@siteed/expo-audio-studio 2.4.1 → 2.6.0

package/android/src/main/java/net/siteed/audiostream/LogUtils.kt

@@ -0,0 +1,65 @@
+package net.siteed.audiostream
+
+import android.util.Log
+
+/**
+ * Utility class for standardized logging across the ExpoAudioStream library.
+ * Provides consistent logging format and tags for easier filtering in logcat.
+ */
+object LogUtils {
+    // Format: [ExpoAudioStream:ClassName]
+    private const val TAG_PREFIX = "ExpoAudioStream"
+    
+    /**
+     * Logs a debug message with a consistent format.
+     * 
+     * @param className The name of the class generating the log
+     * @param message The message to log
+     */
+    fun d(className: String, message: String) {
+        Log.d("$TAG_PREFIX:$className", message)
+    }
+    
+    /**
+     * Logs an error message with a consistent format.
+     * 
+     * @param className The name of the class generating the log
+     * @param message The message to log
+     * @param throwable Optional throwable to include in the log
+     */
+    fun e(className: String, message: String, throwable: Throwable? = null) {
+        Log.e("$TAG_PREFIX:$className", message, throwable)
+    }
+    
+    /**
+     * Logs a warning message with a consistent format.
+     * 
+     * @param className The name of the class generating the log
+     * @param message The message to log
+     * @param throwable Optional throwable to include in the log
+     */
+    fun w(className: String, message: String, throwable: Throwable? = null) {
+        Log.w("$TAG_PREFIX:$className", message, throwable)
+    }
+    
+    /**
+     * Logs an info message with a consistent format.
+     * 
+     * @param className The name of the class generating the log
+     * @param message The message to log
+     */
+    fun i(className: String, message: String) {
+        Log.i("$TAG_PREFIX:$className", message)
+    }
+    
+    /**
+     * Creates a formatted tag for direct use with Android Log methods.
+     * Use this if you need to use the Android Log methods directly.
+     * 
+     * @param className The name of the class generating the log
+     * @return A formatted tag string
+     */
+    fun tag(className: String): String {
+        return "$TAG_PREFIX:$className"
+    }
+} 

package/android/src/main/java/net/siteed/audiostream/RecordingConfig.kt

@@ -23,6 +23,8 @@ data class RecordingConfig(
     val autoResumeAfterInterruption: Boolean = false,
     val outputDirectory: String? = null,
     val filename: String? = null,
+    val deviceId: String? = null,
+    val deviceDisconnectionBehavior: String? = null,
 ) {
     companion object {
         fun fromMap(options: Map<String, Any?>?): Result<Pair<RecordingConfig, AudioFormatInfo>> {
@@ -82,6 +84,10 @@ data class RecordingConfig(
                 }
             }
 
+            // Get device-related settings
+            val deviceId = options["deviceId"] as? String
+            val deviceDisconnectionBehavior = options["deviceDisconnectionBehavior"] as? String
+
             // Initialize the recording configuration with cleaned directory path
             val tempRecordingConfig = RecordingConfig(
                 sampleRate = options.getNumberOrDefault("sampleRate", Constants.DEFAULT_SAMPLE_RATE),
@@ -105,7 +111,9 @@ data class RecordingConfig(
                         .trim('/')
                         .replace("//", "/")
                 },
-                filename = options["filename"] as? String
+                filename = options["filename"] as? String,
+                deviceId = deviceId,
+                deviceDisconnectionBehavior = deviceDisconnectionBehavior
             )
 
             // Validate sample rate and channels

package/build/AudioAnalysis/AudioAnalysis.types.js.map

@@ -1 +1 @@
-{"version":3,"file":"AudioAnalysis.types.js","sourceRoot":"","sources":["../../src/AudioAnalysis/AudioAnalysis.types.ts"],"names":[],"mappings":"AAAA,sEAAsE","sourcesContent":["// packages/expo-audio-stream/src/AudioAnalysis/AudioAnalysis.types.ts\n\nimport { BitDepth, ConsoleLike } from '../ExpoAudioStream.types'\n\n/**\n * Represents the configuration for decoding audio data.\n */\nexport interface DecodingConfig {\n    /** Target sample rate for decoded audio (Android and Web) */\n    targetSampleRate?: number\n    /** Target number of channels (Android and Web) */\n    targetChannels?: number\n    /** Target bit depth (Android and Web) */\n    targetBitDepth?: BitDepth\n    /** Whether to normalize audio levels (Android and Web) */\n    normalizeAudio?: boolean\n}\n\n/**\n * Represents speech-related features extracted from audio.\n */\nexport interface SpeechFeatures {\n    isActive: boolean // Whether speech is detected in this segment\n    speakerId?: number // Optional speaker identification\n    // Could add more speech-related features here like:\n    // confidence: number\n    // language?: string\n    // sentiment?: number\n    // etc.\n}\n\n/**\n * Represents various audio features extracted from an audio signal.\n */\nexport interface AudioFeatures {\n    energy?: number // The infinite integral of the squared signal, representing the overall energy of the audio.\n    mfcc?: number[] // Mel-frequency cepstral coefficients, describing the short-term power spectrum of a sound.\n    rms?: number // Root mean square value, indicating the amplitude of the audio signal.\n    minAmplitude?: number // Minimum amplitude value in the audio signal.\n    maxAmplitude?: number // Maximum amplitude value in the audio signal.\n    zcr?: number // Zero-crossing rate, indicating the rate at which the signal changes sign.\n    spectralCentroid?: number // The center of mass of the spectrum, indicating the brightness of the sound.\n    spectralFlatness?: number // Measure of the flatness of the spectrum, indicating how noise-like the signal is.\n    spectralRolloff?: number // The frequency below which a specified percentage (usually 85%) of the total spectral energy lies.\n    spectralBandwidth?: number // The width of the spectrum, indicating the range of frequencies present.\n    chromagram?: number[] // Chromagram, representing the 12 different pitch classes of the audio.\n    tempo?: number // Estimated tempo of the audio signal, measured in beats per minute (BPM).\n    hnr?: number // Harmonics-to-noise ratio, indicating the proportion of harmonics to noise in the audio signal.\n    melSpectrogram?: number[] // Mel-scaled spectrogram representation of the audio.\n    spectralContrast?: number[] // Spectral contrast features representing the difference between peaks and valleys.\n    tonnetz?: number[] // Tonal network features representing harmonic relationships.\n    pitch?: number // Pitch of the audio signal, measured in Hertz (Hz).\n    crc32?: number // crc32 checksum of the audio signal, used to verify the integrity of the audio.\n}\n\n/**\n * Options to specify which audio features to extract.\n */\nexport interface AudioFeaturesOptions {\n    energy?: boolean\n    mfcc?: boolean\n    rms?: boolean\n    zcr?: boolean\n    spectralCentroid?: boolean\n    spectralFlatness?: boolean\n    spectralRolloff?: boolean\n    spectralBandwidth?: boolean\n    chromagram?: boolean\n    tempo?: boolean\n    hnr?: boolean\n    melSpectrogram?: boolean\n    spectralContrast?: boolean\n    tonnetz?: boolean\n    pitch?: boolean\n    crc32?: boolean\n}\n\n/**\n * Represents a single data point in the audio analysis.\n */\nexport interface DataPoint {\n    id: number\n    amplitude: number // Peak amplitude for the segment\n    rms: number // Root mean square value\n    dB: number // dBFS (decibels relative to full scale) computed from RMS value\n    silent: boolean // Always computed\n    features?: AudioFeatures\n    speech?: SpeechFeatures\n    startTime?: number\n    endTime?: number\n    // start / end position in bytes\n    startPosition?: number\n    endPosition?: number\n    // number of audio samples for this point (samples size depends on bit depth)\n    samples?: number\n}\n\n/**\n * Represents the complete data from the audio analysis.\n */\nexport interface AudioAnalysis {\n    segmentDurationMs: number // Duration of each segment in milliseconds\n    durationMs: number // Duration of the audio in milliseconds\n    bitDepth: number // Bit depth of the audio\n    samples: number // Size of the audio in bytes\n    numberOfChannels: number // Number of audio channels\n    sampleRate: number // Sample rate of the audio\n    dataPoints: DataPoint[] // Array of data points from the analysis.\n    amplitudeRange: {\n        min: number\n        max: number\n    }\n    rmsRange: {\n        min: number\n        max: number\n    }\n    // TODO: speaker changes into a broader speech analysis section\n    speechAnalysis?: {\n        speakerChanges: {\n            timestamp: number\n            speakerId: number\n        }[]\n        // Could add more speech analysis data here like:\n        // dominantSpeaker?: number\n        // totalSpeechDuration?: number\n        // speakerStats?: { [speakerId: number]: { duration: number, segments: number } }\n    }\n}\n\n/**\n * Options for specifying a time range within an audio file.\n */\nexport interface AudioRangeOptions {\n    /** Start time in milliseconds */\n    startTimeMs?: number\n    /** End time in milliseconds */\n    endTimeMs?: number\n}\n\n/**\n * Options for generating a quick preview of audio waveform.\n * This is optimized for UI rendering with a specified number of points.\n */\nexport interface PreviewOptions extends AudioRangeOptions {\n    /** URI of the audio file to analyze */\n    fileUri: string\n    /**\n     * Total number of points to generate for the preview.\n     * @default 100\n     */\n    numberOfPoints?: number\n    /**\n     * Optional logger for debugging.\n     */\n    logger?: ConsoleLike\n    /**\n     * Optional configuration for decoding the audio file.\n     * Defaults to:\n     * - targetSampleRate: undefined (keep original)\n     * - targetChannels: undefined (keep original)\n     * - targetBitDepth: 16\n     * - normalizeAudio: false\n     */\n    decodingOptions?: DecodingConfig\n}\n\n/**\n * Options for mel-spectrogram extraction\n * \n * @experimental This feature is experimental and currently only available on Android.\n * The API may change in future versions.\n */\nexport interface ExtractMelSpectrogramOptions {\n    fileUri?: string // Path to audio file\n    arrayBuffer?: ArrayBuffer // Raw audio buffer\n    windowSizeMs: number // Window size in ms (e.g., 25)\n    hopLengthMs: number // Hop length in ms (e.g., 10)\n    nMels: number // Number of mel filters (e.g., 60)\n    fMin?: number // Min frequency (default: 0)\n    fMax?: number // Max frequency (default: sampleRate / 2)\n    windowType?: 'hann' | 'hamming' // Window function (default: 'hann')\n    normalize?: boolean // Mean normalization (default: false)\n    logScale?: boolean // Log scaling of mel energies (default: true)\n    decodingOptions?: DecodingConfig // Audio decoding settings\n    startTimeMs?: number // Optional start time\n    endTimeMs?: number // Optional end time\n    logger?: ConsoleLike\n}\n\n/**\n * Return type for mel spectrogram extraction\n * \n * @experimental This feature is experimental and currently only available on Android.\n * The API may change in future versions.\n */\nexport interface MelSpectrogram {\n    spectrogram: number[][] // 2D array [time][mel]\n    sampleRate: number // Audio sample rate\n    nMels: number // Number of mel filters\n    timeSteps: number // Number of time frames\n    durationMs: number // Audio duration in ms\n}\n"]}
+{"version":3,"file":"AudioAnalysis.types.js","sourceRoot":"","sources":["../../src/AudioAnalysis/AudioAnalysis.types.ts"],"names":[],"mappings":"AAAA,sEAAsE","sourcesContent":["// packages/expo-audio-stream/src/AudioAnalysis/AudioAnalysis.types.ts\n\nimport { BitDepth, ConsoleLike } from '../ExpoAudioStream.types'\n\n/**\n * Represents the configuration for decoding audio data.\n */\nexport interface DecodingConfig {\n    /** Target sample rate for decoded audio (Android and Web) */\n    targetSampleRate?: number\n    /** Target number of channels (Android and Web) */\n    targetChannels?: number\n    /** Target bit depth (Android and Web) */\n    targetBitDepth?: BitDepth\n    /** Whether to normalize audio levels (Android and Web) */\n    normalizeAudio?: boolean\n}\n\n/**\n * Represents speech-related features extracted from audio.\n */\nexport interface SpeechFeatures {\n    isActive: boolean // Whether speech is detected in this segment\n    speakerId?: number // Optional speaker identification\n    // Could add more speech-related features here like:\n    // confidence: number\n    // language?: string\n    // sentiment?: number\n    // etc.\n}\n\n/**\n * Represents various audio features extracted from an audio signal.\n */\nexport interface AudioFeatures {\n    energy?: number // The infinite integral of the squared signal, representing the overall energy of the audio.\n    mfcc?: number[] // Mel-frequency cepstral coefficients, describing the short-term power spectrum of a sound.\n    rms?: number // Root mean square value, indicating the amplitude of the audio signal.\n    minAmplitude?: number // Minimum amplitude value in the audio signal.\n    maxAmplitude?: number // Maximum amplitude value in the audio signal.\n    zcr?: number // Zero-crossing rate, indicating the rate at which the signal changes sign.\n    spectralCentroid?: number // The center of mass of the spectrum, indicating the brightness of the sound.\n    spectralFlatness?: number // Measure of the flatness of the spectrum, indicating how noise-like the signal is.\n    spectralRolloff?: number // The frequency below which a specified percentage (usually 85%) of the total spectral energy lies.\n    spectralBandwidth?: number // The width of the spectrum, indicating the range of frequencies present.\n    chromagram?: number[] // Chromagram, representing the 12 different pitch classes of the audio.\n    tempo?: number // Estimated tempo of the audio signal, measured in beats per minute (BPM).\n    hnr?: number // Harmonics-to-noise ratio, indicating the proportion of harmonics to noise in the audio signal.\n    melSpectrogram?: number[] // Mel-scaled spectrogram representation of the audio.\n    spectralContrast?: number[] // Spectral contrast features representing the difference between peaks and valleys.\n    tonnetz?: number[] // Tonal network features representing harmonic relationships.\n    pitch?: number // Pitch of the audio signal, measured in Hertz (Hz).\n    crc32?: number // crc32 checksum of the audio signal, used to verify the integrity of the audio.\n}\n\n/**\n * Options to specify which audio features to extract.\n */\nexport interface AudioFeaturesOptions {\n    energy?: boolean\n    mfcc?: boolean\n    rms?: boolean\n    zcr?: boolean\n    spectralCentroid?: boolean\n    spectralFlatness?: boolean\n    spectralRolloff?: boolean\n    spectralBandwidth?: boolean\n    chromagram?: boolean\n    tempo?: boolean\n    hnr?: boolean\n    melSpectrogram?: boolean\n    spectralContrast?: boolean\n    tonnetz?: boolean\n    pitch?: boolean\n    crc32?: boolean\n}\n\n/**\n * Represents a single data point in the audio analysis.\n */\nexport interface DataPoint {\n    id: number\n    amplitude: number // Peak amplitude for the segment\n    rms: number // Root mean square value\n    dB: number // dBFS (decibels relative to full scale) computed from RMS value\n    silent: boolean // Always computed\n    features?: AudioFeatures\n    speech?: SpeechFeatures\n    startTime?: number\n    endTime?: number\n    // start / end position in bytes\n    startPosition?: number\n    endPosition?: number\n    // number of audio samples for this point (samples size depends on bit depth)\n    samples?: number\n}\n\n/**\n * Represents the complete data from the audio analysis.\n */\nexport interface AudioAnalysis {\n    segmentDurationMs: number // Duration of each segment in milliseconds\n    durationMs: number // Duration of the audio in milliseconds\n    bitDepth: number // Bit depth of the audio\n    samples: number // Size of the audio in bytes\n    numberOfChannels: number // Number of audio channels\n    sampleRate: number // Sample rate of the audio\n    dataPoints: DataPoint[] // Array of data points from the analysis.\n    amplitudeRange: {\n        min: number\n        max: number\n    }\n    rmsRange: {\n        min: number\n        max: number\n    }\n    // TODO: speaker changes into a broader speech analysis section\n    speechAnalysis?: {\n        speakerChanges: {\n            timestamp: number\n            speakerId: number\n        }[]\n        // Could add more speech analysis data here like:\n        // dominantSpeaker?: number\n        // totalSpeechDuration?: number\n        // speakerStats?: { [speakerId: number]: { duration: number, segments: number } }\n    }\n}\n\n/**\n * Options for specifying a time range within an audio file.\n */\nexport interface AudioRangeOptions {\n    /** Start time in milliseconds */\n    startTimeMs?: number\n    /** End time in milliseconds */\n    endTimeMs?: number\n}\n\n/**\n * Options for generating a quick preview of audio waveform.\n * This is optimized for UI rendering with a specified number of points.\n */\nexport interface PreviewOptions extends AudioRangeOptions {\n    /** URI of the audio file to analyze */\n    fileUri: string\n    /**\n     * Total number of points to generate for the preview.\n     * @default 100\n     */\n    numberOfPoints?: number\n    /**\n     * Optional logger for debugging.\n     */\n    logger?: ConsoleLike\n    /**\n     * Optional configuration for decoding the audio file.\n     * Defaults to:\n     * - targetSampleRate: undefined (keep original)\n     * - targetChannels: undefined (keep original)\n     * - targetBitDepth: 16\n     * - normalizeAudio: false\n     */\n    decodingOptions?: DecodingConfig\n}\n\n/**\n * Options for mel-spectrogram extraction\n *\n * @experimental This feature is experimental and currently only available on Android.\n * The API may change in future versions.\n */\nexport interface ExtractMelSpectrogramOptions {\n    fileUri?: string // Path to audio file\n    arrayBuffer?: ArrayBuffer // Raw audio buffer\n    windowSizeMs: number // Window size in ms (e.g., 25)\n    hopLengthMs: number // Hop length in ms (e.g., 10)\n    nMels: number // Number of mel filters (e.g., 60)\n    fMin?: number // Min frequency (default: 0)\n    fMax?: number // Max frequency (default: sampleRate / 2)\n    windowType?: 'hann' | 'hamming' // Window function (default: 'hann')\n    normalize?: boolean // Mean normalization (default: false)\n    logScale?: boolean // Log scaling of mel energies (default: true)\n    decodingOptions?: DecodingConfig // Audio decoding settings\n    startTimeMs?: number // Optional start time\n    endTimeMs?: number // Optional end time\n    logger?: ConsoleLike\n}\n\n/**\n * Return type for mel spectrogram extraction\n *\n * @experimental This feature is experimental and currently only available on Android.\n * The API may change in future versions.\n */\nexport interface MelSpectrogram {\n    spectrogram: number[][] // 2D array [time][mel]\n    sampleRate: number // Audio sample rate\n    nMels: number // Number of mel filters\n    timeSteps: number // Number of time frames\n    durationMs: number // Audio duration in ms\n}\n"]}

package/build/AudioDeviceManager.d.ts

@@ -0,0 +1,107 @@
+import { AudioDevice, DeviceDisconnectionBehavior, ConsoleLike } from './ExpoAudioStream.types';
+/**
+ * Class that provides a cross-platform API for managing audio input devices
+ */
+export declare class AudioDeviceManager {
+    private eventEmitter;
+    private currentDeviceId;
+    private availableDevices;
+    private deviceChangeListeners;
+    private deviceListeners;
+    private lastRefreshTime;
+    private refreshInProgress;
+    private refreshDebounceMs;
+    private logger?;
+    constructor(options?: {
+        logger?: ConsoleLike;
+    });
+    /**
+     * Initialize the device manager with a logger
+     * @param logger A logger instance that implements the ConsoleLike interface
+     * @returns The manager instance for chaining
+     */
+    initWithLogger(logger: ConsoleLike): AudioDeviceManager;
+    /**
+     * Set the logger instance
+     * @param logger A logger instance that implements the ConsoleLike interface
+     */
+    setLogger(logger: ConsoleLike): void;
+    /**
+     * Get all available audio input devices
+     * @param options Optional settings to force refresh the device list. Can include a refresh flag.
+     * @returns Promise resolving to an array of audio devices conforming to AudioDevice interface
+     */
+    getAvailableDevices(options?: {
+        refresh?: boolean;
+    }): Promise<AudioDevice[]>;
+    /**
+     * Get the currently selected audio input device
+     * @returns Promise resolving to the current device (conforming to AudioDevice) or null
+     */
+    getCurrentDevice(): Promise<AudioDevice | null>;
+    /**
+     * Select a specific audio input device for recording
+     * @param deviceId The ID of the device to select
+     * @returns Promise resolving to a boolean indicating success
+     */
+    selectDevice(deviceId: string): Promise<boolean>;
+    /**
+     * Reset to the default audio input device
+     * @returns Promise resolving to a boolean indicating success
+     */
+    resetToDefaultDevice(): Promise<boolean>;
+    /**
+     * Register a listener for device changes
+     * @param listener Function to call when devices change (receives AudioDevice[])
+     * @returns Function to remove the listener
+     */
+    addDeviceChangeListener(listener: (devices: AudioDevice[]) => void): () => void;
+    /**
+     * Refresh the list of available devices with debouncing and notify listeners.
+     * @returns Promise resolving to the updated device list (AudioDevice[])
+     */
+    refreshDevices(): Promise<AudioDevice[]>;
+    /**
+     * Get audio input devices using the Web Audio API
+     * @returns Promise resolving to an array of AudioDevice objects
+     */
+    private getWebAudioDevices;
+    /**
+     * Check the current microphone permission status
+     * @returns Permission state ('prompt', 'granted', or 'denied')
+     */
+    private checkMicrophonePermission;
+    /**
+     * Setup listener for device changes in web environment
+     */
+    private setupWebDeviceChangeListener;
+    /**
+     * Check if the current browser is Safari or iOS WebKit
+     */
+    private isSafariOrIOS;
+    /**
+     * Create enhanced device information for Safari and privacy-restricted browsers
+     * @param devices Array of AudioDevice objects, potentially unlabeled
+     * @returns Array of enhanced AudioDevice objects
+     */
+    private enhanceDevicesForSafari;
+    /**
+     * Map a Web MediaDeviceInfo to our AudioDevice format
+     * @param device The MediaDeviceInfo object from the browser
+     * @returns An object conforming to the AudioDevice interface
+     */
+    private mapWebDeviceToAudioDevice;
+    /**
+     * Try to infer the device type from its name
+     * @param deviceName The label of the device
+     * @returns A string representing the inferred device type
+     */
+    private inferDeviceType;
+    /**
+     * Notify all registered listeners about device changes.
+     */
+    private notifyListeners;
+}
+export declare const audioDeviceManager: AudioDeviceManager;
+export { DeviceDisconnectionBehavior };
+//# sourceMappingURL=AudioDeviceManager.d.ts.map

package/build/AudioDeviceManager.d.ts.map

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