Merge "Improve resampler speed for floating point and arm64" (df68f07c) · Commits · e / os / android_frameworks_av

services/audioflinger/AudioResamplerFirOps.h

+1 −1

Original line number	Diff line number	Diff line
		@@ -25,7 +25,7 @@ namespace android {
		#define USE_INLINE_ASSEMBLY (false)
		#endif

		#if USE_INLINE_ASSEMBLY && defined(__ARM_NEON__)
		#if defined(__aarch64__) \|\| defined(__ARM_NEON__)
		#define USE_NEON (true)
		#include <arm_neon.h>
		#else

services/audioflinger/AudioResamplerFirProcess.h

+38 −2

Original line number	Diff line number	Diff line
		@@ -243,6 +243,9 @@ void ProcessBase(TO* const out,
		}
		}

		/* Calculates a single output frame from a polyphase resampling filter.
		* See Process() for parameter details.
		*/
		template <int CHANNELS, int STRIDE, typename TC, typename TI, typename TO>
		static inline
		void ProcessL(TO* const out,
		@@ -256,6 +259,39 @@ void ProcessL(TO* const out,
		ProcessBase<CHANNELS, STRIDE, InterpNull>(out, count, coefsP, coefsN, sP, sN, 0, volumeLR);
		}

		/*
		* Calculates a single output frame from a polyphase resampling filter,
		* with filter phase interpolation.
		*
		* @param out should point to the output buffer with space for at least one output frame.
		*
		* @param count should be half the size of the total filter length (halfNumCoefs), as we
		* use symmetry in filter coefficients to evaluate two dot products.
		*
		* @param coefsP is one phase of the polyphase filter bank of size halfNumCoefs, corresponding
		* to the positive sP.
		*
		* @param coefsN is one phase of the polyphase filter bank of size halfNumCoefs, corresponding
		* to the negative sN.
		*
		* @param coefsP1 is the next phase of coefsP (used for interpolation).
		*
		* @param coefsN1 is the next phase of coefsN (used for interpolation).
		*
		* @param sP is the positive half of the coefficients (as viewed by a convolution),
		* starting at the original samples pointer and decrementing (by CHANNELS).
		*
		* @param sN is the negative half of the samples (as viewed by a convolution),
		* starting at the original samples pointer + CHANNELS and incrementing (by CHANNELS).
		*
		* @param lerpP The fractional siting between the polyphase indices is given by the bits
		* below coefShift. See fir() for details.
		*
		* @param volumeLR is a pointer to an array of two 32 bit volume values, one per stereo channel,
		* expressed as a S32 integer or float. A negative value inverts the channel 180 degrees.
		* The pointer volumeLR should be aligned to a minimum of 8 bytes.
		* A typical value for volume is 0x1000 to align to a unity gain output of 20.12.
		*/
		template <int CHANNELS, int STRIDE, typename TC, typename TI, typename TO, typename TINTERP>
		static inline
		void Process(TO* const out,
		@@ -274,7 +310,7 @@ void Process(TO* const out,
		}

		/*
		* Calculates a single output frame (two samples) from input sample pointer.
		* Calculates a single output frame from input sample pointer.
		*
		* This sets up the params for the accelerated Process() and ProcessL()
		* functions to do the appropriate dot products.
		@@ -309,7 +345,7 @@ void Process(TO* const out,
		* the positive half of the filter is dot product from samples to samples-halfNumCoefs+1.
		*
		* @param volumeLR is a pointer to an array of two 32 bit volume values, one per stereo channel,
		* expressed as a S32 integer. A negative value inverts the channel 180 degrees.
		* expressed as a S32 integer or float. A negative value inverts the channel 180 degrees.
		* The pointer volumeLR should be aligned to a minimum of 8 bytes.
		* A typical value for volume is 0x1000 to align to a unity gain output of 20.12.
		*

services/audioflinger/AudioResamplerFirProcessNeon.h

+577 −1

File changed.

Preview size limit exceeded, changes collapsed.

services/audioflinger/test-resample.cpp

+8 −0

Original line number	Diff line number	Diff line
		@@ -427,6 +427,14 @@ int main(int argc, char* argv[]) {
		printf("quality: %d channels: %d msec: %" PRId64 " Mfrms/s: %.2lf\n",
		quality, channels, time/1000000, output_frames * looplimit / (time / 1e9) / 1e6);
		resampler->reset();

		// TODO fix legacy bug: reset does not clear buffers.
		// delete and recreate resampler here.
		delete resampler;
		resampler = AudioResampler::create(format, channels,
		output_freq, quality);
		resampler->setSampleRate(input_freq);
		resampler->setVolume(AudioResampler::UNITY_GAIN_FLOAT, AudioResampler::UNITY_GAIN_FLOAT);
		}

		memset(output_vaddr, 0, output_size);

services/audioflinger/tests/build_and_run_all_unit_tests.sh

+1 −1

Original line number	Diff line number	Diff line
		@@ -15,7 +15,7 @@ mm
		echo "waiting for device"
		adb root && adb wait-for-device remount
		adb push $OUT/system/lib/libaudioresampler.so /system/lib
		adb push $OUT/system/bin/resampler_tests /system/bin
		adb push $OUT/data/nativetest/resampler_tests /system/bin

		sh $ANDROID_BUILD_TOP/frameworks/av/services/audioflinger/tests/run_all_unit_tests.sh