feat: v137 upgrade (#20)

Author: santhoshvai

android/build.gradle

@@ -70,6 +70,7 @@ android {
         targetSdkVersion safeExtGet('targetSdkVersion', 24)
         versionCode 1
         versionName "1.0"
+        consumerProguardFiles 'consumer-rules.pro'
     }
 
     // WebRTC requires Java 8 features
@@ -85,8 +86,8 @@ def kotlin_version = getExtOrDefault('kotlinVersion', '1.8.10')
 println "Building Stream WebRTC React Native module with Kotlin version: $kotlin_version"
 
 dependencies {
-    api 'io.getstream:stream-webrtc-android:1.3.10'
-    implementation 'com.facebook.react:react-native:+'
+    api 'io.getstream:stream-video-webrtc-android:137.0.1'
+    implementation "com.facebook.react:react-android:+"
     implementation "org.jetbrains.kotlin:kotlin-stdlib:$kotlin_version"
     implementation "androidx.core:core:1.7.0"
 }

android/consumer-rules.pro

@@ -0,0 +1,3 @@
+# WebRTC
+-keep class org.webrtc.** { *; }
+-keep class org.jni_zero.** { *; }

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

@@ -111,6 +111,24 @@ private AudioTrack createAudioTrack(ReadableMap constraints) {
         PeerConnectionFactory pcFactory = webRTCModule.mFactory;
         MediaConstraints peerConstraints = webRTCModule.constraintsForOptions(audioConstraintsMap);
 
+        // Convert given constraints into the internal webrtc media constraints.
+        peerConstraints.optional.add(new MediaConstraints.KeyValuePair("googAutoGainControl",
+                audioConstraintsMap.hasKey("autoGainControl")
+                        ? ReactBridgeUtil.getMapStrValue(audioConstraintsMap, "autoGainControl")
+                        : "true"));
+        peerConstraints.optional.add(new MediaConstraints.KeyValuePair("googNoiseSuppression",
+                audioConstraintsMap.hasKey("noiseSuppression")
+                        ? ReactBridgeUtil.getMapStrValue(audioConstraintsMap, "noiseSuppression")
+                        : "true"));
+        peerConstraints.optional.add(new MediaConstraints.KeyValuePair("googEchoCancellation",
+                audioConstraintsMap.hasKey("echoCancellation")
+                        ? ReactBridgeUtil.getMapStrValue(audioConstraintsMap, "echoCancellation")
+                        : "true"));
+        peerConstraints.optional.add(new MediaConstraints.KeyValuePair("googHighpassFilter",
+                audioConstraintsMap.hasKey("highpassFilter")
+                        ? ReactBridgeUtil.getMapStrValue(audioConstraintsMap, "highpassFilter")
+                        : "true"));
+
         // PeerConnectionFactory.createAudioSource will throw an error when mandatory constraints contain nulls.
         // so, let's check for nulls
         checkMandatoryConstraints(peerConstraints);

ios/RCTWebRTC/WebRTCModule+RTCMediaStream.m

@@ -32,6 +32,10 @@ - (void)setVideoEffectProcessor:(VideoEffectProcessor *)videoEffectProcessor
 
 #pragma mark - getUserMedia
 
+- (NSString *)convertBoolToString:(id)value {
+    return value ? @"true" : @"false";
+}
+
 /**
  * Initializes a new {@link RTCAudioTrack} which satisfies the given constraints.
  *
@@ -40,7 +44,26 @@ - (void)setVideoEffectProcessor:(VideoEffectProcessor *)videoEffectProcessor
  */
 - (RTCAudioTrack *)createAudioTrack:(NSDictionary *)constraints {
     NSString *trackId = [[NSUUID UUID] UUIDString];
-    RTCAudioTrack *audioTrack = [self.peerConnectionFactory audioTrackWithTrackId:trackId];
+    NSDictionary *audioConstraints = constraints[@"audio"];
+    NSMutableDictionary *optionalConstraints = [NSMutableDictionary dictionary];
+    optionalConstraints[@"googAutoGainControl"] = audioConstraints[@"autoGainControl"] != nil
+                                                      ? [self convertBoolToString:audioConstraints[@"autoGainControl"]]
+                                                      : @"true";
+    optionalConstraints[@"googNoiseSuppression"] =
+        audioConstraints[@"noiseSuppression"] != nil ? [self convertBoolToString:audioConstraints[@"noiseSuppression"]]
+                                                     : @"true";
+    optionalConstraints[@"googEchoCancellation"] =
+        audioConstraints[@"echoCancellation"] != nil ? [self convertBoolToString:audioConstraints[@"echoCancellation"]]
+                                                     : @"true";
+    optionalConstraints[@"googHighpassFilter"] = audioConstraints[@"highpassFilter"] != nil
+                                                     ? [self convertBoolToString:audioConstraints[@"highpassFilter"]]
+                                                     : @"true";
+
+    RTCMediaConstraints *mediaConstraints =
+        [[RTCMediaConstraints alloc] initWithMandatoryConstraints:nil optionalConstraints:optionalConstraints];
+
+    RTCAudioSource *audioSource = [self.peerConnectionFactory audioSourceWithConstraints:mediaConstraints];
+    RTCAudioTrack *audioTrack = [self.peerConnectionFactory audioTrackWithSource:audioSource trackId:trackId];
     return audioTrack;
 }
 /**

ios/RCTWebRTC/WebRTCModule.m

@@ -77,15 +77,24 @@ - (instancetype)init {
                 NSLog(@"Both audioProcessingModule and audioDevice are provided, but only one can be used. Ignoring audioDevice.");
             }
             RCTLogInfo(@"Using audio processing module: %@", NSStringFromClass([audioProcessingModule class]));
-            _peerConnectionFactory = [[RTCPeerConnectionFactory alloc] initWithBypassVoiceProcessing:NO
-                                                                                      encoderFactory:encoderFactory
-                                                                                      decoderFactory:decoderFactory
-                                                                               audioProcessingModule:audioProcessingModule];
-        } else {
-            RCTLogInfo(@"Using audio device: %@", NSStringFromClass([audioDevice class]));
+            _peerConnectionFactory =
+                [[RTCPeerConnectionFactory alloc] initWithAudioDeviceModuleType:RTCAudioDeviceModuleTypePlatformDefault
+                                                          bypassVoiceProcessing:NO
+                                                                 encoderFactory:encoderFactory
+                                                                 decoderFactory:decoderFactory
+                                                          audioProcessingModule:audioProcessingModule];
+        } else if (audioDevice != nil) {
+            RCTLogInfo(@"Using custom audio device: %@", NSStringFromClass([audioDevice class]));
             _peerConnectionFactory = [[RTCPeerConnectionFactory alloc] initWithEncoderFactory:encoderFactory
                                                                                decoderFactory:decoderFactory
                                                                                   audioDevice:audioDevice];
+        } else {
+            _peerConnectionFactory =
+                [[RTCPeerConnectionFactory alloc] initWithAudioDeviceModuleType:RTCAudioDeviceModuleTypePlatformDefault
+                                                          bypassVoiceProcessing:NO
+                                                                 encoderFactory:encoderFactory
+                                                                 decoderFactory:decoderFactory
+                                                          audioProcessingModule:nil];
         }
 
         _peerConnections = [NSMutableDictionary new];

package-lock.json

@@ -1,6 +1,6 @@
 {
   "name": "@stream-io/react-native-webrtc",
-  "version": "125.4.5",
+  "version": "137.0.0-alpha.6",
   "repository": {
     "type": "git",
     "url": "git+https://github.com/GetStream/react-native-webrtc.git"

package.json

@@ -2,6 +2,9 @@ require 'json'
 
 package = JSON.parse(File.read(File.join(__dir__, 'package.json')))
 
+# WebRTC version from stream-video-swift-webrtc releases
+webrtc_version = '137.0.42'
+
 Pod::Spec.new do |s|
   s.name                = 'stream-react-native-webrtc'
   s.version             = package['version']
@@ -19,7 +22,12 @@ Pod::Spec.new do |s|
   s.libraries           = 'c', 'sqlite3', 'stdc++'
   s.framework           = 'AudioToolbox','AVFoundation', 'CoreAudio', 'CoreGraphics', 'CoreVideo', 'GLKit', 'VideoToolbox'
   s.dependency          'React-Core'
-  s.dependency          'StreamWebRTC', '~>125.6422.070'
+  
+  s.prepare_command = <<-CMD
+    curl -sL "https://github.com/GetStream/stream-video-swift-webrtc/releases/download/#{webrtc_version}/WebRTC.xcframework.zip" -o w.zip && unzip -oq w.zip && rm w.zip
+  CMD
+  
+  s.vendored_frameworks = 'WebRTC.xcframework'
   # Swift/Objective-C compatibility #https://blog.cocoapods.org/CocoaPods-1.5.0/
   s.pod_target_xcconfig = {
     'DEFINES_MODULE' => 'YES'