| /* |
| * Copyright (C) 2020 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 "SampleBuffer.h" |
| |
| // Resampler Includes |
| #include <resampler/MultiChannelResampler.h> |
| |
| #include "wav/WavStreamReader.h" |
| |
| using namespace RESAMPLER_OUTER_NAMESPACE::resampler; |
| |
| namespace iolib { |
| |
| void SampleBuffer::loadSampleData(parselib::WavStreamReader* reader) { |
| // Although we read this in, at this time we know a-priori that the data is mono |
| mAudioProperties.channelCount = reader->getNumChannels(); |
| mAudioProperties.sampleRate = reader->getSampleRate(); |
| |
| reader->positionToAudio(); |
| |
| mNumSamples = reader->getNumSampleFrames() * reader->getNumChannels(); |
| mSampleData = new float[mNumSamples]; |
| |
| reader->getDataFloat(mSampleData, reader->getNumSampleFrames()); |
| } |
| |
| void SampleBuffer::unloadSampleData() { |
| if (mSampleData != nullptr) { |
| delete[] mSampleData; |
| mSampleData = nullptr; |
| } |
| mNumSamples = 0; |
| } |
| |
| class ResampleBlock { |
| public: |
| int32_t mSampleRate; |
| float* mBuffer; |
| int32_t mNumFrames; |
| }; |
| |
| void resampleData(const ResampleBlock& input, ResampleBlock* output, int numChannels) { |
| // Calculate output buffer size |
| double temp = |
| ((double)input.mNumFrames * (double)output->mSampleRate) / (double)input.mSampleRate; |
| |
| // round up |
| int32_t numOutFrames = (int32_t)(temp + 0.5); |
| // We iterate thousands of times through the loop. Roundoff error could accumulate |
| // so add a few more frames for padding |
| numOutFrames += 8; |
| |
| MultiChannelResampler *resampler = MultiChannelResampler::make( |
| numChannels, // channel count |
| input.mSampleRate, // input sampleRate |
| output->mSampleRate, // output sampleRate |
| MultiChannelResampler::Quality::Medium); // conversion quality |
| |
| float *inputBuffer = input.mBuffer;; // multi-channel buffer to be consumed |
| float *outputBuffer = new float[numOutFrames]; // multi-channel buffer to be filled |
| output->mBuffer = outputBuffer; |
| |
| int numOutputFrames = 0; |
| int inputFramesLeft = input.mNumFrames; |
| while (inputFramesLeft > 0) { |
| if(resampler->isWriteNeeded()) { |
| resampler->writeNextFrame(inputBuffer); |
| inputBuffer += numChannels; |
| inputFramesLeft--; |
| } else { |
| resampler->readNextFrame(outputBuffer); |
| outputBuffer += numChannels; |
| numOutputFrames++; |
| } |
| } |
| output->mNumFrames = numOutputFrames; |
| |
| delete resampler; |
| } |
| |
| void SampleBuffer::resampleData(int sampleRate) { |
| if (mAudioProperties.sampleRate == sampleRate) { |
| // nothing to do |
| return; |
| } |
| |
| ResampleBlock inputBlock; |
| inputBlock.mBuffer = mSampleData; |
| inputBlock.mNumFrames = mNumSamples; |
| inputBlock.mSampleRate = mAudioProperties.sampleRate; |
| |
| ResampleBlock outputBlock; |
| outputBlock.mSampleRate = sampleRate; |
| iolib::resampleData(inputBlock, &outputBlock, mAudioProperties.channelCount); |
| |
| // delete previous samples |
| delete[] mSampleData; |
| |
| // install the resampled data |
| mSampleData = outputBlock.mBuffer; |
| mNumSamples = outputBlock.mNumFrames; |
| mAudioProperties.sampleRate = outputBlock.mSampleRate; |
| } |
| |
| } // namespace iolib |