ios: Add helper methods to access RTCAudioSession.audioSessionDidActivate and deactivate

Author: davidliu

Documentation/iOSInstallation.md

@@ -26,6 +26,33 @@ Navigate to `<ProjectFolder>/ios/<ProjectName>/` and edit `Info.plist`, add the
 <string>Microphone permission description</string>
 ```
 
+## CallKit
+
+If your app uses a CallKit integration to handle incoming calls, then your
+CXProviderDelegate should call through to `RTCAudioSession.sharedInstance.audioSessionDidActivate/Deactivate` accordingly.
+
+```
+#import <WebRTC/RTCAudioSession.h>
+
+- (void) provider:(CXProvider *) provider didActivateAudioSession:(AVAudioSession *) audioSession {
+    [[RTCAudioSession sharedInstance] audioSessionDidActivate:[AVAudioSession sharedInstance]];
+}
+
+- (void) provider:(CXProvider *) provider didDeactivateAudioSession:(AVAudioSession *) audioSession {
+    [[RTCAudioSession sharedInstance] audioSessionDidDeactivate:[AVAudioSession sharedInstance]];
+}
+```
+
+Javascript methods are also provided to call these methods:
+
+```
+import { RTCAudioSession } from '@livekit/react-native-webrtc'
+
+// Call as needed.
+RTCAudioSession.audioSessionDidActivate();
+RTCAudioSession.audioSessionDidDeactivate();
+```
+
 ## Library not loaded/Code signature invalid
 
 This is an issue with iOS 13.3.1.  

ios/RCTWebRTC.xcodeproj/project.pbxproj

@@ -24,6 +24,7 @@
 		4EE3A8D425B841DD00FAA24A /* ScreenCapturer.m in Sources */ = {isa = PBXBuildFile; fileRef = 4EE3A8CD25B841DD00FAA24A /* ScreenCapturer.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 */; };
 		DEC96577264176C10052DB35 /* DataChannelWrapper.m in Sources */ = {isa = PBXBuildFile; fileRef = DEC96576264176C10052DB35 /* DataChannelWrapper.m */; };
 /* End PBXBuildFile section */
 
@@ -75,6 +76,7 @@
 		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; };
 		D74EF94729652169000742E1 /* TrackCapturerEventsEmitter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = TrackCapturerEventsEmitter.h; path = RCTWebRTC/TrackCapturerEventsEmitter.h; sourceTree = SOURCE_ROOT; };
+		D7F0711D2C6DC91F0031F594 /* WebRTCModule+RTCAudioSession.m */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.objc; name = "WebRTCModule+RTCAudioSession.m"; path = "RCTWebRTC/WebRTCModule+RTCAudioSession.m"; sourceTree = SOURCE_ROOT; };
 		D7F99C122938F4E0000A2450 /* WebRTCModule+RTCMediaStream.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = "WebRTCModule+RTCMediaStream.h"; path = "RCTWebRTC/WebRTCModule+RTCMediaStream.h"; sourceTree = SOURCE_ROOT; };
 		DEC96576264176C10052DB35 /* DataChannelWrapper.m */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.objc; name = DataChannelWrapper.m; path = RCTWebRTC/DataChannelWrapper.m; sourceTree = "<group>"; };
 		DEC96579264176DF0052DB35 /* DataChannelWrapper.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = DataChannelWrapper.h; path = RCTWebRTC/DataChannelWrapper.h; sourceTree = "<group>"; };
@@ -131,6 +133,7 @@
 				4EE3A8AF25B8413200FAA24A /* VideoCaptureController.h */,
 				0BDDA6DF20C18B6B00B38B45 /* VideoCaptureController.m */,
 				4EE3A8B525B8414A00FAA24A /* WebRTCModule+Permissions.m */,
+				D7F0711D2C6DC91F0031F594 /* WebRTCModule+RTCAudioSession.m */,
 				4EE3A8B825B8415900FAA24A /* WebRTCModule+RTCDataChannel.h */,
 				4EE3A8B925B8415900FAA24A /* WebRTCModule+RTCDataChannel.m */,
 				D7F99C122938F4E0000A2450 /* WebRTCModule+RTCMediaStream.h */,
@@ -262,6 +265,7 @@
 				4EC498BC25B8777F00E76218 /* ScreenCapturePickerViewManager.m in Sources */,
 				4EE3A8B225B8414000FAA24A /* WebRTCModule.m in Sources */,
 				4EE3A8D225B841DD00FAA24A /* CaptureController.m in Sources */,
+				D7F0711E2C6DC91F0031F594 /* WebRTCModule+RTCAudioSession.m in Sources */,
 				D74EF94829652169000742E1 /* TrackCapturerEventsEmitter.m in Sources */,
 				4EE3A8D125B841DD00FAA24A /* SocketConnection.m in Sources */,
 				4EE3A8BD25B8416500FAA24A /* WebRTCModule+RTCMediaStream.m in Sources */,

ios/RCTWebRTC/WebRTCModule+RTCAudioSession.m

@@ -0,0 +1,20 @@
+#import <objc/runtime.h>
+
+#import <React/RCTBridge.h>
+#import <React/RCTBridgeModule.h>
+
+#import "WebRTCModule.h"
+
+@implementation WebRTCModule (RTCAudioSession)
+
+RCT_EXPORT_BLOCKING_SYNCHRONOUS_METHOD(audioSessionDidActivate) {
+    [[RTCAudioSession sharedInstance] audioSessionDidActivate:[AVAudioSession sharedInstance]];
+    return nil;
+}
+
+RCT_EXPORT_BLOCKING_SYNCHRONOUS_METHOD(audioSessionDidDeactivate) {
+    [[RTCAudioSession sharedInstance] audioSessionDidDeactivate:[AVAudioSession sharedInstance]];
+    return nil;
+}
+
+@end

src/RTCAudioSession.ts

@@ -0,0 +1,25 @@
+import { NativeModules, Platform } from 'react-native';
+
+const { WebRTCModule } = NativeModules;
+
+export default class RTCAudioSession {
+    /**
+     * To be called when CallKit activates the audio session.
+     */
+    static audioSessionDidActivate() {
+        // Only valid for iOS
+        if (Platform.OS === "ios") {
+            WebRTCModule.audioSessionDidActivate()
+        }
+    }
+
+    /**
+     * To be called when CallKit deactivates the audio session.
+     */
+    static audioSessionDidDeactivate() {
+        // Only valid for iOS
+        if (Platform.OS === "ios") {
+            WebRTCModule.audioSessionDidDeactivate()
+        }
+    }
+}

src/index.ts

@@ -15,6 +15,7 @@ import MediaStream from './MediaStream';
 import MediaStreamTrack from './MediaStreamTrack';
 import MediaStreamTrackEvent from './MediaStreamTrackEvent';
 import permissions from './Permissions';
+import RTCAudioSession from './RTCAudioSession';
 import RTCErrorEvent from './RTCErrorEvent';
 import RTCIceCandidate from './RTCIceCandidate';
 import RTCPeerConnection from './RTCPeerConnection';
@@ -40,6 +41,7 @@ export {
     RTCRtpReceiver,
     RTCRtpSender,
     RTCErrorEvent,
+    RTCAudioSession,
     MediaStream,
     MediaStreamTrack,
     mediaDevices,