cordova-rtc · hthetiot · May 12, 2022 · May 12, 2022 · Jun 17, 2022 · Jun 17, 2022
diff --git a/CHANGELOG.md b/CHANGELOG.md
@@ -1,3 +1,7 @@
+#### Version 10.0.0
+* fix: update to WebRTC.framework M100
+* feat: remove i386 build from WebRTC.xcframework
+
 #### Version 8.0.3
 * fix: PluginRTCPeerConnection: cyclic reference leak fixed #754 via @RSATom
 * fix: update npm packages with audit security warnings

diff --git a/README.md b/README.md
@@ -37,7 +37,7 @@ In order to make this Cordova plugin run into a iOS application some requirement
 ### Third-Party Supported Library
 
 * WebRTC W3C v1.0.0
-* WebRTC.framework => M87
+* WebRTC.framework => M100
 * Janus => 0.7.4
 * JSSip => 3.1.2
 * Sip.js => 0.15.6
@@ -60,7 +60,7 @@ $ cordova plugin add cordova-plugin-iosrtc
 
 ## Building
 
-* Last [Tested WebRTC.framework](./lib/WebRTC.framework/) version: M87 on cordova-plugin-iosrtc version 8.0.0+
+* Last [Tested WebRTC.framework](./lib/WebRTC.framework/) version: M100 on cordova-plugin-iosrtc version 10.0.0+
 * [Building](docs/Building.md): Guidelines for building a Cordova iOS application including the *cordova-plugin-iosrtc* plugin.
 * [Building `libwebrtc`](docs/BuildingLibWebRTC.md): Guidelines for building Google's *libwebrtc* with modifications needed by the *cordova-plugin-iosrtc* plugin (just in case you want to use a different version of *libwebrtc* or apply your own changes to it).
 

diff --git a/lib/WebRTC.xcframework/Info.plist b/lib/WebRTC.xcframework/Info.plist
@@ -6,25 +6,24 @@
 	<array>
 		<dict>
 			<key>LibraryIdentifier</key>
-			<string>ios-arm64_armv7</string>
+			<string>ios-arm64</string>
 			<key>LibraryPath</key>
 			<string>WebRTC.framework</string>
 			<key>SupportedArchitectures</key>
 			<array>
 				<string>arm64</string>
-				<string>armv7</string>
 			</array>
 			<key>SupportedPlatform</key>
 			<string>ios</string>
 		</dict>
 		<dict>
 			<key>LibraryIdentifier</key>
-			<string>ios-i386_x86_64-simulator</string>
+			<string>ios-arm64_x86_64-simulator</string>
 			<key>LibraryPath</key>
 			<string>WebRTC.framework</string>
 			<key>SupportedArchitectures</key>
 			<array>
-				<string>i386</string>
+				<string>arm64</string>
 				<string>x86_64</string>
 			</array>
 			<key>SupportedPlatform</key>

diff --git a/...ebRTC.framework/Headers/RTCAudioSession.h → ...ebRTC.framework/Headers/RTCAudioSession.h b/...ebRTC.framework/Headers/RTCAudioSession.h → ...ebRTC.framework/Headers/RTCAudioSession.h
@@ -11,7 +11,7 @@
 #import <AVFoundation/AVFoundation.h>
 #import <Foundation/Foundation.h>
 
-#import "RTCMacros.h"
+#import <WebRTC/RTCMacros.h>
 
 NS_ASSUME_NONNULL_BEGIN
 
@@ -99,6 +99,9 @@ RTC_OBJC_EXPORT
     failedToSetActive:(BOOL)active
                 error:(NSError *)error;
 
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    audioUnitStartFailedWithError:(NSError *)error;
+
 @end
 
 /** This is a protocol used to inform RTCAudioSession when the audio session
@@ -122,7 +125,7 @@ RTC_OBJC_EXPORT
  *  WebRTC and the application layer are avoided.
  *
  *  RTCAudioSession also coordinates activation so that the audio session is
- *  activated only once. See |setActive:error:|.
+ *  activated only once. See `setActive:error:`.
  */
 RTC_OBJC_EXPORT
 @interface RTC_OBJC_TYPE (RTCAudioSession) : NSObject <RTC_OBJC_TYPE(RTCAudioSessionActivationDelegate)>
@@ -137,8 +140,6 @@ RTC_OBJC_EXPORT
  *  AVAudioSession.
  */
 @property(nonatomic, readonly) BOOL isActive;
-/** Whether RTCAudioSession is currently locked for configuration. */
-@property(nonatomic, readonly) BOOL isLocked;
 
 /** If YES, WebRTC will not initialize the audio unit automatically when an
  *  audio track is ready for playout or recording. Instead, applications should
@@ -211,17 +212,17 @@ RTC_OBJC_EXPORT
 /** Relinquishes exclusive access to the audio session. */
 - (void)unlockForConfiguration;
 
-/** If |active|, activates the audio session if it isn't already active.
+/** If `active`, activates the audio session if it isn't already active.
  *  Successful calls must be balanced with a setActive:NO when activation is no
- *  longer required. If not |active|, deactivates the audio session if one is
+ *  longer required. If not `active`, deactivates the audio session if one is
  *  active and this is the last balanced call. When deactivating, the
  *  AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
  *  AVAudioSession.
  */
 - (BOOL)setActive:(BOOL)active error:(NSError **)outError;
 
 // The following methods are proxies for the associated methods on
-// AVAudioSession. |lockForConfiguration| must be called before using them
+// AVAudioSession. `lockForConfiguration` must be called before using them
 // otherwise they will fail with kRTCAudioSessionErrorLockRequired.
 
 - (BOOL)setCategory:(NSString *)category
@@ -247,13 +248,13 @@ RTC_OBJC_EXPORT
     /** Applies the configuration to the current session. Attempts to set all
      *  properties even if previous ones fail. Only the last error will be
      *  returned.
-     *  |lockForConfiguration| must be called first.
+     *  `lockForConfiguration` must be called first.
      */
     - (BOOL)setConfiguration : (RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration error
     : (NSError **)outError;
 
 /** Convenience method that calls both setConfiguration and setActive.
- *  |lockForConfiguration| must be called first.
+ *  `lockForConfiguration` must be called first.
  */
 - (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration
                   active:(BOOL)active

diff --git a/...rk/Headers/RTCAudioSessionConfiguration.h → ...rk/Headers/RTCAudioSessionConfiguration.h b/...rk/Headers/RTCAudioSessionConfiguration.h → ...rk/Headers/RTCAudioSessionConfiguration.h
@@ -11,7 +11,7 @@
 #import <AVFoundation/AVFoundation.h>
 #import <Foundation/Foundation.h>
 
-#import "RTCMacros.h"
+#import <WebRTC/RTCMacros.h>
 
 NS_ASSUME_NONNULL_BEGIN
 

diff --git a/...WebRTC.framework/Headers/RTCAudioSource.h → ...WebRTC.framework/Headers/RTCAudioSource.h b/...WebRTC.framework/Headers/RTCAudioSource.h → ...WebRTC.framework/Headers/RTCAudioSource.h
@@ -10,8 +10,8 @@
 
 #import <Foundation/Foundation.h>
 
-#import "RTCMacros.h"
-#import "RTCMediaSource.h"
+#import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCMediaSource.h>
 
 NS_ASSUME_NONNULL_BEGIN
 
@@ -20,7 +20,7 @@ RTC_OBJC_EXPORT
 
 - (instancetype)init NS_UNAVAILABLE;
 
-// Sets the volume for the RTCMediaSource. |volume| is a gain value in the range
+// Sets the volume for the RTCMediaSource. `volume` is a gain value in the range
 // [0, 10].
 // Temporary fix to be able to modify volume of remote audio tracks.
 // TODO(kthelgason): Property stays here temporarily until a proper volume-api

diff --git a/.../WebRTC.framework/Headers/RTCAudioTrack.h → .../WebRTC.framework/Headers/RTCAudioTrack.h b/.../WebRTC.framework/Headers/RTCAudioTrack.h → .../WebRTC.framework/Headers/RTCAudioTrack.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCMacros.h"
-#import "RTCMediaStreamTrack.h"
+#import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCMediaStreamTrack.h>
 
 NS_ASSUME_NONNULL_BEGIN
 

diff --git a/...bRTC.framework/Headers/RTCCVPixelBuffer.h → ...bRTC.framework/Headers/RTCCVPixelBuffer.h b/...bRTC.framework/Headers/RTCCVPixelBuffer.h → ...bRTC.framework/Headers/RTCCVPixelBuffer.h
@@ -10,8 +10,8 @@
 
 #import <AVFoundation/AVFoundation.h>
 
-#import "RTCMacros.h"
-#import "RTCVideoFrameBuffer.h"
+#import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCVideoFrameBuffer.h>
 
 NS_ASSUME_NONNULL_BEGIN
 
@@ -40,9 +40,9 @@ RTC_OBJC_EXPORT
 - (BOOL)requiresScalingToWidth:(int)width height:(int)height;
 - (int)bufferSizeForCroppingAndScalingToWidth:(int)width height:(int)height;
 
-/** The minimum size of the |tmpBuffer| must be the number of bytes returned from the
+/** The minimum size of the `tmpBuffer` must be the number of bytes returned from the
  * bufferSizeForCroppingAndScalingToWidth:height: method.
- * If that size is 0, the |tmpBuffer| may be nil.
+ * If that size is 0, the `tmpBuffer` may be nil.
  */
 - (BOOL)cropAndScaleTo:(CVPixelBufferRef)outputPixelBuffer
         withTempBuffer:(nullable uint8_t *)tmpBuffer;

diff --git a/...RTC.framework/Headers/RTCCallbackLogger.h → ...RTC.framework/Headers/RTCCallbackLogger.h b/...RTC.framework/Headers/RTCCallbackLogger.h → ...RTC.framework/Headers/RTCCallbackLogger.h
@@ -10,8 +10,8 @@
 
 #import <Foundation/Foundation.h>
 
-#import "RTCLogging.h"
-#import "RTCMacros.h"
+#import <WebRTC/RTCLogging.h>
+#import <WebRTC/RTCMacros.h>
 
 NS_ASSUME_NONNULL_BEGIN
 

diff --git a/....framework/Headers/RTCCameraPreviewView.h → ....framework/Headers/RTCCameraPreviewView.h b/....framework/Headers/RTCCameraPreviewView.h → ....framework/Headers/RTCCameraPreviewView.h
@@ -11,7 +11,7 @@
 #import <Foundation/Foundation.h>
 #import <UIKit/UIKit.h>
 
-#import "RTCMacros.h"
+#import <WebRTC/RTCMacros.h>
 
 @class AVCaptureSession;
 

diff --git a/...ramework/Headers/RTCCameraVideoCapturer.h → ...ramework/Headers/RTCCameraVideoCapturer.h b/...ramework/Headers/RTCCameraVideoCapturer.h → ...ramework/Headers/RTCCameraVideoCapturer.h
@@ -11,8 +11,8 @@
 #import <AVFoundation/AVFoundation.h>
 #import <Foundation/Foundation.h>
 
-#import "RTCMacros.h"
-#import "RTCVideoCapturer.h"
+#import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCVideoCapturer.h>
 
 NS_ASSUME_NONNULL_BEGIN
 
@@ -40,7 +40,7 @@ NS_EXTENSION_UNAVAILABLE_IOS("Camera not available in app extensions.")
 - (void)startCaptureWithDevice:(AVCaptureDevice *)device
                         format:(AVCaptureDeviceFormat *)format
                            fps:(NSInteger)fps
-             completionHandler:(nullable void (^)(NSError *))completionHandler;
+             completionHandler:(nullable void (^)(NSError *_Nullable))completionHandler;
 // Stops the capture session asynchronously and notifies callback on completion.
 - (void)stopCaptureWithCompletionHandler:(nullable void (^)(void))completionHandler;
 

diff --git a/...WebRTC.framework/Headers/RTCCertificate.h → ...WebRTC.framework/Headers/RTCCertificate.h b/...WebRTC.framework/Headers/RTCCertificate.h → ...WebRTC.framework/Headers/RTCCertificate.h
@@ -10,7 +10,7 @@
 
 #import <Foundation/Foundation.h>
 
-#import "RTCMacros.h"
+#import <WebRTC/RTCMacros.h>
 
 NS_ASSUME_NONNULL_BEGIN
 

diff --git a/....framework/Headers/RTCCodecSpecificInfo.h → ....framework/Headers/RTCCodecSpecificInfo.h b/....framework/Headers/RTCCodecSpecificInfo.h → ....framework/Headers/RTCCodecSpecificInfo.h
@@ -10,7 +10,7 @@
 
 #import <Foundation/Foundation.h>
 
-#import "RTCMacros.h"
+#import <WebRTC/RTCMacros.h>
 
 NS_ASSUME_NONNULL_BEGIN
 

diff --git a/...mework/Headers/RTCCodecSpecificInfoH264.h → ...mework/Headers/RTCCodecSpecificInfoH264.h b/...mework/Headers/RTCCodecSpecificInfoH264.h → ...mework/Headers/RTCCodecSpecificInfoH264.h
@@ -10,8 +10,8 @@
 
 #import <Foundation/Foundation.h>
 
-#import "RTCCodecSpecificInfo.h"
-#import "RTCMacros.h"
+#import <WebRTC/RTCCodecSpecificInfo.h>
+#import <WebRTC/RTCMacros.h>
 
 /** Class for H264 specific config. */
 typedef NS_ENUM(NSUInteger, RTCH264PacketizationMode) {