react-native-audio-api 0.5.6 → 0.6.0-rc.0

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

@@ -1,82 +0,0 @@
-#ifndef SIGNALSMITH_DSP_PERF_H
-#define SIGNALSMITH_DSP_PERF_H
-
-#include <complex>
-
-#if defined(__SSE__) || defined(_M_X64)
-#	include <xmmintrin.h>
-#else
-#	include <cstdint> // for uintptr_t
-#endif
-
-namespace signalsmith {
-namespace perf {
-	/**	@defgroup Performance Performance helpers
-		@brief Nothing serious, just some `#defines` and helpers
-
-		@{
-		@file
-	*/
-
-	/// *Really* insist that a function/method is inlined (mostly for performance in DEBUG builds)
-	#ifndef SIGNALSMITH_INLINE
-	#ifdef __GNUC__
-	#define SIGNALSMITH_INLINE __attribute__((always_inline)) inline
-	#elif defined(__MSVC__)
-	#define SIGNALSMITH_INLINE __forceinline inline
-	#else
-	#define SIGNALSMITH_INLINE inline
-	#endif
-	#endif
-
-	/** @brief Complex-multiplication (with optional conjugate second-arg), without handling NaN/Infinity
-		The `std::complex` multiplication has edge-cases around NaNs which slow things down and prevent auto-vectorisation.  Flags like `-ffast-math` sort this out anyway, but this helps with Debug builds.
-	*/
-	template <bool conjugateSecond=false, typename V>
-	SIGNALSMITH_INLINE static std::complex<V> mul(const std::complex<V> &a, const std::complex<V> &b) {
-		return conjugateSecond ? std::complex<V>{
-			b.real()*a.real() + b.imag()*a.imag(),
-			b.real()*a.imag() - b.imag()*a.real()
-		} : std::complex<V>{
-			a.real()*b.real() - a.imag()*b.imag(),
-			a.real()*b.imag() + a.imag()*b.real()
-		};
-	}
-
-#if defined(__SSE__) || defined(_M_X64)
-	class StopDenormals {
-		unsigned int controlStatusRegister;
-	public:
-		StopDenormals() : controlStatusRegister(_mm_getcsr()) {
-			_mm_setcsr(controlStatusRegister|0x8040); // Flush-to-Zero and Denormals-Are-Zero
-		}
-		~StopDenormals() {
-			_mm_setcsr(controlStatusRegister);
-		}
-	};
-#elif (defined (__ARM_NEON) || defined (__ARM_NEON__))
-	class StopDenormals {
-		uintptr_t status;
-	public:
-		StopDenormals() {
-			uintptr_t asmStatus;
-			asm volatile("mrs %0, fpcr" : "=r"(asmStatus));
-			status = asmStatus = asmStatus|0x01000000U; // Flush to Zero
-			asm volatile("msr fpcr, %0" : : "ri"(asmStatus));
-		}
-		~StopDenormals() {
-			uintptr_t asmStatus = status;
-			asm volatile("msr fpcr, %0" : : "ri"(asmStatus));
-		}
-	};
-#else
-#	if __cplusplus >= 202302L
-# 		warning "The `StopDenormals` class doesn't do anything for this architecture"
-#	endif
-	class StopDenormals {}; // FIXME: add for other architectures
-#endif
-
-/** @} */
-}} // signalsmith::perf::
-
-#endif // include guard

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

@@ -1,493 +0,0 @@
-#include <audioapi/core/Constants.h>
-
-#ifndef SIGNALSMITH_DSP_SPECTRAL_H
-#define SIGNALSMITH_DSP_SPECTRAL_H
-
-#include <audioapi/libs/signalsmith-stretch/perf.h>
-#include <audioapi/libs/signalsmith-stretch/fft.h>
-#include <audioapi/libs/signalsmith-stretch/delay.h>
-
-#include <audioapi/dsp/Windows.h>
-
-#include <cmath>
-
-namespace signalsmith {
-namespace spectral {
-	/**	@defgroup Spectral Spectral Processing
-		@brief Tools for frequency-domain manipulation of audio signals
-
-		@{
-		@file
-	*/
-
-	/** @brief An FFT with built-in windowing and round-trip scaling
-
-		This uses a Modified Real FFT, which applies half-bin shift before the transform.  The result therefore has `N/2` bins, centred at the frequencies: `(i + 0.5)/N`.
-
-		This avoids the awkward (real-valued) bands for DC-offset and Nyquist.
-	 */
-	template<typename Sample>
-	class WindowedFFT {
-		using MRFFT = signalsmith::fft::ModifiedRealFFT<Sample>;
-		using Complex = std::complex<Sample>;
-		MRFFT mrfft{2};
-
-		std::vector<Sample> fftWindow;
-		std::vector<Sample> timeBuffer;
-		int offsetSamples = 0;
-	public:
-		/// Returns a fast FFT size <= `size`
-		static int fastSizeAbove(int size, int divisor=1) {
-			return MRFFT::fastSizeAbove(size/divisor)*divisor;
-		}
-		/// Returns a fast FFT size >= `size`
-		static int fastSizeBelow(int size, int divisor=1) {
-			return MRFFT::fastSizeBelow(1 + (size - 1)/divisor)*divisor;
-		}
-
-		WindowedFFT() {}
-		WindowedFFT(int size, int rotateSamples=0) {
-			setSize(size, rotateSamples);
-		}
-		template<class WindowFn>
-		WindowedFFT(int size, WindowFn fn, Sample windowOffset=0.5, int rotateSamples=0) {
-			setSize(size, fn, windowOffset, rotateSamples);
-		}
-
-		/// Sets the size, returning the window for modification (initially all 1s)
-		std::vector<Sample> & setSizeWindow(int size, int rotateSamples=0) {
-			mrfft.setSize(size);
-			fftWindow.assign(size, 1);
-			timeBuffer.resize(size);
-			offsetSamples = rotateSamples;
-			if (offsetSamples < 0) offsetSamples += size; // TODO: for a negative rotation, the other half of the result is inverted
-			return fftWindow;
-		}
-		/// Sets the FFT size, with a user-defined functor for the window
-		template<class WindowFn>
-		void setSize(int size, WindowFn fn, Sample windowOffset=0.5, int rotateSamples=0) {
-			setSizeWindow(size, rotateSamples);
-
-			Sample invSize = 1/(Sample)size;
-			for (int i = 0; i < size; ++i) {
-				Sample r = (i + windowOffset)*invSize;
-				fftWindow[i] = fn(r);
-			}
-		}
-		/// Sets the size (using the default Blackman-Harris window)
-		void setSize(int size, int rotateSamples=0) {
-			setSize(size, [](double x) {
-				double phase = 2 * audioapi::PI * x;
-				// Blackman-Harris
-				return 0.35875 - 0.48829*std::cos(phase) + 0.14128*std::cos(phase*2) - 0.01168*std::cos(phase*3);
-			}, Sample(0.5), rotateSamples);
-		}
-
-		const std::vector<Sample> & window() const {
-			return this->fftWindow;
-		}
-		int size() const {
-			return mrfft.size();
-		}
-
-		/// Performs an FFT, with windowing and rotation (if enabled)
-		template<bool withWindow=true, bool withScaling=false, class Input, class Output>
-		void fft(Input &&input, Output &&output) {
-			int fftSize = size();
-			const Sample norm = (withScaling ? 1/(Sample)fftSize : 1);
-			for (int i = 0; i < offsetSamples; ++i) {
-				// Inverted polarity since we're using the MRFFT
-				timeBuffer[i + fftSize - offsetSamples] = -input[i]*norm*(withWindow ? fftWindow[i] : Sample(1));
-			}
-			for (int i = offsetSamples; i < fftSize; ++i) {
-				timeBuffer[i - offsetSamples] = input[i]*norm*(withWindow ? fftWindow[i] : Sample(1));
-			}
-			mrfft.fft(timeBuffer, output);
-		}
-		/// Performs an FFT (no windowing or rotation)
-		template<class Input, class Output>
-		void fftRaw(Input &&input, Output &&output) {
-			mrfft.fft(input, output);
-		}
-
-		/// Inverse FFT, with windowing, 1/N scaling and rotation (if enabled)
-		template<bool withWindow=true, bool withScaling=true, class Input, class Output>
-		void ifft(Input &&input, Output &&output) {
-			mrfft.ifft(input, timeBuffer);
-			int fftSize = mrfft.size();
-			const Sample norm = (withScaling ? 1/(Sample)fftSize : 1);
-
-			for (int i = 0; i < offsetSamples; ++i) {
-				// Inverted polarity since we're using the MRFFT
-				output[i] = -timeBuffer[i + fftSize - offsetSamples]*norm*(withWindow ? fftWindow[i] : Sample(1));
-			}
-			for (int i = offsetSamples; i < fftSize; ++i) {
-				output[i] = timeBuffer[i - offsetSamples]*norm*(withWindow ? fftWindow[i] : Sample(1));
-			}
-		}
-		/// Performs an IFFT (no windowing, scaling or rotation)
-		template<class Input, class Output>
-		void ifftRaw(Input &&input, Output &&output) {
-			mrfft.ifft(input, output);
-		}
-	};
-
-	/** STFT synthesis, built on a `MultiBuffer`.
-
-		Any window length and block interval is supported, but the FFT size may be rounded up to a faster size (by zero-padding).  It uses a heuristically-optimal Kaiser window modified for perfect-reconstruction.
-
-		\diagram{stft-aliasing-simulated.svg,Simulated bad-case aliasing (random phase-shift for each band) for overlapping ratios}
-
-		There is a "latest valid index", and you can read the output up to one `historyLength` behind this (see `.resize()`).  You can read up to one window-length _ahead_ to get partially-summed future output.
-
-		\diagram{stft-buffer-validity.svg}
-
-		You move the valid index along using `.ensureValid()`, passing in a functor which provides spectra (using `.analyse()` and/or direct modification through `.spectrum[c]`):
-
-		\code
-			void processSample(...) {
-				stft.ensureValid([&](int) {
-					// Here, we introduce (1 - windowSize) of latency
-					stft.analyse(inputBuffer.view(1 - windowSize))
-				});
-				// read as a MultiBuffer
-				auto result = stft.at(0);
-				++stft; // also moves the latest valid index
-			}
-
-			void processBlock(...) {
-				// assuming `historyLength` == blockSize
-				stft.ensureValid(blockSize, [&](int blockStartIndex) {
-					int inputStart = blockStartIndex + (1 - windowSize);
-					stft.analyse(inputBuffer.view(inputStart));
-				});
-				auto earliestValid = stft.at(0);
-				auto latestValid = stft.at(blockSize);
-				stft += blockSize;
-			}
-		\endcode
-
-		The index passed to this functor will be greater than the previous valid index, and `<=` the index you pass in.  Therefore, if you call `.ensureValid()` every sample, it can only ever be `0`.
-	*/
-	template<typename Sample>
-	class STFT : public signalsmith::delay::MultiBuffer<Sample> {
-
-		using Super = signalsmith::delay::MultiBuffer<Sample>;
-		using Complex = std::complex<Sample>;
-
-		int channels = 0, _windowSize = 0, _fftSize = 0, _interval = 1;
-		int validUntilIndex = 0;
-
-		class MultiSpectrum {
-			int channels, stride;
-			std::vector<Complex> buffer;
-		public:
-			MultiSpectrum() : MultiSpectrum(0, 0) {}
-			MultiSpectrum(int channels, int bands) : channels(channels), stride(bands), buffer(channels*bands, 0) {}
-
-			void resize(int nChannels, int nBands) {
-				channels = nChannels;
-				stride = nBands;
-				buffer.assign(channels*stride, 0);
-			}
-
-			void reset() {
-				buffer.assign(buffer.size(), 0);
-			}
-
-			void swap(MultiSpectrum &other) {
-				using std::swap;
-				swap(buffer, other.buffer);
-			}
-
-			Complex * operator [](int channel) {
-				return buffer.data() + channel*stride;
-			}
-			const Complex * operator [](int channel) const {
-				return buffer.data() + channel*stride;
-			}
-		};
-		std::vector<Sample> timeBuffer;
-
-		bool rotate = false;
-		void resizeInternal(int newChannels, int windowSize, int newInterval, int historyLength, int zeroPadding) {
-			Super::resize(newChannels,
-				windowSize /* for output summing */
-				+ newInterval /* so we can read `windowSize` ahead (we'll be at most `interval-1` from the most recent block */
-				+ historyLength);
-
-			int fftSize = fft.fastSizeAbove(windowSize + zeroPadding);
-
-			this->channels = newChannels;
-			_windowSize = windowSize;
-			this->_fftSize = fftSize;
-			this->_interval = newInterval;
-			validUntilIndex = -1;
-
-			setWindow(windowShape, rotate);
-
-			spectrum.resize(channels, fftSize/2);
-			timeBuffer.resize(fftSize);
-		}
-	public:
-		enum class Window {kaiser, acg};
-		/// \deprecated use `.setWindow()` which actually updates the window when you change it
-		Window windowShape = Window::kaiser;
-		// for convenience
-		static constexpr Window kaiser = Window::kaiser;
-		static constexpr Window acg = Window::acg;
-
-		/** Swaps between the default (Kaiser) shape and Approximate Confined Gaussian (ACG).
-		\diagram{stft-windows.svg,Default (Kaiser) windows and partial cumulative sum}
-		The ACG has better rolloff since its edges go to 0:
-		\diagram{stft-windows-acg.svg,ACG windows and partial cumulative sum}
-		However, it generally has worse performance in terms of total sidelobe energy, affecting worst-case aliasing levels for (most) higher overlap ratios:
-		\diagram{stft-aliasing-simulated-acg.svg,Simulated bad-case aliasing for ACG windows - compare with above}*/
-		// TODO: these should both be set before resize()
-		void setWindow(Window shape, bool rotateToZero=false) {
-			windowShape = shape;
-			rotate = rotateToZero;
-
-			auto &window = fft.setSizeWindow(_fftSize, rotateToZero ? _windowSize/2 : 0);
-			if (windowShape == Window::kaiser) {
-				/// Roughly optimal Kaiser for STFT analysis (forced to perfect reconstruction)
-				audioapi::dsp::Kaiser::withBandwidth(_windowSize/_interval, true).apply(window.data(), _windowSize);
-			} else {
-				audioapi::dsp::ApproximateConfinedGaussian::withBandwidth(_windowSize/_interval).apply(window.data(), _windowSize);
-			}
-            audioapi::dsp::WindowFunction::forcePerfectReconstruction(window.data(), _windowSize, _interval);
-
-			// TODO: fill extra bits of an input buffer with NaN/Infinity, to break this, and then fix by adding zero-padding to WindowedFFT (as opposed to zero-valued window sections)
-			for (int i = _windowSize; i < _fftSize; ++i) {
-				window[i] = 0;
-			}
-		}
-
-		using Spectrum = MultiSpectrum;
-		Spectrum spectrum;
-		WindowedFFT<Sample> fft;
-
-		STFT() {}
-		/// Parameters passed straight to `.resize()`
-		STFT(int channels, int windowSize, int interval, int historyLength=0, int zeroPadding=0) {
-			resize(channels, windowSize, interval, historyLength, zeroPadding);
-		}
-
-		/// Sets the channel-count, FFT size and interval.
-		void resize(int nChannels, int windowSize, int interval, int historyLength=0, int zeroPadding=0) {
-			resizeInternal(nChannels, windowSize, interval, historyLength, zeroPadding);
-		}
-
-		int windowSize() const {
-			return _windowSize;
-		}
-		int fftSize() const {
-			return _fftSize;
-		}
-		int interval() const {
-			return _interval;
-		}
-		/// Returns the (analysis and synthesis) window
-		decltype(fft.window()) window() const {
-			return fft.window();
-		}
-		/// Calculates the effective window for the partially-summed future output (relative to the most recent block)
-		std::vector<Sample> partialSumWindow(bool includeLatestBlock=true) const {
-			const auto &w = window();
-			std::vector<Sample> result(_windowSize, 0);
-			int firstOffset = (includeLatestBlock ? 0 : _interval);
-			for (int offset = firstOffset; offset < _windowSize; offset += _interval) {
-				for (int i = 0; i < _windowSize - offset; ++i) {
-					Sample value = w[i + offset];
-					result[i] += value*value;
-				}
-			}
-			return result;
-		}
-
-		/// Resets everything - since we clear the output sum, it will take `windowSize` samples to get proper output.
-		void reset() {
-			Super::reset();
-			spectrum.reset();
-			validUntilIndex = -1;
-		}
-
-		/** Generates valid output up to the specified index (or 0), using the callback as many times as needed.
-
-			The callback should be a functor accepting a single integer argument, which is the index for which a spectrum is required.
-
-			The block created from these spectra will start at this index in the output, plus `.latency()`.
-		*/
-		template<class AnalysisFn>
-		void ensureValid(int i, AnalysisFn fn) {
-			while (validUntilIndex < i) {
-				int blockIndex = validUntilIndex + 1;
-				fn(blockIndex);
-
-				auto output = this->view(blockIndex);
-				for (int c = 0; c < channels; ++c) {
-					auto channel = output[c];
-
-					// Clear out the future sum, a window-length and an interval ahead
-					for (int wi = _windowSize; wi < _windowSize + _interval; ++wi) {
-						channel[wi] = 0;
-					}
-
-					// Add in the IFFT'd result
-					fft.ifft(spectrum[c], timeBuffer);
-					for (int wi = 0; wi < _windowSize; ++wi) {
-						channel[wi] += timeBuffer[wi];
-					}
-				}
-				validUntilIndex += _interval;
-			}
-		}
-		/// The same as above, assuming index 0
-		template<class AnalysisFn>
-		void ensureValid(AnalysisFn fn) {
-			return ensureValid(0, fn);
-		}
-		/// Returns the next invalid index (a.k.a. the index of the next block)
-		int nextInvalid() const {
-			return validUntilIndex + 1;
-		}
-
-		/** Analyse a multi-channel input, for any type where `data[channel][index]` returns samples
-
-		Results can be read/edited using `.spectrum`. */
-		template<class Data>
-		void analyse(Data &&data) {
-			for (int c = 0; c < channels; ++c) {
-				fft.fft(data[c], spectrum[c]);
-			}
-		}
-		template<class Data>
-		void analyse(int c, Data &&data) {
-			fft.fft(data, spectrum[c]);
-		}
-		/// Analyse without windowing or zero-rotation
-		template<class Data>
-		void analyseRaw(Data &&data) {
-			for (int c = 0; c < channels; ++c) {
-				fft.fftRaw(data[c], spectrum[c]);
-			}
-		}
-		template<class Data>
-		void analyseRaw(int c, Data &&data) {
-			fft.fftRaw(data, spectrum[c]);
-		}
-
-		int bands() const {
-			return _fftSize/2;
-		}
-
-		/** Internal latency (between the block-index requested in `.ensureValid()` and its position in the output)
-
-		Currently unused, but it's in here to allow for a future implementation which spreads the FFT calculations out across each interval.*/
-		int latency() {
-			return 0;
-		}
-
-		// @name Shift the underlying buffer (moving the "valid" index accordingly)
-		// @{
-		STFT & operator ++() {
-			Super::operator ++();
-			validUntilIndex--;
-			return *this;
-		}
-		STFT & operator +=(int i) {
-			Super::operator +=(i);
-			validUntilIndex -= i;
-			return *this;
-		}
-		STFT & operator --() {
-			Super::operator --();
-			validUntilIndex++;
-			return *this;
-		}
-		STFT & operator -=(int i) {
-			Super::operator -=(i);
-			validUntilIndex += i;
-			return *this;
-		}
-		// @}
-
-		typename Super::MutableView operator ++(int postIncrement) {
-			auto result = Super::operator ++(postIncrement);
-			validUntilIndex--;
-			return result;
-		}
-		typename Super::MutableView operator --(int postIncrement) {
-			auto result = Super::operator --(postIncrement);
-			validUntilIndex++;
-			return result;
-		}
-	};
-
-	/** STFT processing, with input/output.
-		Before calling `.ensureValid(index)`, you should make sure the input is filled up to `index`.
-	*/
-	template<typename Sample>
-	class ProcessSTFT : public STFT<Sample> {
-		using Super = STFT<Sample>;
-	public:
-		signalsmith::delay::MultiBuffer<Sample> input;
-
-		ProcessSTFT(int inChannels, int outChannels, int windowSize, int interval, int historyLength=0) {
-			resize(inChannels, outChannels, windowSize, interval, historyLength);
-		}
-
-		/** Alter the spectrum, using input up to this point, for the output block starting from this point.
-			Sub-classes should replace this with whatever processing is desired. */
-		virtual void processSpectrum(int /*blockIndex*/) {}
-
-		/// Sets the input/output channels, FFT size and interval.
-		void resize(int inChannels, int outChannels, int windowSize, int interval, int historyLength=0) {
-			Super::resize(outChannels, windowSize, interval, historyLength);
-			input.resize(inChannels, windowSize + interval + historyLength);
-		}
-		void reset(Sample value=Sample()) {
-			Super::reset(value);
-			input.reset(value);
-		}
-
-		/// Internal latency, including buffering samples for analysis.
-		int latency() {
-			return Super::latency() + (this->windowSize() - 1);
-		}
-
-		void ensureValid(int i=0) {
-			Super::ensureValid(i, [&](int blockIndex) {
-				this->analyse(input.view(blockIndex - this->windowSize() + 1));
-				this->processSpectrum(blockIndex);
-			});
-		}
-
-		// @name Shift the output, input, and valid index.
-		// @{
-		ProcessSTFT & operator ++() {
-			Super::operator ++();
-			++input;
-			return *this;
-		}
-		ProcessSTFT & operator +=(int i) {
-			Super::operator +=(i);
-			input += i;
-			return *this;
-		}
-		ProcessSTFT & operator --() {
-			Super::operator --();
-			--input;
-			return *this;
-		}
-		ProcessSTFT & operator -=(int i) {
-			Super::operator -=(i);
-			input -= i;
-			return *this;
-		}
-		// @}
-	};
-
-/** @} */
-}} // signalsmith::spectral::
-#endif // include guard