596 lines
22 KiB
C++
596 lines
22 KiB
C++
/*
|
|
* Copyright (C) 2007 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 "AudioResampler"
|
|
//#define LOG_NDEBUG 0
|
|
|
|
#include <stdint.h>
|
|
#include <stdlib.h>
|
|
#include <sys/types.h>
|
|
#include <cutils/log.h>
|
|
#include <cutils/properties.h>
|
|
#include "AudioResampler.h"
|
|
#include "AudioResamplerSinc.h"
|
|
#include "AudioResamplerCubic.h"
|
|
|
|
namespace android {
|
|
|
|
#ifdef __ARM_ARCH_5E__ // optimized asm option
|
|
#define ASM_ARM_RESAMP1 // enable asm optimisation for ResamplerOrder1
|
|
#endif // __ARM_ARCH_5E__
|
|
// ----------------------------------------------------------------------------
|
|
|
|
class AudioResamplerOrder1 : public AudioResampler {
|
|
public:
|
|
AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) :
|
|
AudioResampler(bitDepth, inChannelCount, sampleRate), mX0L(0), mX0R(0) {
|
|
}
|
|
virtual void resample(int32_t* out, size_t outFrameCount,
|
|
AudioBufferProvider* provider);
|
|
private:
|
|
// number of bits used in interpolation multiply - 15 bits avoids overflow
|
|
static const int kNumInterpBits = 15;
|
|
|
|
// bits to shift the phase fraction down to avoid overflow
|
|
static const int kPreInterpShift = kNumPhaseBits - kNumInterpBits;
|
|
|
|
void init() {}
|
|
void resampleMono16(int32_t* out, size_t outFrameCount,
|
|
AudioBufferProvider* provider);
|
|
void resampleStereo16(int32_t* out, size_t outFrameCount,
|
|
AudioBufferProvider* provider);
|
|
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
|
|
void AsmMono16Loop(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);
|
|
void 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);
|
|
#endif // ASM_ARM_RESAMP1
|
|
|
|
static inline int32_t Interp(int32_t x0, int32_t x1, uint32_t f) {
|
|
return x0 + (((x1 - x0) * (int32_t)(f >> kPreInterpShift)) >> kNumInterpBits);
|
|
}
|
|
static inline void Advance(size_t* index, uint32_t* frac, uint32_t inc) {
|
|
*frac += inc;
|
|
*index += (size_t)(*frac >> kNumPhaseBits);
|
|
*frac &= kPhaseMask;
|
|
}
|
|
int mX0L;
|
|
int mX0R;
|
|
};
|
|
|
|
// ----------------------------------------------------------------------------
|
|
AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
|
|
int32_t sampleRate, int quality) {
|
|
|
|
// can only create low quality resample now
|
|
AudioResampler* resampler;
|
|
|
|
char value[PROPERTY_VALUE_MAX];
|
|
if (property_get("af.resampler.quality", value, 0)) {
|
|
quality = atoi(value);
|
|
LOGD("forcing AudioResampler quality to %d", quality);
|
|
}
|
|
|
|
if (quality == DEFAULT)
|
|
quality = LOW_QUALITY;
|
|
|
|
switch (quality) {
|
|
default:
|
|
case LOW_QUALITY:
|
|
LOGV("Create linear Resampler");
|
|
resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
|
|
break;
|
|
case MED_QUALITY:
|
|
LOGV("Create cubic Resampler");
|
|
resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
|
|
break;
|
|
case HIGH_QUALITY:
|
|
LOGV("Create sinc Resampler");
|
|
resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
|
|
break;
|
|
}
|
|
|
|
// initialize resampler
|
|
resampler->init();
|
|
return resampler;
|
|
}
|
|
|
|
AudioResampler::AudioResampler(int bitDepth, int inChannelCount,
|
|
int32_t sampleRate) :
|
|
mBitDepth(bitDepth), mChannelCount(inChannelCount),
|
|
mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
|
|
mPhaseFraction(0) {
|
|
// sanity check on format
|
|
if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) {
|
|
LOGE("Unsupported sample format, %d bits, %d channels", bitDepth,
|
|
inChannelCount);
|
|
// LOG_ASSERT(0);
|
|
}
|
|
|
|
// initialize common members
|
|
mVolume[0] = mVolume[1] = 0;
|
|
mBuffer.frameCount = 0;
|
|
|
|
// save format for quick lookup
|
|
if (inChannelCount == 1) {
|
|
mFormat = MONO_16_BIT;
|
|
} else {
|
|
mFormat = STEREO_16_BIT;
|
|
}
|
|
}
|
|
|
|
AudioResampler::~AudioResampler() {
|
|
}
|
|
|
|
void AudioResampler::setSampleRate(int32_t inSampleRate) {
|
|
mInSampleRate = inSampleRate;
|
|
mPhaseIncrement = (uint32_t)((kPhaseMultiplier * inSampleRate) / mSampleRate);
|
|
}
|
|
|
|
void AudioResampler::setVolume(int16_t left, int16_t right) {
|
|
// TODO: Implement anti-zipper filter
|
|
mVolume[0] = left;
|
|
mVolume[1] = right;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
|
|
AudioBufferProvider* provider) {
|
|
|
|
// should never happen, but we overflow if it does
|
|
// LOG_ASSERT(outFrameCount < 32767);
|
|
|
|
// select the appropriate resampler
|
|
switch (mChannelCount) {
|
|
case 1:
|
|
resampleMono16(out, outFrameCount, provider);
|
|
break;
|
|
case 2:
|
|
resampleStereo16(out, outFrameCount, provider);
|
|
break;
|
|
}
|
|
}
|
|
|
|
void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
|
|
AudioBufferProvider* provider) {
|
|
|
|
int32_t vl = mVolume[0];
|
|
int32_t vr = mVolume[1];
|
|
|
|
size_t inputIndex = mInputIndex;
|
|
uint32_t phaseFraction = mPhaseFraction;
|
|
uint32_t phaseIncrement = mPhaseIncrement;
|
|
size_t outputIndex = 0;
|
|
size_t outputSampleCount = outFrameCount * 2;
|
|
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
|
|
|
|
// LOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
|
|
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
|
|
|
|
while (outputIndex < outputSampleCount) {
|
|
|
|
// buffer is empty, fetch a new one
|
|
while (mBuffer.frameCount == 0) {
|
|
mBuffer.frameCount = inFrameCount;
|
|
provider->getNextBuffer(&mBuffer);
|
|
if (mBuffer.raw == NULL) {
|
|
goto resampleStereo16_exit;
|
|
}
|
|
|
|
// LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
|
|
if (mBuffer.frameCount > inputIndex) break;
|
|
|
|
inputIndex -= mBuffer.frameCount;
|
|
mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
|
|
mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
|
|
provider->releaseBuffer(&mBuffer);
|
|
// mBuffer.frameCount == 0 now so we reload a new buffer
|
|
}
|
|
|
|
int16_t *in = mBuffer.i16;
|
|
|
|
// handle boundary case
|
|
while (inputIndex == 0) {
|
|
// LOGE("boundary case\n");
|
|
out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
|
|
out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
|
|
Advance(&inputIndex, &phaseFraction, phaseIncrement);
|
|
if (outputIndex == outputSampleCount)
|
|
break;
|
|
}
|
|
|
|
// process input samples
|
|
// LOGE("general case\n");
|
|
|
|
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
|
|
if (inputIndex + 2 < mBuffer.frameCount) {
|
|
int32_t* maxOutPt;
|
|
int32_t maxInIdx;
|
|
|
|
maxOutPt = out + (outputSampleCount - 2); // 2 because 2 frames per loop
|
|
maxInIdx = mBuffer.frameCount - 2;
|
|
AsmStereo16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
|
|
phaseFraction, phaseIncrement);
|
|
}
|
|
#endif // ASM_ARM_RESAMP1
|
|
|
|
while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
|
|
out[outputIndex++] += vl * Interp(in[inputIndex*2-2],
|
|
in[inputIndex*2], phaseFraction);
|
|
out[outputIndex++] += vr * Interp(in[inputIndex*2-1],
|
|
in[inputIndex*2+1], phaseFraction);
|
|
Advance(&inputIndex, &phaseFraction, phaseIncrement);
|
|
}
|
|
|
|
// LOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
|
|
|
|
// if done with buffer, save samples
|
|
if (inputIndex >= mBuffer.frameCount) {
|
|
inputIndex -= mBuffer.frameCount;
|
|
|
|
// LOGE("buffer done, new input index %d", inputIndex);
|
|
|
|
mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
|
|
mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
|
|
provider->releaseBuffer(&mBuffer);
|
|
|
|
// verify that the releaseBuffer resets the buffer frameCount
|
|
// LOG_ASSERT(mBuffer.frameCount == 0);
|
|
}
|
|
}
|
|
|
|
// LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
|
|
|
|
resampleStereo16_exit:
|
|
// save state
|
|
mInputIndex = inputIndex;
|
|
mPhaseFraction = phaseFraction;
|
|
}
|
|
|
|
void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
|
|
AudioBufferProvider* provider) {
|
|
|
|
int32_t vl = mVolume[0];
|
|
int32_t vr = mVolume[1];
|
|
|
|
size_t inputIndex = mInputIndex;
|
|
uint32_t phaseFraction = mPhaseFraction;
|
|
uint32_t phaseIncrement = mPhaseIncrement;
|
|
size_t outputIndex = 0;
|
|
size_t outputSampleCount = outFrameCount * 2;
|
|
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
|
|
|
|
// LOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
|
|
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
|
|
while (outputIndex < outputSampleCount) {
|
|
// buffer is empty, fetch a new one
|
|
while (mBuffer.frameCount == 0) {
|
|
mBuffer.frameCount = inFrameCount;
|
|
provider->getNextBuffer(&mBuffer);
|
|
if (mBuffer.raw == NULL) {
|
|
mInputIndex = inputIndex;
|
|
mPhaseFraction = phaseFraction;
|
|
goto resampleMono16_exit;
|
|
}
|
|
// LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
|
|
if (mBuffer.frameCount > inputIndex) break;
|
|
|
|
inputIndex -= mBuffer.frameCount;
|
|
mX0L = mBuffer.i16[mBuffer.frameCount-1];
|
|
provider->releaseBuffer(&mBuffer);
|
|
// mBuffer.frameCount == 0 now so we reload a new buffer
|
|
}
|
|
int16_t *in = mBuffer.i16;
|
|
|
|
// handle boundary case
|
|
while (inputIndex == 0) {
|
|
// LOGE("boundary case\n");
|
|
int32_t sample = Interp(mX0L, in[0], phaseFraction);
|
|
out[outputIndex++] += vl * sample;
|
|
out[outputIndex++] += vr * sample;
|
|
Advance(&inputIndex, &phaseFraction, phaseIncrement);
|
|
if (outputIndex == outputSampleCount)
|
|
break;
|
|
}
|
|
|
|
// process input samples
|
|
// LOGE("general case\n");
|
|
|
|
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
|
|
if (inputIndex + 2 < mBuffer.frameCount) {
|
|
int32_t* maxOutPt;
|
|
int32_t maxInIdx;
|
|
|
|
maxOutPt = out + (outputSampleCount - 2);
|
|
maxInIdx = (int32_t)mBuffer.frameCount - 2;
|
|
AsmMono16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
|
|
phaseFraction, phaseIncrement);
|
|
}
|
|
#endif // ASM_ARM_RESAMP1
|
|
|
|
while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
|
|
int32_t sample = Interp(in[inputIndex-1], in[inputIndex],
|
|
phaseFraction);
|
|
out[outputIndex++] += vl * sample;
|
|
out[outputIndex++] += vr * sample;
|
|
Advance(&inputIndex, &phaseFraction, phaseIncrement);
|
|
}
|
|
|
|
|
|
// LOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
|
|
|
|
// if done with buffer, save samples
|
|
if (inputIndex >= mBuffer.frameCount) {
|
|
inputIndex -= mBuffer.frameCount;
|
|
|
|
// LOGE("buffer done, new input index %d", inputIndex);
|
|
|
|
mX0L = mBuffer.i16[mBuffer.frameCount-1];
|
|
provider->releaseBuffer(&mBuffer);
|
|
|
|
// verify that the releaseBuffer resets the buffer frameCount
|
|
// LOG_ASSERT(mBuffer.frameCount == 0);
|
|
}
|
|
}
|
|
|
|
// LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
|
|
|
|
resampleMono16_exit:
|
|
// save state
|
|
mInputIndex = inputIndex;
|
|
mPhaseFraction = phaseFraction;
|
|
}
|
|
|
|
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
|
|
|
|
/*******************************************************************
|
|
*
|
|
* AsmMono16Loop
|
|
* asm optimized monotonic 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::AsmMono16Loop(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 MO_PARAM5 "36" // offset of parameter 5 (outputIndex)
|
|
|
|
asm(
|
|
"stmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, lr}\n"
|
|
// get parameters
|
|
" ldr r6, [sp, #" MO_PARAM5 " + 20]\n" // &phaseFraction
|
|
" ldr r6, [r6]\n" // phaseFraction
|
|
" ldr r7, [sp, #" MO_PARAM5 " + 8]\n" // &inputIndex
|
|
" ldr r7, [r7]\n" // inputIndex
|
|
" ldr r8, [sp, #" MO_PARAM5 " + 4]\n" // out
|
|
" ldr r0, [sp, #" MO_PARAM5 " + 0]\n" // &outputIndex
|
|
" ldr r0, [r0]\n" // outputIndex
|
|
" add r8, r0, asl #2\n" // curOut
|
|
" ldr r9, [sp, #" MO_PARAM5 " + 24]\n" // phaseIncrement
|
|
" ldr r10, [sp, #" MO_PARAM5 " + 12]\n" // vl
|
|
" ldr r11, [sp, #" MO_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
|
|
// r13 sp
|
|
// r14
|
|
|
|
// the following loop works on 2 frames
|
|
|
|
".Y4L01:\n"
|
|
" cmp r8, r2\n" // curOut - maxCurOut
|
|
" bcs .Y4L02\n"
|
|
|
|
#define MO_ONE_FRAME \
|
|
" add r0, r1, r7, asl #1\n" /* in + inputIndex */\
|
|
" ldrsh r4, [r0]\n" /* in[inputIndex] */\
|
|
" ldr r5, [r8]\n" /* out[outputIndex] */\
|
|
" ldrsh r0, [r0, #-2]\n" /* in[inputIndex-1] */\
|
|
" bic r6, r6, #0xC0000000\n" /* phaseFraction & ... */\
|
|
" sub r4, r4, r0\n" /* in[inputIndex] - in[inputIndex-1] */\
|
|
" mov r4, r4, lsl #2\n" /* <<2 */\
|
|
" smulwt r4, r4, r6\n" /* (x1-x0)*.. */\
|
|
" add r6, r6, r9\n" /* phaseFraction + phaseIncrement */\
|
|
" add r0, r0, r4\n" /* x0 - (..) */\
|
|
" 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 .Y4L01\n"
|
|
".Y4L02:\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
|
|
|
|
".Y5L01:\n"
|
|
" cmp r8, r2\n" // curOut - maxCurOut
|
|
" bcs .Y5L02\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 .Y5L01\n"
|
|
".Y5L02:\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
|
|
|