185 lines
5.7 KiB
C++
185 lines
5.7 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.
|
|
*/
|
|
|
|
#include <stdint.h>
|
|
#include <string.h>
|
|
#include <sys/types.h>
|
|
#include <cutils/log.h>
|
|
|
|
#include "AudioResampler.h"
|
|
#include "AudioResamplerCubic.h"
|
|
|
|
#define LOG_TAG "AudioSRC"
|
|
|
|
namespace android {
|
|
// ----------------------------------------------------------------------------
|
|
|
|
void AudioResamplerCubic::init() {
|
|
memset(&left, 0, sizeof(state));
|
|
memset(&right, 0, sizeof(state));
|
|
}
|
|
|
|
void AudioResamplerCubic::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 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 = (outFrameCount*mInSampleRate)/mSampleRate;
|
|
|
|
// fetch first buffer
|
|
if (mBuffer.frameCount == 0) {
|
|
mBuffer.frameCount = inFrameCount;
|
|
provider->getNextBuffer(&mBuffer);
|
|
if (mBuffer.raw == NULL)
|
|
return;
|
|
// LOGW("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);
|
|
if (mBuffer.raw == NULL)
|
|
goto save_state; // ugly, but efficient
|
|
in = mBuffer.i16;
|
|
// LOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
|
|
}
|
|
|
|
// advance sample state
|
|
advance(&left, in[inputIndex*2]);
|
|
advance(&right, in[inputIndex*2+1]);
|
|
}
|
|
}
|
|
|
|
save_state:
|
|
// LOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
|
|
mInputIndex = inputIndex;
|
|
mPhaseFraction = phaseFraction;
|
|
}
|
|
|
|
void 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 = (outFrameCount*mInSampleRate)/mSampleRate;
|
|
|
|
// fetch first buffer
|
|
if (mBuffer.frameCount == 0) {
|
|
mBuffer.frameCount = inFrameCount;
|
|
provider->getNextBuffer(&mBuffer);
|
|
if (mBuffer.raw == NULL)
|
|
return;
|
|
// LOGW("New buffer: offset=%p, frames=%d\n", 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);
|
|
if (mBuffer.raw == NULL)
|
|
goto save_state; // ugly, but efficient
|
|
// LOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
|
|
in = mBuffer.i16;
|
|
}
|
|
|
|
// advance sample state
|
|
advance(&left, in[inputIndex]);
|
|
}
|
|
}
|
|
|
|
save_state:
|
|
// LOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
|
|
mInputIndex = inputIndex;
|
|
mPhaseFraction = phaseFraction;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
}
|
|
; // namespace android
|
|
|