mirror of https://github.com/axmolengine/axmol.git
192 lines
6.1 KiB
C++
192 lines
6.1 KiB
C++
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/*
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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 "AudioResamplerCubic"
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#include <stdint.h>
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#include <string.h>
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#include <sys/types.h>
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#include "audio/android/cutils/log.h"
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#include "audio/android/AudioResampler.h"
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#include "audio/android/AudioResamplerCubic.h"
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namespace cocos2d {
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// ----------------------------------------------------------------------------
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void AudioResamplerCubic::init() {
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memset(&left, 0, sizeof(state));
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memset(&right, 0, sizeof(state));
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}
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size_t AudioResamplerCubic::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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// ALOG_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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return resampleMono16(out, outFrameCount, provider);
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case 2:
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return resampleStereo16(out, outFrameCount, provider);
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default:
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LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount);
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return 0;
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}
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}
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size_t AudioResamplerCubic::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 = getInFrameCountRequired(outFrameCount);
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// fetch first buffer
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if (mBuffer.frameCount == 0) {
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mBuffer.frameCount = inFrameCount;
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provider->getNextBuffer(&mBuffer, mPTS);
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if (mBuffer.raw == NULL) {
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return 0;
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}
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// ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
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}
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int16_t *in = mBuffer.i16;
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while (outputIndex < outputSampleCount) {
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int32_t sample;
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int32_t x;
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// calculate output sample
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x = phaseFraction >> kPreInterpShift;
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out[outputIndex++] += vl * interp(&left, x);
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out[outputIndex++] += vr * interp(&right, x);
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// out[outputIndex++] += vr * in[inputIndex*2];
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// increment phase
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phaseFraction += phaseIncrement;
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uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits);
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phaseFraction &= kPhaseMask;
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// time to fetch another sample
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while (indexIncrement--) {
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inputIndex++;
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if (inputIndex == mBuffer.frameCount) {
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inputIndex = 0;
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provider->releaseBuffer(&mBuffer);
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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 save_state; // ugly, but efficient
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}
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in = mBuffer.i16;
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// ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
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}
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// advance sample state
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advance(&left, in[inputIndex*2]);
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advance(&right, in[inputIndex*2+1]);
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}
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}
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save_state:
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// ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
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mInputIndex = inputIndex;
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mPhaseFraction = phaseFraction;
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return outputIndex / 2 /* channels for stereo */;
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}
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size_t AudioResamplerCubic::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 = getInFrameCountRequired(outFrameCount);
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// fetch first buffer
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if (mBuffer.frameCount == 0) {
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mBuffer.frameCount = inFrameCount;
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provider->getNextBuffer(&mBuffer, mPTS);
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if (mBuffer.raw == NULL) {
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return 0;
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}
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// ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
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}
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int16_t *in = mBuffer.i16;
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while (outputIndex < outputSampleCount) {
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int32_t sample;
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int32_t x;
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// calculate output sample
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x = phaseFraction >> kPreInterpShift;
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sample = interp(&left, x);
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out[outputIndex++] += vl * sample;
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out[outputIndex++] += vr * sample;
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// increment phase
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phaseFraction += phaseIncrement;
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uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits);
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phaseFraction &= kPhaseMask;
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// time to fetch another sample
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while (indexIncrement--) {
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inputIndex++;
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if (inputIndex == mBuffer.frameCount) {
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inputIndex = 0;
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provider->releaseBuffer(&mBuffer);
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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 save_state; // ugly, but efficient
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}
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// ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
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in = mBuffer.i16;
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}
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// advance sample state
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advance(&left, in[inputIndex]);
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}
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}
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save_state:
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// ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
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mInputIndex = inputIndex;
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mPhaseFraction = phaseFraction;
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return outputIndex;
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}
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// ----------------------------------------------------------------------------
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} // namespace cocos2d {
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