react-native-audio-api 0.4.10 → 0.4.12-beta.1

package/common/cpp/libs/signalsmith-stretch.h

@@ -0,0 +1,637 @@
+#ifndef SIGNALSMITH_STRETCH_H
+#define SIGNALSMITH_STRETCH_H
+
+#include "dsp/spectral.h"
+#include "dsp/delay.h"
+#include "dsp/perf.h"
+SIGNALSMITH_DSP_VERSION_CHECK(1, 6, 0); // Check version is compatible
+#include <vector>
+#include <algorithm>
+#include <functional>
+#include <random>
+
+namespace signalsmith { namespace stretch {
+
+template<typename Sample=float, class RandomEngine=std::default_random_engine>
+struct SignalsmithStretch {
+	static constexpr size_t version[3] = {1, 1, 1};
+
+	SignalsmithStretch() : randomEngine(std::random_device{}()) {}
+	SignalsmithStretch(long seed) : randomEngine(seed) {}
+
+	int blockSamples() const {
+		return stft.windowSize();
+	}
+	int intervalSamples() const {
+		return stft.interval();
+	}
+	int inputLatency() const {
+		return stft.windowSize()/2;
+	}
+	int outputLatency() const {
+		return stft.windowSize() - inputLatency();
+	}
+	
+	void reset() {
+		stft.reset();
+		inputBuffer.reset();
+		prevInputOffset = -1;
+		channelBands.assign(channelBands.size(), Band());
+		silenceCounter = 0;
+		didSeek = false;
+		flushed = true;
+	}
+
+	// Configures using a default preset
+	void presetDefault(int nChannels, Sample sampleRate) {
+		configure(nChannels, sampleRate*0.12, sampleRate*0.03);
+	}
+	void presetCheaper(int nChannels, Sample sampleRate) {
+		configure(nChannels, sampleRate*0.1, sampleRate*0.04);
+	}
+
+	// Manual setup
+	void configure(int nChannels, int blockSamples, int intervalSamples) {
+		channels = nChannels;
+		stft.setWindow(stft.kaiser, true);
+		stft.resize(channels, blockSamples, intervalSamples);
+		bands = stft.bands();
+		inputBuffer.resize(channels, blockSamples + intervalSamples + 1);
+		timeBuffer.assign(stft.fftSize(), 0);
+		channelBands.assign(bands*channels, Band());
+		
+		peaks.reserve(bands/2);
+		energy.resize(bands);
+		smoothedEnergy.resize(bands);
+		outputMap.resize(bands);
+		channelPredictions.resize(channels*bands);
+	}
+
+	/// Frequency multiplier, and optional tonality limit (as multiple of sample-rate)
+	void setTransposeFactor(Sample multiplier, Sample tonalityLimit=0) {
+		freqMultiplier = multiplier;
+		if (tonalityLimit > 0) {
+			freqTonalityLimit = tonalityLimit/std::sqrt(multiplier); // compromise between input and output limits
+		} else {
+			freqTonalityLimit = 1;
+		}
+		customFreqMap = nullptr;
+	}
+	void setTransposeSemitones(Sample semitones, Sample tonalityLimit=0) {
+		setTransposeFactor(std::pow(2, semitones/12), tonalityLimit);
+		customFreqMap = nullptr;
+	}
+	// Sets a custom frequency map - should be monotonically increasing
+	void setFreqMap(std::function<Sample(Sample)> inputToOutput) {
+		customFreqMap = inputToOutput;
+	}
+
+	// Provide previous input ("pre-roll"), without affecting the speed calculation.  You should ideally feed it one block-length + one interval
+	template<class Inputs>
+	void seek(Inputs &&inputs, int inputSamples, double playbackRate) {
+		inputBuffer.reset();
+		Sample totalEnergy = 0;
+		for (int c = 0; c < channels; ++c) {
+			auto &&inputChannel = inputs[c];
+			auto &&bufferChannel = inputBuffer[c];
+			int startIndex = std::max<int>(0, inputSamples - stft.windowSize() - stft.interval());
+			for (int i = startIndex; i < inputSamples; ++i) {
+				Sample s = inputChannel[i];
+				totalEnergy += s*s;
+				bufferChannel[i] = s;
+			}
+		}
+		if (totalEnergy >= noiseFloor) {
+			silenceCounter = 0;
+			silenceFirst = true;
+		}
+		inputBuffer += inputSamples;
+		didSeek = true;
+		seekTimeFactor = (playbackRate*stft.interval() > 1) ? 1/playbackRate : stft.interval();
+	}
+
+	template<class Inputs, class Outputs>
+	void process(Inputs &&inputs, int inputSamples, Outputs &&outputs, int outputSamples) {
+		Sample totalEnergy = 0;
+		for (int c = 0; c < channels; ++c) {
+			auto &&inputChannel = inputs[c];
+			for (int i = 0; i < inputSamples; ++i) {
+				Sample s = inputChannel[i];
+				totalEnergy += s*s;
+			}
+		}
+		if (totalEnergy < noiseFloor) {
+			if (silenceCounter >= 2*stft.windowSize()) {
+				if (silenceFirst) {
+					silenceFirst = false;
+					for (auto &b : channelBands) {
+						b.input = b.prevInput = b.output = 0;
+						b.inputEnergy = 0;
+					}
+				}
+			
+				if (inputSamples > 0) {
+					// copy from the input, wrapping around if needed
+					for (int outputIndex = 0; outputIndex < outputSamples; ++outputIndex) {
+						int inputIndex = outputIndex%inputSamples;
+						for (int c = 0; c < channels; ++c) {
+							outputs[c][outputIndex] = inputs[c][inputIndex];
+						}
+					}
+				} else {
+					for (int c = 0; c < channels; ++c) {
+						auto &&outputChannel = outputs[c];
+						for (int outputIndex = 0; outputIndex < outputSamples; ++outputIndex) {
+							outputChannel[outputIndex] = 0;
+						}
+					}
+				}
+
+				// Store input in history buffer
+				for (int c = 0; c < channels; ++c) {
+					auto &&inputChannel = inputs[c];
+					auto &&bufferChannel = inputBuffer[c];
+					int startIndex = std::max<int>(0, inputSamples - stft.windowSize() - stft.interval());
+					for (int i = startIndex; i < inputSamples; ++i) {
+						bufferChannel[i] = inputChannel[i];
+					}
+				}
+				inputBuffer += inputSamples;
+				return;
+			} else {
+				silenceCounter += inputSamples;
+			}
+		} else {
+			silenceCounter = 0;
+			silenceFirst = true;
+		}
+
+		for (int outputIndex = 0; outputIndex < outputSamples; ++outputIndex) {
+			stft.ensureValid(outputIndex, [&](int outputOffset) {
+				// Time to process a spectrum!  Where should it come from in the input?
+				int inputOffset = std::round(outputOffset*Sample(inputSamples)/outputSamples) - stft.windowSize();
+				int inputInterval = inputOffset - prevInputOffset;
+				prevInputOffset = inputOffset;
+
+				bool newSpectrum = didSeek || (inputInterval > 0);
+				if (newSpectrum) {
+					for (int c = 0; c < channels; ++c) {
+						// Copy from the history buffer, if needed
+						auto &&bufferChannel = inputBuffer[c];
+						for (int i = 0; i < -inputOffset; ++i) {
+							timeBuffer[i] = bufferChannel[i + inputOffset];
+						}
+						// Copy the rest from the input
+						auto &&inputChannel = inputs[c];
+						for (int i = std::max<int>(0, -inputOffset); i < stft.windowSize(); ++i) {
+							timeBuffer[i] = inputChannel[i + inputOffset];
+						}
+						stft.analyse(c, timeBuffer);
+					}
+					flushed = false; // TODO: first block after a flush should be gain-compensated
+
+					for (int c = 0; c < channels; ++c) {
+						auto channelBands = bandsForChannel(c);
+						auto &&spectrumBands = stft.spectrum[c];
+						for (int b = 0; b < bands; ++b) {
+							channelBands[b].input = spectrumBands[b];
+						}
+					}
+
+					if (didSeek || inputInterval != stft.interval()) { // make sure the previous input is the correct distance in the past
+						int prevIntervalOffset = inputOffset - stft.interval();
+						for (int c = 0; c < channels; ++c) {
+							// Copy from the history buffer, if needed
+							auto &&bufferChannel = inputBuffer[c];
+							for (int i = 0; i < std::min(-prevIntervalOffset, stft.windowSize()); ++i) {
+								timeBuffer[i] = bufferChannel[i + prevIntervalOffset];
+							}
+							// Copy the rest from the input
+							auto &&inputChannel = inputs[c];
+							for (int i = std::max<int>(0, -prevIntervalOffset); i < stft.windowSize(); ++i) {
+								timeBuffer[i] = inputChannel[i + prevIntervalOffset];
+							}
+							stft.analyse(c, timeBuffer);
+						}
+						for (int c = 0; c < channels; ++c) {
+							auto channelBands = bandsForChannel(c);
+							auto &&spectrumBands = stft.spectrum[c];
+							for (int b = 0; b < bands; ++b) {
+								channelBands[b].prevInput = spectrumBands[b];
+							}
+						}
+					}
+				}
+				
+				Sample timeFactor = didSeek ? seekTimeFactor : stft.interval()/std::max<Sample>(1, inputInterval);
+				processSpectrum(newSpectrum, timeFactor);
+				didSeek = false;
+
+				for (int c = 0; c < channels; ++c) {
+					auto channelBands = bandsForChannel(c);
+					auto &&spectrumBands = stft.spectrum[c];
+					for (int b = 0; b < bands; ++b) {
+						spectrumBands[b] = channelBands[b].output;
+					}
+				}
+			});
+
+			for (int c = 0; c < channels; ++c) {
+				auto &&outputChannel = outputs[c];
+				auto &&stftChannel = stft[c];
+				outputChannel[outputIndex] = stftChannel[outputIndex];
+			}
+		}
+
+		// Store input in history buffer
+		for (int c = 0; c < channels; ++c) {
+			auto &&inputChannel = inputs[c];
+			auto &&bufferChannel = inputBuffer[c];
+			int startIndex = std::max<int>(0, inputSamples - stft.windowSize());
+			for (int i = startIndex; i < inputSamples; ++i) {
+				bufferChannel[i] = inputChannel[i];
+			}
+		}
+		inputBuffer += inputSamples;
+		stft += outputSamples;
+		prevInputOffset -= inputSamples;
+	}
+
+	// Read the remaining output, providing no further input.  `outputSamples` should ideally be at least `.outputLatency()`
+	template<class Outputs>
+	void flush(Outputs &&outputs, int outputSamples) {
+		int plainOutput = std::min<int>(outputSamples, stft.windowSize());
+		int foldedBackOutput = std::min<int>(outputSamples, stft.windowSize() - plainOutput);
+		for (int c = 0; c < channels; ++c) {
+			auto &&outputChannel = outputs[c];
+			auto &&stftChannel = stft[c];
+			for (int i = 0; i < plainOutput; ++i) {
+				// TODO: plain output should be gain-
+				outputChannel[i] = stftChannel[i];
+			}
+			for (int i = 0; i < foldedBackOutput; ++i) {
+				outputChannel[outputSamples - 1 - i] -= stftChannel[plainOutput + i];
+			}
+			for (int i = 0; i < plainOutput + foldedBackOutput; ++i) {
+				stftChannel[i] = 0;
+			}
+		}
+		// Skip the output we just used/cleared
+		stft += plainOutput + foldedBackOutput;
+		// Reset the phase-vocoder stuff, so the next block gets a fresh start
+		for (int c = 0; c < channels; ++c) {
+			auto channelBands = bandsForChannel(c);
+			for (int b = 0; b < bands; ++b) {
+				channelBands[b].prevInput = channelBands[b].output = 0;
+			}
+		}
+		flushed = true;
+	}
+private:
+	using Complex = std::complex<Sample>;
+	static constexpr Sample noiseFloor{1e-15};
+	static constexpr Sample maxCleanStretch{2}; // time-stretch ratio before we start randomising phases
+	int silenceCounter = 0;
+	bool silenceFirst = true;
+
+	Sample freqMultiplier = 1, freqTonalityLimit = 0.5;
+	std::function<Sample(Sample)> customFreqMap = nullptr;
+
+	signalsmith::spectral::STFT<Sample> stft{0, 1, 1};
+	signalsmith::delay::MultiBuffer<Sample> inputBuffer;
+	int channels = 0, bands = 0;
+	int prevInputOffset = -1;
+	std::vector<Sample> timeBuffer;
+	bool didSeek = false, flushed = true;
+	Sample seekTimeFactor = 1;
+
+	Sample bandToFreq(Sample b) const {
+		return (b + Sample(0.5))/stft.fftSize();
+	}
+	Sample freqToBand(Sample f) const {
+		return f*stft.fftSize() - Sample(0.5);
+	}
+	
+	struct Band {
+		Complex input, prevInput{0};
+		Complex output{0};
+		Sample inputEnergy;
+	};
+	std::vector<Band> channelBands;
+	Band * bandsForChannel(int channel) {
+		return channelBands.data() + channel*bands;
+	}
+	template<Complex Band::*member>
+	Complex getBand(int channel, int index) {
+		if (index < 0 || index >= bands) return 0;
+		return channelBands[index + channel*bands].*member;
+	}
+	template<Complex Band::*member>
+	Complex getFractional(int channel, int lowIndex, Sample fractional) {
+		Complex low = getBand<member>(channel, lowIndex);
+		Complex high = getBand<member>(channel, lowIndex + 1);
+		return low + (high - low)*fractional;
+	}
+	template<Complex Band::*member>
+	Complex getFractional(int channel, Sample inputIndex) {
+		int lowIndex = std::floor(inputIndex);
+		Sample fracIndex = inputIndex - lowIndex;
+		return getFractional<member>(channel, lowIndex, fracIndex);
+	}
+	template<Sample Band::*member>
+	Sample getBand(int channel, int index) {
+		if (index < 0 || index >= bands) return 0;
+		return channelBands[index + channel*bands].*member;
+	}
+	template<Sample Band::*member>
+	Sample getFractional(int channel, int lowIndex, Sample fractional) {
+		Sample low = getBand<member>(channel, lowIndex);
+		Sample high = getBand<member>(channel, lowIndex + 1);
+		return low + (high - low)*fractional;
+	}
+	template<Sample Band::*member>
+	Sample getFractional(int channel, Sample inputIndex) {
+		int lowIndex = std::floor(inputIndex);
+		Sample fracIndex = inputIndex - lowIndex;
+		return getFractional<member>(channel, lowIndex, fracIndex);
+	}
+
+	struct Peak {
+		Sample input, output;
+	};
+	std::vector<Peak> peaks;
+	std::vector<Sample> energy, smoothedEnergy;
+	struct PitchMapPoint {
+		Sample inputBin, freqGrad;
+	};
+	std::vector<PitchMapPoint> outputMap;
+	
+	struct Prediction {
+		Sample energy = 0;
+		Complex input;
+
+		Complex makeOutput(Complex phase) {
+			Sample phaseNorm = std::norm(phase);
+			if (phaseNorm <= noiseFloor) {
+				phase = input; // prediction is too weak, fall back to the input
+				phaseNorm = std::norm(input) + noiseFloor;
+			}
+			return phase*std::sqrt(energy/phaseNorm);
+		}
+	};
+	std::vector<Prediction> channelPredictions;
+	Prediction * predictionsForChannel(int c) {
+		return channelPredictions.data() + c*bands;
+	}
+
+	RandomEngine randomEngine;
+
+	void processSpectrum(bool newSpectrum, Sample timeFactor) {
+		timeFactor = std::max<Sample>(timeFactor, 1/maxCleanStretch);
+		bool randomTimeFactor = (timeFactor > maxCleanStretch);
+		std::uniform_real_distribution<Sample> timeFactorDist(maxCleanStretch*2*randomTimeFactor - timeFactor, timeFactor);
+		
+		if (newSpectrum) {
+			for (int c = 0; c < channels; ++c) {
+				auto bins = bandsForChannel(c);
+
+				Complex rot = std::polar(Sample(1), bandToFreq(0)*stft.interval()*Sample(2*M_PI));
+				Sample freqStep = bandToFreq(1) - bandToFreq(0);
+				Complex rotStep = std::polar(Sample(1), freqStep*stft.interval()*Sample(2*M_PI));
+				
+				for (int b = 0; b < bands; ++b) {
+					auto &bin = bins[b];
+					bin.output = signalsmith::perf::mul(bin.output, rot);
+					bin.prevInput = signalsmith::perf::mul(bin.prevInput, rot);
+					rot = signalsmith::perf::mul(rot, rotStep);
+				}
+			}
+		}
+
+		Sample smoothingBins = Sample(stft.fftSize())/stft.interval();
+		int longVerticalStep = std::round(smoothingBins);
+		if (customFreqMap || freqMultiplier != 1) {
+			findPeaks(smoothingBins);
+			updateOutputMap();
+		} else { // we're not pitch-shifting, so no need to find peaks etc.
+			for (int c = 0; c < channels; ++c) {
+				Band *bins = bandsForChannel(c);
+				for (int b = 0; b < bands; ++b) {
+					bins[b].inputEnergy = std::norm(bins[b].input);
+				}
+			}
+			for (int b = 0; b < bands; ++b) {
+				outputMap[b] = {Sample(b), 1};
+			}
+		}
+
+		// Preliminary output prediction from phase-vocoder
+		for (int c = 0; c < channels; ++c) {
+			Band *bins = bandsForChannel(c);
+			auto *predictions = predictionsForChannel(c);
+			for (int b = 0; b < bands; ++b) {
+				auto mapPoint = outputMap[b];
+				int lowIndex = std::floor(mapPoint.inputBin);
+				Sample fracIndex = mapPoint.inputBin - lowIndex;
+
+				Prediction &prediction = predictions[b];
+				Sample prevEnergy = prediction.energy;
+				prediction.energy = getFractional<&Band::inputEnergy>(c, lowIndex, fracIndex);
+				prediction.energy *= std::max<Sample>(0, mapPoint.freqGrad); // scale the energy according to local stretch factor
+				prediction.input = getFractional<&Band::input>(c, lowIndex, fracIndex);
+
+				auto &outputBin = bins[b];
+				Complex prevInput = getFractional<&Band::prevInput>(c, lowIndex, fracIndex);
+				Complex freqTwist = signalsmith::perf::mul<true>(prediction.input, prevInput);
+				Complex phase = signalsmith::perf::mul(outputBin.output, freqTwist);
+				outputBin.output = phase/(std::max(prevEnergy, prediction.energy) + noiseFloor);
+			}
+		}
+
+		// Re-predict using phase differences between frequencies
+		for (int b = 0; b < bands; ++b) {
+			// Find maximum-energy channel and calculate that
+			int maxChannel = 0;
+			Sample maxEnergy = predictionsForChannel(0)[b].energy;
+			for (int c = 1; c < channels; ++c) {
+				Sample e = predictionsForChannel(c)[b].energy;
+				if (e > maxEnergy) {
+					maxChannel = c;
+					maxEnergy = e;
+				}
+			}
+
+			auto *predictions = predictionsForChannel(maxChannel);
+			auto &prediction = predictions[b];
+			auto *bins = bandsForChannel(maxChannel);
+			auto &outputBin = bins[b];
+
+			Complex phase = 0;
+			auto mapPoint = outputMap[b];
+
+			// Upwards vertical steps
+			if (b > 0) {
+				Sample binTimeFactor = randomTimeFactor ? timeFactorDist(randomEngine) : timeFactor;
+				Complex downInput = getFractional<&Band::input>(maxChannel, mapPoint.inputBin - binTimeFactor);
+				Complex shortVerticalTwist = signalsmith::perf::mul<true>(prediction.input, downInput);
+
+				auto &downBin = bins[b - 1];
+				phase += signalsmith::perf::mul(downBin.output, shortVerticalTwist);
+				
+				if (b >= longVerticalStep) {
+					Complex longDownInput = getFractional<&Band::input>(maxChannel, mapPoint.inputBin - longVerticalStep*binTimeFactor);
+					Complex longVerticalTwist = signalsmith::perf::mul<true>(prediction.input, longDownInput);
+
+					auto &longDownBin = bins[b - longVerticalStep];
+					phase += signalsmith::perf::mul(longDownBin.output, longVerticalTwist);
+				}
+			}
+			// Downwards vertical steps
+			if (b < bands - 1) {
+				auto &upPrediction = predictions[b + 1];
+				auto &upMapPoint = outputMap[b + 1];
+
+				Sample binTimeFactor = randomTimeFactor ? timeFactorDist(randomEngine) : timeFactor;
+				Complex downInput = getFractional<&Band::input>(maxChannel, upMapPoint.inputBin - binTimeFactor);
+				Complex shortVerticalTwist = signalsmith::perf::mul<true>(upPrediction.input, downInput);
+
+				auto &upBin = bins[b + 1];
+				phase += signalsmith::perf::mul<true>(upBin.output, shortVerticalTwist);
+				
+				if (b < bands - longVerticalStep) {
+					auto &longUpPrediction = predictions[b + longVerticalStep];
+					auto &longUpMapPoint = outputMap[b + longVerticalStep];
+
+					Complex longDownInput = getFractional<&Band::input>(maxChannel, longUpMapPoint.inputBin - longVerticalStep*binTimeFactor);
+					Complex longVerticalTwist = signalsmith::perf::mul<true>(longUpPrediction.input, longDownInput);
+
+					auto &longUpBin = bins[b + longVerticalStep];
+					phase += signalsmith::perf::mul<true>(longUpBin.output, longVerticalTwist);
+				}
+			}
+
+			outputBin.output = prediction.makeOutput(phase);
+			
+			// All other bins are locked in phase
+			for (int c = 0; c < channels; ++c) {
+				if (c != maxChannel) {
+					auto &channelBin = bandsForChannel(c)[b];
+					auto &channelPrediction = predictionsForChannel(c)[b];
+					
+					Complex channelTwist = signalsmith::perf::mul<true>(channelPrediction.input, prediction.input);
+					Complex channelPhase = signalsmith::perf::mul(outputBin.output, channelTwist);
+					channelBin.output = channelPrediction.makeOutput(channelPhase);
+				}
+			}
+		}
+
+		if (newSpectrum) {
+			for (auto &bin : channelBands) {
+				bin.prevInput = bin.input;
+			}
+		}
+	}
+	
+	// Produces smoothed energy across all channels
+	void smoothEnergy(Sample smoothingBins) {
+		Sample smoothingSlew = 1/(1 + smoothingBins*Sample(0.5));
+		for (auto &e : energy) e = 0;
+		for (int c = 0; c < channels; ++c) {
+			Band *bins = bandsForChannel(c);
+			for (int b = 0; b < bands; ++b) {
+				Sample e = std::norm(bins[b].input);
+				bins[b].inputEnergy = e; // Used for interpolating prediction energy
+				energy[b] += e;
+			}
+		}
+		for (int b = 0; b < bands; ++b) {
+			smoothedEnergy[b] = energy[b];
+		}
+		Sample e = 0;
+		for (int repeat = 0; repeat < 2; ++repeat) {
+			for (int b = bands - 1; b >= 0; --b) {
+				e += (smoothedEnergy[b] - e)*smoothingSlew;
+				smoothedEnergy[b] = e;
+			}
+			for (int b = 0; b < bands; ++b) {
+				e += (smoothedEnergy[b] - e)*smoothingSlew;
+				smoothedEnergy[b] = e;
+			}
+		}
+	}
+	
+	Sample mapFreq(Sample freq) const {
+		if (customFreqMap) return customFreqMap(freq);
+		if (freq > freqTonalityLimit) {
+			Sample diff = freq - freqTonalityLimit;
+			return freqTonalityLimit*freqMultiplier + diff;
+		}
+		return freq*freqMultiplier;
+	}
+	
+	// Identifies spectral peaks using energy across all channels
+	void findPeaks(Sample smoothingBins) {
+		smoothEnergy(smoothingBins);
+
+		peaks.resize(0);
+		
+		int start = 0;
+		while (start < bands) {
+			if (energy[start] > smoothedEnergy[start]) {
+				int end = start;
+				Sample bandSum = 0, energySum = 0;
+				while (end < bands && energy[end] > smoothedEnergy[end]) {
+					bandSum += end*energy[end];
+					energySum += energy[end];
+					++end;
+				}
+				Sample avgBand = bandSum/energySum;
+				Sample avgFreq = bandToFreq(avgBand);
+				peaks.emplace_back(Peak{avgBand, freqToBand(mapFreq(avgFreq))});
+
+				start = end;
+			}
+			++start;
+		}
+	}
+	
+	void updateOutputMap() {
+		if (peaks.empty()) {
+			for (int b = 0; b < bands; ++b) {
+				outputMap[b] = {Sample(b), 1};
+			}
+			return;
+		}
+		Sample bottomOffset = peaks[0].input - peaks[0].output;
+		for (int b = 0; b < std::min<int>(bands, std::ceil(peaks[0].output)); ++b) {
+			outputMap[b] = {b + bottomOffset, 1};
+		}
+		// Interpolate between points
+		for (size_t p = 1; p < peaks.size(); ++p) {
+			const Peak &prev = peaks[p - 1], &next = peaks[p];
+			Sample rangeScale = 1/(next.output - prev.output);
+			Sample outOffset = prev.input - prev.output;
+			Sample outScale = next.input - next.output - prev.input + prev.output;
+			Sample gradScale = outScale*rangeScale;
+			int startBin = std::max<int>(0, std::ceil(prev.output));
+			int endBin = std::min<int>(bands, std::ceil(next.output));
+			for (int b = startBin; b < endBin; ++b) {
+				Sample r = (b - prev.output)*rangeScale;
+				Sample h = r*r*(3 - 2*r);
+				Sample outB = b + outOffset + h*outScale;
+				
+				Sample gradH = 6*r*(1 - r);
+				Sample gradB = 1 + gradH*gradScale;
+				
+				outputMap[b] = {outB, gradB};
+			}
+		}
+		Sample topOffset = peaks.back().input - peaks.back().output;
+		for (int b = std::max<int>(0, peaks.back().output); b < bands; ++b) {
+			outputMap[b] = {b + topOffset, 1};
+		}
+	}
+};
+
+}} // namespace
+#endif // include guard

package/common/cpp/types/TimeStretchType.h

@@ -0,0 +1,6 @@
+#pragma once
+
+namespace audioapi {
+
+enum class TimeStretchType {LINEAR, SPEECH, MUSIC};
+} // namespace audioapi

package/ios/AudioAPIModule.h

@@ -9,7 +9,7 @@
 
 @interface AudioAPIModule : RCTEventEmitter
 #ifdef RCT_NEW_ARCH_ENABLED
-                            <RCTCallInvokerModule>
+                            <NativeAudioAPIModuleSpec, RCTCallInvokerModule>
 #else
                             <RCTBridgeModule>
 #endif // RCT_NEW_ARCH_ENABLED

package/ios/AudioAPIModule.mm

@@ -6,7 +6,7 @@
 #import <ReactCommon/RCTTurboModule.h>
 #endif // RCT_NEW_ARCH_ENABLED
 
-#import "AudioAPIInstallerHostObject.h"
+#include "AudioAPIModuleInstaller.h"
 
 @implementation AudioAPIModule
 
@@ -38,11 +38,18 @@ RCT_EXPORT_BLOCKING_SYNCHRONOUS_METHOD(install)
 
   assert(jsiRuntime != nullptr);
 
-  auto hostObject = std::make_shared<audioapi::AudioAPIInstallerHostObject>(jsiRuntime, jsCallInvoker);
-  hostObject->install();
+  audioapi::AudioAPIModuleInstaller::injectJSIBindings(jsiRuntime, jsCallInvoker);
 
   NSLog(@"Successfully installed JSI bindings for react-native-audio-api!");
   return @true;
 }
 
+#ifdef RCT_NEW_ARCH_ENABLED
+- (std::shared_ptr<facebook::react::TurboModule>)getTurboModule:
+    (const facebook::react::ObjCTurboModule::InitParams &)params
+{
+  return std::make_shared<facebook::react::NativeAudioAPIModuleSpecJSI>(params);
+}
+#endif // RCT_NEW_ARCH_ENABLED
+
 @end

package/ios/core/AudioDecoder.mm

@@ -7,7 +7,7 @@
 
 namespace audioapi {
 
-AudioBus *AudioDecoder::decodeWithFilePath(const std::string &path) const
+std::shared_ptr<AudioBus> AudioDecoder::decodeWithFilePath(const std::string &path) const
 {
   ma_decoder decoder;
   ma_decoder_config config = ma_decoder_config_init(ma_format_f32, 2, sampleRate_);
@@ -23,7 +23,7 @@ AudioBus *AudioDecoder::decodeWithFilePath(const std::string &path) const
   ma_uint64 totalFrameCount;
   ma_decoder_get_length_in_pcm_frames(&decoder, &totalFrameCount);
 
-  auto *audioBus = new AudioBus(static_cast<int>(totalFrameCount), 2, sampleRate_);
+  auto audioBus = std::make_shared<AudioBus>(static_cast<int>(totalFrameCount), 2, sampleRate_);
   auto *buffer = new float[totalFrameCount * 2];
 
   ma_uint64 framesDecoded;
@@ -32,7 +32,6 @@ AudioBus *AudioDecoder::decodeWithFilePath(const std::string &path) const
     NSLog(@"Failed to decode audio file: %s", path.c_str());
 
     delete[] buffer;
-    delete audioBus;
     ma_decoder_uninit(&decoder);
 
     return nullptr;