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Commit 6b667dde authored by Andy Hung's avatar Andy Hung
Browse files

Improve resampler speed for floating point and arm64 

Add floating point intrinsics for arm32 and arm64 devices.
Add integer intrinsics for arm64 devices.

Bug: 17366024
Change-Id: Id4240f549033deb262815c7145d69820e5fd7b92
parent bf31b330

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