#include <cstring>
#include <cmath>
#include <VulcanoLE/Audio/AudioGrabber.h>
#include <VUtils/Logging.h>


AudioGrabber::AudioGrabber() = default;
AudioGrabber::~AudioGrabber() = default;

bool AudioGrabber::read(stereoSample *buffer, uint32_t bufferSize) {
    auto buffer_size_bytes = static_cast<size_t>(sizeof(stereoSample) * bufferSize);
    if (m_pulseaudioSimple == nullptr) {
        openPulseaudioSource(static_cast<uint32_t>(buffer_size_bytes));
    }

    if (m_pulseaudioSimple != nullptr) {
        memset(buffer, 0, buffer_size_bytes);
        int32_t error_code;
        auto return_code = pa_simple_read(m_pulseaudioSimple, buffer,
                                          buffer_size_bytes, &error_code);
        if (return_code < 0) {
            WARN("Could not finish reading pulse Audio stream buffer\n bytes read: %d buffer\n size: %d", return_code,
                 buffer_size_bytes)
            // zero out buffer
            memset(buffer, 0, buffer_size_bytes);
            pa_simple_free(m_pulseaudioSimple);
            m_pulseaudioSimple = nullptr;

            return false;
        }

        // Success fully read entire buffer
        return true;
    }
    return false;
}


void AudioGrabber::populateDefaultSourceName() {
    pa_mainloop_api *mainloop_api;
    pa_context *pulseaudio_context;
    m_pulseaudioMainloop = pa_mainloop_new();
    mainloop_api = pa_mainloop_get_api(m_pulseaudioMainloop);
    pulseaudio_context = pa_context_new(mainloop_api, "VulcanoLE device list");
    pa_context_connect(pulseaudio_context, nullptr, PA_CONTEXT_NOFLAGS,
                       nullptr);
    pa_context_set_state_callback(pulseaudio_context,
                                  pulseaudioContextStateCallback,
                                  reinterpret_cast<void *>(this));

    int ret;
    if (pa_mainloop_run(m_pulseaudioMainloop, &ret) < 0) {
        ERR("Could not open pulseaudio mainloop to find default device name: %d", ret)
    }
}

bool AudioGrabber::openPulseaudioSource(uint32_t maxBufferSize) {
    int32_t error_code = 0;
    static const pa_sample_spec sample_spec = { PA_SAMPLE_FLOAT32NE, sampleRate,
                                                channels };
    static const pa_buffer_attr buffer_attr = { maxBufferSize, 0, 0, 0,
                                                (maxBufferSize / 2) };
    populateDefaultSourceName();
    if (!m_PulseaudioDefaultSourceName.empty()) {
        m_pulseaudioSimple =
                pa_simple_new(nullptr, recStreamName, PA_STREAM_RECORD,
                              m_PulseaudioDefaultSourceName.c_str(),
                              recStreamDescription, &sample_spec,
                              nullptr, &buffer_attr, &error_code);
    }
    if (m_pulseaudioSimple == nullptr) {
        m_pulseaudioSimple =
                pa_simple_new(nullptr, recStreamName, PA_STREAM_RECORD,
                              nullptr, recStreamDescription,
                              &sample_spec, nullptr, &buffer_attr, &error_code);
    }
    if (m_pulseaudioSimple == nullptr) {
        m_pulseaudioSimple =
                pa_simple_new(nullptr, recStreamName, PA_STREAM_RECORD,
                              "0", recStreamDescription, &sample_spec,
                              nullptr, &buffer_attr, &error_code);
    }
    if (m_pulseaudioSimple != nullptr) {
        return true;
    }

    ERR("Could not open pulseaudio source: %s", pa_strerror(error_code))
    return false;
}

void AudioGrabber::pulseaudioContextStateCallback(pa_context *c, void *userdata) {
    switch (pa_context_get_state(c)) {
        case PA_CONTEXT_UNCONNECTED:
        case PA_CONTEXT_CONNECTING:
        case PA_CONTEXT_AUTHORIZING:
        case PA_CONTEXT_SETTING_NAME:
            break;

        case PA_CONTEXT_READY: {
            pa_operation_unref(pa_context_get_server_info(
                    c, pulseaudioServerInfoCallback, userdata));
            break;
        }

        case PA_CONTEXT_FAILED:
        case PA_CONTEXT_TERMINATED:
            auto *src =
                    reinterpret_cast<AudioGrabber *>(userdata);
            pa_mainloop_quit(src->m_pulseaudioMainloop, 0);
            break;
    }
}

void AudioGrabber::pulseaudioServerInfoCallback(pa_context *context, const pa_server_info *i, void *userdata) {
    if (i != nullptr) {
        auto *src = reinterpret_cast<AudioGrabber *>(userdata);
        std::string name = i->default_sink_name;
        name.append(defaultMonitorPostfix);

        src->m_PulseaudioDefaultSourceName = name;

        // stop mainloop after finding default name
        pa_mainloop_quit(src->m_pulseaudioMainloop, 0);
    }
}

AudioGrabber *AudioGrabber::createAudioGrabber() {
    auto *grabber = new AudioGrabber();
    return grabber;
}

void AudioGrabber::init() {
    m_buffer = static_cast<stereoSample *>(calloc(BUFFER_SIZE, sizeof(stereoSample)));
    if (env != nullptr)
        m_scale = env->getAsDouble("audio_scale", 1.0);
    DBG("SET Audio Scale: %.3f", m_scale)
    loudness = { 0.0, 0.0 };
}

void AudioGrabber::calculateRMS(stereoSample *pFrame) {
    float squareL = 0, meanL;
    float squareR = 0, meanR;
    for (int i = 0; i < BUFFER_SIZE; i++) {
        squareL += std::pow(pFrame[0].l, 2);
        squareR += std::pow(pFrame[0].r, 2);
    }
    meanL = (squareL / (float) (BUFFER_SIZE));
    meanR = (squareR / (float) (BUFFER_SIZE));
    loudness = { std::sqrt(meanL), std::sqrt(meanR) };
}

void AudioGrabber::calculatePEAK(stereoSample *pFrame) {
    stereoSampleFrame max = { 0, 0 };
    for (int i = 0; i < BUFFER_SIZE; i++) {
        float left = std::abs(pFrame[0].l);
        float right = std::abs(pFrame[0].r);
        if (left > max.l)
            max.l = left;
        if (right > max.r)
            max.r = right;
    }
    loudness = max;
}

stereoSampleFrame AudioGrabber::getLoudness() {
    std::unique_lock<std::mutex> lck(m_mtx);
    return loudness;
}

bool AudioGrabber::work() {
    std::unique_lock<std::mutex> lck(m_mtx);
    if (this->read(m_buffer, BUFFER_SIZE)) {
        switch (requestMode) {
            case ReqMode::FFT:
                // FFT get's better results with maybe a bit scaling.. for RMS and PEAK this gets "worse"
                for (int i = 0; i < BUFFER_SIZE; ++i) {
                    m_buffer[i].l *= m_scale;
                    m_buffer[i].r *= m_scale;
                }
                fft.process(m_buffer);
                break;
            case ReqMode::RMS:
                calculateRMS(m_buffer);
                break;
            case ReqMode::PEAK:
                calculatePEAK(m_buffer);
                break;
            default:
                fft.process(m_buffer);
                calculateRMS(m_buffer);
                calculatePEAK(m_buffer);
        }
        return true;
    } else {
        DBG("Wait for Data")
        return false;
    }
}