Fix multichannel integer resampling and add tests (42b01116) · Commits · e / os / android_frameworks_av

services/audioflinger/AudioResamplerFirProcess.h

+10 −26

Original line number	Diff line number	Diff line
		@@ -109,40 +109,25 @@ public:
		}
		};

		/*
		* Helper template functions for interpolating filter coefficients.
		*/

		template<typename TC, typename T>
		void adjustLerp(T& lerpP __unused)
		{
		}

		template<int32_t, typename T>
		void adjustLerp(T& lerpP)
		{
		lerpP >>= 16; // lerpP is 32bit for NEON int32_t, but always 16 bit for non-NEON path
		}

		template<typename TC, typename TINTERP>
		static inline
		inline
		TC interpolate(TC coef_0, TC coef_1, TINTERP lerp)
		{
		return lerp * (coef_1 - coef_0) + coef_0;
		}

		template<int16_t, uint32_t>
		static inline
		int16_t interpolate(int16_t coef_0, int16_t coef_1, uint32_t lerp)
		{
		return (static_cast<int16_t>(lerp) * ((coef_1-coef_0)<<1)>>16) + coef_0;
		template<>
		inline
		int16_t interpolate<int16_t, uint32_t>(int16_t coef_0, int16_t coef_1, uint32_t lerp)
		{ // in some CPU architectures 16b x 16b multiplies are faster.
		return (static_cast<int16_t>(lerp) * static_cast<int16_t>(coef_1 - coef_0) >> 15) + coef_0;
		}

		template<int32_t, uint32_t>
		static inline
		int32_t interpolate(int32_t coef_0, int32_t coef_1, uint32_t lerp)
		template<>
		inline
		int32_t interpolate<int32_t, uint32_t>(int32_t coef_0, int32_t coef_1, uint32_t lerp)
		{
		return mulAdd(static_cast<int16_t>(lerp), (coef_1-coef_0)<<1, coef_0);
		return (lerp * static_cast<int64_t>(coef_1 - coef_0) >> 31) + coef_0;
		}

		/* class scope for passing in functions into templates */
		@@ -283,7 +268,6 @@ void Process(TO* const out,
		TINTERP lerpP,
		const TO* const volumeLR)
		{
		adjustLerp<TC, TINTERP>(lerpP); // coefficient type adjustment for interpolations
		ProcessBase<CHANNELS, STRIDE, InterpCompute>(out, count, coefsP, coefsN, sP, sN, lerpP, volumeLR);
		}

services/audioflinger/tests/resampler_tests.cpp

+101 −9

Original line number	Diff line number	Diff line
		@@ -29,6 +29,7 @@
		#include <math.h>
		#include <vector>
		#include <utility>
		#include <iostream>
		#include <cutils/log.h>
		#include <gtest/gtest.h>
		#include <media/AudioBufferProvider.h>
		@@ -153,6 +154,9 @@ double signalEnergy(T start, T end, unsigned stride)
		return accum / count;
		}

		// TI = resampler input type, int16_t or float
		// TO = resampler output type, int32_t or float
		template <typename TI, typename TO>
		void testStopbandDownconversion(size_t channels,
		unsigned inputFreq, unsigned outputFreq,
		unsigned passband, unsigned stopband,
		@@ -161,20 +165,21 @@ void testStopbandDownconversion(size_t channels,
		// create the provider
		std::vector<int> inputIncr;
		SignalProvider provider;
		provider.setChirp<int16_t>(channels,
		provider.setChirp<TI>(channels,
		0., inputFreq/2., inputFreq, inputFreq/2000.);
		provider.setIncr(inputIncr);

		// calculate the output size
		size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq;
		size_t outputFrameSize = channels * sizeof(int32_t);
		size_t outputFrameSize = channels * sizeof(TO);
		size_t outputSize = outputFrameSize * outputFrames;
		outputSize &= ~7;

		// create the resampler
		android::AudioResampler* resampler;

		resampler = android::AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT,
		resampler = android::AudioResampler::create(
		is_same<TI, int16_t>::value ? AUDIO_FORMAT_PCM_16_BIT : AUDIO_FORMAT_PCM_FLOAT,
		channels, outputFreq, quality);
		resampler->setSampleRate(inputFreq);
		resampler->setVolume(android::AudioResampler::UNITY_GAIN_FLOAT,
		@@ -186,7 +191,7 @@ void testStopbandDownconversion(size_t channels,
		void* reference = malloc(outputSize);
		resample(channels, reference, outputFrames, refIncr, &provider, resampler);

		int32_t out = reinterpret_cast<int32_t >(reference);
		TO out = reinterpret_cast<TO >(reference);

		// check signal energy in passband
		const unsigned passbandFrame = passband * outputFreq / 1000.;
		@@ -206,10 +211,10 @@ void testStopbandDownconversion(size_t channels,
		provider.getNumFrames(), outputFrames,
		passbandFrame, stopbandFrame, stopbandEnergy, passbandEnergy, dbAtten);
		for (size_t i = 0; i < 10; ++i) {
		printf("%d\n", out[i+passbandFrame*channels]);
		std::cout << out[i+passbandFrame*channels] << std::endl;
		}
		for (size_t i = 0; i < 10; ++i) {
		printf("%d\n", out[i+stopbandFrame*channels]);
		std::cout << out[i+stopbandFrame*channels] << std::endl;
		}
		#endif
		}
		@@ -292,7 +297,35 @@ TEST(audioflinger_resampler, bufferincrement_interpolatedphase_multi_float) {
		* are properly suppressed. It uses downsampling because the stopband can be
		* clearly isolated by input frequencies exceeding the output sample rate (nyquist).
		*/
		TEST(audioflinger_resampler, stopbandresponse) {
		TEST(audioflinger_resampler, stopbandresponse_integer) {
		// not all of these may work (old resamplers fail on downsampling)
		static const enum android::AudioResampler::src_quality kQualityArray[] = {
		//android::AudioResampler::LOW_QUALITY,
		//android::AudioResampler::MED_QUALITY,
		//android::AudioResampler::HIGH_QUALITY,
		//android::AudioResampler::VERY_HIGH_QUALITY,
		android::AudioResampler::DYN_LOW_QUALITY,
		android::AudioResampler::DYN_MED_QUALITY,
		android::AudioResampler::DYN_HIGH_QUALITY,
		};

		// in this test we assume a maximum transition band between 12kHz and 20kHz.
		// there must be at least 60dB relative attenuation between stopband and passband.
		for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
		testStopbandDownconversion<int16_t, int32_t>(
		2, 48000, 32000, 12000, 20000, kQualityArray[i]);
		}

		// in this test we assume a maximum transition band between 7kHz and 15kHz.
		// there must be at least 60dB relative attenuation between stopband and passband.
		// (the weird ratio triggers interpolative resampling)
		for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
		testStopbandDownconversion<int16_t, int32_t>(
		2, 48000, 22101, 7000, 15000, kQualityArray[i]);
		}
		}

		TEST(audioflinger_resampler, stopbandresponse_integer_multichannel) {
		// not all of these may work (old resamplers fail on downsampling)
		static const enum android::AudioResampler::src_quality kQualityArray[] = {
		//android::AudioResampler::LOW_QUALITY,
		@@ -307,13 +340,72 @@ TEST(audioflinger_resampler, stopbandresponse) {
		// in this test we assume a maximum transition band between 12kHz and 20kHz.
		// there must be at least 60dB relative attenuation between stopband and passband.
		for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
		testStopbandDownconversion(2, 48000, 32000, 12000, 20000, kQualityArray[i]);
		testStopbandDownconversion<int16_t, int32_t>(
		8, 48000, 32000, 12000, 20000, kQualityArray[i]);
		}

		// in this test we assume a maximum transition band between 7kHz and 15kHz.
		// there must be at least 60dB relative attenuation between stopband and passband.
		// (the weird ratio triggers interpolative resampling)
		for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
		testStopbandDownconversion(2, 48000, 22101, 7000, 15000, kQualityArray[i]);
		testStopbandDownconversion<int16_t, int32_t>(
		8, 48000, 22101, 7000, 15000, kQualityArray[i]);
		}
		}

		TEST(audioflinger_resampler, stopbandresponse_float) {
		// not all of these may work (old resamplers fail on downsampling)
		static const enum android::AudioResampler::src_quality kQualityArray[] = {
		//android::AudioResampler::LOW_QUALITY,
		//android::AudioResampler::MED_QUALITY,
		//android::AudioResampler::HIGH_QUALITY,
		//android::AudioResampler::VERY_HIGH_QUALITY,
		android::AudioResampler::DYN_LOW_QUALITY,
		android::AudioResampler::DYN_MED_QUALITY,
		android::AudioResampler::DYN_HIGH_QUALITY,
		};

		// in this test we assume a maximum transition band between 12kHz and 20kHz.
		// there must be at least 60dB relative attenuation between stopband and passband.
		for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
		testStopbandDownconversion<float, float>(
		2, 48000, 32000, 12000, 20000, kQualityArray[i]);
		}

		// in this test we assume a maximum transition band between 7kHz and 15kHz.
		// there must be at least 60dB relative attenuation between stopband and passband.
		// (the weird ratio triggers interpolative resampling)
		for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
		testStopbandDownconversion<float, float>(
		2, 48000, 22101, 7000, 15000, kQualityArray[i]);
		}
		}

		TEST(audioflinger_resampler, stopbandresponse_float_multichannel) {
		// not all of these may work (old resamplers fail on downsampling)
		static const enum android::AudioResampler::src_quality kQualityArray[] = {
		//android::AudioResampler::LOW_QUALITY,
		//android::AudioResampler::MED_QUALITY,
		//android::AudioResampler::HIGH_QUALITY,
		//android::AudioResampler::VERY_HIGH_QUALITY,
		android::AudioResampler::DYN_LOW_QUALITY,
		android::AudioResampler::DYN_MED_QUALITY,
		android::AudioResampler::DYN_HIGH_QUALITY,
		};

		// in this test we assume a maximum transition band between 12kHz and 20kHz.
		// there must be at least 60dB relative attenuation between stopband and passband.
		for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
		testStopbandDownconversion<float, float>(
		8, 48000, 32000, 12000, 20000, kQualityArray[i]);
		}

		// in this test we assume a maximum transition band between 7kHz and 15kHz.
		// there must be at least 60dB relative attenuation between stopband and passband.
		// (the weird ratio triggers interpolative resampling)
		for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
		testStopbandDownconversion<float, float>(
		8, 48000, 22101, 7000, 15000, kQualityArray[i]);
		}
		}