From b3da00e081e36b74570a9a0417de471478d2d83f Mon Sep 17 00:00:00 2001 From: Flamefire Date: Sun, 27 Apr 2014 16:41:40 +0200 Subject: [PATCH] Use FFT for calculating AKF --- PitchTracker/FFT/FFT.cpp | 180 ++++++++++ PitchTracker/FFT/FFT.h | 65 ++++ PitchTracker/FFT/License.txt | 299 +++++++++++++++++ PitchTracker/FFT/RealFFTf.cpp | 388 ++++++++++++++++++++++ PitchTracker/FFT/RealFFTf.h | 23 ++ PitchTracker/PitchTracker.vcxproj | 4 + PitchTracker/PitchTracker.vcxproj.filters | 18 + PitchTracker/performous/libda/sample.hpp | 1 - PitchTracker/ptAKF.cpp | 27 +- PitchTracker/ptAKF.h | 13 +- 10 files changed, 1009 insertions(+), 9 deletions(-) create mode 100644 PitchTracker/FFT/FFT.cpp create mode 100644 PitchTracker/FFT/FFT.h create mode 100644 PitchTracker/FFT/License.txt create mode 100644 PitchTracker/FFT/RealFFTf.cpp create mode 100644 PitchTracker/FFT/RealFFTf.h diff --git a/PitchTracker/FFT/FFT.cpp b/PitchTracker/FFT/FFT.cpp new file mode 100644 index 00000000..39fdb52f --- /dev/null +++ b/PitchTracker/FFT/FFT.cpp @@ -0,0 +1,180 @@ +/********************************************************************** + + FFT.cpp + + Dominic Mazzoni + + September 2000 + +*******************************************************************//*! + +\file FFT.cpp +\brief Fast Fourier Transform routines. + + This file contains a few FFT routines, including a real-FFT + routine that is almost twice as fast as a normal complex FFT, + and a power spectrum routine when you know you don't care + about phase information. + + Some of this code was based on a free implementation of an FFT + by Don Cross, available on the web at: + + http://www.intersrv.com/~dcross/fft.html + + The basic algorithm for his code was based on Numerican Recipes + in Fortran. I optimized his code further by reducing array + accesses, caching the bit reversal table, and eliminating + float-to-double conversions, and I added the routines to + calculate a real FFT and a real power spectrum. + +*//*******************************************************************/ + +#include +#include +#include + +#include "FFT.h" + +#include "RealFFTf.h" + +void DeinitFFT() +{ + // Deallocate any unused RealFFTf tables + CleanupFFT(); +} + +/* + * Real Fast Fourier Transform + * + */ +void RealFFT(int NumSamples, float *RealIn, float *RealOut, float *ImagOut) +{ + // Remap to RealFFTf() function + int i; + HFFT hFFT = GetFFT(NumSamples); + float *pFFT = new float[NumSamples]; + // Copy the data into the processing buffer + for(i=0; iBitReversed[i] ]; + ImagOut[i]=pFFT[hFFT->BitReversed[i]+1]; + } + // Handle the (real-only) DC and Fs/2 bins + RealOut[0] = pFFT[0]; + RealOut[i] = pFFT[1]; + ImagOut[0] = ImagOut[i] = 0; + // Fill in the upper half using symmetry properties + for(i++ ; iBitReversed[i]]; + } + // Handle the (real-only) DC + Out[0] = pFFT[0]; + delete [] pFFT; + ReleaseFFT(hFFT); +} + +/* + * PowerSpectrum + * + * This function computes the same as RealFFT, above, but + * adds the squares of the real and imaginary part of each + * coefficient, extracting the power and throwing away the + * phase. + * + * For speed, it does not call RealFFT, but duplicates some + * of its code. + */ + +void PowerSpectrum(int NumSamples, float *In, float *Out) +{ + // Remap to RealFFTf() function + int i; + HFFT hFFT = GetFFT(NumSamples); + float *pFFT = new float[NumSamples]; + // Copy the data into the processing buffer + for(i=0; iBitReversed[i] ]*pFFT[hFFT->BitReversed[i] ]) + + (pFFT[hFFT->BitReversed[i]+1]*pFFT[hFFT->BitReversed[i]+1]); + } + // Handle the (real-only) DC and Fs/2 bins + Out[0] = pFFT[0]*pFFT[0]; + Out[i] = pFFT[1]*pFFT[1]; + delete [] pFFT; + ReleaseFFT(hFFT); +} \ No newline at end of file diff --git a/PitchTracker/FFT/FFT.h b/PitchTracker/FFT/FFT.h new file mode 100644 index 00000000..b8919e61 --- /dev/null +++ b/PitchTracker/FFT/FFT.h @@ -0,0 +1,65 @@ +/********************************************************************** + + FFT.h + + Dominic Mazzoni + + September 2000 + + This file contains a few FFT routines, including a real-FFT + routine that is almost twice as fast as a normal complex FFT, + and a power spectrum routine which is more convenient when + you know you don't care about phase information. It now also + contains a few basic windowing functions. + + Some of this code was based on a free implementation of an FFT + by Don Cross, available on the web at: + + http://www.intersrv.com/~dcross/fft.html + + The basic algorithm for his code was based on Numerical Recipes + in Fortran. I optimized his code further by reducing array + accesses, caching the bit reversal table, and eliminating + float-to-double conversions, and I added the routines to + calculate a real FFT and a real power spectrum. + + Note: all of these routines use single-precision floats. + I have found that in practice, floats work well until you + get above 8192 samples. If you need to do a larger FFT, + you need to use doubles. + +**********************************************************************/ + +/* + * This is the function you will use the most often. + * Given an array of floats, this will compute the power + * spectrum by doing a Real FFT and then computing the + * sum of the squares of the real and imaginary parts. + * Note that the output array is half the length of the + * input array, and that NumSamples must be a power of two. + */ + +void PowerSpectrum(int NumSamples, float *In, float *Out); + +/* + * Computes an FFT when the input data is real but you still + * want complex data as output. The output arrays are the + * same length as the input, but will be conjugate-symmetric + * NumSamples must be a power of two. + */ + +void RealFFT(int NumSamples, + float *RealIn, float *RealOut, float *ImagOut); + +/* + * Computes an Inverse FFT when the input data is conjugate symmetric + * so the output is purely real. NumSamples must be a power of + * two. + * Requires: EXPERIMENTAL_USE_REALFFTF + */ +void InverseRealFFT(int NumSamples, + float *RealIn, float *ImagIn, float *RealOut); + +void RealInverseRealFFT(int NumSamples, float *In, float *Out); + +void DeinitFFT(); diff --git a/PitchTracker/FFT/License.txt b/PitchTracker/FFT/License.txt new file mode 100644 index 00000000..c2e7eafa --- /dev/null +++ b/PitchTracker/FFT/License.txt @@ -0,0 +1,299 @@ +Code in this folder is based on Code from Audacity. +License of that code is below: + +------------------------------------------------------------------------ + +Note: Audacity is distributed under the terms of the GNU GPL. +This includes everything in the source code distribution +except code in the lib-src directory. Code in the lib-src +directory may be released under a different license +(which is GPL-compatible). 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See the +* GNU General Public License for more details. +* +* You should have received a copy of the GNU General Public License +* along with this program; if not, write to the Free Software +* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. +*/ + +#include +#include +#include + +#include "RealFFTf.h" +#ifdef EXPERIMENTAL_EQ_SSE_THREADED +#include "RealFFTf48x.h" +#endif + +#ifndef M_PI +#define M_PI 3.14159265358979323846 /* pi */ +#endif + +/* +* Initialize the Sine table and Twiddle pointers (bit-reversed pointers) +* for the FFT routine. +*/ +HFFT InitializeFFT(int fftlen) +{ + int i; + int temp; + int mask; + HFFT h; + + if((h=(HFFT)malloc(sizeof(FFTParam)))==NULL) + { + fprintf(stderr,"Error allocating memory for FFT\n"); + exit(8); + } + /* + * FFT size is only half the number of data points + * The full FFT output can be reconstructed from this FFT's output. + * (This optimization can be made since the data is real.) + */ + h->Points = fftlen/2; + + if((h->SinTable=(fft_type *)malloc(2*h->Points*sizeof(fft_type)))==NULL) + { + fprintf(stderr,"Error allocating memory for Sine table.\n"); + exit(8); + } + + if((h->BitReversed=(int *)malloc(h->Points*sizeof(int)))==NULL) + { + fprintf(stderr,"Error allocating memory for BitReversed.\n"); + exit(8); + } + + for(i=0;iPoints;i++) + { + temp=0; + for(mask=h->Points/2;mask>0;mask >>= 1) + temp=(temp >> 1) + (i&mask ? h->Points : 0); + + h->BitReversed[i]=temp; + } + + for(i=0;iPoints;i++) + { + h->SinTable[h->BitReversed[i] ]=(fft_type)-sin(2*M_PI*i/(2*h->Points)); + h->SinTable[h->BitReversed[i]+1]=(fft_type)-cos(2*M_PI*i/(2*h->Points)); + } + + return h; +} + +/* +* Free up the memory allotted for Sin table and Twiddle Pointers +*/ +void EndFFT(HFFT h) +{ + if(h->Points>0) { + free(h->BitReversed); + free(h->SinTable); + } + h->Points=0; + free(h); +} + +#define MAX_HFFT 10 +static HFFT hFFTArray[MAX_HFFT] = { NULL }; +static int nFFTLockCount[MAX_HFFT] = { 0 }; + +/* Get a handle to the FFT tables of the desired length */ +/* This version keeps common tables rather than allocating a new table every time */ +HFFT GetFFT(int fftlen) +{ + int h,n = fftlen/2; + for(h=0; (hPoints); h++); + if(hBitReversed[i] ] +* Imag_i = buffer[ h->BitReversed[i]+1 ] ) +* Input is in normal order. +* +* Output buffer[0] is the DC bin, and output buffer[1] is the Fs/2 bin +* - this can be done because both values will always be real only +* - this allows us to not have to allocate an extra complex value for the Fs/2 bin +* +* Note: The scaling on this is done according to the standard FFT definition, +* so a unit amplitude DC signal will output an amplitude of (N) +* (Older revisions would progressively scale the input, so the output +* values would be similar in amplitude to the input values, which is +* good when using fixed point arithmetic) +*/ +void RealFFTf(fft_type *buffer,HFFT h) +{ + fft_type *A,*B; + fft_type *sptr; + fft_type *endptr1,*endptr2; + int *br1,*br2; + fft_type HRplus,HRminus,HIplus,HIminus; + fft_type v1,v2,sin,cos; + + int ButterfliesPerGroup=h->Points/2; + + /* + * Butterfly: + * Ain-----Aout + * \ / + * / \ + * Bin-----Bout + */ + + endptr1=buffer+h->Points*2; + + while(ButterfliesPerGroup>0) + { + A=buffer; + B=buffer+ButterfliesPerGroup*2; + sptr=h->SinTable; + + while(A>= 1; + } + /* Massage output to get the output for a real input sequence. */ + br1=h->BitReversed+1; + br2=h->BitReversed+h->Points-1; + + while(br1SinTable[*br1]; + cos=h->SinTable[*br1+1]; + A=buffer+*br1; + B=buffer+*br2; + HRplus = (HRminus = *A - *B ) + (*B * 2); + HIplus = (HIminus = *(A+1) - *(B+1)) + (*(B+1) * 2); + v1 = (sin*HRminus - cos*HIplus); + v2 = (cos*HRminus + sin*HIplus); + *A = (HRplus + v1) * (fft_type)0.5; + *B = *A - v1; + *(A+1) = (HIminus + v2) * (fft_type)0.5; + *(B+1) = *(A+1) - HIminus; + + br1++; + br2--; + } + /* Handle the center bin (just need a conjugate) */ + A=buffer+*br1+1; + *A=-*A; + /* Handle DC bin separately - and ignore the Fs/2 bin + buffer[0]+=buffer[1]; + buffer[1]=(fft_type)0;*/ + /* Handle DC and Fs/2 bins separately */ + /* Put the Fs/2 value into the imaginary part of the DC bin */ + v1=buffer[0]-buffer[1]; + buffer[0]+=buffer[1]; + buffer[1]=v1; +} + + +/* Description: This routine performs an inverse FFT to real data. +* This code is for floating point data. +* +* Note: Output is BIT-REVERSED! so you must use the BitReversed to +* get legible output, (i.e. wave[2*i] = buffer[ BitReversed[i] ] +* wave[2*i+1] = buffer[ BitReversed[i]+1 ] ) +* Input is in normal order, interleaved (real,imaginary) complex data +* You must call InitializeFFT(fftlen) first to initialize some buffers! +* +* Input buffer[0] is the DC bin, and input buffer[1] is the Fs/2 bin +* - this can be done because both values will always be real only +* - this allows us to not have to allocate an extra complex value for the Fs/2 bin +* +* Note: The scaling on this is done according to the standard FFT definition, +* so a unit amplitude DC signal will output an amplitude of (N) +* (Older revisions would progressively scale the input, so the output +* values would be similar in amplitude to the input values, which is +* good when using fixed point arithmetic) +*/ +void InverseRealFFTf(fft_type *buffer,HFFT h) +{ + fft_type *A,*B; + fft_type *sptr; + fft_type *endptr1,*endptr2; + int *br1; + fft_type HRplus,HRminus,HIplus,HIminus; + fft_type v1,v2,sin,cos; + + int ButterfliesPerGroup=h->Points/2; + + /* Massage input to get the input for a real output sequence. */ + A=buffer+2; + B=buffer+h->Points*2-2; + br1=h->BitReversed+1; + while(ASinTable[*br1]; + cos=h->SinTable[*br1+1]; + HRplus = (HRminus = *A - *B ) + (*B * 2); + HIplus = (HIminus = *(A+1) - *(B+1)) + (*(B+1) * 2); + v1 = (sin*HRminus + cos*HIplus); + v2 = (cos*HRminus - sin*HIplus); + *A = (HRplus + v1) * (fft_type)0.5; + *B = *A - v1; + *(A+1) = (HIminus - v2) * (fft_type)0.5; + *(B+1) = *(A+1) - HIminus; + + A+=2; + B-=2; + br1++; + } + /* Handle center bin (just need conjugate) */ + *(A+1)=-*(A+1); + /* Handle DC bin separately - this ignores any Fs/2 component + buffer[1]=buffer[0]=buffer[0]/2;*/ + /* Handle DC and Fs/2 bins specially */ + /* The DC bin is passed in as the real part of the DC complex value */ + /* The Fs/2 bin is passed in as the imaginary part of the DC complex value */ + /* (v1+v2) = buffer[0] == the DC component */ + /* (v1-v2) = buffer[1] == the Fs/2 component */ + v1=0.5f*(buffer[0]+buffer[1]); + v2=0.5f*(buffer[0]-buffer[1]); + buffer[0]=v1; + buffer[1]=v2; + + /* + * Butterfly: + * Ain-----Aout + * \ / + * / \ + * Bin-----Bout + */ + + endptr1=buffer+h->Points*2; + + while(ButterfliesPerGroup>0) + { + A=buffer; + B=buffer+ButterfliesPerGroup*2; + sptr=h->SinTable; + + while(A>= 1; + } +} + +void ReorderToFreq(HFFT hFFT, fft_type *buffer, fft_type *RealOut, fft_type *ImagOut) +{ + // Copy the data into the real and imaginary outputs + for(int i=1;iPoints;i++) { + RealOut[i]=buffer[hFFT->BitReversed[i] ]; + ImagOut[i]=buffer[hFFT->BitReversed[i]+1]; + } + RealOut[0] = buffer[0]; // DC component + ImagOut[0] = 0; + RealOut[hFFT->Points] = buffer[1]; // Fs/2 component + ImagOut[hFFT->Points] = 0; +} + +void ReorderToTime(HFFT hFFT, fft_type *buffer, fft_type *TimeOut) +{ + // Copy the data into the real outputs + for(int i=0;iPoints;i++) { + TimeOut[i*2 ]=buffer[hFFT->BitReversed[i] ]; + TimeOut[i*2+1]=buffer[hFFT->BitReversed[i]+1]; + } +} diff --git a/PitchTracker/FFT/RealFFTf.h b/PitchTracker/FFT/RealFFTf.h new file mode 100644 index 00000000..f46119ca --- /dev/null +++ b/PitchTracker/FFT/RealFFTf.h @@ -0,0 +1,23 @@ +#ifndef __realfftf_h +#define __realfftf_h + +#define fft_type float +typedef struct FFTParamType { + int *BitReversed; + fft_type *SinTable; + int Points; +} FFTParam; +#define HFFT FFTParam * + +HFFT InitializeFFT(int); +void EndFFT(HFFT); +HFFT GetFFT(int); +void ReleaseFFT(HFFT); +void CleanupFFT(); +void RealFFTf(fft_type *,HFFT); +void InverseRealFFTf(fft_type *,HFFT); +void ReorderToTime(HFFT hFFT, fft_type *buffer, fft_type *TimeOut); +void ReorderToFreq(HFFT hFFT, fft_type *buffer, fft_type *RealOut, fft_type *ImagOut); + +#endif + diff --git a/PitchTracker/PitchTracker.vcxproj b/PitchTracker/PitchTracker.vcxproj index 1b729255..65ce0ada 100644 --- a/PitchTracker/PitchTracker.vcxproj +++ b/PitchTracker/PitchTracker.vcxproj @@ -174,6 +174,8 @@ + + @@ -183,6 +185,8 @@ + + diff --git a/PitchTracker/PitchTracker.vcxproj.filters b/PitchTracker/PitchTracker.vcxproj.filters index 6e3d94f4..81447394 100644 --- a/PitchTracker/PitchTracker.vcxproj.filters +++ b/PitchTracker/PitchTracker.vcxproj.filters @@ -28,6 +28,12 @@ {2a09cee1-51b7-4c24-aa9f-32bec53c3a36} + + {498f64e2-ff83-41fd-be25-c613750f0e34} + + + {0791b62f-ef8d-40e2-b67e-a6d9dead8720} + @@ -51,6 +57,12 @@ Quelldateien + + Quelldateien\FFT + + + Quelldateien\FFT + @@ -77,6 +89,12 @@ Headerdateien\dywapitchtrack + + Headerdateien\FFT + + + Headerdateien\FFT + diff --git a/PitchTracker/performous/libda/sample.hpp b/PitchTracker/performous/libda/sample.hpp index a09bbcb6..da55bda7 100644 --- a/PitchTracker/performous/libda/sample.hpp +++ b/PitchTracker/performous/libda/sample.hpp @@ -1,7 +1,6 @@ #pragma once #include -#include "../compatibility.h" /** * @file sample.hpp Sample format definition and format conversions. diff --git a/PitchTracker/ptAKF.cpp b/PitchTracker/ptAKF.cpp index 335bcde6..7936cad3 100644 --- a/PitchTracker/ptAKF.cpp +++ b/PitchTracker/ptAKF.cpp @@ -17,8 +17,13 @@ #include "ptAKF.h" #include +#ifdef USE_FFT +#include "FFT/FFT.h" +#endif + static constexpr double BaseToneFrequency = 65.4064; // lowest (half-)tone to analyze (C2 = 65.4064 Hz) static constexpr double HalftoneBase = 1.05946309436; // 2^(1/12) -> HalftoneBase^12 = 2 (one octave) + int PtAKF::_InitCount = 0; float* restrict PtAKF::_SamplesPerPeriodPerTone = NULL; float* restrict PtAKF::_SamplesPerPeriodPerToneFine = NULL; @@ -41,9 +46,9 @@ PtAKF::PtAKF(unsigned step){ _SamplesPerPeriodPerToneFine[2 * toneIndex + 1] = static_cast(44100.0 / GetFrequencyFromTone(toneIndex + 1./3.)); // samples for a bit above exact frequency } _Window = new float[_SampleCt]; - double cosMult = 2. * M_PI / (_SampleCt - 1.); //To simplify and speed up + double cosMult = 2. * M_PI / (2 * _SampleCt - 1.); //To simplify and speed up; 2 * _SampleCt because we extend the data by a factor of 2 for (int i = 0; i < _SampleCt; i++) { - _Window[i] = static_cast(0.64 - 0.36 * cos(i * cosMult)); + _Window[i] = static_cast(0.54 - 0.46 * cos(i * cosMult)); } } _Step = step; @@ -149,12 +154,20 @@ int PtAKF::_GetNote(float samples[_SampleCt], float* restrict maxVolume, float w if(maxVolumeL < _VolTreshold) return -1; - float samplesWindowed[_SampleCt]; + float samplesWindowed[_SampleCt * 2]; for(int i = 0; i < _SampleCt; i++){ samplesWindowed[i] = samples[i] * _Window[i]; } +#ifdef USE_FFT + for(int i = _SampleCt; i < _SampleCt * 2; i++){ + samplesWindowed[i] = 0.f; + } + float samplesFFT[_SampleCt+1]; // +1 for middle value! + PowerSpectrum(_SampleCt*2, samplesWindowed, samplesFFT); + RealInverseRealFFT(_SampleCt*2, samplesFFT, _AKFValues); +#endif // Now analyze the samples and get peaks at the most appropriate tones //Attention: We have a peak at lag 0 that might stretch that far, that we detect a wrong "peak" at _MaxHalfTone @@ -163,6 +176,7 @@ int PtAKF::_GetNote(float samples[_SampleCt], float* restrict maxVolume, float w float lastWeight = 1.f; float maxWeight = 0.f; + //TODO: When using FFT this can be speed up a lot by only checking around the AKF peaks #pragma omp parallel for schedule(static) for (int toneIndex = 0; toneIndex <= _MaxHalfTone; toneIndex++){ float curWeight = _AnalyzeByTone(samples, samplesWindowed, toneIndex); @@ -307,6 +321,10 @@ float PtAKF::_AKFBySampleCt(float samples[_SampleCt], float samplesPerPeriodD){ int samplesPerPeriod = static_cast(samplesPerPeriodD); float fHigh = samplesPerPeriodD - samplesPerPeriod; float fLow = 1.0f - fHigh; +#ifdef USE_FFT + float akf2 = _AKFValues[samplesPerPeriod] * fLow + _AKFValues[samplesPerPeriod+1] * fHigh; + return akf2 / (_SampleCt * _SampleCt); +#else float accumDist = 0; // accumulated distances @@ -322,10 +340,11 @@ float PtAKF::_AKFBySampleCt(float samples[_SampleCt], float samplesPerPeriodD){ } return accumDist / _SampleCt; +#endif } float PtAKF::_AMDFByTone(float samples[_SampleCt], int toneIndex){ - return _AKFBySampleCt(samples, _SamplesPerPeriodPerTone[toneIndex]); + return _AMDFBySampleCt(samples, _SamplesPerPeriodPerTone[toneIndex]); } float PtAKF::_AMDFBySampleCt(float samples[_SampleCt], float samplesPerPeriodD){ diff --git a/PitchTracker/ptAKF.h b/PitchTracker/ptAKF.h index b5d9a2b6..00f8aeb5 100644 --- a/PitchTracker/ptAKF.h +++ b/PitchTracker/ptAKF.h @@ -13,6 +13,8 @@ # endif #endif +#define USE_FFT + struct SPeak{ int toneIndex; float weight; @@ -55,14 +57,17 @@ private: float _LastMaxVol; int _LastTones[_SmoothCt]; int _LastToneIndex; +#ifdef USE_FFT + float _AKFValues[_SampleCt*2]; +#endif int _GetNote(float samples[_SampleCt], float* restrict maxVolume, float weights[_MaxHalfTone+1]); int _GetSmoothTone(); - static float _AnalyzeBySampleCt(float samples[_SampleCt], float samplesWindowed[_SampleCt], float samplesPerPeriodD); - static inline float _AnalyzeByTone(float samples[_SampleCt], float samplesWindowed[_SampleCt], int toneIndex); - static float _AKFBySampleCt(float samples[_SampleCt], float samplesPerPeriodD); - static inline float _AKFByTone(float samples[_SampleCt], int toneIndex); + float _AnalyzeBySampleCt(float samples[_SampleCt], float samplesWindowed[_SampleCt], float samplesPerPeriodD); + inline float _AnalyzeByTone(float samples[_SampleCt], float samplesWindowed[_SampleCt], int toneIndex); + float _AKFBySampleCt(float samples[_SampleCt], float samplesPerPeriodD); + inline float _AKFByTone(float samples[_SampleCt], int toneIndex); static float _AMDFBySampleCt(float samples[_SampleCt], float samplesPerPeriodD); static inline float _AMDFByTone(float samples[_SampleCt], int toneIndex); static inline void InitPeaks(SPeak peaks[_MaxPeaks]);