npm - @siteed/expo-audio-stream - Versions diffs - 1.1.6 → 1.1.8 - Mend

@siteed/expo-audio-stream 1.1.6 → 1.1.8

This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.

Files changed (16) hide show

package/CHANGELOG.md +40 -0
package/build/ExpoAudioStream.web.d.ts.map +1 -1
package/build/ExpoAudioStream.web.js +1 -0
package/build/ExpoAudioStream.web.js.map +1 -1
package/build/WebRecorder.web.d.ts.map +1 -1
package/build/WebRecorder.web.js +13 -2
package/build/WebRecorder.web.js.map +1 -1
package/build/utils/writeWavHeader.d.ts +43 -3
package/build/utils/writeWavHeader.d.ts.map +1 -1
package/build/utils/writeWavHeader.js +65 -19
package/build/utils/writeWavHeader.js.map +1 -1
package/package.json +4 -5
package/publish.sh +4 -1
package/src/ExpoAudioStream.web.ts +3 -2
package/src/WebRecorder.web.ts +18 -2
package/src/utils/writeWavHeader.ts +77 -24

package/CHANGELOG.md ADDED Viewed

@@ -0,0 +1,40 @@
+# Changelog
+All notable changes to this project will be documented in this file.
+The format is based on [Keep a Changelog](https://keepachangelog.com/en/1.0.0/),
+and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0.html).
+## [Unreleased]
+### Added
+- For new features.
+### Changed
+- For changes in existing functionality.
+### Deprecated
+- For soon-to-be removed features.
+### Removed
+- For now removed features.
+### Fixed
+- For any bug fixes.
+### Security
+- In case of vulnerabilities.
+## [1.0.0] - YYYY-MM-DD
+### Added
+- Initial release of @siteed/expo-audio-stream.
+- Feature: Real-time audio streaming across iOS, Android, and web.
+- Feature: Configurable intervals for audio buffer receipt.
+- Feature: Automated microphone permissions setup in managed Expo projects.
+- Feature: Background audio recording on iOS.
+- Feature: Audio features extraction during recording.
+- Feature: Consistent WAV PCM recording format across all platforms.
+[Unreleased]: https://github.com/deeeed/expo-audio-stream/compare/v1.0.0...HEAD
+[1.0.0]: https://github.com/deeeed/expo-audio-stream/releases/tag/v1.0.0

package/build/ExpoAudioStream.web.d.ts.map CHANGED Viewed

	@@ -1 +1 @@
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package/build/ExpoAudioStream.web.js CHANGED Viewed

@@ -139,6 +139,7 @@ export class ExpoAudioStreamWeb extends EventEmitter {
         logger.debug(`Stopped recording`, fullPcmBufferArray);
         this.isRecording = false;
         this.currentDurationMs = Date.now() - this.recordingStartTime;
+        // Rewrite wav header with correct data size
         const wavConfig = {
             buffer: fullPcmBufferArray.buffer,
             sampleRate: this.recordingConfig?.sampleRate ?? 44100,

package/build/ExpoAudioStream.web.js.map CHANGED Viewed

	@@ -1 +1 @@
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src/ExpoAudioStreamModule.web.ts\nimport { EventEmitter } from 'expo-modules-core'\n\nimport { AudioAnalysis } from './AudioAnalysis/AudioAnalysis.types'\nimport {\n AudioRecording,\n AudioStreamStatus,\n BitDepth,\n RecordingConfig,\n StartRecordingResult,\n} from './ExpoAudioStream.types'\nimport { WebRecorder } from './WebRecorder.web'\nimport { AudioEventPayload } from './events'\nimport { getLogger } from './logger'\nimport { encodingToBitDepth } from './utils/encodingToBitDepth'\nimport { writeWavHeader } from './utils/writeWavHeader'\n\nexport interface EmitAudioEventProps {\n data: Float32Array\n position: number\n}\nexport type EmitAudioEventFunction = (_: EmitAudioEventProps) => void\nexport type EmitAudioAnalysisFunction = (_: AudioAnalysis) => void\n\nexport interface ExpoAudioStreamWebProps {\n audioWorkletUrl: string\n featuresExtratorUrl: string\n}\n\nconst logger = getLogger('ExpoAudioStreamWeb')\n\nexport class ExpoAudioStreamWeb extends EventEmitter {\n customRecorder: WebRecorder \| null\n audioChunks: ArrayBuffer[]\n isRecording: boolean\n isPaused: boolean\n recordingStartTime: number\n pausedTime: number\n currentDurationMs: number\n currentSize: number\n currentInterval: number\n lastEmittedSize: number\n lastEmittedTime: number\n streamUuid: string \| null\n extension: 'webm' \| 'wav' = 'wav' // Default extension is 'webm'\n recordingConfig?: RecordingConfig\n bitDepth: BitDepth // Bit depth of the audio\n audioWorkletUrl: string\n featuresExtratorUrl: string\n\n constructor({\n audioWorkletUrl,\n featuresExtratorUrl,\n }: ExpoAudioStreamWebProps) {\n const mockNativeModule = {\n addListener: () => {\n // Not used on web\n },\n removeListeners: () => {\n // Not used on web\n },\n }\n super(mockNativeModule) // Pass the mock native module to the parent class\n\n this.customRecorder = null\n this.audioChunks = []\n this.isRecording = false\n this.isPaused = false\n this.recordingStartTime = 0\n this.pausedTime = 0\n this.currentDurationMs = 0\n this.currentSize = 0\n this.bitDepth = 32 // Default\n this.currentInterval = 1000 // Default interval in ms\n this.lastEmittedSize = 0\n this.lastEmittedTime = 0\n this.streamUuid = null // Initialize UUID on first recording start\n this.audioWorkletUrl = audioWorkletUrl\n this.featuresExtratorUrl = featuresExtratorUrl\n }\n\n // Utility to handle user media stream\n async getMediaStream() {\n try {\n return await navigator.mediaDevices.getUserMedia({ audio: true })\n } catch (error) {\n console.error('Failed to get media stream:', error)\n throw error\n }\n }\n\n // Start recording with options\n async startRecording(recordingConfig: RecordingConfig = {}) {\n if (this.isRecording) {\n throw new Error('Recording is already in progress')\n }\n\n this.bitDepth = encodingToBitDepth({\n encoding: recordingConfig.encoding ?? 'pcm_32bit',\n })\n\n const audioContext = new (window.AudioContext \|\|\n // eslint-disable-next-line @typescript-eslint/ban-ts-comment\n // @ts-ignore - Allow webkitAudioContext for Safari\n window.webkitAudioContext)()\n const stream = await this.getMediaStream()\n\n const source = audioContext.createMediaStreamSource(stream)\n\n this.customRecorder = new WebRecorder({\n audioContext,\n source,\n recordingConfig,\n audioWorkletUrl: this.audioWorkletUrl,\n emitAudioEventCallback: ({\n data,\n position,\n }: EmitAudioEventProps) => {\n this.audioChunks.push(data)\n this.currentSize += data.byteLength\n this.emitAudioEvent({ data, position })\n this.lastEmittedTime = Date.now()\n this.lastEmittedSize = this.currentSize\n },\n emitAudioAnalysisCallback: (audioAnalysisData: AudioAnalysis) => {\n logger.log(`Emitted AudioAnalysis:`, audioAnalysisData)\n this.emit('AudioAnalysis', audioAnalysisData)\n },\n })\n await this.customRecorder.init()\n this.customRecorder.start()\n\n // // Set a timer to stop recording after 5 seconds\n // setTimeout(() => {\n // logger.log(\"AUTO Stopping recording\");\n // this.customRecorder?.stopAndPlay();\n // this.isRecording = false;\n // }, 3000);\n\n this.isRecording = true\n this.recordingConfig = recordingConfig\n this.recordingStartTime = Date.now()\n this.pausedTime = 0\n this.lastEmittedSize = 0\n this.lastEmittedTime = 0\n this.streamUuid = Date.now().toString()\n const fileUri = `${this.streamUuid}.${this.extension}`\n const streamConfig: StartRecordingResult = {\n fileUri,\n mimeType: `audio/${this.extension}`,\n bitDepth: this.bitDepth,\n channels: recordingConfig.channels ?? 1,\n sampleRate: recordingConfig.sampleRate ?? 44100,\n }\n return streamConfig\n }\n\n emitAudioEvent({ data, position }: EmitAudioEventProps) {\n const fileUri = `${this.streamUuid}.${this.extension}`\n const audioEventPayload: AudioEventPayload = {\n fileUri,\n mimeType: `audio/${this.extension}`,\n lastEmittedSize: this.lastEmittedSize, // Since this might be continuously streaming, adjust accordingly\n deltaSize: data.byteLength,\n position,\n totalSize: this.currentSize,\n buffer: data,\n streamUuid: this.streamUuid ?? '', // Generate or manage UUID for stream identification\n }\n\n this.emit('AudioData', audioEventPayload)\n }\n\n // Stop recording\n async stopRecording(): Promise<AudioRecording> {\n if (!this.customRecorder) {\n throw new Error('Recorder is not initialized')\n }\n\n const fullPcmBufferArray = await this.customRecorder.stop()\n\n // concat all audio chunks\n logger.debug(`Stopped recording`, fullPcmBufferArray)\n this.isRecording = false\n this.currentDurationMs = Date.now() - this.recordingStartTime\n\n const wavConfig = {\n buffer: fullPcmBufferArray.buffer,\n sampleRate: this.recordingConfig?.sampleRate ?? 44100,\n numChannels: this.recordingConfig?.channels ?? 1,\n bitDepth: this.bitDepth,\n }\n logger.debug(`Writing wav header`, wavConfig)\n const wavBuffer = writeWavHeader(wavConfig).slice(0)\n\n // Create blob fileUri from audio chunks\n const blob = new Blob([wavBuffer], {\n type: `audio/${this.extension}`,\n })\n const fileUri = URL.createObjectURL(blob)\n\n const result: AudioRecording = {\n fileUri,\n filename: `${this.streamUuid}.${this.extension}`,\n wavPCMData: fullPcmBufferArray,\n bitDepth: this.bitDepth,\n channels: this.recordingConfig?.channels ?? 1,\n sampleRate: this.recordingConfig?.sampleRate ?? 44100,\n durationMs: this.currentDurationMs,\n size: this.currentSize,\n mimeType: `audio/${this.extension}`,\n }\n\n return result\n }\n\n // Pause recording\n async pauseRecording() {\n if (!this.isRecording \|\| this.isPaused) {\n throw new Error('Recording is not active or already paused')\n }\n\n if (this.customRecorder) {\n this.customRecorder.pause()\n }\n this.isPaused = true\n this.pausedTime = Date.now()\n }\n\n // Resume recording\n async resumeRecording() {\n if (!this.isPaused) {\n throw new Error('Recording is not paused')\n }\n\n if (this.customRecorder) {\n this.customRecorder.resume()\n }\n this.isPaused = false\n this.recordingStartTime += Date.now() - this.pausedTime\n }\n\n // Get current status\n status() {\n const status: AudioStreamStatus = {\n isRecording: this.isRecording,\n isPaused: this.isPaused,\n durationMs: Date.now() - this.recordingStartTime,\n size: this.currentSize,\n interval: this.currentInterval,\n mimeType: `audio/${this.extension}`,\n }\n return status\n }\n}\n"]}
1	+ 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src/ExpoAudioStreamModule.web.ts\nimport { EventEmitter } from 'expo-modules-core'\n\nimport { AudioAnalysis } from './AudioAnalysis/AudioAnalysis.types'\nimport {\n AudioRecording,\n AudioStreamStatus,\n BitDepth,\n RecordingConfig,\n StartRecordingResult,\n} from './ExpoAudioStream.types'\nimport { WebRecorder } from './WebRecorder.web'\nimport { AudioEventPayload } from './events'\nimport { getLogger } from './logger'\nimport { encodingToBitDepth } from './utils/encodingToBitDepth'\nimport { WavHeaderOptions, writeWavHeader } from './utils/writeWavHeader'\n\nexport interface EmitAudioEventProps {\n data: Float32Array\n position: number\n}\nexport type EmitAudioEventFunction = (_: EmitAudioEventProps) => void\nexport type EmitAudioAnalysisFunction = (_: AudioAnalysis) => void\n\nexport interface ExpoAudioStreamWebProps {\n audioWorkletUrl: string\n featuresExtratorUrl: string\n}\n\nconst logger = getLogger('ExpoAudioStreamWeb')\n\nexport class ExpoAudioStreamWeb extends EventEmitter {\n customRecorder: WebRecorder \| null\n audioChunks: ArrayBuffer[]\n isRecording: boolean\n isPaused: boolean\n recordingStartTime: number\n pausedTime: number\n currentDurationMs: number\n currentSize: number\n currentInterval: number\n lastEmittedSize: number\n lastEmittedTime: number\n streamUuid: string \| null\n extension: 'webm' \| 'wav' = 'wav' // Default extension is 'webm'\n recordingConfig?: RecordingConfig\n bitDepth: BitDepth // Bit depth of the audio\n audioWorkletUrl: string\n featuresExtratorUrl: string\n\n constructor({\n audioWorkletUrl,\n featuresExtratorUrl,\n }: ExpoAudioStreamWebProps) {\n const mockNativeModule = {\n addListener: () => {\n // Not used on web\n },\n removeListeners: () => {\n // Not used on web\n },\n }\n super(mockNativeModule) // Pass the mock native module to the parent class\n\n this.customRecorder = null\n this.audioChunks = []\n this.isRecording = false\n this.isPaused = false\n this.recordingStartTime = 0\n this.pausedTime = 0\n this.currentDurationMs = 0\n this.currentSize = 0\n this.bitDepth = 32 // Default\n this.currentInterval = 1000 // Default interval in ms\n this.lastEmittedSize = 0\n this.lastEmittedTime = 0\n this.streamUuid = null // Initialize UUID on first recording start\n this.audioWorkletUrl = audioWorkletUrl\n this.featuresExtratorUrl = featuresExtratorUrl\n }\n\n // Utility to handle user media stream\n async getMediaStream() {\n try {\n return await navigator.mediaDevices.getUserMedia({ audio: true })\n } catch (error) {\n console.error('Failed to get media stream:', error)\n throw error\n }\n }\n\n // Start recording with options\n async startRecording(recordingConfig: RecordingConfig = {}) {\n if (this.isRecording) {\n throw new Error('Recording is already in progress')\n }\n\n this.bitDepth = encodingToBitDepth({\n encoding: recordingConfig.encoding ?? 'pcm_32bit',\n })\n\n const audioContext = new (window.AudioContext \|\|\n // eslint-disable-next-line @typescript-eslint/ban-ts-comment\n // @ts-ignore - Allow webkitAudioContext for Safari\n window.webkitAudioContext)()\n const stream = await this.getMediaStream()\n\n const source = audioContext.createMediaStreamSource(stream)\n\n this.customRecorder = new WebRecorder({\n audioContext,\n source,\n recordingConfig,\n audioWorkletUrl: this.audioWorkletUrl,\n emitAudioEventCallback: ({\n data,\n position,\n }: EmitAudioEventProps) => {\n this.audioChunks.push(data)\n this.currentSize += data.byteLength\n this.emitAudioEvent({ data, position })\n this.lastEmittedTime = Date.now()\n this.lastEmittedSize = this.currentSize\n },\n emitAudioAnalysisCallback: (audioAnalysisData: AudioAnalysis) => {\n logger.log(`Emitted AudioAnalysis:`, audioAnalysisData)\n this.emit('AudioAnalysis', audioAnalysisData)\n },\n })\n await this.customRecorder.init()\n this.customRecorder.start()\n\n // // Set a timer to stop recording after 5 seconds\n // setTimeout(() => {\n // logger.log(\"AUTO Stopping recording\");\n // this.customRecorder?.stopAndPlay();\n // this.isRecording = false;\n // }, 3000);\n\n this.isRecording = true\n this.recordingConfig = recordingConfig\n this.recordingStartTime = Date.now()\n this.pausedTime = 0\n this.lastEmittedSize = 0\n this.lastEmittedTime = 0\n this.streamUuid = Date.now().toString()\n const fileUri = `${this.streamUuid}.${this.extension}`\n const streamConfig: StartRecordingResult = {\n fileUri,\n mimeType: `audio/${this.extension}`,\n bitDepth: this.bitDepth,\n channels: recordingConfig.channels ?? 1,\n sampleRate: recordingConfig.sampleRate ?? 44100,\n }\n return streamConfig\n }\n\n emitAudioEvent({ data, position }: EmitAudioEventProps) {\n const fileUri = `${this.streamUuid}.${this.extension}`\n const audioEventPayload: AudioEventPayload = {\n fileUri,\n mimeType: `audio/${this.extension}`,\n lastEmittedSize: this.lastEmittedSize, // Since this might be continuously streaming, adjust accordingly\n deltaSize: data.byteLength,\n position,\n totalSize: this.currentSize,\n buffer: data,\n streamUuid: this.streamUuid ?? '', // Generate or manage UUID for stream identification\n }\n\n this.emit('AudioData', audioEventPayload)\n }\n\n // Stop recording\n async stopRecording(): Promise<AudioRecording> {\n if (!this.customRecorder) {\n throw new Error('Recorder is not initialized')\n }\n\n const fullPcmBufferArray = await this.customRecorder.stop()\n\n // concat all audio chunks\n logger.debug(`Stopped recording`, fullPcmBufferArray)\n this.isRecording = false\n this.currentDurationMs = Date.now() - this.recordingStartTime\n\n // Rewrite wav header with correct data size\n const wavConfig: WavHeaderOptions = {\n buffer: fullPcmBufferArray.buffer,\n sampleRate: this.recordingConfig?.sampleRate ?? 44100,\n numChannels: this.recordingConfig?.channels ?? 1,\n bitDepth: this.bitDepth,\n }\n logger.debug(`Writing wav header`, wavConfig)\n const wavBuffer = writeWavHeader(wavConfig).slice(0)\n\n // Create blob fileUri from audio chunks\n const blob = new Blob([wavBuffer], {\n type: `audio/${this.extension}`,\n })\n const fileUri = URL.createObjectURL(blob)\n\n const result: AudioRecording = {\n fileUri,\n filename: `${this.streamUuid}.${this.extension}`,\n wavPCMData: fullPcmBufferArray,\n bitDepth: this.bitDepth,\n channels: this.recordingConfig?.channels ?? 1,\n sampleRate: this.recordingConfig?.sampleRate ?? 44100,\n durationMs: this.currentDurationMs,\n size: this.currentSize,\n mimeType: `audio/${this.extension}`,\n }\n\n return result\n }\n\n // Pause recording\n async pauseRecording() {\n if (!this.isRecording \|\| this.isPaused) {\n throw new Error('Recording is not active or already paused')\n }\n\n if (this.customRecorder) {\n this.customRecorder.pause()\n }\n this.isPaused = true\n this.pausedTime = Date.now()\n }\n\n // Resume recording\n async resumeRecording() {\n if (!this.isPaused) {\n throw new Error('Recording is not paused')\n }\n\n if (this.customRecorder) {\n this.customRecorder.resume()\n }\n this.isPaused = false\n this.recordingStartTime += Date.now() - this.pausedTime\n }\n\n // Get current status\n status() {\n const status: AudioStreamStatus = {\n isRecording: this.isRecording,\n isPaused: this.isPaused,\n durationMs: Date.now() - this.recordingStartTime,\n size: this.currentSize,\n interval: this.currentInterval,\n mimeType: `audio/${this.extension}`,\n }\n return status\n }\n}\n"]}

package/build/WebRecorder.web.d.ts.map CHANGED Viewed

	@@ -1 +1 @@
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package/build/WebRecorder.web.js CHANGED Viewed

@@ -1,5 +1,6 @@
 import { getLogger } from './logger';
 import { encodingToBitDepth } from './utils/encodingToBitDepth';
+import { writeWavHeader } from './utils/writeWavHeader';
 import { InlineFeaturesExtractor } from './workers/InlineFeaturesExtractor.web';
 import { InlineAudioWebWorker } from './workers/inlineAudioWebWorker.web';
 const DEFAULT_WEB_BITDEPTH = 32;
@@ -33,8 +34,6 @@ export class WebRecorder {
         this.emitAudioAnalysisCallback = emitAudioAnalysisCallback;
         this.config = recordingConfig;
         this.position = 0;
-        this.bufferSize = 0;
-        this.buffer = new Float32Array(0); // Initialize the buffer
         const audioContextFormat = this.checkAudioContextFormat({
             sampleRate: this.audioContext.sampleRate,
         });
@@ -53,6 +52,18 @@ export class WebRecorder {
             }) ||
                 audioContextFormat.bitDepth ||
                 DEFAULT_WEB_BITDEPTH;
+        // Initialize WAV header
+        const wavHeader = writeWavHeader({
+            sampleRate: this.audioContext.sampleRate,
+            numChannels: this.numberOfChannels,
+            bitDepth: this.exportBitDepth,
+        });
+        logger.debug(`WAV header length: ${wavHeader.byteLength}`, wavHeader);
+        // Initialize the buffer with WAV header
+        this.buffer = new Float32Array(wavHeader.byteLength / Float32Array.BYTES_PER_ELEMENT);
+        this.bufferSize = this.buffer.byteLength;
+        // Copy WAV header to Float32Array buffer
+        new Uint8Array(this.buffer.buffer).set(new Uint8Array(wavHeader));
         this.audioAnalysisData = {
             amplitudeRange: { min: 0, max: 0 },
             dataPoints: [],

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src/WebRecorder.ts\nimport { AudioAnalysis } from './AudioAnalysis/AudioAnalysis.types'\nimport { RecordingConfig } from './ExpoAudioStream.types'\nimport {\n EmitAudioAnalysisFunction,\n EmitAudioEventFunction,\n} from './ExpoAudioStream.web'\nimport { getLogger } from './logger'\nimport { encodingToBitDepth } from './utils/encodingToBitDepth'\nimport { InlineFeaturesExtractor } from './workers/InlineFeaturesExtractor.web'\nimport { InlineAudioWebWorker } from './workers/inlineAudioWebWorker.web'\n\ninterface AudioWorkletEvent {\n data: {\n command: string\n recordedData?: Float32Array\n sampleRate?: number\n }\n}\n\ninterface AudioFeaturesEvent {\n data: {\n command: string\n result: AudioAnalysis\n }\n}\n\nconst DEFAULT_WEB_BITDEPTH = 32\nconst DEFAULT_WEB_POINTS_PER_SECOND = 10\nconst DEFAULT_WEB_INTERVAL = 500\nconst DEFAULT_WEB_NUMBER_OF_CHANNELS = 1\nconst DEFAULT_ALGORITHM = 'rms'\n\nconst TAG = 'WebRecorder'\nconst logger = getLogger(TAG)\n\nexport class WebRecorder {\n private audioContext: AudioContext\n private audioWorkletNode!: AudioWorkletNode\n private featureExtractorWorker?: Worker\n private source: MediaStreamAudioSourceNode\n private audioWorkletUrl: string\n private emitAudioEventCallback: EmitAudioEventFunction\n private emitAudioAnalysisCallback: EmitAudioAnalysisFunction\n private config: RecordingConfig\n private position: number // Track the cumulative position\n private numberOfChannels: number // Number of audio channels\n private bitDepth: number // Bit depth of the audio\n private exportBitDepth: number // Bit depth of the audio\n private buffer: Float32Array // Single buffer to store the audio data\n private bufferSize: number // Keep track of the buffer size\n private audioAnalysisData: AudioAnalysis // Keep updating the full audio analysis data with latest events\n\n constructor({\n audioContext,\n source,\n recordingConfig,\n audioWorkletUrl,\n emitAudioEventCallback,\n emitAudioAnalysisCallback,\n }: {\n audioContext: AudioContext\n source: MediaStreamAudioSourceNode\n recordingConfig: RecordingConfig\n audioWorkletUrl: string\n emitAudioEventCallback: EmitAudioEventFunction\n emitAudioAnalysisCallback: EmitAudioAnalysisFunction\n }) {\n this.audioContext = audioContext\n this.source = source\n this.audioWorkletUrl = audioWorkletUrl\n this.emitAudioEventCallback = emitAudioEventCallback\n this.emitAudioAnalysisCallback = emitAudioAnalysisCallback\n this.config = recordingConfig\n this.position = 0\n this.bufferSize = 0\n this.buffer = new Float32Array(0) // Initialize the buffer\n\n const audioContextFormat = this.checkAudioContextFormat({\n sampleRate: this.audioContext.sampleRate,\n })\n logger.debug('Initialized WebRecorder with config:', {\n sampleRate: audioContextFormat.sampleRate,\n bitDepth: audioContextFormat.bitDepth,\n numberOfChannels: audioContextFormat.numberOfChannels,\n })\n\n this.bitDepth = audioContextFormat.bitDepth\n this.numberOfChannels =\n audioContextFormat.numberOfChannels \|\|\n DEFAULT_WEB_NUMBER_OF_CHANNELS // Default to 1 if not available\n this.exportBitDepth =\n encodingToBitDepth({\n encoding: recordingConfig.encoding ?? 'pcm_32bit',\n }) \|\|\n audioContextFormat.bitDepth \|\|\n DEFAULT_WEB_BITDEPTH\n\n this.audioAnalysisData = {\n amplitudeRange: { min: 0, max: 0 },\n dataPoints: [],\n durationMs: 0,\n samples: 0,\n amplitudeAlgorithm: recordingConfig.algorithm \|\| DEFAULT_ALGORITHM,\n bitDepth: this.bitDepth,\n numberOfChannels: this.numberOfChannels,\n sampleRate: this.config.sampleRate \|\| this.audioContext.sampleRate,\n pointsPerSecond:\n this.config.pointsPerSecond \|\| DEFAULT_WEB_POINTS_PER_SECOND,\n speakerChanges: [],\n }\n\n if (recordingConfig.enableProcessing) {\n this.initFeatureExtractorWorker()\n }\n }\n\n async init() {\n try {\n if (!this.audioWorkletUrl) {\n const blob = new Blob([InlineAudioWebWorker], {\n type: 'application/javascript',\n })\n const url = URL.createObjectURL(blob)\n await this.audioContext.audioWorklet.addModule(url)\n } else {\n await this.audioContext.audioWorklet.addModule(\n this.audioWorkletUrl\n )\n }\n this.audioWorkletNode = new AudioWorkletNode(\n this.audioContext,\n 'recorder-processor'\n )\n\n this.audioWorkletNode.port.onmessage = async (\n event: AudioWorkletEvent\n ) => {\n const command = event.data.command\n if (command !== 'newData') {\n return\n }\n // Handle the audio blob (e.g., send it to the server or process it further)\n logger.debug('Received audio blob from processor', event)\n const pcmBufferFloat = event.data.recordedData\n\n if (!pcmBufferFloat) {\n return\n }\n\n // Concatenate the incoming Float32Array to the existing buffer\n const newBuffer = new Float32Array(\n this.bufferSize + pcmBufferFloat.length\n )\n newBuffer.set(this.buffer, 0)\n newBuffer.set(pcmBufferFloat, this.bufferSize)\n this.buffer = newBuffer\n this.bufferSize += pcmBufferFloat.length\n\n const sampleRate =\n event.data.sampleRate ?? this.audioContext.sampleRate\n const duration = pcmBufferFloat.length / sampleRate // Calculate duration of the current buffer\n\n this.emitAudioEventCallback({\n data: pcmBufferFloat,\n position: this.position,\n })\n this.position += duration // Update position\n\n this.featureExtractorWorker?.postMessage(\n {\n command: 'process',\n channelData: pcmBufferFloat,\n sampleRate,\n pointsPerSecond:\n this.config.pointsPerSecond \|\|\n DEFAULT_WEB_POINTS_PER_SECOND,\n algorithm: this.config.algorithm \|\| 'rms',\n bitDepth: this.bitDepth,\n fullAudioDurationMs: this.position * 1000,\n numberOfChannels: this.numberOfChannels,\n features: this.config.features,\n },\n []\n )\n }\n\n logger.debug(\n `WebRecorder initialized -- recordSampleRate=${this.audioContext.sampleRate}`,\n this.config\n )\n this.audioWorkletNode.port.postMessage({\n command: 'init',\n recordSampleRate: this.audioContext.sampleRate, // Pass the original sample rate\n exportSampleRate:\n this.config.sampleRate ?? this.audioContext.sampleRate,\n bitDepth: this.bitDepth,\n exportBitDepth: this.exportBitDepth,\n channels: this.numberOfChannels,\n interval: this.config.interval ?? DEFAULT_WEB_INTERVAL,\n })\n\n // Connect the source to the AudioWorkletNode and start recording\n this.source.connect(this.audioWorkletNode)\n this.audioWorkletNode.connect(this.audioContext.destination)\n } catch (error) {\n console.error(`[${TAG}] Failed to initialize WebRecorder`, error)\n }\n }\n\n initFeatureExtractorWorker(featuresExtratorUrl?: string) {\n try {\n if (featuresExtratorUrl) {\n // Initialize the feature extractor worker\n //TODO: create audio feature extractor from a Blob instead of url since we cannot include the url directly in the library\n // We keep the url during dev and use the blob in production.\n this.featureExtractorWorker = new Worker(\n new URL(featuresExtratorUrl, window.location.href)\n )\n this.featureExtractorWorker.onmessage =\n this.handleFeatureExtractorMessage.bind(this)\n this.featureExtractorWorker.onerror =\n this.handleWorkerError.bind(this)\n } else {\n // Fallback to the inline worker if the URL is not provided\n this.initFallbackWorker()\n }\n } catch (error) {\n console.error(\n `[${TAG}] Failed to initialize feature extractor worker`,\n error\n )\n this.initFallbackWorker()\n }\n }\n\n initFallbackWorker() {\n try {\n const blob = new Blob([InlineFeaturesExtractor], {\n type: 'application/javascript',\n })\n const url = URL.createObjectURL(blob)\n this.featureExtractorWorker = new Worker(url)\n this.featureExtractorWorker.onmessage =\n this.handleFeatureExtractorMessage.bind(this)\n this.featureExtractorWorker.onerror = (error) => {\n console.error(`[${TAG}] Default Inline worker failed`, error)\n }\n logger.log('Inline worker initialized successfully')\n } catch (error) {\n console.error(\n `[${TAG}] Failed to initialize Inline Feature Extractor worker`,\n error\n )\n }\n }\n\n handleWorkerError(error: ErrorEvent) {\n console.error(`[${TAG}] Feature extractor worker error:`, error)\n }\n\n handleFeatureExtractorMessage(event: AudioFeaturesEvent) {\n if (event.data.command === 'features') {\n const segmentResult = event.data.result\n\n // Merge the segment result with the full audio analysis data\n this.audioAnalysisData.dataPoints.push(...segmentResult.dataPoints)\n this.audioAnalysisData.speakerChanges?.push(\n ...(segmentResult.speakerChanges ?? [])\n )\n this.audioAnalysisData.durationMs = segmentResult.durationMs\n if (segmentResult.amplitudeRange) {\n this.audioAnalysisData.amplitudeRange = {\n min: Math.min(\n this.audioAnalysisData.amplitudeRange.min,\n segmentResult.amplitudeRange.min\n ),\n max: Math.max(\n this.audioAnalysisData.amplitudeRange.max,\n segmentResult.amplitudeRange.max\n ),\n }\n }\n // Handle the extracted features (e.g., emit an event or log them)\n logger.debug('features event segmentResult', segmentResult)\n logger.debug(\n 'features event audioAnalysisData',\n this.audioAnalysisData\n )\n this.emitAudioAnalysisCallback(segmentResult)\n }\n }\n\n start() {\n this.source.connect(this.audioWorkletNode)\n this.audioWorkletNode.connect(this.audioContext.destination)\n }\n\n stop(): Promise<Float32Array> {\n return new Promise((resolve, reject) => {\n try {\n if (this.audioWorkletNode) {\n // this.source.disconnect(this.audioWorkletNode);\n // this.audioWorkletNode.disconnect(this.audioContext.destination);\n this.audioWorkletNode.port.postMessage({ command: 'stop' })\n\n // Set a timeout to reject the promise if no message is received within 5 seconds\n const timeout = setTimeout(() => {\n this.audioWorkletNode.port.removeEventListener(\n 'message',\n onMessage\n )\n reject(\n new Error(\n \"Timeout error, audioWorkletNode didn't complete.\"\n )\n )\n }, 5000)\n\n // Listen for the recordedData message to confirm stopping\n const onMessage = async (event: AudioWorkletEvent) => {\n const command = event.data.command\n if (command === 'recordedData') {\n clearTimeout(timeout) // Clear the timeout\n\n const rawPCMDataFull =\n event.data.recordedData?.slice(0)\n\n if (!rawPCMDataFull) {\n reject(new Error('Failed to get recorded data'))\n return\n }\n\n // Compute duration of the recorded data\n const duration =\n rawPCMDataFull.byteLength /\n (this.audioContext.sampleRate \n (this.exportBitDepth /\n this.numberOfChannels))\n logger.debug(\n `Received recorded data -- Duration: ${duration} vs ${rawPCMDataFull.byteLength / this.audioContext.sampleRate} seconds`\n )\n logger.debug(\n `recordedData.length=${rawPCMDataFull.byteLength} vs transmittedData.length=${this.bufferSize}`\n )\n\n // Remove the event listener after receiving the final data\n this.audioWorkletNode.port.removeEventListener(\n 'message',\n onMessage\n )\n resolve(this.buffer) // Resolve the promise with the collected buffers\n }\n }\n this.audioWorkletNode.port.addEventListener(\n 'message',\n onMessage\n )\n }\n\n // Stop all media stream tracks to stop the browser recording\n this.stopMediaStreamTracks()\n } catch (error) {\n reject(error)\n }\n })\n }\n\n pause() {\n this.source.disconnect(this.audioWorkletNode) // Disconnect the source from the AudioWorkletNode\n this.audioWorkletNode.disconnect(this.audioContext.destination) // Disconnect the AudioWorkletNode from the destination\n this.audioWorkletNode.port.postMessage({ command: 'pause' })\n }\n\n stopMediaStreamTracks() {\n // Stop all audio tracks to stop the recording icon\n const tracks = this.source.mediaStream.getTracks()\n tracks.forEach((track) => track.stop())\n }\n\n async playRecordedData({\n recordedData,\n }: {\n recordedData: ArrayBuffer\n mimeType?: string\n }) {\n try {\n const blob = new Blob([recordedData])\n const url = URL.createObjectURL(blob)\n const response = await fetch(url)\n const arrayBuffer = await response.arrayBuffer()\n\n // Decode the audio data\n const audioBuffer =\n await this.audioContext.decodeAudioData(arrayBuffer)\n\n // Create a buffer source node and play the audio\n const bufferSource = this.audioContext.createBufferSource()\n bufferSource.buffer = audioBuffer\n bufferSource.connect(this.audioContext.destination)\n bufferSource.start()\n logger.debug('Playing recorded data', recordedData)\n } catch (error) {\n console.error(`[${TAG}] Failed to play recorded data:`, error)\n }\n }\n\n private checkAudioContextFormat({ sampleRate }: { sampleRate: number }) {\n // Create a silent AudioBuffer\n const frameCount = sampleRate 1.0 // 1 second buffer\n const audioBuffer = this.audioContext.createBuffer(\n 1,\n frameCount,\n sampleRate\n )\n\n // Check the format\n const channelData = audioBuffer.getChannelData(0)\n const bitDepth = channelData.BYTES_PER_ELEMENT * 8 // 4 bytes per element means 32-bit\n\n return {\n sampleRate: audioBuffer.sampleRate,\n bitDepth,\n numberOfChannels: audioBuffer.numberOfChannels,\n }\n }\n\n resume() {\n this.source.connect(this.audioWorkletNode)\n this.audioWorkletNode.connect(this.audioContext.destination)\n this.audioWorkletNode.port.postMessage({ command: 'resume' })\n }\n}\n"]}
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src/WebRecorder.ts\nimport { AudioAnalysis } from './AudioAnalysis/AudioAnalysis.types'\nimport { RecordingConfig } from './ExpoAudioStream.types'\nimport {\n EmitAudioAnalysisFunction,\n EmitAudioEventFunction,\n} from './ExpoAudioStream.web'\nimport { getLogger } from './logger'\nimport { encodingToBitDepth } from './utils/encodingToBitDepth'\nimport { writeWavHeader } from './utils/writeWavHeader'\nimport { InlineFeaturesExtractor } from './workers/InlineFeaturesExtractor.web'\nimport { InlineAudioWebWorker } from './workers/inlineAudioWebWorker.web'\n\ninterface AudioWorkletEvent {\n data: {\n command: string\n recordedData?: Float32Array\n sampleRate?: number\n }\n}\n\ninterface AudioFeaturesEvent {\n data: {\n command: string\n result: AudioAnalysis\n }\n}\n\nconst DEFAULT_WEB_BITDEPTH = 32\nconst DEFAULT_WEB_POINTS_PER_SECOND = 10\nconst DEFAULT_WEB_INTERVAL = 500\nconst DEFAULT_WEB_NUMBER_OF_CHANNELS = 1\nconst DEFAULT_ALGORITHM = 'rms'\n\nconst TAG = 'WebRecorder'\nconst logger = getLogger(TAG)\n\nexport class WebRecorder {\n private audioContext: AudioContext\n private audioWorkletNode!: AudioWorkletNode\n private featureExtractorWorker?: Worker\n private source: MediaStreamAudioSourceNode\n private audioWorkletUrl: string\n private emitAudioEventCallback: EmitAudioEventFunction\n private emitAudioAnalysisCallback: EmitAudioAnalysisFunction\n private config: RecordingConfig\n private position: number // Track the cumulative position\n private numberOfChannels: number // Number of audio channels\n private bitDepth: number // Bit depth of the audio\n private exportBitDepth: number // Bit depth of the audio\n private buffer: Float32Array // Single buffer to store the audio data\n private bufferSize: number // Keep track of the buffer size\n private audioAnalysisData: AudioAnalysis // Keep updating the full audio analysis data with latest events\n\n constructor({\n audioContext,\n source,\n recordingConfig,\n audioWorkletUrl,\n emitAudioEventCallback,\n emitAudioAnalysisCallback,\n }: {\n audioContext: AudioContext\n source: MediaStreamAudioSourceNode\n recordingConfig: RecordingConfig\n audioWorkletUrl: string\n emitAudioEventCallback: EmitAudioEventFunction\n emitAudioAnalysisCallback: EmitAudioAnalysisFunction\n }) {\n this.audioContext = audioContext\n this.source = source\n this.audioWorkletUrl = audioWorkletUrl\n this.emitAudioEventCallback = emitAudioEventCallback\n this.emitAudioAnalysisCallback = emitAudioAnalysisCallback\n this.config = recordingConfig\n this.position = 0\n\n const audioContextFormat = this.checkAudioContextFormat({\n sampleRate: this.audioContext.sampleRate,\n })\n logger.debug('Initialized WebRecorder with config:', {\n sampleRate: audioContextFormat.sampleRate,\n bitDepth: audioContextFormat.bitDepth,\n numberOfChannels: audioContextFormat.numberOfChannels,\n })\n\n this.bitDepth = audioContextFormat.bitDepth\n this.numberOfChannels =\n audioContextFormat.numberOfChannels \|\|\n DEFAULT_WEB_NUMBER_OF_CHANNELS // Default to 1 if not available\n this.exportBitDepth =\n encodingToBitDepth({\n encoding: recordingConfig.encoding ?? 'pcm_32bit',\n }) \|\|\n audioContextFormat.bitDepth \|\|\n DEFAULT_WEB_BITDEPTH\n\n // Initialize WAV header\n const wavHeader = writeWavHeader({\n sampleRate: this.audioContext.sampleRate,\n numChannels: this.numberOfChannels,\n bitDepth: this.exportBitDepth,\n })\n\n logger.debug(`WAV header length: ${wavHeader.byteLength}`, wavHeader)\n // Initialize the buffer with WAV header\n this.buffer = new Float32Array(\n wavHeader.byteLength / Float32Array.BYTES_PER_ELEMENT\n )\n this.bufferSize = this.buffer.byteLength\n\n // Copy WAV header to Float32Array buffer\n new Uint8Array(this.buffer.buffer).set(new Uint8Array(wavHeader))\n\n this.audioAnalysisData = {\n amplitudeRange: { min: 0, max: 0 },\n dataPoints: [],\n durationMs: 0,\n samples: 0,\n amplitudeAlgorithm: recordingConfig.algorithm \|\| DEFAULT_ALGORITHM,\n bitDepth: this.bitDepth,\n numberOfChannels: this.numberOfChannels,\n sampleRate: this.config.sampleRate \|\| this.audioContext.sampleRate,\n pointsPerSecond:\n this.config.pointsPerSecond \|\| DEFAULT_WEB_POINTS_PER_SECOND,\n speakerChanges: [],\n }\n\n if (recordingConfig.enableProcessing) {\n this.initFeatureExtractorWorker()\n }\n }\n\n async init() {\n try {\n if (!this.audioWorkletUrl) {\n const blob = new Blob([InlineAudioWebWorker], {\n type: 'application/javascript',\n })\n const url = URL.createObjectURL(blob)\n await this.audioContext.audioWorklet.addModule(url)\n } else {\n await this.audioContext.audioWorklet.addModule(\n this.audioWorkletUrl\n )\n }\n this.audioWorkletNode = new AudioWorkletNode(\n this.audioContext,\n 'recorder-processor'\n )\n\n this.audioWorkletNode.port.onmessage = async (\n event: AudioWorkletEvent\n ) => {\n const command = event.data.command\n if (command !== 'newData') {\n return\n }\n // Handle the audio blob (e.g., send it to the server or process it further)\n logger.debug('Received audio blob from processor', event)\n const pcmBufferFloat = event.data.recordedData\n\n if (!pcmBufferFloat) {\n return\n }\n\n // Concatenate the incoming Float32Array to the existing buffer\n const newBuffer = new Float32Array(\n this.bufferSize + pcmBufferFloat.length\n )\n newBuffer.set(this.buffer, 0)\n newBuffer.set(pcmBufferFloat, this.bufferSize)\n this.buffer = newBuffer\n this.bufferSize += pcmBufferFloat.length\n\n const sampleRate =\n event.data.sampleRate ?? this.audioContext.sampleRate\n const duration = pcmBufferFloat.length / sampleRate // Calculate duration of the current buffer\n\n this.emitAudioEventCallback({\n data: pcmBufferFloat,\n position: this.position,\n })\n this.position += duration // Update position\n\n this.featureExtractorWorker?.postMessage(\n {\n command: 'process',\n channelData: pcmBufferFloat,\n sampleRate,\n pointsPerSecond:\n this.config.pointsPerSecond \|\|\n DEFAULT_WEB_POINTS_PER_SECOND,\n algorithm: this.config.algorithm \|\| 'rms',\n bitDepth: this.bitDepth,\n fullAudioDurationMs: this.position * 1000,\n numberOfChannels: this.numberOfChannels,\n features: this.config.features,\n },\n []\n )\n }\n\n logger.debug(\n `WebRecorder initialized -- recordSampleRate=${this.audioContext.sampleRate}`,\n this.config\n )\n this.audioWorkletNode.port.postMessage({\n command: 'init',\n recordSampleRate: this.audioContext.sampleRate, // Pass the original sample rate\n exportSampleRate:\n this.config.sampleRate ?? this.audioContext.sampleRate,\n bitDepth: this.bitDepth,\n exportBitDepth: this.exportBitDepth,\n channels: this.numberOfChannels,\n interval: this.config.interval ?? DEFAULT_WEB_INTERVAL,\n })\n\n // Connect the source to the AudioWorkletNode and start recording\n this.source.connect(this.audioWorkletNode)\n this.audioWorkletNode.connect(this.audioContext.destination)\n } catch (error) {\n console.error(`[${TAG}] Failed to initialize WebRecorder`, error)\n }\n }\n\n initFeatureExtractorWorker(featuresExtratorUrl?: string) {\n try {\n if (featuresExtratorUrl) {\n // Initialize the feature extractor worker\n //TODO: create audio feature extractor from a Blob instead of url since we cannot include the url directly in the library\n // We keep the url during dev and use the blob in production.\n this.featureExtractorWorker = new Worker(\n new URL(featuresExtratorUrl, window.location.href)\n )\n this.featureExtractorWorker.onmessage =\n this.handleFeatureExtractorMessage.bind(this)\n this.featureExtractorWorker.onerror =\n this.handleWorkerError.bind(this)\n } else {\n // Fallback to the inline worker if the URL is not provided\n this.initFallbackWorker()\n }\n } catch (error) {\n console.error(\n `[${TAG}] Failed to initialize feature extractor worker`,\n error\n )\n this.initFallbackWorker()\n }\n }\n\n initFallbackWorker() {\n try {\n const blob = new Blob([InlineFeaturesExtractor], {\n type: 'application/javascript',\n })\n const url = URL.createObjectURL(blob)\n this.featureExtractorWorker = new Worker(url)\n this.featureExtractorWorker.onmessage =\n this.handleFeatureExtractorMessage.bind(this)\n this.featureExtractorWorker.onerror = (error) => {\n console.error(`[${TAG}] Default Inline worker failed`, error)\n }\n logger.log('Inline worker initialized successfully')\n } catch (error) {\n console.error(\n `[${TAG}] Failed to initialize Inline Feature Extractor worker`,\n error\n )\n }\n }\n\n handleWorkerError(error: ErrorEvent) {\n console.error(`[${TAG}] Feature extractor worker error:`, error)\n }\n\n handleFeatureExtractorMessage(event: AudioFeaturesEvent) {\n if (event.data.command === 'features') {\n const segmentResult = event.data.result\n\n // Merge the segment result with the full audio analysis data\n this.audioAnalysisData.dataPoints.push(...segmentResult.dataPoints)\n this.audioAnalysisData.speakerChanges?.push(\n ...(segmentResult.speakerChanges ?? [])\n )\n this.audioAnalysisData.durationMs = segmentResult.durationMs\n if (segmentResult.amplitudeRange) {\n this.audioAnalysisData.amplitudeRange = {\n min: Math.min(\n this.audioAnalysisData.amplitudeRange.min,\n segmentResult.amplitudeRange.min\n ),\n max: Math.max(\n this.audioAnalysisData.amplitudeRange.max,\n segmentResult.amplitudeRange.max\n ),\n }\n }\n // Handle the extracted features (e.g., emit an event or log them)\n logger.debug('features event segmentResult', segmentResult)\n logger.debug(\n 'features event audioAnalysisData',\n this.audioAnalysisData\n )\n this.emitAudioAnalysisCallback(segmentResult)\n }\n }\n\n start() {\n this.source.connect(this.audioWorkletNode)\n this.audioWorkletNode.connect(this.audioContext.destination)\n }\n\n stop(): Promise<Float32Array> {\n return new Promise((resolve, reject) => {\n try {\n if (this.audioWorkletNode) {\n // this.source.disconnect(this.audioWorkletNode);\n // this.audioWorkletNode.disconnect(this.audioContext.destination);\n this.audioWorkletNode.port.postMessage({ command: 'stop' })\n\n // Set a timeout to reject the promise if no message is received within 5 seconds\n const timeout = setTimeout(() => {\n this.audioWorkletNode.port.removeEventListener(\n 'message',\n onMessage\n )\n reject(\n new Error(\n \"Timeout error, audioWorkletNode didn't complete.\"\n )\n )\n }, 5000)\n\n // Listen for the recordedData message to confirm stopping\n const onMessage = async (event: AudioWorkletEvent) => {\n const command = event.data.command\n if (command === 'recordedData') {\n clearTimeout(timeout) // Clear the timeout\n\n const rawPCMDataFull =\n event.data.recordedData?.slice(0)\n\n if (!rawPCMDataFull) {\n reject(new Error('Failed to get recorded data'))\n return\n }\n\n // Compute duration of the recorded data\n const duration =\n rawPCMDataFull.byteLength /\n (this.audioContext.sampleRate \n (this.exportBitDepth /\n this.numberOfChannels))\n logger.debug(\n `Received recorded data -- Duration: ${duration} vs ${rawPCMDataFull.byteLength / this.audioContext.sampleRate} seconds`\n )\n logger.debug(\n `recordedData.length=${rawPCMDataFull.byteLength} vs transmittedData.length=${this.bufferSize}`\n )\n\n // Remove the event listener after receiving the final data\n this.audioWorkletNode.port.removeEventListener(\n 'message',\n onMessage\n )\n resolve(this.buffer) // Resolve the promise with the collected buffers\n }\n }\n this.audioWorkletNode.port.addEventListener(\n 'message',\n onMessage\n )\n }\n\n // Stop all media stream tracks to stop the browser recording\n this.stopMediaStreamTracks()\n } catch (error) {\n reject(error)\n }\n })\n }\n\n pause() {\n this.source.disconnect(this.audioWorkletNode) // Disconnect the source from the AudioWorkletNode\n this.audioWorkletNode.disconnect(this.audioContext.destination) // Disconnect the AudioWorkletNode from the destination\n this.audioWorkletNode.port.postMessage({ command: 'pause' })\n }\n\n stopMediaStreamTracks() {\n // Stop all audio tracks to stop the recording icon\n const tracks = this.source.mediaStream.getTracks()\n tracks.forEach((track) => track.stop())\n }\n\n async playRecordedData({\n recordedData,\n }: {\n recordedData: ArrayBuffer\n mimeType?: string\n }) {\n try {\n const blob = new Blob([recordedData])\n const url = URL.createObjectURL(blob)\n const response = await fetch(url)\n const arrayBuffer = await response.arrayBuffer()\n\n // Decode the audio data\n const audioBuffer =\n await this.audioContext.decodeAudioData(arrayBuffer)\n\n // Create a buffer source node and play the audio\n const bufferSource = this.audioContext.createBufferSource()\n bufferSource.buffer = audioBuffer\n bufferSource.connect(this.audioContext.destination)\n bufferSource.start()\n logger.debug('Playing recorded data', recordedData)\n } catch (error) {\n console.error(`[${TAG}] Failed to play recorded data:`, error)\n }\n }\n\n private checkAudioContextFormat({ sampleRate }: { sampleRate: number }) {\n // Create a silent AudioBuffer\n const frameCount = sampleRate 1.0 // 1 second buffer\n const audioBuffer = this.audioContext.createBuffer(\n 1,\n frameCount,\n sampleRate\n )\n\n // Check the format\n const channelData = audioBuffer.getChannelData(0)\n const bitDepth = channelData.BYTES_PER_ELEMENT * 8 // 4 bytes per element means 32-bit\n\n return {\n sampleRate: audioBuffer.sampleRate,\n bitDepth,\n numberOfChannels: audioBuffer.numberOfChannels,\n }\n }\n\n resume() {\n this.source.connect(this.audioWorkletNode)\n this.audioWorkletNode.connect(this.audioContext.destination)\n this.audioWorkletNode.port.postMessage({ command: 'resume' })\n }\n}\n"]}

package/build/utils/writeWavHeader.d.ts CHANGED Viewed

@@ -1,9 +1,49 @@
-interface WavHeaderOptions {
-    buffer: ArrayBuffer;
+/**
+ * Options for creating a WAV header.
+ */
+export interface WavHeaderOptions {
+    /** Optional buffer containing audio data. If provided, it will be combined with the header. */
+    buffer?: ArrayBuffer;
+    /** The sample rate of the audio in Hz (e.g., 44100). */
     sampleRate: number;
+    /** The number of audio channels (e.g., 1 for mono, 2 for stereo). */
     numChannels: number;
+    /** The bit depth of the audio (e.g., 16, 24, or 32). */
     bitDepth: number;
 }
+/**
+ * Writes or updates a WAV (RIFF) header based on the provided options.
+ *
+ * This function can be used in three ways:
+ * 1. To create a standalone WAV header (when no buffer is provided).
+ * 2. To create a WAV header and combine it with existing audio data (when a buffer without a header is provided).
+ * 3. To update an existing WAV header in the provided buffer.
+ *
+ * For streaming audio where the final size is unknown, this function sets the size fields
+ * to the maximum 32-bit value (0xFFFFFFFF). These can be updated later using the
+ * `updateWavHeaderSize` function once the final size is known.
+ *
+ * @param options - The options for creating or updating the WAV header.
+ * @returns An ArrayBuffer containing the WAV header, or the header combined with the provided audio data.
+ *
+ * @throws {Error} Throws an error if the provided options are invalid or if the buffer is too small.
+ *
+ * @example
+ * // Create a standalone WAV header
+ * const header = writeWavHeader({
+ *   sampleRate: 44100,
+ *   numChannels: 2,
+ *   bitDepth: 16
+ * });
+ *
+ * @example
+ * // Create a WAV header and combine it with audio data
+ * const completeWav = writeWavHeader({
+ *   buffer: audioData,
+ *   sampleRate: 44100,
+ *   numChannels: 2,
+ *   bitDepth: 16
+ * });
+ */
 export declare const writeWavHeader: ({ buffer, sampleRate, numChannels, bitDepth, }: WavHeaderOptions) => ArrayBuffer;
-export {};
 //# sourceMappingURL=writeWavHeader.d.ts.map

package/build/utils/writeWavHeader.d.ts.map CHANGED Viewed

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package/build/utils/writeWavHeader.js CHANGED Viewed

@@ -1,7 +1,40 @@
+// packages/expo-audio-stream/src/utils/writeWavHeader.ts
+/**
+ * Writes or updates a WAV (RIFF) header based on the provided options.
+ *
+ * This function can be used in three ways:
+ * 1. To create a standalone WAV header (when no buffer is provided).
+ * 2. To create a WAV header and combine it with existing audio data (when a buffer without a header is provided).
+ * 3. To update an existing WAV header in the provided buffer.
+ *
+ * For streaming audio where the final size is unknown, this function sets the size fields
+ * to the maximum 32-bit value (0xFFFFFFFF). These can be updated later using the
+ * `updateWavHeaderSize` function once the final size is known.
+ *
+ * @param options - The options for creating or updating the WAV header.
+ * @returns An ArrayBuffer containing the WAV header, or the header combined with the provided audio data.
+ *
+ * @throws {Error} Throws an error if the provided options are invalid or if the buffer is too small.
+ *
+ * @example
+ * // Create a standalone WAV header
+ * const header = writeWavHeader({
+ *   sampleRate: 44100,
+ *   numChannels: 2,
+ *   bitDepth: 16
+ * });
+ *
+ * @example
+ * // Create a WAV header and combine it with audio data
+ * const completeWav = writeWavHeader({
+ *   buffer: audioData,
+ *   sampleRate: 44100,
+ *   numChannels: 2,
+ *   bitDepth: 16
+ * });
+ */
 export const writeWavHeader = ({ buffer, sampleRate, numChannels, bitDepth, }) => {
     const bytesPerSample = bitDepth / 8;
-    const numSamples = buffer.byteLength / (numChannels * bytesPerSample);
-    const view = new DataView(buffer);
     const blockAlign = numChannels * bytesPerSample;
     const byteRate = sampleRate * blockAlign;
     // Function to write a string to the DataView
@@ -10,16 +43,10 @@ export const writeWavHeader = ({ buffer, sampleRate, numChannels, bitDepth, }) =
             view.setUint8(offset + i, string.charCodeAt(i));
         }
     };
-    // Check if the buffer already has a WAV header by looking for "RIFF" at the start
-    const existingHeader = view.getUint32(0, false) === 0x52494646; // "RIFF" in ASCII
-    if (!existingHeader) {
-        // Ensure the buffer is large enough for the WAV header
-        if (buffer.byteLength < 44) {
-            throw new Error('Buffer is too small to contain a valid WAV header');
-        }
-        // Write the WAV header
+    // Function to write or update the header
+    const writeHeader = (view, dataSize = 0xffffffff) => {
         writeString(view, 0, 'RIFF'); // ChunkID
-        view.setUint32(4, 36 + numSamples * blockAlign, true); // ChunkSize
+        view.setUint32(4, 36 + dataSize, true); // ChunkSize
         writeString(view, 8, 'WAVE'); // Format
         writeString(view, 12, 'fmt '); // Subchunk1ID
         view.setUint32(16, 16, true); // Subchunk1Size (16 for PCM)
@@ -30,16 +57,35 @@ export const writeWavHeader = ({ buffer, sampleRate, numChannels, bitDepth, }) =
         view.setUint16(32, blockAlign, true); // BlockAlign
         view.setUint16(34, bitDepth, true); // BitsPerSample
         writeString(view, 36, 'data'); // Subchunk2ID
-        view.setUint32(40, numSamples * blockAlign, true); // Subchunk2Size
+        view.setUint32(40, dataSize, true); // Subchunk2Size
+    };
+    if (buffer) {
+        if (buffer.byteLength < 44) {
+            throw new Error('Buffer is too small to contain a valid WAV header');
+        }
+        const view = new DataView(buffer);
+        // Check if the buffer already has a WAV header by looking for "RIFF" at the start
+        const existingHeader = view.getUint32(0, false) === 0x52494646; // "RIFF" in ASCII
+        if (existingHeader) {
+            // Update the existing header
+            writeHeader(view, buffer.byteLength - 44);
+            return buffer;
+        }
+        else {
+            // Combine the new header with the existing buffer
+            const newBuffer = new ArrayBuffer(44 + buffer.byteLength);
+            const newView = new DataView(newBuffer);
+            writeHeader(newView, buffer.byteLength);
+            new Uint8Array(newBuffer).set(new Uint8Array(buffer), 44);
+            return newBuffer;
+        }
     }
     else {
-        // Update the existing WAV header if necessary
-        view.setUint32(4, 36 + numSamples * blockAlign, true); // Update ChunkSize
-        view.setUint32(24, sampleRate, true); // Update SampleRate
-        view.setUint32(28, byteRate, true); // Update ByteRate
-        view.setUint16(32, blockAlign, true); // Update BlockAlign
-        view.setUint32(40, numSamples * blockAlign, true); // Update Subchunk2Size
+        // Create a standalone header
+        const headerBuffer = new ArrayBuffer(44);
+        const view = new DataView(headerBuffer);
+        writeHeader(view);
+        return headerBuffer;
     }
-    return buffer;
 };
 //# sourceMappingURL=writeWavHeader.js.map

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3 : 1, true) // AudioFormat (3 for float, 1 for PCM)\n view.setUint16(22, numChannels, true) // NumChannels\n view.setUint32(24, sampleRate, true) // SampleRate\n view.setUint32(28, byteRate, true) // ByteRate\n view.setUint16(32, blockAlign, true) // BlockAlign\n view.setUint16(34, bitDepth, true) // BitsPerSample\n writeString(view, 36, 'data') // Subchunk2ID\n view.setUint32(40, ~~numSamples * blockAlign~~, true) // Subchunk2Size\n } ~~else~~ {\n // ~~Update~~ the ~~existing~~ WAV header if ~~necessary~~\n view.~~setUint32~~(4, 36 + ~~numSamples~~ * ~~blockAlign,~~ ~~true~~) // Update ~~ChunkSize~~\n view~~.setUint32(24~~, ~~sampleRate,~~ ~~true~~) // ~~Update~~ ~~SampleRate~~\n ~~view.setUint32~~(~~28,~~ ~~byteRate,~~ ~~true~~) // ~~Update~~ ~~ByteRate~~\n ~~view.setUint16~~(32, ~~blockAlign~~, ~~true~~) // ~~Update~~ ~~BlockAlign~~\n ~~view.setUint32(40,~~ ~~numSamples~~ * ~~blockAlign,~~ ~~true~~) // ~~Update~~ ~~Subchunk2Size~~\n }\n\n ~~return buffer~~\n}\n"]}
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If provided, it will be combined with the header. /\n buffer?: ArrayBuffer\n /* The sample rate of the audio in Hz (e.g., 44100). /\n sampleRate: number\n /* The number of audio channels (e.g., 1 for mono, 2 for stereo). /\n numChannels: number\n /* The bit depth of the audio (e.g., 16, 24, or 32). /\n bitDepth: number\n}\n\n/\n Writes or updates a WAV (RIFF) header based on the provided options.\n \n This function can be used in three ways:\n * 1. To create a standalone WAV header (when no buffer is provided).\n * 2. To create a WAV header and combine it with existing audio data (when a buffer without a header is provided).\n * 3. To update an existing WAV header in the provided buffer.\n \n For streaming audio where the final size is unknown, this function sets the size fields\n * to the maximum 32-bit value (0xFFFFFFFF). These can be updated later using the\n * `updateWavHeaderSize` function once the final size is known.\n \n @param options - The options for creating or updating the WAV header.\n * @returns An ArrayBuffer containing the WAV header, or the header combined with the provided audio data.\n \n @throws {Error} Throws an error if the provided options are invalid or if the buffer is too small.\n \n @example\n * // Create a standalone WAV header\n * const header = writeWavHeader({\n * sampleRate: 44100,\n * numChannels: 2,\n * bitDepth: 16\n * });\n \n @example\n * // Create a WAV header and combine it with audio data\n * const completeWav = writeWavHeader({\n * buffer: audioData,\n * sampleRate: 44100,\n * numChannels: 2,\n * bitDepth: 16\n * });\n /\nexport const writeWavHeader = ({\n buffer,\n sampleRate,\n numChannels,\n bitDepth,\n}: WavHeaderOptions): ArrayBuffer => {\n const bytesPerSample = bitDepth / 8\n const blockAlign = numChannels bytesPerSample\n const byteRate = sampleRate * blockAlign\n\n // Function to write a string to the DataView\n const writeString = (view: DataView, offset: number, string: string) => {\n for (let i = 0; i < string.length; i++) {\n view.setUint8(offset + i, string.charCodeAt(i))\n }\n }\n\n // Function to write or update the header\n const writeHeader = (view: DataView, dataSize: number = 0xffffffff) => {\n writeString(view, 0, 'RIFF') // ChunkID\n view.setUint32(4, 36 + dataSize, true) // ChunkSize\n writeString(view, 8, 'WAVE') // Format\n writeString(view, 12, 'fmt ') // Subchunk1ID\n view.setUint32(16, 16, true) // Subchunk1Size (16 for PCM)\n view.setUint16(20, bitDepth === 32 ? 3 : 1, true) // AudioFormat (3 for float, 1 for PCM)\n view.setUint16(22, numChannels, true) // NumChannels\n view.setUint32(24, sampleRate, true) // SampleRate\n view.setUint32(28, byteRate, true) // ByteRate\n view.setUint16(32, blockAlign, true) // BlockAlign\n view.setUint16(34, bitDepth, true) // BitsPerSample\n writeString(view, 36, 'data') // Subchunk2ID\n view.setUint32(40, dataSize, true) // Subchunk2Size\n }\n\n if (buffer) {\n if (buffer.byteLength < 44) {\n throw new Error('Buffer is too small to contain a valid WAV header')\n }\n\n const view = new DataView(buffer)\n\n // Check if the buffer already has a WAV header by looking for \"RIFF\" at the start\n const existingHeader = view.getUint32(0, false) === 0x52494646 // \"RIFF\" in ASCII\n\n if (existingHeader) {\n // Update the existing header\n writeHeader(view, buffer.byteLength - 44)\n return buffer\n } else {\n // Combine the new header with the existing buffer\n const newBuffer = new ArrayBuffer(44 + buffer.byteLength)\n const newView = new DataView(newBuffer)\n writeHeader(newView, buffer.byteLength)\n new Uint8Array(newBuffer).set(new Uint8Array(buffer), 44)\n return newBuffer\n }\n } else {\n // Create a standalone header\n const headerBuffer = new ArrayBuffer(44)\n const view = new DataView(headerBuffer)\n writeHeader(view)\n return headerBuffer\n }\n}\n"]}

package/package.json CHANGED Viewed

@@ -1,12 +1,12 @@
 {
   "name": "@siteed/expo-audio-stream",
-  "version": "1.1.6",
+  "version": "1.1.8",
   "description": "stream audio crossplatform",
   "license": "MIT",
   "main": "build/index.js",
   "types": "build/index.d.ts",
   "author": "Arthur Breton <abreton@siteed.net> (https://github.com/deeeed)",
-  "homepage": "https://github.com/deeeed/expo-audio-stream#readme",
+  "homepage": "https://github.com/deeeed/expo-audio-stream/blob/main/packages/expo-audio-stream/README.md",
   "repository": {
     "type": "git",
     "url": "git+https://github.com/deeeed/expo-audio-stream.git",
@@ -38,6 +38,7 @@
   },
   "devDependencies": {
     "@expo/config-plugins": "^7.9.1",
+    "@siteed/react-native-logger": "^0.10.0",
     "@size-limit/preset-big-lib": "^11.1.4",
     "@types/jest": "^29.5.12",
     "@types/node": "^20.12.7",
@@ -63,6 +64,7 @@
     "typescript": "^5.5.4"
   },
   "peerDependencies": {
+    "@siteed/react-native-logger": "*",
     "expo": "*",
     "react": "*",
     "react-native": "*"
@@ -70,8 +72,5 @@
   "publishConfig": {
     "access": "public",
     "registry": "https://registry.npmjs.org"
-  },
-  "dependencies": {
-    "@siteed/react-native-logger": "^0.9.3"
   }
 }

package/publish.sh CHANGED Viewed

@@ -5,4 +5,7 @@ yarn version patch
 version=$(node -p "require('./package.json').version")
 git add .
 git commit -m "feat: bump version to $version"
-yarn clean && yarn prepare && yarn npm publish
+echo "Generated updated documentation for version $version"
+yarn docgen
+yarn clean && yarn prepare && yarn npm publish
+rm package-lock.json

package/src/ExpoAudioStream.web.ts CHANGED Viewed

@@ -13,7 +13,7 @@ import { WebRecorder } from './WebRecorder.web'
 import { AudioEventPayload } from './events'
 import { getLogger } from './logger'
 import { encodingToBitDepth } from './utils/encodingToBitDepth'
-import { writeWavHeader } from './utils/writeWavHeader'
+import { WavHeaderOptions, writeWavHeader } from './utils/writeWavHeader'
 export interface EmitAudioEventProps {
     data: Float32Array
@@ -184,7 +184,8 @@ export class ExpoAudioStreamWeb extends EventEmitter {
         this.isRecording = false
         this.currentDurationMs = Date.now() - this.recordingStartTime
-        const wavConfig = {
+        // Rewrite wav header with correct data size
+        const wavConfig: WavHeaderOptions = {
             buffer: fullPcmBufferArray.buffer,
             sampleRate: this.recordingConfig?.sampleRate ?? 44100,
             numChannels: this.recordingConfig?.channels ?? 1,

package/src/WebRecorder.web.ts CHANGED Viewed

@@ -7,6 +7,7 @@ import {
 } from './ExpoAudioStream.web'
 import { getLogger } from './logger'
 import { encodingToBitDepth } from './utils/encodingToBitDepth'
+import { writeWavHeader } from './utils/writeWavHeader'
 import { InlineFeaturesExtractor } from './workers/InlineFeaturesExtractor.web'
 import { InlineAudioWebWorker } from './workers/inlineAudioWebWorker.web'
@@ -73,8 +74,6 @@ export class WebRecorder {
         this.emitAudioAnalysisCallback = emitAudioAnalysisCallback
         this.config = recordingConfig
         this.position = 0
-        this.bufferSize = 0
-        this.buffer = new Float32Array(0) // Initialize the buffer
         const audioContextFormat = this.checkAudioContextFormat({
             sampleRate: this.audioContext.sampleRate,
@@ -96,6 +95,23 @@ export class WebRecorder {
             audioContextFormat.bitDepth ||
             DEFAULT_WEB_BITDEPTH
+        // Initialize WAV header
+        const wavHeader = writeWavHeader({
+            sampleRate: this.audioContext.sampleRate,
+            numChannels: this.numberOfChannels,
+            bitDepth: this.exportBitDepth,
+        })
+        logger.debug(`WAV header length: ${wavHeader.byteLength}`, wavHeader)
+        // Initialize the buffer with WAV header
+        this.buffer = new Float32Array(
+            wavHeader.byteLength / Float32Array.BYTES_PER_ELEMENT
+        )
+        this.bufferSize = this.buffer.byteLength
+        // Copy WAV header to Float32Array buffer
+        new Uint8Array(this.buffer.buffer).set(new Uint8Array(wavHeader))
         this.audioAnalysisData = {
             amplitudeRange: { min: 0, max: 0 },
             dataPoints: [],

package/src/utils/writeWavHeader.ts CHANGED Viewed

@@ -1,11 +1,53 @@
 // packages/expo-audio-stream/src/utils/writeWavHeader.ts
-interface WavHeaderOptions {
-    buffer: ArrayBuffer
+/**
+ * Options for creating a WAV header.
+ */
+export interface WavHeaderOptions {
+    /** Optional buffer containing audio data. If provided, it will be combined with the header. */
+    buffer?: ArrayBuffer
+    /** The sample rate of the audio in Hz (e.g., 44100). */
     sampleRate: number
+    /** The number of audio channels (e.g., 1 for mono, 2 for stereo). */
     numChannels: number
+    /** The bit depth of the audio (e.g., 16, 24, or 32). */
     bitDepth: number
 }
+/**
+ * Writes or updates a WAV (RIFF) header based on the provided options.
+ *
+ * This function can be used in three ways:
+ * 1. To create a standalone WAV header (when no buffer is provided).
+ * 2. To create a WAV header and combine it with existing audio data (when a buffer without a header is provided).
+ * 3. To update an existing WAV header in the provided buffer.
+ *
+ * For streaming audio where the final size is unknown, this function sets the size fields
+ * to the maximum 32-bit value (0xFFFFFFFF). These can be updated later using the
+ * `updateWavHeaderSize` function once the final size is known.
+ *
+ * @param options - The options for creating or updating the WAV header.
+ * @returns An ArrayBuffer containing the WAV header, or the header combined with the provided audio data.
+ *
+ * @throws {Error} Throws an error if the provided options are invalid or if the buffer is too small.
+ *
+ * @example
+ * // Create a standalone WAV header
+ * const header = writeWavHeader({
+ *   sampleRate: 44100,
+ *   numChannels: 2,
+ *   bitDepth: 16
+ * });
+ *
+ * @example
+ * // Create a WAV header and combine it with audio data
+ * const completeWav = writeWavHeader({
+ *   buffer: audioData,
+ *   sampleRate: 44100,
+ *   numChannels: 2,
+ *   bitDepth: 16
+ * });
+ */
 export const writeWavHeader = ({
     buffer,
     sampleRate,
@@ -13,8 +55,6 @@ export const writeWavHeader = ({
     bitDepth,
 }: WavHeaderOptions): ArrayBuffer => {
     const bytesPerSample = bitDepth / 8
-    const numSamples = buffer.byteLength / (numChannels * bytesPerSample)
-    const view = new DataView(buffer)
     const blockAlign = numChannels * bytesPerSample
     const byteRate = sampleRate * blockAlign
@@ -25,18 +65,10 @@ export const writeWavHeader = ({
         }
     }
-    // Check if the buffer already has a WAV header by looking for "RIFF" at the start
-    const existingHeader = view.getUint32(0, false) === 0x52494646 // "RIFF" in ASCII
-    if (!existingHeader) {
-        // Ensure the buffer is large enough for the WAV header
-        if (buffer.byteLength < 44) {
-            throw new Error('Buffer is too small to contain a valid WAV header')
-        }
-        // Write the WAV header
+    // Function to write or update the header
+    const writeHeader = (view: DataView, dataSize: number = 0xffffffff) => {
         writeString(view, 0, 'RIFF') // ChunkID
-        view.setUint32(4, 36 + numSamples * blockAlign, true) // ChunkSize
+        view.setUint32(4, 36 + dataSize, true) // ChunkSize
         writeString(view, 8, 'WAVE') // Format
         writeString(view, 12, 'fmt ') // Subchunk1ID
         view.setUint32(16, 16, true) // Subchunk1Size (16 for PCM)
@@ -47,15 +79,36 @@ export const writeWavHeader = ({
         view.setUint16(32, blockAlign, true) // BlockAlign
         view.setUint16(34, bitDepth, true) // BitsPerSample
         writeString(view, 36, 'data') // Subchunk2ID
-        view.setUint32(40, numSamples * blockAlign, true) // Subchunk2Size
-    } else {
-        // Update the existing WAV header if necessary
-        view.setUint32(4, 36 + numSamples * blockAlign, true) // Update ChunkSize
-        view.setUint32(24, sampleRate, true) // Update SampleRate
-        view.setUint32(28, byteRate, true) // Update ByteRate
-        view.setUint16(32, blockAlign, true) // Update BlockAlign
-        view.setUint32(40, numSamples * blockAlign, true) // Update Subchunk2Size
+        view.setUint32(40, dataSize, true) // Subchunk2Size
     }
-    return buffer
+    if (buffer) {
+        if (buffer.byteLength < 44) {
+            throw new Error('Buffer is too small to contain a valid WAV header')
+        }
+        const view = new DataView(buffer)
+        // Check if the buffer already has a WAV header by looking for "RIFF" at the start
+        const existingHeader = view.getUint32(0, false) === 0x52494646 // "RIFF" in ASCII
+        if (existingHeader) {
+            // Update the existing header
+            writeHeader(view, buffer.byteLength - 44)
+            return buffer
+        } else {
+            // Combine the new header with the existing buffer
+            const newBuffer = new ArrayBuffer(44 + buffer.byteLength)
+            const newView = new DataView(newBuffer)
+            writeHeader(newView, buffer.byteLength)
+            new Uint8Array(newBuffer).set(new Uint8Array(buffer), 44)
+            return newBuffer
+        }
+    } else {
+        // Create a standalone header
+        const headerBuffer = new ArrayBuffer(44)
+        const view = new DataView(headerBuffer)
+        writeHeader(view)
+        return headerBuffer
+    }
 }