@siteed/expo-audio-stream 2.0.0 → 2.1.0

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

@@ -4,6 +4,7 @@ object Constants {
     const val AUDIO_EVENT_NAME = "AudioData"
     const val AUDIO_ANALYSIS_EVENT_NAME = "AudioAnalysis"
     const val RECORDING_INTERRUPTED_EVENT_NAME = "onRecordingInterrupted"
+    const val TRIM_PROGRESS_EVENT = "TrimProgress"
     const val DEFAULT_SAMPLE_RATE = 16000 // Default sample rate for audio recording
     const val DEFAULT_CHANNEL_CONFIG = 1 // Mono
     const val DEFAULT_AUDIO_FORMAT = 16 // 16-bit PCM

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

@@ -2,8 +2,6 @@
 package net.siteed.audiostream
 
 import android.Manifest
-import android.app.ActivityManager
-import android.content.Context
 import android.os.Build
 import android.os.Bundle
 import android.util.Log
@@ -19,6 +17,17 @@ class ExpoAudioStreamModule : Module(), EventSender {
     private lateinit var audioRecorderManager: AudioRecorderManager
     private lateinit var audioProcessor: AudioProcessor
 
+    private val audioFileHandler by lazy { 
+        AudioFileHandler(appContext.reactContext?.filesDir ?: throw IllegalStateException("React context not available")) 
+    }
+    
+    private val audioTrimmer by lazy { 
+        AudioTrimmer(
+            appContext.reactContext ?: throw IllegalStateException("React context not available"), 
+            audioFileHandler
+        ) 
+    }
+
     @RequiresApi(Build.VERSION_CODES.R)
     override fun definition() = ModuleDefinition {
         // The module will be accessible from `requireNativeModule('ExpoAudioStream')` in JavaScript.
@@ -27,7 +36,8 @@ class ExpoAudioStreamModule : Module(), EventSender {
         Events(
             Constants.AUDIO_EVENT_NAME,
             Constants.AUDIO_ANALYSIS_EVENT_NAME,
-            Constants.RECORDING_INTERRUPTED_EVENT_NAME
+            Constants.RECORDING_INTERRUPTED_EVENT_NAME,
+            Constants.TRIM_PROGRESS_EVENT
         )
 
         // Initialize AudioRecorderManager
@@ -241,52 +251,276 @@ class ExpoAudioStreamModule : Module(), EventSender {
 
         AsyncFunction("trimAudio") { options: Map<String, Any>, promise: Promise ->
             try {
-                val fileUri = requireNotNull(options["fileUri"] as? String) { "fileUri is required" }
-                val startTimeMs = requireNotNull(options["startTimeMs"] as? Number)?.toLong() 
-                    ?: throw IllegalArgumentException("startTimeMs is required")
-                val endTimeMs = requireNotNull(options["endTimeMs"] as? Number)?.toLong()
-                    ?: throw IllegalArgumentException("endTimeMs is required")
+                val fileUri = options["fileUri"] as? String ?: run {
+                    promise.reject("INVALID_URI", "fileUri is required", null)
+                    return@AsyncFunction
+                }
+
+                Log.d(Constants.TAG, "trimAudio called with fileUri: $fileUri")
+                Log.d(Constants.TAG, "Full options: $options")
+
+                val mode = options["mode"] as? String ?: "single"
+                val startTimeMs = (options["startTimeMs"] as? Number)?.toLong()
+                val endTimeMs = (options["endTimeMs"] as? Number)?.toLong()
+                
+                @Suppress("UNCHECKED_CAST")
+                val ranges = options["ranges"] as? List<Map<String, Long>>
+                
                 val outputFileName = options["outputFileName"] as? String
                 
-                // Get decoding options
-                val decodingOptionsMap = options["decodingOptions"] as? Map<String, Any>
-                val decodingConfig = if (decodingOptionsMap != null) {
-                    DecodingConfig(
-                        targetSampleRate = decodingOptionsMap["targetSampleRate"] as? Int,
-                        targetChannels = decodingOptionsMap["targetChannels"] as? Int,
-                        targetBitDepth = (decodingOptionsMap["targetBitDepth"] as? Int) ?: 16,
-                        normalizeAudio = (decodingOptionsMap["normalizeAudio"] as? Boolean) ?: false
-                    )
-                } else null
+                @Suppress("UNCHECKED_CAST")
+                var outputFormatMap = options["outputFormat"] as? Map<String, Any>
+                
+                // Validate output format if provided
+                if (outputFormatMap != null) {
+                    val format = outputFormatMap["format"] as? String
+                    if (format != null && format != "wav" && format != "aac" && format != "opus") {
+                        Log.w(Constants.TAG, "Requested format '$format' is not fully supported. Using 'aac' instead.")
+                        // Create a new map with the corrected format
+                        val newOutputFormat = HashMap<String, Any>(outputFormatMap)
+                        newOutputFormat["format"] = "aac"
+                        outputFormatMap = newOutputFormat
+                    }
+                }
+                
+                Log.d(Constants.TAG, "Output format options: $outputFormatMap")
+                
+                // Create progress listener
+                val progressListener = object : AudioTrimmer.ProgressListener {
+                    override fun onProgress(progress: Float, bytesProcessed: Long, totalBytes: Long) {
+                        sendEvent(Constants.TRIM_PROGRESS_EVENT, mapOf(
+                            "progress" to progress,
+                            "bytesProcessed" to bytesProcessed,
+                            "totalBytes" to totalBytes
+                        ))
+                    }
+                }
 
-                Log.d(Constants.TAG, """
-                    Trimming audio with params:
-                    - fileUri: $fileUri
-                    - startTimeMs: $startTimeMs
-                    - endTimeMs: $endTimeMs
-                    - outputFileName: ${outputFileName ?: "auto-generated"}
-                """.trimIndent())
+                // Record start time
+                val startTime = System.currentTimeMillis()
 
-                val trimmedAudio = audioProcessor.trimAudio(
+                // Perform the trim operation
+                val result = audioTrimmer.trimAudio(
                     fileUri = fileUri,
+                    mode = mode,
                     startTimeMs = startTimeMs,
                     endTimeMs = endTimeMs,
-                    config = decodingConfig,
-                    outputFileName = outputFileName
-                ) ?: throw IllegalStateException("Failed to trim audio")
-
-                // Create a map with the available data
-                val resultMap = mapOf<String, Any>(
-                    "sampleRate" to trimmedAudio.sampleRate,
-                    "channels" to trimmedAudio.channels,
-                    "bitDepth" to trimmedAudio.bitDepth,
-                    "dataSize" to trimmedAudio.data.size
+                    ranges = ranges,
+                    outputFileName = outputFileName,
+                    outputFormat = outputFormatMap,
+                    progressListener = progressListener
                 )
 
-                promise.resolve(resultMap)
+                // Calculate processing time
+                val processingTimeMs = System.currentTimeMillis() - startTime
+                
+                // Add processing time to result
+                val resultWithProcessingTime = result.toMutableMap()
+                resultWithProcessingTime["processingInfo"] = mapOf(
+                    "durationMs" to processingTimeMs
+                )
+
+                Log.d(Constants.TAG, "Trim operation completed successfully in ${processingTimeMs}ms: $result")
+                promise.resolve(resultWithProcessingTime)
             } catch (e: Exception) {
-                Log.e(Constants.TAG, "Failed to trim audio: ${e.message}", e)
-                promise.reject("TRIM_ERROR", e.message ?: "Unknown error", e)
+                Log.e(Constants.TAG, "Error trimming audio: ${e.message}", e)
+                promise.reject("TRIM_ERROR", "Error trimming audio: ${e.message}", e)
+            }
+        }
+
+        AsyncFunction("extractMelSpectrogram") { options: Map<String, Any>, promise: Promise ->
+            try {
+                // Log all incoming options for debugging
+                Log.d(Constants.TAG, "extractMelSpectrogram called with options: $options")
+                
+                // Extract required parameters with detailed logging
+                val fileUri = options["fileUri"] as? String
+                Log.d(Constants.TAG, "fileUri: $fileUri")
+                if (fileUri == null) {
+                    Log.e(Constants.TAG, "Missing required parameter: fileUri")
+                    throw IllegalArgumentException("fileUri is required")
+                }
+                
+                val windowSizeMs = options["windowSizeMs"] as? Double
+                Log.d(Constants.TAG, "windowSizeMs: $windowSizeMs")
+                if (windowSizeMs == null) {
+                    Log.e(Constants.TAG, "Missing required parameter: windowSizeMs")
+                    throw IllegalArgumentException("windowSizeMs is required")
+                }
+                
+                val hopLengthMs = options["hopLengthMs"] as? Double
+                Log.d(Constants.TAG, "hopLengthMs: $hopLengthMs")
+                if (hopLengthMs == null) {
+                    Log.e(Constants.TAG, "Missing required parameter: hopLengthMs")
+                    throw IllegalArgumentException("hopLengthMs is required")
+                }
+                
+                // Handle nMels which might come as Double from JavaScript
+                val nMelsValue = options["nMels"]
+                Log.d(Constants.TAG, "Raw nMels value: $nMelsValue (type: ${nMelsValue?.javaClass?.name})")
+                
+                val nMels = when (nMelsValue) {
+                    is Int -> nMelsValue
+                    is Double -> nMelsValue.toInt()
+                    is Number -> nMelsValue.toInt()
+                    else -> {
+                        Log.e(Constants.TAG, "Missing or invalid required parameter: nMels")
+                        throw IllegalArgumentException("nMels is required and must be a number")
+                    }
+                }
+                
+                Log.d(Constants.TAG, "Converted nMels: $nMels (from ${nMelsValue?.javaClass?.name})")
+
+                // Extract optional parameters with defaults
+                val fMin = options["fMin"] as? Double ?: 0.0
+                val fMax = options["fMax"] as? Double
+                val windowType = options["windowType"] as? String ?: "hann"
+                val normalize = options["normalize"] as? Boolean ?: false
+                val logScale = options["logScale"] as? Boolean ?: true
+                
+                // Fix the conversion from Number to Long to preserve decimal values
+                val startTimeMsNumber = options["startTimeMs"] as? Number
+                val endTimeMsNumber = options["endTimeMs"] as? Number
+                val startTimeMs = startTimeMsNumber?.toLong() ?: startTimeMsNumber?.toDouble()?.toLong()
+                val endTimeMs = endTimeMsNumber?.toLong() ?: endTimeMsNumber?.toDouble()?.toLong()
+
+                Log.d(Constants.TAG, """
+                    Optional parameters:
+                    - fMin: $fMin
+                    - fMax: $fMax
+                    - windowType: $windowType
+                    - normalize: $normalize
+                    - logScale: $logScale
+                    - startTimeMs: $startTimeMs (original: $startTimeMsNumber)
+                    - endTimeMs: $endTimeMs (original: $endTimeMsNumber)
+                """.trimIndent())
+
+                // Handle decoding options
+                val decodingOptions = options["decodingOptions"] as? Map<String, Any>
+                Log.d(Constants.TAG, "Decoding options: $decodingOptions")
+                
+                val config = decodingOptions?.let {
+                    val targetSampleRateValue = it["targetSampleRate"]
+                    val targetSampleRate = when (targetSampleRateValue) {
+                        is Int -> targetSampleRateValue
+                        is Double -> targetSampleRateValue.toInt()
+                        is Number -> targetSampleRateValue.toInt()
+                        else -> null
+                    }
+                    
+                    val targetChannelsValue = it["targetChannels"]
+                    val targetChannels = when (targetChannelsValue) {
+                        is Int -> targetChannelsValue
+                        is Double -> targetChannelsValue.toInt()
+                        is Number -> targetChannelsValue.toInt()
+                        else -> 1
+                    }
+                    
+                    val targetBitDepthValue = it["targetBitDepth"]
+                    val targetBitDepth = when (targetBitDepthValue) {
+                        is Int -> targetBitDepthValue
+                        is Double -> targetBitDepthValue.toInt()
+                        is Number -> targetBitDepthValue.toInt()
+                        else -> 16
+                    }
+                    
+                    val normalizeAudio = it["normalizeAudio"] as? Boolean ?: false
+                    
+                    DecodingConfig(
+                        targetSampleRate = targetSampleRate,
+                        targetChannels = targetChannels,
+                        targetBitDepth = targetBitDepth,
+                        normalizeAudio = normalizeAudio
+                    ).also { config ->
+                        Log.d(Constants.TAG, """
+                            Using decoding config:
+                            - targetSampleRate: ${config.targetSampleRate ?: "original"}
+                            - targetChannels: ${config.targetChannels ?: "original"}
+                            - targetBitDepth: ${config.targetBitDepth}
+                            - normalizeAudio: ${config.normalizeAudio}
+                        """.trimIndent())
+                    }
+                } ?: DecodingConfig(targetSampleRate = null, targetChannels = 1, targetBitDepth = 16).also {
+                    Log.d(Constants.TAG, "Using default decoding config")
+                }
+
+                // Check if the audio data is too short
+                if (startTimeMs != null && endTimeMs != null) {
+                    val durationMs = endTimeMs - startTimeMs
+                    Log.d(Constants.TAG, "Audio duration for spectrogram: $durationMs ms")
+                    if (durationMs < 25) {  // 25ms is minimum for a single window
+                        Log.w(Constants.TAG, "Audio duration is too short for spectrogram analysis: $durationMs ms")
+                        throw IllegalArgumentException("Audio duration must be at least 25ms for spectrogram analysis")
+                    }
+                }
+
+                // Load audio data with optional time range
+                Log.d(Constants.TAG, "Loading audio data...")
+                val audioData = when {
+                    startTimeMs != null && endTimeMs != null -> {
+                        Log.d(Constants.TAG, "Loading audio range: $startTimeMs to $endTimeMs ms")
+                        audioProcessor.loadAudioRange(fileUri, startTimeMs, endTimeMs, config)
+                    }
+                    else -> {
+                        Log.d(Constants.TAG, "Loading entire audio file")
+                        audioProcessor.loadAudioFromAnyFormat(fileUri, config)
+                    }
+                }
+                
+                if (audioData == null) {
+                    Log.e(Constants.TAG, "Failed to load audio data")
+                    throw IllegalStateException("Failed to load audio data")
+                }
+                
+                Log.d(Constants.TAG, """
+                    Audio data loaded successfully:
+                    - data size: ${audioData.data.size} bytes
+                    - sampleRate: ${audioData.sampleRate}
+                    - channels: ${audioData.channels}
+                    - bitDepth: ${audioData.bitDepth}
+                    - durationMs: ${audioData.durationMs}
+                """.trimIndent())
+
+                // Validate that we have enough audio data for processing
+                if (audioData.data.size == 0 || audioData.durationMs < windowSizeMs) {
+                    Log.e(Constants.TAG, "Audio data is too short for spectrogram analysis: ${audioData.durationMs}ms, data size: ${audioData.data.size} bytes")
+                    throw IllegalArgumentException(
+                        "Audio data is too short for spectrogram analysis. " +
+                        "Duration: ${audioData.durationMs}ms, minimum required: ${windowSizeMs}ms"
+                    )
+                }
+
+                // Compute mel-spectrogram
+                Log.d(Constants.TAG, "Computing mel-spectrogram...")
+                val spectrogramData = audioProcessor.extractMelSpectrogram(
+                    audioData = audioData,
+                    windowSizeMs = windowSizeMs.toFloat(),
+                    hopLengthMs = hopLengthMs.toFloat(),
+                    nMels = nMels,
+                    fMin = fMin.toFloat(),
+                    fMax = fMax?.toFloat() ?: (audioData.sampleRate.toFloat() / 2),
+                    normalize = normalize,
+                    logScaling = logScale,
+                    windowType = windowType
+                )
+                
+                Log.d(Constants.TAG, "Mel-spectrogram computed successfully with ${spectrogramData.spectrogram.size} time steps")
+
+                // Convert to map for React Native
+                val result = mapOf(
+                    "spectrogram" to spectrogramData.spectrogram.map { it.toList() },
+                    "sampleRate" to audioData.sampleRate,
+                    "nMels" to nMels,
+                    "timeSteps" to spectrogramData.spectrogram.size,
+                    "durationMs" to audioData.durationMs
+                )
+                
+                Log.d(Constants.TAG, "Returning result with ${result["timeSteps"]} time steps and $nMels mel bands")
+                promise.resolve(result)
+            } catch (e: Exception) {
+                Log.e(Constants.TAG, "Failed to extract mel-spectrogram: ${e.message}")
+                Log.e(Constants.TAG, "Stack trace: ${e.stackTraceToString()}")
+                promise.reject("SPECTROGRAM_ERROR", e.message ?: "Unknown error", e)
             }
         }
 
@@ -296,11 +530,7 @@ class ExpoAudioStreamModule : Module(), EventSender {
 
         // Add a new function to check if recording is actually running
         AsyncFunction("checkRecordingStatus") { promise: Promise ->
-            val isServiceRunning = AudioRecordingService::class.java.name.let { className ->
-                val manager = appContext.reactContext?.getSystemService(Context.ACTIVITY_SERVICE) as? ActivityManager
-                manager?.getRunningServices(Integer.MAX_VALUE)
-                    ?.any { it.service.className == className }
-            } ?: false
+            val isServiceRunning = AudioRecordingService.isServiceRunning()
 
             val status = audioRecorderManager.getStatus()
             

package/build/AudioAnalysis/AudioAnalysis.types.d.ts

@@ -141,4 +141,39 @@ export interface PreviewOptions extends AudioRangeOptions {
      */
     decodingOptions?: DecodingConfig;
 }
+/**
+ * Options for mel-spectrogram extraction
+ *
+ * @experimental This feature is experimental and currently only available on Android.
+ * The API may change in future versions.
+ */
+export interface ExtractMelSpectrogramOptions {
+    fileUri?: string;
+    arrayBuffer?: ArrayBuffer;
+    windowSizeMs: number;
+    hopLengthMs: number;
+    nMels: number;
+    fMin?: number;
+    fMax?: number;
+    windowType?: 'hann' | 'hamming';
+    normalize?: boolean;
+    logScale?: boolean;
+    decodingOptions?: DecodingConfig;
+    startTimeMs?: number;
+    endTimeMs?: number;
+    logger?: ConsoleLike;
+}
+/**
+ * Return type for mel spectrogram extraction
+ *
+ * @experimental This feature is experimental and currently only available on Android.
+ * The API may change in future versions.
+ */
+export interface MelSpectrogram {
+    spectrogram: number[][];
+    sampleRate: number;
+    nMels: number;
+    timeSteps: number;
+    durationMs: number;
+}
 //# sourceMappingURL=AudioAnalysis.types.d.ts.map

package/build/AudioAnalysis/AudioAnalysis.types.d.ts.map

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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 // Always computed\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"]}
+{"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/AudioAnalysis/extractAudioAnalysis.d.ts

@@ -1,5 +1,5 @@
-import { ConsoleLike, ExtractAudioDataOptions } from '../ExpoAudioStream.types';
-import { AudioAnalysis, AudioFeaturesOptions, DecodingConfig, PreviewOptions } from './AudioAnalysis.types';
+import { ConsoleLike } from '../ExpoAudioStream.types';
+import { AudioAnalysis, AudioFeaturesOptions, DecodingConfig } from './AudioAnalysis.types';
 import { WavFileInfo } from '../utils/getWavFileInfo';
 export interface ExtractWavAudioAnalysisProps {
     fileUri?: string;
@@ -64,15 +64,5 @@ export declare function extractAudioAnalysis(props: ExtractAudioAnalysisProps):
  * @returns A promise that resolves to the audio analysis data.
  */
 export declare const extractRawWavAnalysis: ({ fileUri, segmentDurationMs, arrayBuffer, bitDepth, durationMs, sampleRate, numberOfChannels, features, logger, position, length, }: ExtractWavAudioAnalysisProps) => Promise<AudioAnalysis>;
-/**
- * Generates a simplified preview of the audio waveform for quick visualization.
- * Ideal for UI rendering with a specified number of points.
- *
- * @param options - The options for the preview, including file URI and time range.
- * @returns A promise that resolves to the audio preview data.
- */
-export declare function extractPreview({ fileUri, numberOfPoints, startTimeMs, endTimeMs, // First 30 seconds
-decodingOptions, logger, }: PreviewOptions): Promise<AudioAnalysis>;
-export declare const extractAudioData: (props: ExtractAudioDataOptions) => Promise<any>;
 export {};
 //# sourceMappingURL=extractAudioAnalysis.d.ts.map

package/build/AudioAnalysis/extractAudioAnalysis.d.ts.map

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package/build/AudioAnalysis/extractAudioAnalysis.js

@@ -200,30 +200,4 @@ arrayBuffer, bitDepth, durationMs, sampleRate, numberOfChannels, features, logge
         return res;
     }
 };
-/**
- * Generates a simplified preview of the audio waveform for quick visualization.
- * Ideal for UI rendering with a specified number of points.
- *
- * @param options - The options for the preview, including file URI and time range.
- * @returns A promise that resolves to the audio preview data.
- */
-export async function extractPreview({ fileUri, numberOfPoints = 100, startTimeMs = 0, endTimeMs = 30000, // First 30 seconds
-decodingOptions, logger, }) {
-    const durationMs = endTimeMs - startTimeMs;
-    const segmentDurationMs = Math.floor(durationMs / numberOfPoints);
-    // Call extractAudioAnalysis with calculated parameters
-    const analysis = await extractAudioAnalysis({
-        fileUri,
-        startTimeMs,
-        endTimeMs,
-        logger,
-        segmentDurationMs,
-        decodingOptions,
-    });
-    // Transform the result into AudioPreview format
-    return analysis;
-}
-export const extractAudioData = async (props) => {
-    return await ExpoAudioStreamModule.extractAudioData(props);
-};
 //# sourceMappingURL=extractAudioAnalysis.js.map

package/build/AudioAnalysis/extractAudioAnalysis.js.map

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packages/expo-audio-stream/src/AudioAnalysis/extractAudioAnalysis.ts\n/**\n * This module provides functions for extracting and analyzing audio data.\n * - `extractAudioAnalysis`: For detailed analysis with customizable ranges and decoding options.\n * - `extractWavAudioAnalysis`: For analyzing WAV files without decoding, preserving original PCM values.\n * - `extractPreview`: For generating quick previews of audio waveforms, optimized for UI rendering.\n */\nimport crc32 from 'crc-32'\n\nimport { ConsoleLike, ExtractAudioDataOptions } from '../ExpoAudioStream.types'\nimport ExpoAudioStreamModule from '../ExpoAudioStreamModule'\nimport { isWeb } from '../constants'\nimport {\n    AudioAnalysis,\n    AudioFeaturesOptions,\n    DataPoint,\n    DecodingConfig,\n    PreviewOptions,\n} from './AudioAnalysis.types'\nimport { processAudioBuffer } from '../utils/audioProcessing'\nimport { convertPCMToFloat32 } from '../utils/convertPCMToFloat32'\nimport { getWavFileInfo, WavFileInfo } from '../utils/getWavFileInfo'\nimport { InlineFeaturesExtractor } from '../workers/InlineFeaturesExtractor.web'\n\nfunction calculateCRC32ForDataPoint(data: Float32Array): number {\n    // Convert float array to byte array for CRC32\n    const byteArray = new Uint8Array(data.length * 4)\n    const dataView = new DataView(byteArray.buffer)\n\n    for (let i = 0; i < data.length; i++) {\n        dataView.setFloat32(i * 4, data[i], true)\n    }\n\n    return crc32.buf(byteArray)\n}\n\nexport interface ExtractWavAudioAnalysisProps {\n    fileUri?: string // should provide either fileUri or arrayBuffer\n    wavMetadata?: WavFileInfo\n    arrayBuffer?: ArrayBuffer\n    bitDepth?: number\n    durationMs?: number\n    sampleRate?: number\n    numberOfChannels?: number\n    position?: number // Optional number of bytes to skip. Default is 0\n    length?: number // Optional number of bytes to read.\n    segmentDurationMs?: number // Optional number of points per second. Use to reduce the number of points and compute the number of datapoints to return.\n    features?: AudioFeaturesOptions\n    featuresExtratorUrl?: string\n    logger?: ConsoleLike\n    decodingOptions?: DecodingConfig\n}\n\n// Define base options interface with common properties\ninterface BaseExtractOptions {\n    fileUri?: string\n    arrayBuffer?: ArrayBuffer\n    /**\n     * Duration of each analysis segment in milliseconds. Defaults to 100ms if not specified.\n     */\n    segmentDurationMs?: number\n    features?: AudioFeaturesOptions\n    decodingOptions?: DecodingConfig\n    logger?: ConsoleLike\n}\n\n// Time-based range options\ninterface TimeRangeOptions extends BaseExtractOptions {\n    startTimeMs?: number\n    endTimeMs?: number\n    position?: never\n    length?: never\n}\n\n// Byte-based range options\ninterface ByteRangeOptions extends BaseExtractOptions {\n    position?: number\n    length?: number\n    startTimeMs?: never\n    endTimeMs?: never\n}\n\n/**\n * Options for extracting audio analysis.\n * - For time-based analysis, provide `startTimeMs` and `endTimeMs`.\n * - For byte-based analysis, provide `position` and `length`.\n * - Do not mix time and byte ranges.\n */\nexport type ExtractAudioAnalysisProps = TimeRangeOptions | ByteRangeOptions\n\n/**\n * Extracts detailed audio analysis from the specified audio file or buffer.\n * Supports either time-based or byte-based ranges for flexibility in analysis.\n *\n * @param props - The options for extraction, including file URI, ranges, and decoding settings.\n * @returns A promise that resolves to the audio analysis data.\n * @throws {Error} If both time and byte ranges are provided or if required parameters are missing.\n */\nexport async function extractAudioAnalysis(\n    props: ExtractAudioAnalysisProps\n): Promise<AudioAnalysis> {\n    const {\n        fileUri,\n        arrayBuffer,\n        decodingOptions,\n        logger,\n        segmentDurationMs = 100,\n        features,\n    } = props\n\n    if (isWeb) {\n        try {\n            // Create AudioContext here\n            const audioContext = new (window.AudioContext ||\n                (window as any).webkitAudioContext)({\n                sampleRate: decodingOptions?.targetSampleRate ?? 16000,\n            })\n\n            try {\n                const processedBuffer = await processAudioBuffer({\n                    arrayBuffer,\n                    fileUri,\n                    targetSampleRate:\n                        decodingOptions?.targetSampleRate ?? 16000,\n                    targetChannels: decodingOptions?.targetChannels ?? 1,\n                    normalizeAudio: decodingOptions?.normalizeAudio ?? false,\n                    startTimeMs:\n                        'startTimeMs' in props ? props.startTimeMs : undefined,\n                    endTimeMs:\n                        'endTimeMs' in props ? props.endTimeMs : undefined,\n                    position: 'position' in props ? props.position : undefined,\n                    length: 'length' in props ? props.length : undefined,\n                    audioContext, // Pass the context we created\n                    logger,\n                })\n\n                const channelData = processedBuffer.buffer.getChannelData(0)\n\n                // Create and initialize the worker\n                const blob = new Blob([InlineFeaturesExtractor], {\n                    type: 'application/javascript',\n                })\n                const workerUrl = URL.createObjectURL(blob)\n                const worker = new Worker(workerUrl)\n\n                return new Promise((resolve, reject) => {\n                    worker.onmessage = (event) => {\n                        if (event.data.error) {\n                            reject(new Error(event.data.error))\n                            return\n                        }\n\n                        const result: AudioAnalysis = event.data.result\n                        // Calculate CRC32 after worker completes if requested\n                        if (features?.crc32) {\n                            const samplesPerSegment = Math.floor(\n                                (processedBuffer.sampleRate *\n                                    segmentDurationMs) /\n                                    1000\n                            )\n\n                            result.dataPoints = result.dataPoints.map(\n                                (point: DataPoint, index: number) => {\n                                    const startSample =\n                                        index * samplesPerSegment\n                                    const segmentData = channelData.slice(\n                                        startSample,\n                                        startSample + samplesPerSegment\n                                    )\n\n                                    return {\n                                        ...point,\n                                        features: {\n                                            ...point.features,\n                                            crc32: calculateCRC32ForDataPoint(\n                                                segmentData\n                                            ),\n                                        },\n                                    }\n                                }\n                            )\n                        }\n\n                        URL.revokeObjectURL(workerUrl)\n                        worker.terminate()\n                        resolve(result)\n                    }\n\n                    worker.onerror = (error) => {\n                        URL.revokeObjectURL(workerUrl)\n                        worker.terminate()\n                        reject(error)\n                    }\n\n                    worker.postMessage({\n                        channelData,\n                        sampleRate: processedBuffer.sampleRate,\n                        segmentDurationMs,\n                        bitDepth: decodingOptions?.targetBitDepth ?? 32,\n                        numberOfChannels: processedBuffer.channels,\n                        // enableLogging: !!logger,\n                        features,\n                    })\n                })\n            } finally {\n                await audioContext.close()\n            }\n        } catch (error) {\n            logger?.error('Failed to process audio:', error)\n            throw error\n        }\n    } else {\n        return await ExpoAudioStreamModule.extractAudioAnalysis(props)\n    }\n}\n\n/**\n * Analyzes WAV files without decoding, preserving original PCM values.\n * Use this function when you need to ensure the analysis matches other software by avoiding any transformations.\n *\n * @param props - The options for WAV analysis, including file URI and range.\n * @returns A promise that resolves to the audio analysis data.\n */\nexport const extractRawWavAnalysis = async ({\n    fileUri,\n    segmentDurationMs = 100, // Default to 100ms\n    arrayBuffer,\n    bitDepth,\n    durationMs,\n    sampleRate,\n    numberOfChannels,\n    features,\n    logger,\n    position = 0,\n    length,\n}: ExtractWavAudioAnalysisProps): Promise<AudioAnalysis> => {\n    if (isWeb) {\n        if (!arrayBuffer && !fileUri) {\n            throw new Error('Either arrayBuffer or fileUri must be provided')\n        }\n\n        if (!arrayBuffer) {\n            logger?.log(`fetching fileUri`, fileUri)\n            const response = await fetch(fileUri!)\n\n            if (!response.ok) {\n                throw new Error(\n                    `Failed to fetch fileUri: ${response.statusText}`\n                )\n            }\n\n            arrayBuffer = await response.arrayBuffer()\n            logger?.log(`fetched fileUri`, arrayBuffer.byteLength, arrayBuffer)\n        }\n\n        // Create a new copy of the ArrayBuffer to avoid detachment issues\n        const bufferCopy = arrayBuffer.slice(0)\n        logger?.log(\n            `extractAudioAnalysis bitDepth=${bitDepth} len=${bufferCopy.byteLength}`,\n            bufferCopy.slice(0, 100)\n        )\n\n        let actualBitDepth = bitDepth\n        if (!actualBitDepth) {\n            logger?.log(\n                `extractAudioAnalysis bitDepth not provided -- getting wav file info`\n            )\n            const fileInfo = await getWavFileInfo(bufferCopy)\n            actualBitDepth = fileInfo.bitDepth\n        }\n        logger?.log(`extractAudioAnalysis actualBitDepth=${actualBitDepth}`)\n\n        const {\n            pcmValues: channelData,\n            min,\n            max,\n        } = await convertPCMToFloat32({\n            buffer: arrayBuffer,\n            bitDepth: actualBitDepth,\n        })\n        logger?.log(\n            `extractAudioAnalysis convertPCMToFloat32 length=${channelData.length} range: [ ${min} :: ${max} ]`\n        )\n\n        // Apply position and length constraints to channelData if specified\n        const startIndex = position\n        const endIndex = length ? startIndex + length : channelData.length\n        const constrainedChannelData = channelData.slice(startIndex, endIndex)\n\n        return new Promise((resolve, reject) => {\n            const blob = new Blob([InlineFeaturesExtractor], {\n                type: 'application/javascript',\n            })\n            const url = URL.createObjectURL(blob)\n            const worker = new Worker(url)\n\n            worker.onmessage = (event) => {\n                resolve(event.data.result)\n            }\n\n            worker.onerror = (error) => {\n                reject(error)\n            }\n\n            worker.postMessage({\n                command: 'process',\n                channelData: constrainedChannelData,\n                sampleRate,\n                segmentDurationMs,\n                logger,\n                bitDepth,\n                fullAudioDurationMs: durationMs,\n                numberOfChannels,\n            })\n        })\n    } else {\n        if (!fileUri) {\n            throw new Error('fileUri is required')\n        }\n        logger?.log(`extractAudioAnalysis`, {\n            fileUri,\n            segmentDurationMs,\n        })\n        const res = await ExpoAudioStreamModule.extractAudioAnalysis({\n            fileUri,\n            segmentDurationMs,\n            features,\n            position,\n            length,\n        })\n        logger?.log(`extractAudioAnalysis`, res)\n        return res\n    }\n}\n\n/**\n * Generates a simplified preview of the audio waveform for quick visualization.\n * Ideal for UI rendering with a specified number of points.\n *\n * @param options - The options for the preview, including file URI and time range.\n * @returns A promise that resolves to the audio preview data.\n */\nexport async function extractPreview({\n    fileUri,\n    numberOfPoints = 100,\n    startTimeMs = 0,\n    endTimeMs = 30000, // First 30 seconds\n    decodingOptions,\n    logger,\n}: PreviewOptions): Promise<AudioAnalysis> {\n    const durationMs = endTimeMs - startTimeMs\n    const segmentDurationMs = Math.floor(durationMs / numberOfPoints)\n\n    // Call extractAudioAnalysis with calculated parameters\n    const analysis = await extractAudioAnalysis({\n        fileUri,\n        startTimeMs,\n        endTimeMs,\n        logger,\n        segmentDurationMs,\n        decodingOptions,\n    })\n\n    // Transform the result into AudioPreview format\n    return analysis\n}\n\nexport const extractAudioData = async (props: ExtractAudioDataOptions) => {\n    return await ExpoAudioStreamModule.extractAudioData(props)\n}"]}
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packages/expo-audio-stream/src/AudioAnalysis/extractAudioAnalysis.ts\n/**\n * This module provides functions for extracting and analyzing audio data.\n * - `extractAudioAnalysis`: For detailed analysis with customizable ranges and decoding options.\n * - `extractWavAudioAnalysis`: For analyzing WAV files without decoding, preserving original PCM values.\n * - `extractPreview`: For generating quick previews of audio waveforms, optimized for UI rendering.\n */\nimport crc32 from 'crc-32'\n\nimport { ConsoleLike } from '../ExpoAudioStream.types'\nimport ExpoAudioStreamModule from '../ExpoAudioStreamModule'\nimport { isWeb } from '../constants'\nimport {\n    AudioAnalysis,\n    AudioFeaturesOptions,\n    DataPoint,\n    DecodingConfig,\n} from './AudioAnalysis.types'\nimport { processAudioBuffer } from '../utils/audioProcessing'\nimport { convertPCMToFloat32 } from '../utils/convertPCMToFloat32'\nimport { getWavFileInfo, WavFileInfo } from '../utils/getWavFileInfo'\nimport { InlineFeaturesExtractor } from '../workers/InlineFeaturesExtractor.web'\n\nfunction calculateCRC32ForDataPoint(data: Float32Array): number {\n    // Convert float array to byte array for CRC32\n    const byteArray = new Uint8Array(data.length * 4)\n    const dataView = new DataView(byteArray.buffer)\n\n    for (let i = 0; i < data.length; i++) {\n        dataView.setFloat32(i * 4, data[i], true)\n    }\n\n    return crc32.buf(byteArray)\n}\n\nexport interface ExtractWavAudioAnalysisProps {\n    fileUri?: string // should provide either fileUri or arrayBuffer\n    wavMetadata?: WavFileInfo\n    arrayBuffer?: ArrayBuffer\n    bitDepth?: number\n    durationMs?: number\n    sampleRate?: number\n    numberOfChannels?: number\n    position?: number // Optional number of bytes to skip. Default is 0\n    length?: number // Optional number of bytes to read.\n    segmentDurationMs?: number // Optional number of points per second. Use to reduce the number of points and compute the number of datapoints to return.\n    features?: AudioFeaturesOptions\n    featuresExtratorUrl?: string\n    logger?: ConsoleLike\n    decodingOptions?: DecodingConfig\n}\n\n// Define base options interface with common properties\ninterface BaseExtractOptions {\n    fileUri?: string\n    arrayBuffer?: ArrayBuffer\n    /**\n     * Duration of each analysis segment in milliseconds. Defaults to 100ms if not specified.\n     */\n    segmentDurationMs?: number\n    features?: AudioFeaturesOptions\n    decodingOptions?: DecodingConfig\n    logger?: ConsoleLike\n}\n\n// Time-based range options\ninterface TimeRangeOptions extends BaseExtractOptions {\n    startTimeMs?: number\n    endTimeMs?: number\n    position?: never\n    length?: never\n}\n\n// Byte-based range options\ninterface ByteRangeOptions extends BaseExtractOptions {\n    position?: number\n    length?: number\n    startTimeMs?: never\n    endTimeMs?: never\n}\n\n/**\n * Options for extracting audio analysis.\n * - For time-based analysis, provide `startTimeMs` and `endTimeMs`.\n * - For byte-based analysis, provide `position` and `length`.\n * - Do not mix time and byte ranges.\n */\nexport type ExtractAudioAnalysisProps = TimeRangeOptions | ByteRangeOptions\n\n/**\n * Extracts detailed audio analysis from the specified audio file or buffer.\n * Supports either time-based or byte-based ranges for flexibility in analysis.\n *\n * @param props - The options for extraction, including file URI, ranges, and decoding settings.\n * @returns A promise that resolves to the audio analysis data.\n * @throws {Error} If both time and byte ranges are provided or if required parameters are missing.\n */\nexport async function extractAudioAnalysis(\n    props: ExtractAudioAnalysisProps\n): Promise<AudioAnalysis> {\n    const {\n        fileUri,\n        arrayBuffer,\n        decodingOptions,\n        logger,\n        segmentDurationMs = 100,\n        features,\n    } = props\n\n    if (isWeb) {\n        try {\n            // Create AudioContext here\n            const audioContext = new (window.AudioContext ||\n                (window as any).webkitAudioContext)({\n                sampleRate: decodingOptions?.targetSampleRate ?? 16000,\n            })\n\n            try {\n                const processedBuffer = await processAudioBuffer({\n                    arrayBuffer,\n                    fileUri,\n                    targetSampleRate:\n                        decodingOptions?.targetSampleRate ?? 16000,\n                    targetChannels: decodingOptions?.targetChannels ?? 1,\n                    normalizeAudio: decodingOptions?.normalizeAudio ?? false,\n                    startTimeMs:\n                        'startTimeMs' in props ? props.startTimeMs : undefined,\n                    endTimeMs:\n                        'endTimeMs' in props ? props.endTimeMs : undefined,\n                    position: 'position' in props ? props.position : undefined,\n                    length: 'length' in props ? props.length : undefined,\n                    audioContext, // Pass the context we created\n                    logger,\n                })\n\n                const channelData = processedBuffer.buffer.getChannelData(0)\n\n                // Create and initialize the worker\n                const blob = new Blob([InlineFeaturesExtractor], {\n                    type: 'application/javascript',\n                })\n                const workerUrl = URL.createObjectURL(blob)\n                const worker = new Worker(workerUrl)\n\n                return new Promise((resolve, reject) => {\n                    worker.onmessage = (event) => {\n                        if (event.data.error) {\n                            reject(new Error(event.data.error))\n                            return\n                        }\n\n                        const result: AudioAnalysis = event.data.result\n                        // Calculate CRC32 after worker completes if requested\n                        if (features?.crc32) {\n                            const samplesPerSegment = Math.floor(\n                                (processedBuffer.sampleRate *\n                                    segmentDurationMs) /\n                                    1000\n                            )\n\n                            result.dataPoints = result.dataPoints.map(\n                                (point: DataPoint, index: number) => {\n                                    const startSample =\n                                        index * samplesPerSegment\n                                    const segmentData = channelData.slice(\n                                        startSample,\n                                        startSample + samplesPerSegment\n                                    )\n\n                                    return {\n                                        ...point,\n                                        features: {\n                                            ...point.features,\n                                            crc32: calculateCRC32ForDataPoint(\n                                                segmentData\n                                            ),\n                                        },\n                                    }\n                                }\n                            )\n                        }\n\n                        URL.revokeObjectURL(workerUrl)\n                        worker.terminate()\n                        resolve(result)\n                    }\n\n                    worker.onerror = (error) => {\n                        URL.revokeObjectURL(workerUrl)\n                        worker.terminate()\n                        reject(error)\n                    }\n\n                    worker.postMessage({\n                        channelData,\n                        sampleRate: processedBuffer.sampleRate,\n                        segmentDurationMs,\n                        bitDepth: decodingOptions?.targetBitDepth ?? 32,\n                        numberOfChannels: processedBuffer.channels,\n                        // enableLogging: !!logger,\n                        features,\n                    })\n                })\n            } finally {\n                await audioContext.close()\n            }\n        } catch (error) {\n            logger?.error('Failed to process audio:', error)\n            throw error\n        }\n    } else {\n        return await ExpoAudioStreamModule.extractAudioAnalysis(props)\n    }\n}\n\n/**\n * Analyzes WAV files without decoding, preserving original PCM values.\n * Use this function when you need to ensure the analysis matches other software by avoiding any transformations.\n *\n * @param props - The options for WAV analysis, including file URI and range.\n * @returns A promise that resolves to the audio analysis data.\n */\nexport const extractRawWavAnalysis = async ({\n    fileUri,\n    segmentDurationMs = 100, // Default to 100ms\n    arrayBuffer,\n    bitDepth,\n    durationMs,\n    sampleRate,\n    numberOfChannels,\n    features,\n    logger,\n    position = 0,\n    length,\n}: ExtractWavAudioAnalysisProps): Promise<AudioAnalysis> => {\n    if (isWeb) {\n        if (!arrayBuffer && !fileUri) {\n            throw new Error('Either arrayBuffer or fileUri must be provided')\n        }\n\n        if (!arrayBuffer) {\n            logger?.log(`fetching fileUri`, fileUri)\n            const response = await fetch(fileUri!)\n\n            if (!response.ok) {\n                throw new Error(\n                    `Failed to fetch fileUri: ${response.statusText}`\n                )\n            }\n\n            arrayBuffer = await response.arrayBuffer()\n            logger?.log(`fetched fileUri`, arrayBuffer.byteLength, arrayBuffer)\n        }\n\n        // Create a new copy of the ArrayBuffer to avoid detachment issues\n        const bufferCopy = arrayBuffer.slice(0)\n        logger?.log(\n            `extractAudioAnalysis bitDepth=${bitDepth} len=${bufferCopy.byteLength}`,\n            bufferCopy.slice(0, 100)\n        )\n\n        let actualBitDepth = bitDepth\n        if (!actualBitDepth) {\n            logger?.log(\n                `extractAudioAnalysis bitDepth not provided -- getting wav file info`\n            )\n            const fileInfo = await getWavFileInfo(bufferCopy)\n            actualBitDepth = fileInfo.bitDepth\n        }\n        logger?.log(`extractAudioAnalysis actualBitDepth=${actualBitDepth}`)\n\n        const {\n            pcmValues: channelData,\n            min,\n            max,\n        } = await convertPCMToFloat32({\n            buffer: arrayBuffer,\n            bitDepth: actualBitDepth,\n        })\n        logger?.log(\n            `extractAudioAnalysis convertPCMToFloat32 length=${channelData.length} range: [ ${min} :: ${max} ]`\n        )\n\n        // Apply position and length constraints to channelData if specified\n        const startIndex = position\n        const endIndex = length ? startIndex + length : channelData.length\n        const constrainedChannelData = channelData.slice(startIndex, endIndex)\n\n        return new Promise((resolve, reject) => {\n            const blob = new Blob([InlineFeaturesExtractor], {\n                type: 'application/javascript',\n            })\n            const url = URL.createObjectURL(blob)\n            const worker = new Worker(url)\n\n            worker.onmessage = (event) => {\n                resolve(event.data.result)\n            }\n\n            worker.onerror = (error) => {\n                reject(error)\n            }\n\n            worker.postMessage({\n                command: 'process',\n                channelData: constrainedChannelData,\n                sampleRate,\n                segmentDurationMs,\n                logger,\n                bitDepth,\n                fullAudioDurationMs: durationMs,\n                numberOfChannels,\n            })\n        })\n    } else {\n        if (!fileUri) {\n            throw new Error('fileUri is required')\n        }\n        logger?.log(`extractAudioAnalysis`, {\n            fileUri,\n            segmentDurationMs,\n        })\n        const res = await ExpoAudioStreamModule.extractAudioAnalysis({\n            fileUri,\n            segmentDurationMs,\n            features,\n            position,\n            length,\n        })\n        logger?.log(`extractAudioAnalysis`, res)\n        return res\n    }\n}\n"]}

package/build/AudioAnalysis/extractAudioData.d.ts

@@ -0,0 +1,3 @@
+import { ExtractAudioDataOptions } from '../ExpoAudioStream.types';
+export declare const extractAudioData: (props: ExtractAudioDataOptions) => Promise<any>;
+//# sourceMappingURL=extractAudioData.d.ts.map