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diff --git a/services/audioflinger/AudioResamplerFirGen.h b/services/audioflinger/AudioResamplerFirGen.h new file mode 100644 index 0000000..d024b2f --- /dev/null +++ b/services/audioflinger/AudioResamplerFirGen.h @@ -0,0 +1,709 @@ +/* + * Copyright (C) 2013 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef ANDROID_AUDIO_RESAMPLER_FIR_GEN_H +#define ANDROID_AUDIO_RESAMPLER_FIR_GEN_H + +namespace android { + +/* + * generates a sine wave at equal steps. + * + * As most of our functions use sine or cosine at equal steps, + * it is very efficient to compute them that way (single multiply and subtract), + * rather than invoking the math library sin() or cos() each time. + * + * SineGen uses Goertzel's Algorithm (as a generator not a filter) + * to calculate sine(wstart + n * wstep) or cosine(wstart + n * wstep) + * by stepping through 0, 1, ... n. + * + * e^i(wstart+wstep) = 2cos(wstep) * e^i(wstart) - e^i(wstart-wstep) + * + * or looking at just the imaginary sine term, as the cosine follows identically: + * + * sin(wstart+wstep) = 2cos(wstep) * sin(wstart) - sin(wstart-wstep) + * + * Goertzel's algorithm is more efficient than the angle addition formula, + * e^i(wstart+wstep) = e^i(wstart) * e^i(wstep), which takes up to + * 4 multiplies and 2 adds (or 3* and 3+) and requires both sine and + * cosine generation due to the complex * complex multiply (full rotation). + * + * See: http://en.wikipedia.org/wiki/Goertzel_algorithm + * + */ + +class SineGen { +public: + SineGen(double wstart, double wstep, bool cosine = false) { + if (cosine) { + mCurrent = cos(wstart); + mPrevious = cos(wstart - wstep); + } else { + mCurrent = sin(wstart); + mPrevious = sin(wstart - wstep); + } + mTwoCos = 2.*cos(wstep); + } + SineGen(double expNow, double expPrev, double twoCosStep) { + mCurrent = expNow; + mPrevious = expPrev; + mTwoCos = twoCosStep; + } + inline double value() const { + return mCurrent; + } + inline void advance() { + double tmp = mCurrent; + mCurrent = mCurrent*mTwoCos - mPrevious; + mPrevious = tmp; + } + inline double valueAdvance() { + double tmp = mCurrent; + mCurrent = mCurrent*mTwoCos - mPrevious; + mPrevious = tmp; + return tmp; + } + +private: + double mCurrent; // current value of sine/cosine + double mPrevious; // previous value of sine/cosine + double mTwoCos; // stepping factor +}; + +/* + * generates a series of sine generators, phase offset by fixed steps. + * + * This is used to generate polyphase sine generators, one per polyphase + * in the filter code below. + * + * The SineGen returned by value() starts at innerStart = outerStart + n*outerStep; + * increments by innerStep. + * + */ + +class SineGenGen { +public: + SineGenGen(double outerStart, double outerStep, double innerStep, bool cosine = false) + : mSineInnerCur(outerStart, outerStep, cosine), + mSineInnerPrev(outerStart-innerStep, outerStep, cosine) + { + mTwoCos = 2.*cos(innerStep); + } + inline SineGen value() { + return SineGen(mSineInnerCur.value(), mSineInnerPrev.value(), mTwoCos); + } + inline void advance() { + mSineInnerCur.advance(); + mSineInnerPrev.advance(); + } + inline SineGen valueAdvance() { + return SineGen(mSineInnerCur.valueAdvance(), mSineInnerPrev.valueAdvance(), mTwoCos); + } + +private: + SineGen mSineInnerCur; // generate the inner sine values (stepped by outerStep). + SineGen mSineInnerPrev; // generate the inner sine previous values + // (behind by innerStep, stepped by outerStep). + double mTwoCos; // the inner stepping factor for the returned SineGen. +}; + +static inline double sqr(double x) { + return x * x; +} + +/* + * rounds a double to the nearest integer for FIR coefficients. + * + * One variant uses noise shaping, which must keep error history + * to work (the err parameter, initialized to 0). + * The other variant is a non-noise shaped version for + * S32 coefficients (noise shaping doesn't gain much). + * + * Caution: No bounds saturation is applied, but isn't needed in this case. + * + * @param x is the value to round. + * + * @param maxval is the maximum integer scale factor expressed as an int64 (for headroom). + * Typically this may be the maximum positive integer+1 (using the fact that double precision + * FIR coefficients generated here are never that close to 1.0 to pose an overflow condition). + * + * @param err is the previous error (actual - rounded) for the previous rounding op. + * For 16b coefficients this can improve stopband dB performance by up to 2dB. + * + * Many variants exist for the noise shaping: http://en.wikipedia.org/wiki/Noise_shaping + * + */ + +static inline int64_t toint(double x, int64_t maxval, double& err) { + double val = x * maxval; + double ival = floor(val + 0.5 + err*0.2); + err = val - ival; + return static_cast<int64_t>(ival); +} + +static inline int64_t toint(double x, int64_t maxval) { + return static_cast<int64_t>(floor(x * maxval + 0.5)); +} + +/* + * Modified Bessel function of the first kind + * http://en.wikipedia.org/wiki/Bessel_function + * + * The formulas are taken from Abramowitz and Stegun, + * _Handbook of Mathematical Functions_ (links below): + * + * http://people.math.sfu.ca/~cbm/aands/page_375.htm + * http://people.math.sfu.ca/~cbm/aands/page_378.htm + * + * http://dlmf.nist.gov/10.25 + * http://dlmf.nist.gov/10.40 + * + * Note we assume x is nonnegative (the function is symmetric, + * pass in the absolute value as needed). + * + * Constants are compile time derived with templates I0Term<> and + * I0ATerm<> to the precision of the compiler. The series can be expanded + * to any precision needed, but currently set around 24b precision. + * + * We use a bit of template math here, constexpr would probably be + * more appropriate for a C++11 compiler. + * + * For the intermediate range 3.75 < x < 15, we use minimax polynomial fit. + * + */ + +template <int N> +struct I0Term { + static const double value = I0Term<N-1>::value / (4. * N * N); +}; + +template <> +struct I0Term<0> { + static const double value = 1.; +}; + +template <int N> +struct I0ATerm { + static const double value = I0ATerm<N-1>::value * (2.*N-1.) * (2.*N-1.) / (8. * N); +}; + +template <> +struct I0ATerm<0> { // 1/sqrt(2*PI); + static const double value = 0.398942280401432677939946059934381868475858631164934657665925; +}; + +#if USE_HORNERS_METHOD +/* Polynomial evaluation of A + Bx + Cx^2 + Dx^3 + ... + * using Horner's Method: http://en.wikipedia.org/wiki/Horner's_method + * + * This has fewer multiplications than Estrin's method below, but has back to back + * floating point dependencies. + * + * On ARM this appears to work slower, so USE_HORNERS_METHOD is not default enabled. + */ + +inline double Poly2(double A, double B, double x) { + return A + x * B; +} + +inline double Poly4(double A, double B, double C, double D, double x) { + return A + x * (B + x * (C + x * (D))); +} + +inline double Poly7(double A, double B, double C, double D, double E, double F, double G, + double x) { + return A + x * (B + x * (C + x * (D + x * (E + x * (F + x * (G)))))); +} + +inline double Poly9(double A, double B, double C, double D, double E, double F, double G, + double H, double I, double x) { + return A + x * (B + x * (C + x * (D + x * (E + x * (F + x * (G + x * (H + x * (I)))))))); +} + +#else +/* Polynomial evaluation of A + Bx + Cx^2 + Dx^3 + ... + * using Estrin's Method: http://en.wikipedia.org/wiki/Estrin's_scheme + * + * This is typically faster, perhaps gains about 5-10% overall on ARM processors + * over Horner's method above. + */ + +inline double Poly2(double A, double B, double x) { + return A + B * x; +} + +inline double Poly3(double A, double B, double C, double x, double x2) { + return Poly2(A, B, x) + C * x2; +} + +inline double Poly3(double A, double B, double C, double x) { + return Poly2(A, B, x) + C * x * x; +} + +inline double Poly4(double A, double B, double C, double D, double x, double x2) { + return Poly2(A, B, x) + Poly2(C, D, x) * x2; // same as poly2(poly2, poly2, x2); +} + +inline double Poly4(double A, double B, double C, double D, double x) { + return Poly4(A, B, C, D, x, x * x); +} + +inline double Poly7(double A, double B, double C, double D, double E, double F, double G, + double x) { + double x2 = x * x; + return Poly4(A, B, C, D, x, x2) + Poly3(E, F, G, x, x2) * (x2 * x2); +} + +inline double Poly8(double A, double B, double C, double D, double E, double F, double G, + double H, double x, double x2, double x4) { + return Poly4(A, B, C, D, x, x2) + Poly4(E, F, G, H, x, x2) * x4; +} + +inline double Poly9(double A, double B, double C, double D, double E, double F, double G, + double H, double I, double x) { + double x2 = x * x; +#if 1 + // It does not seem faster to explicitly decompose Poly8 into Poly4, but + // could depend on compiler floating point scheduling. + double x4 = x2 * x2; + return Poly8(A, B, C, D, E, F, G, H, x, x2, x4) + I * (x4 * x4); +#else + double val = Poly4(A, B, C, D, x, x2); + double x4 = x2 * x2; + return val + Poly4(E, F, G, H, x, x2) * x4 + I * (x4 * x4); +#endif +} +#endif + +static inline double I0(double x) { + if (x < 3.75) { + x *= x; + return Poly7(I0Term<0>::value, I0Term<1>::value, + I0Term<2>::value, I0Term<3>::value, + I0Term<4>::value, I0Term<5>::value, + I0Term<6>::value, x); // e < 1.6e-7 + } + if (1) { + /* + * Series expansion coefs are easy to calculate, but are expanded around 0, + * so error is unequal over the interval 0 < x < 3.75, the error being + * significantly better near 0. + * + * A better solution is to use precise minimax polynomial fits. + * + * We use a slightly more complicated solution for 3.75 < x < 15, based on + * the tables in Blair and Edwards, "Stable Rational Minimax Approximations + * to the Modified Bessel Functions I0(x) and I1(x)", Chalk Hill Nuclear Laboratory, + * AECL-4928. + * + * http://www.iaea.org/inis/collection/NCLCollectionStore/_Public/06/178/6178667.pdf + * + * See Table 11 for 0 < x < 15; e < 10^(-7.13). + * + * Note: Beta cannot exceed 15 (hence Stopband cannot exceed 144dB = 24b). + * + * This speeds up overall computation by about 40% over using the else clause below, + * which requires sqrt and exp. + * + */ + + x *= x; + double num = Poly9(-0.13544938430e9, -0.33153754512e8, + -0.19406631946e7, -0.48058318783e5, + -0.63269783360e3, -0.49520779070e1, + -0.24970910370e-1, -0.74741159550e-4, + -0.18257612460e-6, x); + double y = x - 225.; // reflection around 15 (squared) + double den = Poly4(-0.34598737196e8, 0.23852643181e6, + -0.70699387620e3, 0.10000000000e1, y); + return num / den; + +#if IO_EXTENDED_BETA + /* Table 42 for x > 15; e < 10^(-8.11). + * This is used for Beta>15, but is disabled here as + * we never use Beta that high. + * + * NOTE: This should be enabled only for x > 15. + */ + + double y = 1./x; + double z = y - (1./15); + double num = Poly2(0.415079861746e1, -0.5149092496e1, z); + double den = Poly3(0.103150763823e2, -0.14181687413e2, + 0.1000000000e1, z); + return exp(x) * sqrt(y) * num / den; +#endif + } else { + /* + * NOT USED, but reference for large Beta. + * + * Abramowitz and Stegun asymptotic formula. + * works for x > 3.75. + */ + double y = 1./x; + return exp(x) * sqrt(y) * + // note: reciprocal squareroot may be easier! + // http://en.wikipedia.org/wiki/Fast_inverse_square_root + Poly9(I0ATerm<0>::value, I0ATerm<1>::value, + I0ATerm<2>::value, I0ATerm<3>::value, + I0ATerm<4>::value, I0ATerm<5>::value, + I0ATerm<6>::value, I0ATerm<7>::value, + I0ATerm<8>::value, y); // (... e) < 1.9e-7 + } +} + +/* A speed optimized version of the Modified Bessel I0() which incorporates + * the sqrt and numerator multiply and denominator divide into the computation. + * This speeds up filter computation by about 10-15%. + */ +static inline double I0SqrRat(double x2, double num, double den) { + if (x2 < (3.75 * 3.75)) { + return Poly7(I0Term<0>::value, I0Term<1>::value, + I0Term<2>::value, I0Term<3>::value, + I0Term<4>::value, I0Term<5>::value, + I0Term<6>::value, x2) * num / den; // e < 1.6e-7 + } + num *= Poly9(-0.13544938430e9, -0.33153754512e8, + -0.19406631946e7, -0.48058318783e5, + -0.63269783360e3, -0.49520779070e1, + -0.24970910370e-1, -0.74741159550e-4, + -0.18257612460e-6, x2); // e < 10^(-7.13). + double y = x2 - 225.; // reflection around 15 (squared) + den *= Poly4(-0.34598737196e8, 0.23852643181e6, + -0.70699387620e3, 0.10000000000e1, y); + return num / den; +} + +/* + * calculates the transition bandwidth for a Kaiser filter + * + * Formula 3.2.8, Vaidyanathan, _Multirate Systems and Filter Banks_, p. 48 + * Formula 7.76, Oppenheim and Schafer, _Discrete-time Signal Processing, 3e_, p. 542 + * + * @param halfNumCoef is half the number of coefficients per filter phase. + * + * @param stopBandAtten is the stop band attenuation desired. + * + * @return the transition bandwidth in normalized frequency (0 <= f <= 0.5) + */ +static inline double firKaiserTbw(int halfNumCoef, double stopBandAtten) { + return (stopBandAtten - 7.95)/((2.*14.36)*halfNumCoef); +} + +/* + * calculates the fir transfer response of the overall polyphase filter at w. + * + * Calculates the DTFT transfer coefficient H(w) for 0 <= w <= PI, utilizing the + * fact that h[n] is symmetric (cosines only, no complex arithmetic). + * + * We use Goertzel's algorithm to accelerate the computation to essentially + * a single multiply and 2 adds per filter coefficient h[]. + * + * Be careful be careful to consider that h[n] is the overall polyphase filter, + * with L phases, so rescaling H(w)/L is probably what you expect for "unity gain", + * as you only use one of the polyphases at a time. + */ +template <typename T> +static inline double firTransfer(const T* coef, int L, int halfNumCoef, double w) { + double accum = static_cast<double>(coef[0])*0.5; // "center coefficient" from first bank + coef += halfNumCoef; // skip first filterbank (picked up by the last filterbank). +#if SLOW_FIRTRANSFER + /* Original code for reference. This is equivalent to the code below, but slower. */ + for (int i=1 ; i<=L ; ++i) { + for (int j=0, ix=i ; j<halfNumCoef ; ++j, ix+=L) { + accum += cos(ix*w)*static_cast<double>(*coef++); + } + } +#else + /* + * Our overall filter is stored striped by polyphases, not a contiguous h[n]. + * We could fetch coefficients in a non-contiguous fashion + * but that will not scale to vector processing. + * + * We apply Goertzel's algorithm directly to each polyphase filter bank instead of + * using cosine generation/multiplication, thereby saving one multiply per inner loop. + * + * See: http://en.wikipedia.org/wiki/Goertzel_algorithm + * Also: Oppenheim and Schafer, _Discrete Time Signal Processing, 3e_, p. 720. + * + * We use the basic recursion to incorporate the cosine steps into real sequence x[n]: + * s[n] = x[n] + (2cosw)*s[n-1] + s[n-2] + * + * y[n] = s[n] - e^(iw)s[n-1] + * = sum_{k=-\infty}^{n} x[k]e^(-iw(n-k)) + * = e^(-iwn) sum_{k=0}^{n} x[k]e^(iwk) + * + * The summation contains the frequency steps we want multiplied by the source + * (similar to a DTFT). + * + * Using symmetry, and just the real part (be careful, this must happen + * after any internal complex multiplications), the polyphase filterbank + * transfer function is: + * + * Hpp[n, w, w_0] = sum_{k=0}^{n} x[k] * cos(wk + w_0) + * = Re{ e^(iwn + iw_0) y[n]} + * = cos(wn+w_0) * s[n] - cos(w(n+1)+w_0) * s[n-1] + * + * using the fact that s[n] of real x[n] is real. + * + */ + double dcos = 2. * cos(L*w); + int start = ((halfNumCoef)*L + 1); + SineGen cc((start - L) * w, w, true); // cosine + SineGen cp(start * w, w, true); // cosine + for (int i=1 ; i<=L ; ++i) { + double sc = 0; + double sp = 0; + for (int j=0 ; j<halfNumCoef ; ++j) { + double tmp = sc; + sc = static_cast<double>(*coef++) + dcos*sc - sp; + sp = tmp; + } + // If we are awfully clever, we can apply Goertzel's algorithm + // again on the sc and sp sequences returned here. + accum += cc.valueAdvance() * sc - cp.valueAdvance() * sp; + } +#endif + return accum*2.; +} + +/* + * evaluates the minimum and maximum |H(f)| bound in a band region. + * + * This is usually done with equally spaced increments in the target band in question. + * The passband is often very small, and sampled that way. The stopband is often much + * larger. + * + * We use the fact that the overall polyphase filter has an additional bank at the end + * for interpolation; hence it is overspecified for the H(f) computation. Thus the + * first polyphase is never actually checked, excepting its first term. + * + * In this code we use the firTransfer() evaluator above, which uses Goertzel's + * algorithm to calculate the transfer function at each point. + * + * TODO: An alternative with equal spacing is the FFT/DFT. An alternative with unequal + * spacing is a chirp transform. + * + * @param coef is the designed polyphase filter banks + * + * @param L is the number of phases (for interpolation) + * + * @param halfNumCoef should be half the number of coefficients for a single + * polyphase. + * + * @param fstart is the normalized frequency start. + * + * @param fend is the normalized frequency end. + * + * @param steps is the number of steps to take (sampling) between frequency start and end + * + * @param firMin returns the minimum transfer |H(f)| found + * + * @param firMax returns the maximum transfer |H(f)| found + * + * 0 <= f <= 0.5. + * This is used to test passband and stopband performance. + */ +template <typename T> +static void testFir(const T* coef, int L, int halfNumCoef, + double fstart, double fend, int steps, double &firMin, double &firMax) { + double wstart = fstart*(2.*M_PI); + double wend = fend*(2.*M_PI); + double wstep = (wend - wstart)/steps; + double fmax, fmin; + double trf = firTransfer(coef, L, halfNumCoef, wstart); + if (trf<0) { + trf = -trf; + } + fmin = fmax = trf; + wstart += wstep; + for (int i=1; i<steps; ++i) { + trf = firTransfer(coef, L, halfNumCoef, wstart); + if (trf<0) { + trf = -trf; + } + if (trf>fmax) { + fmax = trf; + } + else if (trf<fmin) { + fmin = trf; + } + wstart += wstep; + } + // renormalize - this is only needed for integer filter types + double norm = 1./((1ULL<<(sizeof(T)*8-1))*L); + + firMin = fmin * norm; + firMax = fmax * norm; +} + +/* + * evaluates the |H(f)| lowpass band characteristics. + * + * This function tests the lowpass characteristics for the overall polyphase filter, + * and is used to verify the design. For this case, fp should be set to the + * passband normalized frequency from 0 to 0.5 for the overall filter (thus it + * is the designed polyphase bank value / L). Likewise for fs. + * + * @param coef is the designed polyphase filter banks + * + * @param L is the number of phases (for interpolation) + * + * @param halfNumCoef should be half the number of coefficients for a single + * polyphase. + * + * @param fp is the passband normalized frequency, 0 < fp < fs < 0.5. + * + * @param fs is the stopband normalized frequency, 0 < fp < fs < 0.5. + * + * @param passSteps is the number of passband sampling steps. + * + * @param stopSteps is the number of stopband sampling steps. + * + * @param passMin is the minimum value in the passband + * + * @param passMax is the maximum value in the passband (useful for scaling). This should + * be less than 1., to avoid sine wave test overflow. + * + * @param passRipple is the passband ripple. Typically this should be less than 0.1 for + * an audio filter. Generally speaker/headphone device characteristics will dominate + * the passband term. + * + * @param stopMax is the maximum value in the stopband. + * + * @param stopRipple is the stopband ripple, also known as stopband attenuation. + * Typically this should be greater than ~80dB for low quality, and greater than + * ~100dB for full 16b quality, otherwise aliasing may become noticeable. + * + */ +template <typename T> +static void testFir(const T* coef, int L, int halfNumCoef, + double fp, double fs, int passSteps, int stopSteps, + double &passMin, double &passMax, double &passRipple, + double &stopMax, double &stopRipple) { + double fmin, fmax; + testFir(coef, L, halfNumCoef, 0., fp, passSteps, fmin, fmax); + double d1 = (fmax - fmin)/2.; + passMin = fmin; + passMax = fmax; + passRipple = -20.*log10(1. - d1); // passband ripple + testFir(coef, L, halfNumCoef, fs, 0.5, stopSteps, fmin, fmax); + // fmin is really not important for the stopband. + stopMax = fmax; + stopRipple = -20.*log10(fmax); // stopband ripple/attenuation +} + +/* + * Calculates the overall polyphase filter based on a windowed sinc function. + * + * The windowed sinc is an odd length symmetric filter of exactly L*halfNumCoef*2+1 + * taps for the entire kernel. This is then decomposed into L+1 polyphase filterbanks. + * The last filterbank is used for interpolation purposes (and is mostly composed + * of the first bank shifted by one sample), and is unnecessary if one does + * not do interpolation. + * + * We use the last filterbank for some transfer function calculation purposes, + * so it needs to be generated anyways. + * + * @param coef is the caller allocated space for coefficients. This should be + * exactly (L+1)*halfNumCoef in size. + * + * @param L is the number of phases (for interpolation) + * + * @param halfNumCoef should be half the number of coefficients for a single + * polyphase. + * + * @param stopBandAtten is the stopband value, should be >50dB. + * + * @param fcr is cutoff frequency/sampling rate (<0.5). At this point, the energy + * should be 6dB less. (fcr is where the amplitude drops by half). Use the + * firKaiserTbw() to calculate the transition bandwidth. fcr is the midpoint + * between the stop band and the pass band (fstop+fpass)/2. + * + * @param atten is the attenuation (generally slightly less than 1). + */ + +template <typename T> +static inline void firKaiserGen(T* coef, int L, int halfNumCoef, + double stopBandAtten, double fcr, double atten) { + // + // Formula 3.2.5, 3.2.7, Vaidyanathan, _Multirate Systems and Filter Banks_, p. 48 + // Formula 7.75, Oppenheim and Schafer, _Discrete-time Signal Processing, 3e_, p. 542 + // + // See also: http://melodi.ee.washington.edu/courses/ee518/notes/lec17.pdf + // + // Kaiser window and beta parameter + // + // | 0.1102*(A - 8.7) A > 50 + // beta = | 0.5842*(A - 21)^0.4 + 0.07886*(A - 21) 21 <= A <= 50 + // | 0. A < 21 + // + // with A is the desired stop-band attenuation in dBFS + // + // 30 dB 2.210 + // 40 dB 3.384 + // 50 dB 4.538 + // 60 dB 5.658 + // 70 dB 6.764 + // 80 dB 7.865 + // 90 dB 8.960 + // 100 dB 10.056 + + const int N = L * halfNumCoef; // non-negative half + const double beta = 0.1102 * (stopBandAtten - 8.7); // >= 50dB always + const double xstep = (2. * M_PI) * fcr / L; + const double xfrac = 1. / N; + const double yscale = atten * L / (I0(beta) * M_PI); + const double sqrbeta = sqr(beta); + + // We use sine generators, which computes sines on regular step intervals. + // This speeds up overall computation about 40% from computing the sine directly. + + SineGenGen sgg(0., xstep, L*xstep); // generates sine generators (one per polyphase) + + for (int i=0 ; i<=L ; ++i) { // generate an extra set of coefs for interpolation + + // computation for a single polyphase of the overall filter. + SineGen sg = sgg.valueAdvance(); // current sine generator for "j" inner loop. + double err = 0; // for noise shaping on int16_t coefficients (over each polyphase) + + for (int j=0, ix=i ; j<halfNumCoef ; ++j, ix+=L) { + double y; + if (CC_LIKELY(ix)) { + double x = static_cast<double>(ix); + + // sine generator: sg.valueAdvance() returns sin(ix*xstep); + // y = I0(beta * sqrt(1.0 - sqr(x * xfrac))) * yscale * sg.valueAdvance() / x; + y = I0SqrRat(sqrbeta * (1.0 - sqr(x * xfrac)), yscale * sg.valueAdvance(), x); + } else { + y = 2. * atten * fcr; // center of filter, sinc(0) = 1. + sg.advance(); + } + + if (is_same<T, int16_t>::value) { // int16_t needs noise shaping + *coef++ = static_cast<T>(toint(y, 1ULL<<(sizeof(T)*8-1), err)); + } else if (is_same<T, int32_t>::value) { + *coef++ = static_cast<T>(toint(y, 1ULL<<(sizeof(T)*8-1))); + } else { // assumed float or double + *coef++ = static_cast<T>(y); + } + } + } +} + +}; // namespace android + +#endif /*ANDROID_AUDIO_RESAMPLER_FIR_GEN_H*/ |