*pBufOut = int16_t(lrintf(clamp(pFloatOut[i], -32768, 32767)));
|
|
pBufOut += iChannels;
|
|
}
|
|
if( iChannel == 0 )
|
|
iFramesRead += iGotFrames;
|
|
}
|
|
}
|
|
|
|
return iFramesRead * iBytesPerFrame;
|
|
}
|
|
|
|
SoundReader *RageSoundReader_Resample_Good::Copy() const
|
|
{
|
|
SoundReader *pSource = m_pSource->Copy();
|
|
RageSoundReader_Resample_Good *ret = new RageSoundReader_Resample_Good;
|
|
|
|
for( size_t i = 0; i < resamplers.size(); ++i )
|
|
ret->resamplers.push_back( new RageSoundResampler_Polyphase(*resamplers[i]) );
|
|
ret->m_pSource = pSource;
|
|
ret->m_iSampleRate = m_iSampleRate;
|
|
|
|
return ret;
|
|
}
|
|
|
|
/*
|
|
* (c) 2006 Glenn Maynard
|
|
* All rights reserved.
|
|
*
|
|
* Permission is hereby granted, free of charge, to any person obtaining a
|
|
* copy of this software and associated documentation files (the
|
|
* "Software"), to deal in the Software without restriction, including
|
|
* without limitation the rights to use, copy, modify, merge, publish,
|
|
* distribute, and/or sell copies of the Software, and to permit persons to
|
|
* whom the Software is furnished to do so, provided that the above
|
|
* copyright notice(s) and this permission notice appear in all copies of
|
|
* the Software and that both the above copyright notice(s) and this
|
|
* permission notice appear in supporting documentation.
|
|
*
|
|
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
|
|
* OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
|
|
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OF
|
|
* THIRD PARTY RIGHTS. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR HOLDERS
|
|
* INCLUDED IN THIS NOTICE BE LIABLE FOR ANY CLAIM, OR ANY SPECIAL INDIRECT
|
|
* OR CONSEQUENTIAL DAMAGES, OR ANY DAMAGES WHATSOEVER RESULTING FROM LOSS
|
|
* OF USE, DATA OR PROFITS, WHETHER IN AN ACTION OF CONTRACT, NEGLIGENCE OR
|
|
* OTHER TORTIOUS ACTION, ARISING OUT OF OR IN CONNECTION WITH THE USE OR
|
|
* PERFORMANCE OF THIS SOFTWARE.
|
|
*/
|