e55b072c93
The assembly expects arguments to live at fixed offsets from the stack pointer which are invalid if the code is inlined. Change-Id: Ie93e93c5c69774079112345754fbc85896fc2f64
627 lines
23 KiB
C++
627 lines
23 KiB
C++
/*
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* Copyright (C) 2007 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#define LOG_TAG "AudioResampler"
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//#define LOG_NDEBUG 0
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#include <stdint.h>
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#include <stdlib.h>
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#include <sys/types.h>
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#include <cutils/log.h>
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#include <cutils/properties.h>
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#include "AudioResampler.h"
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#include "AudioResamplerSinc.h"
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#include "AudioResamplerCubic.h"
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#ifdef __arm__
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#include <machine/cpu-features.h>
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#endif
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namespace android {
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#ifdef __ARM_HAVE_HALFWORD_MULTIPLY // optimized asm option
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#define ASM_ARM_RESAMP1 // enable asm optimisation for ResamplerOrder1
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#endif // __ARM_HAVE_HALFWORD_MULTIPLY
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// ----------------------------------------------------------------------------
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class AudioResamplerOrder1 : public AudioResampler {
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public:
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AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) :
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AudioResampler(bitDepth, inChannelCount, sampleRate), mX0L(0), mX0R(0) {
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}
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virtual void resample(int32_t* out, size_t outFrameCount,
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AudioBufferProvider* provider);
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private:
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// number of bits used in interpolation multiply - 15 bits avoids overflow
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static const int kNumInterpBits = 15;
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// bits to shift the phase fraction down to avoid overflow
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static const int kPreInterpShift = kNumPhaseBits - kNumInterpBits;
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void init() {}
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void resampleMono16(int32_t* out, size_t outFrameCount,
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AudioBufferProvider* provider);
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void resampleStereo16(int32_t* out, size_t outFrameCount,
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AudioBufferProvider* provider);
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#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
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void AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
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size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
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uint32_t &phaseFraction, uint32_t phaseIncrement)
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__attribute__((noinline));
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void AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
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size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
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uint32_t &phaseFraction, uint32_t phaseIncrement)
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__attribute__((noinline));
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#endif // ASM_ARM_RESAMP1
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static inline int32_t Interp(int32_t x0, int32_t x1, uint32_t f) {
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return x0 + (((x1 - x0) * (int32_t)(f >> kPreInterpShift)) >> kNumInterpBits);
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}
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static inline void Advance(size_t* index, uint32_t* frac, uint32_t inc) {
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*frac += inc;
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*index += (size_t)(*frac >> kNumPhaseBits);
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*frac &= kPhaseMask;
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}
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int mX0L;
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int mX0R;
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};
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// ----------------------------------------------------------------------------
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AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
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int32_t sampleRate, int quality) {
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// can only create low quality resample now
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AudioResampler* resampler;
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char value[PROPERTY_VALUE_MAX];
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if (property_get("af.resampler.quality", value, 0)) {
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quality = atoi(value);
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LOGD("forcing AudioResampler quality to %d", quality);
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}
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if (quality == DEFAULT)
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quality = LOW_QUALITY;
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switch (quality) {
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default:
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case LOW_QUALITY:
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LOGV("Create linear Resampler");
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resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
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break;
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case MED_QUALITY:
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LOGV("Create cubic Resampler");
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resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
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break;
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case HIGH_QUALITY:
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LOGV("Create sinc Resampler");
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resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
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break;
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}
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// initialize resampler
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resampler->init();
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return resampler;
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}
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AudioResampler::AudioResampler(int bitDepth, int inChannelCount,
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int32_t sampleRate) :
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mBitDepth(bitDepth), mChannelCount(inChannelCount),
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mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
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mPhaseFraction(0), mLocalTimeFreq(0),
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mPTS(AudioBufferProvider::kInvalidPTS) {
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// sanity check on format
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if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) {
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LOGE("Unsupported sample format, %d bits, %d channels", bitDepth,
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inChannelCount);
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// LOG_ASSERT(0);
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}
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// initialize common members
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mVolume[0] = mVolume[1] = 0;
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mBuffer.frameCount = 0;
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// save format for quick lookup
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if (inChannelCount == 1) {
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mFormat = MONO_16_BIT;
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} else {
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mFormat = STEREO_16_BIT;
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}
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}
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AudioResampler::~AudioResampler() {
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}
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void AudioResampler::setSampleRate(int32_t inSampleRate) {
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mInSampleRate = inSampleRate;
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mPhaseIncrement = (uint32_t)((kPhaseMultiplier * inSampleRate) / mSampleRate);
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}
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void AudioResampler::setVolume(int16_t left, int16_t right) {
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// TODO: Implement anti-zipper filter
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mVolume[0] = left;
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mVolume[1] = right;
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}
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void AudioResampler::setLocalTimeFreq(uint64_t freq) {
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mLocalTimeFreq = freq;
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}
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void AudioResampler::setPTS(int64_t pts) {
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mPTS = pts;
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}
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int64_t AudioResampler::calculateOutputPTS(int outputFrameIndex) {
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if (mPTS == AudioBufferProvider::kInvalidPTS) {
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return AudioBufferProvider::kInvalidPTS;
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} else {
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return mPTS + ((outputFrameIndex * mLocalTimeFreq) / mSampleRate);
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}
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}
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void AudioResampler::reset() {
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mInputIndex = 0;
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mPhaseFraction = 0;
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mBuffer.frameCount = 0;
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}
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// ----------------------------------------------------------------------------
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void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
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AudioBufferProvider* provider) {
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// should never happen, but we overflow if it does
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// LOG_ASSERT(outFrameCount < 32767);
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// select the appropriate resampler
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switch (mChannelCount) {
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case 1:
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resampleMono16(out, outFrameCount, provider);
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break;
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case 2:
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resampleStereo16(out, outFrameCount, provider);
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break;
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}
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}
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void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
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AudioBufferProvider* provider) {
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int32_t vl = mVolume[0];
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int32_t vr = mVolume[1];
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size_t inputIndex = mInputIndex;
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uint32_t phaseFraction = mPhaseFraction;
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uint32_t phaseIncrement = mPhaseIncrement;
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size_t outputIndex = 0;
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size_t outputSampleCount = outFrameCount * 2;
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size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
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// LOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
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// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
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while (outputIndex < outputSampleCount) {
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// buffer is empty, fetch a new one
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while (mBuffer.frameCount == 0) {
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mBuffer.frameCount = inFrameCount;
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provider->getNextBuffer(&mBuffer,
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calculateOutputPTS(outputIndex / 2));
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if (mBuffer.raw == NULL) {
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goto resampleStereo16_exit;
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}
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// LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
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if (mBuffer.frameCount > inputIndex) break;
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inputIndex -= mBuffer.frameCount;
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mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
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mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
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provider->releaseBuffer(&mBuffer);
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// mBuffer.frameCount == 0 now so we reload a new buffer
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}
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int16_t *in = mBuffer.i16;
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// handle boundary case
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while (inputIndex == 0) {
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// LOGE("boundary case\n");
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out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
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out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
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Advance(&inputIndex, &phaseFraction, phaseIncrement);
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if (outputIndex == outputSampleCount)
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break;
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}
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// process input samples
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// LOGE("general case\n");
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#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
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if (inputIndex + 2 < mBuffer.frameCount) {
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int32_t* maxOutPt;
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int32_t maxInIdx;
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maxOutPt = out + (outputSampleCount - 2); // 2 because 2 frames per loop
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maxInIdx = mBuffer.frameCount - 2;
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AsmStereo16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
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phaseFraction, phaseIncrement);
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}
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#endif // ASM_ARM_RESAMP1
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while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
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out[outputIndex++] += vl * Interp(in[inputIndex*2-2],
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in[inputIndex*2], phaseFraction);
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out[outputIndex++] += vr * Interp(in[inputIndex*2-1],
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in[inputIndex*2+1], phaseFraction);
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Advance(&inputIndex, &phaseFraction, phaseIncrement);
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}
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// LOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
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// if done with buffer, save samples
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if (inputIndex >= mBuffer.frameCount) {
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inputIndex -= mBuffer.frameCount;
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// LOGE("buffer done, new input index %d", inputIndex);
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mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
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mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
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provider->releaseBuffer(&mBuffer);
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// verify that the releaseBuffer resets the buffer frameCount
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// LOG_ASSERT(mBuffer.frameCount == 0);
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}
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}
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// LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
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resampleStereo16_exit:
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// save state
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mInputIndex = inputIndex;
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mPhaseFraction = phaseFraction;
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}
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void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
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AudioBufferProvider* provider) {
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int32_t vl = mVolume[0];
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int32_t vr = mVolume[1];
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size_t inputIndex = mInputIndex;
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uint32_t phaseFraction = mPhaseFraction;
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uint32_t phaseIncrement = mPhaseIncrement;
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size_t outputIndex = 0;
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size_t outputSampleCount = outFrameCount * 2;
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size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
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// LOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
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// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
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while (outputIndex < outputSampleCount) {
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// buffer is empty, fetch a new one
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while (mBuffer.frameCount == 0) {
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mBuffer.frameCount = inFrameCount;
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provider->getNextBuffer(&mBuffer,
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calculateOutputPTS(outputIndex / 2));
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if (mBuffer.raw == NULL) {
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mInputIndex = inputIndex;
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mPhaseFraction = phaseFraction;
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goto resampleMono16_exit;
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}
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// LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
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if (mBuffer.frameCount > inputIndex) break;
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inputIndex -= mBuffer.frameCount;
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mX0L = mBuffer.i16[mBuffer.frameCount-1];
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provider->releaseBuffer(&mBuffer);
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// mBuffer.frameCount == 0 now so we reload a new buffer
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}
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int16_t *in = mBuffer.i16;
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// handle boundary case
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while (inputIndex == 0) {
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// LOGE("boundary case\n");
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int32_t sample = Interp(mX0L, in[0], phaseFraction);
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out[outputIndex++] += vl * sample;
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out[outputIndex++] += vr * sample;
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Advance(&inputIndex, &phaseFraction, phaseIncrement);
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if (outputIndex == outputSampleCount)
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break;
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}
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// process input samples
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// LOGE("general case\n");
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#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
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if (inputIndex + 2 < mBuffer.frameCount) {
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int32_t* maxOutPt;
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int32_t maxInIdx;
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maxOutPt = out + (outputSampleCount - 2);
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maxInIdx = (int32_t)mBuffer.frameCount - 2;
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AsmMono16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
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phaseFraction, phaseIncrement);
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}
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#endif // ASM_ARM_RESAMP1
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while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
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int32_t sample = Interp(in[inputIndex-1], in[inputIndex],
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phaseFraction);
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out[outputIndex++] += vl * sample;
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out[outputIndex++] += vr * sample;
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Advance(&inputIndex, &phaseFraction, phaseIncrement);
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}
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// LOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
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// if done with buffer, save samples
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if (inputIndex >= mBuffer.frameCount) {
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inputIndex -= mBuffer.frameCount;
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// LOGE("buffer done, new input index %d", inputIndex);
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mX0L = mBuffer.i16[mBuffer.frameCount-1];
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provider->releaseBuffer(&mBuffer);
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// verify that the releaseBuffer resets the buffer frameCount
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// LOG_ASSERT(mBuffer.frameCount == 0);
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}
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}
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// LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
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resampleMono16_exit:
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// save state
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mInputIndex = inputIndex;
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mPhaseFraction = phaseFraction;
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}
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#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
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/*******************************************************************
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*
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* AsmMono16Loop
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* asm optimized monotonic loop version; one loop is 2 frames
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* Input:
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* in : pointer on input samples
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* maxOutPt : pointer on first not filled
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* maxInIdx : index on first not used
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* outputIndex : pointer on current output index
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* out : pointer on output buffer
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* inputIndex : pointer on current input index
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* vl, vr : left and right gain
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* phaseFraction : pointer on current phase fraction
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* phaseIncrement
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* Ouput:
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* outputIndex :
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* out : updated buffer
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* inputIndex : index of next to use
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* phaseFraction : phase fraction for next interpolation
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*
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*******************************************************************/
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void AudioResamplerOrder1::AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
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size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
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uint32_t &phaseFraction, uint32_t phaseIncrement)
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{
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#define MO_PARAM5 "36" // offset of parameter 5 (outputIndex)
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asm(
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"stmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, lr}\n"
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// get parameters
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" ldr r6, [sp, #" MO_PARAM5 " + 20]\n" // &phaseFraction
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" ldr r6, [r6]\n" // phaseFraction
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" ldr r7, [sp, #" MO_PARAM5 " + 8]\n" // &inputIndex
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" ldr r7, [r7]\n" // inputIndex
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" ldr r8, [sp, #" MO_PARAM5 " + 4]\n" // out
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" ldr r0, [sp, #" MO_PARAM5 " + 0]\n" // &outputIndex
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" ldr r0, [r0]\n" // outputIndex
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" add r8, r0, asl #2\n" // curOut
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" ldr r9, [sp, #" MO_PARAM5 " + 24]\n" // phaseIncrement
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" ldr r10, [sp, #" MO_PARAM5 " + 12]\n" // vl
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" ldr r11, [sp, #" MO_PARAM5 " + 16]\n" // vr
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// r0 pin, x0, Samp
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// r1 in
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// r2 maxOutPt
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// r3 maxInIdx
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// r4 x1, i1, i3, Out1
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// r5 out0
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// r6 frac
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// r7 inputIndex
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// r8 curOut
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// r9 inc
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// r10 vl
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// r11 vr
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// r12
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// r13 sp
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// r14
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// the following loop works on 2 frames
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"1:\n"
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" cmp r8, r2\n" // curOut - maxCurOut
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" bcs 2f\n"
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#define MO_ONE_FRAME \
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" add r0, r1, r7, asl #1\n" /* in + inputIndex */\
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" ldrsh r4, [r0]\n" /* in[inputIndex] */\
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" ldr r5, [r8]\n" /* out[outputIndex] */\
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" ldrsh r0, [r0, #-2]\n" /* in[inputIndex-1] */\
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" bic r6, r6, #0xC0000000\n" /* phaseFraction & ... */\
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" sub r4, r4, r0\n" /* in[inputIndex] - in[inputIndex-1] */\
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" mov r4, r4, lsl #2\n" /* <<2 */\
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" smulwt r4, r4, r6\n" /* (x1-x0)*.. */\
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" add r6, r6, r9\n" /* phaseFraction + phaseIncrement */\
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" add r0, r0, r4\n" /* x0 - (..) */\
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" mla r5, r0, r10, r5\n" /* vl*interp + out[] */\
|
|
" ldr r4, [r8, #4]\n" /* out[outputIndex+1] */\
|
|
" str r5, [r8], #4\n" /* out[outputIndex++] = ... */\
|
|
" mla r4, r0, r11, r4\n" /* vr*interp + out[] */\
|
|
" add r7, r7, r6, lsr #30\n" /* inputIndex + phaseFraction>>30 */\
|
|
" str r4, [r8], #4\n" /* out[outputIndex++] = ... */
|
|
|
|
MO_ONE_FRAME // frame 1
|
|
MO_ONE_FRAME // frame 2
|
|
|
|
" cmp r7, r3\n" // inputIndex - maxInIdx
|
|
" bcc 1b\n"
|
|
"2:\n"
|
|
|
|
" bic r6, r6, #0xC0000000\n" // phaseFraction & ...
|
|
// save modified values
|
|
" ldr r0, [sp, #" MO_PARAM5 " + 20]\n" // &phaseFraction
|
|
" str r6, [r0]\n" // phaseFraction
|
|
" ldr r0, [sp, #" MO_PARAM5 " + 8]\n" // &inputIndex
|
|
" str r7, [r0]\n" // inputIndex
|
|
" ldr r0, [sp, #" MO_PARAM5 " + 4]\n" // out
|
|
" sub r8, r0\n" // curOut - out
|
|
" asr r8, #2\n" // new outputIndex
|
|
" ldr r0, [sp, #" MO_PARAM5 " + 0]\n" // &outputIndex
|
|
" str r8, [r0]\n" // save outputIndex
|
|
|
|
" ldmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, pc}\n"
|
|
);
|
|
}
|
|
|
|
/*******************************************************************
|
|
*
|
|
* AsmStereo16Loop
|
|
* asm optimized stereo loop version; one loop is 2 frames
|
|
* Input:
|
|
* in : pointer on input samples
|
|
* maxOutPt : pointer on first not filled
|
|
* maxInIdx : index on first not used
|
|
* outputIndex : pointer on current output index
|
|
* out : pointer on output buffer
|
|
* inputIndex : pointer on current input index
|
|
* vl, vr : left and right gain
|
|
* phaseFraction : pointer on current phase fraction
|
|
* phaseIncrement
|
|
* Ouput:
|
|
* outputIndex :
|
|
* out : updated buffer
|
|
* inputIndex : index of next to use
|
|
* phaseFraction : phase fraction for next interpolation
|
|
*
|
|
*******************************************************************/
|
|
void AudioResamplerOrder1::AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
|
|
size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
|
|
uint32_t &phaseFraction, uint32_t phaseIncrement)
|
|
{
|
|
#define ST_PARAM5 "40" // offset of parameter 5 (outputIndex)
|
|
asm(
|
|
"stmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, lr}\n"
|
|
// get parameters
|
|
" ldr r6, [sp, #" ST_PARAM5 " + 20]\n" // &phaseFraction
|
|
" ldr r6, [r6]\n" // phaseFraction
|
|
" ldr r7, [sp, #" ST_PARAM5 " + 8]\n" // &inputIndex
|
|
" ldr r7, [r7]\n" // inputIndex
|
|
" ldr r8, [sp, #" ST_PARAM5 " + 4]\n" // out
|
|
" ldr r0, [sp, #" ST_PARAM5 " + 0]\n" // &outputIndex
|
|
" ldr r0, [r0]\n" // outputIndex
|
|
" add r8, r0, asl #2\n" // curOut
|
|
" ldr r9, [sp, #" ST_PARAM5 " + 24]\n" // phaseIncrement
|
|
" ldr r10, [sp, #" ST_PARAM5 " + 12]\n" // vl
|
|
" ldr r11, [sp, #" ST_PARAM5 " + 16]\n" // vr
|
|
|
|
// r0 pin, x0, Samp
|
|
|
|
// r1 in
|
|
// r2 maxOutPt
|
|
// r3 maxInIdx
|
|
|
|
// r4 x1, i1, i3, out1
|
|
// r5 out0
|
|
|
|
// r6 frac
|
|
// r7 inputIndex
|
|
// r8 curOut
|
|
|
|
// r9 inc
|
|
// r10 vl
|
|
// r11 vr
|
|
|
|
// r12 temporary
|
|
// r13 sp
|
|
// r14
|
|
|
|
"3:\n"
|
|
" cmp r8, r2\n" // curOut - maxCurOut
|
|
" bcs 4f\n"
|
|
|
|
#define ST_ONE_FRAME \
|
|
" bic r6, r6, #0xC0000000\n" /* phaseFraction & ... */\
|
|
\
|
|
" add r0, r1, r7, asl #2\n" /* in + 2*inputIndex */\
|
|
\
|
|
" ldrsh r4, [r0]\n" /* in[2*inputIndex] */\
|
|
" ldr r5, [r8]\n" /* out[outputIndex] */\
|
|
" ldrsh r12, [r0, #-4]\n" /* in[2*inputIndex-2] */\
|
|
" sub r4, r4, r12\n" /* in[2*InputIndex] - in[2*InputIndex-2] */\
|
|
" mov r4, r4, lsl #2\n" /* <<2 */\
|
|
" smulwt r4, r4, r6\n" /* (x1-x0)*.. */\
|
|
" add r12, r12, r4\n" /* x0 - (..) */\
|
|
" mla r5, r12, r10, r5\n" /* vl*interp + out[] */\
|
|
" ldr r4, [r8, #4]\n" /* out[outputIndex+1] */\
|
|
" str r5, [r8], #4\n" /* out[outputIndex++] = ... */\
|
|
\
|
|
" ldrsh r12, [r0, #+2]\n" /* in[2*inputIndex+1] */\
|
|
" ldrsh r0, [r0, #-2]\n" /* in[2*inputIndex-1] */\
|
|
" sub r12, r12, r0\n" /* in[2*InputIndex] - in[2*InputIndex-2] */\
|
|
" mov r12, r12, lsl #2\n" /* <<2 */\
|
|
" smulwt r12, r12, r6\n" /* (x1-x0)*.. */\
|
|
" add r12, r0, r12\n" /* x0 - (..) */\
|
|
" mla r4, r12, r11, r4\n" /* vr*interp + out[] */\
|
|
" str r4, [r8], #4\n" /* out[outputIndex++] = ... */\
|
|
\
|
|
" add r6, r6, r9\n" /* phaseFraction + phaseIncrement */\
|
|
" add r7, r7, r6, lsr #30\n" /* inputIndex + phaseFraction>>30 */
|
|
|
|
ST_ONE_FRAME // frame 1
|
|
ST_ONE_FRAME // frame 1
|
|
|
|
" cmp r7, r3\n" // inputIndex - maxInIdx
|
|
" bcc 3b\n"
|
|
"4:\n"
|
|
|
|
" bic r6, r6, #0xC0000000\n" // phaseFraction & ...
|
|
// save modified values
|
|
" ldr r0, [sp, #" ST_PARAM5 " + 20]\n" // &phaseFraction
|
|
" str r6, [r0]\n" // phaseFraction
|
|
" ldr r0, [sp, #" ST_PARAM5 " + 8]\n" // &inputIndex
|
|
" str r7, [r0]\n" // inputIndex
|
|
" ldr r0, [sp, #" ST_PARAM5 " + 4]\n" // out
|
|
" sub r8, r0\n" // curOut - out
|
|
" asr r8, #2\n" // new outputIndex
|
|
" ldr r0, [sp, #" ST_PARAM5 " + 0]\n" // &outputIndex
|
|
" str r8, [r0]\n" // save outputIndex
|
|
|
|
" ldmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, pc}\n"
|
|
);
|
|
}
|
|
|
|
#endif // ASM_ARM_RESAMP1
|
|
|
|
|
|
// ----------------------------------------------------------------------------
|
|
}
|
|
; // namespace android
|