@sbhjt-gr/react-native-webrtc 137.0.3 → 137.0.5

package/android/src/main/java/com/oney/WebRTCModule/WebRTCModule.java

@@ -27,6 +27,7 @@ import com.oney.WebRTCModule.webrtcutils.H264AndSoftwareVideoEncoderFactory;
 import org.webrtc.AddIceObserver;
 import org.webrtc.AudioProcessingFactory;
 import org.webrtc.AudioTrack;
+import org.webrtc.AudioTrackSink;
 import org.webrtc.CryptoOptions;
 import org.webrtc.EglBase;
 import org.webrtc.IceCandidate;
@@ -52,12 +53,17 @@ import org.webrtc.VideoTrack;
 import org.webrtc.audio.AudioDeviceModule;
 import org.webrtc.audio.JavaAudioDeviceModule;
 
+import java.nio.ByteBuffer;
+import java.nio.ByteOrder;
+import java.nio.IntBuffer;
+import java.nio.ShortBuffer;
 import java.util.ArrayList;
 import java.util.HashMap;
 import java.util.List;
 import java.util.Map;
 import java.util.Objects;
 import java.util.concurrent.Callable;
+import java.util.concurrent.ConcurrentHashMap;
 import java.util.concurrent.ExecutionException;
 
 @ReactModule(name = "WebRTCModule")
@@ -72,6 +78,7 @@ public class WebRTCModule extends ReactContextBaseJavaModule {
     // Need to expose the peer connection codec factories here to get capabilities
     private final SparseArray<PeerConnectionObserver> mPeerConnectionObservers;
     final Map<String, MediaStream> localStreams;
+    private final Map<String, AudioTrackSink> audioSinkMap = new ConcurrentHashMap<>();
 
     private final GetUserMediaImpl getUserMediaImpl;
 
@@ -479,6 +486,10 @@ public class WebRTCModule extends ReactContextBaseJavaModule {
         return pco.remoteTracks.get(trackId);
     }
 
+    private String audioSinkKey(int pcId, String trackId) {
+        return pcId + ":" + trackId;
+    }
+
     MediaStreamTrack getLocalTrack(String trackId) {
         return getUserMediaImpl.getTrack(trackId);
     }
@@ -971,9 +982,98 @@ public class WebRTCModule extends ReactContextBaseJavaModule {
         });
     }
 
+    private class AudioSamplesInterceptor implements AudioTrackSink {
+        private final String trackId;
+
+        AudioSamplesInterceptor(String trackId) {
+            this.trackId = trackId;
+        }
+
+        @Override
+        public void onData(
+                ByteBuffer audioData,
+                int bitsPerSample,
+                int sampleRate,
+                int numberOfChannels,
+                int numberOfFrames,
+                long absoluteCaptureTimestampMs) {
+            WritableMap params = Arguments.createMap();
+            params.putString("trackId", trackId);
+            params.putArray("samples", convertSamples(audioData, bitsPerSample));
+            params.putInt("sampleRate", sampleRate);
+            params.putInt("channels", numberOfChannels);
+            params.putInt("bitsPerSample", bitsPerSample);
+            params.putInt("framesPerBuffer", numberOfFrames);
+            params.putDouble("timestamp", (double) absoluteCaptureTimestampMs);
+
+            sendEvent("audioSamples", params);
+        }
+
+        private WritableArray convertSamples(ByteBuffer audioData, int bitsPerSample) {
+            WritableArray samples = Arguments.createArray();
+            if (audioData == null) {
+                return samples;
+            }
+
+            ByteBuffer buffer = audioData.slice().order(ByteOrder.LITTLE_ENDIAN);
+
+            if (bitsPerSample == 16) {
+                ShortBuffer shortBuffer = buffer.asShortBuffer();
+                while (shortBuffer.hasRemaining()) {
+                    samples.pushDouble(shortBuffer.get());
+                }
+            } else if (bitsPerSample == 32) {
+                IntBuffer intBuffer = buffer.asIntBuffer();
+                while (intBuffer.hasRemaining()) {
+                    samples.pushDouble(intBuffer.get());
+                }
+            } else {
+                while (buffer.hasRemaining()) {
+                    samples.pushDouble(buffer.get());
+                }
+            }
+
+            return samples;
+        }
+    }
+
     @ReactMethod
     public void mediaStreamTrackEnableAudioSink(int pcId, String id, boolean enabled) {
-        Log.d(TAG, "mediaStreamTrackEnableAudioSink() pcId=" + pcId + " trackId=" + id + " enabled=" + enabled + " - Not fully implemented on Android");
+        ThreadUtils.runOnExecutor(() -> {
+            String sinkKey = audioSinkKey(pcId, id);
+
+            if (enabled) {
+                if (audioSinkMap.containsKey(sinkKey)) {
+                    return;
+                }
+
+                MediaStreamTrack track = getTrack(pcId, id);
+                if (track == null) {
+                    Log.d(TAG, "mediaStreamTrackEnableAudioSink() could not find track " + id);
+                    return;
+                }
+
+                if (!(track instanceof AudioTrack)) {
+                    Log.d(TAG, "mediaStreamTrackEnableAudioSink() track is not an AudioTrack!");
+                    return;
+                }
+
+                AudioSamplesInterceptor sink = new AudioSamplesInterceptor(id);
+                ((AudioTrack) track).addSink(sink);
+                audioSinkMap.put(sinkKey, sink);
+            } else {
+                AudioTrackSink sink = audioSinkMap.remove(sinkKey);
+
+                if (sink == null) {
+                    return;
+                }
+
+                MediaStreamTrack track = getTrack(pcId, id);
+                if (track instanceof AudioTrack) {
+                    ((AudioTrack) track).removeSink(sink);
+                }
+            }
+        });
     }
 
     /**

package/ios/RCTWebRTC/WLVAudioDevice.h

@@ -0,0 +1,8 @@
+#import <Foundation/Foundation.h>
+#import <WebRTC/WebRTC.h>
+
+@interface WLVAudioDevice : NSObject<RTCAudioDevice>
+
+- (void)pushAudioData:(NSData *)data sampleRate:(double)sampleRate channels:(NSInteger)channels;
+
+@end

package/ios/RCTWebRTC/WLVAudioDevice.m

@@ -0,0 +1,201 @@
+#import "WLVAudioDevice.h"
+#import <AudioToolbox/AudioToolbox.h>
+#import <math.h>
+
+static const double kDefaultSampleRate = 48000.0;
+static const NSInteger kDefaultChannels = 1;
+static const NSTimeInterval kDefaultBufferDuration = 0.02;
+
+@interface WLVAudioDevice ()
+
+@property(nonatomic, weak) id<RTCAudioDeviceDelegate> delegate;
+@property(nonatomic) BOOL initialized;
+@property(nonatomic) BOOL playoutInitialized;
+@property(nonatomic) BOOL recordingInitialized;
+@property(nonatomic) BOOL playing;
+@property(nonatomic) BOOL recording;
+@property(nonatomic) double inRate;
+@property(nonatomic) NSInteger inChannels;
+@property(nonatomic) double outRate;
+@property(nonatomic) NSInteger outChannels;
+@property(nonatomic) NSTimeInterval inBufferDuration;
+@property(nonatomic) NSTimeInterval outBufferDuration;
+@property(nonatomic) NSTimeInterval inLatency;
+@property(nonatomic) NSTimeInterval outLatency;
+@property(nonatomic) double sampleTime;
+@property(nonatomic, strong) dispatch_queue_t queue;
+
+@end
+
+@implementation WLVAudioDevice
+
+- (instancetype)init {
+    self = [super init];
+    if (self) {
+        _inRate = kDefaultSampleRate;
+        _outRate = kDefaultSampleRate;
+        _inChannels = kDefaultChannels;
+        _outChannels = kDefaultChannels;
+        _inBufferDuration = kDefaultBufferDuration;
+        _outBufferDuration = kDefaultBufferDuration;
+        _queue = dispatch_queue_create("webrtc.virtual.audio", DISPATCH_QUEUE_SERIAL);
+    }
+    return self;
+}
+
+- (void)pushAudioData:(NSData *)data sampleRate:(double)sampleRate channels:(NSInteger)channels {
+    if (!self.recording || !self.delegate || data.length == 0) {
+        return;
+    }
+    NSInteger bytesPerFrame = channels * sizeof(int16_t);
+    if (bytesPerFrame == 0) {
+        return;
+    }
+    UInt32 frames = (UInt32)(data.length / bytesPerFrame);
+    if (frames == 0) {
+        return;
+    }
+    [self updateInputRate:sampleRate channels:channels];
+    NSData *chunk = [data copy];
+    dispatch_async(self.queue, ^{
+        if (!self.recording || !self.delegate) {
+            return;
+        }
+        AudioBuffer buffer;
+        buffer.mNumberChannels = (UInt32)channels;
+        buffer.mDataByteSize = (UInt32)chunk.length;
+        buffer.mData = (void *)chunk.bytes;
+        AudioBufferList list;
+        list.mNumberBuffers = 1;
+        list.mBuffers[0] = buffer;
+        AudioUnitRenderActionFlags flags = 0;
+        AudioTimeStamp stamp;
+        memset(&stamp, 0, sizeof(AudioTimeStamp));
+        stamp.mFlags = kAudioTimeStampSampleTimeValid;
+        stamp.mSampleTime = self.sampleTime;
+        self.sampleTime += frames;
+        RTC_OBJC_TYPE(RTCAudioDeviceDeliverRecordedDataBlock) block = self.delegate.deliverRecordedData;
+        if (block) {
+            block(&flags, &stamp, 0, frames, &list, NULL, nil);
+        }
+    });
+}
+
+- (void)updateInputRate:(double)rate channels:(NSInteger)channels {
+    BOOL rateChanged = fabs(self.inRate - rate) > 0.1;
+    BOOL channelChanged = self.inChannels != channels;
+    if (rateChanged) {
+        self.inRate = rate;
+    }
+    if (channelChanged) {
+        self.inChannels = channels;
+    }
+    if ((rateChanged || channelChanged) && self.delegate) {
+        [self.delegate dispatchAsync:^{
+            [self.delegate notifyAudioInputParametersChange];
+        }];
+    }
+}
+
+#pragma mark - RTCAudioDevice
+
+- (double)deviceInputSampleRate {
+    return self.inRate;
+}
+
+- (NSTimeInterval)inputIOBufferDuration {
+    return self.inBufferDuration;
+}
+
+- (NSInteger)inputNumberOfChannels {
+    return self.inChannels;
+}
+
+- (NSTimeInterval)inputLatency {
+    return self.inLatency;
+}
+
+- (double)deviceOutputSampleRate {
+    return self.outRate;
+}
+
+- (NSTimeInterval)outputIOBufferDuration {
+    return self.outBufferDuration;
+}
+
+- (NSInteger)outputNumberOfChannels {
+    return self.outChannels;
+}
+
+- (NSTimeInterval)outputLatency {
+    return self.outLatency;
+}
+
+- (BOOL)isInitialized {
+    return self.initialized;
+}
+
+- (BOOL)initializeWithDelegate:(id<RTCAudioDeviceDelegate>)delegate {
+    self.delegate = delegate;
+    self.initialized = YES;
+    return YES;
+}
+
+- (BOOL)terminateDevice {
+    self.delegate = nil;
+    self.initialized = NO;
+    self.playoutInitialized = NO;
+    self.recordingInitialized = NO;
+    self.playing = NO;
+    self.recording = NO;
+    self.sampleTime = 0;
+    return YES;
+}
+
+- (BOOL)isPlayoutInitialized {
+    return self.playoutInitialized;
+}
+
+- (BOOL)initializePlayout {
+    self.playoutInitialized = YES;
+    return YES;
+}
+
+- (BOOL)isPlaying {
+    return self.playing;
+}
+
+- (BOOL)startPlayout {
+    self.playing = YES;
+    return YES;
+}
+
+- (BOOL)stopPlayout {
+    self.playing = NO;
+    return YES;
+}
+
+- (BOOL)isRecordingInitialized {
+    return self.recordingInitialized;
+}
+
+- (BOOL)initializeRecording {
+    self.recordingInitialized = YES;
+    return YES;
+}
+
+- (BOOL)isRecording {
+    return self.recording;
+}
+
+- (BOOL)startRecording {
+    self.recording = YES;
+    return YES;
+}
+
+- (BOOL)stopRecording {
+    self.recording = NO;
+    return YES;
+}
+
+@end

package/ios/RCTWebRTC/WebRTCModule+RTCMediaStream.m

@@ -1,5 +1,10 @@
 #import <objc/runtime.h>
 
+#import <AVFoundation/AVFoundation.h>
+#import <math.h>
+#import <limits.h>
+#import <stdint.h>
+
 #import <WebRTC/RTCCameraVideoCapturer.h>
 #import <WebRTC/RTCMediaConstraints.h>
 #import <WebRTC/RTCMediaStreamTrack.h>
@@ -16,8 +21,109 @@
 #import "TrackCapturerEventsEmitter.h"
 #import "VideoCaptureController.h"
 
+@interface AudioSamplesRenderer : NSObject<RTCAudioRenderer>
+
+- (instancetype)initWithModule:(WebRTCModule *)module trackId:(NSString *)trackId;
+
+@end
+
+@implementation AudioSamplesRenderer {
+    __weak WebRTCModule *_module;
+    NSString *_trackId;
+}
+
+- (instancetype)initWithModule:(WebRTCModule *)module trackId:(NSString *)trackId {
+    self = [super init];
+    if (self) {
+        _module = module;
+        _trackId = [trackId copy];
+    }
+    return self;
+}
+
+- (void)render:(AVAudioPCMBuffer *)pcmBuffer {
+    WebRTCModule *module = _module;
+    if (!module) {
+        return;
+    }
+
+    AVAudioFrameCount frameLength = pcmBuffer.frameLength;
+    if (frameLength == 0) {
+        return;
+    }
+
+    UInt32 channelCount = pcmBuffer.format.channelCount;
+    if (channelCount == 0) {
+        return;
+    }
+
+    AVAudioCommonFormat format = pcmBuffer.format.commonFormat;
+    NSUInteger totalSamples = (NSUInteger)frameLength * (NSUInteger)channelCount;
+    NSMutableArray<NSNumber *> *samples = [NSMutableArray arrayWithCapacity:totalSamples];
+    int bitsPerSample = (int)pcmBuffer.format.streamDescription->mBitsPerChannel;
+
+    if (format == AVAudioPCMFormatInt16) {
+        int16_t **channelData = pcmBuffer.int16ChannelData;
+        if (!channelData) {
+            return;
+        }
+        for (AVAudioFrameCount frame = 0; frame < frameLength; frame++) {
+            for (UInt32 channel = 0; channel < channelCount; channel++) {
+                [samples addObject:@(channelData[channel][frame])];
+            }
+        }
+    } else if (format == AVAudioPCMFormatFloat32) {
+        float **channelData = pcmBuffer.floatChannelData;
+        if (!channelData) {
+            return;
+        }
+        bitsPerSample = 16;
+        for (AVAudioFrameCount frame = 0; frame < frameLength; frame++) {
+            for (UInt32 channel = 0; channel < channelCount; channel++) {
+                float value = channelData[channel][frame];
+                float clipped = fminf(fmaxf(value, -1.0f), 1.0f);
+                int16_t sample = (int16_t)lrintf(clipped * (float)INT16_MAX);
+                [samples addObject:@(sample)];
+            }
+        }
+    } else if (format == AVAudioPCMFormatInt32) {
+        int32_t **channelData = pcmBuffer.int32ChannelData;
+        if (!channelData) {
+            return;
+        }
+        bitsPerSample = 32;
+        for (AVAudioFrameCount frame = 0; frame < frameLength; frame++) {
+            for (UInt32 channel = 0; channel < channelCount; channel++) {
+                int32_t value = channelData[channel][frame];
+                [samples addObject:@(value)];
+            }
+        }
+    } else {
+        return;
+    }
+
+    NSTimeInterval timestampMs = [[NSDate date] timeIntervalSince1970] * 1000.0;
+
+    [module sendEventWithName:kEventAudioSamples
+                         body:@{
+                             @"trackId" : _trackId,
+                             @"samples" : samples,
+                             @"sampleRate" : @(pcmBuffer.format.sampleRate),
+                             @"channels" : @(channelCount),
+                             @"bitsPerSample" : @(bitsPerSample),
+                             @"framesPerBuffer" : @(frameLength),
+                             @"timestamp" : @(timestampMs)
+                         }];
+}
+
+@end
+
 @implementation WebRTCModule (RTCMediaStream)
 
+- (NSString *)audioSinkKeyForPcId:(NSNumber *)pcId trackId:(NSString *)trackId {
+    return [NSString stringWithFormat:@"%@:%@", pcId, trackId];
+}
+
 - (VideoEffectProcessor *)videoEffectProcessor {
     return objc_getAssociatedObject(self, _cmd);
 }
@@ -217,6 +323,53 @@ RCT_EXPORT_METHOD(getDisplayMedia : (RCTPromiseResolveBlock)resolve rejecter : (
 #endif
 }
 
+RCT_EXPORT_METHOD(createVirtualAudioTrack : (RCTPromiseResolveBlock)resolve rejecter : (RCTPromiseRejectBlock)reject) {
+#if TARGET_OS_TV
+    reject(@"unsupported_platform", @"tvOS is not supported", nil);
+    return;
+#else
+    RTCPeerConnectionFactory *factory = [self virtualFactory];
+    if (!factory) {
+        reject(@"virtual_factory_error", @"Factory unavailable", nil);
+        return;
+    }
+    NSString *trackId = [[NSUUID UUID] UUIDString];
+    RTCAudioSource *source = [factory audioSourceWithConstraints:nil];
+    RTCAudioTrack *track = [factory audioTrackWithSource:source trackId:trackId];
+    NSArray *components = [self createMediaStream:@[ track ]];
+    resolve(@{ @"streamId" : components[0], @"tracks" : components[1] });
+#endif
+}
+
+RCT_EXPORT_METHOD(pushVirtualAudioSamples
+                  : (nonnull NSArray<NSNumber *> *)samples
+                  sampleRate
+                  : (nonnull NSNumber *)sampleRate
+                  channels
+                  : (nonnull NSNumber *)channels) {
+#if TARGET_OS_TV
+    return;
+#else
+    if (samples.count == 0) {
+        return;
+    }
+    if (!self.virtualAudioDevice) {
+        [self virtualFactory];
+    }
+    if (!self.virtualAudioDevice) {
+        return;
+    }
+    NSUInteger length = samples.count * sizeof(int16_t);
+    NSMutableData *data = [NSMutableData dataWithLength:length];
+    int16_t *buffer = data.mutableBytes;
+    NSUInteger idx = 0;
+    for (NSNumber *value in samples) {
+        buffer[idx++] = (int16_t)value.shortValue;
+    }
+    [self.virtualAudioDevice pushAudioData:data sampleRate:sampleRate.doubleValue channels:channels.integerValue];
+#endif
+}
+
 /**
  * Implements {@code getUserMedia}. Note that at this point constraints have
  * been normalized and permissions have been granted. The constraints only
@@ -497,7 +650,36 @@ RCT_EXPORT_METHOD(mediaStreamTrackSetPlaybackEnabled : (nonnull NSNumber *)pcId
 }
 
 RCT_EXPORT_METHOD(mediaStreamTrackEnableAudioSink : (nonnull NSNumber *)pcId : (nonnull NSString *)trackID : (BOOL)enabled) {
-    RCTLogInfo(@"mediaStreamTrackEnableAudioSink: pcId=%@, trackId=%@, enabled=%d - Not fully implemented on iOS", pcId, trackID, enabled);
+    NSString *key = [self audioSinkKeyForPcId:pcId trackId:trackID];
+
+    if (enabled) {
+        if (self.audioRenderers[key]) {
+            return;
+        }
+
+        RTCMediaStreamTrack *track = [self trackForId:trackID pcId:pcId];
+        if (track == nil || ![track.kind isEqualToString:@"audio"]) {
+            return;
+        }
+
+        RTCAudioTrack *audioTrack = (RTCAudioTrack *)track;
+        AudioSamplesRenderer *renderer = [[AudioSamplesRenderer alloc] initWithModule:self trackId:trackID];
+        self.audioRenderers[key] = renderer;
+        [audioTrack addRenderer:renderer];
+    } else {
+        id<RTCAudioRenderer> renderer = self.audioRenderers[key];
+        if (!renderer) {
+            return;
+        }
+
+        RTCMediaStreamTrack *track = [self trackForId:trackID pcId:pcId];
+        if ([track isKindOfClass:[RTCAudioTrack class]]) {
+            RTCAudioTrack *audioTrack = (RTCAudioTrack *)track;
+            [audioTrack removeRenderer:renderer];
+        }
+
+        [self.audioRenderers removeObjectForKey:key];
+    }
 }
 
 RCT_EXPORT_METHOD(mediaStreamTrackSetVideoEffects

package/ios/RCTWebRTC/WebRTCModule.h

@@ -36,10 +36,13 @@ static NSString *const kEventAudioSamples = @"audioSamples";
 @property(nonatomic, strong) NSMutableDictionary<NSString *, RTCMediaStream *> *localStreams;
 @property(nonatomic, strong) NSMutableDictionary<NSString *, RTCMediaStreamTrack *> *localTracks;
 
+@property(nonatomic, strong) NSMutableDictionary<NSString *, id<RTCAudioRenderer>> *audioRenderers;
+
 @property(nonatomic, strong) NSMutableDictionary<NSString *, RTCFrameCryptor *> *frameCryptors;
 @property(nonatomic, strong) NSMutableDictionary<NSString *, RTCFrameCryptorKeyProvider *> *keyProviders;
 @property(nonatomic, strong) NSMutableDictionary<NSString *, RTCDataPacketCryptor *> *dataPacketCryptors;
 
 - (RTCMediaStream *)streamForReactTag:(NSString *)reactTag;
+- (RTCPeerConnectionFactory *)virtualFactory;
 
 @end

package/ios/RCTWebRTC/WebRTCModule.m

@@ -10,8 +10,13 @@
 #import "WebRTCModule+RTCPeerConnection.h"
 #import "WebRTCModule.h"
 #import "WebRTCModuleOptions.h"
+#import "WLVAudioDevice.h"
 
 @interface WebRTCModule ()
+
+@property(nonatomic, strong) WLVAudioDevice *virtualAudioDevice;
+@property(nonatomic, strong) RTCPeerConnectionFactory *virtualPeerConnectionFactory;
+
 @end
 
 @implementation WebRTCModule
@@ -23,6 +28,8 @@
 - (void)dealloc {
     [_localTracks removeAllObjects];
     _localTracks = nil;
+    [_audioRenderers removeAllObjects];
+    _audioRenderers = nil;
     [_localStreams removeAllObjects];
     _localStreams = nil;
 
@@ -92,6 +99,7 @@
         _peerConnections = [NSMutableDictionary new];
         _localStreams = [NSMutableDictionary new];
         _localTracks = [NSMutableDictionary new];
+    _audioRenderers = [NSMutableDictionary new];
 
         _frameCryptors = [NSMutableDictionary new];
         _keyProviders = [NSMutableDictionary new];
@@ -125,6 +133,16 @@ RCT_EXPORT_MODULE();
     return _workerQueue;
 }
 
+- (RTCPeerConnectionFactory *)virtualFactory {
+    if (_virtualPeerConnectionFactory == nil) {
+        self.virtualAudioDevice = [[WLVAudioDevice alloc] init];
+        _virtualPeerConnectionFactory = [[RTCPeerConnectionFactory alloc] initWithEncoderFactory:_encoderFactory
+                                                                                       decoderFactory:_decoderFactory
+                                                                                          audioDevice:self.virtualAudioDevice];
+    }
+    return _virtualPeerConnectionFactory;
+}
+
 - (NSArray<NSString *> *)supportedEvents {
     return @[
         kEventPeerConnectionSignalingStateChanged,

package/ios/RCTWebRTC.xcodeproj/project.pbxproj

@@ -22,6 +22,7 @@
 		4EE3A8D225B841DD00FAA24A /* CaptureController.m in Sources */ = {isa = PBXBuildFile; fileRef = 4EE3A8C925B841DD00FAA24A /* CaptureController.m */; };
 		4EE3A8D325B841DD00FAA24A /* ScreenCaptureController.m in Sources */ = {isa = PBXBuildFile; fileRef = 4EE3A8CC25B841DD00FAA24A /* ScreenCaptureController.m */; };
 		4EE3A8D425B841DD00FAA24A /* ScreenCapturer.m in Sources */ = {isa = PBXBuildFile; fileRef = 4EE3A8CD25B841DD00FAA24A /* ScreenCapturer.m */; };
+		E0C37C1F2D8D1C6A009E2E63 /* WLVAudioDevice.m in Sources */ = {isa = PBXBuildFile; fileRef = E0C37C1E2D8D1C6A009E2E63 /* WLVAudioDevice.m */; };
 		D3FF699919D2664B25C9D458 /* Pods_RCTWebRTC.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = A20F721AD842563B66292D5B /* Pods_RCTWebRTC.framework */; };
 		D74EF94829652169000742E1 /* TrackCapturerEventsEmitter.m in Sources */ = {isa = PBXBuildFile; fileRef = D74EF94629652169000742E1 /* TrackCapturerEventsEmitter.m */; };
 		D7F0711E2C6DC91F0031F594 /* WebRTCModule+RTCAudioSession.m in Sources */ = {isa = PBXBuildFile; fileRef = D7F0711D2C6DC91F0031F594 /* WebRTCModule+RTCAudioSession.m */; };
@@ -72,6 +73,8 @@
 		4EE3A8CC25B841DD00FAA24A /* ScreenCaptureController.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; name = ScreenCaptureController.m; path = RCTWebRTC/ScreenCaptureController.m; sourceTree = SOURCE_ROOT; };
 		4EE3A8CD25B841DD00FAA24A /* ScreenCapturer.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; name = ScreenCapturer.m; path = RCTWebRTC/ScreenCapturer.m; sourceTree = SOURCE_ROOT; };
 		4EE3A8CE25B841DD00FAA24A /* SocketConnection.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = SocketConnection.h; path = RCTWebRTC/SocketConnection.h; sourceTree = SOURCE_ROOT; };
+		E0C37C1D2D8D1C6A009E2E63 /* WLVAudioDevice.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = WLVAudioDevice.h; path = RCTWebRTC/WLVAudioDevice.h; sourceTree = SOURCE_ROOT; };
+		E0C37C1E2D8D1C6A009E2E63 /* WLVAudioDevice.m */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.objc; name = WLVAudioDevice.m; path = RCTWebRTC/WLVAudioDevice.m; sourceTree = SOURCE_ROOT; };
 		A20F721AD842563B66292D5B /* Pods_RCTWebRTC.framework */ = {isa = PBXFileReference; explicitFileType = wrapper.framework; includeInIndex = 0; path = Pods_RCTWebRTC.framework; sourceTree = BUILT_PRODUCTS_DIR; };
 		D74EF94529652148000742E1 /* CapturerEventsDelegate.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = CapturerEventsDelegate.h; path = RCTWebRTC/CapturerEventsDelegate.h; sourceTree = SOURCE_ROOT; };
 		D74EF94629652169000742E1 /* TrackCapturerEventsEmitter.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; name = TrackCapturerEventsEmitter.m; path = RCTWebRTC/TrackCapturerEventsEmitter.m; sourceTree = SOURCE_ROOT; };
@@ -157,6 +160,8 @@
 				4EE3A8CD25B841DD00FAA24A /* ScreenCapturer.m */,
 				4EE3A8CE25B841DD00FAA24A /* SocketConnection.h */,
 				4EE3A8C825B841DD00FAA24A /* SocketConnection.m */,
+				E0C37C1D2D8D1C6A009E2E63 /* WLVAudioDevice.h */,
+				E0C37C1E2D8D1C6A009E2E63 /* WLVAudioDevice.m */,
 				DEC96579264176DF0052DB35 /* DataChannelWrapper.h */,
 				DEC96576264176C10052DB35 /* DataChannelWrapper.m */,
 			);
@@ -255,6 +260,7 @@
 			isa = PBXSourcesBuildPhase;
 			buildActionMask = 2147483647;
 			files = (
+				E0C37C1F2D8D1C6A009E2E63 /* WLVAudioDevice.m in Sources */,
 				4EE3A8C525B8417800FAA24A /* WebRTCModule+VideoTrackAdapter.m in Sources */,
 				4EE3A8BA25B8415900FAA24A /* WebRTCModule+RTCDataChannel.m in Sources */,
 				4EE3A8B625B8414A00FAA24A /* WebRTCModule+Permissions.m in Sources */,

package/lib/commonjs/EventEmitter.js

@@ -51,7 +51,9 @@ function removeListener(listener) {
 }
 function removeListenerForEvent(listener, eventName) {
   const subs = _subscriptions.get(listener);
-  if (!subs) return;
+  if (!subs) {
+    return;
+  }
   const remaining = [];
   subs.forEach(sub => {
     if (sub.eventType === eventName) {

package/lib/commonjs/EventEmitter.js.map

@@ -1 +1 @@
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nativeEmitter = new NativeEventEmitter(WebRTCModule);\n\nconst NATIVE_EVENTS = [\n    'peerConnectionSignalingStateChanged',\n    'peerConnectionStateChanged',\n    'peerConnectionOnRenegotiationNeeded',\n    'peerConnectionIceConnectionChanged',\n    'peerConnectionIceGatheringChanged',\n    'peerConnectionGotICECandidate',\n    'peerConnectionDidOpenDataChannel',\n    'peerConnectionOnRemoveTrack',\n    'peerConnectionOnTrack',\n    'dataChannelStateChanged',\n    'dataChannelReceiveMessage',\n    'dataChannelDidChangeBufferedAmount',\n    'mediaStreamTrackMuteChanged',\n    'mediaStreamTrackEnded',\n    'frameCryptionStateChanged',\n    'audioSamples',\n];\n\nconst eventEmitter = new EventEmitter();\n\nexport function setupNativeEvents() {\n    for (const eventName of NATIVE_EVENTS) {\n        nativeEmitter.addListener(eventName, (...args) => {\n            eventEmitter.emit(eventName, ...args);\n        });\n    }\n}\n\ntype EventHandler = (event: unknown) => void;\ntype Listener = unknown;\n\nconst _subscriptions: Map<Listener, EmitterSubscription[]> = new Map();\n\nexport function addListener(listener: Listener, eventName: string, eventHandler: EventHandler): void {\n    if (!NATIVE_EVENTS.includes(eventName)) {\n        throw new Error(`Invalid event: ${eventName}`);\n    }\n\n    if (!_subscriptions.has(listener)) {\n        _subscriptions.set(listener, []);\n    }\n\n    _subscriptions.get(listener)?.push(eventEmitter.addListener(eventName, eventHandler));\n}\n\nexport function removeListener(listener: Listener): void {\n    _subscriptions.get(listener)?.forEach(sub => {\n        sub.remove();\n    });\n\n    _subscriptions.delete(listener);\n}\n\nexport function removeListenerForEvent(listener: Listener, eventName: string): void {\n    const subs = _subscriptions.get(listener);\n    if (!subs) return;\n    \n    const remaining: EmitterSubscription[] = [];\n    subs.forEach(sub => {\n        if ((sub as any).eventType === eventName) {\n            sub.remove();\n        } else {\n            remaining.push(sub);\n        }\n    });\n    \n    if (remaining.length === 0) {\n        _subscriptions.delete(listener);\n    } else {\n        _subscriptions.set(listener, remaining);\n    }\n}\n"],"mappings":";;;;;;;;;AAAA,IAAAA,YAAA,GAAAC,OAAA;AAEA,IAAAC,aAAA,GAAAC,sBAAA,CAAAF,OAAA;AAA8E,SAAAE,uBAAAC,GAAA,WAAAA,GAAA,IAAAA,GAAA,CAAAC,UAAA,GAAAD,GAAA,KAAAE,OAAA,EAAAF,GAAA;AAD9E;;AAGA,MAAM;EAAEG;AAAa,CAAC,GAAGC,0BAAa;;AAEtC;AACA;AACA,MAAMC,aAAa,GAAG,IAAIC,+BAAkB,CAACH,YAAY,CAAC;AAE1D,MAAMI,aAAa,GAAG,CAClB,qCAAqC,EACrC,4BAA4B,EAC5B,qCAAqC,EACrC,oCAAoC,EACpC,mCAAmC,EACnC,+BAA+B,EAC/B,kCAAkC,EAClC,6BAA6B,EAC7B,uBAAuB,EACvB,yBAAyB,EACzB,2BAA2B,EAC3B,oCAAoC,EACpC,6BAA6B,EAC7B,uBAAuB,EACvB,2BAA2B,EAC3B,cAAc,CACjB;AAED,MAAMC,YAAY,GAAG,IAAIC,qBAAY,CAAC,CAAC;AAEhC,SAASC,iBAAiBA,CAAA,EAAG;EAChC,KAAK,MAAMC,SAAS,IAAIJ,aAAa,EAAE;IACnCF,aAAa,CAACO,WAAW,CAACD,SAAS,EAAE,YAAa;MAAA,SAAAE,IAAA,GAAAC,SAAA,CAAAC,MAAA,EAATC,IAAI,OAAAC,KAAA,CAAAJ,IAAA,GAAAK,IAAA,MAAAA,IAAA,GAAAL,IAAA,EAAAK,IAAA;QAAJF,IAAI,CAAAE,IAAA,IAAAJ,SAAA,CAAAI,IAAA;MAAA;MACzCV,YAAY,CAACW,IAAI,CAACR,SAAS,EAAE,GAAGK,IAAI,CAAC;IACzC,CAAC,CAAC;EACN;AACJ;AAKA,MAAMI,cAAoD,GAAG,IAAIC,GAAG,CAAC,CAAC;AAE/D,SAAST,WAAWA,CAACU,QAAkB,EAAEX,SAAiB,EAAEY,YAA0B,EAAQ;EAAA,IAAAC,kBAAA;EACjG,IAAI,CAACjB,aAAa,CAACkB,QAAQ,CAACd,SAAS,CAAC,EAAE;IACpC,MAAM,IAAIe,KAAK,CAAE,kBAAiBf,SAAU,EAAC,CAAC;EAClD;EAEA,IAAI,CAACS,cAAc,CAACO,GAAG,CAACL,QAAQ,CAAC,EAAE;IAC/BF,cAAc,CAACQ,GAAG,CAACN,QAAQ,EAAE,EAAE,CAAC;EACpC;EAEA,CAAAE,kBAAA,GAAAJ,cAAc,CAACS,GAAG,CAACP,QAAQ,CAAC,cAAAE,kBAAA,uBAA5BA,kBAAA,CAA8BM,IAAI,CAACtB,YAAY,CAACI,WAAW,CAACD,SAAS,EAAEY,YAAY,CAAC,CAAC;AACzF;AAEO,SAASQ,cAAcA,CAACT,QAAkB,EAAQ;EAAA,IAAAU,mBAAA;EACrD,CAAAA,mBAAA,GAAAZ,cAAc,CAACS,GAAG,CAACP,QAAQ,CAAC,cAAAU,mBAAA,uBAA5BA,mBAAA,CAA8BC,OAAO,CAACC,GAAG,IAAI;IACzCA,GAAG,CAACC,MAAM,CAAC,CAAC;EAChB,CAAC,CAAC;EAEFf,cAAc,CAACgB,MAAM,CAACd,QAAQ,CAAC;AACnC;AAEO,SAASe,sBAAsBA,CAACf,QAAkB,EAAEX,SAAiB,EAAQ;EAChF,MAAM2B,IAAI,GAAGlB,cAAc,CAACS,GAAG,CAACP,QAAQ,CAAC;EACzC,IAAI,CAACgB,IAAI,EAAE;EAEX,MAAMC,SAAgC,GAAG,EAAE;EAC3CD,IAAI,CAACL,OAAO,CAACC,GAAG,IAAI;IAChB,IAAKA,GAAG,CAASM,SAAS,KAAK7B,SAAS,EAAE;MACtCuB,GAAG,CAACC,MAAM,CAAC,CAAC;IAChB,CAAC,MAAM;MACHI,SAAS,CAACT,IAAI,CAACI,GAAG,CAAC;IACvB;EACJ,CAAC,CAAC;EAEF,IAAIK,SAAS,CAACxB,MAAM,KAAK,CAAC,EAAE;IACxBK,cAAc,CAACgB,MAAM,CAACd,QAAQ,CAAC;EACnC,CAAC,MAAM;IACHF,cAAc,CAACQ,GAAG,CAACN,QAAQ,EAAEiB,SAAS,CAAC;EAC3C;AACJ"}
+{"version":3,"names":["_reactNative","require","_EventEmitter","_interopRequireDefault","obj","__esModule","default","WebRTCModule","NativeModules","nativeEmitter","NativeEventEmitter","NATIVE_EVENTS","eventEmitter","EventEmitter","setupNativeEvents","eventName","addListener","_len","arguments","length","args","Array","_key","emit","_subscriptions","Map","listener","eventHandler","_subscriptions$get","includes","Error","has","set","get","push","removeListener","_subscriptions$get2","forEach","sub","remove","delete","removeListenerForEvent","subs","remaining","eventType"],"sources":["EventEmitter.ts"],"sourcesContent":["import { NativeModules, NativeEventEmitter, EmitterSubscription } from 'react-native';\n// @ts-ignore\nimport EventEmitter from 'react-native/Libraries/vendor/emitter/EventEmitter';\n\nconst { WebRTCModule } = NativeModules;\n\n// This emitter is going to be used to listen to all the native events (once) and then\n// re-emit them on a JS-only emitter.\nconst nativeEmitter = new NativeEventEmitter(WebRTCModule);\n\nconst NATIVE_EVENTS = [\n    'peerConnectionSignalingStateChanged',\n    'peerConnectionStateChanged',\n    'peerConnectionOnRenegotiationNeeded',\n    'peerConnectionIceConnectionChanged',\n    'peerConnectionIceGatheringChanged',\n    'peerConnectionGotICECandidate',\n    'peerConnectionDidOpenDataChannel',\n    'peerConnectionOnRemoveTrack',\n    'peerConnectionOnTrack',\n    'dataChannelStateChanged',\n    'dataChannelReceiveMessage',\n    'dataChannelDidChangeBufferedAmount',\n    'mediaStreamTrackMuteChanged',\n    'mediaStreamTrackEnded',\n    'frameCryptionStateChanged',\n    'audioSamples',\n];\n\nconst eventEmitter = new EventEmitter();\n\nexport function setupNativeEvents() {\n    for (const eventName of NATIVE_EVENTS) {\n        nativeEmitter.addListener(eventName, (...args) => {\n            eventEmitter.emit(eventName, ...args);\n        });\n    }\n}\n\ntype EventHandler = (event: unknown) => void;\ntype Listener = unknown;\n\nconst _subscriptions: Map<Listener, EmitterSubscription[]> = new Map();\n\nexport function addListener(listener: Listener, eventName: string, eventHandler: EventHandler): void {\n    if (!NATIVE_EVENTS.includes(eventName)) {\n        throw new Error(`Invalid event: ${eventName}`);\n    }\n\n    if (!_subscriptions.has(listener)) {\n        _subscriptions.set(listener, []);\n    }\n\n    _subscriptions.get(listener)?.push(eventEmitter.addListener(eventName, eventHandler));\n}\n\nexport function removeListener(listener: Listener): void {\n    _subscriptions.get(listener)?.forEach(sub => {\n        sub.remove();\n    });\n\n    _subscriptions.delete(listener);\n}\n\nexport function removeListenerForEvent(listener: Listener, eventName: string): void {\n    const subs = _subscriptions.get(listener);\n\n    if (!subs) {\n        return;\n    }\n\n    const remaining: EmitterSubscription[] = [];\n\n    subs.forEach(sub => {\n        if ((sub as any).eventType === eventName) {\n            sub.remove();\n        } else {\n            remaining.push(sub);\n        }\n    });\n\n    if (remaining.length === 0) {\n        _subscriptions.delete(listener);\n    } else {\n        _subscriptions.set(listener, remaining);\n    }\n}\n"],"mappings":";;;;;;;;;AAAA,IAAAA,YAAA,GAAAC,OAAA;AAEA,IAAAC,aAAA,GAAAC,sBAAA,CAAAF,OAAA;AAA8E,SAAAE,uBAAAC,GAAA,WAAAA,GAAA,IAAAA,GAAA,CAAAC,UAAA,GAAAD,GAAA,KAAAE,OAAA,EAAAF,GAAA;AAD9E;;AAGA,MAAM;EAAEG;AAAa,CAAC,GAAGC,0BAAa;;AAEtC;AACA;AACA,MAAMC,aAAa,GAAG,IAAIC,+BAAkB,CAACH,YAAY,CAAC;AAE1D,MAAMI,aAAa,GAAG,CAClB,qCAAqC,EACrC,4BAA4B,EAC5B,qCAAqC,EACrC,oCAAoC,EACpC,mCAAmC,EACnC,+BAA+B,EAC/B,kCAAkC,EAClC,6BAA6B,EAC7B,uBAAuB,EACvB,yBAAyB,EACzB,2BAA2B,EAC3B,oCAAoC,EACpC,6BAA6B,EAC7B,uBAAuB,EACvB,2BAA2B,EAC3B,cAAc,CACjB;AAED,MAAMC,YAAY,GAAG,IAAIC,qBAAY,CAAC,CAAC;AAEhC,SAASC,iBAAiBA,CAAA,EAAG;EAChC,KAAK,MAAMC,SAAS,IAAIJ,aAAa,EAAE;IACnCF,aAAa,CAACO,WAAW,CAACD,SAAS,EAAE,YAAa;MAAA,SAAAE,IAAA,GAAAC,SAAA,CAAAC,MAAA,EAATC,IAAI,OAAAC,KAAA,CAAAJ,IAAA,GAAAK,IAAA,MAAAA,IAAA,GAAAL,IAAA,EAAAK,IAAA;QAAJF,IAAI,CAAAE,IAAA,IAAAJ,SAAA,CAAAI,IAAA;MAAA;MACzCV,YAAY,CAACW,IAAI,CAACR,SAAS,EAAE,GAAGK,IAAI,CAAC;IACzC,CAAC,CAAC;EACN;AACJ;AAKA,MAAMI,cAAoD,GAAG,IAAIC,GAAG,CAAC,CAAC;AAE/D,SAAST,WAAWA,CAACU,QAAkB,EAAEX,SAAiB,EAAEY,YAA0B,EAAQ;EAAA,IAAAC,kBAAA;EACjG,IAAI,CAACjB,aAAa,CAACkB,QAAQ,CAACd,SAAS,CAAC,EAAE;IACpC,MAAM,IAAIe,KAAK,CAAE,kBAAiBf,SAAU,EAAC,CAAC;EAClD;EAEA,IAAI,CAACS,cAAc,CAACO,GAAG,CAACL,QAAQ,CAAC,EAAE;IAC/BF,cAAc,CAACQ,GAAG,CAACN,QAAQ,EAAE,EAAE,CAAC;EACpC;EAEA,CAAAE,kBAAA,GAAAJ,cAAc,CAACS,GAAG,CAACP,QAAQ,CAAC,cAAAE,kBAAA,uBAA5BA,kBAAA,CAA8BM,IAAI,CAACtB,YAAY,CAACI,WAAW,CAACD,SAAS,EAAEY,YAAY,CAAC,CAAC;AACzF;AAEO,SAASQ,cAAcA,CAACT,QAAkB,EAAQ;EAAA,IAAAU,mBAAA;EACrD,CAAAA,mBAAA,GAAAZ,cAAc,CAACS,GAAG,CAACP,QAAQ,CAAC,cAAAU,mBAAA,uBAA5BA,mBAAA,CAA8BC,OAAO,CAACC,GAAG,IAAI;IACzCA,GAAG,CAACC,MAAM,CAAC,CAAC;EAChB,CAAC,CAAC;EAEFf,cAAc,CAACgB,MAAM,CAACd,QAAQ,CAAC;AACnC;AAEO,SAASe,sBAAsBA,CAACf,QAAkB,EAAEX,SAAiB,EAAQ;EAChF,MAAM2B,IAAI,GAAGlB,cAAc,CAACS,GAAG,CAACP,QAAQ,CAAC;EAEzC,IAAI,CAACgB,IAAI,EAAE;IACP;EACJ;EAEA,MAAMC,SAAgC,GAAG,EAAE;EAE3CD,IAAI,CAACL,OAAO,CAACC,GAAG,IAAI;IAChB,IAAKA,GAAG,CAASM,SAAS,KAAK7B,SAAS,EAAE;MACtCuB,GAAG,CAACC,MAAM,CAAC,CAAC;IAChB,CAAC,MAAM;MACHI,SAAS,CAACT,IAAI,CAACI,GAAG,CAAC;IACvB;EACJ,CAAC,CAAC;EAEF,IAAIK,SAAS,CAACxB,MAAM,KAAK,CAAC,EAAE;IACxBK,cAAc,CAACgB,MAAM,CAACd,QAAQ,CAAC;EACnC,CAAC,MAAM;IACHF,cAAc,CAACQ,GAAG,CAACN,QAAQ,EAAEiB,SAAS,CAAC;EAC3C;AACJ"}

package/lib/commonjs/MediaDevices.js

@@ -42,6 +42,19 @@ class MediaDevices extends _index.EventTarget {
   getUserMedia(constraints) {
     return (0, _getUserMedia.default)(constraints);
   }
+  createVirtualAudioTrack() {
+    if (!WebRTCModule.createVirtualAudioTrack) {
+      return Promise.reject(new Error('virtual audio unavailable'));
+    }
+    return WebRTCModule.createVirtualAudioTrack();
+  }
+  pushVirtualAudioSamples(samples, sampleRate, channels) {
+    if (!WebRTCModule.pushVirtualAudioSamples) {
+      return;
+    }
+    const array = samples instanceof Int16Array ? Array.from(samples) : samples.slice();
+    WebRTCModule.pushVirtualAudioSamples(array, sampleRate, channels);
+  }
 }
 
 /**

package/lib/commonjs/MediaDevices.js.map

@@ -1 +1 @@
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package/lib/commonjs/MediaStreamTrack.js.map

@@ -1 +1 @@
-{"version":3,"names":["_index","require","_reactNative","_EventEmitter","_Logger","_interopRequireDefault","_RTCUtil","obj","__esModule","default","_defineProperty","key","value","Object","defineProperty","enumerable","configurable","writable","log","Logger","WebRTCModule","NativeModules","MediaStreamTrack","EventTarget","constructor","info","_constraints","constraints","_enabled","enabled","_settings","settings","_muted","_peerConnectionId","peerConnectionId","_readyState","readyState","id","kind","remote","_registerEvents","Boolean","mediaStreamTrackSetEnabled","muted","stop","_switchCamera","Error","deepClone","deviceId","facingMode","applyConstraints","_setVideoEffects","names","mediaStreamTrackSetVideoEffects","_setVideoEffect","name","_setMutedInternal","dispatchEvent","Event","_setVolume","volume","mediaStreamTrackSetVolume","normalized","normalizeConstraints","video","mediaStreamTrackApplyConstraints","clone","getCapabilities","getConstraints","getSettings","addListener","ev","trackId","debug","_setAudioSinkCallback","callback","_audioSinkCallback","_handleAudioSamples","data","setAudioSink","_audioSinkListenerActive","mediaStreamTrackEnableAudioSink","removeListenerForEvent","setPlaybackEnabled","_playbackEnabled","mediaStreamTrackSetPlaybackEnabled","isPlaybackEnabled","release","removeListener","mediaStreamTrackRelease","exports","proto","prototype","defineEventAttribute"],"sources":["MediaStreamTrack.ts"],"sourcesContent":["import { EventTarget, Event, defineEventAttribute } from 'event-target-shim/index';\nimport { NativeModules, NativeEventEmitter, Platform } from 'react-native';\n\nimport { MediaTrackConstraints } from './Constraints';\nimport { addListener, removeListener, removeListenerForEvent } from './EventEmitter';\nimport Logger from './Logger';\nimport { deepClone, normalizeConstraints } from './RTCUtil';\n\nconst log = new Logger('pc');\nconst { WebRTCModule } = NativeModules;\n\nexport type AudioSampleData = {\n    trackId: string;\n    samples: number[];\n    sampleRate: number;\n    channels: number;\n    timestamp: number;\n}\n\nexport type AudioSinkCallback = (data: AudioSampleData) => void;\n\n\ntype MediaStreamTrackState = 'live' | 'ended';\n\nexport type MediaStreamTrackInfo = {\n    id: string;\n    kind: string;\n    remote: boolean;\n    constraints: object;\n    enabled: boolean;\n    settings: object;\n    peerConnectionId: number;\n    readyState: MediaStreamTrackState;\n}\n\nexport type MediaTrackSettings = {\n    width?: number;\n    height?: number;\n    frameRate?: number;\n    facingMode?: string;\n    deviceId?: string;\n    groupId?: string;\n}\n\ntype MediaStreamTrackEventMap = {\n    ended: Event<'ended'>;\n    mute: Event<'mute'>;\n    unmute: Event<'unmute'>;\n}\n\nexport default class MediaStreamTrack extends EventTarget<MediaStreamTrackEventMap> {\n    _constraints: MediaTrackConstraints;\n    _enabled: boolean;\n    _settings: MediaTrackSettings;\n    _muted: boolean;\n    _peerConnectionId: number;\n    _readyState: MediaStreamTrackState;\n\n    readonly id: string;\n    readonly kind: string;\n    readonly label: string = '';\n    readonly remote: boolean;\n\n    constructor(info: MediaStreamTrackInfo) {\n        super();\n\n        this._constraints = info.constraints || {};\n        this._enabled = info.enabled;\n        this._settings = info.settings || {};\n        this._muted = false;\n        this._peerConnectionId = info.peerConnectionId;\n        this._readyState = info.readyState;\n\n        this.id = info.id;\n        this.kind = info.kind;\n        this.remote = info.remote;\n\n        if (!this.remote) {\n            this._registerEvents();\n        }\n    }\n\n    get enabled(): boolean {\n        return this._enabled;\n    }\n\n    set enabled(enabled: boolean) {\n        if (enabled === this._enabled) {\n            return;\n        }\n\n        this._enabled = Boolean(enabled);\n\n        if (this._readyState === 'ended') {\n            return;\n        }\n\n        WebRTCModule.mediaStreamTrackSetEnabled(this.remote ? this._peerConnectionId : -1, this.id, this._enabled);\n    }\n\n    get muted(): boolean {\n        return this._muted;\n    }\n\n    get readyState(): string {\n        return this._readyState;\n    }\n\n    stop(): void {\n        this.enabled = false;\n        this._readyState = 'ended';\n    }\n\n    /**\n     * Private / custom API for switching the cameras on the fly, without the\n     * need for adding / removing tracks or doing any SDP renegotiation.\n     *\n     * This is how the reference application (AppRTCMobile) implements camera\n     * switching.\n     *\n     * @deprecated Use applyConstraints instead.\n     */\n    _switchCamera(): void {\n        if (this.remote) {\n            throw new Error('Not implemented for remote tracks');\n        }\n\n        if (this.kind !== 'video') {\n            throw new Error('Only implemented for video tracks');\n        }\n\n        const constraints = deepClone(this._settings);\n\n        delete constraints.deviceId;\n        constraints.facingMode = this._settings.facingMode === 'user' ? 'environment' : 'user';\n\n        this.applyConstraints(constraints);\n    }\n\n    _setVideoEffects(names: string[]) {\n        if (this.remote) {\n            throw new Error('Not implemented for remote tracks');\n        }\n\n        if (this.kind !== 'video') {\n            throw new Error('Only implemented for video tracks');\n        }\n\n        WebRTCModule.mediaStreamTrackSetVideoEffects(this.id, names);\n    }\n\n    _setVideoEffect(name: string) {\n        this._setVideoEffects([ name ]);\n    }\n\n    /**\n     * Internal function which is used to set the muted state on remote tracks and\n     * emit the mute / unmute event.\n     *\n     * @param muted Whether the track should be marked as muted / unmuted.\n     */\n    _setMutedInternal(muted: boolean) {\n        if (!this.remote) {\n            throw new Error('Track is not remote!');\n        }\n\n        this._muted = muted;\n        this.dispatchEvent(new Event(muted ? 'mute' : 'unmute'));\n    }\n\n    /**\n     * Custom API for setting the volume on an individual audio track.\n     *\n     * @param volume a gain value in the range of 0-10. defaults to 1.0\n     */\n    _setVolume(volume: number) {\n        if (this.kind !== 'audio') {\n            throw new Error('Only implemented for audio tracks');\n        }\n\n        WebRTCModule.mediaStreamTrackSetVolume(this.remote ? this._peerConnectionId : -1, this.id, volume);\n    }\n\n    /**\n     * Applies a new set of constraints to the track.\n     *\n     * @param constraints An object listing the constraints\n     * to apply to the track's constrainable properties; any existing\n     * constraints are replaced with the new values specified, and any\n     * constrainable properties not included are restored to their default\n     * constraints. If this parameter is omitted, all currently set custom\n     * constraints are cleared.\n     */\n    async applyConstraints(constraints?: MediaTrackConstraints): Promise<void> {\n        if (this.kind !== 'video') {\n            log.info(`Only implemented for video tracks, ignoring applyConstraints for ${this.id}`);\n\n            return;\n        }\n\n        const normalized = normalizeConstraints({ video: constraints ?? true });\n\n        this._settings = await WebRTCModule.mediaStreamTrackApplyConstraints(this.id, normalized.video);\n        this._constraints = constraints ?? {};\n    }\n\n    clone(): never {\n        throw new Error('Not implemented.');\n    }\n\n    getCapabilities(): never {\n        throw new Error('Not implemented.');\n    }\n\n    getConstraints() {\n        return deepClone(this._constraints);\n    }\n\n    getSettings(): MediaTrackSettings {\n        return deepClone(this._settings);\n    }\n\n    _registerEvents(): void {\n        addListener(this, 'mediaStreamTrackEnded', (ev: any) => {\n            if (ev.trackId !== this.id || this._readyState === 'ended') {\n                return;\n            }\n\n            log.debug(`${this.id} mediaStreamTrackEnded`);\n            this._readyState = 'ended';\n\n            this.dispatchEvent(new Event('ended'));\n        });\n    }\n\n    private _audioSinkCallback: AudioSinkCallback | null = null;\n    private _audioSinkListenerActive: boolean = false;\n    private _playbackEnabled: boolean = true;\n\n    _setAudioSinkCallback(callback: AudioSinkCallback | null): void {\n        this._audioSinkCallback = callback;\n    }\n\n    _handleAudioSamples(data: AudioSampleData): void {\n        if (this._audioSinkCallback) {\n            this._audioSinkCallback(data);\n        }\n    }\n\n    setAudioSink(callback: AudioSinkCallback | null): void {\n        if (this.kind !== 'audio') {\n            throw new Error('Audio sink only available for audio tracks');\n        }\n\n        if (!this.remote) {\n            throw new Error('Audio sink only available for remote tracks');\n        }\n\n        this._setAudioSinkCallback(callback);\n\n        if (callback && !this._audioSinkListenerActive) {\n            this._audioSinkListenerActive = true;\n            WebRTCModule.mediaStreamTrackEnableAudioSink(\n                this._peerConnectionId,\n                this.id,\n                true\n            );\n            addListener(this, 'audioSamples', (ev: any) => {\n                if (ev.trackId === this.id) {\n                    this._handleAudioSamples(ev);\n                }\n            });\n        } else if (!callback && this._audioSinkListenerActive) {\n            this._audioSinkListenerActive = false;\n            WebRTCModule.mediaStreamTrackEnableAudioSink(\n                this._peerConnectionId,\n                this.id,\n                false\n            );\n            removeListenerForEvent(this, 'audioSamples');\n        }\n    }\n\n    setPlaybackEnabled(enabled: boolean): void {\n        if (this.kind !== 'audio') {\n            throw new Error('Playback control only available for audio tracks');\n        }\n\n        if (!this.remote) {\n            throw new Error('Playback control only available for remote tracks');\n        }\n\n        this._playbackEnabled = enabled;\n        WebRTCModule.mediaStreamTrackSetPlaybackEnabled(\n            this._peerConnectionId,\n            this.id,\n            enabled\n        );\n    }\n\n    isPlaybackEnabled(): boolean {\n        return this._playbackEnabled;\n    }\n\n    release(): void {\n        if (this.remote) {\n            if (this._audioSinkListenerActive) {\n                this._audioSinkListenerActive = false;\n                WebRTCModule.mediaStreamTrackEnableAudioSink(\n                    this._peerConnectionId,\n                    this.id,\n                    false\n                );\n                removeListenerForEvent(this, 'audioSamples');\n       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this._peerConnectionId : -1, this.id, this._enabled);\n    }\n\n    get muted(): boolean {\n        return this._muted;\n    }\n\n    get readyState(): string {\n        return this._readyState;\n    }\n\n    stop(): void {\n        this.enabled = false;\n        this._readyState = 'ended';\n    }\n\n    /**\n     * Private / custom API for switching the cameras on the fly, without the\n     * need for adding / removing tracks or doing any SDP renegotiation.\n     *\n     * This is how the reference application (AppRTCMobile) implements camera\n     * switching.\n     *\n     * @deprecated Use applyConstraints instead.\n     */\n    _switchCamera(): void {\n        if (this.remote) {\n            throw new Error('Not implemented for remote tracks');\n        }\n\n        if (this.kind !== 'video') {\n            throw new Error('Only implemented for video tracks');\n        }\n\n        const constraints = deepClone(this._settings);\n\n        delete constraints.deviceId;\n        constraints.facingMode = this._settings.facingMode === 'user' ? 'environment' : 'user';\n\n        this.applyConstraints(constraints);\n    }\n\n    _setVideoEffects(names: string[]) {\n        if (this.remote) {\n            throw new Error('Not implemented for remote tracks');\n        }\n\n        if (this.kind !== 'video') {\n            throw new Error('Only implemented for video tracks');\n        }\n\n        WebRTCModule.mediaStreamTrackSetVideoEffects(this.id, names);\n    }\n\n    _setVideoEffect(name: string) {\n        this._setVideoEffects([ name ]);\n    }\n\n    /**\n     * Internal function which is used to set the muted state on remote tracks and\n     * emit the mute / unmute event.\n     *\n     * @param muted Whether the track should be marked as muted / unmuted.\n     */\n    _setMutedInternal(muted: boolean) {\n        if (!this.remote) {\n            throw new Error('Track is not remote!');\n        }\n\n        this._muted = muted;\n        this.dispatchEvent(new Event(muted ? 'mute' : 'unmute'));\n    }\n\n    /**\n     * Custom API for setting the volume on an individual audio track.\n     *\n     * @param volume a gain value in the range of 0-10. defaults to 1.0\n     */\n    _setVolume(volume: number) {\n        if (this.kind !== 'audio') {\n            throw new Error('Only implemented for audio tracks');\n        }\n\n        WebRTCModule.mediaStreamTrackSetVolume(this.remote ? this._peerConnectionId : -1, this.id, volume);\n    }\n\n    /**\n     * Applies a new set of constraints to the track.\n     *\n     * @param constraints An object listing the constraints\n     * to apply to the track's constrainable properties; any existing\n     * constraints are replaced with the new values specified, and any\n     * constrainable properties not included are restored to their default\n     * constraints. If this parameter is omitted, all currently set custom\n     * constraints are cleared.\n     */\n    async applyConstraints(constraints?: MediaTrackConstraints): Promise<void> {\n        if (this.kind !== 'video') {\n            log.info(`Only implemented for video tracks, ignoring applyConstraints for ${this.id}`);\n\n            return;\n        }\n\n        const normalized = normalizeConstraints({ video: constraints ?? true });\n\n        this._settings = await WebRTCModule.mediaStreamTrackApplyConstraints(this.id, normalized.video);\n        this._constraints = constraints ?? {};\n    }\n\n    clone(): never {\n        throw new Error('Not implemented.');\n    }\n\n    getCapabilities(): never {\n        throw new Error('Not implemented.');\n    }\n\n    getConstraints() {\n        return deepClone(this._constraints);\n    }\n\n    getSettings(): MediaTrackSettings {\n        return deepClone(this._settings);\n    }\n\n    _registerEvents(): void {\n        addListener(this, 'mediaStreamTrackEnded', (ev: any) => {\n            if (ev.trackId !== this.id || this._readyState === 'ended') {\n                return;\n            }\n\n            log.debug(`${this.id} mediaStreamTrackEnded`);\n            this._readyState = 'ended';\n\n            this.dispatchEvent(new Event('ended'));\n        });\n    }\n\n    private _audioSinkCallback: AudioSinkCallback | null = null;\n    private _audioSinkListenerActive = false;\n    private _playbackEnabled = true;\n\n    _setAudioSinkCallback(callback: AudioSinkCallback | null): void {\n        this._audioSinkCallback = callback;\n    }\n\n    _handleAudioSamples(data: AudioSampleData): void {\n        if (this._audioSinkCallback) {\n            this._audioSinkCallback(data);\n        }\n    }\n\n    setAudioSink(callback: AudioSinkCallback | null): void {\n        if (this.kind !== 'audio') {\n            throw new Error('Audio sink only available for audio tracks');\n        }\n\n        if (!this.remote) {\n            throw new Error('Audio sink only available for remote tracks');\n        }\n\n        this._setAudioSinkCallback(callback);\n\n        if (callback && !this._audioSinkListenerActive) {\n            this._audioSinkListenerActive = true;\n            WebRTCModule.mediaStreamTrackEnableAudioSink(\n                this._peerConnectionId,\n                this.id,\n                true\n            );\n            addListener(this, 'audioSamples', (ev: any) => {\n                if (ev.trackId === this.id) {\n                    this._handleAudioSamples(ev);\n                }\n            });\n        } else if (!callback && this._audioSinkListenerActive) {\n            this._audioSinkListenerActive = false;\n            WebRTCModule.mediaStreamTrackEnableAudioSink(\n                this._peerConnectionId,\n                this.id,\n                false\n            );\n            removeListenerForEvent(this, 'audioSamples');\n        }\n    }\n\n    setPlaybackEnabled(enabled: boolean): void {\n        if (this.kind !== 'audio') {\n            throw new Error('Playback control only available for audio tracks');\n        }\n\n        if (!this.remote) {\n            throw new Error('Playback control only available for remote tracks');\n        }\n\n        this._playbackEnabled = enabled;\n        WebRTCModule.mediaStreamTrackSetPlaybackEnabled(\n            this._peerConnectionId,\n            this.id,\n            enabled\n        );\n    }\n\n    isPlaybackEnabled(): boolean {\n        return this._playbackEnabled;\n    }\n\n    release(): void {\n        if (this.remote) {\n            if (this._audioSinkListenerActive) {\n                this._audioSinkListenerActive = false;\n                WebRTCModule.mediaStreamTrackEnableAudioSink(\n                    this._peerConnectionId,\n                    this.id,\n                    false\n                );\n                removeListenerForEvent(this, 'audioSamples');\n            }\n\n            return;\n        }\n\n        removeListener(this);\n        WebRTCModule.mediaStreamTrackRelease(this.id);\n    }\n}\n\n/**\n * Define the `onxxx` event handlers.\n */\nconst proto = MediaStreamTrack.prototype;\n\ndefineEventAttribute(proto, 'ended');\ndefineEventAttribute(proto, 'mute');\ndefineEventAttribute(proto, 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package/lib/module/EventEmitter.js

@@ -40,7 +40,9 @@ export function removeListener(listener) {
 }
 export function removeListenerForEvent(listener, eventName) {
   const subs = _subscriptions.get(listener);
-  if (!subs) return;
+  if (!subs) {
+    return;
+  }
   const remaining = [];
   subs.forEach(sub => {
     if (sub.eventType === eventName) {

package/lib/module/EventEmitter.js.map

@@ -1 +1 @@
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Map();\n\nexport function addListener(listener: Listener, eventName: string, eventHandler: EventHandler): void {\n    if (!NATIVE_EVENTS.includes(eventName)) {\n        throw new Error(`Invalid event: ${eventName}`);\n    }\n\n    if (!_subscriptions.has(listener)) {\n        _subscriptions.set(listener, []);\n    }\n\n    _subscriptions.get(listener)?.push(eventEmitter.addListener(eventName, eventHandler));\n}\n\nexport function removeListener(listener: Listener): void {\n    _subscriptions.get(listener)?.forEach(sub => {\n        sub.remove();\n    });\n\n    _subscriptions.delete(listener);\n}\n\nexport function removeListenerForEvent(listener: Listener, eventName: string): void {\n    const subs = _subscriptions.get(listener);\n    if (!subs) return;\n    \n    const remaining: EmitterSubscription[] = [];\n    subs.forEach(sub => {\n        if ((sub as any).eventType === eventName) {\n            sub.remove();\n        } else {\n            remaining.push(sub);\n        }\n    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+{"version":3,"names":["NativeModules","NativeEventEmitter","EventEmitter","WebRTCModule","nativeEmitter","NATIVE_EVENTS","eventEmitter","setupNativeEvents","eventName","addListener","_len","arguments","length","args","Array","_key","emit","_subscriptions","Map","listener","eventHandler","_subscriptions$get","includes","Error","has","set","get","push","removeListener","_subscriptions$get2","forEach","sub","remove","delete","removeListenerForEvent","subs","remaining","eventType"],"sources":["EventEmitter.ts"],"sourcesContent":["import { NativeModules, NativeEventEmitter, EmitterSubscription } from 'react-native';\n// @ts-ignore\nimport EventEmitter from 'react-native/Libraries/vendor/emitter/EventEmitter';\n\nconst { WebRTCModule } = NativeModules;\n\n// This emitter is going to be used to listen to all the native events (once) and then\n// re-emit them on a JS-only emitter.\nconst nativeEmitter = new NativeEventEmitter(WebRTCModule);\n\nconst NATIVE_EVENTS = [\n    'peerConnectionSignalingStateChanged',\n    'peerConnectionStateChanged',\n    'peerConnectionOnRenegotiationNeeded',\n    'peerConnectionIceConnectionChanged',\n    'peerConnectionIceGatheringChanged',\n    'peerConnectionGotICECandidate',\n    'peerConnectionDidOpenDataChannel',\n    'peerConnectionOnRemoveTrack',\n    'peerConnectionOnTrack',\n    'dataChannelStateChanged',\n    'dataChannelReceiveMessage',\n    'dataChannelDidChangeBufferedAmount',\n    'mediaStreamTrackMuteChanged',\n    'mediaStreamTrackEnded',\n    'frameCryptionStateChanged',\n    'audioSamples',\n];\n\nconst eventEmitter = new EventEmitter();\n\nexport function setupNativeEvents() {\n    for (const eventName of NATIVE_EVENTS) {\n        nativeEmitter.addListener(eventName, (...args) => {\n            eventEmitter.emit(eventName, ...args);\n        });\n    }\n}\n\ntype EventHandler = (event: unknown) => void;\ntype Listener = unknown;\n\nconst _subscriptions: Map<Listener, EmitterSubscription[]> = new Map();\n\nexport function addListener(listener: Listener, eventName: string, eventHandler: EventHandler): void {\n    if (!NATIVE_EVENTS.includes(eventName)) {\n        throw new Error(`Invalid event: ${eventName}`);\n    }\n\n    if (!_subscriptions.has(listener)) {\n        _subscriptions.set(listener, []);\n    }\n\n    _subscriptions.get(listener)?.push(eventEmitter.addListener(eventName, eventHandler));\n}\n\nexport function removeListener(listener: Listener): void {\n    _subscriptions.get(listener)?.forEach(sub => {\n        sub.remove();\n    });\n\n    _subscriptions.delete(listener);\n}\n\nexport function removeListenerForEvent(listener: Listener, eventName: string): void {\n    const subs = _subscriptions.get(listener);\n\n    if (!subs) {\n        return;\n    }\n\n    const remaining: EmitterSubscription[] = [];\n\n    subs.forEach(sub => {\n        if ((sub as any).eventType === eventName) {\n            sub.remove();\n        } else {\n            remaining.push(sub);\n        }\n    });\n\n    if (remaining.length === 0) {\n        _subscriptions.delete(listener);\n    } else {\n        _subscriptions.set(listener, remaining);\n    }\n}\n"],"mappings":"AAAA,SAASA,aAAa,EAAEC,kBAAkB,QAA6B,cAAc;AACrF;AACA,OAAOC,YAAY,MAAM,oDAAoD;AAE7E,MAAM;EAAEC;AAAa,CAAC,GAAGH,aAAa;;AAEtC;AACA;AACA,MAAMI,aAAa,GAAG,IAAIH,kBAAkB,CAACE,YAAY,CAAC;AAE1D,MAAME,aAAa,GAAG,CAClB,qCAAqC,EACrC,4BAA4B,EAC5B,qCAAqC,EACrC,oCAAoC,EACpC,mCAAmC,EACnC,+BAA+B,EAC/B,kCAAkC,EAClC,6BAA6B,EAC7B,uBAAuB,EACvB,yBAAyB,EACzB,2BAA2B,EAC3B,oCAAoC,EACpC,6BAA6B,EAC7B,uBAAuB,EACvB,2BAA2B,EAC3B,cAAc,CACjB;AAED,MAAMC,YAAY,GAAG,IAAIJ,YAAY,CAAC,CAAC;AAEvC,OAAO,SAASK,iBAAiBA,CAAA,EAAG;EAChC,KAAK,MAAMC,SAAS,IAAIH,aAAa,EAAE;IACnCD,aAAa,CAACK,WAAW,CAACD,SAAS,EAAE,YAAa;MAAA,SAAAE,IAAA,GAAAC,SAAA,CAAAC,MAAA,EAATC,IAAI,OAAAC,KAAA,CAAAJ,IAAA,GAAAK,IAAA,MAAAA,IAAA,GAAAL,IAAA,EAAAK,IAAA;QAAJF,IAAI,CAAAE,IAAA,IAAAJ,SAAA,CAAAI,IAAA;MAAA;MACzCT,YAAY,CAACU,IAAI,CAACR,SAAS,EAAE,GAAGK,IAAI,CAAC;IACzC,CAAC,CAAC;EACN;AACJ;AAKA,MAAMI,cAAoD,GAAG,IAAIC,GAAG,CAAC,CAAC;AAEtE,OAAO,SAAST,WAAWA,CAACU,QAAkB,EAAEX,SAAiB,EAAEY,YAA0B,EAAQ;EAAA,IAAAC,kBAAA;EACjG,IAAI,CAAChB,aAAa,CAACiB,QAAQ,CAACd,SAAS,CAAC,EAAE;IACpC,MAAM,IAAIe,KAAK,CAAE,kBAAiBf,SAAU,EAAC,CAAC;EAClD;EAEA,IAAI,CAACS,cAAc,CAACO,GAAG,CAACL,QAAQ,CAAC,EAAE;IAC/BF,cAAc,CAACQ,GAAG,CAACN,QAAQ,EAAE,EAAE,CAAC;EACpC;EAEA,CAAAE,kBAAA,GAAAJ,cAAc,CAACS,GAAG,CAACP,QAAQ,CAAC,cAAAE,kBAAA,uBAA5BA,kBAAA,CAA8BM,IAAI,CAACrB,YAAY,CAACG,WAAW,CAACD,SAAS,EAAEY,YAAY,CAAC,CAAC;AACzF;AAEA,OAAO,SAASQ,cAAcA,CAACT,QAAkB,EAAQ;EAAA,IAAAU,mBAAA;EACrD,CAAAA,mBAAA,GAAAZ,cAAc,CAACS,GAAG,CAACP,QAAQ,CAAC,cAAAU,mBAAA,uBAA5BA,mBAAA,CAA8BC,OAAO,CAACC,GAAG,IAAI;IACzCA,GAAG,CAACC,MAAM,CAAC,CAAC;EAChB,CAAC,CAAC;EAEFf,cAAc,CAACgB,MAAM,CAACd,QAAQ,CAAC;AACnC;AAEA,OAAO,SAASe,sBAAsBA,CAACf,QAAkB,EAAEX,SAAiB,EAAQ;EAChF,MAAM2B,IAAI,GAAGlB,cAAc,CAACS,GAAG,CAACP,QAAQ,CAAC;EAEzC,IAAI,CAACgB,IAAI,EAAE;IACP;EACJ;EAEA,MAAMC,SAAgC,GAAG,EAAE;EAE3CD,IAAI,CAACL,OAAO,CAACC,GAAG,IAAI;IAChB,IAAKA,GAAG,CAASM,SAAS,KAAK7B,SAAS,EAAE;MACtCuB,GAAG,CAACC,MAAM,CAAC,CAAC;IAChB,CAAC,MAAM;MACHI,SAAS,CAACT,IAAI,CAACI,GAAG,CAAC;IACvB;EACJ,CAAC,CAAC;EAEF,IAAIK,SAAS,CAACxB,MAAM,KAAK,CAAC,EAAE;IACxBK,cAAc,CAACgB,MAAM,CAACd,QAAQ,CAAC;EACnC,CAAC,MAAM;IACHF,cAAc,CAACQ,GAAG,CAACN,QAAQ,EAAEiB,SAAS,CAAC;EAC3C;AACJ"}

package/lib/module/MediaDevices.js

@@ -35,6 +35,19 @@ class MediaDevices extends EventTarget {
   getUserMedia(constraints) {
     return getUserMedia(constraints);
   }
+  createVirtualAudioTrack() {
+    if (!WebRTCModule.createVirtualAudioTrack) {
+      return Promise.reject(new Error('virtual audio unavailable'));
+    }
+    return WebRTCModule.createVirtualAudioTrack();
+  }
+  pushVirtualAudioSamples(samples, sampleRate, channels) {
+    if (!WebRTCModule.pushVirtualAudioSamples) {
+      return;
+    }
+    const array = samples instanceof Int16Array ? Array.from(samples) : samples.slice();
+    WebRTCModule.pushVirtualAudioSamples(array, sampleRate, channels);
+  }
 }
 
 /**

package/lib/module/MediaDevices.js.map

@@ -1 +1 @@
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+{"version":3,"names":["EventTarget","defineEventAttribute","NativeModules","getDisplayMedia","getUserMedia","WebRTCModule","MediaDevices","enumerateDevices","Promise","resolve","constraints","createVirtualAudioTrack","reject","Error","pushVirtualAudioSamples","samples","sampleRate","channels","array","Int16Array","Array","from","slice","proto","prototype"],"sources":["MediaDevices.ts"],"sourcesContent":["import { EventTarget, Event, defineEventAttribute } from 'event-target-shim/index';\nimport { NativeModules } from 'react-native';\n\nimport { type MediaTrackSettings } from './MediaStreamTrack';\nimport getDisplayMedia from './getDisplayMedia';\nimport getUserMedia, { Constraints } from './getUserMedia';\n\nconst { WebRTCModule } = NativeModules;\n\ntype MediaDevicesEventMap = {\n    devicechange: Event<'devicechange'>\n}\n\ntype VirtualTrackInfo = {\n    enabled: boolean;\n    id: string;\n    kind: string;\n    readyState: string;\n    remote: boolean;\n    settings: MediaTrackSettings;\n}\n\ntype VirtualAudioResponse = {\n    streamId: string;\n    tracks: VirtualTrackInfo[];\n}\n\nclass MediaDevices extends EventTarget<MediaDevicesEventMap> {\n    /**\n     * W3C \"Media Capture and Streams\" compatible {@code enumerateDevices}\n     * implementation.\n     */\n    enumerateDevices() {\n        return new Promise(resolve => WebRTCModule.enumerateDevices(resolve));\n    }\n\n    /**\n     * W3C \"Screen Capture\" compatible {@code getDisplayMedia} implementation.\n     * See: https://w3c.github.io/mediacapture-screen-share/\n     *\n     * @returns {Promise}\n     */\n    getDisplayMedia() {\n        return getDisplayMedia();\n    }\n\n    /**\n     * W3C \"Media Capture and Streams\" compatible {@code getUserMedia}\n     * implementation.\n     * See: https://www.w3.org/TR/mediacapture-streams/#dom-mediadevices-enumeratedevices\n     *\n     * @param {*} constraints\n     * @returns {Promise}\n     */\n    getUserMedia(constraints: Constraints) {\n        return getUserMedia(constraints);\n    }\n\n    createVirtualAudioTrack(): Promise<VirtualAudioResponse> {\n        if (!WebRTCModule.createVirtualAudioTrack) {\n            return Promise.reject(new Error('virtual audio unavailable'));\n        }\n\n        return WebRTCModule.createVirtualAudioTrack();\n    }\n\n    pushVirtualAudioSamples(samples: Int16Array | number[], sampleRate: number, channels: number) {\n        if (!WebRTCModule.pushVirtualAudioSamples) {\n            return;\n        }\n\n        const array = samples instanceof Int16Array ? Array.from(samples) : samples.slice();\n\n        WebRTCModule.pushVirtualAudioSamples(array, sampleRate, channels);\n    }\n}\n\n/**\n * Define the `onxxx` event handlers.\n */\nconst proto = MediaDevices.prototype;\n\ndefineEventAttribute(proto, 'devicechange');\n\n\nexport default new MediaDevices();\n"],"mappings":"AAAA,SAASA,WAAW,EAASC,oBAAoB,QAAQ,yBAAyB;AAClF,SAASC,aAAa,QAAQ,cAAc;AAG5C,OAAOC,eAAe,MAAM,mBAAmB;AAC/C,OAAOC,YAAY,MAAuB,gBAAgB;AAE1D,MAAM;EAAEC;AAAa,CAAC,GAAGH,aAAa;AAoBtC,MAAMI,YAAY,SAASN,WAAW,CAAuB;EACzD;AACJ;AACA;AACA;EACIO,gBAAgBA,CAAA,EAAG;IACf,OAAO,IAAIC,OAAO,CAACC,OAAO,IAAIJ,YAAY,CAACE,gBAAgB,CAACE,OAAO,CAAC,CAAC;EACzE;;EAEA;AACJ;AACA;AACA;AACA;AACA;EACIN,eAAeA,CAAA,EAAG;IACd,OAAOA,eAAe,CAAC,CAAC;EAC5B;;EAEA;AACJ;AACA;AACA;AACA;AACA;AACA;AACA;EACIC,YAAYA,CAACM,WAAwB,EAAE;IACnC,OAAON,YAAY,CAACM,WAAW,CAAC;EACpC;EAEAC,uBAAuBA,CAAA,EAAkC;IACrD,IAAI,CAACN,YAAY,CAACM,uBAAuB,EAAE;MACvC,OAAOH,OAAO,CAACI,MAAM,CAAC,IAAIC,KAAK,CAAC,2BAA2B,CAAC,CAAC;IACjE;IAEA,OAAOR,YAAY,CAACM,uBAAuB,CAAC,CAAC;EACjD;EAEAG,uBAAuBA,CAACC,OAA8B,EAAEC,UAAkB,EAAEC,QAAgB,EAAE;IAC1F,IAAI,CAACZ,YAAY,CAACS,uBAAuB,EAAE;MACvC;IACJ;IAEA,MAAMI,KAAK,GAAGH,OAAO,YAAYI,UAAU,GAAGC,KAAK,CAACC,IAAI,CAACN,OAAO,CAAC,GAAGA,OAAO,CAACO,KAAK,CAAC,CAAC;IAEnFjB,YAAY,CAACS,uBAAuB,CAACI,KAAK,EAAEF,UAAU,EAAEC,QAAQ,CAAC;EACrE;AACJ;;AAEA;AACA;AACA;AACA,MAAMM,KAAK,GAAGjB,YAAY,CAACkB,SAAS;AAEpCvB,oBAAoB,CAACsB,KAAK,EAAE,cAAc,CAAC;AAG3C,eAAe,IAAIjB,YAAY,CAAC,CAAC"}

package/lib/module/MediaStreamTrack.js.map

@@ -1 +1 @@
-{"version":3,"names":["EventTarget","Event","defineEventAttribute","NativeModules","addListener","removeListener","removeListenerForEvent","Logger","deepClone","normalizeConstraints","log","WebRTCModule","MediaStreamTrack","constructor","info","_defineProperty","_constraints","constraints","_enabled","enabled","_settings","settings","_muted","_peerConnectionId","peerConnectionId","_readyState","readyState","id","kind","remote","_registerEvents","Boolean","mediaStreamTrackSetEnabled","muted","stop","_switchCamera","Error","deviceId","facingMode","applyConstraints","_setVideoEffects","names","mediaStreamTrackSetVideoEffects","_setVideoEffect","name","_setMutedInternal","dispatchEvent","_setVolume","volume","mediaStreamTrackSetVolume","normalized","video","mediaStreamTrackApplyConstraints","clone","getCapabilities","getConstraints","getSettings","ev","trackId","debug","_setAudioSinkCallback","callback","_audioSinkCallback","_handleAudioSamples","data","setAudioSink","_audioSinkListenerActive","mediaStreamTrackEnableAudioSink","setPlaybackEnabled","_playbackEnabled","mediaStreamTrackSetPlaybackEnabled","isPlaybackEnabled","release","mediaStreamTrackRelease","proto","prototype"],"sources":["MediaStreamTrack.ts"],"sourcesContent":["import { EventTarget, Event, defineEventAttribute } from 'event-target-shim/index';\nimport { NativeModules, NativeEventEmitter, Platform } from 'react-native';\n\nimport { MediaTrackConstraints } from './Constraints';\nimport { addListener, removeListener, removeListenerForEvent } from './EventEmitter';\nimport Logger from './Logger';\nimport { deepClone, normalizeConstraints } from './RTCUtil';\n\nconst log = new Logger('pc');\nconst { WebRTCModule } = NativeModules;\n\nexport type AudioSampleData = {\n    trackId: string;\n    samples: number[];\n    sampleRate: number;\n    channels: number;\n    timestamp: number;\n}\n\nexport type AudioSinkCallback = (data: AudioSampleData) => void;\n\n\ntype MediaStreamTrackState = 'live' | 'ended';\n\nexport type MediaStreamTrackInfo = {\n    id: string;\n    kind: string;\n    remote: boolean;\n    constraints: object;\n    enabled: boolean;\n    settings: object;\n    peerConnectionId: number;\n    readyState: MediaStreamTrackState;\n}\n\nexport type MediaTrackSettings = {\n    width?: number;\n    height?: number;\n    frameRate?: number;\n    facingMode?: string;\n    deviceId?: string;\n    groupId?: string;\n}\n\ntype MediaStreamTrackEventMap = {\n    ended: Event<'ended'>;\n    mute: Event<'mute'>;\n    unmute: Event<'unmute'>;\n}\n\nexport default class MediaStreamTrack extends EventTarget<MediaStreamTrackEventMap> {\n    _constraints: MediaTrackConstraints;\n    _enabled: boolean;\n    _settings: MediaTrackSettings;\n    _muted: boolean;\n    _peerConnectionId: number;\n    _readyState: MediaStreamTrackState;\n\n    readonly id: string;\n    readonly kind: string;\n    readonly label: string = '';\n    readonly remote: boolean;\n\n    constructor(info: MediaStreamTrackInfo) {\n        super();\n\n        this._constraints = info.constraints || {};\n        this._enabled = info.enabled;\n        this._settings = info.settings || {};\n        this._muted = false;\n        this._peerConnectionId = info.peerConnectionId;\n        this._readyState = info.readyState;\n\n        this.id = info.id;\n        this.kind = info.kind;\n        this.remote = info.remote;\n\n        if (!this.remote) {\n            this._registerEvents();\n        }\n    }\n\n    get enabled(): boolean {\n        return this._enabled;\n    }\n\n    set enabled(enabled: boolean) {\n        if (enabled === this._enabled) {\n            return;\n        }\n\n        this._enabled = Boolean(enabled);\n\n        if (this._readyState === 'ended') {\n            return;\n        }\n\n        WebRTCModule.mediaStreamTrackSetEnabled(this.remote ? this._peerConnectionId : -1, this.id, this._enabled);\n    }\n\n    get muted(): boolean {\n        return this._muted;\n    }\n\n    get readyState(): string {\n        return this._readyState;\n    }\n\n    stop(): void {\n        this.enabled = false;\n        this._readyState = 'ended';\n    }\n\n    /**\n     * Private / custom API for switching the cameras on the fly, without the\n     * need for adding / removing tracks or doing any SDP renegotiation.\n     *\n     * This is how the reference application (AppRTCMobile) implements camera\n     * switching.\n     *\n     * @deprecated Use applyConstraints instead.\n     */\n    _switchCamera(): void {\n        if (this.remote) {\n            throw new Error('Not implemented for remote tracks');\n        }\n\n        if (this.kind !== 'video') {\n            throw new Error('Only implemented for video tracks');\n        }\n\n        const constraints = deepClone(this._settings);\n\n        delete constraints.deviceId;\n        constraints.facingMode = this._settings.facingMode === 'user' ? 'environment' : 'user';\n\n        this.applyConstraints(constraints);\n    }\n\n    _setVideoEffects(names: string[]) {\n        if (this.remote) {\n            throw new Error('Not implemented for remote tracks');\n        }\n\n        if (this.kind !== 'video') {\n            throw new Error('Only implemented for video tracks');\n        }\n\n        WebRTCModule.mediaStreamTrackSetVideoEffects(this.id, names);\n    }\n\n    _setVideoEffect(name: string) {\n        this._setVideoEffects([ name ]);\n    }\n\n    /**\n     * Internal function which is used to set the muted state on remote tracks and\n     * emit the mute / unmute event.\n     *\n     * @param muted Whether the track should be marked as muted / unmuted.\n     */\n    _setMutedInternal(muted: boolean) {\n        if (!this.remote) {\n            throw new Error('Track is not remote!');\n        }\n\n        this._muted = muted;\n        this.dispatchEvent(new Event(muted ? 'mute' : 'unmute'));\n    }\n\n    /**\n     * Custom API for setting the volume on an individual audio track.\n     *\n     * @param volume a gain value in the range of 0-10. defaults to 1.0\n     */\n    _setVolume(volume: number) {\n        if (this.kind !== 'audio') {\n            throw new Error('Only implemented for audio tracks');\n        }\n\n        WebRTCModule.mediaStreamTrackSetVolume(this.remote ? this._peerConnectionId : -1, this.id, volume);\n    }\n\n    /**\n     * Applies a new set of constraints to the track.\n     *\n     * @param constraints An object listing the constraints\n     * to apply to the track's constrainable properties; any existing\n     * constraints are replaced with the new values specified, and any\n     * constrainable properties not included are restored to their default\n     * constraints. If this parameter is omitted, all currently set custom\n     * constraints are cleared.\n     */\n    async applyConstraints(constraints?: MediaTrackConstraints): Promise<void> {\n        if (this.kind !== 'video') {\n            log.info(`Only implemented for video tracks, ignoring applyConstraints for ${this.id}`);\n\n            return;\n        }\n\n        const normalized = normalizeConstraints({ video: constraints ?? true });\n\n        this._settings = await WebRTCModule.mediaStreamTrackApplyConstraints(this.id, normalized.video);\n        this._constraints = constraints ?? {};\n    }\n\n    clone(): never {\n        throw new Error('Not implemented.');\n    }\n\n    getCapabilities(): never {\n        throw new Error('Not implemented.');\n    }\n\n    getConstraints() {\n        return deepClone(this._constraints);\n    }\n\n    getSettings(): MediaTrackSettings {\n        return deepClone(this._settings);\n    }\n\n    _registerEvents(): void {\n        addListener(this, 'mediaStreamTrackEnded', (ev: any) => {\n            if (ev.trackId !== this.id || this._readyState === 'ended') {\n                return;\n            }\n\n            log.debug(`${this.id} mediaStreamTrackEnded`);\n            this._readyState = 'ended';\n\n            this.dispatchEvent(new Event('ended'));\n        });\n    }\n\n    private _audioSinkCallback: AudioSinkCallback | null = null;\n    private _audioSinkListenerActive: boolean = false;\n    private _playbackEnabled: boolean = true;\n\n    _setAudioSinkCallback(callback: AudioSinkCallback | null): void {\n        this._audioSinkCallback = callback;\n    }\n\n    _handleAudioSamples(data: AudioSampleData): void {\n        if (this._audioSinkCallback) {\n            this._audioSinkCallback(data);\n        }\n    }\n\n    setAudioSink(callback: AudioSinkCallback | null): void {\n        if (this.kind !== 'audio') {\n            throw new Error('Audio sink only available for audio tracks');\n        }\n\n        if (!this.remote) {\n            throw new Error('Audio sink only available for remote tracks');\n        }\n\n        this._setAudioSinkCallback(callback);\n\n        if (callback && !this._audioSinkListenerActive) {\n            this._audioSinkListenerActive = true;\n            WebRTCModule.mediaStreamTrackEnableAudioSink(\n                this._peerConnectionId,\n                this.id,\n                true\n            );\n            addListener(this, 'audioSamples', (ev: any) => {\n                if (ev.trackId === this.id) {\n                    this._handleAudioSamples(ev);\n                }\n            });\n        } else if (!callback && this._audioSinkListenerActive) {\n            this._audioSinkListenerActive = false;\n            WebRTCModule.mediaStreamTrackEnableAudioSink(\n                this._peerConnectionId,\n                this.id,\n                false\n            );\n            removeListenerForEvent(this, 'audioSamples');\n        }\n    }\n\n    setPlaybackEnabled(enabled: boolean): void {\n        if (this.kind !== 'audio') {\n            throw new Error('Playback control only available for audio tracks');\n        }\n\n        if (!this.remote) {\n            throw new Error('Playback control only available for remote tracks');\n        }\n\n        this._playbackEnabled = enabled;\n        WebRTCModule.mediaStreamTrackSetPlaybackEnabled(\n            this._peerConnectionId,\n            this.id,\n            enabled\n        );\n    }\n\n    isPlaybackEnabled(): boolean {\n        return this._playbackEnabled;\n    }\n\n    release(): void {\n        if (this.remote) {\n            if (this._audioSinkListenerActive) {\n                this._audioSinkListenerActive = false;\n                WebRTCModule.mediaStreamTrackEnableAudioSink(\n                    this._peerConnectionId,\n                    this.id,\n                    false\n                );\n                removeListenerForEvent(this, 'audioSamples');\n       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+{"version":3,"names":["EventTarget","Event","defineEventAttribute","NativeModules","addListener","removeListener","removeListenerForEvent","Logger","deepClone","normalizeConstraints","log","WebRTCModule","MediaStreamTrack","constructor","info","_defineProperty","_constraints","constraints","_enabled","enabled","_settings","settings","_muted","_peerConnectionId","peerConnectionId","_readyState","readyState","id","kind","remote","_registerEvents","Boolean","mediaStreamTrackSetEnabled","muted","stop","_switchCamera","Error","deviceId","facingMode","applyConstraints","_setVideoEffects","names","mediaStreamTrackSetVideoEffects","_setVideoEffect","name","_setMutedInternal","dispatchEvent","_setVolume","volume","mediaStreamTrackSetVolume","normalized","video","mediaStreamTrackApplyConstraints","clone","getCapabilities","getConstraints","getSettings","ev","trackId","debug","_setAudioSinkCallback","callback","_audioSinkCallback","_handleAudioSamples","data","setAudioSink","_audioSinkListenerActive","mediaStreamTrackEnableAudioSink","setPlaybackEnabled","_playbackEnabled","mediaStreamTrackSetPlaybackEnabled","isPlaybackEnabled","release","mediaStreamTrackRelease","proto","prototype"],"sources":["MediaStreamTrack.ts"],"sourcesContent":["import { EventTarget, Event, defineEventAttribute } from 'event-target-shim/index';\nimport { NativeModules } from 'react-native';\n\nimport { MediaTrackConstraints } from './Constraints';\nimport { addListener, removeListener, removeListenerForEvent } from './EventEmitter';\nimport Logger from './Logger';\nimport { deepClone, normalizeConstraints } from './RTCUtil';\n\nconst log = new Logger('pc');\nconst { WebRTCModule } = NativeModules;\n\nexport type AudioSampleData = {\n    trackId: string;\n    samples: number[];\n    sampleRate: number;\n    channels: number;\n    bitsPerSample: number;\n    framesPerBuffer: number;\n    timestamp: number;\n}\n\nexport type AudioSinkCallback = (data: AudioSampleData) => void;\n\n\ntype MediaStreamTrackState = 'live' | 'ended';\n\nexport type MediaStreamTrackInfo = {\n    id: string;\n    kind: string;\n    remote: boolean;\n    constraints: object;\n    enabled: boolean;\n    settings: object;\n    peerConnectionId: number;\n    readyState: MediaStreamTrackState;\n}\n\nexport type MediaTrackSettings = {\n    width?: number;\n    height?: number;\n    frameRate?: number;\n    facingMode?: string;\n    deviceId?: string;\n    groupId?: string;\n}\n\ntype MediaStreamTrackEventMap = {\n    ended: Event<'ended'>;\n    mute: Event<'mute'>;\n    unmute: Event<'unmute'>;\n}\n\nexport default class MediaStreamTrack extends EventTarget<MediaStreamTrackEventMap> {\n    _constraints: MediaTrackConstraints;\n    _enabled: boolean;\n    _settings: MediaTrackSettings;\n    _muted: boolean;\n    _peerConnectionId: number;\n    _readyState: MediaStreamTrackState;\n\n    readonly id: string;\n    readonly kind: string;\n    readonly label: string = '';\n    readonly remote: boolean;\n\n    constructor(info: MediaStreamTrackInfo) {\n        super();\n\n        this._constraints = info.constraints || {};\n        this._enabled = info.enabled;\n        this._settings = info.settings || {};\n        this._muted = false;\n        this._peerConnectionId = info.peerConnectionId;\n        this._readyState = info.readyState;\n\n        this.id = info.id;\n        this.kind = info.kind;\n        this.remote = info.remote;\n\n        if (!this.remote) {\n            this._registerEvents();\n        }\n    }\n\n    get enabled(): boolean {\n        return this._enabled;\n    }\n\n    set enabled(enabled: boolean) {\n        if (enabled === this._enabled) {\n            return;\n        }\n\n        this._enabled = Boolean(enabled);\n\n        if (this._readyState === 'ended') {\n            return;\n        }\n\n        WebRTCModule.mediaStreamTrackSetEnabled(this.remote ? this._peerConnectionId : -1, this.id, this._enabled);\n    }\n\n    get muted(): boolean {\n        return this._muted;\n    }\n\n    get readyState(): string {\n        return this._readyState;\n    }\n\n    stop(): void {\n        this.enabled = false;\n        this._readyState = 'ended';\n    }\n\n    /**\n     * Private / custom API for switching the cameras on the fly, without the\n     * need for adding / removing tracks or doing any SDP renegotiation.\n     *\n     * This is how the reference application (AppRTCMobile) implements camera\n     * switching.\n     *\n     * @deprecated Use applyConstraints instead.\n     */\n    _switchCamera(): void {\n        if (this.remote) {\n            throw new Error('Not implemented for remote tracks');\n        }\n\n        if (this.kind !== 'video') {\n            throw new Error('Only implemented for video tracks');\n        }\n\n        const constraints = deepClone(this._settings);\n\n        delete constraints.deviceId;\n        constraints.facingMode = this._settings.facingMode === 'user' ? 'environment' : 'user';\n\n        this.applyConstraints(constraints);\n    }\n\n    _setVideoEffects(names: string[]) {\n        if (this.remote) {\n            throw new Error('Not implemented for remote tracks');\n        }\n\n        if (this.kind !== 'video') {\n            throw new Error('Only implemented for video tracks');\n        }\n\n        WebRTCModule.mediaStreamTrackSetVideoEffects(this.id, names);\n    }\n\n    _setVideoEffect(name: string) {\n        this._setVideoEffects([ name ]);\n    }\n\n    /**\n     * Internal function which is used to set the muted state on remote tracks and\n     * emit the mute / unmute event.\n     *\n     * @param muted Whether the track should be marked as muted / unmuted.\n     */\n    _setMutedInternal(muted: boolean) {\n        if (!this.remote) {\n            throw new Error('Track is not remote!');\n        }\n\n        this._muted = muted;\n        this.dispatchEvent(new Event(muted ? 'mute' : 'unmute'));\n    }\n\n    /**\n     * Custom API for setting the volume on an individual audio track.\n     *\n     * @param volume a gain value in the range of 0-10. defaults to 1.0\n     */\n    _setVolume(volume: number) {\n        if (this.kind !== 'audio') {\n            throw new Error('Only implemented for audio tracks');\n        }\n\n        WebRTCModule.mediaStreamTrackSetVolume(this.remote ? this._peerConnectionId : -1, this.id, volume);\n    }\n\n    /**\n     * Applies a new set of constraints to the track.\n     *\n     * @param constraints An object listing the constraints\n     * to apply to the track's constrainable properties; any existing\n     * constraints are replaced with the new values specified, and any\n     * constrainable properties not included are restored to their default\n     * constraints. If this parameter is omitted, all currently set custom\n     * constraints are cleared.\n     */\n    async applyConstraints(constraints?: MediaTrackConstraints): Promise<void> {\n        if (this.kind !== 'video') {\n            log.info(`Only implemented for video tracks, ignoring applyConstraints for ${this.id}`);\n\n            return;\n        }\n\n        const normalized = normalizeConstraints({ video: constraints ?? true });\n\n        this._settings = await WebRTCModule.mediaStreamTrackApplyConstraints(this.id, normalized.video);\n        this._constraints = constraints ?? {};\n    }\n\n    clone(): never {\n        throw new Error('Not implemented.');\n    }\n\n    getCapabilities(): never {\n        throw new Error('Not implemented.');\n    }\n\n    getConstraints() {\n        return deepClone(this._constraints);\n    }\n\n    getSettings(): MediaTrackSettings {\n        return deepClone(this._settings);\n    }\n\n    _registerEvents(): void {\n        addListener(this, 'mediaStreamTrackEnded', (ev: any) => {\n            if (ev.trackId !== this.id || this._readyState === 'ended') {\n                return;\n            }\n\n            log.debug(`${this.id} mediaStreamTrackEnded`);\n            this._readyState = 'ended';\n\n            this.dispatchEvent(new Event('ended'));\n        });\n    }\n\n    private _audioSinkCallback: AudioSinkCallback | null = null;\n    private _audioSinkListenerActive = false;\n    private _playbackEnabled = true;\n\n    _setAudioSinkCallback(callback: AudioSinkCallback | null): void {\n        this._audioSinkCallback = callback;\n    }\n\n    _handleAudioSamples(data: AudioSampleData): void {\n        if (this._audioSinkCallback) {\n            this._audioSinkCallback(data);\n        }\n    }\n\n    setAudioSink(callback: AudioSinkCallback | null): void {\n        if (this.kind !== 'audio') {\n            throw new Error('Audio sink only available for audio tracks');\n        }\n\n        if (!this.remote) {\n            throw new Error('Audio sink only available for remote tracks');\n        }\n\n        this._setAudioSinkCallback(callback);\n\n        if (callback && !this._audioSinkListenerActive) {\n            this._audioSinkListenerActive = true;\n            WebRTCModule.mediaStreamTrackEnableAudioSink(\n                this._peerConnectionId,\n                this.id,\n                true\n            );\n            addListener(this, 'audioSamples', (ev: any) => {\n                if (ev.trackId === this.id) {\n                    this._handleAudioSamples(ev);\n                }\n            });\n        } else if (!callback && this._audioSinkListenerActive) {\n            this._audioSinkListenerActive = false;\n            WebRTCModule.mediaStreamTrackEnableAudioSink(\n                this._peerConnectionId,\n                this.id,\n                false\n            );\n            removeListenerForEvent(this, 'audioSamples');\n        }\n    }\n\n    setPlaybackEnabled(enabled: boolean): void {\n        if (this.kind !== 'audio') {\n            throw new Error('Playback control only available for audio tracks');\n        }\n\n        if (!this.remote) {\n            throw new Error('Playback control only available for remote tracks');\n        }\n\n        this._playbackEnabled = enabled;\n        WebRTCModule.mediaStreamTrackSetPlaybackEnabled(\n            this._peerConnectionId,\n            this.id,\n            enabled\n        );\n    }\n\n    isPlaybackEnabled(): boolean {\n        return this._playbackEnabled;\n    }\n\n    release(): void {\n        if (this.remote) {\n            if (this._audioSinkListenerActive) {\n                this._audioSinkListenerActive = false;\n                WebRTCModule.mediaStreamTrackEnableAudioSink(\n                    this._peerConnectionId,\n                    this.id,\n                    false\n                );\n                removeListenerForEvent(this, 'audioSamples');\n            }\n\n            return;\n        }\n\n        removeListener(this);\n        WebRTCModule.mediaStreamTrackRelease(this.id);\n    }\n}\n\n/**\n * Define the `onxxx` event handlers.\n */\nconst proto = MediaStreamTrack.prototype;\n\ndefineEventAttribute(proto, 'ended');\ndefineEventAttribute(proto, 'mute');\ndefineEventAttribute(proto, 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package/lib/typescript/MediaDevices.d.ts

@@ -1,8 +1,21 @@
 import { EventTarget, Event } from 'event-target-shim/index';
+import { type MediaTrackSettings } from './MediaStreamTrack';
 import { Constraints } from './getUserMedia';
 declare type MediaDevicesEventMap = {
     devicechange: Event<'devicechange'>;
 };
+declare type VirtualTrackInfo = {
+    enabled: boolean;
+    id: string;
+    kind: string;
+    readyState: string;
+    remote: boolean;
+    settings: MediaTrackSettings;
+};
+declare type VirtualAudioResponse = {
+    streamId: string;
+    tracks: VirtualTrackInfo[];
+};
 declare class MediaDevices extends EventTarget<MediaDevicesEventMap> {
     /**
      * W3C "Media Capture and Streams" compatible {@code enumerateDevices}
@@ -25,6 +38,8 @@ declare class MediaDevices extends EventTarget<MediaDevicesEventMap> {
      * @returns {Promise}
      */
     getUserMedia(constraints: Constraints): Promise<import("./MediaStream").default>;
+    createVirtualAudioTrack(): Promise<VirtualAudioResponse>;
+    pushVirtualAudioSamples(samples: Int16Array | number[], sampleRate: number, channels: number): void;
 }
 declare const _default: MediaDevices;
 export default _default;

package/lib/typescript/MediaStreamTrack.d.ts

@@ -5,6 +5,8 @@ export declare type AudioSampleData = {
     samples: number[];
     sampleRate: number;
     channels: number;
+    bitsPerSample: number;
+    framesPerBuffer: number;
     timestamp: number;
 };
 export declare type AudioSinkCallback = (data: AudioSampleData) => void;

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@sbhjt-gr/react-native-webrtc",
-  "version": "137.0.3",
+  "version": "137.0.5",
   "repository": {
     "type": "git",
     "url": "git+https://github.com/sbhjt-gr/react-native-webrtc.git"

package/src/EventEmitter.ts

@@ -64,9 +64,13 @@ export function removeListener(listener: Listener): void {
 
 export function removeListenerForEvent(listener: Listener, eventName: string): void {
     const subs = _subscriptions.get(listener);
-    if (!subs) return;
-    
+
+    if (!subs) {
+        return;
+    }
+
     const remaining: EmitterSubscription[] = [];
+
     subs.forEach(sub => {
         if ((sub as any).eventType === eventName) {
             sub.remove();
@@ -74,7 +78,7 @@ export function removeListenerForEvent(listener: Listener, eventName: string): v
             remaining.push(sub);
         }
     });
-    
+
     if (remaining.length === 0) {
         _subscriptions.delete(listener);
     } else {

package/src/MediaDevices.ts

@@ -1,6 +1,7 @@
 import { EventTarget, Event, defineEventAttribute } from 'event-target-shim/index';
 import { NativeModules } from 'react-native';
 
+import { type MediaTrackSettings } from './MediaStreamTrack';
 import getDisplayMedia from './getDisplayMedia';
 import getUserMedia, { Constraints } from './getUserMedia';
 
@@ -10,6 +11,20 @@ type MediaDevicesEventMap = {
     devicechange: Event<'devicechange'>
 }
 
+type VirtualTrackInfo = {
+    enabled: boolean;
+    id: string;
+    kind: string;
+    readyState: string;
+    remote: boolean;
+    settings: MediaTrackSettings;
+}
+
+type VirtualAudioResponse = {
+    streamId: string;
+    tracks: VirtualTrackInfo[];
+}
+
 class MediaDevices extends EventTarget<MediaDevicesEventMap> {
     /**
      * W3C "Media Capture and Streams" compatible {@code enumerateDevices}
@@ -40,6 +55,24 @@ class MediaDevices extends EventTarget<MediaDevicesEventMap> {
     getUserMedia(constraints: Constraints) {
         return getUserMedia(constraints);
     }
+
+    createVirtualAudioTrack(): Promise<VirtualAudioResponse> {
+        if (!WebRTCModule.createVirtualAudioTrack) {
+            return Promise.reject(new Error('virtual audio unavailable'));
+        }
+
+        return WebRTCModule.createVirtualAudioTrack();
+    }
+
+    pushVirtualAudioSamples(samples: Int16Array | number[], sampleRate: number, channels: number) {
+        if (!WebRTCModule.pushVirtualAudioSamples) {
+            return;
+        }
+
+        const array = samples instanceof Int16Array ? Array.from(samples) : samples.slice();
+
+        WebRTCModule.pushVirtualAudioSamples(array, sampleRate, channels);
+    }
 }
 
 /**

package/src/MediaStreamTrack.ts

@@ -1,5 +1,5 @@
 import { EventTarget, Event, defineEventAttribute } from 'event-target-shim/index';
-import { NativeModules, NativeEventEmitter, Platform } from 'react-native';
+import { NativeModules } from 'react-native';
 
 import { MediaTrackConstraints } from './Constraints';
 import { addListener, removeListener, removeListenerForEvent } from './EventEmitter';
@@ -14,6 +14,8 @@ export type AudioSampleData = {
     samples: number[];
     sampleRate: number;
     channels: number;
+    bitsPerSample: number;
+    framesPerBuffer: number;
     timestamp: number;
 }
 
@@ -234,8 +236,8 @@ export default class MediaStreamTrack extends EventTarget<MediaStreamTrackEventM
     }
 
     private _audioSinkCallback: AudioSinkCallback | null = null;
-    private _audioSinkListenerActive: boolean = false;
-    private _playbackEnabled: boolean = true;
+    private _audioSinkListenerActive = false;
+    private _playbackEnabled = true;
 
     _setAudioSinkCallback(callback: AudioSinkCallback | null): void {
         this._audioSinkCallback = callback;
@@ -313,6 +315,7 @@ export default class MediaStreamTrack extends EventTarget<MediaStreamTrackEventM
                 );
                 removeListenerForEvent(this, 'audioSamples');
             }
+
             return;
         }