RedCore/cocos2d/cocos/audio/android/AudioDecoder.cpp

/****************************************************************************
Copyright (c) 2016-2017 Chukong Technologies Inc.

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#define LOG_TAG "AudioDecoder"

#include "audio/android/AudioDecoder.h"
#include "audio/android/AudioResampler.h"
#include "audio/android/PcmBufferProvider.h"
#include "audio/android/AudioResampler.h"

#include <thread>
#include <chrono>

namespace cocos2d { namespace experimental {

size_t AudioDecoder::fileRead(void* ptr, size_t size, size_t nmemb, void* datasource)
{
    AudioDecoder* thiz = (AudioDecoder*)datasource;
    ssize_t toReadBytes = std::min((ssize_t)(thiz->_fileData.getSize() - thiz->_fileCurrPos), (ssize_t)(nmemb * size));
    if (toReadBytes > 0)
    {
        memcpy(ptr, (unsigned char*) thiz->_fileData.getBytes() + thiz->_fileCurrPos, toReadBytes);
        thiz->_fileCurrPos += toReadBytes;
    }
    // ALOGD("File size: %d, After fileRead _fileCurrPos %d", (int)thiz->_fileData.getSize(), thiz->_fileCurrPos);
    return toReadBytes;
}

int AudioDecoder::fileSeek(void* datasource, int64_t offset, int whence)
{
    AudioDecoder* thiz = (AudioDecoder*)datasource;
    if (whence == SEEK_SET)
        thiz->_fileCurrPos = offset;
    else if (whence == SEEK_CUR)
        thiz->_fileCurrPos = thiz->_fileCurrPos + offset;
    else if (whence == SEEK_END)
        thiz->_fileCurrPos = thiz->_fileData.getSize();
    return 0;
}

int AudioDecoder::fileClose(void* datasource)
{
    return 0;
}

long AudioDecoder::fileTell(void* datasource)
{
    AudioDecoder* thiz = (AudioDecoder*)datasource;
    return (long) thiz->_fileCurrPos;
}

AudioDecoder::AudioDecoder()
        : _fileCurrPos(0), _sampleRate(-1)
{
    auto pcmBuffer = std::make_shared<std::vector<char>>();
    pcmBuffer->reserve(4096);
    _result.pcmBuffer = pcmBuffer;
}

AudioDecoder::~AudioDecoder()
{
    ALOGV("~AudioDecoder() %p", this);
}

bool AudioDecoder::init(const std::string &url, int sampleRate)
{
    _url = url;
    _sampleRate = sampleRate;
    return true;
}

bool AudioDecoder::start()
{
    auto oldTime = clockNow();
    auto nowTime = oldTime;
    bool ret;
    do
    {
        ret = decodeToPcm();
        if (!ret) 
        {
            ALOGE("decodeToPcm (%s) failed!", _url.c_str());
            break;
        }

        nowTime = clockNow();
        ALOGD("Decoding (%s) to pcm data wasted %fms", _url.c_str(), intervalInMS(oldTime, nowTime));
        oldTime = nowTime;

        ret = resample();
        if (!ret) 
        {
            ALOGE("resample (%s) failed!", _url.c_str());
            break;
        }

        nowTime = clockNow();
        ALOGD("Resampling (%s) wasted %fms", _url.c_str(), intervalInMS(oldTime, nowTime));
        oldTime = nowTime;

        ret = interleave();
        if (!ret) 
        {
            ALOGE("interleave (%s) failed!", _url.c_str());
            break;
        }
        
        nowTime = clockNow();
        ALOGD("Interleave (%s) wasted %fms", _url.c_str(), intervalInMS(oldTime, nowTime));

    } while(false);

    ALOGV_IF(!ret, "%s returns false, decode (%s)", __FUNCTION__, _url.c_str());
    return ret;
}

bool AudioDecoder::resample()
{
    if (_result.sampleRate == _sampleRate)
    {
        ALOGI("No need to resample since the sample rate (%d) of the decoded pcm data is the same as the device output sample rate",
              _sampleRate);
        return true;
    }

    ALOGV("Resample: %d --> %d", _result.sampleRate, _sampleRate);

    auto r = _result;
    PcmBufferProvider provider;
    provider.init(r.pcmBuffer->data(), r.numFrames, r.pcmBuffer->size() / r.numFrames);

    const int outFrameRate = _sampleRate;
    int outputChannels = 2;
    size_t outputFrameSize = outputChannels * sizeof(int32_t);
    size_t outputFrames = ((int64_t) r.numFrames * outFrameRate) / r.sampleRate;
    size_t outputSize = outputFrames * outputFrameSize;
    void *outputVAddr = malloc(outputSize);

    auto resampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT, r.numChannels, outFrameRate,
                                            AudioResampler::MED_QUALITY);
    resampler->setSampleRate(r.sampleRate);
    resampler->setVolume(AudioResampler::UNITY_GAIN_FLOAT, AudioResampler::UNITY_GAIN_FLOAT);

    memset(outputVAddr, 0, outputSize);

    ALOGV("resample() %zu output frames", outputFrames);

    std::vector<int> Ovalues;

    if (Ovalues.empty())
    {
        Ovalues.push_back(outputFrames);
    }
    for (size_t i = 0, j = 0; i < outputFrames;)
    {
        size_t thisFrames = Ovalues[j++];
        if (j >= Ovalues.size())
        {
            j = 0;
        }
        if (thisFrames == 0 || thisFrames > outputFrames - i)
        {
            thisFrames = outputFrames - i;
        }
        int outFrames = resampler->resample((int *) outputVAddr + outputChannels * i, thisFrames,
                                            &provider);
        ALOGV("outFrames: %d", outFrames);
        i += thisFrames;
    }

    ALOGV("resample() complete");

    resampler->reset();

    ALOGV("reset() complete");

    delete resampler;
    resampler = nullptr;

    // mono takes left channel only (out of stereo output pair)
    // stereo and multichannel preserve all channels.

    int channels = r.numChannels;
    int32_t *out = (int32_t *) outputVAddr;
    int16_t *convert = (int16_t *) malloc(outputFrames * channels * sizeof(int16_t));

    const int volumeShift = 12; // shift requirement for Q4.27 to Q.15
    // round to half towards zero and saturate at int16 (non-dithered)
    const int roundVal = (1 << (volumeShift - 1)) - 1; // volumePrecision > 0

    for (size_t i = 0; i < outputFrames; i++)
    {
        for (int j = 0; j < channels; j++)
        {
            int32_t s = out[i * outputChannels + j] + roundVal; // add offset here
            if (s < 0)
            {
                s = (s + 1) >> volumeShift; // round to 0
                if (s < -32768)
                {
                    s = -32768;
                }
            } else
            {
                s = s >> volumeShift;
                if (s > 32767)
                {
                    s = 32767;
                }
            }
            convert[i * channels + j] = int16_t(s);
        }
    }

    // Reset result
    _result.numFrames = outputFrames;
    _result.sampleRate = outFrameRate;

    auto buffer = std::make_shared<std::vector<char>>();
    buffer->reserve(_result.numFrames * _result.bitsPerSample / 8);
    buffer->insert(buffer->end(), (char *) convert,
                   (char *) convert + outputFrames * channels * sizeof(int16_t));
    _result.pcmBuffer = buffer;

    ALOGV("pcm buffer size: %d", (int)_result.pcmBuffer->size());

    free(convert);
    free(outputVAddr);
    return true;
}

//-----------------------------------------------------------------
bool AudioDecoder::interleave()
{
    if (_result.numChannels == 2)
    {
        ALOGI("Audio channel count is 2, no need to interleave");
        return true;
    }
    else if (_result.numChannels == 1)
    {
        // If it's a mono audio, try to compose a fake stereo buffer
        size_t newBufferSize = _result.pcmBuffer->size() * 2;
        auto newBuffer = std::make_shared<std::vector<char>>();
        newBuffer->reserve(newBufferSize);
        size_t totalFrameSizeInBytes = (size_t) (_result.numFrames * _result.bitsPerSample / 8);

        for (size_t i = 0; i < totalFrameSizeInBytes; i += 2)
        {
            // get one short value
            char byte1 = _result.pcmBuffer->at(i);
            char byte2 = _result.pcmBuffer->at(i + 1);

            // push two short value
            for (int j = 0; j < 2; ++j)
            {
                newBuffer->push_back(byte1);
                newBuffer->push_back(byte2);
            }
        }
        _result.numChannels = 2;
        _result.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
        _result.pcmBuffer = newBuffer;
        return true;
    }

    ALOGE("Audio channel count (%d) is wrong, interleave only supports converting mono to stereo!", _result.numChannels);
    return false;
}

}} // namespace cocos2d { namespace experimental {