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/*
* 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 "AudioResamplerCubic"
#include <stdint.h>
#include <string.h>
#include <sys/types.h>
#include <cutils/log.h>
#include "AudioResampler.h"
#include "AudioResamplerCubic.h"
namespace android {
// ----------------------------------------------------------------------------
void AudioResamplerCubic::init() {
memset(&left, 0, sizeof(state));
memset(&right, 0, sizeof(state));
}
size_t AudioResamplerCubic::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:
return resampleMono16(out, outFrameCount, provider);
case 2:
return resampleStereo16(out, outFrameCount, provider);
default:
LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount);
return 0;
}
}
size_t AudioResamplerCubic::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 = getInFrameCountRequired(outFrameCount);
// fetch first buffer
if (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer, mPTS);
if (mBuffer.raw == NULL) {
return 0;
}
// ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
}
int16_t *in = mBuffer.i16;
while (outputIndex < outputSampleCount) {
int32_t sample;
int32_t x;
// calculate output sample
x = phaseFraction >> kPreInterpShift;
out[outputIndex++] += vl * interp(&left, x);
out[outputIndex++] += vr * interp(&right, x);
// out[outputIndex++] += vr * in[inputIndex*2];
// increment phase
phaseFraction += phaseIncrement;
uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits);
phaseFraction &= kPhaseMask;
// time to fetch another sample
while (indexIncrement--) {
inputIndex++;
if (inputIndex == mBuffer.frameCount) {
inputIndex = 0;
provider->releaseBuffer(&mBuffer);
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer,
calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL) {
goto save_state; // ugly, but efficient
}
in = mBuffer.i16;
// ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
}
// advance sample state
advance(&left, in[inputIndex*2]);
advance(&right, in[inputIndex*2+1]);
}
}
save_state:
// ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
return outputIndex / 2 /* channels for stereo */;
}
size_t AudioResamplerCubic::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 = getInFrameCountRequired(outFrameCount);
// fetch first buffer
if (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer, mPTS);
if (mBuffer.raw == NULL) {
return 0;
}
// ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
}
int16_t *in = mBuffer.i16;
while (outputIndex < outputSampleCount) {
int32_t sample;
int32_t x;
// calculate output sample
x = phaseFraction >> kPreInterpShift;
sample = interp(&left, x);
out[outputIndex++] += vl * sample;
out[outputIndex++] += vr * sample;
// increment phase
phaseFraction += phaseIncrement;
uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits);
phaseFraction &= kPhaseMask;
// time to fetch another sample
while (indexIncrement--) {
inputIndex++;
if (inputIndex == mBuffer.frameCount) {
inputIndex = 0;
provider->releaseBuffer(&mBuffer);
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer,
calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL) {
goto save_state; // ugly, but efficient
}
// ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
in = mBuffer.i16;
}
// advance sample state
advance(&left, in[inputIndex]);
}
}
save_state:
// ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
return outputIndex;
}
// ----------------------------------------------------------------------------
} // namespace android
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