/*
* 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 <audio_utils/primitives.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 {
/*
* 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
size_t stateCount = halfNumCoefs * CHANNELS * 2 * kStateSizeMultipleOfFilterLength;
// check if buffer needs resizing
if (mState
&& stateCount == mStateCount
&& mRingFull-mState == (ssize_t) (mStateCount-halfNumCoefs*CHANNELS)) {
return;
}
// create new buffer
TI* state = NULL;
(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 inChannelCount, int32_t sampleRate, src_quality quality)
: AudioResampler(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(float left, float right)
{
AudioResampler::setVolume(left, right);
if (is_same<TO, float>::value || is_same<TO, double>::value) {
mVolumeSimd[0] = static_cast<TO>(left);
mVolumeSimd[1] = static_cast<TO>(right);
} else { // integer requires scaling to U4_28 (rounding down)
// integer volumes are clamped to 0 to UNITY_GAIN so there
// are no issues with signed overflow.
mVolumeSimd[0] = u4_28_from_float(clampFloatVol(left));
mVolumeSimd[1] = u4_28_from_float(clampFloatVol(right));
}
}
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 = NULL;
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;
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<uint64_t>(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;
if (locked) {
mPhaseFraction = mPhaseFraction >> c.mShift << c.mShift; // remove fractional phase
}
// 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 < 1 || mChannelCount > 8,
"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",
stride, useS32 ? "S32" : "S16", 2*c.mHalfNumCoefs, c.mShift);
#endif
}
template<typename TC, typename TI, typename TO>
size_t AudioResamplerDyn<TC, TI, TO>::resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider)
{
return (this->*mResampleFunc)(reinterpret_cast<TO*>(out), outFrameCount, provider);
}
template<typename TC, typename TI, typename TO>
template<int CHANNELS, bool LOCKED, int STRIDE>
size_t 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();
size_t inputIndex = 0;
uint32_t phaseFraction = mPhaseFraction;
const uint32_t phaseIncrement = mPhaseIncrement;
size_t outputIndex = 0;
size_t outputSampleCount = outFrameCount * OUTPUT_CHANNELS;
const uint32_t phaseWrapLimit = c.mL << c.mShift;
size_t inFrameCount = (phaseIncrement * (uint64_t)outFrameCount + phaseFraction)
/ phaseWrapLimit;
// sanity check that inFrameCount is in signed 32 bit integer range.
ALOG_ASSERT(0 <= inFrameCount && inFrameCount < (1U << 31));
//ALOGV("inFrameCount:%d outFrameCount:%d"
// " phaseIncrement:%u phaseFraction:%u phaseWrapLimit:%u",
// inFrameCount, outFrameCount, phaseIncrement, phaseFraction, phaseWrapLimit);
// 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) {
//ALOGV("LOOP: inFrameCount:%d outputIndex:%d outFrameCount:%d"
// " phaseFraction:%u phaseWrapLimit:%u",
// inFrameCount, outputIndex, outFrameCount, phaseFraction, phaseWrapLimit);
// check inputIndex overflow
ALOG_ASSERT(inputIndex <= mBuffer.frameCount, "inputIndex%d > frameCount%d",
inputIndex, mBuffer.frameCount);
// Buffer is empty, fetch a new one if necessary (inFrameCount > 0).
// We may not fetch a new buffer if the existing data is sufficient.
while (mBuffer.frameCount == 0 && inFrameCount > 0) {
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer);
if (mBuffer.raw == NULL) {
goto resample_exit;
}
inFrameCount -= mBuffer.frameCount;
if (phaseFraction >= phaseWrapLimit) { // read in data
mInBuffer.template readAdvance<CHANNELS>(
impulse, c.mHalfNumCoefs,
reinterpret_cast<TI*>(mBuffer.raw), inputIndex);
inputIndex++;
phaseFraction -= phaseWrapLimit;
while (phaseFraction >= phaseWrapLimit) {
if (inputIndex >= mBuffer.frameCount) {
inputIndex = 0;
provider->releaseBuffer(&mBuffer);
break;
}
mInBuffer.template readAdvance<CHANNELS>(
impulse, c.mHalfNumCoefs,
reinterpret_cast<TI*>(mBuffer.raw), inputIndex);
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;
// 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.
//
//ALOGV("LOOP2: inFrameCount:%d outputIndex:%d outFrameCount:%d"
// " phaseFraction:%u phaseWrapLimit:%u",
// inFrameCount, outputIndex, outFrameCount, phaseFraction, phaseWrapLimit);
ALOG_ASSERT(phaseFraction < phaseWrapLimit);
fir<CHANNELS, LOCKED, STRIDE>(
&out[outputIndex],
phaseFraction, phaseWrapLimit,
coefShift, halfNumCoefs, coefs,
impulse, volumeSimd);
outputIndex += OUTPUT_CHANNELS;
phaseFraction += phaseIncrement;
while (phaseFraction >= phaseWrapLimit) {
if (inputIndex >= frameCount) {
goto done; // need a new buffer
}
mInBuffer.template readAdvance<CHANNELS>(impulse, halfNumCoefs, in, inputIndex);
inputIndex++;
phaseFraction -= phaseWrapLimit;
}
}
done:
// We arrive here when we're finished or when the input buffer runs out.
// Regardless we need to release the input buffer if we've acquired it.
if (inputIndex > 0) { // we've acquired a buffer (alternatively could check frameCount)
ALOG_ASSERT(inputIndex == frameCount, "inputIndex(%d) != frameCount(%d)",
inputIndex, frameCount); // must have been fully read.
inputIndex = 0;
provider->releaseBuffer(&mBuffer);
ALOG_ASSERT(mBuffer.frameCount == 0);
}
}
resample_exit:
// inputIndex must be zero in all three cases:
// (1) the buffer never was been acquired; (2) the buffer was
// released at "done:"; or (3) getNextBuffer() failed.
ALOG_ASSERT(inputIndex == 0, "Releasing: inputindex:%d frameCount:%d phaseFraction:%u",
inputIndex, mBuffer.frameCount, phaseFraction);
ALOG_ASSERT(mBuffer.frameCount == 0); // there must be no frames in the buffer
mInBuffer.setImpulse(impulse);
mPhaseFraction = phaseFraction;
return outputIndex / OUTPUT_CHANNELS;
}
/* 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