AudioResampler: Add configurable resampler design
Can be modified through properties.
Parameters enabled for 48kHz output sample rate and higher.
Default 90dB stopband attenuation.
Default cutoff frequency at Nyquist.
Test: native gtest resampler_tests
Bug: 66091148
Change-Id: Ieb8a959b7473613ef21121ccea0aa7ea3e2f1210
diff --git a/media/libaudioprocessing/AudioResamplerFirGen.h b/media/libaudioprocessing/AudioResamplerFirGen.h
index ad18965..39cafeb 100644
--- a/media/libaudioprocessing/AudioResamplerFirGen.h
+++ b/media/libaudioprocessing/AudioResamplerFirGen.h
@@ -546,8 +546,9 @@
}
wstart += wstep;
}
- // renormalize - this is only needed for integer filter types
- double norm = 1./((1ULL<<(sizeof(T)*8-1))*L);
+ // renormalize - this is needed for integer filter types, use 1 for float or double.
+ constexpr int64_t integralShift = std::is_integral<T>::value ? (sizeof(T) * 8 - 1) : 0;
+ const double norm = 1. / (L << integralShift);
firMin = fmin * norm;
firMax = fmax * norm;
@@ -557,9 +558,12 @@
* 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
+ * and is used to verify the design.
+ *
+ * For a polyphase filter (L > 1), typically 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.
+ * Similarly the stopSteps should be L * passSteps for equivalent accuracy.
*
* @param coef is the designed polyphase filter banks
*
@@ -610,6 +614,74 @@
}
/*
+ * Estimate the windowed sinc minimum passband value.
+ *
+ * This is the minimum value for a windowed sinc filter in its passband,
+ * which is identical to the scaling required not to cause overflow of a 0dBFS signal.
+ * The actual value used to attenuate the filter amplitude should be slightly
+ * smaller than this (suggest squaring) as this is just an estimate.
+ *
+ * As a windowed sinc has a passband ripple commensurate to the stopband attenuation
+ * due to Gibb's phenomenon from truncating the sinc, we derive this value from
+ * the design stopbandAttenuationDb (a positive value).
+ */
+static inline double computeWindowedSincMinimumPassbandValue(
+ double stopBandAttenuationDb) {
+ return 1. - pow(10. /* base */, stopBandAttenuationDb * (-1. / 20.));
+}
+
+/*
+ * Compute the windowed sinc passband ripple from stopband attenuation.
+ *
+ * As a windowed sinc has an passband ripple commensurate to the stopband attenuation
+ * due to Gibb's phenomenon from truncating the sinc, we derive this value from
+ * the design stopbandAttenuationDb (a positive value).
+ */
+static inline double computeWindowedSincPassbandRippleDb(
+ double stopBandAttenuationDb) {
+ return -20. * log10(computeWindowedSincMinimumPassbandValue(stopBandAttenuationDb));
+}
+
+/*
+ * Kaiser window Beta value
+ *
+ * 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 positive 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
+ *
+ * For some values of stopBandAttenuationDb the function may be computed
+ * at compile time.
+ */
+static inline constexpr double computeBeta(double stopBandAttenuationDb) {
+ if (stopBandAttenuationDb > 50.) {
+ return 0.1102 * (stopBandAttenuationDb - 8.7);
+ }
+ const double offset = stopBandAttenuationDb - 21.;
+ if (offset > 0.) {
+ return 0.5842 * pow(offset, 0.4) + 0.07886 * offset;
+ }
+ return 0.;
+}
+
+/*
* 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
@@ -642,31 +714,8 @@
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 beta = computeBeta(stopBandAtten);
const double xstep = (2. * M_PI) * fcr / L;
const double xfrac = 1. / N;
const double yscale = atten * L / (I0(beta) * M_PI);
@@ -696,9 +745,9 @@
sg.advance();
}
- if (is_same<T, int16_t>::value) { // int16_t needs noise shaping
+ if (std::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) {
+ } else if (std::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);