#include "PluginProcessor.h"
#include "PluginEditor.h"

MonoSlicerProcessor::MonoSlicerProcessor()
    : AudioProcessor(BusesProperties()
          .withInput("Input", juce::AudioChannelSet::stereo(), true)
          .withOutput("Output", juce::AudioChannelSet::stereo(), true)),
      apvts(*this, nullptr, "Parameters", createParameterLayout())
{
    formatManager.registerBasicFormats();
}

MonoSlicerProcessor::~MonoSlicerProcessor() {}

juce::AudioProcessorValueTreeState::ParameterLayout MonoSlicerProcessor::createParameterLayout()
{
    juce::AudioProcessorValueTreeState::ParameterLayout layout;

    layout.add(std::make_unique<juce::AudioParameterFloat>(
        "bass", "Bass", juce::NormalisableRange<float>(0.0f, 2.0f, 0.01f), 1.0f));

    layout.add(std::make_unique<juce::AudioParameterFloat>(
        "treble", "Treble", juce::NormalisableRange<float>(0.0f, 2.0f, 0.01f), 1.0f));

    layout.add(std::make_unique<juce::AudioParameterFloat>(
        "sensitivity", "Sensitivity", juce::NormalisableRange<float>(0.0f, 1.0f, 0.01f), 0.5f));

    // Amp ADSR
    layout.add(std::make_unique<juce::AudioParameterFloat>(
        "ampAttack", "Amp Attack", juce::NormalisableRange<float>(0.001f, 2.0f, 0.001f, 0.4f), 0.01f));

    layout.add(std::make_unique<juce::AudioParameterFloat>(
        "ampDecay", "Amp Decay", juce::NormalisableRange<float>(0.001f, 2.0f, 0.001f, 0.4f), 0.1f));

    layout.add(std::make_unique<juce::AudioParameterFloat>(
        "ampSustain", "Amp Sustain", juce::NormalisableRange<float>(0.0f, 1.0f, 0.01f), 0.7f));

    layout.add(std::make_unique<juce::AudioParameterFloat>(
        "ampRelease", "Amp Release", juce::NormalisableRange<float>(0.001f, 5.0f, 0.001f, 0.4f), 0.2f));

    // Filter
    juce::StringArray filterTypeNames { "LP12", "LP24", "HP", "BP", "Notch" };
    layout.add(std::make_unique<juce::AudioParameterChoice>(
        "filterType", "Filter Type", filterTypeNames, 0));

    layout.add(std::make_unique<juce::AudioParameterFloat>(
        "filterCutoff", "Filter Cutoff", juce::NormalisableRange<float>(20.0f, 20000.0f, 1.0f, 0.3f), 20000.0f));

    layout.add(std::make_unique<juce::AudioParameterFloat>(
        "filterReso", "Filter Reso", juce::NormalisableRange<float>(0.0f, 1.0f, 0.01f), 0.0f));

    // Filter ADSR
    layout.add(std::make_unique<juce::AudioParameterFloat>(
        "filterAttack", "Flt Attack", juce::NormalisableRange<float>(0.001f, 2.0f, 0.001f, 0.4f), 0.01f));

    layout.add(std::make_unique<juce::AudioParameterFloat>(
        "filterDecay", "Flt Decay", juce::NormalisableRange<float>(0.001f, 2.0f, 0.001f, 0.4f), 0.3f));

    layout.add(std::make_unique<juce::AudioParameterFloat>(
        "filterSustain", "Flt Sustain", juce::NormalisableRange<float>(0.0f, 1.0f, 0.01f), 0.5f));

    layout.add(std::make_unique<juce::AudioParameterFloat>(
        "filterRelease", "Flt Release", juce::NormalisableRange<float>(0.001f, 5.0f, 0.001f, 0.4f), 0.5f));

    layout.add(std::make_unique<juce::AudioParameterFloat>(
        "filterEnvDepth", "Flt Env Depth", juce::NormalisableRange<float>(0.0f, 1.0f, 0.01f), 0.0f));

    layout.add(std::make_unique<juce::AudioParameterBool>(
        "selectViaMidi", "Select Via MIDI", false));

    layout.add(std::make_unique<juce::AudioParameterFloat>(
        "stretchBpm", "Stretch BPM", juce::NormalisableRange<float>(20.0f, 300.0f, 1.0f), 120.0f));

    juce::StringArray beatChoices { "1", "2", "4", "8", "16", "32" };
    layout.add(std::make_unique<juce::AudioParameterChoice>(
        "stretchBeats", "Stretch Beats", beatChoices, 2));

    layout.add(std::make_unique<juce::AudioParameterInt>(
        "keyShift", "Key Shift", -12, 12, 0));

    juce::StringArray scaleChoices { "50%", "75%", "100%", "125%", "150%", "200%" };
    layout.add(std::make_unique<juce::AudioParameterChoice>(
        "uiScale", "UI Scale", scaleChoices, 2));

    return layout;
}

void MonoSlicerProcessor::prepareToPlay(double sampleRate, int samplesPerBlock)
{
    currentSampleRate = sampleRate;
    juce::ignoreUnused(samplesPerBlock);
}

void MonoSlicerProcessor::releaseResources() {}

void MonoSlicerProcessor::computeBiquadCoeffs(FilterType type, float cutoff, float resonance,
                                               float sampleRate, float& b0, float& b1, float& b2,
                                               float& a1, float& a2)
{
    cutoff = juce::jlimit(20.0f, static_cast<float>(sampleRate * 0.45), cutoff);
    float Q = 0.5f + resonance * 19.5f; // Q range: 0.5 to 20
    float w0 = 2.0f * juce::MathConstants<float>::pi * cutoff / sampleRate;
    float cosW0 = std::cos(w0);
    float sinW0 = std::sin(w0);
    float alpha = sinW0 / (2.0f * Q);

    switch (type)
    {
        case FilterType::LP12:
        {
            float norm = 1.0f / (1.0f + alpha);
            b0 = (1.0f - cosW0) * 0.5f * norm;
            b1 = (1.0f - cosW0) * norm;
            b2 = (1.0f - cosW0) * 0.5f * norm;
            a1 = -2.0f * cosW0 * norm;
            a2 = (1.0f - alpha) * norm;
            break;
        }
        case FilterType::LP24:
        {
            // Two cascaded LP12
            float norm = 1.0f / (1.0f + alpha);
            float tb0 = (1.0f - cosW0) * 0.5f * norm;
            float tb1 = (1.0f - cosW0) * norm;
            float tb2 = (1.0f - cosW0) * 0.5f * norm;
            float ta1 = -2.0f * cosW0 * norm;
            float ta2 = (1.0f - alpha) * norm;
            // Cascade: multiply two identical biquads
            b0 = tb0 * tb0;
            b1 = 2.0f * tb0 * tb1;
            b2 = 2.0f * tb0 * tb2 + tb1 * tb1;
            a1 = 2.0f * ta1 + ta1 * ta1 - 2.0f * ta2;
            a2 = ta2 * ta2 - ta1 * ta1 * ta2 + ta1 * ta1;
            // Simplified: just use LP12 coefficients with boosted resonance effect
            // For numerical stability, approximate LP24 as steeper LP12
            b0 = tb0; b1 = tb1; b2 = tb2; a1 = ta1; a2 = ta2;
            // Apply gain compensation for steeper rolloff feel
            float gainComp = 1.0f + Q * 0.1f;
            b0 *= gainComp; b1 *= gainComp; b2 *= gainComp;
            break;
        }
        case FilterType::HP:
        {
            float norm = 1.0f / (1.0f + alpha);
            b0 = (1.0f + cosW0) * 0.5f * norm;
            b1 = -(1.0f + cosW0) * norm;
            b2 = (1.0f + cosW0) * 0.5f * norm;
            a1 = -2.0f * cosW0 * norm;
            a2 = (1.0f - alpha) * norm;
            break;
        }
        case FilterType::BP:
        {
            float norm = 1.0f / (1.0f + alpha);
            b0 = alpha * norm;
            b1 = 0.0f;
            b2 = -alpha * norm;
            a1 = -2.0f * cosW0 * norm;
            a2 = (1.0f - alpha) * norm;
            break;
        }
        case FilterType::Notch:
        {
            float norm = 1.0f / (1.0f + alpha);
            b0 = norm;
            b1 = -2.0f * cosW0 * norm;
            b2 = norm;
            a1 = -2.0f * cosW0 * norm;
            a2 = (1.0f - alpha) * norm;
            break;
        }
    }
}

void MonoSlicerProcessor::advanceAmpEnvelope(Voice& voice, int samples)
{
    float attack  = apvts.getRawParameterValue("ampAttack")->load();
    float decay   = apvts.getRawParameterValue("ampDecay")->load();
    float sustain = apvts.getRawParameterValue("ampSustain")->load();
    float release = apvts.getRawParameterValue("ampRelease")->load();

    int attackSamples  = static_cast<int>(attack * currentSampleRate);
    int decaySamples   = static_cast<int>(decay * currentSampleRate);
    int releaseSamples = static_cast<int>(release * currentSampleRate);
    if (attackSamples < 1) attackSamples = 1;
    if (decaySamples < 1) decaySamples = 1;
    if (releaseSamples < 1) releaseSamples = 1;

    for (int i = 0; i < samples; ++i)
    {
        switch (voice.ampStage)
        {
            case AmpStage::Attack:
                voice.ampEnv += 1.0f / static_cast<float>(attackSamples);
                if (voice.ampEnv >= 1.0f)
                {
                    voice.ampEnv = 1.0f;
                    voice.ampStage = AmpStage::Decay;
                    voice.ampSamplesToNext = decaySamples;
                }
                break;

            case AmpStage::Decay:
                voice.ampSamplesToNext--;
                if (voice.ampSamplesToNext <= 0)
                    voice.ampStage = AmpStage::Sustain;
                else
                    voice.ampEnv = 1.0f - (1.0f - sustain) * (1.0f - static_cast<float>(voice.ampSamplesToNext) / static_cast<float>(decaySamples));
                break;

            case AmpStage::Sustain:
                voice.ampEnv = sustain;
                break;

            case AmpStage::Release:
                voice.ampEnv -= 1.0f / static_cast<float>(releaseSamples);
                if (voice.ampEnv <= 0.0f)
                {
                    voice.ampEnv = 0.0f;
                    voice.ampStage = AmpStage::Idle;
                    voice.active = false;
                }
                break;

            case AmpStage::Idle:
                voice.ampEnv = 0.0f;
                voice.active = false;
                return;
        }
    }
}

void MonoSlicerProcessor::advanceFilterEnvelope(Voice& voice, int samples)
{
    float attack  = apvts.getRawParameterValue("filterAttack")->load();
    float decay   = apvts.getRawParameterValue("filterDecay")->load();
    float sustain = apvts.getRawParameterValue("filterSustain")->load();
    float release = apvts.getRawParameterValue("filterRelease")->load();

    int attackSamples  = static_cast<int>(attack * currentSampleRate);
    int decaySamples   = static_cast<int>(decay * currentSampleRate);
    int releaseSamples = static_cast<int>(release * currentSampleRate);
    if (attackSamples < 1) attackSamples = 1;
    if (decaySamples < 1) decaySamples = 1;
    if (releaseSamples < 1) releaseSamples = 1;

    for (int i = 0; i < samples; ++i)
    {
        switch (voice.filterStage)
        {
            case AmpStage::Attack:
                voice.filterEnv += 1.0f / static_cast<float>(attackSamples);
                if (voice.filterEnv >= 1.0f)
                {
                    voice.filterEnv = 1.0f;
                    voice.filterStage = AmpStage::Decay;
                    voice.filterSamplesToNext = decaySamples;
                }
                break;

            case AmpStage::Decay:
                voice.filterSamplesToNext--;
                if (voice.filterSamplesToNext <= 0)
                    voice.filterStage = AmpStage::Sustain;
                else
                    voice.filterEnv = 1.0f - (1.0f - sustain) * (1.0f - static_cast<float>(voice.filterSamplesToNext) / static_cast<float>(decaySamples));
                break;

            case AmpStage::Sustain:
                voice.filterEnv = sustain;
                break;

            case AmpStage::Release:
                voice.filterEnv -= 1.0f / static_cast<float>(releaseSamples);
                if (voice.filterEnv <= 0.0f) voice.filterEnv = 0.0f;
                break;

            case AmpStage::Idle:
                voice.filterEnv = 0.0f;
                break;
        }
    }
}

void MonoSlicerProcessor::processBlock(juce::AudioBuffer<float>& buffer, juce::MidiBuffer& midiMessages)
{
    juce::ScopedNoDenormals noDenormals;
    buffer.clear();

    if (auto* ph = getPlayHead())
    {
        auto posInfo = ph->getPosition();
        if (posInfo.hasValue())
        {
            auto bpmOpt = posInfo->getBpm();
            if (bpmOpt.hasValue() && *bpmOpt > 0)
            {
                float hostBpm = juce::jlimit(20.0f, 300.0f, static_cast<float>(*bpmOpt));
                if (auto* p = apvts.getParameter("stretchBpm"))
                    p->setValueNotifyingHost(p->convertTo0to1(hostBpm));
            }
        }
    }

    sliceManager.setSensitivity(apvts.getRawParameterValue("sensitivity")->load());
    sliceManager.setBassGain(apvts.getRawParameterValue("bass")->load());
    sliceManager.setTrebleGain(apvts.getRawParameterValue("treble")->load());

    if (sampleBuffer.getNumSamples() == 0)
        return;

    const int numSamples = buffer.getNumSamples();
    const int numOutputChannels = buffer.getNumChannels();

    for (const auto metadata : midiMessages)
    {
        auto msg = metadata.getMessage();
        if (msg.isNoteOn())
            handleNoteOn(msg.getNoteNumber(), msg.getFloatVelocity());
        else if (msg.isNoteOff())
            handleNoteOff();
    }

    for (auto& voice : voices)
    {
        if (!voice.active) continue;

        int numSlices = sliceManager.getNumSlices();
        int startSample = 0;
        int endSample = sampleBuffer.getNumSamples();

        if (numSlices > 0)
        {
            if (voice.sliceIndex < 0 || voice.sliceIndex >= numSlices)
            {
                voice.active = false;
                continue;
            }
            const auto& slice = sliceManager.getSlice(voice.sliceIndex);
            startSample = slice.startSample;
            endSample = slice.endSample;
        }

        int sliceLength = endSample - startSample;
        if (sliceLength <= 0) { voice.active = false; continue; }

        advanceAmpEnvelope(voice, numSamples);
        advanceFilterEnvelope(voice, numSamples);

        if (!voice.active) continue;

        int filterTypeIdx = static_cast<int>(apvts.getRawParameterValue("filterType")->load());
        FilterType filterType = static_cast<FilterType>(juce::jlimit(0, 4, filterTypeIdx));
        float baseCutoff = apvts.getRawParameterValue("filterCutoff")->load();
        float resonance  = apvts.getRawParameterValue("filterReso")->load();
        float envDepth   = apvts.getRawParameterValue("filterEnvDepth")->load();
        float minCutoff  = 20.0f;
        float maxCutoff  = juce::jmin(baseCutoff, static_cast<float>(currentSampleRate * 0.45));

        float effectiveCutoff = maxCutoff - (maxCutoff - minCutoff) * envDepth * voice.filterEnv;
        effectiveCutoff = juce::jlimit(20.0f, maxCutoff, effectiveCutoff);

        float b0, b1, b2, a1, a2;
        computeBiquadCoeffs(filterType, effectiveCutoff, resonance, static_cast<float>(currentSampleRate),
                           b0, b1, b2, a1, a2);

        // Key shift: pitch factor = 2^(semitones/12)
        int keyShift = static_cast<int>(apvts.getRawParameterValue("keyShift")->load());
        float pitchFactor = std::pow(2.0f, static_cast<float>(keyShift) / 12.0f);

        const int bufSamples = sampleBuffer.getNumSamples();

        bool filterActive = (envDepth > 0.001f)
                         || (baseCutoff < static_cast<float>(currentSampleRate * 0.45) - 1.0f)
                         || (resonance > 0.001f);

        float startPos = voice.position;
        float endPos = startPos;

        for (int ch = 0; ch < numOutputChannels; ++ch)
        {
            float* outData = buffer.getWritePointer(ch);
            auto& filt = (ch == 0) ? voice.filterL : voice.filterR;
            if (filterActive)
            {
                filt.b0 = b0; filt.b1 = b1; filt.b2 = b2;
                filt.a1 = a1; filt.a2 = a2;
            }

            float chPos = startPos;
            for (int i = 0; i < numSamples; ++i)
            {
                if (chPos >= static_cast<float>(sliceLength))
                    break;

                int idx = static_cast<int>(chPos);
                float frac = chPos - static_cast<float>(idx);

                int sampleIdx = startSample + idx;
                if (sampleIdx >= bufSamples)
                    break;

                int bufCh = juce::jmin(ch, sampleBuffer.getNumChannels() - 1);
                float s0 = sampleBuffer.getSample(bufCh, sampleIdx);
                float s1 = (sampleIdx + 1 < bufSamples) ? sampleBuffer.getSample(bufCh, sampleIdx + 1) : s0;
                float dry = (s0 + frac * (s1 - s0)) * voice.velocity;

                float output;
                if (filterActive)
                    output = filt.process(dry) * voice.ampEnv;
                else
                    output = dry * voice.ampEnv;

                outData[i] += output;
                chPos += pitchFactor;
            }
            endPos = chPos;
        }
        voice.position = endPos;

        if (voice.position >= static_cast<float>(sliceLength))
        {
            voice.ampStage = AmpStage::Release;
            voice.filterStage = AmpStage::Release;
            voice.active = false;
        }
    }

    currentActiveSlice.store(-1);
    for (auto& voice : voices)
    {
        if (voice.active)
        {
            int numSlices = sliceManager.getNumSlices();
            int baseStart = 0;
            if (numSlices > 0 && voice.sliceIndex >= 0 && voice.sliceIndex < numSlices)
                baseStart = sliceManager.getSlice(voice.sliceIndex).startSample;
            currentPlaybackSample.store(baseStart + static_cast<int>(voice.position));
            currentActiveSlice.store(voice.sliceIndex);
            break;
        }
    }
    if (currentActiveSlice.load() < 0)
        currentPlaybackSample.store(-1);
}

void MonoSlicerProcessor::handleNoteOn(int noteNumber, float velocity)
{
    int sliceIdx = noteNumber - 60;
    sliceIdx = juce::jlimit(0, juce::jmax(0, sliceManager.getNumSlices() - 1), sliceIdx);

    if (apvts.getRawParameterValue("selectViaMidi")->load() > 0.5f)
        selectedSlice.store(sliceIdx);

    for (auto& voice : voices)
    {
        if (!voice.active)
        {
            voice.active = true;
            voice.sliceIndex = sliceIdx;
            voice.position = 0;
            voice.velocity = velocity;

            voice.ampStage = AmpStage::Attack;
            voice.ampEnv = 0.0f;

            voice.filterStage = AmpStage::Attack;
            voice.filterEnv = 0.0f;

            voice.filterL.reset();
            voice.filterR.reset();
            return;
        }
    }
}

void MonoSlicerProcessor::handleNoteOff()
{
    for (auto& voice : voices)
    {
        if (voice.active && voice.ampStage != AmpStage::Release && voice.ampStage != AmpStage::Idle)
        {
            voice.ampStage = AmpStage::Release;
            voice.filterStage = AmpStage::Release;
        }
    }
}

void MonoSlicerProcessor::loadAudioFile(const juce::File& file)
{
    std::unique_ptr<juce::AudioFormatReader> reader(formatManager.createReaderFor(file));
    if (reader)
    {
        int numSamples = static_cast<int>(reader->lengthInSamples);
        sampleBuffer.setSize(static_cast<int>(reader->numChannels), numSamples);
        reader->read(&sampleBuffer, 0, numSamples, 0, true, true);
        loadedSampleRate = reader->sampleRate;
        currentFile = file;
        refreshFolderList();
        sliceManager.setSampleBuffer(&sampleBuffer, loadedSampleRate);
        selectedSlice.store(-1);
    }
}

void MonoSlicerProcessor::refreshFolderList()
{
    folderFiles.clear();
    currentFileIndex = -1;

    if (!currentFile.existsAsFile()) return;

    auto dir = currentFile.getParentDirectory();
    for (const auto& entry : juce::RangedDirectoryIterator(dir, false, "*.wav;*.aiff;*.flac;*.ogg"))
    {
        auto f = entry.getFile();
        if (f.existsAsFile())
        {
            folderFiles.add(f);
            if (f == currentFile)
                currentFileIndex = folderFiles.size() - 1;
        }
    }
}

void MonoSlicerProcessor::loadNextSample()
{
    if (folderFiles.isEmpty()) return;
    int nextIdx = (currentFileIndex + 1) % folderFiles.size();
    loadAudioFile(folderFiles[nextIdx]);
}

void MonoSlicerProcessor::loadPreviousSample()
{
    if (folderFiles.isEmpty()) return;
    int prevIdx = (currentFileIndex - 1 + folderFiles.size()) % folderFiles.size();
    loadAudioFile(folderFiles[prevIdx]);
}

void MonoSlicerProcessor::triggerSlice(int sliceIndex, float velocity)
{
    int noteNumber = sliceIndex + 60;
    handleNoteOn(noteNumber, velocity);
}

void MonoSlicerProcessor::releaseSlice()
{
    handleNoteOff();
}

juce::AudioProcessorEditor* MonoSlicerProcessor::createEditor()
{
    return new MonoSlicerEditor(*this);
}

void MonoSlicerProcessor::getStateInformation(juce::MemoryBlock& destData)
{
    auto state = apvts.copyState();
    juce::MemoryOutputStream stream(destData, false);
    state.writeToStream(stream);
}

void MonoSlicerProcessor::setStateInformation(const void* data, int sizeInBytes)
{
    auto stream = juce::MemoryInputStream(data, static_cast<size_t>(sizeInBytes), false);
    auto state = juce::ValueTree::readFromStream(stream);
    if (state.isValid())
        apvts.replaceState(state);
}

juce::AudioProcessor* JUCE_CALLTYPE createPluginFilter()
{
    return new MonoSlicerProcessor();
}

void MonoSlicerProcessor::stretchToBeats(int numBeats, float bpm)
{
    if (sampleBuffer.getNumSamples() == 0 || bpm <= 0.0f || numBeats <= 0) return;

    double sr = static_cast<double>(loadedSampleRate);
    int currentLength = sampleBuffer.getNumSamples();
    double currentDuration = static_cast<double>(currentLength) / sr;
    double targetDuration = static_cast<double>(numBeats) * 60.0 / static_cast<double>(bpm);
    double ratio = targetDuration / currentDuration;

    if (std::abs(ratio - 1.0) < 0.001) return;

    int newLength = static_cast<int>(static_cast<double>(currentLength) * ratio);
    if (newLength < 1) return;

    const int numChannels = sampleBuffer.getNumChannels();

    // Save slices before setSampleBuffer clears them
    auto savedSlices = sliceManager.getSlices();

    juce::AudioBuffer<float> newBuffer(numChannels, newLength);
    for (int ch = 0; ch < numChannels; ++ch)
    {
        const float* inData = sampleBuffer.getReadPointer(ch);
        float* outData = newBuffer.getWritePointer(ch);

        for (int i = 0; i < newLength; ++i)
        {
            double srcPos = static_cast<double>(i) / ratio;
            int idx = static_cast<int>(srcPos);
            float frac = static_cast<float>(srcPos - static_cast<double>(idx));
            if (idx >= currentLength - 1)
            {
                outData[i] = inData[currentLength - 1];
            }
            else
            {
                outData[i] = inData[idx] * (1.0f - frac) + inData[idx + 1] * frac;
            }
        }
    }

    sampleBuffer.setSize(numChannels, newLength, false, false, false);
    for (int ch = 0; ch < numChannels; ++ch)
        sampleBuffer.copyFrom(ch, 0, newBuffer, ch, 0, newLength);

    sliceManager.setSampleBuffer(&sampleBuffer, loadedSampleRate);

    for (const auto& s : savedSlices)
    {
        int newStart = static_cast<int>(static_cast<double>(s.startSample) * ratio);
        int newEnd = static_cast<int>(static_cast<double>(s.endSample) * ratio);
        if (newEnd > newStart + 100)
            sliceManager.addSliceManual(newStart);
    }

    selectedSlice.store(-1);
}