Fix multichannel integer resampling and add tests
Change-Id: I384bf8317d4f03616bf9f2b458a8700965d5cf56
diff --git a/services/audioflinger/AudioResamplerFirProcess.h b/services/audioflinger/AudioResamplerFirProcess.h
index bb0f1c9..d130013 100644
--- a/services/audioflinger/AudioResamplerFirProcess.h
+++ b/services/audioflinger/AudioResamplerFirProcess.h
@@ -109,40 +109,25 @@
}
};
-/*
- * 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 @@
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);
}
diff --git a/services/audioflinger/tests/resampler_tests.cpp b/services/audioflinger/tests/resampler_tests.cpp
index 8624b62..169ce02 100644
--- a/services/audioflinger/tests/resampler_tests.cpp
+++ b/services/audioflinger/tests/resampler_tests.cpp
@@ -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 @@
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 @@
// 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* 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 @@
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 +297,7 @@
* 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,
@@ -307,13 +312,100 @@
// 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>(
+ 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(2, 48000, 22101, 7000, 15000, 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,
+ //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>(
+ 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<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]);
+ }
+}
+