diff options
Diffstat (limited to 'ofxMaxim/ofxMaximiPhoneExample')
18 files changed, 3316 insertions, 0 deletions
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+ defaultConfigurationIsVisible = 0; + defaultConfigurationName = Release; + }; + C01FCF4E08A954540054247B /* Build configuration list for PBXProject "iPhoneEmptyExample" */ = { + isa = XCConfigurationList; + buildConfigurations = ( + C01FCF4F08A954540054247B /* Debug */, + C01FCF5008A954540054247B /* Release */, + E49BCC2F12554938009F19D0 /* ReleaseArmv7 */, + ); + defaultConfigurationIsVisible = 0; + defaultConfigurationName = Release; + }; +/* End XCConfigurationList section */ + }; + rootObject = 29B97313FDCFA39411CA2CEA /* Project object */; +} diff --git a/ofxMaxim/ofxMaximiPhoneExample/iPhone_Prefix.pch b/ofxMaxim/ofxMaximiPhoneExample/iPhone_Prefix.pch new file mode 100644 index 0000000..fd3401c --- /dev/null +++ b/ofxMaxim/ofxMaximiPhoneExample/iPhone_Prefix.pch @@ -0,0 +1,8 @@ +// +// Prefix header for all source files of the 'iPhone' target in the 'iPhone' project +// + +#ifdef __OBJC__ + #import <Foundation/Foundation.h> + #import <UIKit/UIKit.h> +#endif diff --git a/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/install.xml b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/install.xml new file mode 100644 index 0000000..58923be --- /dev/null +++ b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/install.xml @@ -0,0 +1,35 @@ +<install> + <version>0.1</version> + <author>Goldsmiths Creative Computing</author> + <url>http://www.maximilian.strangeloop.co.uk/</url> + <requires></requires> + + + <add> + <!-- ===================================================================== --> + <!-- ========================== add per project using this addon ========= --> + <!-- ===================================================================== --> + + <src> + + <folder name="addons/ofxMaxim/src"> + <file>../../../addons/ofxMaxim/src/ofxMaxim.h</file> + </folder> + + <folder name="addons/ofxMaxim/libs"> + <file>../../../addons/ofxMaxim/libs/fft.h</file> + <file>../../../addons/ofxMaxim/libs/fft.cpp</file> + <file>../../../addons/ofxMaxim/libs/fftOctaveAnalyzer.h</file> + <file>../../../addons/ofxMaxim/libs/fftOctaveAnalyzer.cpp</file> + <file>../../../addons/ofxMaxim/libs/maximilian.h</file> + <file>../../../addons/ofxMaxim/libs/maximilian.cpp</file> + </folder> + + </src> + + <include> + <path>../../../addons/ofxMaxim/src</path> + <path>../../../addons/ofxMaxim/libs</path> + </include> + </add> +</install>
\ No newline at end of file diff --git a/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/fft.cpp b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/fft.cpp new file mode 100644 index 0000000..e19498d --- /dev/null +++ b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/fft.cpp @@ -0,0 +1,592 @@ +/* + * maximilian + * platform independent synthesis library using portaudio or rtaudio + * + * Created by Mick Grierson on 29/12/2009. + * Copyright 2009 Mick Grierson & Strangeloop Limited. All rights reserved. + * Thanks to the Goldsmiths Creative Computing Team. + * Special thanks to Arturo Castro for the PortAudio implementation. + * + * 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, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following + * conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * 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. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + * + */ +/* + + fft.cpp + + Based on K+R Numerical recipes in C and some other stuff hacked about. + + */ + +#include "fft.h" +#include <stdlib.h> +#include <stdio.h> +#include <math.h> +#include <string.h> + +int **gFFTBitTable = NULL; +const int MaxFastBits = 16; + +int IsPowerOfTwo(int x) +{ + if (x < 2) + return false; + + if (x & (x - 1)) + return false; + + return true; +} + +int NumberOfBitsNeeded(int PowerOfTwo) +{ + int i; + + if (PowerOfTwo < 2) { + fprintf(stderr, "Error: FFT called with size %d\n", PowerOfTwo); + exit(1); + } + + for (i = 0;; i++) + if (PowerOfTwo & (1 << i)) + return i; +} + +int ReverseBits(int index, int NumBits) +{ + int i, rev; + + for (i = rev = 0; i < NumBits; i++) { + rev = (rev << 1) | (index & 1); + index >>= 1; + } + + return rev; +} + +void InitFFT() +{ +// gFFTBitTable = new int *[MaxFastBits]; + //use malloc for 16 byte alignment + gFFTBitTable = (int**) malloc(MaxFastBits * sizeof(int*)); + + int len = 2; + for (int b = 1; b <= MaxFastBits; b++) { + +// gFFTBitTable[b - 1] = new int[len]; + gFFTBitTable[b - 1] = (int*) malloc(len * sizeof(int)); + + for (int i = 0; i < len; i++) + gFFTBitTable[b - 1][i] = ReverseBits(i, b); + + len <<= 1; + } +} + +inline int FastReverseBits(int i, int NumBits) +{ + if (NumBits <= MaxFastBits) + return gFFTBitTable[NumBits - 1][i]; + else + return ReverseBits(i, NumBits); +} + +/* + * Complex Fast Fourier Transform + */ + +void FFT(int NumSamples, + bool InverseTransform, + float *RealIn, float *ImagIn, float *RealOut, float *ImagOut) +{ + int NumBits; /* Number of bits needed to store indices */ + int i, j, k, n; + int BlockSize, BlockEnd; + + double angle_numerator = 2.0 * M_PI; + float tr, ti; /* temp real, temp imaginary */ + + if (!IsPowerOfTwo(NumSamples)) { + fprintf(stderr, "%d is not a power of two\n", NumSamples); + exit(1); + } + + if (!gFFTBitTable) + InitFFT(); + + if (InverseTransform) + angle_numerator = -angle_numerator; + + NumBits = NumberOfBitsNeeded(NumSamples); + + /* + ** Do simultaneous data copy and bit-reversal ordering into outputs... + */ + + for (i = 0; i < NumSamples; i++) { + j = FastReverseBits(i, NumBits); + RealOut[j] = RealIn[i]; + ImagOut[j] = (ImagIn == NULL) ? 0.0 : ImagIn[i]; + } + + /* + ** Do the FFT itself... + */ + + BlockEnd = 1; + for (BlockSize = 2; BlockSize <= NumSamples; BlockSize <<= 1) { + + double delta_angle = angle_numerator / (double) BlockSize; + + float sm2 = sin(-2 * delta_angle); + float sm1 = sin(-delta_angle); + float cm2 = cos(-2 * delta_angle); + float cm1 = cos(-delta_angle); + float w = 2 * cm1; + float ar0, ar1, ar2, ai0, ai1, ai2; + + for (i = 0; i < NumSamples; i += BlockSize) { + ar2 = cm2; + ar1 = cm1; + + ai2 = sm2; + ai1 = sm1; + + for (j = i, n = 0; n < BlockEnd; j++, n++) { + ar0 = w * ar1 - ar2; + ar2 = ar1; + ar1 = ar0; + + ai0 = w * ai1 - ai2; + ai2 = ai1; + ai1 = ai0; + + k = j + BlockEnd; + tr = ar0 * RealOut[k] - ai0 * ImagOut[k]; + ti = ar0 * ImagOut[k] + ai0 * RealOut[k]; + + RealOut[k] = RealOut[j] - tr; + ImagOut[k] = ImagOut[j] - ti; + + RealOut[j] += tr; + ImagOut[j] += ti; + } + } + + BlockEnd = BlockSize; + } + + /* + ** Need to normalize if inverse transform... + */ + + if (InverseTransform) { + float denom = (float) NumSamples; + + for (i = 0; i < NumSamples; i++) { + RealOut[i] /= denom; + ImagOut[i] /= denom; + } + } +} + +/* + * Real Fast Fourier Transform + * + * This function was based on the code in Numerical Recipes in C. + * In Num. Rec., the inner loop is based on a single 1-based array + * of interleaved real and imaginary numbers. Because we have two + * separate zero-based arrays, our indices are quite different. + * Here is the correspondence between Num. Rec. indices and our indices: + * + * i1 <-> real[i] + * i2 <-> imag[i] + * i3 <-> real[n/2-i] + * i4 <-> imag[n/2-i] + */ + +void RealFFT(int NumSamples, float *RealIn, float *RealOut, float *ImagOut) +{ + int Half = NumSamples / 2; + int i; + + float theta = M_PI / Half; + + float *tmpReal = (float*) malloc(Half * sizeof(float)); + float *tmpImag = (float*) malloc(Half * sizeof(float)); + + for (i = 0; i < Half; i++) { + tmpReal[i] = RealIn[2 * i]; + tmpImag[i] = RealIn[2 * i + 1]; + } + + FFT(Half, 0, tmpReal, tmpImag, RealOut, ImagOut); + + float wtemp = float (sin(0.5 * theta)); + + float wpr = -2.0 * wtemp * wtemp; + float wpi = float (sin(theta)); + float wr = 1.0 + wpr; + float wi = wpi; + + int i3; + + float h1r, h1i, h2r, h2i; + + for (i = 1; i < Half / 2; i++) { + + i3 = Half - i; + + h1r = 0.5 * (RealOut[i] + RealOut[i3]); + h1i = 0.5 * (ImagOut[i] - ImagOut[i3]); + h2r = 0.5 * (ImagOut[i] + ImagOut[i3]); + h2i = -0.5 * (RealOut[i] - RealOut[i3]); + + RealOut[i] = h1r + wr * h2r - wi * h2i; + ImagOut[i] = h1i + wr * h2i + wi * h2r; + RealOut[i3] = h1r - wr * h2r + wi * h2i; + ImagOut[i3] = -h1i + wr * h2i + wi * h2r; + + wr = (wtemp = wr) * wpr - wi * wpi + wr; + wi = wi * wpr + wtemp * wpi + wi; + } + + RealOut[0] = (h1r = RealOut[0]) + ImagOut[0]; + ImagOut[0] = h1r - ImagOut[0]; + + delete[]tmpReal; + delete[]tmpImag; +} + +/* + * 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) +{ + int Half = NumSamples / 2; + int i; + + float theta = M_PI / Half; + + float *tmpReal = new float[Half]; + float *tmpImag = new float[Half]; + float *RealOut = new float[Half]; + float *ImagOut = new float[Half]; + + for (i = 0; i < Half; i++) { + tmpReal[i] = In[2 * i]; + tmpImag[i] = In[2 * i + 1]; + } + + FFT(Half, 0, tmpReal, tmpImag, RealOut, ImagOut); + + float wtemp = float (sin(0.5 * theta)); + + float wpr = -2.0 * wtemp * wtemp; + float wpi = float (sin(theta)); + float wr = 1.0 + wpr; + float wi = wpi; + + int i3; + + float h1r, h1i, h2r, h2i, rt, it; + //float total=0; + + for (i = 1; i < Half / 2; i++) { + + i3 = Half - i; + + h1r = 0.5 * (RealOut[i] + RealOut[i3]); + h1i = 0.5 * (ImagOut[i] - ImagOut[i3]); + h2r = 0.5 * (ImagOut[i] + ImagOut[i3]); + h2i = -0.5 * (RealOut[i] - RealOut[i3]); + + rt = h1r + wr * h2r - wi * h2i; //printf("Realout%i = %f",i,rt);total+=fabs(rt); + it = h1i + wr * h2i + wi * h2r; // printf(" Imageout%i = %f\n",i,it); + + Out[i] = rt * rt + it * it; + + rt = h1r - wr * h2r + wi * h2i; + it = -h1i + wr * h2i + wi * h2r; + + Out[i3] = rt * rt + it * it; + + wr = (wtemp = wr) * wpr - wi * wpi + wr; + wi = wi * wpr + wtemp * wpi + wi; + } + //printf("total = %f\n",total); + rt = (h1r = RealOut[0]) + ImagOut[0]; + it = h1r - ImagOut[0]; + Out[0] = rt * rt + it * it; + + rt = RealOut[Half / 2]; + it = ImagOut[Half / 2]; + Out[Half / 2] = rt * rt + it * it; + + delete[]tmpReal; + delete[]tmpImag; + delete[]RealOut; + delete[]ImagOut; +} + +/* + * Windowing Functions + */ + +int NumWindowFuncs() +{ + return 4; +} + +char *WindowFuncName(int whichFunction) +{ + switch (whichFunction) { + default: + case 0: + return "Rectangular"; + case 1: + return "Bartlett"; + case 2: + return "Hamming"; + case 3: + return "Hanning"; + } +} + +void WindowFunc(int whichFunction, int NumSamples, float *in) +{ + int i; + + if (whichFunction == 1) { + // Bartlett (triangular) window + for (i = 0; i < NumSamples / 2; i++) { + in[i] *= (i / (float) (NumSamples / 2)); + in[i + (NumSamples / 2)] *= + (1.0 - (i / (float) (NumSamples / 2))); + } + } + + if (whichFunction == 2) { + // Hamming + for (i = 0; i < NumSamples; i++) + in[i] *= 0.54 - 0.46 * cos(2 * M_PI * i / (NumSamples - 1)); + } + + if (whichFunction == 3) { + // Hanning + for (i = 0; i < NumSamples; i++) + in[i] *= 0.50 - 0.50 * cos(2 * M_PI * i / (NumSamples - 1)); + } +} + +void fft::genWindow(int whichFunction, int NumSamples, float *window) +{ + int i; + + if (whichFunction == 1) { + // Bartlett (triangular) window + for (i = 0; i < NumSamples / 2; i++) { + window[i] = (i / (float) (NumSamples / 2)); + window[i + (NumSamples / 2)] = + (1.0 - (i / (float) (NumSamples / 2))); + } + } + + if (whichFunction == 2) { + // Hamming + for (i = 0; i < NumSamples; i++) + window[i] = 0.54 - 0.46 * cos(2 * M_PI * i / (NumSamples - 1)); + } + + if (whichFunction == 3) { + // Hanning + for (i = 0; i < NumSamples; i++) + window[i] = 0.50 - 0.50 * cos(2 * M_PI * i / (NumSamples - 1)); + } +} + +/* constructor */ +fft::fft(int fftSize) { + n = fftSize; + half = fftSize / 2; + //use malloc for 16 byte alignment + in_real = (float *) malloc(n * sizeof(float)); + in_img = (float *) malloc(n * sizeof(float)); + out_real = (float *) malloc(n * sizeof(float)); + out_img = (float *) malloc(n * sizeof(float)); + +#ifdef __APPLE_CC__ + log2n = log2(n); + A.realp = (float *) malloc(half * sizeof(float)); + A.imagp = (float *) malloc(half * sizeof(float)); + setupReal = vDSP_create_fftsetup(log2n, FFT_RADIX2); + if (setupReal == NULL) { + printf("\nFFT_Setup failed to allocate enough memory for" + "the real FFT.\n"); + } + polar = (float *) malloc(n * sizeof(float)); +#endif +} + +/* destructor */ +fft::~fft() { + delete[] in_real, out_real, in_img, out_img; +#ifdef __APPLE_CC__ + vDSP_destroy_fftsetup(setupReal); + delete[] A.realp; + delete[] A.imagp; + delete[] polar; +#endif + +} + +/* Calculate the power spectrum */ +void fft::powerSpectrum(int start, float *data, float *window, float *magnitude,float *phase, float *power, float *avg_power) { + int i; + float total_power = 0.0f; + + + //windowing + for (i = 0; i < n; i++) { + in_real[i] = data[start + i] * window[i]; + } + + + RealFFT(n, in_real, out_real, out_img); + + for (i = 0; i < half; i++) { + /* compute power */ + power[i] = out_real[i]*out_real[i] + out_img[i]*out_img[i]; + total_power += power[i]; + /* compute magnitude and phase */ + magnitude[i] = 2.0*sqrt(power[i]); + + if (magnitude[i] < 0.000001){ // less than 0.1 nV + magnitude[i] = 0; // out of range + } else { + magnitude[i] = 20.0*log10(magnitude[i] + 1); // get to to db scale + } + phase[i] = atan2(out_img[i],out_real[i]); + } + /* calculate average power */ + *(avg_power) = total_power / (float) half; + +} + +#ifdef __APPLE_CC__ + +/* Calculate the power spectrum */ +void fft::powerSpectrum_vdsp(int start, float *data, float *window, float *magnitude,float *phase) { + + uint32_t i; + + //multiply by window + vDSP_vmul(data, 1, window, 1, in_real, 1, n); + + //convert to split complex format - evens and odds + vDSP_ctoz((COMPLEX *) in_real, 2, &A, 1, half); + + + //calc fft + vDSP_fft_zrip(setupReal, &A, 1, log2n, FFT_FORWARD); + + //scale by 2 (see vDSP docs) + static float scale=0.5; + vDSP_vsmul(A.realp, 1, &scale, A.realp, 1, half); + vDSP_vsmul(A.imagp, 1, &scale, A.imagp, 1, half); + + //back to split complex format + vDSP_ztoc(&A, 1, (COMPLEX*) out_real, 2, half); + + //convert to polar + vDSP_polar(out_real, 2, polar, 2, half); + + for (i = 0; i < half; i++) { + magnitude[i]=polar[2*i]+1.0; + phase[i]=polar[2*i + 1]; + } + float ref = 1.0; + vDSP_vdbcon(magnitude, 1, &ref, magnitude, 1, half, 1); + + + +} + +#endif + +void fft::inversePowerSpectrum(int start, float *finalOut, float *window, float *magnitude,float *phase) { + int i; + + /* get real and imag part */ + for (i = 0; i < half; i++) { + float mag = pow(10.0, magnitude[i] / 20.0) - 1.0; + in_real[i] = mag *cos(phase[i]); + in_img[i] = mag *sin(phase[i]); + } + + /* zero negative frequencies */ + memset(in_real+half, half, 0.0); + memset(in_img+half, half, 0.0); + + FFT(n, 1, in_real, in_img, out_real, out_img); // second parameter indicates inverse transform + + for (i = 0; i < n; i++) { + finalOut[start + i] += out_real[i] * window[i] + ; + } + +} + + +#ifdef __APPLE_CC__ +void fft::inversePowerSpectrum_vdsp(int start, float *finalOut, float *window, float *magnitude,float *phase) { + uint32_t i; + + for (i = 0; i < half; i++) { + polar[2*i] = pow(10.0, magnitude[i] / 20.0) - 1.0; + polar[2*i + 1] = phase[i]; + } + + vDSP_rect(polar, 2, in_real, 2, half); + + vDSP_ctoz((COMPLEX*) in_real, 2, &A, 1, half); + vDSP_fft_zrip(setupReal, &A, 1, log2n, FFT_INVERSE); + vDSP_ztoc(&A, 1, (COMPLEX*) out_real, 2, half); + + static float scale = 1./(2. * n); + vDSP_vsmul(out_real, 1, &scale, out_real, 1, n); + + //multiply by window + vDSP_vmul(out_real, 1, window, 1, finalOut, 1, n); + +} + +#endif diff --git a/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/fft.h b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/fft.h new file mode 100644 index 0000000..77aa777 --- /dev/null +++ b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/fft.h @@ -0,0 +1,76 @@ +/* + * maximilian + * platform independent synthesis library using portaudio or rtaudio + * + * Created by Mick Grierson on 29/12/2009. + * Copyright 2009 Mick Grierson & Strangeloop Limited. All rights reserved. + * Thanks to the Goldsmiths Creative Computing Team. + * Special thanks to Arturo Castro for the PortAudio implementation. + * + * 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, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following + * conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * 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. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + * + */ + +#ifndef _FFT +#define _FFT + +#ifndef M_PI +#define M_PI 3.14159265358979323846 /* pi */ +#endif + +#ifdef __APPLE_CC__ +#include <Accelerate/Accelerate.h> +#endif + + +class fft { + +public: + + fft(int fftSize); + ~fft(); + + int n; //fftSize + int half; //halfFFTSize + + float *in_real, *out_real, *in_img, *out_img; + +#ifdef __APPLE_CC__ + int log2n; //log2(n); + FFTSetup setupReal; + COMPLEX_SPLIT A; + float *polar; + void powerSpectrum_vdsp(int start, float *data, float *window, float *magnitude,float *phase); + void inversePowerSpectrum_vdsp(int start, float *finalOut, float *window, float *magnitude,float *phase); +#endif; + + /* Calculate the power spectrum */ + void powerSpectrum(int start, float *data, float *window, float *magnitude, float *phase, float *power, float *avg_power); + /* ... the inverse */ + void inversePowerSpectrum(int start, float *finalOut, float *window, float *magnitude,float *phase); + + static void genWindow(int whichFunction, int NumSamples, float *window); + +}; + + +#endif diff --git a/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/maxiFFT.cpp b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/maxiFFT.cpp new file mode 100644 index 0000000..eee8ee8 --- /dev/null +++ b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/maxiFFT.cpp @@ -0,0 +1,257 @@ +/* + * maximilian + * platform independent synthesis library using portaudio or rtaudio + * + * Created by Mick Grierson on 29/12/2009. + * Copyright 2009 Mick Grierson & Strangeloop Limited. All rights reserved. + * Thanks to the Goldsmiths Creative Computing Team. + * Special thanks to Arturo Castro for the PortAudio implementation. + * + * 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, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following + * conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * 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. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + * + */ + +#include "maxiFFT.h" +#include <iostream> +#include "math.h" + +using namespace std; + + +////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// +//F F T +////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + +void maxiFFT::setup(int _fftSize, int _windowSize, int _hopSize) { + _fft = new fft(_fftSize); + fftSize = _fftSize; + windowSize = _windowSize; + bins = fftSize / 2; + hopSize = _hopSize; + buffer = (float *) malloc(fftSize * sizeof(float)); + magnitudes = (float *) malloc(bins * sizeof(float)); + phases = (float *) malloc(bins * sizeof(float)); + powers = (float *) malloc(bins * sizeof(float)); + avgPower = new float; + memset(buffer, 0, fftSize * sizeof(float)); + memset(magnitudes, 0, bins * sizeof(float)); + memset(phases, 0, bins * sizeof(float)); + memset(powers, 0, bins * sizeof(float)); + *avgPower = 0; + pos =windowSize - hopSize; + newFFT = 0; + window = (float*) malloc(fftSize * sizeof(float)); + memset(window, 0, fftSize * sizeof(float)); + fft::genWindow(3, windowSize, window); +} + +bool maxiFFT::process(float value) { + //add value to buffer at current pos + buffer[pos++] = value; + //if buffer full, run fft + newFFT = pos == windowSize; + if (newFFT) { +#if defined(__APPLE_CC__) && !defined(_NO_VDSP) + _fft->powerSpectrum_vdsp(0, buffer, window, magnitudes, phases); +#else + _fft->powerSpectrum(0, buffer, window, magnitudes, phases, powers, avgPower); +#endif + //shift buffer back by one hop size + memcpy(buffer, buffer + hopSize, (windowSize - hopSize) * sizeof(float)); + //set the new data to zero (is this necessary?, it will get overwritten) + memset(buffer + windowSize - hopSize, 0, hopSize * sizeof(float)); + //reset pos to start of hop + pos= windowSize - hopSize; + } + return newFFT; +} + +maxiFFT::~maxiFFT() { + delete _fft; + delete[] buffer,magnitudes,phases,powers, window; + delete avgPower; +} + + + + +////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// +//I N V E R S E F F T +////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + +void maxiIFFT::setup(int _fftSize, int _windowSize, int _hopSize) { + _fft = new fft(_fftSize); + fftSize = _fftSize; + windowSize = _windowSize; + bins = fftSize / 2; + hopSize = _hopSize; + buffer = (float *) malloc(fftSize * sizeof(float)); + ifftOut = (float *) malloc(fftSize * sizeof(float)); + memset(buffer, 0, windowSize * sizeof(float)); + pos =0; + window = (float*) malloc(fftSize * sizeof(float)); + memset(window, 0, fftSize * sizeof(float)); + fft::genWindow(3, windowSize, window); +} + +float maxiIFFT::process(float *magnitudes, float *phases) { + if (0==pos) { + //do ifft + memset(ifftOut, 0, fftSize * sizeof(float)); +#if defined(__APPLE_CC__) && !defined(_NO_VDSP) + _fft->inversePowerSpectrum_vdsp(0, ifftOut, window, magnitudes, phases); +#else + _fft->inversePowerSpectrum(0, ifftOut, window, magnitudes, phases); +#endif + //add to output + //shift back by one hop + memcpy(buffer, buffer+hopSize, (fftSize - hopSize) * sizeof(float)); + //clear the end chunk + memset(buffer + (fftSize - hopSize), 0, hopSize * sizeof(float)); + //merge new output + for(int i=0; i < fftSize; i++) { + buffer[i] += ifftOut[i]; + } + } + + nextValue = buffer[pos]; + //limit the values, this alg seems to spike occasionally (and break the audio drivers) + if (nextValue > 0.99999f) nextValue = 0.99999f; + if (nextValue < -0.99999f) nextValue = -0.99999f; + if (hopSize == ++pos ) { + pos=0; + } + + return nextValue; +} + +maxiIFFT::~maxiIFFT() { + delete _fft; + delete[] ifftOut, buffer, window; +}; + + + + + + + +////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// +//O C T A V E A N A L Y S E R +////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + + + +void maxiFFTOctaveAnalyzer::setup(float samplingRate, int nBandsInTheFFT, int nAveragesPerOctave){ + + samplingRate = samplingRate; + nSpectrum = nBandsInTheFFT; + spectrumFrequencySpan = (samplingRate / 2.0f) / (float)(nSpectrum); + nAverages = nBandsInTheFFT; + // fe: 2f for octave bands, sqrt(2) for half-octave bands, cuberoot(2) for third-octave bands, etc + if (nAveragesPerOctave==0) // um, wtf? + nAveragesPerOctave = 1; + nAveragesPerOctave = nAveragesPerOctave; + averageFrequencyIncrement = pow(2.0f, 1.0f/(float)(nAveragesPerOctave)); + // this isn't currently configurable (used once here then no effect), but here's some reasoning: + // 43 is a good value if you want to approximate "computer" octaves: 44100/2/2/2/2/2/2/2/2/2/2 + // 55 is a good value if you'd rather approximate A-440 octaves: 440/2/2/2 + // 65 is a good value if you'd rather approximate "middle-C" octaves: ~262/2/2 + // you could easily double it if you felt the lowest band was just rumble noise (as it probably is) + // but don't go much smaller unless you have a huge fft window size (see below for more why) + // keep in mind, if you change it, that the number of actual bands may change +/-1, and + // for some values, the last averaging band may not be very useful (may extend above nyquist) + firstOctaveFrequency = 55.0f; + // for each spectrum[] bin, calculate the mapping into the appropriate average[] bin. + // this gives us roughly log-sized averaging bins, subject to how "fine" the spectrum bins are. + // with more spectrum bins, you can better map into the averaging bins (especially at low + // frequencies) or use more averaging bins per octave. with an fft window size of 2048, + // sampling rate of 44100, and first octave around 55, that's about enough to do half-octave + // analysis. if you don't have enough spectrum bins to map adequately into averaging bins + // at the requested number per octave then you'll end up with "empty" averaging bins, where + // there is no spectrum available to map into it. (so... if you have "nonreactive" averages, + // either increase fft buffer size, or decrease number of averages per octave, etc) + spe2avg = new int[nSpectrum]; + int avgidx = 0; + float averageFreq = firstOctaveFrequency; // the "top" of the first averaging bin + // we're looking for the "top" of the first spectrum bin, and i'm just sort of + // guessing that this is where it is (or possibly spectrumFrequencySpan/2?) + // ... either way it's probably close enough for these purposes + float spectrumFreq = spectrumFrequencySpan; + for (int speidx=0; speidx < nSpectrum; speidx++) { + while (spectrumFreq > averageFreq) { + avgidx++; + averageFreq *= averageFrequencyIncrement; + } + spe2avg[speidx] = avgidx; + spectrumFreq += spectrumFrequencySpan; + } + nAverages = avgidx; + averages = new float[nAverages]; + peaks = new float[nAverages]; + peakHoldTimes = new int[nAverages]; + peakHoldTime = 0; // arbitrary + peakDecayRate = 0.9f; // arbitrary + linearEQIntercept = 1.0f; // unity -- no eq by default + linearEQSlope = 0.0f; // unity -- no eq by default +} + +void maxiFFTOctaveAnalyzer::calculate(float * fftData){ + + int last_avgidx = 0; // tracks when we've crossed into a new averaging bin, so store current average + float sum = 0.0f; // running total of spectrum data + int count = 0; // count of spectrums accumulated (for averaging) + for (int speidx=0; speidx < nSpectrum; speidx++) { + count++; + sum += fftData[speidx] * (linearEQIntercept + (float)(speidx) * linearEQSlope); + int avgidx = spe2avg[speidx]; + if (avgidx != last_avgidx) { + + for (int j = last_avgidx; j < avgidx; j++){ + averages[j] = sum / (float)(count); + } + count = 0; + sum = 0.0f; + } + last_avgidx = avgidx; + } + // the last average was probably not calculated... + if ((count > 0) && (last_avgidx < nAverages)){ + averages[last_avgidx] = sum / (float)(count); + } + + // update the peaks separately + for (int i=0; i < nAverages; i++) { + if (averages[i] >= peaks[i]) { + // save new peak level, also reset the hold timer + peaks[i] = averages[i]; + peakHoldTimes[i] = peakHoldTime; + } else { + // current average does not exceed peak, so hold or decay the peak + if (peakHoldTimes[i] > 0) { + peakHoldTimes[i]--; + } else { + peaks[i] *= peakDecayRate; + } + } + } +} diff --git a/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/maxiFFT.h b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/maxiFFT.h new file mode 100644 index 0000000..4d73441 --- /dev/null +++ b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/maxiFFT.h @@ -0,0 +1,120 @@ +/* + * maximilian + * platform independent synthesis library using portaudio or rtaudio + * + * Created by Mick Grierson on 29/12/2009. + * Copyright 2009 Mick Grierson & Strangeloop Limited. All rights reserved. + * Thanks to the Goldsmiths Creative Computing Team. + * Special thanks to Arturo Castro for the PortAudio implementation. + * + * 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, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following + * conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * 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. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + * + */ + +#ifndef _MAXI_FFT +#define _MAXI_FFT + +//#define _NO_VDSP //set this if you don't want to use apple's vDSP fft functions + + +#include "fft.h" + +class maxiFFT { + +public: + maxiFFT(){ + _fft = 0; + }; + ~maxiFFT(); + void setup(int fftSize, int windowSize, int hopSize); + bool process(float value); + float *magnitudes, *phases, *powers; + float *avgPower; + int windowSize; + int hopSize; + +private: + float *buffer, *window; + int bins; + int pos; + float nextValue; + int fftSize; + fft *_fft; + bool newFFT; +}; + +class maxiIFFT { + +public: + maxiIFFT(){ + _fft=0; + }; + ~maxiIFFT(); + void setup(int fftSize, int windowSize, int hopSize); + float process(float *magnitudes, float *phases); + +private: + float *ifftOut, *buffer, *window; + int windowSize; + int bins; + int hopSize; + int pos; + float nextValue; + int fftSize; + fft *_fft; +}; + + +class maxiFFTOctaveAnalyzer { +/*based on code by David Bollinger, http://www.davebollinger.com/ + */ +public: + + float samplingRate; // sampling rate in Hz (needed to calculate frequency spans) + int nSpectrum; // number of spectrum bins in the fft + int nAverages; // number of averaging bins here + int nAveragesPerOctave; // number of averages per octave as requested by user + float spectrumFrequencySpan; // the "width" of an fft spectrum bin in Hz + float firstOctaveFrequency; // the "top" of the first averaging bin here in Hz + float averageFrequencyIncrement; // the root-of-two multiplier between averaging bin frequencies + float * averages; // the actual averages + float * peaks; // peaks of the averages, aka "maxAverages" in other implementations + int * peakHoldTimes; // how long to hold THIS peak meter? decay if == 0 + int peakHoldTime; // how long do we hold peaks? (in fft frames) + float peakDecayRate; // how quickly the peaks decay: 0f=instantly .. 1f=not at all + int * spe2avg; // the mapping between spectrum[] indices and averages[] indices + // the fft's log equalizer() is no longer of any use (it would be nonsense to log scale + // the spectrum values into log-sized average bins) so here's a quick-and-dirty linear + // equalizer instead: + float linearEQSlope; // the rate of linear eq + float linearEQIntercept; // the base linear scaling used at the first averaging bin + // the formula is: spectrum[i] * (linearEQIntercept + i * linearEQSlope) + // so.. note that clever use of it can also provide a "gain" control of sorts + // (fe: set intercept to 2f and slope to 0f to double gain) + + void setup(float samplingRate, int nBandsInTheFFT, int nAveragesPerOctave); + + void calculate(float * fftData); + +}; + +#endif
\ No newline at end of file diff --git a/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/maximilian.cpp b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/maximilian.cpp new file mode 100644 index 0000000..efd6d28 --- /dev/null +++ b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/maximilian.cpp @@ -0,0 +1,712 @@ + +/* + * maximilian.cpp + * platform independent synthesis library using portaudio or rtaudio + * + * Created by Mick Grierson on 29/12/2009. + * Copyright 2009 Mick Grierson & Strangeloop Limited. All rights reserved. + * Thanks to the Goldsmiths Creative Computing Team. + * Special thanks to Arturo Castro for the PortAudio implementation. + * + * 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, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following + * conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * 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. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + * + */ + +#include "maximilian.h" +#include "math.h" + +int channels=2; +int samplerate=44100; +int buffersize=1024; +float chandiv= 1; + +double sineBuffer[514]={0,0.012268,0.024536,0.036804,0.049042,0.06131,0.073547,0.085785,0.097992,0.1102,0.12241,0.13455,0.1467,0.15884,0.17093,0.18301,0.19507,0.20709,0.21909,0.23105,0.24295,0.25485,0.26669,0.2785,0.29025,0.30197,0.31366,0.32529,0.33685,0.34839,0.35986,0.37128,0.38266,0.39395,0.40521,0.41641,0.42752,0.4386,0.44958,0.46051,0.47137,0.48215,0.49286,0.50351,0.51407,0.52457,0.53497,0.54529,0.55554,0.5657,0.57578,0.58575,0.59567,0.60547,0.6152,0.62482,0.63437,0.6438,0.65314,0.66238,0.67151,0.68057,0.68951,0.69833,0.70706,0.7157,0.72421,0.7326,0.74091,0.74908,0.75717,0.76514,0.77298,0.7807,0.7883,0.79581,0.80316,0.81042,0.81754,0.82455,0.83142,0.8382,0.84482,0.85132,0.8577,0.86392,0.87006,0.87604,0.88187,0.8876,0.89319,0.89862,0.90396,0.90912,0.91415,0.91907,0.92383,0.92847,0.93295,0.93729,0.9415,0.94556,0.94949,0.95325,0.95691,0.96039,0.96375,0.96692,0.97,0.9729,0.97565,0.97827,0.98074,0.98306,0.98523,0.98724,0.98914,0.99084,0.99243,0.99387,0.99515,0.99628,0.99725,0.99808,0.99875,0.99927,0.99966,0.99988,0.99997,0.99988,0.99966,0.99927,0.99875,0.99808,0.99725,0.99628,0.99515,0.99387,0.99243,0.99084,0.98914,0.98724,0.98523,0.98306,0.98074,0.97827,0.97565,0.9729,0.97,0.96692,0.96375,0.96039,0.95691,0.95325,0.94949,0.94556,0.9415,0.93729,0.93295,0.92847,0.92383,0.91907,0.91415,0.90912,0.90396,0.89862,0.89319,0.8876,0.88187,0.87604,0.87006,0.86392,0.8577,0.85132,0.84482,0.8382,0.83142,0.82455,0.81754,0.81042,0.80316,0.79581,0.7883,0.7807,0.77298,0.76514,0.75717,0.74908,0.74091,0.7326,0.72421,0.7157,0.70706,0.69833,0.68951,0.68057,0.67151,0.66238,0.65314,0.6438,0.63437,0.62482,0.6152,0.60547,0.59567,0.58575,0.57578,0.5657,0.55554,0.54529,0.53497,0.52457,0.51407,0.50351,0.49286,0.48215,0.47137,0.46051,0.44958,0.4386,0.42752,0.41641,0.40521,0.39395,0.38266,0.37128,0.35986,0.34839,0.33685,0.32529,0.31366,0.30197,0.29025,0.2785,0.26669,0.25485,0.24295,0.23105,0.21909,0.20709,0.19507,0.18301,0.17093,0.15884,0.1467,0.13455,0.12241,0.1102,0.097992,0.085785,0.073547,0.06131,0.049042,0.036804,0.024536,0.012268,0,-0.012268,-0.024536,-0.036804,-0.049042,-0.06131,-0.073547,-0.085785,-0.097992,-0.1102,-0.12241,-0.13455,-0.1467,-0.15884,-0.17093,-0.18301,-0.19507,-0.20709,-0.21909,-0.23105,-0.24295,-0.25485,-0.26669,-0.2785,-0.29025,-0.30197,-0.31366,-0.32529,-0.33685,-0.34839,-0.35986,-0.37128,-0.38266,-0.39395,-0.40521,-0.41641,-0.42752,-0.4386,-0.44958,-0.46051,-0.47137,-0.48215,-0.49286,-0.50351,-0.51407,-0.52457,-0.53497,-0.54529,-0.55554,-0.5657,-0.57578,-0.58575,-0.59567,-0.60547,-0.6152,-0.62482,-0.63437,-0.6438,-0.65314,-0.66238,-0.67151,-0.68057,-0.68951,-0.69833,-0.70706,-0.7157,-0.72421,-0.7326,-0.74091,-0.74908,-0.75717,-0.76514,-0.77298,-0.7807,-0.7883,-0.79581,-0.80316,-0.81042,-0.81754,-0.82455,-0.83142,-0.8382,-0.84482,-0.85132,-0.8577,-0.86392,-0.87006,-0.87604,-0.88187,-0.8876,-0.89319,-0.89862,-0.90396,-0.90912,-0.91415,-0.91907,-0.92383,-0.92847,-0.93295,-0.93729,-0.9415,-0.94556,-0.94949,-0.95325,-0.95691,-0.96039,-0.96375,-0.96692,-0.97,-0.9729,-0.97565,-0.97827,-0.98074,-0.98306,-0.98523,-0.98724,-0.98914,-0.99084,-0.99243,-0.99387,-0.99515,-0.99628,-0.99725,-0.99808,-0.99875,-0.99927,-0.99966,-0.99988,-0.99997,-0.99988,-0.99966,-0.99927,-0.99875,-0.99808,-0.99725,-0.99628,-0.99515,-0.99387,-0.99243,-0.99084,-0.98914,-0.98724,-0.98523,-0.98306,-0.98074,-0.97827,-0.97565,-0.9729,-0.97,-0.96692,-0.96375,-0.96039,-0.95691,-0.95325,-0.94949,-0.94556,-0.9415,-0.93729,-0.93295,-0.92847,-0.92383,-0.91907,-0.91415,-0.90912,-0.90396,-0.89862,-0.89319,-0.8876,-0.88187,-0.87604,-0.87006,-0.86392,-0.8577,-0.85132,-0.84482,-0.8382,-0.83142,-0.82455,-0.81754,-0.81042,-0.80316,-0.79581,-0.7883,-0.7807,-0.77298,-0.76514,-0.75717,-0.74908,-0.74091,-0.7326,-0.72421,-0.7157,-0.70706,-0.69833,-0.68951,-0.68057,-0.67151,-0.66238,-0.65314,-0.6438,-0.63437,-0.62482,-0.6152,-0.60547,-0.59567,-0.58575,-0.57578,-0.5657,-0.55554,-0.54529,-0.53497,-0.52457,-0.51407,-0.50351,-0.49286,-0.48215,-0.47137,-0.46051,-0.44958,-0.4386,-0.42752,-0.41641,-0.40521,-0.39395,-0.38266,-0.37128,-0.35986,-0.34839,-0.33685,-0.32529,-0.31366,-0.30197,-0.29025,-0.2785,-0.26669,-0.25485,-0.24295,-0.23105,-0.21909,-0.20709,-0.19507,-0.18301,-0.17093,-0.15884,-0.1467,-0.13455,-0.12241,-0.1102,-0.097992,-0.085785,-0.073547,-0.06131,-0.049042,-0.036804,-0.024536,-0.012268,0,0.012268 +}; + +void setup();//use this to do any initialisation if you want. + +void play(double *channels);//run dac! + +maxiOsc::maxiOsc(){ + phase = 0.0; +// memset(phases,0,500); +// memset(freqs,0,500); +} + +double maxiOsc::noise() { + //always the same unless you seed it. + float r = rand()/(float)RAND_MAX; + output=r*2-1; + return(output); +} + +double maxiOsc::sinewave(double frequency) { + output=sin (phase*(TWOPI)); + if ( phase >= 1.0 ) phase -= 1.0; + phase += (1./(samplerate/(frequency))); + return(output); + +} + +double maxiOsc::sinebuf4(double frequency) { + double remainder; + double a,b,c,d,a1,a2,a3; + phase += 512./(samplerate/(frequency)); + if ( phase >= 511 ) phase -=512; + remainder = phase - floor(phase); + + if (phase==0) { + a=sineBuffer[(long) 512]; + b=sineBuffer[(long) phase]; + c=sineBuffer[(long) phase+1]; + d=sineBuffer[(long) phase+2]; + + } else { + a=sineBuffer[(long) phase-1]; + b=sineBuffer[(long) phase]; + c=sineBuffer[(long) phase+1]; + d=sineBuffer[(long) phase+2]; + + } + + a1 = 0.5f * (c - a); + a2 = a - 2.5 * b + 2.f * c - 0.5f * d; + a3 = 0.5f * (d - a) + 1.5f * (b - c); + output = double (((a3 * remainder + a2) * remainder + a1) * remainder + b); + return(output); +} + +double maxiOsc::sinebuf(double frequency) { + double remainder; + phase += 512./(samplerate/(frequency*chandiv)); + if ( phase >= 511 ) phase -=512; + remainder = phase - floor(phase); + output = (double) ((1-remainder) * sineBuffer[1+ (long) phase] + remainder * sineBuffer[2+(long) phase]); + return(output); +} + +double maxiOsc::coswave(double frequency) { + output=cos (phase*(TWOPI)); + if ( phase >= 1.0 ) phase -= 1.0; + phase += (1./(samplerate/(frequency))); + return(output); + +} + +double maxiOsc::phasor(double frequency) { + output=phase; + if ( phase >= 1.0 ) phase -= 1.0; + phase += (1./(samplerate/(frequency))); + return(output); +} + +double maxiOsc::square(double frequency) { + if (phase<0.5) output=-1; + if (phase>0.5) output=1; + if ( phase >= 1.0 ) phase -= 1.0; + phase += (1./(samplerate/(frequency))); + return(output); +} + +double maxiOsc::pulse(double frequency, double duty) { + if (duty<0.) duty=0; + if (duty>1.) duty=1; + if ( phase >= 1.0 ) phase -= 1.0; + phase += (1./(samplerate/(frequency))); + if (phase<duty) output=-1.; + if (phase>duty) output=1.; + return(output); +} + +double maxiOsc::phasor(double frequency, double startphase, double endphase) { + output=phase; + if (phase<startphase) { + phase=startphase; + } + if ( phase >= endphase ) phase = startphase; + phase += ((endphase-startphase)/(samplerate/(frequency))); + return(output); +} + + +double maxiOsc::saw(double frequency) { + + output=phase; + if ( phase >= 1.0 ) phase -= 2.0; + phase += (1./(samplerate/(frequency))); + return(output); + +} + +double maxiOsc::triangle(double frequency, double phase) { + output=tri*2; + if ( phase >= 1.0 ) phase -= 1.0; + phase += (1./(samplerate/(frequency*chandiv))); + if (phase <= 0.5 ) { + tri = phase; + } else { + tri =(1-phase); + } + return(output); + +} + +//I like this. +double maxiEnvelope::line(int numberofsegments,double segments[1000]) { + period=2./(segments[valindex+1]*0.004); + nextval=segments[valindex+2]; + currentval=segments[valindex]; + if (currentval-amplitude > 0.0000001 && valindex < numberofsegments) { + amplitude += ((currentval-startval)/(samplerate/period)); + } else if (currentval-amplitude < -0.0000001 && valindex < numberofsegments) { + amplitude -= (((currentval-startval)*(-1))/(samplerate/period)); + } else if (valindex >numberofsegments-1) { + valindex=numberofsegments-2; + } else { + valindex=valindex+2; + startval=currentval; + } + output=amplitude; + return(output); +} + +//and this +void maxiEnvelope::trigger(int index,double amp) { + valindex=index; + amplitude=amp; + +} + +//and this +double maxiDelayline::dl(double input, int size, double feedback) { + if ( phase >=size ) phase = 0; + memory[phase]=(memory[phase]*feedback)+(input*feedback)*0.5; + phase+=1; + output=memory[phase+1]; + return(output); + +} + +double maxiDelayline::dl(double input, int size, double feedback, int position) { + if ( phase >=size ) phase = 0; + if ( position >=size ) position = 0; + memory[phase]=(memory[phase]*feedback)+(input*feedback)*chandiv; + phase+=1; + output=memory[position]; + return(output); + +} + +//I particularly like these. cutoff between 0 and 1 +double maxiFilter::lopass(double input, double cutoff) { + output=outputs[0] + cutoff*(input-outputs[0]); + outputs[0]=output; + return(output); +} +//as above +double maxiFilter::hipass(double input, double cutoff) { + output=input-(outputs[0] + cutoff*(input-outputs[0])); + outputs[0]=output; + return(output); +} +//awesome. cuttof is freq in hz. res is between 1 and whatever. Watch out! +double maxiFilter::lores(double input,double cutoff1, double resonance) { + cutoff=cutoff1*0.5; + if (cutoff>(samplerate*0.25)) cutoff=(samplerate*0.25); + if (resonance<1.) resonance = 1.; + z=cos(TWOPI*cutoff/samplerate); + c=2-2*z; + double r=(sqrt(2.0)*sqrt(-pow((z-1.0),3.0))+resonance*(z-1))/(resonance*(z-1)); + x=x+(input-y)*c; + y=y+x; + x=x*r; + output=y; + return(output); +} + +//this is just stupid. Kindof works. +double maxiFilter::hires(double input,double cutoff1, double resonance) { + cutoff=cutoff1*0.5; + if (cutoff>(samplerate*0.25)) cutoff=(samplerate*0.25); + if (resonance<1.) resonance = 1.; + z=cos(TWOPI*cutoff/samplerate); + c=2-2*z; + double r=(sqrt(2.0)*sqrt(-pow((z-1.0),3.0))+resonance*(z-1))/(resonance*(z-1)); + x=x+(input-y)*c; + y=y+x; + x=x*r; + output=input-y; + return(output); +} + +//This works a bit. Needs attention. +double maxiFilter::bandpass(double input,double cutoff1, double resonance) { + cutoff=cutoff1; + if (cutoff>(samplerate*0.5)) cutoff=(samplerate*0.5); + if (resonance>=1.) resonance=0.999999; + z=cos(TWOPI*cutoff/samplerate); + inputs[0] = (1-resonance)*(sqrt(resonance*(resonance-4.0*pow(z,2.0)+2.0)+1)); + inputs[1] = 2*z*resonance; + inputs[2] = pow((resonance*-1),2); + + output=inputs[0]*input+inputs[1]*outputs[1]+inputs[2]*outputs[2]; + outputs[2]=outputs[1]; + outputs[1]=output; + return(output); +} + +//stereo bus +double *maxiMix::stereo(double input,double two[2],double x) { + if (x>1) x=1; + if (x<0) x=0; + two[0]=input*sqrt(1.0-x); + two[1]=input*sqrt(x); + return(two); +} + +//quad bus +double *maxiMix::quad(double input,double four[4],double x,double y) { + if (x>1) x=1; + if (x<0) x=0; + if (y>1) y=1; + if (y<0) y=0; + four[0]=input*sqrt((1.0-x)*y); + four[1]=input*sqrt((1.0-x)*(1.0-y)); + four[2]=input*sqrt(x*y); + four[3]=input*sqrt(x*(1.0-y)); + return(four); +} + +//ambisonic bus +double *maxiMix::ambisonic(double input,double eight[8],double x,double y,double z) { + if (x>1) x=1; + if (x<0) x=0; + if (y>1) y=1; + if (y<0) y=0; + if (z>1) y=1; + if (z<0) y=0; + eight[0]=input*(sqrt((1.0-x)*y)*1.0-z); + eight[1]=input*(sqrt((1.0-x)*(1.0-y))*1.0-z); + eight[2]=input*(sqrt(x*y)*1.0-z); + eight[3]=input*(sqrt(x*(1.0-y))*1.0-z); + eight[4]=input*(sqrt((1.0-x)*y)*z); + eight[5]=input*(sqrt((1.0-x)*(1.0-y))*z); + eight[6]=input*sqrt((x*y)*z); + eight[7]=input*sqrt((x*(1.0-y))*z); + return(eight); +} + + +bool maxiSample::load(string fileName) { + myPath = fileName; + return read(); +} + +void maxiSample::trigger() { + position = 0; +} + +double maxiSample::play() { + long length=myDataSize*(1./myChannels); + double remainder; + short* buffer = (short *)myData; + position=(position+1); + remainder = position - (long) position; + if ((long) position>length) position=0; + output = (double) ((1-remainder) * buffer[1+ (long) position] + remainder * buffer[2+(long) position])/32767;//linear interpolation + return(output); +} + +double maxiSample::play(double speed) { + double remainder; + long a,b; +// long length=myDataSize*0.5; + short* buffer = (short *)myData; + position=position+((speed*chandiv*myChannels)/(samplerate/mySampleRate)); + if (speed >=0) { + + if ((long) position>=length-1) position=1; + remainder = position - floor(position); + if (position+1<length) { + a=position+1; + + } + else { + a=length-1; + } + if (position+2<length) + { + b=position+2; + } + else { + b=length-1; + } + + output = (double) ((1-remainder) * buffer[a] + remainder * buffer[b])/32767;//linear interpolation +} else { + if ((long) position<0) position=length; + remainder = position - floor(position); + if (position-1>=0) { + a=position-1; + + } + else { + a=0; + } + if (position-2>=0) { + b=position-2; + } + else { + b=0; + } + output = (double) ((-1-remainder) * buffer[a] + remainder * buffer[b])/32767;//linear interpolation + } + return(output); +} + +double maxiSample::play(double frequency, double start, double end) { + double remainder; +// long length=myDataSize; + if (end>=length) end=length-1; + long a,b; + short* buffer = (short *)myData; + if (frequency >0.) { + if (position<start) { + position=start; + } + + if ( position >= end ) position = start; + position += ((end-start)/(samplerate/(frequency*chandiv))); + remainder = position - floor(position); + long pos = floor(position); + if (pos+1<length) { + a=pos+1; + + } + else { + a=pos-1; + } + if (pos+2<length) { + b=pos+2; + } + else { + b=length-1; + } + + output = (double) ((1-remainder) * buffer[a] + + remainder * buffer[b])/32767;//linear interpolation + } else { + frequency=frequency-(frequency+frequency); + if ( position <= start ) position = end; + position -= ((end-start)/(samplerate/(frequency*chandiv))); + remainder = position - floor(position); + long pos = floor(position); + if (pos-1>=0) { + a=pos-1; + } + else { + a=0; + } + if (pos-2>=0) { + b=pos-2; + } + else { + b=0; + } + output = (double) ((-1-remainder) * buffer[a] + + remainder * buffer[b])/32767;//linear interpolation + + } + + return(output); +} + +//better cubic inerpolation. Cobbled together from various (pd externals, yehar, other places). +double maxiSample::play4(double frequency, double start, double end) { + double remainder; + double a,b,c,d,a1,a2,a3; + short* buffer = (short*)myData; + if (frequency >0.) { + if (position<start) { + position=start; + } + if ( position >= end ) position = start; + position += ((end-start)/(samplerate/(frequency*chandiv))); + remainder = position - floor(position); + if (position>0) { + a=buffer[(int)(floor(position))-1]; + + } else { + a=buffer[0]; + + } + + b=buffer[(long) position]; + if (position<end-2) { + c=buffer[(long) position+1]; + + } else { + c=buffer[0]; + + } + if (position<end-3) { + d=buffer[(long) position+2]; + + } else { + d=buffer[0]; + } + a1 = 0.5f * (c - a); + a2 = a - 2.5 * b + 2.f * c - 0.5f * d; + a3 = 0.5f * (d - a) + 1.5f * (b - c); + output = (double) (((a3 * remainder + a2) * remainder + a1) * remainder + b) / 32767; + + } else { + frequency=frequency-(frequency+frequency); + if ( position <= start ) position = end; + position -= ((end-start)/(samplerate/(frequency*chandiv))); + remainder = position - floor(position); + if (position>start && position < end-1) { + a=buffer[(long) position+1]; + + } else { + a=buffer[0]; + + } + + b=buffer[(long) position]; + if (position>start) { + c=buffer[(long) position-1]; + + } else { + c=buffer[0]; + + } + if (position>start+1) { + d=buffer[(long) position-2]; + + } else { + d=buffer[0]; + } + a1 = 0.5f * (c - a); + a2 = a - 2.5 * b + 2.f * c - 0.5f * d; + a3 = 0.5f * (d - a) + 1.5f * (b - c); + output = (double) (((a3 * remainder + a2) * -remainder + a1) * -remainder + b) / 32767; + + } + + return(output); +} + +double maxiSample::bufferPlay(unsigned char &bufferin,long length) { + double remainder; + short* buffer = (short *)bufferin; + position=(position+1); + remainder = position - (long) position; + if ((long) position>length) position=0; + output = (double) ((1-remainder) * buffer[1+ (long) position] + remainder * buffer[2+(long) position])/32767;//linear interpolation + return(output); +} + +double maxiSample::bufferPlay(unsigned char &bufferin,double speed,long length) { + double remainder; + long a,b; + short* buffer = (short *)bufferin; + position=position+((speed*chandiv*myChannels)/(samplerate/mySampleRate)); + if (speed >=0) { + + if ((long) position>=length-1) position=1; + remainder = position - floor(position); + if (position+1<length) { + a=position+1; + + } + else { + a=length-1; + } + if (position+2<length) + { + b=position+2; + } + else { + b=length-1; + } + + output = (double) ((1-remainder) * buffer[a] + remainder * buffer[b])/32767;//linear interpolation + } else { + if ((long) position<0) position=length; + remainder = position - floor(position); + if (position-1>=0) { + a=position-1; + + } + else { + a=0; + } + if (position-2>=0) { + b=position-2; + } + else { + b=0; + } + output = (double) ((-1-remainder) * buffer[a] + remainder * buffer[b])/32767;//linear interpolation + } + return(output); +} + +double maxiSample::bufferPlay(unsigned char &bufferin,double frequency, double start, double end) { + double remainder; + length=end; + long a,b; + short* buffer = (short *)bufferin; + if (frequency >0.) { + if (position<start) { + position=start; + } + + if ( position >= end ) position = start; + position += ((end-start)/(samplerate/(frequency*chandiv))); + remainder = position - floor(position); + long pos = floor(position); + if (pos+1<length) { + a=pos+1; + + } + else { + a=pos-1; + } + if (pos+2<length) { + b=pos+2; + } + else { + b=length-1; + } + + output = (double) ((1-remainder) * buffer[a] + + remainder * buffer[b])/32767;//linear interpolation + } else { + frequency=frequency-(frequency+frequency); + if ( position <= start ) position = end; + position -= ((end-start)/(samplerate/(frequency*chandiv))); + remainder = position - floor(position); + long pos = floor(position); + if (pos-1>=0) { + a=pos-1; + } + else { + a=0; + } + if (pos-2>=0) { + b=pos-2; + } + else { + b=0; + } + output = (double) ((-1-remainder) * buffer[a] + + remainder * buffer[b])/32767;//linear interpolation + + } + + return(output); +} + +//better cubic inerpolation. Cobbled together from various (pd externals, yehar, other places). +double maxiSample::bufferPlay4(unsigned char &bufferin,double frequency, double start, double end) { + double remainder; + double a,b,c,d,a1,a2,a3; + short* buffer = (short*)bufferin; + if (frequency >0.) { + if (position<start) { + position=start; + } + if ( position >= end ) position = start; + position += ((end-start)/(samplerate/(frequency*chandiv))); + remainder = position - floor(position); + if (position>0) { + a=buffer[(int)(floor(position))-1]; + + } else { + a=buffer[0]; + + } + + b=buffer[(long) position]; + if (position<end-2) { + c=buffer[(long) position+1]; + + } else { + c=buffer[0]; + + } + if (position<end-3) { + d=buffer[(long) position+2]; + + } else { + d=buffer[0]; + } + a1 = 0.5f * (c - a); + a2 = a - 2.5 * b + 2.f * c - 0.5f * d; + a3 = 0.5f * (d - a) + 1.5f * (b - c); + output = (double) (((a3 * remainder + a2) * remainder + a1) * remainder + b) / 32767; + + } else { + frequency=frequency-(frequency+frequency); + if ( position <= start ) position = end; + position -= ((end-start)/(samplerate/(frequency*chandiv))); + remainder = position - floor(position); + if (position>start && position < end-1) { + a=buffer[(long) position+1]; + + } else { + a=buffer[0]; + + } + + b=buffer[(long) position]; + if (position>start) { + c=buffer[(long) position-1]; + + } else { + c=buffer[0]; + + } + if (position>start+1) { + d=buffer[(long) position-2]; + + } else { + d=buffer[0]; + } + a1 = 0.5f * (c - a); + a2 = a - 2.5 * b + 2.f * c - 0.5f * d; + a3 = 0.5f * (d - a) + 1.5f * (b - c); + output = (double) (((a3 * remainder + a2) * -remainder + a1) * -remainder + b) / 32767; + + } + + return(output); +} + + +void maxiSample::getLength() { + length=myDataSize*0.5; +} + + diff --git a/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/maximilian.h b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/maximilian.h new file mode 100755 index 0000000..86fcef4 --- /dev/null +++ b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/libs/maximilian.h @@ -0,0 +1,355 @@ +/* + * maximilian.h + * platform independent synthesis library using portaudio or rtaudio + * + * Created by Mick Grierson on 29/12/2009. + * Copyright 2009 Mick Grierson & Strangeloop Limited. All rights reserved. + * Thanks to the Goldsmiths Creative Computing Team. + * Special thanks to Arturo Castro for the PortAudio implementation. + * + * 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, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following + * conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * 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. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + * + */ + +#ifndef MAXIMILIAN_H +#define MAXIMILIAN_H + +//#define MAXIMILIAN_PORTAUDIO +#define MAXIMILIAN_RT_AUDIO + +#include "ofMain.h" +#include "ofUtils.h" + + +#include <iostream> +#include <fstream> +#include <string> +#include <cstdlib> + +using namespace std; +#define TWOPI 3.14159*2 + +class maxiOsc { + + double frequency; + double phase; + double startphase; + double endphase; + double output; + double tri; + + +public: + maxiOsc(); + double sinewave(double frequency); + double coswave(double frequency); + double phasor(double frequency); + double phasor(double frequency, double startphase, double endphase); + double saw(double frequency); + double triangle(double frequency,double phase); + double square(double frequency); + double pulse(double frequency, double duty); + double noise(); + double sinebuf(double frequency); + double sinebuf4(double frequency); + +}; + + +class maxiEnvelope { + + double period; + double output; + double startval; + double currentval; + double nextval; + +public: + double line(int numberofsegments,double segments[100]); + void trigger(int index,double amp); + int valindex; + double amplitude; + +}; + + +class maxiDelayline { + double frequency; + int phase; + double startphase; + double endphase; + double output; + double memory[88200]; + +public: + double dl(double input, int size, double feedback); + double dl(double input, int size, double feedback, int position); + + +}; + + +class maxiFilter { + double gain; + double input; + double output; + double inputs[10]; + double outputs[10]; + double cutoff1; + double x;//speed + double y;//pos + double z;//pole + double c;//filter coefficient +public: + double cutoff; + double resonance; + double lores(double input,double cutoff1, double resonance); + double hires(double input,double cutoff1, double resonance); + double bandpass(double input,double cutoff1, double resonance); + double lopass(double input,double cutoff); + double hipass(double input,double cutoff); + +}; + +class maxiMix { + double input; + double two[2]; + double four[4]; + double eight[8]; +public: + double x; + double y; + double z; + double *stereo(double input,double two[2],double x); + double *quad(double input,double four[4], double x,double y); + double *ambisonic(double input,double eight[8],double x,double y, double z); + +}; + + +class maxiSample { + +private: + string myPath; + int myChunkSize; + int mySubChunk1Size; + short myFormat; + int myByteRate; + short myBlockAlign; + short myBitsPerSample; + int myDataSize; + double position; + double speed; + double output; + +public: + short myChannels; + int mySampleRate; + long length; + void getLength(); + + char* myData; + + // get/set for the Path property + + ~maxiSample() + { + delete myData; + myChunkSize = NULL; + mySubChunk1Size = NULL; + myFormat = NULL; + myChannels = NULL; + mySampleRate = NULL; + myByteRate = NULL; + myBlockAlign = NULL; + myBitsPerSample = NULL; + myDataSize = NULL; + } + +// maxiSample(); + + bool load(string fileName); + + void trigger(); + + // read a wav file into this class + bool read() + { + bool result; + ifstream inFile( myPath.c_str(), ios::in | ios::binary); + result = inFile; + + if (inFile) { + //printf("Reading wav file...\n"); // for debugging only + + inFile.seekg(4, ios::beg); + inFile.read( (char*) &myChunkSize, 4 ); // read the ChunkSize + + inFile.seekg(16, ios::beg); + inFile.read( (char*) &mySubChunk1Size, 4 ); // read the SubChunk1Size + + //inFile.seekg(20, ios::beg); + inFile.read( (char*) &myFormat, sizeof(short) ); // read the file format. This should be 1 for PCM + + //inFile.seekg(22, ios::beg); + inFile.read( (char*) &myChannels, sizeof(short) ); // read the # of channels (1 or 2) + + //inFile.seekg(24, ios::beg); + inFile.read( (char*) &mySampleRate, sizeof(int) ); // read the samplerate + + //inFile.seekg(28, ios::beg); + inFile.read( (char*) &myByteRate, sizeof(int) ); // read the byterate + + //inFile.seekg(32, ios::beg); + inFile.read( (char*) &myBlockAlign, sizeof(short) ); // read the blockalign + + //inFile.seekg(34, ios::beg); + inFile.read( (char*) &myBitsPerSample, sizeof(short) ); // read the bitspersample + + //ignore any extra chunks + char chunkID[4]=""; + int filePos = 36; + bool found = false; + while(!found && !inFile.eof()) { + inFile.seekg(filePos, ios::beg); + inFile.read((char*) &chunkID, sizeof(char) * 4); + inFile.seekg(filePos + 4, ios::beg); + inFile.read( (char*) &myDataSize, sizeof(int) ); // read the size of the data + filePos += 8; + if (strcmp(chunkID,"data") == 0) { + found = true; + }else{ + filePos += myDataSize; + } + } + + + + // read the data chunk + myData = (char*) malloc(myDataSize * sizeof(char)); + inFile.seekg(filePos, ios::beg); + inFile.read(myData, myDataSize); + length=myDataSize*(0.5/myChannels); + + inFile.close(); // close the input file + } + + + return result; // this should probably be something more descriptive + } + + + double play(); + + double play(double speed); + + double play(double frequency, double start, double end); + + double play4(double frequency, double start, double end); + + double bufferPlay(unsigned char &bufferin,long length); + + double bufferPlay(unsigned char &bufferin,double speed,long length); + + double bufferPlay(unsigned char &bufferin,double frequency, double start, double end); + + double bufferPlay4(unsigned char &bufferin,double frequency, double start, double end); + + //int open(char *filename) { + // // open the wav file + // char *path = new char[50]; + // strcpy(path, filename); + // sample *myWav = new sample(path); + // return 0; + // } + // // print a summary of the wav file + // int info() { + // char *summary = myWav->getSummary(); + // printf("Summary:\n%s", summary); + // return 0; + // } + // int savefile() { + // // write the summary back out + // strcpy(path, "testout.wav"); + // myWav->setPath(path); + // myWav->save(); + // return 0: + // } + // int close() { + // // collect the garbage + // delete summary; + // delete path; + // delete myWav; + // + // return 0; + // } + // write out the wav file + bool save() + { + fstream myFile (myPath.c_str(), ios::out | ios::binary); + + // write the wav file per the wav file format + myFile.seekp (0, ios::beg); + myFile.write ("RIFF", 4); + myFile.write ((char*) &myChunkSize, 4); + myFile.write ("WAVE", 4); + myFile.write ("fmt ", 4); + myFile.write ((char*) &mySubChunk1Size, 4); + myFile.write ((char*) &myFormat, 2); + myFile.write ((char*) &myChannels, 2); + myFile.write ((char*) &mySampleRate, 4); + myFile.write ((char*) &myByteRate, 4); + myFile.write ((char*) &myBlockAlign, 2); + myFile.write ((char*) &myBitsPerSample, 2); + myFile.write ("data", 4); + myFile.write ((char*) &myDataSize, 4); + myFile.write (myData, myDataSize); + + return true; + } + + // return a printable summary of the wav file + char *getSummary() + { + char *summary = new char[250]; + sprintf(summary, " Format: %d\n Channels: %d\n SampleRate: %d\n ByteRate: %d\n BlockAlign: %d\n BitsPerSample: %d\n DataSize: %d\n", myFormat, myChannels, mySampleRate, myByteRate, myBlockAlign, myBitsPerSample, myDataSize); + std::cout << myDataSize; + return summary; + } +}; + + +class maxiMap { +public: + static double inline linlin(double val, double inMin, double inMax, double outMin, double outMax) { + return ((val - inMin) / (inMax - inMin) * (outMax - outMin)) + outMin; + } + + static double inline linexp(double val, double inMin, double inMax, double outMin, double outMax) { + //clipping + val = max(min(val, inMax), inMin); + return pow((outMax / outMin), (val - inMin) / (inMax - inMin)) * outMin; + } +}; + + +#endif diff --git a/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/src/ofxMaxim.h b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/src/ofxMaxim.h new file mode 100644 index 0000000..317ba34 --- /dev/null +++ b/ofxMaxim/ofxMaximiPhoneExample/ofxMaxim/src/ofxMaxim.h @@ -0,0 +1,50 @@ +/* + * maximilian + * platform independent synthesis library using portaudio or rtaudio + * + * Created by Mick Grierson on 29/12/2009. + * Copyright 2009 Mick Grierson & Strangeloop Limited. All rights reserved. + * Thanks to the Goldsmiths Creative Computing Team. + * Special thanks to Arturo Castro for the PortAudio implementation. + * + * 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, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following + * conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * 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. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + * + */ + +#ifndef _OFMAXIM_H +#define _OFMAXIM_H + +#include "maximilian.h" +//#include "maxiFFT.h" + + +typedef maxiMix ofxMaxiMix; +typedef maxiOsc ofxMaxiOsc; +typedef maxiEnvelope ofxMaxiEnvelope; +typedef maxiDelayline ofxMaxiDelayline; +typedef maxiFilter ofxMaxiFilter; +typedef maxiSample ofxMaxiSample; +//typedef maxiFFT ofxMaxiFFT; +//typedef maxiIFFT ofxMaxiIFFT; +//typedef maxiFFTOctaveAnalyzer ofxMaxiFFTOctaveAnalyzer; + +#endif diff --git a/ofxMaxim/ofxMaximiPhoneExample/ofxiphone-Info.plist b/ofxMaxim/ofxMaximiPhoneExample/ofxiphone-Info.plist new file mode 100644 index 0000000..43db2fa --- /dev/null +++ b/ofxMaxim/ofxMaximiPhoneExample/ofxiphone-Info.plist @@ -0,0 +1,36 @@ +<?xml version="1.0" encoding="UTF-8"?> +<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd"> +<plist version="1.0"> +<dict> + <key>CFBundleVersion</key> + <string>1.0</string> + <key>CFBundleShortVersionString</key> + <string>1.0</string> + <key>CFBundleIdentifier</key> + <string>${PRODUCT_NAME:identifier}</string> + <key>UIStatusBarHidden</key> + <true/> + <key>CFBundleDevelopmentRegion</key> + <string>English</string> + <key>CFBundleDisplayName</key> + <string>${PRODUCT_NAME}</string> + <key>CFBundleExecutable</key> + <string>${EXECUTABLE_NAME}</string> + <key>CFBundleIconFile</key> + <string></string> + <key>CFBundleInfoDictionaryVersion</key> + <string>6.0</string> + <key>CFBundleName</key> + <string>${PRODUCT_NAME}</string> + <key>CFBundlePackageType</key> + <string>APPL</string> + <key>CFBundleSignature</key> + <string>????</string> + <key>LSRequiresIPhoneOS</key> + <true/> + <key>UIApplicationExitsOnSuspend</key> + <true/> + <key>UIInterfaceOrientation</key> + <string>UIInterfaceOrientationPortrait</string> +</dict> +</plist> diff --git a/ofxMaxim/ofxMaximiPhoneExample/src/main.mm b/ofxMaxim/ofxMaximiPhoneExample/src/main.mm new file mode 100644 index 0000000..a0b6b4c --- /dev/null +++ b/ofxMaxim/ofxMaximiPhoneExample/src/main.mm @@ -0,0 +1,8 @@ +#include "ofMain.h" +#include "testApp.h" + +int main(){ + ofSetupOpenGL(1024,768, OF_FULLSCREEN); // <-------- setup the GL context + + ofRunApp(new testApp); +} diff --git a/ofxMaxim/ofxMaximiPhoneExample/src/testApp.h b/ofxMaxim/ofxMaximiPhoneExample/src/testApp.h new file mode 100755 index 0000000..ac944e5 --- /dev/null +++ b/ofxMaxim/ofxMaximiPhoneExample/src/testApp.h @@ -0,0 +1,42 @@ +#pragma once + +#include "ofMain.h" +#include "ofxiPhone.h" +#include "ofxiPhoneExtras.h" +#include "ofxMaxim.h" + + +class testApp : public ofxiPhoneApp { + +public: + ~testApp(); + void setup(); + void update(); + void draw(); + void exit(); + + void touchDown(ofTouchEventArgs &touch); + void touchMoved(ofTouchEventArgs &touch); + void touchUp(ofTouchEventArgs &touch); + void touchDoubleTap(ofTouchEventArgs &touch); + + void lostFocus(); + void gotFocus(); + void gotMemoryWarning(); + void deviceOrientationChanged(int newOrientation); + + void audioReceived( float * input, int bufferSize, int nChannels ); + void audioRequested( float * output, int bufferSize, int nChannels ); + + int initialBufferSize; + int sampleRate; + + + ofxMaxiOsc myOsc1; + ofxMaxiSample sample1; + double patch1,sample,outputs1[2]; + ofxMaxiMix channel1; + +}; + + diff --git a/ofxMaxim/ofxMaximiPhoneExample/src/testApp.mm b/ofxMaxim/ofxMaximiPhoneExample/src/testApp.mm new file mode 100755 index 0000000..8dc9e95 --- /dev/null +++ b/ofxMaxim/ofxMaximiPhoneExample/src/testApp.mm @@ -0,0 +1,116 @@ +#include "testApp.h" +#include "ofxMaxim.h" + +testApp::~testApp() { + + delete sample1.myData; + +} + + +//-------------------------------------------------------------- +void testApp::setup(){ + ofEnableAlphaBlending(); + // register touch events + ofRegisterTouchEvents(this); + + // initialize the accelerometer + ofxAccelerometer.setup(); + + //iPhoneAlerts will be sent to this. + ofxiPhoneAlerts.addListener(this); + + sampleRate = 44100; + initialBufferSize = 512; + + sample1.load(ofToDataPath("beat2.wav")); + + ofSoundStreamSetup(2,0,this, sampleRate, initialBufferSize, 4); + + //If you want a landscape oreintation + //iPhoneSetOrientation(OFXIPHONE_ORIENTATION_LANDSCAPE_RIGHT); + + ofBackground(255,255,255); +} + +//-------------------------------------------------------------- +void testApp::update(){ + +} + +//-------------------------------------------------------------- +void testApp::draw(){ + +} + +//-------------------------------------------------------------- +void testApp::exit(){ + +} + + +//-------------------------------------------------------------- +void testApp::audioRequested(float * output, int bufferSize, int nChannels){ + + if( initialBufferSize != bufferSize ){ + ofLog(OF_LOG_ERROR, "your buffer size was set to %i - but the stream needs a buffer size of %i", initialBufferSize, bufferSize); + return; + } + + for (int i = 0; i < bufferSize; i++){ + + sample=sample1.play(1.); + channel1.stereo(sample,outputs1,0.5); + + output[i*nChannels ] = outputs1[0]; + output[i*nChannels + 1] = outputs1[1]; + + } + +} + +//-------------------------------------------------------------- +void testApp::audioReceived(float * input, int bufferSize, int nChannels){ + +} + +//-------------------------------------------------------------- +void testApp::touchDown(ofTouchEventArgs &touch){ + +} + +//-------------------------------------------------------------- +void testApp::touchMoved(ofTouchEventArgs &touch){ + +} + +//-------------------------------------------------------------- +void testApp::touchUp(ofTouchEventArgs &touch){ + +} + +//-------------------------------------------------------------- +void testApp::touchDoubleTap(ofTouchEventArgs &touch){ + +} + +//-------------------------------------------------------------- +void testApp::lostFocus(){ + +} + +//-------------------------------------------------------------- +void testApp::gotFocus(){ + +} + +//-------------------------------------------------------------- +void testApp::gotMemoryWarning(){ + +} + +//-------------------------------------------------------------- +void testApp::deviceOrientationChanged(int newOrientation){ + +} + |
