dspx 0.1.1-alpha.0

package/src/native/IDspStage.h

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+#pragma once
+#include <napi.h>
+
+namespace dsp
+{
+    // This abstract class is the key.
+    // Every filter you add will implement this.
+    class IDspStage
+    {
+    public:
+        virtual ~IDspStage() = default;
+
+        /**
+         * @brief Returns the type identifier of this stage.
+         *
+         * @return A string identifying the stage type (e.g., "movingAverage", "notchFilter").
+         */
+        virtual const char *getType() const = 0;
+
+        /**
+         * @brief Processes a chunk of audio data in-place.
+         *
+         * @param buffer The interleaved audio buffer.
+         * @param numSamples The total number of samples (e.g., 1024).
+         * @param numChannels The number of channels (e.g., 1, 2, 4).
+         * @param timestamps Optional array of timestamps (in milliseconds) for each sample.
+         *                   If nullptr, uses sample-based processing (legacy mode).
+         *                   If provided, must have length equal to numSamples.
+         */
+        virtual void process(float *buffer, size_t numSamples, int numChannels, const float *timestamps = nullptr) = 0;
+
+        /**
+         * @brief Serializes the stage's internal state to a Napi::Object.
+         *
+         * @param env The N-API environment for creating JavaScript objects.
+         * @return Napi::Object containing the serialized state.
+         */
+        virtual Napi::Object serializeState(Napi::Env env) const = 0;
+
+        /**
+         * @brief Deserializes and restores the stage's internal state.
+         *
+         * @param state The Napi::Object containing the serialized state.
+         */
+        virtual void deserializeState(const Napi::Object &state) = 0;
+
+        /**
+         * @brief Resets the stage's internal state to initial values.
+         */
+        virtual void reset() = 0;
+    };
+
+} // namespace dsp

package/src/native/adapters/InterpolatorStage.h

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+/**
+ * InterpolatorStage.h
+ *
+ * Polyphase FIR interpolator for efficient upsampling by integer factor L.
+ * Inserts (L-1) zero samples between each input sample and applies anti-imaging
+ * low-pass filter to smooth the result.
+ *
+ * Algorithm:
+ * 1. Insert L-1 zeros between input samples (zero-stuffing)
+ * 2. Design anti-imaging FIR filter with cutoff at π/L
+ * 3. Use polyphase decomposition for efficiency (avoids computing on zeros)
+ * 4. Maintain state across process() calls for streaming
+ *
+ * Efficiency: Polyphase structure avoids multiplying by zeros, reducing
+ * computation significantly.
+ */
+
+#pragma once
+
+#include "../IDspStage.h"
+#include "../core/FirFilter.h"
+#include <algorithm>
+#include <cmath>
+#include <stdexcept>
+#include <vector>
+
+namespace dsp
+{
+
+    /**
+     * Interpolator stage: Upsample signal by integer factor L
+     * Includes anti-imaging FIR low-pass filter
+     */
+    class InterpolatorStage : public IDspStage
+    {
+    public:
+        /**
+         * Construct interpolator
+         * @param factor Interpolation factor L (output rate = input rate * L)
+         * @param order FIR filter order (must be odd)
+         * @param sampleRate Input sample rate in Hz
+         */
+        InterpolatorStage(int factor, int order, double sampleRate)
+            : interpolationFactor_(factor), filterOrder_(order), sampleRate_(sampleRate), phaseIndex_(0)
+        {
+            if (factor < 2)
+            {
+                throw std::invalid_argument("Interpolation factor must be >= 2");
+            }
+            if (order < 3 || order % 2 == 0)
+            {
+                throw std::invalid_argument("Filter order must be odd and >= 3");
+            }
+
+            // Design anti-imaging filter: cutoff at Fs/(2*L) to prevent imaging
+            double outputSampleRate = sampleRate * factor;
+            double cutoffFreq = sampleRate / 2.0; // Nyquist of input rate
+
+            // Create FIR low-pass filter (operates at output rate)
+            filter_ = std::make_unique<FirFilter>(order, outputSampleRate);
+            designLowPassFilter(cutoffFreq, outputSampleRate);
+
+            // State buffer for polyphase filtering
+            stateBuffer_.resize(filterOrder_, 0.0f);
+            stateIndex_ = 0;
+        }
+
+        /**
+         * Process input samples and produce interpolated output
+         * Output size will be input size * L
+         */
+        void process(const float *input, size_t inputSize,
+                     float *output, size_t &outputSize) override
+        {
+            outputSize = 0;
+
+            if (inputSize == 0)
+            {
+                return;
+            }
+
+            // Process each input sample, producing L output samples
+            for (size_t i = 0; i < inputSize; ++i)
+            {
+                // Add input sample to state buffer
+                stateBuffer_[stateIndex_] = input[i];
+                stateIndex_ = (stateIndex_ + 1) % filterOrder_;
+
+                // Generate L output samples using polyphase filter
+                for (int phase = 0; phase < interpolationFactor_; ++phase)
+                {
+                    float sum = 0.0f;
+
+                    // Apply polyphase filter coefficients
+                    // Each phase uses every L-th coefficient
+                    for (int tap = 0; tap < filterOrder_; ++tap)
+                    {
+                        int bufferIdx = (stateIndex_ - 1 - tap + filterOrder_) % filterOrder_;
+                        int coeffIdx = phase + tap * interpolationFactor_;
+
+                        if (coeffIdx < filterOrder_ * interpolationFactor_)
+                        {
+                            sum += stateBuffer_[bufferIdx] * polyphaseCoeffs_[coeffIdx];
+                        }
+                    }
+
+                    output[outputSize++] = sum * interpolationFactor_; // Gain correction
+                }
+            }
+        }
+
+        void reset() override
+        {
+            std::fill(stateBuffer_.begin(), stateBuffer_.end(), 0.0f);
+            stateIndex_ = 0;
+            phaseIndex_ = 0;
+        }
+
+        std::string getName() const override
+        {
+            return "Interpolator(L=" + std::to_string(interpolationFactor_) + ")";
+        }
+
+        /**
+         * Get interpolation factor
+         */
+        int getFactor() const
+        {
+            return interpolationFactor_;
+        }
+
+        /**
+         * Get filter order
+         */
+        int getOrder() const
+        {
+            return filterOrder_;
+        }
+
+    private:
+        /**
+         * Design anti-imaging low-pass filter using windowed sinc method
+         * Creates polyphase decomposition of filter
+         */
+        void designLowPassFilter(double cutoffFreq, double outputSampleRate)
+        {
+            const int M = filterOrder_;
+            const int L = interpolationFactor_;
+            const int totalTaps = M * L;
+
+            polyphaseCoeffs_.resize(totalTaps);
+
+            const int center = totalTaps / 2;
+            const double fc = cutoffFreq / outputSampleRate;
+            const double omega_c = 2.0 * M_PI * fc;
+
+            // Design prototype filter at output rate
+            for (int n = 0; n < totalTaps; ++n)
+            {
+                // Sinc function
+                double t = n - center;
+                double sinc_val;
+                if (std::abs(t) < 1e-10)
+                {
+                    sinc_val = omega_c / M_PI;
+                }
+                else
+                {
+                    sinc_val = std::sin(omega_c * t) / (M_PI * t);
+                }
+
+                // Hamming window
+                double window = 0.54 - 0.46 * std::cos(2.0 * M_PI * n / (totalTaps - 1));
+
+                polyphaseCoeffs_[n] = static_cast<float>(sinc_val * window);
+            }
+
+            // Normalize coefficients
+            float sum = 0.0f;
+            for (float c : polyphaseCoeffs_)
+            {
+                sum += c;
+            }
+            for (float &c : polyphaseCoeffs_)
+            {
+                c /= sum;
+            }
+        }
+
+        int interpolationFactor_;
+        int filterOrder_;
+        double sampleRate_;
+
+        std::unique_ptr<FirFilter> filter_;
+        std::vector<float> stateBuffer_;
+        std::vector<float> polyphaseCoeffs_;
+        size_t stateIndex_;
+        int phaseIndex_;
+    };
+
+} // namespace dsp

package/src/native/adapters/MeanAbsoluteValueStage.h

@@ -0,0 +1,289 @@
+#pragma once
+
+#include "../IDspStage.h"
+#include "../core/MovingAbsoluteValueFilter.h" // Include the new core filter
+#include <vector>
+#include <stdexcept>
+#include <cmath>
+#include <string>
+#include <algorithm>
+#include <numeric> // For std::accumulate
+
+namespace dsp::adapters
+{
+    enum class MavMode
+    {
+        Batch,
+        Moving
+    };
+
+    class MeanAbsoluteValueStage : public IDspStage
+    {
+    public:
+        /**
+         * @brief Constructs a new Mean Absolute Value Stage.
+         * @param mode The MAV mode (Batch or Moving).
+         * @param window_size The window size in samples (0 if using duration-based).
+         * @param window_duration_ms The window duration in milliseconds (0 if using size-based).
+         */
+        explicit MeanAbsoluteValueStage(MavMode mode, size_t window_size = 0, double window_duration_ms = 0.0)
+            : m_mode(mode),
+              m_window_size(window_size),
+              m_window_duration_ms(window_duration_ms),
+              m_is_initialized(window_size > 0)
+        {
+            if (m_mode == MavMode::Moving && window_size == 0 && window_duration_ms == 0.0)
+            {
+                throw std::invalid_argument("MeanAbsoluteValue: either window size or window duration must be greater than 0 for 'moving' mode");
+            }
+        }
+
+        // Return the type identifier for this stage
+        const char *getType() const override
+        {
+            return "meanAbsoluteValue";
+        }
+
+        // Implementation of the interface method
+        void process(float *buffer, size_t numSamples, int numChannels, const float *timestamps = nullptr) override
+        {
+            if (m_mode == MavMode::Batch)
+            {
+                processBatch(buffer, numSamples, numChannels);
+            }
+            else // MavMode::Moving
+            {
+                processMoving(buffer, numSamples, numChannels, timestamps);
+            }
+        }
+
+        // Serialize the stage's state
+        Napi::Object serializeState(Napi::Env env) const override
+        {
+            Napi::Object state = Napi::Object::New(env);
+            std::string modeStr = (m_mode == MavMode::Moving) ? "moving" : "batch";
+            state.Set("mode", modeStr);
+
+            if (m_mode == MavMode::Moving)
+            {
+                state.Set("windowSize", static_cast<uint32_t>(m_window_size));
+                state.Set("numChannels", static_cast<uint32_t>(m_filters.size()));
+
+                // Serialize each channel's filter state
+                Napi::Array channelsArray = Napi::Array::New(env, m_filters.size());
+                for (size_t i = 0; i < m_filters.size(); ++i)
+                {
+                    Napi::Object channelState = Napi::Object::New(env);
+
+                    // Get the filter's internal state
+                    auto [bufferData, runningSumOfAbs] = m_filters[i].getState();
+
+                    // Convert buffer data to JavaScript array
+                    Napi::Array bufferArray = Napi::Array::New(env, bufferData.size());
+                    for (size_t j = 0; j < bufferData.size(); ++j)
+                    {
+                        bufferArray.Set(j, Napi::Number::New(env, bufferData[j]));
+                    }
+
+                    channelState.Set("buffer", bufferArray);
+                    channelState.Set("runningSum", Napi::Number::New(env, runningSumOfAbs));
+
+                    channelsArray.Set(static_cast<uint32_t>(i), channelState);
+                }
+                state.Set("channels", channelsArray);
+            }
+
+            return state;
+        }
+
+        // Deserialize and restore the stage's state
+        void deserializeState(const Napi::Object &state) override
+        {
+            std::string modeStr = state.Get("mode").As<Napi::String>().Utf8Value();
+            MavMode newMode = (modeStr == "moving") ? MavMode::Moving : MavMode::Batch;
+
+            if (newMode != m_mode)
+            {
+                throw std::runtime_error("MeanAbsoluteValue mode mismatch during deserialization");
+            }
+
+            if (m_mode == MavMode::Moving)
+            {
+                // Get window size and validate
+                size_t windowSize = state.Get("windowSize").As<Napi::Number>().Uint32Value();
+                if (windowSize != m_window_size)
+                {
+                    throw std::runtime_error("Window size mismatch during deserialization");
+                }
+
+                // Get number of channels
+                uint32_t numChannels = state.Get("channels").As<Napi::Array>().Length();
+
+                // Recreate filters
+                m_filters.clear();
+                for (uint32_t i = 0; i < numChannels; ++i)
+                {
+                    m_filters.emplace_back(m_window_size);
+                }
+
+                // Restore each channel's state
+                Napi::Array channelsArray = state.Get("channels").As<Napi::Array>();
+                for (uint32_t i = 0; i < numChannels; ++i)
+                {
+                    Napi::Object channelState = channelsArray.Get(i).As<Napi::Object>();
+
+                    // Get buffer data
+                    Napi::Array bufferArray = channelState.Get("buffer").As<Napi::Array>();
+                    std::vector<float> bufferData;
+                    bufferData.reserve(bufferArray.Length());
+                    for (uint32_t j = 0; j < bufferArray.Length(); ++j)
+                    {
+                        bufferData.push_back(bufferArray.Get(j).As<Napi::Number>().FloatValue());
+                    }
+
+                    // Get running sum (of absolute values)
+                    float runningSum = channelState.Get("runningSum").As<Napi::Number>().FloatValue();
+
+                    // --- Validation ---
+                    // We must re-calculate the sum of absolute values from the original buffer data
+                    float actualSumOfAbs = 0.0f;
+                    for (const auto &val : bufferData)
+                    {
+                        actualSumOfAbs += std::abs(val);
+                    }
+
+                    const float tolerance = 0.0001f * std::max(1.0f, std::abs(actualSumOfAbs));
+                    if (std::abs(runningSum - actualSumOfAbs) > tolerance)
+                    {
+                        throw std::runtime_error(
+                            "Running sum of absolute values validation failed: expected " +
+                            std::to_string(actualSumOfAbs) + " but got " +
+                            std::to_string(runningSum));
+                    }
+                    // --- End Validation ---
+
+                    // Restore the filter's state
+                    m_filters[i].setState(bufferData, runningSum);
+                }
+            }
+        }
+
+        // Reset all filters to initial state
+        void reset() override
+        {
+            for (auto &filter : m_filters)
+            {
+                filter.clear();
+            }
+        }
+
+    private:
+        /**
+         * @brief Statelessly calculates the MAV for each channel
+         * and overwrites all samples in that channel with the result.
+         */
+        void processBatch(float *buffer, size_t numSamples, int numChannels)
+        {
+            for (int c = 0; c < numChannels; ++c)
+            {
+                size_t numSamplesPerChannel = numSamples / numChannels;
+                if (numSamplesPerChannel == 0)
+                    continue;
+
+                double sum_abs = 0.0;
+
+                // First pass: Calculate sum of absolute values
+                for (size_t i = c; i < numSamples; i += numChannels)
+                {
+                    sum_abs += static_cast<double>(std::abs(buffer[i]));
+                }
+
+                // Calculate MAV
+                float mav = static_cast<float>(sum_abs / numSamplesPerChannel);
+
+                // Second pass: Fill this channel's buffer with the single MAV value
+                for (size_t i = c; i < numSamples; i += numChannels)
+                {
+                    buffer[i] = mav;
+                }
+            }
+        }
+
+        /**
+         * @brief Statefully processes samples using the moving MAV filters.
+         */
+        void processMoving(float *buffer, size_t numSamples, int numChannels, const float *timestamps)
+        {
+            // Determine if we're in time-aware mode
+            bool useTimeAware = (m_window_duration_ms > 0.0) && timestamps != nullptr;
+
+            // Lazy initialization: convert windowDuration to windowSize if needed
+            if (!m_is_initialized && m_window_duration_ms > 0.0)
+            {
+                if (timestamps != nullptr && numSamples > 1)
+                {
+                    // Estimate sample rate from timestamps
+                    size_t samples_to_check = std::min(numSamples, size_t(10));
+                    double total_time_ms = timestamps[samples_to_check - 1] - timestamps[0];
+                    double avg_sample_period_ms = total_time_ms / (samples_to_check - 1);
+                    double estimated_sample_rate = 1000.0 / avg_sample_period_ms;
+
+                    // Use 3x the estimated size for time-aware mode
+                    size_t estimated_size = static_cast<size_t>((m_window_duration_ms / 1000.0) * estimated_sample_rate);
+                    m_window_size = std::max(size_t(1), estimated_size * 3);
+
+                    m_is_initialized = true;
+                }
+                else
+                {
+                    throw std::runtime_error("MeanAbsoluteValue: windowDuration was set, but timestamps are not available to derive sample rate");
+                }
+            }
+
+            // Lazily initialize our filters, one for each channel
+            if (m_filters.size() != static_cast<size_t>(numChannels))
+            {
+                m_filters.clear();
+                for (int i = 0; i < numChannels; ++i)
+                {
+                    if (useTimeAware)
+                    {
+                        // Create time-aware filter
+                        m_filters.emplace_back(m_window_size, m_window_duration_ms);
+                    }
+                    else
+                    {
+                        // Create regular filter
+                        m_filters.emplace_back(m_window_size);
+                    }
+                }
+            }
+
+            // Process the buffer sample by sample, de-interleaving
+            for (size_t i = 0; i < numSamples; ++i)
+            {
+                int channel = i % numChannels;
+                size_t sample_index = i / numChannels;
+
+                if (useTimeAware)
+                {
+                    // Use time-aware processing
+                    buffer[i] = m_filters[channel].addSampleWithTimestamp(buffer[i], timestamps[sample_index]);
+                }
+                else
+                {
+                    // Use sample-count processing
+                    buffer[i] = m_filters[channel].addSample(buffer[i]);
+                }
+            }
+        }
+
+        MavMode m_mode;
+        size_t m_window_size;
+        double m_window_duration_ms;
+        bool m_is_initialized;
+        // We need a separate filter instance for each channel's state
+        std::vector<dsp::core::MovingAbsoluteValueFilter<float>> m_filters;
+    };
+
+} // namespace dsp::adapters