@imcooder/opuslib 0.2.1 → 2.2.1

package/README.md

@@ -29,28 +29,36 @@
 >   });
 >   ```
 >
-> **New Fields**
-> - **`audioLevel`** — Each `audioChunk` event includes a normalized `audioLevel` (0.0~1.0), computed via configurable RMS sliding window (default 360ms) with dBFS-to-linear mapping (IEC 61606).
+> **`packetDuration` now works (was ignored in original)**
+> - In the original library, `packetDuration` was accepted but had no effect. Now it controls how many Opus frames are batched before emitting an `audioChunk` event, reducing JS bridge calls.
+> - Example: `frameSize=20ms, packetDuration=100ms` → 5 frames encoded individually, batched into one `audioChunk` event (80% fewer bridge calls).
+>
+> **New `audioChunk` fields**
+> - **`audioLevel`** — Normalized audio level (0.0~1.0), computed via configurable RMS sliding window (default 360ms) with dBFS-to-linear mapping (IEC 61606).
+> - **`duration`** — Duration of this packet in milliseconds (`frameSize * frameCount`).
+> - **`frameCount`** — Number of Opus frames contained in this packet.
+> - **`preSkip`** — (in `audioStarted` event) Opus encoder lookahead in samples. Decoders should skip this many samples at the beginning of the stream.
 >   ```typescript
 >   Opuslib.addListener('audioChunk', (event) => {
->     // event.data: ArrayBuffer            (Opus encoded packet)
+>     // event.data: ArrayBuffer            (batched Opus encoded frames)
 >     // event.timestamp: 1711000000100     (ms since epoch)
 >     // event.sequenceNumber: 5            (packet counter)
 >     // event.audioLevel: 0.72            (0=silence, 1=loud)
+>     // event.duration: 100               (ms, = frameSize * frameCount)
+>     // event.frameCount: 5               (number of Opus frames in this packet)
 >   });
 >   ```
-> - **`preSkip`** — Opus encoder lookahead in samples, returned in `audioStarted` event. Decoders should skip this many samples at the beginning of the stream.
 >
 > **New Config Options**
-> - **`audioLevelWindow`** — RMS window duration in milliseconds for audio level calculation (default: 360ms). Shorter window = more responsive, longer window = smoother.
+> - **`audioLevelWindow`** — RMS window duration in milliseconds for audio level calculation (default: 360ms). Shorter = more responsive, longer = smoother.
 >   ```typescript
 >   await Opuslib.startStreaming({
 >     sampleRate: 16000,
 >     channels: 1,
 >     bitrate: 24000,
 >     frameSize: 20,
->     packetDuration: 20,
->     audioLevelWindow: 200,  // 200ms window (default: 360ms)
+>     packetDuration: 100,   // batch 5 frames per event (was ignored before)
+>     audioLevelWindow: 200, // 200ms RMS window (default: 360ms)
 >   });
 >   ```
 
@@ -269,8 +277,10 @@ Emitted when an encoded Opus packet is ready.
 
 ```typescript
 Opuslib.addListener('audioChunk', (event: AudioChunkEvent) => {
-  // event.data: ArrayBuffer - Raw Opus packet (ready to send/save)
+  // event.data: ArrayBuffer - Batched Opus frames (ready to send/save)
   // event.audioLevel: number - Audio level 0.0~1.0 (0=silence, 1=loud)
+  // event.duration: number - Duration in ms (frameSize * frameCount)
+  // event.frameCount: number - Number of Opus frames in this packet
 });
 ```
 
@@ -278,10 +288,12 @@ Opuslib.addListener('audioChunk', (event: AudioChunkEvent) => {
 
 ```typescript
 interface AudioChunkEvent {
-  data: ArrayBuffer;         // Raw Opus-encoded audio packet
+  data: ArrayBuffer;         // Batched Opus-encoded frames
   timestamp: number;         // Milliseconds since epoch
   sequenceNumber: number;    // Incrementing packet counter
   audioLevel: number;        // Audio level 0.0~1.0 (360ms RMS window, 0=silence, 1=loud)
+  duration: number;          // Packet duration in ms (frameSize * frameCount)
+  frameCount: number;        // Number of Opus frames in this packet
 }
 ```
 

package/android/src/main/java/expo/modules/opuslib/AudioProcessor.kt

@@ -31,6 +31,9 @@ class AudioProcessor(private val config: AudioConfig) {
   private var opusEncoder: OpusEncoder? = null
   private val pendingSamples = mutableListOf<Short>()
   private val samplesPerFrame: Int = (config.sampleRate * config.frameSize / 1000.0).toInt()
+  private val framesPerPacket: Int = Math.max(1, (config.packetDuration / config.frameSize).toInt())
+  private var packetBuffer = java.io.ByteArrayOutputStream()  // accumulates encoded frames
+  private var packetFrameCount: Int = 0
   private var sequenceNumber: Int = 0
   private var startTime: Double = 0.0
 
@@ -44,7 +47,8 @@ class AudioProcessor(private val config: AudioConfig) {
   private var pcmFileOutputStream: FileOutputStream? = null
 
   // Event callbacks (all invoked on encoding thread)
-  private var onAudioChunk: ((ByteArray, Double, Int, Float) -> Unit)? = null
+  // onAudioChunk: (data, timestamp, sequenceNumber, audioLevel, duration, frameCount)
+  private var onAudioChunk: ((ByteArray, Double, Int, Float, Double, Int) -> Unit)? = null
   private var onStarted: ((timestamp: Double, sampleRate: Int, channels: Int, bitrate: Int, frameSize: Double, preSkip: Int) -> Unit)? = null
   private var onEnd: ((timestamp: Double, totalDuration: Double, totalPackets: Int) -> Unit)? = null
 
@@ -138,7 +142,7 @@ class AudioProcessor(private val config: AudioConfig) {
 
   // MARK: - Event callback setters
 
-  fun setOnAudioChunk(callback: (ByteArray, Double, Int, Float) -> Unit) {
+  fun setOnAudioChunk(callback: (ByteArray, Double, Int, Float, Double, Int) -> Unit) {
     this.onAudioChunk = callback
   }
 
@@ -179,6 +183,7 @@ class AudioProcessor(private val config: AudioConfig) {
         fos.write(bytes)
       }
 
+      // Encode single frame to Opus
       val opusData = try {
         encoder.encode(frameData, samplesPerFrame)
       } catch (e: Exception) {
@@ -191,7 +196,11 @@ class AudioProcessor(private val config: AudioConfig) {
         continue
       }
 
-      // Accumulate energy for RMS over ~360ms window
+      // Accumulate encoded frame into packet buffer
+      packetBuffer.write(opusData)
+      packetFrameCount++
+
+      // Accumulate energy for RMS level
       for (sample in frameData) {
         val s = sample.toDouble() / 32768.0
         levelSumSquares += s * s
@@ -208,20 +217,29 @@ class AudioProcessor(private val config: AudioConfig) {
         levelSampleCount = 0
       }
 
-      val timestampMs = System.currentTimeMillis().toDouble()
-      onAudioChunk?.invoke(opusData, timestampMs, sequenceNumber, currentLevel)
-      sequenceNumber++
+      // Emit when we have enough frames for one packet (packetDuration)
+      if (packetFrameCount >= framesPerPacket) {
+        val timestampMs = System.currentTimeMillis().toDouble()
+        val duration = packetFrameCount * config.frameSize
+        onAudioChunk?.invoke(packetBuffer.toByteArray(), timestampMs, sequenceNumber, currentLevel, duration, packetFrameCount)
+        sequenceNumber++
+        packetBuffer.reset()
+        packetFrameCount = 0
+      }
     }
   }
 
   private fun _flushRemainingFrames() {
     val encoder = opusEncoder ?: return
-    if (pendingSamples.isEmpty()) return
 
-    while (pendingSamples.size < samplesPerFrame) {
-      pendingSamples.add(0)
+    // Pad remaining PCM with silence to fill the last frame
+    if (pendingSamples.isNotEmpty() && pendingSamples.size < samplesPerFrame) {
+      while (pendingSamples.size < samplesPerFrame) {
+        pendingSamples.add(0)
+      }
     }
 
+    // Encode remaining frames
     while (pendingSamples.size >= samplesPerFrame) {
       val frameData = ShortArray(samplesPerFrame)
       for (i in 0 until samplesPerFrame) {
@@ -235,10 +253,18 @@ class AudioProcessor(private val config: AudioConfig) {
       }
 
       if (opusData == null || opusData.isEmpty()) continue
+      packetBuffer.write(opusData)
+      packetFrameCount++
+    }
 
+    // Flush any remaining packet buffer (even if less than framesPerPacket)
+    if (packetBuffer.size() > 0) {
       val timestampMs = System.currentTimeMillis().toDouble()
-      onAudioChunk?.invoke(opusData, timestampMs, sequenceNumber, currentLevel)
+      val duration = packetFrameCount * config.frameSize
+      onAudioChunk?.invoke(packetBuffer.toByteArray(), timestampMs, sequenceNumber, currentLevel, duration, packetFrameCount)
       sequenceNumber++
+      packetBuffer.reset()
+      packetFrameCount = 0
     }
   }
 }

package/android/src/main/java/expo/modules/opuslib/AudioRecordManager.kt

@@ -37,7 +37,7 @@ class AudioRecordManager(
   private var loggedFirstBuffer = false
 
   // Event callbacks
-  private var onAudioChunk: ((ByteArray, Double, Int, Float) -> Unit)? = null
+  private var onAudioChunk: ((ByteArray, Double, Int, Float, Double, Int) -> Unit)? = null
   private var onStarted: ((timestamp: Double, sampleRate: Int, channels: Int, bitrate: Int, frameSize: Double, preSkip: Int) -> Unit)? = null
   private var onEnd: ((timestamp: Double, totalDuration: Double, totalPackets: Int) -> Unit)? = null
   private var onAmplitude: ((Float, Float, Double) -> Unit)? = null
@@ -91,8 +91,8 @@ class AudioRecordManager(
 
     // Create and start AudioProcessor (encoding thread)
     val proc = AudioProcessor(config)
-    proc.setOnAudioChunk { data, timestamp, seq, level ->
-      onAudioChunk?.invoke(data, timestamp, seq, level)
+    proc.setOnAudioChunk { data, timestamp, seq, level, duration, frameCount ->
+      onAudioChunk?.invoke(data, timestamp, seq, level, duration, frameCount)
     }
     proc.setOnStarted { timestamp, sampleRate, channels, bitrate, frameSize, preSkip ->
       onStarted?.invoke(timestamp, sampleRate, channels, bitrate, frameSize, preSkip)
@@ -172,7 +172,7 @@ class AudioRecordManager(
   }
 
   // Event handlers
-  fun setOnAudioChunk(callback: (ByteArray, Double, Int, Float) -> Unit) {
+  fun setOnAudioChunk(callback: (ByteArray, Double, Int, Float, Double, Int) -> Unit) {
     this.onAudioChunk = callback
   }
 

package/android/src/main/java/expo/modules/opuslib/OpuslibModule.kt

@@ -81,12 +81,14 @@ class OpuslibModule : Module() {
 
     // Set up event callbacks — audioStarted/audioEnd come from encoding thread
     android.util.Log.d(TAG, "🔗 Setting up event callbacks...")
-    manager.setOnAudioChunk { data, timestamp, sequenceNumber, audioLevel ->
+    manager.setOnAudioChunk { data, timestamp, sequenceNumber, audioLevel, duration, frameCount ->
       sendEvent("audioChunk", mapOf(
         "data" to data,
         "timestamp" to timestamp,
         "sequenceNumber" to sequenceNumber,
-        "audioLevel" to audioLevel
+        "audioLevel" to audioLevel,
+        "duration" to duration,
+        "frameCount" to frameCount
       ))
     }
 

package/build/Opuslib.types.d.ts

@@ -35,6 +35,10 @@ export interface AudioChunkEvent {
     sequenceNumber: number;
     /** Audio level normalized to 0.0~1.0 (mapped from dBFS, 0 = silence, 1 = loud) */
     audioLevel: number;
+    /** Duration of this packet in milliseconds (frameSize * frameCount) */
+    duration: number;
+    /** Number of Opus frames in this packet */
+    frameCount: number;
 }
 /**
  * Amplitude event payload (for waveform visualization)

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

@@ -1 +1 @@
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+{"version":3,"file":"Opuslib.types.d.ts","sourceRoot":"","sources":["../src/Opuslib.types.ts"],"names":[],"mappings":"AAAA;;GAEG;AACH,MAAM,WAAW,WAAW;IAC1B,2DAA2D;IAC3D,UAAU,EAAE,MAAM,CAAA;IAClB,gDAAgD;IAChD,QAAQ,EAAE,MAAM,CAAA;IAChB,6DAA6D;IAC7D,OAAO,EAAE,MAAM,CAAA;IACf,8DAA8D;IAC9D,SAAS,EAAE,MAAM,CAAA;IACjB,2DAA2D;IAC3D,cAAc,EAAE,MAAM,CAAA;IACtB,oFAAoF;IACpF,YAAY,CAAC,EAAE,MAAM,CAAA;IACrB,yDAAyD;IACzD,qBAAqB,CAAC,EAAE,OAAO,CAAA;IAC/B,4DAA4D;IAC5D,sBAAsB,CAAC,EAAE,MAAM,CAAA;IAC/B,oEAAoE;IACpE,gBAAgB,CAAC,EAAE,MAAM,CAAA;IACzB,sDAAsD;IACtD,cAAc,CAAC,EAAE,OAAO,CAAA;CACzB;AAED;;GAEG;AACH,MAAM,WAAW,eAAe;IAC9B,6CAA6C;IAC7C,IAAI,EAAE,WAAW,CAAA;IACjB,gCAAgC;IAChC,SAAS,EAAE,MAAM,CAAA;IACjB,oDAAoD;IACpD,cAAc,EAAE,MAAM,CAAA;IACtB,kFAAkF;IAClF,UAAU,EAAE,MAAM,CAAA;IAClB,uEAAuE;IACvE,QAAQ,EAAE,MAAM,CAAA;IAChB,2CAA2C;IAC3C,UAAU,EAAE,MAAM,CAAA;CACnB;AAED;;GAEG;AACH,MAAM,WAAW,cAAc;IAC7B,6CAA6C;IAC7C,GAAG,EAAE,MAAM,CAAA;IACX,iCAAiC;IACjC,IAAI,EAAE,MAAM,CAAA;IACZ,gCAAgC;IAChC,SAAS,EAAE,MAAM,CAAA;CAClB;AAED;;;GAGG;AACH,MAAM,WAAW,iBAAiB;IAChC,uDAAuD;IACvD,SAAS,EAAE,MAAM,CAAA;IACjB,oCAAoC;IACpC,UAAU,EAAE,MAAM,CAAA;IAClB,yBAAyB;IACzB,QAAQ,EAAE,MAAM,CAAA;IAChB,wCAAwC;IACxC,OAAO,EAAE,MAAM,CAAA;IACf,qCAAqC;IACrC,SAAS,EAAE,MAAM,CAAA;IACjB,yFAAyF;IACzF,OAAO,EAAE,MAAM,CAAA;CAChB;AAED;;;GAGG;AACH,MAAM,WAAW,aAAa;IAC5B,uDAAuD;IACvD,SAAS,EAAE,MAAM,CAAA;IACjB,8DAA8D;IAC9D,aAAa,EAAE,MAAM,CAAA;IACrB,yDAAyD;IACzD,YAAY,EAAE,MAAM,CAAA;CACrB;AAED;;GAEG;AACH,MAAM,WAAW,UAAU;IACzB,iBAAiB;IACjB,IAAI,EAAE,MAAM,CAAA;IACZ,oBAAoB;IACpB,OAAO,EAAE,MAAM,CAAA;CAChB;AAED;;GAEG;AACH,MAAM,WAAW,YAAY;IAC3B,MAAM,EAAE,MAAM,IAAI,CAAA;CACnB"}

package/build/Opuslib.types.js.map

@@ -1 +1 @@
-{"version":3,"file":"Opuslib.types.js","sourceRoot":"","sources":["../src/Opuslib.types.ts"],"names":[],"mappings":"","sourcesContent":["/**\n * Audio configuration for Opus encoding\n */\nexport interface AudioConfig {\n  /** Sample rate in Hz (8000, 12000, 16000, 24000, 48000) */\n  sampleRate: number\n  /** Number of channels (1 = mono, 2 = stereo) */\n  channels: number\n  /** Target bitrate in bits/second (e.g., 24000 for 24kbps) */\n  bitrate: number\n  /** Frame duration in milliseconds (2.5, 5, 10, 20, 40, 60) */\n  frameSize: number\n  /** Packet duration in milliseconds (typically 20-100ms) */\n  packetDuration: number\n  /** DRED recovery duration in milliseconds (0-100, default 100) - NEW in Opus 1.6 */\n  dredDuration?: number\n  /** Enable amplitude events for waveform visualization */\n  enableAmplitudeEvents?: boolean\n  /** Amplitude event interval in milliseconds (default 16) */\n  amplitudeEventInterval?: number\n  /** Audio level RMS window duration in milliseconds (default 360) */\n  audioLevelWindow?: number\n  /** Save debug PCM audio to file (development only) */\n  saveDebugAudio?: boolean\n}\n\n/**\n * Audio chunk event payload (Opus-encoded data)\n */\nexport interface AudioChunkEvent {\n  /** Opus-encoded audio data as ArrayBuffer */\n  data: ArrayBuffer\n  /** Timestamp in milliseconds */\n  timestamp: number\n  /** Sequence number (increments with each packet) */\n  sequenceNumber: number\n  /** Audio level normalized to 0.0~1.0 (mapped from dBFS, 0 = silence, 1 = loud) */\n  audioLevel: number\n}\n\n/**\n * Amplitude event payload (for waveform visualization)\n */\nexport interface AmplitudeEvent {\n  /** Root mean square amplitude (0.0 - 1.0) */\n  rms: number\n  /** Peak amplitude (0.0 - 1.0) */\n  peak: number\n  /** Timestamp in milliseconds */\n  timestamp: number\n}\n\n/**\n * Audio started event payload\n * Emitted when audio streaming successfully starts\n */\nexport interface AudioStartedEvent {\n  /** Timestamp in milliseconds when streaming started */\n  timestamp: number\n  /** Actual sample rate being used */\n  sampleRate: number\n  /** Number of channels */\n  channels: number\n  /** Configured bitrate in bits/second */\n  bitrate: number\n  /** Frame duration in milliseconds */\n  frameSize: number\n  /** Opus encoder lookahead in samples (decoder should skip this many samples at start) */\n  preSkip: number\n}\n\n/**\n * Audio end event payload\n * Emitted when audio streaming stops\n */\nexport interface AudioEndEvent {\n  /** Timestamp in milliseconds when streaming stopped */\n  timestamp: number\n  /** Total duration of the streaming session in milliseconds */\n  totalDuration: number\n  /** Total number of packets encoded during the session */\n  totalPackets: number\n}\n\n/**\n * Error event payload\n */\nexport interface ErrorEvent {\n  /** Error code */\n  code: string\n  /** Error message */\n  message: string\n}\n\n/**\n * Event subscription\n */\nexport interface Subscription {\n  remove: () => void\n}\n"]}
+{"version":3,"file":"Opuslib.types.js","sourceRoot":"","sources":["../src/Opuslib.types.ts"],"names":[],"mappings":"","sourcesContent":["/**\n * Audio configuration for Opus encoding\n */\nexport interface AudioConfig {\n  /** Sample rate in Hz (8000, 12000, 16000, 24000, 48000) */\n  sampleRate: number\n  /** Number of channels (1 = mono, 2 = stereo) */\n  channels: number\n  /** Target bitrate in bits/second (e.g., 24000 for 24kbps) */\n  bitrate: number\n  /** Frame duration in milliseconds (2.5, 5, 10, 20, 40, 60) */\n  frameSize: number\n  /** Packet duration in milliseconds (typically 20-100ms) */\n  packetDuration: number\n  /** DRED recovery duration in milliseconds (0-100, default 100) - NEW in Opus 1.6 */\n  dredDuration?: number\n  /** Enable amplitude events for waveform visualization */\n  enableAmplitudeEvents?: boolean\n  /** Amplitude event interval in milliseconds (default 16) */\n  amplitudeEventInterval?: number\n  /** Audio level RMS window duration in milliseconds (default 360) */\n  audioLevelWindow?: number\n  /** Save debug PCM audio to file (development only) */\n  saveDebugAudio?: boolean\n}\n\n/**\n * Audio chunk event payload (Opus-encoded data)\n */\nexport interface AudioChunkEvent {\n  /** Opus-encoded audio data as ArrayBuffer */\n  data: ArrayBuffer\n  /** Timestamp in milliseconds */\n  timestamp: number\n  /** Sequence number (increments with each packet) */\n  sequenceNumber: number\n  /** Audio level normalized to 0.0~1.0 (mapped from dBFS, 0 = silence, 1 = loud) */\n  audioLevel: number\n  /** Duration of this packet in milliseconds (frameSize * frameCount) */\n  duration: number\n  /** Number of Opus frames in this packet */\n  frameCount: number\n}\n\n/**\n * Amplitude event payload (for waveform visualization)\n */\nexport interface AmplitudeEvent {\n  /** Root mean square amplitude (0.0 - 1.0) */\n  rms: number\n  /** Peak amplitude (0.0 - 1.0) */\n  peak: number\n  /** Timestamp in milliseconds */\n  timestamp: number\n}\n\n/**\n * Audio started event payload\n * Emitted when audio streaming successfully starts\n */\nexport interface AudioStartedEvent {\n  /** Timestamp in milliseconds when streaming started */\n  timestamp: number\n  /** Actual sample rate being used */\n  sampleRate: number\n  /** Number of channels */\n  channels: number\n  /** Configured bitrate in bits/second */\n  bitrate: number\n  /** Frame duration in milliseconds */\n  frameSize: number\n  /** Opus encoder lookahead in samples (decoder should skip this many samples at start) */\n  preSkip: number\n}\n\n/**\n * Audio end event payload\n * Emitted when audio streaming stops\n */\nexport interface AudioEndEvent {\n  /** Timestamp in milliseconds when streaming stopped */\n  timestamp: number\n  /** Total duration of the streaming session in milliseconds */\n  totalDuration: number\n  /** Total number of packets encoded during the session */\n  totalPackets: number\n}\n\n/**\n * Error event payload\n */\nexport interface ErrorEvent {\n  /** Error code */\n  code: string\n  /** Error message */\n  message: string\n}\n\n/**\n * Event subscription\n */\nexport interface Subscription {\n  remove: () => void\n}\n"]}

package/ios/AudioEngineManager.swift

@@ -29,7 +29,7 @@ class AudioEngineManager {
   private var loggedFirstBuffer = false
 
   // Event callbacks
-  private var onAudioChunk: ((Data, Double, Int, Float) -> Void)?
+  private var onAudioChunk: ((Data, Double, Int, Float, Double, Int) -> Void)?
   private var onStarted: ((_ timestamp: Double, _ sampleRate: Int, _ channels: Int, _ bitrate: Int, _ frameSize: Double, _ preSkip: Int) -> Void)?
   private var onEnd: ((_ timestamp: Double, _ totalDuration: Double, _ totalPackets: Int) -> Void)?
   private var onAmplitude: ((Float, Float, Double) -> Void)?
@@ -59,8 +59,8 @@ class AudioEngineManager {
 
     // Create and start AudioProcessor (encoding thread)
     let proc = AudioProcessor(config: config)
-    proc.setOnAudioChunk { [weak self] data, timestamp, seq, level in
-      self?.onAudioChunk?(data, timestamp, seq, level)
+    proc.setOnAudioChunk { [weak self] data, timestamp, seq, level, duration, frameCount in
+      self?.onAudioChunk?(data, timestamp, seq, level, duration, frameCount)
     }
     proc.setOnStarted { [weak self] timestamp, sampleRate, channels, bitrate, frameSize, preSkip in
       self?.onStarted?(timestamp, sampleRate, channels, bitrate, frameSize, preSkip)
@@ -159,7 +159,7 @@ class AudioEngineManager {
 
   // MARK: - Event Handlers
 
-  func setOnAudioChunk(_ callback: @escaping (Data, Double, Int, Float) -> Void) {
+  func setOnAudioChunk(_ callback: @escaping (Data, Double, Int, Float, Double, Int) -> Void) {
     self.onAudioChunk = callback
   }
 

package/ios/AudioProcessor.swift

@@ -19,6 +19,9 @@ class AudioProcessor {
   private var opusEncoder: OpusEncoder?
   private var pendingSamples: [Int16] = []
   private let samplesPerFrame: Int
+  private let framesPerPacket: Int  // how many frames to batch before emitting
+  private var packetBuffer: Data = Data()  // accumulates encoded frames
+  private var packetFrameCount: Int = 0
   private var sequenceNumber: Int = 0
   private var startTime: Double = 0
 
@@ -32,7 +35,8 @@ class AudioProcessor {
   private var pcmFileHandle: FileHandle?
 
   // Event callbacks (all invoked on encoding queue)
-  private var onAudioChunk: ((Data, Double, Int, Float) -> Void)?
+  // onAudioChunk: (data, timestamp, sequenceNumber, audioLevel, duration, frameCount)
+  private var onAudioChunk: ((Data, Double, Int, Float, Double, Int) -> Void)?
   private var onStarted: ((_ timestamp: Double, _ sampleRate: Int, _ channels: Int, _ bitrate: Int, _ frameSize: Double, _ preSkip: Int) -> Void)?
   private var onEnd: ((_ timestamp: Double, _ totalDuration: Double, _ totalPackets: Int) -> Void)?
 
@@ -42,6 +46,7 @@ class AudioProcessor {
   init(config: AudioConfig) {
     self.config = config
     self.samplesPerFrame = Int(Double(config.sampleRate) * config.frameSize / 1000.0)
+    self.framesPerPacket = max(1, Int(config.packetDuration / config.frameSize))
     let windowMs = config.audioLevelWindow ?? 360
     self.levelUpdateSamples = config.sampleRate * config.channels * windowMs / 1000
   }
@@ -128,7 +133,7 @@ class AudioProcessor {
 
   // MARK: - Event callbacks
 
-  func setOnAudioChunk(_ callback: @escaping (Data, Double, Int, Float) -> Void) {
+  func setOnAudioChunk(_ callback: @escaping (Data, Double, Int, Float, Double, Int) -> Void) {
     self.onAudioChunk = callback
   }
 
@@ -157,7 +162,7 @@ class AudioProcessor {
         }
       }
 
-      // Encode to Opus
+      // Encode single frame to Opus
       var encodedPacket: Data?
       frameData.withUnsafeBufferPointer { bufferPointer in
         guard let baseAddress = bufferPointer.baseAddress else { return }
@@ -169,7 +174,11 @@ class AudioProcessor {
         continue
       }
 
-      // Accumulate energy for RMS over ~360ms window
+      // Accumulate encoded frame into packet buffer
+      packetBuffer.append(opusData)
+      packetFrameCount += 1
+
+      // Accumulate energy for RMS level
       for sample in frameData {
         let s = Double(sample) / 32768.0
         levelSumSquares += s * s
@@ -186,20 +195,27 @@ class AudioProcessor {
         levelSampleCount = 0
       }
 
-      let timestampMs = Date().timeIntervalSince1970 * 1000
-      onAudioChunk?(opusData, timestampMs, sequenceNumber, currentLevel)
-      sequenceNumber += 1
+      // Emit when we have enough frames for one packet (packetDuration)
+      if packetFrameCount >= framesPerPacket {
+        let timestampMs = Date().timeIntervalSince1970 * 1000
+        let duration = Double(packetFrameCount) * config.frameSize
+        onAudioChunk?(packetBuffer, timestampMs, sequenceNumber, currentLevel, duration, packetFrameCount)
+        sequenceNumber += 1
+        packetBuffer = Data()
+        packetFrameCount = 0
+      }
     }
   }
 
   private func _flushRemainingFrames() {
     guard let opusEncoder = opusEncoder else { return }
-    guard !pendingSamples.isEmpty else { return }
 
-    if pendingSamples.count < samplesPerFrame {
+    // Pad remaining PCM with silence to fill the last frame
+    if !pendingSamples.isEmpty && pendingSamples.count < samplesPerFrame {
       pendingSamples.append(contentsOf: [Int16](repeating: 0, count: samplesPerFrame - pendingSamples.count))
     }
 
+    // Encode remaining frames
     while pendingSamples.count >= samplesPerFrame {
       let frameData = Array(pendingSamples.prefix(samplesPerFrame))
       pendingSamples.removeFirst(samplesPerFrame)
@@ -211,10 +227,18 @@ class AudioProcessor {
       }
 
       guard let opusData = encodedPacket, !opusData.isEmpty else { continue }
+      packetBuffer.append(opusData)
+      packetFrameCount += 1
+    }
 
+    // Flush any remaining packet buffer (even if less than framesPerPacket)
+    if !packetBuffer.isEmpty {
       let timestampMs = Date().timeIntervalSince1970 * 1000
-      onAudioChunk?(opusData, timestampMs, sequenceNumber, currentLevel)
+      let duration = Double(packetFrameCount) * config.frameSize
+      onAudioChunk?(packetBuffer, timestampMs, sequenceNumber, currentLevel, duration, packetFrameCount)
       sequenceNumber += 1
+      packetBuffer = Data()
+      packetFrameCount = 0
     }
   }
 }

package/ios/OpuslibModule.swift

@@ -58,12 +58,14 @@ public class OpuslibModule: Module {
     print("[OpuslibModule] ✅ AudioEngineManager created")
 
     // Set up event callbacks — audioStarted/audioEnd come from encoding thread
-    manager.setOnAudioChunk { [weak self] data, timestamp, sequenceNumber, audioLevel in
+    manager.setOnAudioChunk { [weak self] data, timestamp, sequenceNumber, audioLevel, duration, frameCount in
       self?.sendEvent("audioChunk", [
         "data": data,
         "timestamp": timestamp,
         "sequenceNumber": sequenceNumber,
-        "audioLevel": audioLevel
+        "audioLevel": audioLevel,
+        "duration": duration,
+        "frameCount": frameCount
       ])
     }
 

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@imcooder/opuslib",
-  "version": "0.2.1",
+  "version": "2.2.1",
   "description": "Opus 1.6 audio encoding for React Native and Expo with audio level metering and lifecycle events. Forked from Scdales/opuslib.",
   "main": "build/index.js",
   "types": "build/index.d.ts",

package/src/Opuslib.types.ts

@@ -36,6 +36,10 @@ export interface AudioChunkEvent {
   sequenceNumber: number
   /** Audio level normalized to 0.0~1.0 (mapped from dBFS, 0 = silence, 1 = loud) */
   audioLevel: number
+  /** Duration of this packet in milliseconds (frameSize * frameCount) */
+  duration: number
+  /** Number of Opus frames in this packet */
+  frameCount: number
 }
 
 /**