/*
* 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"
#if 0
#include "AudioResamplerSinc.h"
#include "AudioResamplerCubic.h"
#endif
#ifdef __arm__
#include <machine/cpu-features.h>
#endif
namespace android {
#ifdef __ARM_HAVE_HALFWORD_MULTIPLY // optimized asm option
#define ASM_ARM_RESAMP1 // enable asm optimisation for ResamplerOrder1
#endif // __ARM_HAVE_HALFWORD_MULTIPLY
// ----------------------------------------------------------------------------
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);
ALOGD("forcing AudioResampler quality to %d", quality);
}
if (quality == DEFAULT)
quality = LOW_QUALITY;
switch (quality) {
default:
case LOW_QUALITY:
ALOGV("Create linear Resampler");
resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
break;
#if 0
case MED_QUALITY:
ALOGV("Create cubic Resampler");
resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
break;
case HIGH_QUALITY:
ALOGV("Create sinc Resampler");
resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
break;
#endif
}
// 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), mLocalTimeFreq(0),
mPTS(AudioBufferProvider::kInvalidPTS) {
// sanity check on format
if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) {
ALOGE("Unsupported sample format, %d bits, %d channels", bitDepth,
inChannelCount);
// ALOG_ASSERT(0);
}
// initialize common members
mVolume[0] = mVolume[1] = 0;
mBuffer.frameCount = 0;
}
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 AudioResampler::setLocalTimeFreq(uint64_t freq) {
mLocalTimeFreq = freq;
}
void AudioResampler::setPTS(int64_t pts) {
mPTS = pts;
}
int64_t AudioResampler::calculateOutputPTS(int outputFrameIndex) {
if (mPTS == AudioBufferProvider::kInvalidPTS) {
return AudioBufferProvider::kInvalidPTS;
} else {
return mPTS + ((outputFrameIndex * mLocalTimeFreq) / mSampleRate);
}
}
void AudioResampler::reset() {
mInputIndex = 0;
mPhaseFraction = 0;
mBuffer.frameCount = 0;
}
// ----------------------------------------------------------------------------
void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) {
// should never happen, but we overflow if it does
// ALOG_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;
// ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
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 resampleStereo16_exit;
}
// ALOGE("New buffer fetched: %d frames", 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) {
// ALOGE("boundary case");
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
// ALOGE("general case");
#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);
}
// ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
// if done with buffer, save samples
if (inputIndex >= mBuffer.frameCount) {
inputIndex -= mBuffer.frameCount;
// ALOGE("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
// ALOG_ASSERT(mBuffer.frameCount == 0);
}
}
// ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", 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;
// ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
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) {
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
goto resampleMono16_exit;
}
// ALOGE("New buffer fetched: %d frames", 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) {
// ALOGE("boundary case");
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
// ALOGE("general case");
#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);
}
// ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
// if done with buffer, save samples
if (inputIndex >= mBuffer.frameCount) {
inputIndex -= mBuffer.frameCount;
// ALOGE("buffer done, new input index %d", inputIndex);
mX0L = mBuffer.i16[mBuffer.frameCount-1];
provider->releaseBuffer(&mBuffer);
// verify that the releaseBuffer resets the buffer frameCount
// ALOG_ASSERT(mBuffer.frameCount == 0);
}
}
// ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", 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
*
*******************************************************************/
__attribute__((noinline))
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
"1:\n"
" cmp r8, r2\n" // curOut - maxCurOut
" bcs 2f\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 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
*
*******************************************************************/
__attribute__((noinline))
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