react-native-audio-api 0.4.12 → 0.4.14

package/src/api.ts

@@ -30,6 +30,7 @@ export { default as BiquadFilterNode } from './core/BiquadFilterNode';
 export { default as GainNode } from './core/GainNode';
 export { default as OscillatorNode } from './core/OscillatorNode';
 export { default as StereoPannerNode } from './core/StereoPannerNode';
+export { default as StretcherNode } from './core/StretcherNode';
 
 export {
   OscillatorType,
@@ -39,7 +40,6 @@ export {
   ContextState,
   WindowType,
   PeriodicWaveConstraints,
-  TimeStretchType,
 } from './types';
 
 export {

package/src/api.web.ts

@@ -23,7 +23,6 @@ export {
   ContextState,
   WindowType,
   PeriodicWaveConstraints,
-  TimeStretchType,
 } from './types';
 
 export {

package/src/core/AudioBufferSourceNode.ts

@@ -4,7 +4,6 @@ import BaseAudioContext from './BaseAudioContext';
 import AudioBuffer from './AudioBuffer';
 import AudioParam from './AudioParam';
 import { InvalidStateError, RangeError } from '../errors';
-import { TimeStretchType } from '../types';
 
 export default class AudioBufferSourceNode extends AudioScheduledSourceNode {
   readonly playbackRate: AudioParam;
@@ -58,14 +57,6 @@ export default class AudioBufferSourceNode extends AudioScheduledSourceNode {
     (this.node as IAudioBufferSourceNode).loopEnd = value;
   }
 
-  public get timeStretch(): TimeStretchType {
-    return (this.node as IAudioBufferSourceNode).timeStretch;
-  }
-
-  public set timeStretch(value: TimeStretchType) {
-    (this.node as IAudioBufferSourceNode).timeStretch = value;
-  }
-
   public start(when: number = 0, offset: number = 0, duration?: number): void {
     if (when < 0) {
       throw new RangeError(

package/src/core/AudioScheduledSourceNode.ts

@@ -35,9 +35,4 @@ export default class AudioScheduledSourceNode extends AudioNode {
 
     (this.node as IAudioScheduledSourceNode).stop(when);
   }
-
-  // eslint-disable-next-line accessor-pairs
-  public set onended(callback: (arg?: number) => void) {
-    (this.node as IAudioScheduledSourceNode).onended = callback;
-  }
 }

package/src/core/BaseAudioContext.ts

@@ -9,6 +9,7 @@ import AudioBufferSourceNode from './AudioBufferSourceNode';
 import AudioBuffer from './AudioBuffer';
 import PeriodicWave from './PeriodicWave';
 import AnalyserNode from './AnalyserNode';
+import StretcherNode from './StretcherNode';
 import { InvalidAccessError, NotSupportedError } from '../errors';
 
 export default class BaseAudioContext {
@@ -100,6 +101,10 @@ export default class BaseAudioContext {
     return new AnalyserNode(this, this.context.createAnalyser());
   }
 
+  createStretcher(): StretcherNode {
+    return new StretcherNode(this, this.context.createStretcher());
+  }
+
   async decodeAudioDataSource(sourcePath: string): Promise<AudioBuffer> {
     // Remove the file:// prefix if it exists
     if (sourcePath.startsWith('file://')) {

package/src/core/StretcherNode.ts

@@ -0,0 +1,15 @@
+import { IStretcherNode } from '../interfaces';
+import AudioNode from './AudioNode';
+import AudioParam from './AudioParam';
+import BaseAudioContext from './BaseAudioContext';
+
+export default class StretcherNode extends AudioNode {
+  readonly rate: AudioParam;
+  readonly semitones: AudioParam;
+
+  constructor(context: BaseAudioContext, stretcher: IStretcherNode) {
+    super(context, stretcher);
+    this.rate = new AudioParam(stretcher.rate);
+    this.semitones = new AudioParam(stretcher.semitones);
+  }
+}

package/src/interfaces.ts

@@ -5,7 +5,6 @@ import {
   ChannelCountMode,
   ChannelInterpretation,
   WindowType,
-  TimeStretchType,
 } from './types';
 
 export interface AudioAPIInstaller {
@@ -34,6 +33,7 @@ export interface IBaseAudioContext {
     disableNormalization: boolean
   ) => IPeriodicWave;
   createAnalyser: () => IAnalyserNode;
+  createStretcher: () => IStretcherNode;
   decodeAudioDataSource: (sourcePath: string) => Promise<IAudioBuffer>;
 }
 
@@ -82,7 +82,6 @@ export interface IAudioDestinationNode extends IAudioNode {}
 export interface IAudioScheduledSourceNode extends IAudioNode {
   start(when?: number): void;
   stop: (when: number) => void;
-  onended: (arg?: number) => void | null;
 }
 
 export interface IOscillatorNode extends IAudioScheduledSourceNode {
@@ -100,7 +99,6 @@ export interface IAudioBufferSourceNode extends IAudioScheduledSourceNode {
   loopEnd: number;
   detune: IAudioParam;
   playbackRate: IAudioParam;
-  timeStretch: TimeStretchType;
 
   start: (when?: number, offset?: number, duration?: number) => void;
 }
@@ -162,3 +160,8 @@ export interface IAnalyserNode extends IAudioNode {
   getFloatTimeDomainData: (array: number[]) => void;
   getByteTimeDomainData: (array: number[]) => void;
 }
+
+export interface IStretcherNode extends IAudioNode {
+  readonly rate: IAudioParam;
+  readonly semitones: IAudioParam;
+}

package/src/types.ts

@@ -30,5 +30,3 @@ export interface AudioContextOptions {
 }
 
 export type WindowType = 'blackman' | 'hann';
-
-export type TimeStretchType = 'linear' | 'speech-music';

package/src/web-core/AudioBufferSourceNode.tsx

@@ -3,7 +3,6 @@ import AudioParam from './AudioParam';
 import AudioBuffer from './AudioBuffer';
 import { InvalidStateError, RangeError } from '../errors';
 import BaseAudioContext from './BaseAudioContext';
-import { TimeStretchType } from '../types';
 
 export default class AudioBufferSourceNode extends AudioScheduledSourceNode {
   readonly playbackRate: AudioParam;
@@ -62,16 +61,6 @@ export default class AudioBufferSourceNode extends AudioScheduledSourceNode {
     (this.node as globalThis.AudioBufferSourceNode).loopEnd = value;
   }
 
-  public get timeStretch(): TimeStretchType {
-    return 'linear';
-  }
-
-  public set timeStretch(value: TimeStretchType) {
-    console.log(
-      'React Native Audio API: setting timeStretch is not supported on web'
-    );
-  }
-
   public start(when?: number, offset?: number, duration?: number): void {
     if (when && when < 0) {
       throw new RangeError(

package/src/web-core/AudioScheduledSourceNode.tsx

@@ -34,13 +34,4 @@ export default class AudioScheduledSourceNode extends AudioNode {
 
     (this.node as globalThis.AudioScheduledSourceNode).stop(when);
   }
-
-  // eslint-disable-next-line accessor-pairs
-  public set onended(callback: (arg?: number) => void) {
-    const eventCallback = (_event: Event) => {
-      callback();
-    };
-
-    (this.node as globalThis.AudioScheduledSourceNode).onended = eventCallback;
-  }
 }

package/common/cpp/audioapi/core/types/TimeStretchType.h

@@ -1,7 +0,0 @@
-#pragma once
-
-namespace audioapi {
-
-enum class TimeStretchType { LINEAR, SPEECH_MUSIC };
-
-}

package/common/cpp/audioapi/dsp/FFT.cpp

@@ -1,41 +0,0 @@
-#include <audioapi/dsp/FFT.h>
-
-namespace audioapi::dsp {
-
-FFT::FFT(int size) : size_(size) {
-  pffftSetup_ = pffft_new_setup(size_, PFFFT_REAL);
-  work_ = (float *)pffft_aligned_malloc(size_ * sizeof(float));
-}
-
-FFT::~FFT() {
-  pffft_destroy_setup(pffftSetup_);
-  pffft_aligned_free(work_);
-}
-
-void FFT::doFFT(float *in, std::vector<std::complex<float>> &out) {
-  pffft_transform_ordered(
-      pffftSetup_,
-      in,
-      reinterpret_cast<float *>(&out[0]),
-      work_,
-      PFFFT_FORWARD);
-
-  dsp::multiplyByScalar(
-      reinterpret_cast<float *>(&out[0]),
-      0.5f,
-      reinterpret_cast<float *>(&out[0]),
-      size_ * 2);
-}
-
-void FFT::doInverseFFT(std::vector<std::complex<float>> &in, float *out) {
-  pffft_transform_ordered(
-      pffftSetup_,
-      reinterpret_cast<float *>(&in[0]),
-      out,
-      work_,
-      PFFFT_BACKWARD);
-
-  dsp::multiplyByScalar(out, 1.0f / static_cast<float>(size_), out, size_);
-}
-
-} // namespace audioapi::dsp

package/common/cpp/audioapi/dsp/FFT.h

@@ -1,29 +0,0 @@
-#pragma once
-
-#include <audioapi/dsp/VectorMath.h>
-#include <audioapi/libs/pffft/pffft.h>
-
-#include <algorithm>
-#include <cmath>
-#include <utility>
-#include <complex>
-#include <vector>
-
-namespace audioapi::dsp {
-
-class FFT {
- public:
-  explicit FFT(int size);
-  ~FFT();
-
-  void doFFT(float *in, std::vector<std::complex<float>> &out);
-  void doInverseFFT(std::vector<std::complex<float>> &in, float *out);
-
- private:
-  int size_;
-
-  PFFFT_Setup *pffftSetup_;
-  float *work_;
-};
-
-} // namespace audioapi::dsp

package/common/cpp/audioapi/dsp/Windows.cpp

@@ -1,80 +0,0 @@
-#include <audioapi/core/Constants.h>
-#include <audioapi/dsp/Windows.h>
-
-namespace audioapi::dsp {
-
-void WindowFunction::forcePerfectReconstruction(
-    float *data,
-    int windowLength,
-    int interval) {
-  for (int i = 0; i < interval; ++i) {
-    float sum2 = 0;
-
-    for (int index = i; index < windowLength; index += interval) {
-      sum2 += data[index] * data[index];
-    }
-
-    float factor = 1 / std::sqrt(sum2);
-
-    for (int index = i; index < windowLength; index += interval) {
-      data[index] *= factor;
-    }
-  }
-}
-
-void Hann::apply(float *data, int size) const {
-  for (int i = 0; i < size; ++i) {
-    auto x = static_cast<float>(i) / static_cast<float>(size - 1);
-    auto window = 0.5f - 0.5f * cos(2 * PI * x);
-    data[i] = window * amplitude_;
-  }
-}
-
-void Blackman::apply(float *data, int size) const {
-  for (int i = 0; i < size; ++i) {
-    auto x = static_cast<float>(i) / static_cast<float>(size - 1);
-    auto window = 0.42f - 0.5f * cos(2 * PI * x) + 0.08f * cos(4 * PI * x);
-    data[i] = window * amplitude_;
-  }
-}
-
-void Kaiser::apply(float *data, int size) const {
-  for (int i = 0; i < size; ++i) {
-    auto r = static_cast<float>(2 * i + 1) / static_cast<float>(size) - 1.0f;
-    auto arg = std::sqrt(1 - r * r);
-    data[i] = bessel0(beta_ * arg) * invB0_ * amplitude_;
-  }
-}
-
-float Kaiser::bandwidthToBeta(float bandwidth, bool heuristicOptimal) {
-  if (heuristicOptimal) { // Heuristic based on numerical search
-    return bandwidth + 8.0f / (bandwidth + 3.0f) * (bandwidth + 3.0f) +
-        0.25f * std::max(3.0f - bandwidth, 0.0f);
-  }
-
-  bandwidth = std::max(bandwidth, 2.0f);
-  auto alpha = std::sqrt(bandwidth * bandwidth * 0.25f - 1.0f);
-  return alpha * PI;
-}
-
-void ApproximateConfinedGaussian::apply(float *data, int size) const {
-  auto offsetScale =
-      getGaussian(1.0f) / (getGaussian(3.0f) + getGaussian(-1.0f));
-  auto norm = 1 / (getGaussian(1.0f) - 2 * offsetScale * getGaussian(2.0f));
-  for (int i = 0; i < size; ++i) {
-    auto r = static_cast<float>(2 * i + 1) / static_cast<float>(size) - 1.0f;
-    data[i] = norm *
-        (getGaussian(r) -
-         offsetScale * (getGaussian(r - 2) + getGaussian(r + 2)));
-  }
-}
-
-float ApproximateConfinedGaussian::bandwidthToSigma(float bandwidth) {
-  return 0.3f / std::sqrt(bandwidth);
-}
-
-float ApproximateConfinedGaussian::getGaussian(float x) const {
-  return std::exp(-x * x * gaussianFactor_);
-}
-
-} // namespace audioapi::dsp

package/common/cpp/audioapi/dsp/Windows.h

@@ -1,95 +0,0 @@
-#pragma once
-
-#include <cmath>
-#include <algorithm>
-
-namespace audioapi::dsp {
-
-// https://en.wikipedia.org/wiki/Window_function
-// https://personalpages.hs-kempten.de/~vollratj/InEl/pdf/Window%20function%20-%20Wikipedia.pdf
-class WindowFunction {
- public:
-  explicit WindowFunction(float amplitude = 1.0f): amplitude_(amplitude) {}
-
-  virtual void apply(float *data, int size) const = 0;
-  // forces STFT perfect-reconstruction (WOLA) on an existing window, for a given STFT interval.
-  static void forcePerfectReconstruction(float *data, int windowLength, int interval);
-
- protected:
-  // 1/L = amplitude
-  float amplitude_;
-};
-
-//https://en.wikipedia.org/wiki/Hann_function
-// https://www.sciencedirect.com/topics/engineering/hanning-window
-// https://docs.scipy.org/doc//scipy-1.2.3/reference/generated/scipy.signal.windows.hann.html#scipy.signal.windows.hann
-class Hann: public WindowFunction {
- public:
-  explicit Hann(float amplitude = 1.0f): WindowFunction(amplitude) {}
-
-  void apply(float *data, int size) const override;
-};
-
-// https://www.sciencedirect.com/topics/engineering/blackman-window
-// https://docs.scipy.org/doc//scipy-1.2.3/reference/generated/scipy.signal.windows.blackman.html#scipy.signal.windows.blackman
-class Blackman: public WindowFunction {
- public:
-  explicit Blackman(float amplitude = 1.0f): WindowFunction(amplitude) {}
-
-  void apply(float *data, int size) const override;
-};
-
-
-// https://en.wikipedia.org/wiki/Kaiser_window
-class Kaiser: public WindowFunction {
- public:
-  explicit Kaiser(float beta, float amplitude = 1.0f): WindowFunction(amplitude), beta_(beta), invB0_(1.0f / bessel0(beta)) {}
-
-  static Kaiser withBandwidth(float bandwidth, bool heuristicOptimal = false, float amplitude = 1.0f) {
-    return Kaiser(bandwidthToBeta(bandwidth, heuristicOptimal), amplitude);
-  }
-
-  void apply(float *data, int size) const override;
-
- private:
-  // beta = pi * alpha
-  // invB0 = 1 / I0(beta)
-  float beta_;
-  float invB0_;
-
-  // https://en.wikipedia.org/wiki/Bessel_function#Modified_Bessel_functions:_I%CE%B1,_K%CE%B1
-  static inline float bessel0(float x) {
-      const double significanceLimit = 1e-4;
-      auto result = 0.0f;
-      auto term = 1.0f;
-      auto m = 1.0f;
-      while (term > significanceLimit) {
-          result += term;
-          ++m;
-          term *= (x * x) / (4 * m * m);
-      }
-
-      return result;
-  }
-  static float bandwidthToBeta(float bandwidth, bool heuristicOptimal = false);
-};
-
-// https://www.recordingblogs.com/wiki/gaussian-window
-class ApproximateConfinedGaussian: public WindowFunction {
- public:
-  explicit ApproximateConfinedGaussian(float sigma, float amplitude = 1.0f): WindowFunction(amplitude), gaussianFactor_(0.0625f/(sigma*sigma)) {}
-
-  static ApproximateConfinedGaussian withBandwidth(float bandwidth, float amplitude = 1.0f) {
-    return ApproximateConfinedGaussian(bandwidthToSigma(bandwidth), amplitude);
-  }
-
-  void apply(float *data, int size) const override;
-
- private:
-  float gaussianFactor_;
-
-  static float bandwidthToSigma(float bandwidth);
-
-  [[nodiscard]] float getGaussian(float x) const;
-};
-} // namespace audioapi::dsp