@stream-io/video-react-native-sdk 1.39.0 → 1.40.0

package/dist/typescript/hooks/useModeration.d.ts.map

@@ -1 +1 @@
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+{"version":3,"file":"useModeration.d.ts","sourceRoot":"","sources":["../../../src/hooks/useModeration.ts"],"names":[],"mappings":"AAIA,MAAM,WAAW,iBAAiB;IAChC;;;OAGG;IACH,QAAQ,CAAC,EAAE,MAAM,CAAC;CACnB;AAED,eAAO,MAAM,aAAa,GAAI,UAAU,iBAAiB,SAyExD,CAAC"}

package/dist/typescript/icons/index.d.ts

@@ -4,22 +4,16 @@ export * from './Mic';
 export * from './MicOff';
 export * from './Phone';
 export * from './PhoneDown';
-export * from './Settings';
 export * from './Video';
 export * from './VideoSlash';
-export * from './ThreeDots';
 export * from './PinVertical';
-export * from './Spotlight';
 export * from './ScreenShareIndicator';
 export * from './ScreenShare';
 export * from './Reaction';
-export * from './TopViewBackground';
 export * from './Eye';
-export * from './ShieldBadge';
 export * from './StartStreamIcon';
 export * from './StopScreenShare';
 export * from './EndStreamIcon';
-export * from './LeaveStreamIcon';
 export * from './IconWrapper';
 export * from './Maximize';
 //# sourceMappingURL=index.d.ts.map

package/dist/typescript/icons/index.d.ts.map

@@ -1 +1 @@
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package/dist/typescript/providers/StreamCall/AppStateListener.d.ts.map

@@ -1 +1 @@
-{"version":3,"file":"AppStateListener.d.ts","sourceRoot":"","sources":["../../../../src/providers/StreamCall/AppStateListener.tsx"],"names":[],"mappings":"AAsBA,eAAO,MAAM,gBAAgB,YAuN5B,CAAC"}
+{"version":3,"file":"AppStateListener.d.ts","sourceRoot":"","sources":["../../../../src/providers/StreamCall/AppStateListener.tsx"],"names":[],"mappings":"AAsBA,eAAO,MAAM,gBAAgB,YA0N5B,CAAC"}

package/dist/typescript/providers/StreamCall/DeviceStats.d.ts.map

@@ -1 +1 @@
-{"version":3,"file":"DeviceStats.d.ts","sourceRoot":"","sources":["../../../../src/providers/StreamCall/DeviceStats.tsx"],"names":[],"mappings":"AAWA;;GAEG;AACH,eAAO,MAAM,WAAW,YA0EvB,CAAC"}
+{"version":3,"file":"DeviceStats.d.ts","sourceRoot":"","sources":["../../../../src/providers/StreamCall/DeviceStats.tsx"],"names":[],"mappings":"AAUA;;GAEG;AACH,eAAO,MAAM,WAAW,YA4EvB,CAAC"}

package/dist/typescript/utils/StreamVideoRN/index.d.ts

@@ -4,7 +4,6 @@ type DeepPartial<T> = {
 };
 export declare class StreamVideoRN {
     private static config;
-    private static busyToneTimeout;
     /**
      * Update the global config for StreamVideoRN except for push config.
      * To set push config use `StreamVideoRN.setPushConfig` instead.

package/dist/typescript/utils/StreamVideoRN/index.d.ts.map

@@ -1 +1 @@
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+{"version":3,"file":"index.d.ts","sourceRoot":"","sources":["../../../../src/utils/StreamVideoRN/index.ts"],"names":[],"mappings":"AAAA,OAAO,KAAK,EAAE,iBAAiB,EAAE,MAAM,SAAS,CAAC;AAYjD,KAAK,WAAW,CAAC,CAAC,IAAI;KACnB,CAAC,IAAI,MAAM,CAAC,CAAC,CAAC,EAAE,CAAC,CAAC,CAAC,CAAC,SAAS,MAAM,GAAG,WAAW,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,GAAG,CAAC,CAAC,CAAC,CAAC;CAChE,CAAC;AAgDF,qBAAa,aAAa;IACxB,OAAO,CAAC,MAAM,CAAC,MAAM,CAA+B;IAEpD;;;;OAIG;IACH,MAAM,CAAC,YAAY,CACjB,YAAY,EAAE,WAAW,CAAC,IAAI,CAAC,iBAAiB,EAAE,MAAM,CAAC,CAAC;IAK5D;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;OAmCG;IACH,MAAM,CAAC,aAAa,CAAC,UAAU,EAAE,WAAW,CAAC,iBAAiB,CAAC,MAAM,CAAC,CAAC;IAyBvE,MAAM,CAAC,SAAS;IAIhB;;;OAGG;IACH,MAAM,CAAC,YAAY;IASnB,MAAM,CAAC,wBAAwB;IAI/B;;OAEG;WACU,YAAY;IAIzB;;OAEG;WACU,YAAY;IAIzB;;;OAGG;WACU,6BAA6B,IAAI,OAAO,CAAC,OAAO,CAAC;IAQ9D;;;OAGG;WACU,4BAA4B,IAAI,OAAO,CAAC,OAAO,CAAC;IAQ7D;;;OAGG;WACU,wBAAwB,IAAI,OAAO,CAAC,OAAO,CAAC;CAO1D"}

package/dist/typescript/utils/internal/callingx/callingx.d.ts.map

@@ -1 +1 @@
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+{"version":3,"file":"callingx.d.ts","sourceRoot":"","sources":["../../../../../src/utils/internal/callingx/callingx.ts"],"names":[],"mappings":"AAKA,OAAO,KAAK,EAAE,aAAa,EAAE,MAAM,kCAAkC,CAAC;AAGtE,OAAO,KAAK,EACV,IAAI,EACJ,cAAc,EACd,sBAAsB,EACvB,MAAM,yBAAyB,CAAC;AAKjC;;GAEG;AACH,wBAAgB,kBAAkB,CAChC,WAAW,EAAE,cAAc,EAAE,GAAG,SAAS,EACzC,YAAY,EAAE,sBAAsB,EAAE,GAAG,SAAS,EAClD,aAAa,EAAE,MAAM,GAAG,SAAS,GAChC,MAAM,CAgCR;AAaD,wBAAsB,oBAAoB,CAAC,IAAI,EAAE,IAAI,iBA+BpD;AAED;;;;;;;GAOG;AACH,wBAAsB,gBAAgB,CAAC,IAAI,EAAE,IAAI,EAAE,WAAW,EAAE,IAAI,EAAE,iBAiFrE;AAED,wBAAsB,eAAe,CAAC,IAAI,EAAE,IAAI,EAAE,MAAM,CAAC,EAAE,aAAa,iBAmBvE"}

package/dist/typescript/utils/push/libs/firebaseMessaging/index.d.ts

@@ -1,20 +1,6 @@
 import { type Type } from './lib';
 export type { FirebaseMessagingTypes } from '@react-native-firebase/messaging';
 export type FirebaseMessagingType = Type;
-export declare function getFirebaseMessagingLib(): {
-    (app?: import("@react-native-firebase/app").ReactNativeFirebase.FirebaseApp): import("@react-native-firebase/messaging").Messaging;
-    readonly SDK_VERSION: string;
-} & import("@react-native-firebase/messaging/dist/typescript/lib/types/messaging").Statics & {
-    messaging: import("@react-native-firebase/app").ReactNativeFirebase.FirebaseModuleWithStaticsAndApp<import("@react-native-firebase/messaging").Messaging, import("@react-native-firebase/messaging/dist/typescript/lib/types/messaging").Statics>;
-    firebase: import("@react-native-firebase/app").ReactNativeFirebase.Module;
-    app(name?: string): import("@react-native-firebase/app").ReactNativeFirebase.FirebaseApp;
-};
-export declare function getFirebaseMessagingLibNoThrow(isExpo: boolean): ({
-    (app?: import("@react-native-firebase/app").ReactNativeFirebase.FirebaseApp): import("@react-native-firebase/messaging").Messaging;
-    readonly SDK_VERSION: string;
-} & import("@react-native-firebase/messaging/dist/typescript/lib/types/messaging").Statics & {
-    messaging: import("@react-native-firebase/app").ReactNativeFirebase.FirebaseModuleWithStaticsAndApp<import("@react-native-firebase/messaging").Messaging, import("@react-native-firebase/messaging/dist/typescript/lib/types/messaging").Statics>;
-    firebase: import("@react-native-firebase/app").ReactNativeFirebase.Module;
-    app(name?: string): import("@react-native-firebase/app").ReactNativeFirebase.FirebaseApp;
-}) | undefined;
+export declare function getFirebaseMessagingLib(): FirebaseMessagingType;
+export declare function getFirebaseMessagingLibNoThrow(isExpo: boolean): FirebaseMessagingType | undefined;
 //# sourceMappingURL=index.d.ts.map

package/dist/typescript/utils/push/libs/firebaseMessaging/index.d.ts.map

@@ -1 +1 @@
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+{"version":3,"file":"index.d.ts","sourceRoot":"","sources":["../../../../../../src/utils/push/libs/firebaseMessaging/index.ts"],"names":[],"mappings":"AAAA,OAAO,EAAO,KAAK,IAAI,EAAE,MAAM,OAAO,CAAC;AAGvC,YAAY,EAAE,sBAAsB,EAAE,MAAM,kCAAkC,CAAC;AAC/E,MAAM,MAAM,qBAAqB,GAAG,IAAI,CAAC;AAKzC,wBAAgB,uBAAuB,IAAI,qBAAqB,CAQ/D;AAED,wBAAgB,8BAA8B,CAC5C,MAAM,EAAE,OAAO,GACd,qBAAqB,GAAG,SAAS,CAcnC"}

package/dist/typescript/version.d.ts

@@ -1,2 +1,2 @@
-export declare const version = "1.39.0";
+export declare const version = "1.40.0";
 //# sourceMappingURL=version.d.ts.map

package/ios/StreamVideoReactNative-Bridging-Header.h

@@ -11,6 +11,8 @@
 #import <WebRTC/RTCVideoTrack.h>
 #import <WebRTC/RTCVideoRenderer.h>
 #import <WebRTC/RTCVideoFrameBuffer.h>
+#import <WebRTC/RTCAudioTrack.h>
+#import <WebRTC/RTCAudioRenderer.h>
 #import "WebRTCModule.h"
 #import "WebRTCModuleOptions.h"
 

package/ios/StreamVideoReactNative.m

@@ -20,6 +20,13 @@
 #import <stream_react_native_webrtc/stream_react_native_webrtc-Swift.h>
 #endif
 
+// Import Swift-generated header for TracksRecorderManager and friends.
+#if __has_include("stream_video_react_native-Swift.h")
+#import "stream_video_react_native-Swift.h"
+#elif __has_include(<stream_video_react_native/stream_video_react_native-Swift.h>)
+#import <stream_video_react_native/stream_video_react_native-Swift.h>
+#endif
+
 // Do not change these consts, it is what is used react-native-webrtc
 NSNotificationName const kBroadcastStartedNotification = @"iOS_BroadcastStarted";
 NSNotificationName const kBroadcastStoppedNotification = @"iOS_BroadcastStopped";
@@ -685,4 +692,78 @@ RCT_EXPORT_METHOD(stopScreenShareAudioMixing:(RCTPromiseResolveBlock)resolve
     resolve(nil);
 }
 
+#pragma mark - Track Recording
+
+RCT_EXPORT_METHOD(startTrackRecording:(NSDictionary *)options
+                  resolver:(RCTPromiseResolveBlock)resolve
+                  rejecter:(RCTPromiseRejectBlock)reject)
+{
+    WebRTCModule *webrtcModule = [self.bridge moduleForClass:[WebRTCModule class]];
+    if (!webrtcModule) {
+        reject(@"recording_error", @"WebRTCModule not available", nil);
+        return;
+    }
+
+    NSString *videoTrackId = options[@"videoTrackId"];
+    if (![videoTrackId isKindOfClass:[NSString class]]) videoTrackId = nil;
+
+    NSNumber *maxDuration = options[@"maxDurationMs"];
+    NSInteger maxDurationMs = ([maxDuration isKindOfClass:[NSNumber class]])
+        ? [maxDuration integerValue] : 5000;
+
+    NSNumber *targetW = options[@"targetWidth"];
+    NSInteger targetWidth = ([targetW isKindOfClass:[NSNumber class]])
+        ? [targetW integerValue] : 0;
+
+    NSNumber *targetH = options[@"targetHeight"];
+    NSInteger targetHeight = ([targetH isKindOfClass:[NSNumber class]])
+        ? [targetH integerValue] : 0;
+
+    [[TracksRecorderManager shared]
+        startRecordingWithVideoTrackId:videoTrackId
+                         maxDurationMs:maxDurationMs
+                           targetWidth:targetWidth
+                          targetHeight:targetHeight
+                          webRTCModule:webrtcModule
+                            completion:^(NSURL * _Nullable fileURL, NSError * _Nullable err) {
+        if (err) {
+            reject(@"recording_error", err.localizedDescription, err);
+        } else {
+            resolve(fileURL ? fileURL.absoluteString : [NSNull null]);
+        }
+    }];
+}
+
+RCT_EXPORT_METHOD(stopTrackRecording:(RCTPromiseResolveBlock)resolve
+                  rejecter:(RCTPromiseRejectBlock)reject)
+{
+    [[TracksRecorderManager shared] stopRecordingWithCompletion:^{
+        resolve(nil);
+    }];
+}
+
+RCT_EXPORT_METHOD(clearStreamRecordings:(RCTPromiseResolveBlock)resolve
+                  rejecter:(RCTPromiseRejectBlock)reject)
+{
+    [[TracksRecorderManager shared] clearRecordingsDirectoryWithCompletion:^(NSError * _Nullable err) {
+        if (err) {
+            reject(@"clear_error", err.localizedDescription, err);
+        } else {
+            resolve(nil);
+        }
+    }];
+}
+
+RCT_EXPORT_METHOD(getStreamRecordings:(RCTPromiseResolveBlock)resolve
+                  rejecter:(RCTPromiseRejectBlock)reject)
+{
+    NSArray<NSURL *> *urls = [[TracksRecorderManager shared] listRecordings];
+    NSMutableArray<NSString *> *result = [NSMutableArray arrayWithCapacity:urls.count];
+    for (NSURL *url in urls) {
+        NSString *abs = url.absoluteString;
+        if (abs) [result addObject:abs];
+    }
+    resolve(result);
+}
+
 @end

package/ios/TracksRecorder/AudioPipeline.swift

@@ -0,0 +1,270 @@
+//
+// Copyright © 2026 Stream.io Inc. All rights reserved.
+//
+
+import AVFoundation
+import CoreMedia
+import Foundation
+import WebRTC
+
+/// Audio pipeline owned by `TracksRecorderManager`. Encapsulates the AAC audio path:
+///  - the `RecorderAudioRenderTap` installed on
+///    `RTCDefaultAudioProcessingModule.renderPreProcessingDelegate`
+///    (post-mix decoded audio, no per-track lookup required),
+///  - the in-place speaker mute (`muteOriginal: true` on the tap; the tap
+///    zero-fills the buffer after copying for recording),
+///  - the AAC `AVAssetWriterInput` (writer-driven encode via
+///    `outputSettings`),
+///  - per-recording counters / PTS range surfaced via `logSummary` at stop.
+///
+/// All state mutation runs on the host's serial queue. The tap's
+/// callback runs on a WebRTC audio thread and re-dispatches onto
+/// `host.queue` after copying the PCM buffer.
+internal final class AudioPipeline {
+
+    private static let aacBitRate: NSNumber = NSNumber(value: 64_000)
+
+    private weak var host: PipelineHost?
+
+    private let apm: RTCDefaultAudioProcessingModule
+
+    private var renderTap: RecorderAudioRenderTap?
+    private var audioInput: AVAssetWriterInput?
+    private var inputAdded = false
+
+    // Diagnostic counters + PTS range, surfaced via [logSummary] at stop.
+    private var buffersReceived = 0
+    private var samplesAppended = 0
+    private var buffersDropped = 0
+    private var firstSamplePtsUs: Int64 = -1
+    private var lastSamplePtsUs: Int64 = -1
+
+    // MARK: - Init
+
+    init(host: PipelineHost, apm: RTCDefaultAudioProcessingModule) {
+        self.host = host
+        self.apm = apm
+    }
+
+    // MARK: - Public API
+
+    /// Install the render-tap as the APM's `renderPreProcessingDelegate`.
+    /// The tap copies PCM into a new buffer for recording AND zero-fills the
+    /// original (post-mix decoded audio) so the speaker plays silence —
+    /// this gives "audio in the file, silence at the speaker" without
+    /// disrupting the recording. The standard `track.setVolume(0)` /
+    /// `track.isEnabled = false` mutes apply *before* this tap and would
+    /// silence the recording too.
+    func start() {
+        let tap = RecorderAudioRenderTap(muteOriginal: true) { [weak self] pcmBuffer in
+            self?.handleAudioBuffer(pcmBuffer: pcmBuffer)
+        }
+        renderTap = tap
+        apm.renderPreProcessingDelegate = tap
+    }
+
+    /// On-queue. Clear the render-tap delegate slot — only if it still
+    /// points to this pipeline's tap. If another consumer has rotated in,
+    /// leave theirs alone.
+    func detachSink() {
+        if let tap = renderTap, apm.renderPreProcessingDelegate === tap {
+            apm.renderPreProcessingDelegate = nil
+        }
+        renderTap = nil
+    }
+
+    /// On-queue. Marks the asset-writer input as finished so the writer can
+    /// finalise.
+    func markInputAsFinished() {
+        audioInput?.markAsFinished()
+    }
+
+    func logSummary() {
+        let tapCalls = renderTap?.callCount ?? -1
+        let durationMs: Int64
+        if firstSamplePtsUs >= 0 && lastSamplePtsUs >= firstSamplePtsUs {
+            durationMs = (lastSamplePtsUs - firstSamplePtsUs) / 1000
+        } else {
+            durationMs = -1
+        }
+        NSLog(
+            "[TracksRecorder.Audio] summary received=%d appended=%d dropped=%d tapCalls=%d firstPtsUs=%lld lastPtsUs=%lld durationMs=%lld",
+            buffersReceived,
+            samplesAppended,
+            buffersDropped,
+            tapCalls,
+            firstSamplePtsUs,
+            lastSamplePtsUs,
+            durationMs
+        )
+    }
+
+    // MARK: - Tap → queue bridge
+
+    private func handleAudioBuffer(pcmBuffer: AVAudioPCMBuffer) {
+        // Unlike `VideoPipeline`'s `CVPixelBuffer` closure capture, an
+        // ARC-retained `AVAudioPCMBuffer` does *not* extend the lifetime
+        // of the underlying PCM samples — those live in WebRTC's
+        // render-buffer pool and are reused the moment this callback
+        // returns. A deep copy before the queue hop is mandatory.
+        guard let copy = AudioPipeline.deepCopyPCMBuffer(pcmBuffer) else { return }
+        guard let host = host else { return }
+
+        // `DispatchTime.now().uptimeNanoseconds` is the monotonic clock
+        // that matches `RTCVideoFrame.timeStampNs` on iOS — both reduce
+        // to `mach_absolute_time()` converted to nanoseconds, so the
+        // shared time origin works coherently across both pipelines.
+        let captureTimeNs = DispatchTime.now().uptimeNanoseconds
+        host.queue.async { [weak self] in
+            self?.handleAudioBufferOnQueue(pcmBuffer: copy, captureTimeNs: captureTimeNs)
+        }
+    }
+
+    private func handleAudioBufferOnQueue(pcmBuffer: AVAudioPCMBuffer, captureTimeNs: UInt64) {
+        guard let host = host, host.isRecording, let writer = host.assetWriter else { return }
+
+        // Lazy-create the writer's audio input on the first buffer. The
+        // input's settings depend on the runtime PCM format reported by
+        // WebRTC.
+        if audioInput == nil {
+            configureAudioInput(format: pcmBuffer.format, writer: writer)
+        }
+
+        let pts = presentationTime(host: host, timestampNs: captureTimeNs)
+
+        guard writer.status == .writing,
+              let audioInput = audioInput,
+              audioInput.isReadyForMoreMediaData else {
+            buffersDropped += 1
+            return
+        }
+
+        guard let sampleBuffer = AudioPipeline.makeSampleBuffer(from: pcmBuffer, pts: pts) else {
+            buffersDropped += 1
+            return
+        }
+
+        if audioInput.append(sampleBuffer) {
+            buffersReceived += 1
+            samplesAppended += 1
+            let ptsUs = Int64(CMTimeGetSeconds(pts) * 1_000_000)
+            if firstSamplePtsUs < 0 || ptsUs < firstSamplePtsUs {
+                firstSamplePtsUs = ptsUs
+            }
+            if ptsUs > lastSamplePtsUs {
+                lastSamplePtsUs = ptsUs
+            }
+        } else {
+            buffersDropped += 1
+        }
+    }
+
+    // MARK: - Asset writer input setup
+
+    private func configureAudioInput(format: AVAudioFormat, writer: AVAssetWriter) {
+        let settings: [String: Any] = [
+            AVFormatIDKey: NSNumber(value: kAudioFormatMPEG4AAC),
+            AVSampleRateKey: NSNumber(value: format.sampleRate),
+            AVNumberOfChannelsKey: NSNumber(value: format.channelCount),
+            AVEncoderBitRateKey: AudioPipeline.aacBitRate,
+        ]
+        let input = AVAssetWriterInput(mediaType: .audio, outputSettings: settings)
+        input.expectsMediaDataInRealTime = true
+
+        guard writer.canAdd(input) else {
+            NSLog("[TracksRecorder.Audio] writer cannot add audio input")
+            host?.onFatalError(makeRecorderError("audio_input_add_failed", code: 4))
+            return
+        }
+         
+        writer.add(input)
+        audioInput = input
+        inputAdded = true
+        host?.onTrackAdded()
+    }
+
+    // MARK: - PCM → CMSampleBuffer helper
+
+    /// Converts an `AVAudioPCMBuffer` into a `CMSampleBuffer` suitable for
+    /// `AVAssetWriterInput.append`. Returns `nil` if any Core Media call
+    /// fails; the caller treats that as a dropped buffer.
+    private static func makeSampleBuffer(
+        from pcmBuffer: AVAudioPCMBuffer,
+        pts: CMTime
+    ) -> CMSampleBuffer? {
+        var formatDescription: CMAudioFormatDescription?
+        let createDescStatus = CMAudioFormatDescriptionCreate(
+            allocator: kCFAllocatorDefault,
+            asbd: pcmBuffer.format.streamDescription,
+            layoutSize: 0,
+            layout: nil,
+            magicCookieSize: 0,
+            magicCookie: nil,
+            extensions: nil,
+            formatDescriptionOut: &formatDescription
+        )
+        guard createDescStatus == noErr, let formatDesc = formatDescription else { return nil }
+
+        var sampleBuffer: CMSampleBuffer?
+        var timing = CMSampleTimingInfo(
+            duration: CMTime(value: 1, timescale: Int32(pcmBuffer.format.sampleRate)),
+            presentationTimeStamp: pts,
+            decodeTimeStamp: .invalid
+        )
+        let createStatus = CMSampleBufferCreate(
+            allocator: kCFAllocatorDefault,
+            dataBuffer: nil,
+            dataReady: false,
+            makeDataReadyCallback: nil,
+            refcon: nil,
+            formatDescription: formatDesc,
+            sampleCount: CMItemCount(pcmBuffer.frameLength),
+            sampleTimingEntryCount: 1,
+            sampleTimingArray: &timing,
+            sampleSizeEntryCount: 0,
+            sampleSizeArray: nil,
+            sampleBufferOut: &sampleBuffer
+        )
+        guard createStatus == noErr, let sb = sampleBuffer else { return nil }
+
+        let setStatus = CMSampleBufferSetDataBufferFromAudioBufferList(
+            sb,
+            blockBufferAllocator: kCFAllocatorDefault,
+            blockBufferMemoryAllocator: kCFAllocatorDefault,
+            flags: 0,
+            bufferList: pcmBuffer.audioBufferList
+        )
+        guard setStatus == noErr else { return nil }
+        return sb
+    }
+
+    /// Returns a deep copy of the supplied `AVAudioPCMBuffer`. WebRTC owns
+    /// the source buffer's backing memory only for the duration of the
+    /// render-tap callback; ARC retains the wrapper across the queue hop
+    /// but not the underlying PCM samples. Copying here lets the recorder
+    /// queue read the data later without racing WebRTC's render-buffer
+    /// reuse.
+    private static func deepCopyPCMBuffer(_ source: AVAudioPCMBuffer) -> AVAudioPCMBuffer? {
+        guard let copy = AVAudioPCMBuffer(
+            pcmFormat: source.format,
+            frameCapacity: source.frameCapacity
+        ) else { return nil }
+        copy.frameLength = source.frameLength
+        let frameLength = Int(source.frameLength)
+        let channelCount = Int(source.format.channelCount)
+        if let src = source.int16ChannelData, let dst = copy.int16ChannelData {
+            for ch in 0..<channelCount {
+                memcpy(dst[ch], src[ch], frameLength * MemoryLayout<Int16>.size)
+            }
+        } else if let src = source.floatChannelData, let dst = copy.floatChannelData {
+            for ch in 0..<channelCount {
+                memcpy(dst[ch], src[ch], frameLength * MemoryLayout<Float>.size)
+            }
+        } else if let src = source.int32ChannelData, let dst = copy.int32ChannelData {
+            for ch in 0..<channelCount {
+                memcpy(dst[ch], src[ch], frameLength * MemoryLayout<Int32>.size)
+            }
+        }
+        return copy
+    }
+}

package/ios/TracksRecorder/PipelineHost.swift

@@ -0,0 +1,56 @@
+//
+// Copyright © 2026 Stream.io Inc. All rights reserved.
+//
+
+import AVFoundation
+import CoreMedia
+import Foundation
+
+/// Internal coordination contract between `TracksRecorderManager` and its
+/// per-kind pipelines (`VideoPipeline`, `AudioPipeline`). The pipelines own
+/// their encoder + sink + drain logic; the host owns lifecycle, the asset
+/// writer, the writer-start gate, the shared time origin, and the terminal-
+/// completion barrier.
+///
+/// Every method on this protocol is called from the host's serial queue —
+/// pipelines must `host.queue.async { ... }` before calling back into the
+/// host. The protocol is class-bound so pipelines can hold a `weak`
+/// reference and avoid retain cycles.
+internal protocol PipelineHost: AnyObject {
+    /// The recorder's serial dispatch queue.
+    var queue: DispatchQueue { get }
+
+    var assetWriter: AVAssetWriter? { get }
+
+    var isRecording: Bool { get }
+
+    /// Returns the recording's shared time origin in nanoseconds. The first
+    /// pipeline to deliver a sample seeds the origin with its timestamp;
+    /// subsequent calls return the established value.
+    func seedOriginNs(_ timestampNs: UInt64) -> UInt64
+
+    /// Pipeline has added an input to the writer. The host decrements its
+    /// pending-pipeline counter and starts the writer once all expected
+    /// pipelines have reported their input.
+    func onTrackAdded()
+
+    func onFatalError(_ error: NSError)
+}
+
+/// Maps an absolute monotonic timestamp (nanoseconds) to presentation time
+/// relative to the recording's shared origin. The first sample from either
+/// pipeline seeds the origin via `host.seedOriginNs`; later samples use
+/// elapsed = timestamp − origin (clamped to 0).
+internal func presentationTime(host: PipelineHost, timestampNs: UInt64) -> CMTime {
+    let origin = host.seedOriginNs(timestampNs)
+    let elapsed: Int64 = timestampNs >= origin ? Int64(timestampNs - origin) : 0
+    return CMTime(value: elapsed, timescale: 1_000_000_000)
+}
+
+internal func makeRecorderError(_ message: String, code: Int) -> NSError {
+    NSError(
+        domain: "io.stream.video.tracks-recorder",
+        code: code,
+        userInfo: [NSLocalizedDescriptionKey: message]
+    )
+}