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Diffstat (limited to 'services/audioflinger/AudioResamplerDyn.cpp')
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1 files changed, 556 insertions, 0 deletions
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp new file mode 100644 index 0000000..3abe8fd --- /dev/null +++ b/services/audioflinger/AudioResamplerDyn.cpp @@ -0,0 +1,556 @@ +/* + * Copyright (C) 2013 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "AudioResamplerDyn" +//#define LOG_NDEBUG 0 + +#include <malloc.h> +#include <string.h> +#include <stdlib.h> +#include <dlfcn.h> +#include <math.h> + +#include <cutils/compiler.h> +#include <cutils/properties.h> +#include <utils/Debug.h> +#include <utils/Log.h> + +#include "AudioResamplerFirOps.h" // USE_NEON and USE_INLINE_ASSEMBLY defined here +#include "AudioResamplerFirProcess.h" +#include "AudioResamplerFirProcessNeon.h" +#include "AudioResamplerFirGen.h" // requires math.h +#include "AudioResamplerDyn.h" + +//#define DEBUG_RESAMPLER + +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. + * + * Layout of the state buffer for halfNumCoefs=8. + * + * [rrrrrrppppppppnnnnnnnnrrrrrrrrrrrrrrrrrrr.... rrrrrrr] + * S I R + * + * S = mState + * I = mImpulse + * R = mRingFull + * p = past samples, convoluted with the (p)ositive side of sinc() + * n = future samples, convoluted with the (n)egative side of sinc() + * r = extra space for implementing the ring buffer + */ + +template<typename TC, typename TI, typename TO> +AudioResamplerDyn<TC, TI, TO>::InBuffer::InBuffer() + : mState(NULL), mImpulse(NULL), mRingFull(NULL), mStateCount(0) +{ +} + +template<typename TC, typename TI, typename TO> +AudioResamplerDyn<TC, TI, TO>::InBuffer::~InBuffer() +{ + init(); +} + +template<typename TC, typename TI, typename TO> +void AudioResamplerDyn<TC, TI, TO>::InBuffer::init() +{ + free(mState); + mState = NULL; + mImpulse = NULL; + mRingFull = NULL; + mStateCount = 0; +} + +// resizes the state buffer to accommodate the appropriate filter length +template<typename TC, typename TI, typename TO> +void AudioResamplerDyn<TC, TI, TO>::InBuffer::resize(int CHANNELS, int halfNumCoefs) +{ + // calculate desired state size + int stateCount = halfNumCoefs * CHANNELS * 2 * kStateSizeMultipleOfFilterLength; + + // check if buffer needs resizing + if (mState + && stateCount == mStateCount + && mRingFull-mState == mStateCount-halfNumCoefs*CHANNELS) { + return; + } + + // create new buffer + TI* state; + (void)posix_memalign(reinterpret_cast<void**>(&state), 32, stateCount*sizeof(*state)); + memset(state, 0, stateCount*sizeof(*state)); + + // attempt to preserve state + if (mState) { + TI* srcLo = mImpulse - halfNumCoefs*CHANNELS; + TI* srcHi = mImpulse + halfNumCoefs*CHANNELS; + TI* dst = state; + + if (srcLo < mState) { + dst += mState-srcLo; + srcLo = mState; + } + if (srcHi > mState + mStateCount) { + srcHi = mState + mStateCount; + } + memcpy(dst, srcLo, (srcHi - srcLo) * sizeof(*srcLo)); + free(mState); + } + + // set class member vars + mState = state; + mStateCount = stateCount; + mImpulse = state + halfNumCoefs*CHANNELS; // actually one sample greater than needed + mRingFull = state + mStateCount - halfNumCoefs*CHANNELS; +} + +// copy in the input data into the head (impulse+halfNumCoefs) of the buffer. +template<typename TC, typename TI, typename TO> +template<int CHANNELS> +void AudioResamplerDyn<TC, TI, TO>::InBuffer::readAgain(TI*& impulse, const int halfNumCoefs, + const TI* const in, const size_t inputIndex) +{ + TI* head = impulse + halfNumCoefs*CHANNELS; + for (size_t i=0 ; i<CHANNELS ; i++) { + head[i] = in[inputIndex*CHANNELS + i]; + } +} + +// advance the impulse pointer, and load in data into the head (impulse+halfNumCoefs) +template<typename TC, typename TI, typename TO> +template<int CHANNELS> +void AudioResamplerDyn<TC, TI, TO>::InBuffer::readAdvance(TI*& impulse, const int halfNumCoefs, + const TI* const in, const size_t inputIndex) +{ + impulse += CHANNELS; + + if (CC_UNLIKELY(impulse >= mRingFull)) { + const size_t shiftDown = mRingFull - mState - halfNumCoefs*CHANNELS; + memcpy(mState, mState+shiftDown, halfNumCoefs*CHANNELS*2*sizeof(TI)); + impulse -= shiftDown; + } + readAgain<CHANNELS>(impulse, halfNumCoefs, in, inputIndex); +} + +template<typename TC, typename TI, typename TO> +void AudioResamplerDyn<TC, TI, TO>::Constants::set( + int L, int halfNumCoefs, int inSampleRate, int outSampleRate) +{ + int bits = 0; + int lscale = inSampleRate/outSampleRate < 2 ? L - 1 : + static_cast<int>(static_cast<uint64_t>(L)*inSampleRate/outSampleRate); + for (int i=lscale; i; ++bits, i>>=1) + ; + mL = L; + mShift = kNumPhaseBits - bits; + mHalfNumCoefs = halfNumCoefs; +} + +template<typename TC, typename TI, typename TO> +AudioResamplerDyn<TC, TI, TO>::AudioResamplerDyn(int bitDepth, + int inChannelCount, int32_t sampleRate, src_quality quality) + : AudioResampler(bitDepth, inChannelCount, sampleRate, quality), + mResampleFunc(0), mFilterSampleRate(0), mFilterQuality(DEFAULT_QUALITY), + mCoefBuffer(NULL) +{ + mVolumeSimd[0] = mVolumeSimd[1] = 0; + // The AudioResampler base class assumes we are always ready for 1:1 resampling. + // We reset mInSampleRate to 0, so setSampleRate() will calculate filters for + // setSampleRate() for 1:1. (May be removed if precalculated filters are used.) + mInSampleRate = 0; + mConstants.set(128, 8, mSampleRate, mSampleRate); // TODO: set better +} + +template<typename TC, typename TI, typename TO> +AudioResamplerDyn<TC, TI, TO>::~AudioResamplerDyn() +{ + free(mCoefBuffer); +} + +template<typename TC, typename TI, typename TO> +void AudioResamplerDyn<TC, TI, TO>::init() +{ + mFilterSampleRate = 0; // always trigger new filter generation + mInBuffer.init(); +} + +template<typename TC, typename TI, typename TO> +void AudioResamplerDyn<TC, TI, TO>::setVolume(int16_t left, int16_t right) +{ + AudioResampler::setVolume(left, right); + // volume is applied on the output type. + if (is_same<TO, float>::value || is_same<TO, double>::value) { + const TO scale = 1. / (1UL << 12); + mVolumeSimd[0] = static_cast<TO>(left) * scale; + mVolumeSimd[1] = static_cast<TO>(right) * scale; + } else { + mVolumeSimd[0] = static_cast<int32_t>(left) << 16; + mVolumeSimd[1] = static_cast<int32_t>(right) << 16; + } +} + +template<typename T> T max(T a, T b) {return a > b ? a : b;} + +template<typename T> T absdiff(T a, T b) {return a > b ? a - b : b - a;} + +template<typename TC, typename TI, typename TO> +void AudioResamplerDyn<TC, TI, TO>::createKaiserFir(Constants &c, + double stopBandAtten, int inSampleRate, int outSampleRate, double tbwCheat) +{ + TC* buf; + static const double atten = 0.9998; // to avoid ripple overflow + double fcr; + double tbw = firKaiserTbw(c.mHalfNumCoefs, stopBandAtten); + + (void)posix_memalign(reinterpret_cast<void**>(&buf), 32, (c.mL+1)*c.mHalfNumCoefs*sizeof(TC)); + if (inSampleRate < outSampleRate) { // upsample + fcr = max(0.5*tbwCheat - tbw/2, tbw/2); + } else { // downsample + fcr = max(0.5*tbwCheat*outSampleRate/inSampleRate - tbw/2, tbw/2); + } + // create and set filter + firKaiserGen(buf, c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, atten); + c.mFirCoefs = buf; + if (mCoefBuffer) { + free(mCoefBuffer); + } + mCoefBuffer = buf; +#ifdef DEBUG_RESAMPLER + // print basic filter stats + printf("L:%d hnc:%d stopBandAtten:%lf fcr:%lf atten:%lf tbw:%lf\n", + c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, atten, tbw); + // test the filter and report results + double fp = (fcr - tbw/2)/c.mL; + double fs = (fcr + tbw/2)/c.mL; + double passMin, passMax, passRipple; + double stopMax, stopRipple; + testFir(buf, c.mL, c.mHalfNumCoefs, fp, fs, /*passSteps*/ 1000, /*stopSteps*/ 100000, + passMin, passMax, passRipple, stopMax, stopRipple); + printf("passband(%lf, %lf): %.8lf %.8lf %.8lf\n", 0., fp, passMin, passMax, passRipple); + printf("stopband(%lf, %lf): %.8lf %.3lf\n", fs, 0.5, stopMax, stopRipple); +#endif +} + +// recursive gcd. Using objdump, it appears the tail recursion is converted to a while loop. +static int gcd(int n, int m) +{ + if (m == 0) { + return n; + } + return gcd(m, n % m); +} + +static bool isClose(int32_t newSampleRate, int32_t prevSampleRate, + int32_t filterSampleRate, int32_t outSampleRate) +{ + + // different upsampling ratios do not need a filter change. + if (filterSampleRate != 0 + && filterSampleRate < outSampleRate + && newSampleRate < outSampleRate) + return true; + + // check design criteria again if downsampling is detected. + int pdiff = absdiff(newSampleRate, prevSampleRate); + int adiff = absdiff(newSampleRate, filterSampleRate); + + // allow up to 6% relative change increments. + // allow up to 12% absolute change increments (from filter design) + return pdiff < prevSampleRate>>4 && adiff < filterSampleRate>>3; +} + +template<typename TC, typename TI, typename TO> +void AudioResamplerDyn<TC, TI, TO>::setSampleRate(int32_t inSampleRate) +{ + if (mInSampleRate == inSampleRate) { + return; + } + int32_t oldSampleRate = mInSampleRate; + int32_t oldHalfNumCoefs = mConstants.mHalfNumCoefs; + uint32_t oldPhaseWrapLimit = mConstants.mL << mConstants.mShift; + bool useS32 = false; + + mInSampleRate = inSampleRate; + + // TODO: Add precalculated Equiripple filters + + if (mFilterQuality != getQuality() || + !isClose(inSampleRate, oldSampleRate, mFilterSampleRate, mSampleRate)) { + mFilterSampleRate = inSampleRate; + mFilterQuality = getQuality(); + + // Begin Kaiser Filter computation + // + // The quantization floor for S16 is about 96db - 10*log_10(#length) + 3dB. + // Keep the stop band attenuation no greater than 84-85dB for 32 length S16 filters + // + // For s32 we keep the stop band attenuation at the same as 16b resolution, about + // 96-98dB + // + + double stopBandAtten; + double tbwCheat = 1.; // how much we "cheat" into aliasing + int halfLength; + if (mFilterQuality == DYN_HIGH_QUALITY) { + // 32b coefficients, 64 length + useS32 = true; + stopBandAtten = 98.; + if (inSampleRate >= mSampleRate * 4) { + halfLength = 48; + } else if (inSampleRate >= mSampleRate * 2) { + halfLength = 40; + } else { + halfLength = 32; + } + } else if (mFilterQuality == DYN_LOW_QUALITY) { + // 16b coefficients, 16-32 length + useS32 = false; + stopBandAtten = 80.; + if (inSampleRate >= mSampleRate * 4) { + halfLength = 24; + } else if (inSampleRate >= mSampleRate * 2) { + halfLength = 16; + } else { + halfLength = 8; + } + if (inSampleRate <= mSampleRate) { + tbwCheat = 1.05; + } else { + tbwCheat = 1.03; + } + } else { // DYN_MED_QUALITY + // 16b coefficients, 32-64 length + // note: > 64 length filters with 16b coefs can have quantization noise problems + useS32 = false; + stopBandAtten = 84.; + if (inSampleRate >= mSampleRate * 4) { + halfLength = 32; + } else if (inSampleRate >= mSampleRate * 2) { + halfLength = 24; + } else { + halfLength = 16; + } + if (inSampleRate <= mSampleRate) { + tbwCheat = 1.03; + } else { + tbwCheat = 1.01; + } + } + + // determine the number of polyphases in the filterbank. + // for 16b, it is desirable to have 2^(16/2) = 256 phases. + // https://ccrma.stanford.edu/~jos/resample/Relation_Interpolation_Error_Quantization.html + // + // We are a bit more lax on this. + + int phases = mSampleRate / gcd(mSampleRate, inSampleRate); + + // TODO: Once dynamic sample rate change is an option, the code below + // should be modified to execute only when dynamic sample rate change is enabled. + // + // as above, #phases less than 63 is too few phases for accurate linear interpolation. + // we increase the phases to compensate, but more phases means more memory per + // filter and more time to compute the filter. + // + // if we know that the filter will be used for dynamic sample rate changes, + // that would allow us skip this part for fixed sample rate resamplers. + // + while (phases<63) { + phases *= 2; // this code only needed to support dynamic rate changes + } + + if (phases>=256) { // too many phases, always interpolate + phases = 127; + } + + // create the filter + mConstants.set(phases, halfLength, inSampleRate, mSampleRate); + createKaiserFir(mConstants, stopBandAtten, + inSampleRate, mSampleRate, tbwCheat); + } // End Kaiser filter + + // update phase and state based on the new filter. + const Constants& c(mConstants); + mInBuffer.resize(mChannelCount, c.mHalfNumCoefs); + const uint32_t phaseWrapLimit = c.mL << c.mShift; + // try to preserve as much of the phase fraction as possible for on-the-fly changes + mPhaseFraction = static_cast<unsigned long long>(mPhaseFraction) + * phaseWrapLimit / oldPhaseWrapLimit; + mPhaseFraction %= phaseWrapLimit; // should not do anything, but just in case. + mPhaseIncrement = static_cast<uint32_t>(static_cast<double>(phaseWrapLimit) + * inSampleRate / mSampleRate); + + // 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)); +#ifdef DEBUG_RESAMPLER + printf("channels:%d %s stride:%d %s coef:%d shift:%d\n", + mChannelCount, locked ? "locked" : "interpolated", + stride, useS32 ? "S32" : "S16", 2*c.mHalfNumCoefs, c.mShift); +#endif +} + +template<typename TC, typename TI, typename TO> +void AudioResamplerDyn<TC, TI, TO>::resample(int32_t* out, size_t outFrameCount, + AudioBufferProvider* provider) +{ + (this->*mResampleFunc)(reinterpret_cast<TO*>(out), outFrameCount, provider); +} + +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) +{ + const Constants& c(mConstants); + const TC* const coefs = mConstants.mFirCoefs; + TI* impulse = mInBuffer.getImpulse(); + size_t inputIndex = mInputIndex; + uint32_t phaseFraction = mPhaseFraction; + const uint32_t phaseIncrement = mPhaseIncrement; + size_t outputIndex = 0; + size_t outputSampleCount = outFrameCount * 2; // stereo output + size_t inFrameCount = getInFrameCountRequired(outFrameCount); + const uint32_t phaseWrapLimit = c.mL << c.mShift; + + // NOTE: be very careful when modifying the code here. register + // pressure is very high and a small change might cause the compiler + // to generate far less efficient code. + // Always sanity check the result with objdump or test-resample. + + // the following logic is a bit convoluted to keep the main processing loop + // as tight as possible with register allocation. + while (outputIndex < outputSampleCount) { + // buffer is empty, fetch a new one + while (mBuffer.frameCount == 0) { + mBuffer.frameCount = inFrameCount; + provider->getNextBuffer(&mBuffer, + calculateOutputPTS(outputIndex / 2)); + if (mBuffer.raw == NULL) { + goto resample_exit; + } + if (phaseFraction >= phaseWrapLimit) { // read in data + mInBuffer.template readAdvance<CHANNELS>( + impulse, c.mHalfNumCoefs, + reinterpret_cast<TI*>(mBuffer.raw), inputIndex); + phaseFraction -= phaseWrapLimit; + while (phaseFraction >= phaseWrapLimit) { + inputIndex++; + if (inputIndex >= mBuffer.frameCount) { + inputIndex -= mBuffer.frameCount; + provider->releaseBuffer(&mBuffer); + break; + } + mInBuffer.template readAdvance<CHANNELS>( + impulse, c.mHalfNumCoefs, + reinterpret_cast<TI*>(mBuffer.raw), inputIndex); + phaseFraction -= phaseWrapLimit; + } + } + } + const TI* const in = reinterpret_cast<const TI*>(mBuffer.raw); + const size_t frameCount = mBuffer.frameCount; + const int coefShift = c.mShift; + const int halfNumCoefs = c.mHalfNumCoefs; + const TO* const volumeSimd = mVolumeSimd; + + // reread the last input in. + mInBuffer.template readAgain<CHANNELS>(impulse, halfNumCoefs, in, inputIndex); + + // main processing loop + while (CC_LIKELY(outputIndex < outputSampleCount)) { + // caution: fir() is inlined and may be large. + // output will be loaded with the appropriate values + // + // from the input samples in impulse[-halfNumCoefs+1]... impulse[halfNumCoefs] + // from the polyphase filter of (phaseFraction / phaseWrapLimit) in coefs. + // + fir<CHANNELS, LOCKED, STRIDE>( + &out[outputIndex], + phaseFraction, phaseWrapLimit, + coefShift, halfNumCoefs, coefs, + impulse, volumeSimd); + outputIndex += 2; + + phaseFraction += phaseIncrement; + while (phaseFraction >= phaseWrapLimit) { + inputIndex++; + if (inputIndex >= frameCount) { + goto done; // need a new buffer + } + mInBuffer.template readAdvance<CHANNELS>(impulse, halfNumCoefs, in, inputIndex); + phaseFraction -= phaseWrapLimit; + } + } +done: + // often arrives here when input buffer runs out + if (inputIndex >= frameCount) { + inputIndex -= frameCount; + provider->releaseBuffer(&mBuffer); + // mBuffer.frameCount MUST be zero here. + } + } + +resample_exit: + mInBuffer.setImpulse(impulse); + mInputIndex = inputIndex; + mPhaseFraction = phaseFraction; +} + +/* instantiate templates used by AudioResampler::create */ +template class AudioResamplerDyn<float, float, float>; +template class AudioResamplerDyn<int16_t, int16_t, int32_t>; +template class AudioResamplerDyn<int32_t, int16_t, int32_t>; + +// ---------------------------------------------------------------------------- +}; // namespace android |