diff options
| -rw-r--r-- | services/audioflinger/AudioResampler.cpp | 11 | ||||
| -rw-r--r-- | services/audioflinger/AudioResamplerDyn.cpp | 106 | ||||
| -rw-r--r-- | services/audioflinger/AudioResamplerDyn.h | 4 | ||||
| -rw-r--r-- | services/audioflinger/tests/resampler_tests.cpp | 65 |
4 files changed, 130 insertions, 56 deletions
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp index 562c4ea..b8a0357 100644 --- a/services/audioflinger/AudioResampler.cpp +++ b/services/audioflinger/AudioResampler.cpp @@ -259,13 +259,14 @@ AudioResampler::AudioResampler(int bitDepth, int inChannelCount, mPhaseFraction(0), mLocalTimeFreq(0), mPTS(AudioBufferProvider::kInvalidPTS), mQuality(quality) { // sanity check on format - if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) { - ALOGE("Unsupported sample format, %d bits, %d channels", bitDepth, - inChannelCount); - // ALOG_ASSERT(0); + if ((bitDepth != 16 && (quality < DYN_LOW_QUALITY || bitDepth != 32)) + || inChannelCount < 1 + || inChannelCount > (quality < DYN_LOW_QUALITY ? 2 : 8)) { + LOG_ALWAYS_FATAL("Unsupported sample format %d quality %d bits, %d channels", + quality, bitDepth, inChannelCount); } if (sampleRate <= 0) { - ALOGE("Unsupported sample rate %d Hz", sampleRate); + LOG_ALWAYS_FATAL("Unsupported sample rate %d Hz", sampleRate); } // initialize common members diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp index 318eb57..7ca10c1 100644 --- a/services/audioflinger/AudioResamplerDyn.cpp +++ b/services/audioflinger/AudioResamplerDyn.cpp @@ -38,11 +38,6 @@ namespace android { -// generate a unique resample type compile-time constant (constexpr) -#define RESAMPLETYPE(CHANNELS, LOCKED, STRIDE) \ - ((((CHANNELS)-1)&1) | !!(LOCKED)<<1 \ - | ((STRIDE)==8 ? 1 : (STRIDE)==16 ? 2 : 0)<<2) - /* * InBuffer is a type agnostic input buffer. * @@ -403,12 +398,76 @@ void AudioResamplerDyn<TC, TI, TO>::setSampleRate(int32_t inSampleRate) // determine which resampler to use // check if locked phase (works only if mPhaseIncrement has no "fractional phase bits") int locked = (mPhaseIncrement << (sizeof(mPhaseIncrement)*8 - c.mShift)) == 0; - int stride = (c.mHalfNumCoefs&7)==0 ? 16 : (c.mHalfNumCoefs&3)==0 ? 8 : 2; if (locked) { mPhaseFraction = mPhaseFraction >> c.mShift << c.mShift; // remove fractional phase } - setResampler(RESAMPLETYPE(mChannelCount, locked, stride)); + // stride is the minimum number of filter coefficients processed per loop iteration. + // We currently only allow a stride of 16 to match with SIMD processing. + // This means that the filter length must be a multiple of 16, + // or half the filter length (mHalfNumCoefs) must be a multiple of 8. + // + // Note: A stride of 2 is achieved with non-SIMD processing. + int stride = ((c.mHalfNumCoefs & 7) == 0) ? 16 : 2; + LOG_ALWAYS_FATAL_IF(stride < 16, "Resampler stride must be 16 or more"); + LOG_ALWAYS_FATAL_IF(mChannelCount > 8 || mChannelCount < 1, + "Resampler channels(%d) must be between 1 to 8", mChannelCount); + // stride 16 (falls back to stride 2 for machines that do not support NEON) + if (locked) { + switch (mChannelCount) { + case 1: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, true, 16>; + break; + case 2: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, true, 16>; + break; + case 3: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<3, true, 16>; + break; + case 4: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<4, true, 16>; + break; + case 5: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<5, true, 16>; + break; + case 6: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<6, true, 16>; + break; + case 7: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<7, true, 16>; + break; + case 8: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<8, true, 16>; + break; + } + } else { + switch (mChannelCount) { + case 1: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, false, 16>; + break; + case 2: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, false, 16>; + break; + case 3: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<3, false, 16>; + break; + case 4: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<4, false, 16>; + break; + case 5: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<5, false, 16>; + break; + case 6: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<6, false, 16>; + break; + case 7: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<7, false, 16>; + break; + case 8: + mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<8, false, 16>; + break; + } + } #ifdef DEBUG_RESAMPLER printf("channels:%d %s stride:%d %s coef:%d shift:%d\n", mChannelCount, locked ? "locked" : "interpolated", @@ -424,34 +483,12 @@ void AudioResamplerDyn<TC, TI, TO>::resample(int32_t* out, size_t outFrameCount, } template<typename TC, typename TI, typename TO> -void AudioResamplerDyn<TC, TI, TO>::setResampler(unsigned resampleType) -{ - // stride 16 (falls back to stride 2 for machines that do not support NEON) - switch (resampleType) { - case RESAMPLETYPE(1, true, 16): - mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, true, 16>; - return; - case RESAMPLETYPE(2, true, 16): - mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, true, 16>; - return; - case RESAMPLETYPE(1, false, 16): - mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, false, 16>; - return; - case RESAMPLETYPE(2, false, 16): - mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, false, 16>; - return; - default: - LOG_ALWAYS_FATAL("Invalid resampler type: %u", resampleType); - mResampleFunc = NULL; - return; - } -} - -template<typename TC, typename TI, typename TO> template<int CHANNELS, bool LOCKED, int STRIDE> void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider) { + // TODO Mono -> Mono is not supported. OUTPUT_CHANNELS reflects minimum of stereo out. + const int OUTPUT_CHANNELS = (CHANNELS < 2) ? 2 : CHANNELS; const Constants& c(mConstants); const TC* const coefs = mConstants.mFirCoefs; TI* impulse = mInBuffer.getImpulse(); @@ -459,7 +496,7 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount, uint32_t phaseFraction = mPhaseFraction; const uint32_t phaseIncrement = mPhaseIncrement; size_t outputIndex = 0; - size_t outputSampleCount = outFrameCount * 2; // stereo output + size_t outputSampleCount = outFrameCount * OUTPUT_CHANNELS; const uint32_t phaseWrapLimit = c.mL << c.mShift; size_t inFrameCount = (phaseIncrement * (uint64_t)outFrameCount + phaseFraction) / phaseWrapLimit; @@ -490,7 +527,7 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount, while (mBuffer.frameCount == 0 && inFrameCount > 0) { mBuffer.frameCount = inFrameCount; provider->getNextBuffer(&mBuffer, - calculateOutputPTS(outputIndex / 2)); + calculateOutputPTS(outputIndex / OUTPUT_CHANNELS)); if (mBuffer.raw == NULL) { goto resample_exit; } @@ -538,7 +575,8 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount, phaseFraction, phaseWrapLimit, coefShift, halfNumCoefs, coefs, impulse, volumeSimd); - outputIndex += 2; + + outputIndex += OUTPUT_CHANNELS; phaseFraction += phaseIncrement; while (phaseFraction >= phaseWrapLimit) { diff --git a/services/audioflinger/AudioResamplerDyn.h b/services/audioflinger/AudioResamplerDyn.h index 8c56319..3dced8a 100644 --- a/services/audioflinger/AudioResamplerDyn.h +++ b/services/audioflinger/AudioResamplerDyn.h @@ -110,12 +110,10 @@ private: void createKaiserFir(Constants &c, double stopBandAtten, int inSampleRate, int outSampleRate, double tbwCheat); - void setResampler(unsigned resampleType); - template<int CHANNELS, bool LOCKED, int STRIDE> void resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider); - // declare a pointer to member function for resample + // define a pointer to member function type for resample typedef void (AudioResamplerDyn<TC, TI, TO>::*resample_ABP_t)(TO* out, size_t outFrameCount, AudioBufferProvider* provider); diff --git a/services/audioflinger/tests/resampler_tests.cpp b/services/audioflinger/tests/resampler_tests.cpp index 4a67d0b..d76c376 100644 --- a/services/audioflinger/tests/resampler_tests.cpp +++ b/services/audioflinger/tests/resampler_tests.cpp @@ -35,7 +35,8 @@ #include "AudioResampler.h" #include "test_utils.h" -void resample(void *output, size_t outputFrames, const std::vector<size_t> &outputIncr, +void resample(int channels, void *output, + size_t outputFrames, const std::vector<size_t> &outputIncr, android::AudioBufferProvider *provider, android::AudioResampler *resampler) { for (size_t i = 0, j = 0; i < outputFrames; ) { @@ -46,7 +47,7 @@ void resample(void *output, size_t outputFrames, const std::vector<size_t> &outp if (thisFrames == 0 || thisFrames > outputFrames - i) { thisFrames = outputFrames - i; } - resampler->resample((int32_t*) output + 2*i, thisFrames, provider); + resampler->resample((int32_t*) output + channels*i, thisFrames, provider); i += thisFrames; } } @@ -64,19 +65,26 @@ void buffercmp(const void *reference, const void *test, } } -void testBufferIncrement(size_t channels, unsigned inputFreq, unsigned outputFreq, +void testBufferIncrement(size_t channels, bool useFloat, + unsigned inputFreq, unsigned outputFreq, enum android::AudioResampler::src_quality quality) { + const int bits = useFloat ? 32 : 16; // create the provider std::vector<int> inputIncr; SignalProvider provider; - provider.setChirp<int16_t>(channels, - 0., outputFreq/2., outputFreq, outputFreq/2000.); + if (useFloat) { + provider.setChirp<float>(channels, + 0., outputFreq/2., outputFreq, outputFreq/2000.); + } else { + provider.setChirp<int16_t>(channels, + 0., outputFreq/2., outputFreq, outputFreq/2000.); + } provider.setIncr(inputIncr); // calculate the output size size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq; - size_t outputFrameSize = 2 * sizeof(int32_t); + size_t outputFrameSize = channels * (useFloat ? sizeof(float) : sizeof(int32_t)); size_t outputSize = outputFrameSize * outputFrames; outputSize &= ~7; @@ -84,7 +92,7 @@ void testBufferIncrement(size_t channels, unsigned inputFreq, unsigned outputFre const int volumePrecision = 12; /* typical unity gain */ android::AudioResampler* resampler; - resampler = android::AudioResampler::create(16, channels, outputFreq, quality); + resampler = android::AudioResampler::create(bits, channels, outputFreq, quality); resampler->setSampleRate(inputFreq); resampler->setVolume(1 << volumePrecision, 1 << volumePrecision); @@ -92,7 +100,7 @@ void testBufferIncrement(size_t channels, unsigned inputFreq, unsigned outputFre std::vector<size_t> refIncr; refIncr.push_back(outputFrames); void* reference = malloc(outputSize); - resample(reference, outputFrames, refIncr, &provider, resampler); + resample(channels, reference, outputFrames, refIncr, &provider, resampler); provider.reset(); @@ -101,7 +109,7 @@ void testBufferIncrement(size_t channels, unsigned inputFreq, unsigned outputFre resampler->reset(); #else delete resampler; - resampler = android::AudioResampler::create(16, channels, outputFreq, quality); + resampler = android::AudioResampler::create(bits, channels, outputFreq, quality); resampler->setSampleRate(inputFreq); resampler->setVolume(1 << volumePrecision, 1 << volumePrecision); #endif @@ -112,7 +120,10 @@ void testBufferIncrement(size_t channels, unsigned inputFreq, unsigned outputFre outIncr.push_back(2); outIncr.push_back(3); void* test = malloc(outputSize); - resample(test, outputFrames, outIncr, &provider, resampler); + inputIncr.push_back(1); + inputIncr.push_back(3); + provider.setIncr(inputIncr); + resample(channels, test, outputFrames, outIncr, &provider, resampler); // check buffercmp(reference, test, outputFrameSize, outputFrames); @@ -155,7 +166,7 @@ void testStopbandDownconversion(size_t channels, // calculate the output size size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq; - size_t outputFrameSize = 2 * sizeof(int32_t); + size_t outputFrameSize = channels * sizeof(int32_t); size_t outputSize = outputFrameSize * outputFrames; outputSize &= ~7; @@ -171,7 +182,7 @@ void testStopbandDownconversion(size_t channels, std::vector<size_t> refIncr; refIncr.push_back(outputFrames); void* reference = malloc(outputSize); - resample(reference, outputFrames, refIncr, &provider, resampler); + resample(channels, reference, outputFrames, refIncr, &provider, resampler); int32_t *out = reinterpret_cast<int32_t *>(reference); @@ -226,7 +237,7 @@ TEST(audioflinger_resampler, bufferincrement_fixedphase) { }; for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { - testBufferIncrement(2, 48000, 32000, kQualityArray[i]); + testBufferIncrement(2, false, 48000, 32000, kQualityArray[i]); } } @@ -243,7 +254,33 @@ TEST(audioflinger_resampler, bufferincrement_interpolatedphase) { }; for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { - testBufferIncrement(2, 22050, 48000, kQualityArray[i]); + testBufferIncrement(2, false, 22050, 48000, kQualityArray[i]); + } +} + +TEST(audioflinger_resampler, bufferincrement_fixedphase_multi) { + // only dynamic quality + static const enum android::AudioResampler::src_quality kQualityArray[] = { + android::AudioResampler::DYN_LOW_QUALITY, + android::AudioResampler::DYN_MED_QUALITY, + android::AudioResampler::DYN_HIGH_QUALITY, + }; + + for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { + testBufferIncrement(4, false, 48000, 32000, kQualityArray[i]); + } +} + +TEST(audioflinger_resampler, bufferincrement_interpolatedphase_multi_float) { + // only dynamic quality + static const enum android::AudioResampler::src_quality kQualityArray[] = { + android::AudioResampler::DYN_LOW_QUALITY, + android::AudioResampler::DYN_MED_QUALITY, + android::AudioResampler::DYN_HIGH_QUALITY, + }; + + for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { + testBufferIncrement(8, true, 22050, 48000, kQualityArray[i]); } } |