Format /sdk/objc/api folder

There are a lot of changes in /sdk so I'm splitting it

Formatting done via:

git ls-files | grep -E '^sdk\/objc\/api\/.*\.(h|cc|mm)' | xargs clang-format -i

No-Iwyu: Includes didn't change and it isn't related to formatting
Bug: webrtc:42225392
Change-Id: Ieebcd026e77db31f94df2b5dd5cd18ccc4f06674
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/373883
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43682}

sdk/objc/api/RTCVideoRendererAdapter+Private.h

@@ -23,17 +23,20 @@ NS_ASSUME_NONNULL_BEGIN
  * Calls made to the webrtc::VideoRenderInterface will be adapted and passed to
  * this video renderer.
  */
-@property(nonatomic, readonly) id<RTC_OBJC_TYPE(RTCVideoRenderer)> videoRenderer;
+@property(nonatomic, readonly) id<RTC_OBJC_TYPE(RTCVideoRenderer)>
+    videoRenderer;
 
 /**
  * The native VideoSinkInterface surface exposed by this adapter. Calls made
  * to this interface will be adapted and passed to the RTCVideoRenderer supplied
  * during construction. This pointer is unsafe and owned by this class.
  */
-@property(nonatomic, readonly) rtc::VideoSinkInterface<webrtc::VideoFrame> *nativeVideoRenderer;
+@property(nonatomic, readonly)
+    rtc::VideoSinkInterface<webrtc::VideoFrame> *nativeVideoRenderer;
 
 /** Initialize an RTCVideoRendererAdapter with an RTCVideoRenderer. */
-- (instancetype)initWithNativeRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)videoRenderer
+- (instancetype)initWithNativeRenderer:
+    (id<RTC_OBJC_TYPE(RTCVideoRenderer)>)videoRenderer
     NS_DESIGNATED_INITIALIZER;
 
 @end

sdk/objc/api/RTCVideoRendererAdapter.mm

@@ -26,11 +26,12 @@ class VideoRendererAdapter
   }
 
   void OnFrame(const webrtc::VideoFrame& nativeVideoFrame) override {
-    RTC_OBJC_TYPE(RTCVideoFrame)* videoFrame = NativeToObjCVideoFrame(nativeVideoFrame);
+    RTC_OBJC_TYPE(RTCVideoFrame)* videoFrame =
+        NativeToObjCVideoFrame(nativeVideoFrame);
 
-    CGSize current_size = (videoFrame.rotation % 180 == 0)
+    CGSize current_size = (videoFrame.rotation % 180 == 0) ?
-                              ? CGSizeMake(videoFrame.width, videoFrame.height)
+        CGSizeMake(videoFrame.width, videoFrame.height) :
-                              : CGSizeMake(videoFrame.height, videoFrame.width);
+        CGSizeMake(videoFrame.height, videoFrame.width);
 
     if (!CGSizeEqualToSize(size_, current_size)) {
       size_ = current_size;
@@ -40,10 +41,10 @@ class VideoRendererAdapter
   }
 
  private:
-  __weak RTCVideoRendererAdapter *adapter_;
+  __weak RTCVideoRendererAdapter* adapter_;
   CGSize size_;
 };
-}
+}  // namespace webrtc
 
 @implementation RTCVideoRendererAdapter {
   std::unique_ptr<webrtc::VideoRendererAdapter> _adapter;
@@ -51,7 +52,8 @@ class VideoRendererAdapter
 
 @synthesize videoRenderer = _videoRenderer;
 
-- (instancetype)initWithNativeRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)videoRenderer {
+- (instancetype)initWithNativeRenderer:
+    (id<RTC_OBJC_TYPE(RTCVideoRenderer)>)videoRenderer {
   NSParameterAssert(videoRenderer);
   self = [super init];
   if (self) {
@@ -61,7 +63,7 @@ class VideoRendererAdapter
   return self;
 }
 
-- (rtc::VideoSinkInterface<webrtc::VideoFrame> *)nativeVideoRenderer {
+- (rtc::VideoSinkInterface<webrtc::VideoFrame>*)nativeVideoRenderer {
   return _adapter.get();
 }
 

sdk/objc/api/logging/RTCCallbackLogger.h

@@ -16,8 +16,8 @@
 NS_ASSUME_NONNULL_BEGIN
 
 typedef void (^RTCCallbackLoggerMessageHandler)(NSString *message);
-typedef void (^RTCCallbackLoggerMessageAndSeverityHandler)(NSString *message,
+typedef void (^RTCCallbackLoggerMessageAndSeverityHandler)(
-                                                           RTCLoggingSeverity severity);
+    NSString *message, RTCLoggingSeverity severity);
 
 // This class intercepts WebRTC logs and forwards them to a registered block.
 // This class is not threadsafe.

sdk/objc/api/logging/RTCCallbackLogger.mm

@@ -42,24 +42,32 @@ class CallbackLogSink final : public rtc::LogSink {
 
 class CallbackWithSeverityLogSink final : public rtc::LogSink {
  public:
-  CallbackWithSeverityLogSink(RTCCallbackLoggerMessageAndSeverityHandler callbackHandler)
+  CallbackWithSeverityLogSink(
+      RTCCallbackLoggerMessageAndSeverityHandler callbackHandler)
       : callback_handler_(callbackHandler) {}
 
-  void OnLogMessage(const std::string& message) override { RTC_DCHECK_NOTREACHED(); }
+  void OnLogMessage(const std::string& message) override {
+    RTC_DCHECK_NOTREACHED();
+  }
 
-  void OnLogMessage(const std::string& message, rtc::LoggingSeverity severity) override {
+  void OnLogMessage(const std::string& message,
+                    rtc::LoggingSeverity severity) override {
     OnLogMessage(absl::string_view(message), severity);
   }
 
-  void OnLogMessage(absl::string_view message, rtc::LoggingSeverity severity) override {
+  void OnLogMessage(absl::string_view message,
+                    rtc::LoggingSeverity severity) override {
     if (callback_handler_) {
-      RTCLoggingSeverity loggingSeverity = NativeSeverityToObjcSeverity(severity);
+      RTCLoggingSeverity loggingSeverity =
-      callback_handler_([NSString stringForAbslStringView:message], loggingSeverity);
+          NativeSeverityToObjcSeverity(severity);
+      callback_handler_([NSString stringForAbslStringView:message],
+                        loggingSeverity);
     }
   }
 
  private:
-  static RTCLoggingSeverity NativeSeverityToObjcSeverity(rtc::LoggingSeverity severity) {
+  static RTCLoggingSeverity NativeSeverityToObjcSeverity(
+      rtc::LoggingSeverity severity) {
     switch (severity) {
       case rtc::LS_VERBOSE:
         return RTCLoggingSeverityVerbose;
@@ -77,7 +85,7 @@ class CallbackWithSeverityLogSink final : public rtc::LogSink {
   RTCCallbackLoggerMessageAndSeverityHandler callback_handler_;
 };
 
-}
+}  // namespace
 
 @implementation RTC_OBJC_TYPE (RTCCallbackLogger) {
   BOOL _hasStarted;

sdk/objc/api/peerconnection/RTCAudioSource+Private.h

@@ -19,16 +19,20 @@
      * The AudioSourceInterface object passed to this RTCAudioSource during
      * construction.
      */
-    @property(nonatomic,
+    @property(nonatomic, readonly)
-              readonly) rtc::scoped_refptr<webrtc::AudioSourceInterface> nativeAudioSource;
+        rtc::scoped_refptr<webrtc::AudioSourceInterface> nativeAudioSource;
 
 /** Initialize an RTCAudioSource from a native AudioSourceInterface. */
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
-              nativeAudioSource:(rtc::scoped_refptr<webrtc::AudioSourceInterface>)nativeAudioSource
+      initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+    nativeAudioSource:
+        (rtc::scoped_refptr<webrtc::AudioSourceInterface>)nativeAudioSource
     NS_DESIGNATED_INITIALIZER;
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
-              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+      initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+    nativeMediaSource:
+        (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
                  type:(RTCMediaSourceType)type NS_UNAVAILABLE;
 
 @end

sdk/objc/api/peerconnection/RTCAudioSource.mm

@@ -18,7 +18,8 @@
 @synthesize volume = _volume;
 @synthesize nativeAudioSource = _nativeAudioSource;
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
+      initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
     nativeAudioSource:
         (rtc::scoped_refptr<webrtc::AudioSourceInterface>)nativeAudioSource {
   RTC_DCHECK(factory);
@@ -33,8 +34,10 @@
   return self;
 }
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
-              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+      initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+    nativeMediaSource:
+        (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
                  type:(RTCMediaSourceType)type {
   RTC_DCHECK_NOTREACHED();
   return nil;
@@ -42,7 +45,9 @@
 
 - (NSString *)description {
   NSString *stateString = [[self class] stringForState:self.state];
-  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCAudioSource)( %p ): %@", self, stateString];
+  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCAudioSource)( %p ): %@",
+                                    self,
+                                    stateString];
 }
 
 - (void)setVolume:(double)volume {

sdk/objc/api/peerconnection/RTCAudioTrack+Private.h

@@ -19,10 +19,12 @@ NS_ASSUME_NONNULL_BEGIN
 ()
 
     /** AudioTrackInterface created or passed in at construction. */
-    @property(nonatomic, readonly) rtc::scoped_refptr<webrtc::AudioTrackInterface> nativeAudioTrack;
+    @property(nonatomic, readonly)
+        rtc::scoped_refptr<webrtc::AudioTrackInterface> nativeAudioTrack;
 
 /** Initialize an RTCAudioTrack with an id. */
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)initWithFactory:
+                    (RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
                          source:(RTC_OBJC_TYPE(RTCAudioSource) *)source
                         trackId:(NSString *)trackId;
 

sdk/objc/api/peerconnection/RTCAudioTrack.mm

@@ -21,7 +21,8 @@
 
 @synthesize source = _source;
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)initWithFactory:
+                    (RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
                          source:(RTC_OBJC_TYPE(RTCAudioSource) *)source
                         trackId:(NSString *)trackId {
   RTC_DCHECK(factory);
@@ -30,16 +31,21 @@
 
   std::string nativeId = [NSString stdStringForString:trackId];
   rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
-      factory.nativeFactory->CreateAudioTrack(nativeId, source.nativeAudioSource.get());
+      factory.nativeFactory->CreateAudioTrack(nativeId,
-  self = [self initWithFactory:factory nativeTrack:track type:RTCMediaStreamTrackTypeAudio];
+                                              source.nativeAudioSource.get());
+  self = [self initWithFactory:factory
+                   nativeTrack:track
+                          type:RTCMediaStreamTrackTypeAudio];
   if (self) {
     _source = source;
   }
   return self;
 }
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
-                    nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+    initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+        nativeTrack:
+            (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
                type:(RTCMediaStreamTrackType)type {
   NSParameterAssert(factory);
   NSParameterAssert(nativeTrack);
@@ -49,9 +55,11 @@
 
 - (RTC_OBJC_TYPE(RTCAudioSource) *)source {
   if (!_source) {
-    rtc::scoped_refptr<webrtc::AudioSourceInterface> source(self.nativeAudioTrack->GetSource());
+    rtc::scoped_refptr<webrtc::AudioSourceInterface> source(
+        self.nativeAudioTrack->GetSource());
     if (source) {
-      _source = [[RTC_OBJC_TYPE(RTCAudioSource) alloc] initWithFactory:self.factory
+      _source =
+          [[RTC_OBJC_TYPE(RTCAudioSource) alloc] initWithFactory:self.factory
                                                nativeAudioSource:source];
     }
   }

sdk/objc/api/peerconnection/RTCCertificate.h

@@ -24,10 +24,12 @@ RTC_OBJC_EXPORT
 @property(nonatomic, readonly, copy) NSString *certificate;
 
 /**
- * Initialize an RTCCertificate with PEM strings for private_key and certificate.
+ * Initialize an RTCCertificate with PEM strings for private_key and
+ * certificate.
  */
 - (instancetype)initWithPrivateKey:(NSString *)private_key
-                       certificate:(NSString *)certificate NS_DESIGNATED_INITIALIZER;
+                       certificate:(NSString *)certificate
+    NS_DESIGNATED_INITIALIZER;
 
 - (instancetype)init NS_UNAVAILABLE;
 
@@ -37,7 +39,8 @@ RTC_OBJC_EXPORT
  *  provided.
  *  - name: "ECDSA" or "RSASSA-PKCS1-v1_5"
  */
-+ (nullable RTC_OBJC_TYPE(RTCCertificate) *)generateCertificateWithParams:(NSDictionary *)params;
++ (nullable RTC_OBJC_TYPE(RTCCertificate) *)generateCertificateWithParams:
+    (NSDictionary *)params;
 
 @end
 

sdk/objc/api/peerconnection/RTCCertificate.mm

@@ -22,12 +22,14 @@
 @synthesize certificate = _certificate;
 
 - (id)copyWithZone:(NSZone *)zone {
-  id copy = [[[self class] alloc] initWithPrivateKey:[self.private_key copyWithZone:zone]
+  id copy = [[[self class] alloc]
+      initWithPrivateKey:[self.private_key copyWithZone:zone]
              certificate:[self.certificate copyWithZone:zone]];
   return copy;
 }
 
-- (instancetype)initWithPrivateKey:(NSString *)private_key certificate:(NSString *)certificate {
+- (instancetype)initWithPrivateKey:(NSString *)private_key
+                       certificate:(NSString *)certificate {
   self = [super init];
   if (self) {
     _private_key = [private_key copy];
@@ -36,7 +38,8 @@
   return self;
 }
 
-+ (nullable RTC_OBJC_TYPE(RTCCertificate) *)generateCertificateWithParams:(NSDictionary *)params {
++ (nullable RTC_OBJC_TYPE(RTCCertificate) *)generateCertificateWithParams:
+    (NSDictionary *)params {
   rtc::KeyType keyType = rtc::KT_ECDSA;
   NSString *keyTypeString = [params valueForKey:@"name"];
   if (keyTypeString && [keyTypeString isEqualToString:@"RSASSA-PKCS1-v1_5"]) {
@@ -47,11 +50,11 @@
   rtc::scoped_refptr<rtc::RTCCertificate> cc_certificate = nullptr;
   if (expires != nil) {
     uint64_t expirationTimestamp = [expires unsignedLongLongValue];
-    cc_certificate = rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType),
+    cc_certificate = rtc::RTCCertificateGenerator::GenerateCertificate(
-                                                                       expirationTimestamp);
+        rtc::KeyParams(keyType), expirationTimestamp);
   } else {
-    cc_certificate =
+    cc_certificate = rtc::RTCCertificateGenerator::GenerateCertificate(
-        rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType), std::nullopt);
+        rtc::KeyParams(keyType), std::nullopt);
   }
   if (!cc_certificate) {
     RTCLogError(@"Failed to generate certificate.");
@@ -64,8 +67,8 @@
   RTC_LOG(LS_INFO) << "CERT PEM ";
   RTC_LOG(LS_INFO) << pem_certificate;
 
-  RTC_OBJC_TYPE(RTCCertificate) *cert =
+  RTC_OBJC_TYPE(RTCCertificate) *cert = [[RTC_OBJC_TYPE(RTCCertificate) alloc]
-      [[RTC_OBJC_TYPE(RTCCertificate) alloc] initWithPrivateKey:@(pem_private_key.c_str())
+      initWithPrivateKey:@(pem_private_key.c_str())
              certificate:@(pem_certificate.c_str())];
   return cert;
 }

sdk/objc/api/peerconnection/RTCConfiguration+Private.h

@@ -17,7 +17,8 @@ NS_ASSUME_NONNULL_BEGIN
 @interface RTC_OBJC_TYPE (RTCConfiguration)
 ()
 
-    + (webrtc::PeerConnectionInterface::IceTransportsType)nativeTransportsTypeForTransportPolicy
+    + (webrtc::PeerConnectionInterface::IceTransportsType)
+          nativeTransportsTypeForTransportPolicy
     : (RTCIceTransportPolicy)policy;
 
 + (RTCIceTransportPolicy)transportPolicyForTransportsType:
@@ -41,16 +42,16 @@ NS_ASSUME_NONNULL_BEGIN
 
 + (NSString *)stringForRtcpMuxPolicy:(RTCRtcpMuxPolicy)policy;
 
-+ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativeTcpCandidatePolicyForPolicy:
++ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)
-    (RTCTcpCandidatePolicy)policy;
+    nativeTcpCandidatePolicyForPolicy:(RTCTcpCandidatePolicy)policy;
 
 + (RTCTcpCandidatePolicy)tcpCandidatePolicyForNativePolicy:
     (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativePolicy;
 
 + (NSString *)stringForTcpCandidatePolicy:(RTCTcpCandidatePolicy)policy;
 
-+ (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativeCandidateNetworkPolicyForPolicy:
++ (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)
-    (RTCCandidateNetworkPolicy)policy;
+    nativeCandidateNetworkPolicyForPolicy:(RTCCandidateNetworkPolicy)policy;
 
 + (RTCCandidateNetworkPolicy)candidateNetworkPolicyForNativePolicy:
     (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativePolicy;
@@ -59,9 +60,11 @@ NS_ASSUME_NONNULL_BEGIN
 
 + (rtc::KeyType)nativeEncryptionKeyTypeForKeyType:(RTCEncryptionKeyType)keyType;
 
-+ (webrtc::SdpSemantics)nativeSdpSemanticsForSdpSemantics:(RTCSdpSemantics)sdpSemantics;
++ (webrtc::SdpSemantics)nativeSdpSemanticsForSdpSemantics:
+    (RTCSdpSemantics)sdpSemantics;
 
-+ (RTCSdpSemantics)sdpSemanticsForNativeSdpSemantics:(webrtc::SdpSemantics)sdpSemantics;
++ (RTCSdpSemantics)sdpSemanticsForNativeSdpSemantics:
+    (webrtc::SdpSemantics)sdpSemantics;
 
 + (NSString *)stringForSdpSemantics:(RTCSdpSemantics)sdpSemantics;
 
@@ -69,10 +72,12 @@ NS_ASSUME_NONNULL_BEGIN
  * RTCConfiguration struct representation of this RTCConfiguration.
  * This is needed to pass to the underlying C++ APIs.
  */
-- (nullable webrtc::PeerConnectionInterface::RTCConfiguration *)createNativeConfiguration;
+- (nullable webrtc::PeerConnectionInterface::RTCConfiguration *)
+    createNativeConfiguration;
 
 - (instancetype)initWithNativeConfiguration:
-    (const webrtc::PeerConnectionInterface::RTCConfiguration &)config NS_DESIGNATED_INITIALIZER;
+    (const webrtc::PeerConnectionInterface::RTCConfiguration &)config
+    NS_DESIGNATED_INITIALIZER;
 
 @end
 

sdk/objc/api/peerconnection/RTCConfiguration.h

@@ -35,7 +35,10 @@ typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
 };
 
 /** Represents the rtcp mux policy. */
-typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) { RTCRtcpMuxPolicyNegotiate, RTCRtcpMuxPolicyRequire };
+typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) {
+  RTCRtcpMuxPolicyNegotiate,
+  RTCRtcpMuxPolicyRequire
+};
 
 /** Represents the tcp candidate policy. */
 typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
@@ -95,7 +98,8 @@ RTC_OBJC_EXPORT
 @property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
 @property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
 @property(nonatomic, assign) RTCCandidateNetworkPolicy candidateNetworkPolicy;
-@property(nonatomic, assign) RTCContinualGatheringPolicy continualGatheringPolicy;
+@property(nonatomic, assign)
+    RTCContinualGatheringPolicy continualGatheringPolicy;
 
 /** If set to YES, don't gather IPv6 ICE candidates on Wi-Fi.
  *  Only intended to be used on specific devices. Certain phones disable IPv6
@@ -150,7 +154,8 @@ RTC_OBJC_EXPORT
  * transport type and as a result not observed by PeerConnectionDelegateAdapter,
  * will be surfaced to the delegate.
  */
-@property(nonatomic, assign) BOOL shouldSurfaceIceCandidatesOnIceTransportTypeChanged;
+@property(nonatomic, assign)
+    BOOL shouldSurfaceIceCandidatesOnIceTransportTypeChanged;
 
 /** If set to non-nil, controls the minimal interval between consecutive ICE
  *  check packets.
@@ -225,7 +230,8 @@ RTC_OBJC_EXPORT
  * when ICE is strongly connected, and it overrides the
  * default value of this interval in the ICE implementation;
  */
-@property(nonatomic, copy, nullable) NSNumber *iceCheckIntervalStrongConnectivity;
+@property(nonatomic, copy, nullable)
+    NSNumber *iceCheckIntervalStrongConnectivity;
 
 /**
  * Defines the counterpart for ALL pairs when ICE is

sdk/objc/api/peerconnection/RTCConfiguration.mm

@@ -57,8 +57,10 @@
 @synthesize rtcpVideoReportIntervalMs = _rtcpVideoReportIntervalMs;
 @synthesize enableImplicitRollback = _enableImplicitRollback;
 @synthesize offerExtmapAllowMixed = _offerExtmapAllowMixed;
-@synthesize iceCheckIntervalStrongConnectivity = _iceCheckIntervalStrongConnectivity;
+@synthesize iceCheckIntervalStrongConnectivity =
-@synthesize iceCheckIntervalWeakConnectivity = _iceCheckIntervalWeakConnectivity;
+    _iceCheckIntervalStrongConnectivity;
+@synthesize iceCheckIntervalWeakConnectivity =
+    _iceCheckIntervalWeakConnectivity;
 @synthesize iceUnwritableTimeout = _iceUnwritableTimeout;
 @synthesize iceUnwritableMinChecks = _iceUnwritableMinChecks;
 @synthesize iceInactiveTimeout = _iceInactiveTimeout;
@@ -76,7 +78,8 @@
   if (self) {
     _enableDscp = config.dscp();
     NSMutableArray *iceServers = [NSMutableArray array];
-    for (const webrtc::PeerConnectionInterface::IceServer& server : config.servers) {
+    for (const webrtc::PeerConnectionInterface::IceServer &server :
+         config.servers) {
       RTC_OBJC_TYPE(RTCIceServer) *iceServer =
           [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithNativeServer:server];
       [iceServers addObject:iceServer];
@@ -96,18 +99,20 @@
         [[self class] bundlePolicyForNativePolicy:config.bundle_policy];
     _rtcpMuxPolicy =
         [[self class] rtcpMuxPolicyForNativePolicy:config.rtcp_mux_policy];
-    _tcpCandidatePolicy = [[self class] tcpCandidatePolicyForNativePolicy:
+    _tcpCandidatePolicy = [[self class]
-        config.tcp_candidate_policy];
+        tcpCandidatePolicyForNativePolicy:config.tcp_candidate_policy];
     _candidateNetworkPolicy = [[self class]
         candidateNetworkPolicyForNativePolicy:config.candidate_network_policy];
     webrtc::PeerConnectionInterface::ContinualGatheringPolicy nativePolicy =
         config.continual_gathering_policy;
-    _continualGatheringPolicy = [[self class] continualGatheringPolicyForNativePolicy:nativePolicy];
+    _continualGatheringPolicy =
+        [[self class] continualGatheringPolicyForNativePolicy:nativePolicy];
     _disableIPV6OnWiFi = config.disable_ipv6_on_wifi;
     _maxIPv6Networks = config.max_ipv6_networks;
     _disableLinkLocalNetworks = config.disable_link_local_networks;
     _audioJitterBufferMaxPackets = config.audio_jitter_buffer_max_packets;
-    _audioJitterBufferFastAccelerate = config.audio_jitter_buffer_fast_accelerate;
+    _audioJitterBufferFastAccelerate =
+        config.audio_jitter_buffer_fast_accelerate;
     _iceConnectionReceivingTimeout = config.ice_connection_receiving_timeout;
     _iceBackupCandidatePairPingInterval =
         config.ice_backup_candidate_pair_ping_interval;
@@ -122,30 +127,34 @@
       _iceCheckMinInterval =
           [NSNumber numberWithInt:*config.ice_check_min_interval];
     }
-    _sdpSemantics = [[self class] sdpSemanticsForNativeSdpSemantics:config.sdp_semantics];
+    _sdpSemantics =
+        [[self class] sdpSemanticsForNativeSdpSemantics:config.sdp_semantics];
     _turnCustomizer = config.turn_customizer;
     _activeResetSrtpParams = config.active_reset_srtp_params;
     if (config.crypto_options) {
       _cryptoOptions = [[RTC_OBJC_TYPE(RTCCryptoOptions) alloc]
                initWithSrtpEnableGcmCryptoSuites:config.crypto_options->srtp
                                                      .enable_gcm_crypto_suites
-             srtpEnableAes128Sha1_32CryptoCipher:config.crypto_options->srtp
+             srtpEnableAes128Sha1_32CryptoCipher:
-                                                     .enable_aes128_sha1_32_crypto_cipher
+                 config.crypto_options->srtp.enable_aes128_sha1_32_crypto_cipher
-          srtpEnableEncryptedRtpHeaderExtensions:config.crypto_options->srtp
+          srtpEnableEncryptedRtpHeaderExtensions:
-                                                     .enable_encrypted_rtp_header_extensions
+              config.crypto_options->srtp.enable_encrypted_rtp_header_extensions
                     sframeRequireFrameEncryption:config.crypto_options->sframe
                                                      .require_frame_encryption];
     }
-    _turnLoggingId = [NSString stringWithUTF8String:config.turn_logging_id.c_str()];
+    _turnLoggingId =
+        [NSString stringWithUTF8String:config.turn_logging_id.c_str()];
     _rtcpAudioReportIntervalMs = config.audio_rtcp_report_interval_ms();
     _rtcpVideoReportIntervalMs = config.video_rtcp_report_interval_ms();
     _enableImplicitRollback = config.enable_implicit_rollback;
     _offerExtmapAllowMixed = config.offer_extmap_allow_mixed;
     _iceCheckIntervalStrongConnectivity =
         config.ice_check_interval_strong_connectivity.has_value() ?
-        [NSNumber numberWithInt:*config.ice_check_interval_strong_connectivity] :
+        [NSNumber
+            numberWithInt:*config.ice_check_interval_strong_connectivity] :
         nil;
-    _iceCheckIntervalWeakConnectivity = config.ice_check_interval_weak_connectivity.has_value() ?
+    _iceCheckIntervalWeakConnectivity =
+        config.ice_check_interval_weak_connectivity.has_value() ?
         [NSNumber numberWithInt:*config.ice_check_interval_weak_connectivity] :
         nil;
     _iceUnwritableTimeout = config.ice_unwritable_timeout.has_value() ?
@@ -162,19 +171,23 @@
 }
 
 - (NSString *)description {
-  static NSString *formatString = @"RTC_OBJC_TYPE(RTCConfiguration): "
+  static NSString *formatString =
+      @"RTC_OBJC_TYPE(RTCConfiguration): "
       @"{\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n"
       @"%d\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n%d\n}\n";
 
   return [NSString
-      stringWithFormat:formatString,
+      stringWithFormat:
+          formatString,
           _iceServers,
           [[self class] stringForTransportPolicy:_iceTransportPolicy],
           [[self class] stringForBundlePolicy:_bundlePolicy],
           [[self class] stringForRtcpMuxPolicy:_rtcpMuxPolicy],
           [[self class] stringForTcpCandidatePolicy:_tcpCandidatePolicy],
-                       [[self class] stringForCandidateNetworkPolicy:_candidateNetworkPolicy],
+          [[self class]
-                       [[self class] stringForContinualGatheringPolicy:_continualGatheringPolicy],
+              stringForCandidateNetworkPolicy:_candidateNetworkPolicy],
+          [[self class]
+              stringForContinualGatheringPolicy:_continualGatheringPolicy],
           [[self class] stringForSdpSemantics:_sdpSemantics],
           _audioJitterBufferMaxPackets,
           _audioJitterBufferFastAccelerate,
@@ -215,8 +228,8 @@
       [[self class] nativeTcpCandidatePolicyForPolicy:_tcpCandidatePolicy];
   nativeConfig->candidate_network_policy = [[self class]
       nativeCandidateNetworkPolicyForPolicy:_candidateNetworkPolicy];
-  nativeConfig->continual_gathering_policy =
+  nativeConfig->continual_gathering_policy = [[self class]
-      [[self class] nativeContinualGatheringPolicyForPolicy:_continualGatheringPolicy];
+      nativeContinualGatheringPolicyForPolicy:_continualGatheringPolicy];
   nativeConfig->disable_ipv6_on_wifi = _disableIPV6OnWiFi;
   nativeConfig->max_ipv6_networks = _maxIPv6Networks;
   nativeConfig->disable_link_local_networks = _disableLinkLocalNetworks;
@@ -234,8 +247,10 @@
     RTC_LOG(LS_INFO) << "Have configured cert - using it.";
     std::string pem_private_key = [[_certificate private_key] UTF8String];
     std::string pem_certificate = [[_certificate certificate] UTF8String];
-    rtc::RTCCertificatePEM pem = rtc::RTCCertificatePEM(pem_private_key, pem_certificate);
+    rtc::RTCCertificatePEM pem =
-    rtc::scoped_refptr<rtc::RTCCertificate> certificate = rtc::RTCCertificate::FromPEM(pem);
+        rtc::RTCCertificatePEM(pem_private_key, pem_certificate);
+    rtc::scoped_refptr<rtc::RTCCertificate> certificate =
+        rtc::RTCCertificate::FromPEM(pem);
     RTC_LOG(LS_INFO) << "Created cert from PEM strings.";
     if (!certificate) {
       RTC_LOG(LS_ERROR) << "Failed to generate certificate from PEM.";
@@ -247,8 +262,8 @@
     // Generate non-default certificate.
     if (keyType != rtc::KT_DEFAULT) {
       rtc::scoped_refptr<rtc::RTCCertificate> certificate =
-          rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType),
+          rtc::RTCCertificateGenerator::GenerateCertificate(
-                                                            std::optional<uint64_t>());
+              rtc::KeyParams(keyType), std::optional<uint64_t>());
       if (!certificate) {
         RTCLogError(@"Failed to generate certificate.");
         return nullptr;
@@ -263,13 +278,16 @@
   nativeConfig->surface_ice_candidates_on_ice_transport_type_changed =
       _shouldSurfaceIceCandidatesOnIceTransportTypeChanged ? true : false;
   if (_iceCheckMinInterval != nil) {
-    nativeConfig->ice_check_min_interval = std::optional<int>(_iceCheckMinInterval.intValue);
+    nativeConfig->ice_check_min_interval =
+        std::optional<int>(_iceCheckMinInterval.intValue);
   }
-  nativeConfig->sdp_semantics = [[self class] nativeSdpSemanticsForSdpSemantics:_sdpSemantics];
+  nativeConfig->sdp_semantics =
+      [[self class] nativeSdpSemanticsForSdpSemantics:_sdpSemantics];
   if (_turnCustomizer) {
     nativeConfig->turn_customizer = _turnCustomizer;
   }
-  nativeConfig->active_reset_srtp_params = _activeResetSrtpParams ? true : false;
+  nativeConfig->active_reset_srtp_params =
+      _activeResetSrtpParams ? true : false;
   if (_cryptoOptions) {
     webrtc::CryptoOptions nativeCryptoOptions;
     nativeCryptoOptions.srtp.enable_gcm_crypto_suites =
@@ -280,7 +298,8 @@
         _cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions ? true : false;
     nativeCryptoOptions.sframe.require_frame_encryption =
         _cryptoOptions.sframeRequireFrameEncryption ? true : false;
-    nativeConfig->crypto_options = std::optional<webrtc::CryptoOptions>(nativeCryptoOptions);
+    nativeConfig->crypto_options =
+        std::optional<webrtc::CryptoOptions>(nativeCryptoOptions);
   }
   nativeConfig->turn_logging_id = [_turnLoggingId UTF8String];
   nativeConfig->set_audio_rtcp_report_interval_ms(_rtcpAudioReportIntervalMs);
@@ -296,13 +315,16 @@
         std::optional<int>(_iceCheckIntervalWeakConnectivity.intValue);
   }
   if (_iceUnwritableTimeout != nil) {
-    nativeConfig->ice_unwritable_timeout = std::optional<int>(_iceUnwritableTimeout.intValue);
+    nativeConfig->ice_unwritable_timeout =
+        std::optional<int>(_iceUnwritableTimeout.intValue);
   }
   if (_iceUnwritableMinChecks != nil) {
-    nativeConfig->ice_unwritable_min_checks = std::optional<int>(_iceUnwritableMinChecks.intValue);
+    nativeConfig->ice_unwritable_min_checks =
+        std::optional<int>(_iceUnwritableMinChecks.intValue);
   }
   if (_iceInactiveTimeout != nil) {
-    nativeConfig->ice_inactive_timeout = std::optional<int>(_iceInactiveTimeout.intValue);
+    nativeConfig->ice_inactive_timeout =
+        std::optional<int>(_iceInactiveTimeout.intValue);
   }
   return nativeConfig.release();
 }
@@ -512,7 +534,8 @@
   }
 }
 
-+ (webrtc::SdpSemantics)nativeSdpSemanticsForSdpSemantics:(RTCSdpSemantics)sdpSemantics {
++ (webrtc::SdpSemantics)nativeSdpSemanticsForSdpSemantics:
+    (RTCSdpSemantics)sdpSemantics {
   switch (sdpSemantics) {
     case RTCSdpSemanticsPlanB:
       return webrtc::SdpSemantics::kPlanB_DEPRECATED;
@@ -521,7 +544,8 @@
   }
 }
 
-+ (RTCSdpSemantics)sdpSemanticsForNativeSdpSemantics:(webrtc::SdpSemantics)sdpSemantics {
++ (RTCSdpSemantics)sdpSemanticsForNativeSdpSemantics:
+    (webrtc::SdpSemantics)sdpSemantics {
   switch (sdpSemantics) {
     case webrtc::SdpSemantics::kPlanB_DEPRECATED:
       return RTCSdpSemanticsPlanB;

sdk/objc/api/peerconnection/RTCCryptoOptions.h

@@ -50,9 +50,12 @@ RTC_OBJC_EXPORT
  * Initializes CryptoOptions with all possible options set explicitly. This
  * is done when converting from a native RTCConfiguration.crypto_options.
  */
-- (instancetype)initWithSrtpEnableGcmCryptoSuites:(BOOL)srtpEnableGcmCryptoSuites
+- (instancetype)
-              srtpEnableAes128Sha1_32CryptoCipher:(BOOL)srtpEnableAes128Sha1_32CryptoCipher
+         initWithSrtpEnableGcmCryptoSuites:(BOOL)srtpEnableGcmCryptoSuites
-           srtpEnableEncryptedRtpHeaderExtensions:(BOOL)srtpEnableEncryptedRtpHeaderExtensions
+       srtpEnableAes128Sha1_32CryptoCipher:
+           (BOOL)srtpEnableAes128Sha1_32CryptoCipher
+    srtpEnableEncryptedRtpHeaderExtensions:
+        (BOOL)srtpEnableEncryptedRtpHeaderExtensions
               sframeRequireFrameEncryption:(BOOL)sframeRequireFrameEncryption
     NS_DESIGNATED_INITIALIZER;
 

sdk/objc/api/peerconnection/RTCCryptoOptions.mm

@@ -13,19 +13,25 @@
 @implementation RTC_OBJC_TYPE (RTCCryptoOptions)
 
 @synthesize srtpEnableGcmCryptoSuites = _srtpEnableGcmCryptoSuites;
-@synthesize srtpEnableAes128Sha1_32CryptoCipher = _srtpEnableAes128Sha1_32CryptoCipher;
+@synthesize srtpEnableAes128Sha1_32CryptoCipher =
-@synthesize srtpEnableEncryptedRtpHeaderExtensions = _srtpEnableEncryptedRtpHeaderExtensions;
+    _srtpEnableAes128Sha1_32CryptoCipher;
+@synthesize srtpEnableEncryptedRtpHeaderExtensions =
+    _srtpEnableEncryptedRtpHeaderExtensions;
 @synthesize sframeRequireFrameEncryption = _sframeRequireFrameEncryption;
 
-- (instancetype)initWithSrtpEnableGcmCryptoSuites:(BOOL)srtpEnableGcmCryptoSuites
+- (instancetype)
-              srtpEnableAes128Sha1_32CryptoCipher:(BOOL)srtpEnableAes128Sha1_32CryptoCipher
+         initWithSrtpEnableGcmCryptoSuites:(BOOL)srtpEnableGcmCryptoSuites
-           srtpEnableEncryptedRtpHeaderExtensions:(BOOL)srtpEnableEncryptedRtpHeaderExtensions
+       srtpEnableAes128Sha1_32CryptoCipher:
+           (BOOL)srtpEnableAes128Sha1_32CryptoCipher
+    srtpEnableEncryptedRtpHeaderExtensions:
+        (BOOL)srtpEnableEncryptedRtpHeaderExtensions
               sframeRequireFrameEncryption:(BOOL)sframeRequireFrameEncryption {
   self = [super init];
   if (self) {
     _srtpEnableGcmCryptoSuites = srtpEnableGcmCryptoSuites;
     _srtpEnableAes128Sha1_32CryptoCipher = srtpEnableAes128Sha1_32CryptoCipher;
-    _srtpEnableEncryptedRtpHeaderExtensions = srtpEnableEncryptedRtpHeaderExtensions;
+    _srtpEnableEncryptedRtpHeaderExtensions =
+        srtpEnableEncryptedRtpHeaderExtensions;
     _sframeRequireFrameEncryption = sframeRequireFrameEncryption;
   }
   return self;

sdk/objc/api/peerconnection/RTCDataChannel+Private.h

@@ -21,8 +21,8 @@ NS_ASSUME_NONNULL_BEGIN
 ()
 
     /**
-     * The native DataBuffer representation of this RTCDatabuffer object. This is
+     * The native DataBuffer representation of this RTCDatabuffer object. This
-     * needed to pass to the underlying C++ APIs.
+     * is needed to pass to the underlying C++ APIs.
      */
     @property(nonatomic, readonly) const webrtc::DataBuffer *nativeDataBuffer;
 
@@ -37,7 +37,8 @@ NS_ASSUME_NONNULL_BEGIN
     /** Initialize an RTCDataChannel from a native DataChannelInterface. */
     - (instancetype)initWithFactory
     : (RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory nativeDataChannel
-    : (rtc::scoped_refptr<webrtc::DataChannelInterface>)nativeDataChannel NS_DESIGNATED_INITIALIZER;
+    : (rtc::scoped_refptr<webrtc::DataChannelInterface>)
+          nativeDataChannel NS_DESIGNATED_INITIALIZER;
 
 + (webrtc::DataChannelInterface::DataState)nativeDataChannelStateForState:
     (RTCDataChannelState)state;

sdk/objc/api/peerconnection/RTCDataChannel.h

@@ -40,7 +40,8 @@ RTC_OBJC_EXPORT
 (RTCDataChannelDelegate)<NSObject>
 
     /** The data channel state changed. */
-    - (void)dataChannelDidChangeState : (RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel;
+    - (void)dataChannelDidChangeState
+    : (RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel;
 
 /** The data channel successfully received a data buffer. */
 - (void)dataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel
@@ -77,7 +78,8 @@ RTC_OBJC_EXPORT
 @property(nonatomic, readonly) BOOL isOrdered;
 
 /** Deprecated. Use maxPacketLifeTime. */
-@property(nonatomic, readonly) NSUInteger maxRetransmitTime DEPRECATED_ATTRIBUTE;
+@property(nonatomic, readonly)
+    NSUInteger maxRetransmitTime DEPRECATED_ATTRIBUTE;
 
 /**
  * The length of the time window (in milliseconds) during which transmissions

sdk/objc/api/peerconnection/RTCDataChannel.mm

@@ -18,13 +18,15 @@ namespace webrtc {
 
 class DataChannelDelegateAdapter : public DataChannelObserver {
  public:
-  DataChannelDelegateAdapter(RTC_OBJC_TYPE(RTCDataChannel) * channel) { channel_ = channel; }
+  DataChannelDelegateAdapter(RTC_OBJC_TYPE(RTCDataChannel) * channel) {
+    channel_ = channel;
+  }
 
   void OnStateChange() override {
     [channel_.delegate dataChannelDidChangeState:channel_];
   }
 
-  void OnMessage(const DataBuffer& buffer) override {
+  void OnMessage(const DataBuffer &buffer) override {
     RTC_OBJC_TYPE(RTCDataBuffer) *data_buffer =
         [[RTC_OBJC_TYPE(RTCDataBuffer) alloc] initWithNativeBuffer:buffer];
     [channel_.delegate dataChannel:channel_
@@ -42,7 +44,7 @@ class DataChannelDelegateAdapter : public DataChannelObserver {
  private:
   __weak RTC_OBJC_TYPE(RTCDataChannel) * channel_;
 };
-}
+}  // namespace webrtc
 
 @implementation RTC_OBJC_TYPE (RTCDataBuffer) {
   std::unique_ptr<webrtc::DataBuffer> _dataBuffer;
@@ -52,8 +54,8 @@ class DataChannelDelegateAdapter : public DataChannelObserver {
   NSParameterAssert(data);
   self = [super init];
   if (self) {
-    rtc::CopyOnWriteBuffer buffer(
+    rtc::CopyOnWriteBuffer buffer(reinterpret_cast<const uint8_t *>(data.bytes),
-        reinterpret_cast<const uint8_t*>(data.bytes), data.length);
+                                  data.length);
     _dataBuffer.reset(new webrtc::DataBuffer(buffer, isBinary));
   }
   return self;
@@ -70,7 +72,7 @@ class DataChannelDelegateAdapter : public DataChannelObserver {
 
 #pragma mark - Private
 
-- (instancetype)initWithNativeBuffer:(const webrtc::DataBuffer&)nativeBuffer {
+- (instancetype)initWithNativeBuffer:(const webrtc::DataBuffer &)nativeBuffer {
   self = [super init];
   if (self) {
     _dataBuffer.reset(new webrtc::DataBuffer(nativeBuffer));
@@ -140,8 +142,8 @@ class DataChannelDelegateAdapter : public DataChannelObserver {
 }
 
 - (RTCDataChannelState)readyState {
-  return [[self class] dataChannelStateForNativeState:
+  return
-      _nativeDataChannel->state()];
+      [[self class] dataChannelStateForNativeState:_nativeDataChannel->state()];
 }
 
 - (uint64_t)bufferedAmount {
@@ -157,7 +159,8 @@ class DataChannelDelegateAdapter : public DataChannelObserver {
 }
 
 - (NSString *)description {
-  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCDataChannel):\n%ld\n%@\n%@",
+  return
+      [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCDataChannel):\n%ld\n%@\n%@",
                                  (long)self.channelId,
                                  self.label,
                                  [[self class] stringForState:self.readyState]];
@@ -165,7 +168,8 @@ class DataChannelDelegateAdapter : public DataChannelObserver {
 
 #pragma mark - Private
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
+      initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
     nativeDataChannel:
         (rtc::scoped_refptr<webrtc::DataChannelInterface>)nativeDataChannel {
   NSParameterAssert(nativeDataChannel);
@@ -179,8 +183,8 @@ class DataChannelDelegateAdapter : public DataChannelObserver {
   return self;
 }
 
-+ (webrtc::DataChannelInterface::DataState)
++ (webrtc::DataChannelInterface::DataState)nativeDataChannelStateForState:
-    nativeDataChannelStateForState:(RTCDataChannelState)state {
+    (RTCDataChannelState)state {
   switch (state) {
     case RTCDataChannelStateConnecting:
       return webrtc::DataChannelInterface::DataState::kConnecting;

sdk/objc/api/peerconnection/RTCDataChannelConfiguration+Private.h

@@ -17,7 +17,8 @@ NS_ASSUME_NONNULL_BEGIN
 @interface RTC_OBJC_TYPE (RTCDataChannelConfiguration)
 ()
 
-    @property(nonatomic, readonly) webrtc::DataChannelInit nativeDataChannelInit;
+    @property(nonatomic,
+              readonly) webrtc::DataChannelInit nativeDataChannelInit;
 
 @end
 

sdk/objc/api/peerconnection/RTCDtmfSender+Private.h

@@ -23,7 +23,8 @@ NS_ASSUME_NONNULL_BEGIN
 
 /** Initialize an RTCDtmfSender with a native DtmfSenderInterface. */
 - (instancetype)initWithNativeDtmfSender:
-    (rtc::scoped_refptr<webrtc::DtmfSenderInterface>)nativeDtmfSender NS_DESIGNATED_INITIALIZER;
+    (rtc::scoped_refptr<webrtc::DtmfSenderInterface>)nativeDtmfSender
+    NS_DESIGNATED_INITIALIZER;
 
 @end
 

sdk/objc/api/peerconnection/RTCDtmfSender.h

@@ -20,8 +20,9 @@ RTC_OBJC_EXPORT
 
     /**
      * Returns true if this RTCDtmfSender is capable of sending DTMF. Otherwise
-     * returns false. To be able to send DTMF, the associated RTCRtpSender must be
+     * returns false. To be able to send DTMF, the associated RTCRtpSender must
-     * able to send packets, and a "telephone-event" codec must be negotiated.
+     * be able to send packets, and a "telephone-event" codec must be
+     * negotiated.
      */
     @property(nonatomic, readonly) BOOL canInsertDtmf;
 
@@ -54,15 +55,16 @@ RTC_OBJC_EXPORT
 - (nonnull NSString *)remainingTones;
 
 /**
- * The current tone duration value. This value will be the value last set via the
+ * The current tone duration value. This value will be the value last set via
- * insertDtmf method, or the default value of 100 ms if insertDtmf was never called.
+ * the insertDtmf method, or the default value of 100 ms if insertDtmf was never
+ * called.
  */
 - (NSTimeInterval)duration;
 
 /**
- * The current value of the between-tone gap. This value will be the value last set
+ * The current value of the between-tone gap. This value will be the value last
- * via the insertDtmf() method, or the default value of 50 ms if insertDtmf() was never
+ * set via the insertDtmf() method, or the default value of 50 ms if
- * called.
+ * insertDtmf() was never called.
  */
 - (NSTimeInterval)interToneGap;
 

sdk/objc/api/peerconnection/RTCDtmfSender.mm

@@ -30,7 +30,8 @@
   RTC_DCHECK(tones != nil);
 
   int durationMs = static_cast<int>(duration * rtc::kNumMillisecsPerSec);
-  int interToneGapMs = static_cast<int>(interToneGap * rtc::kNumMillisecsPerSec);
+  int interToneGapMs =
+      static_cast<int>(interToneGap * rtc::kNumMillisecsPerSec);
   return _nativeDtmfSender->InsertDtmf(
       [NSString stdStringForString:tones], durationMs, interToneGapMs);
 }
@@ -40,7 +41,8 @@
 }
 
 - (NSTimeInterval)duration {
-  return static_cast<NSTimeInterval>(_nativeDtmfSender->duration()) / rtc::kNumMillisecsPerSec;
+  return static_cast<NSTimeInterval>(_nativeDtmfSender->duration()) /
+      rtc::kNumMillisecsPerSec;
 }
 
 - (NSTimeInterval)interToneGap {
@@ -49,7 +51,9 @@
 }
 
 - (NSString *)description {
-  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCDtmfSender) {\n  remainingTones: %@\n  "
+  return
+      [NSString stringWithFormat:
+                    @"RTC_OBJC_TYPE(RTCDtmfSender) {\n  remainingTones: %@\n  "
                     @"duration: %f sec\n  interToneGap: %f sec\n}",
                     [self remainingTones],
                     [self duration],
@@ -68,8 +72,9 @@
   self = [super init];
   if (self) {
     _nativeDtmfSender = nativeDtmfSender;
-    RTCLogInfo(
+    RTCLogInfo(@"RTC_OBJC_TYPE(RTCDtmfSender)(%p): created DTMF sender: %@",
-        @"RTC_OBJC_TYPE(RTCDtmfSender)(%p): created DTMF sender: %@", self, self.description);
+               self,
+               self.description);
   }
   return self;
 }

sdk/objc/api/peerconnection/RTCEncodedImage+Private.h

@@ -18,7 +18,8 @@ NS_ASSUME_NONNULL_BEGIN
 @interface RTC_OBJC_TYPE (RTCEncodedImage)
 (Private)
 
-    - (instancetype)initWithNativeEncodedImage : (const webrtc::EncodedImage &)encodedImage;
+    - (instancetype)initWithNativeEncodedImage
+    : (const webrtc::EncodedImage &)encodedImage;
 - (webrtc::EncodedImage)nativeEncodedImage;
 
 @end

sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm

@@ -15,13 +15,16 @@
 #include "rtc_base/numerics/safe_conversions.h"
 
 namespace {
-// An implementation of EncodedImageBufferInterface that doesn't perform any copies.
+// An implementation of EncodedImageBufferInterface that doesn't perform any
+// copies.
 class ObjCEncodedImageBuffer : public webrtc::EncodedImageBufferInterface {
  public:
   static rtc::scoped_refptr<ObjCEncodedImageBuffer> Create(NSData *data) {
     return rtc::make_ref_counted<ObjCEncodedImageBuffer>(data);
   }
-  const uint8_t *data() const override { return static_cast<const uint8_t *>(data_.bytes); }
+  const uint8_t *data() const override {
+    return static_cast<const uint8_t *>(data_.bytes);
+  }
   // TODO(bugs.webrtc.org/9378): delete this non-const data method.
   uint8_t *data() override {
     return const_cast<uint8_t *>(static_cast<const uint8_t *>(data_.bytes));
@@ -34,12 +37,13 @@ class ObjCEncodedImageBuffer : public webrtc::EncodedImageBufferInterface {
 
   NSData *data_;
 };
-}
+}  // namespace
 
-// A simple wrapper around webrtc::EncodedImageBufferInterface to make it usable with associated
+// A simple wrapper around webrtc::EncodedImageBufferInterface to make it usable
-// objects.
+// with associated objects.
 @interface RTCWrappedEncodedImageBuffer : NSObject
-@property(nonatomic) rtc::scoped_refptr<webrtc::EncodedImageBufferInterface> buffer;
+@property(nonatomic) rtc::scoped_refptr<webrtc::EncodedImageBufferInterface>
+    buffer;
 - (instancetype)initWithEncodedImageBuffer:
     (rtc::scoped_refptr<webrtc::EncodedImageBufferInterface>)buffer;
 @end
@@ -64,7 +68,8 @@ class ObjCEncodedImageBuffer : public webrtc::EncodedImageBufferInterface {
   return wrappedBuffer.buffer;
 }
 
-- (void)setEncodedData:(rtc::scoped_refptr<webrtc::EncodedImageBufferInterface>)buffer {
+- (void)setEncodedData:
+    (rtc::scoped_refptr<webrtc::EncodedImageBufferInterface>)buffer {
   return objc_setAssociatedObject(
       self,
       @selector(encodedData),
@@ -72,18 +77,20 @@ class ObjCEncodedImageBuffer : public webrtc::EncodedImageBufferInterface {
       OBJC_ASSOCIATION_RETAIN_NONATOMIC);
 }
 
-- (instancetype)initWithNativeEncodedImage:(const webrtc::EncodedImage &)encodedImage {
+- (instancetype)initWithNativeEncodedImage:
+    (const webrtc::EncodedImage &)encodedImage {
   self = [super init];
   if (self) {
-    // A reference to the encodedData must be stored so that it's kept alive as long
+    // A reference to the encodedData must be stored so that it's kept alive as
-    // self.buffer references its underlying data.
+    // long self.buffer references its underlying data.
     self.encodedData = encodedImage.GetEncodedData();
     // Wrap the buffer in NSData without copying, do not take ownership.
     self.buffer = [NSData dataWithBytesNoCopy:self.encodedData->data()
                                        length:encodedImage.size()
                                  freeWhenDone:NO];
     self.encodedWidth = rtc::dchecked_cast<int32_t>(encodedImage._encodedWidth);
-    self.encodedHeight = rtc::dchecked_cast<int32_t>(encodedImage._encodedHeight);
+    self.encodedHeight =
+        rtc::dchecked_cast<int32_t>(encodedImage._encodedHeight);
     self.timeStamp = encodedImage.RtpTimestamp();
     self.captureTimeMs = encodedImage.capture_time_ms_;
     self.ntpTimeMs = encodedImage.ntp_time_ms_;
@@ -93,7 +100,8 @@ class ObjCEncodedImageBuffer : public webrtc::EncodedImageBufferInterface {
     self.frameType = static_cast<RTCFrameType>(encodedImage._frameType);
     self.rotation = static_cast<RTCVideoRotation>(encodedImage.rotation_);
     self.qp = @(encodedImage.qp_);
-    self.contentType = (encodedImage.content_type_ == webrtc::VideoContentType::SCREENSHARE) ?
+    self.contentType =
+        (encodedImage.content_type_ == webrtc::VideoContentType::SCREENSHARE) ?
         RTCVideoContentTypeScreenshare :
         RTCVideoContentTypeUnspecified;
   }
@@ -111,7 +119,8 @@ class ObjCEncodedImageBuffer : public webrtc::EncodedImageBufferInterface {
   }
   encodedImage.set_size(self.buffer.length);
   encodedImage._encodedWidth = rtc::dchecked_cast<uint32_t>(self.encodedWidth);
-  encodedImage._encodedHeight = rtc::dchecked_cast<uint32_t>(self.encodedHeight);
+  encodedImage._encodedHeight =
+      rtc::dchecked_cast<uint32_t>(self.encodedHeight);
   encodedImage.SetRtpTimestamp(self.timeStamp);
   encodedImage.capture_time_ms_ = self.captureTimeMs;
   encodedImage.ntp_time_ms_ = self.ntpTimeMs;
@@ -121,7 +130,8 @@ class ObjCEncodedImageBuffer : public webrtc::EncodedImageBufferInterface {
   encodedImage._frameType = webrtc::VideoFrameType(self.frameType);
   encodedImage.rotation_ = webrtc::VideoRotation(self.rotation);
   encodedImage.qp_ = self.qp ? self.qp.intValue : -1;
-  encodedImage.content_type_ = (self.contentType == RTCVideoContentTypeScreenshare) ?
+  encodedImage.content_type_ =
+      (self.contentType == RTCVideoContentTypeScreenshare) ?
       webrtc::VideoContentType::SCREENSHARE :
       webrtc::VideoContentType::UNSPECIFIED;
 

sdk/objc/api/peerconnection/RTCFieldTrials.h

@@ -12,7 +12,8 @@
 
 #import "sdk/objc/base/RTCMacros.h"
 
-/** The only valid value for the following if set is kRTCFieldTrialEnabledValue. */
+/** The only valid value for the following if set is kRTCFieldTrialEnabledValue.
+ */
 RTC_EXTERN NSString *const kRTCFieldTrialAudioForceABWENoTWCCKey;
 RTC_EXTERN NSString *const kRTCFieldTrialFlexFec03AdvertisedKey;
 RTC_EXTERN NSString *const kRTCFieldTrialFlexFec03Key;
@@ -27,4 +28,5 @@ RTC_EXTERN NSString *const kRTCFieldTrialEnabledValue;
  * values. See above for valid keys and values. Must be called before any other
  * call into WebRTC. See: webrtc/system_wrappers/include/field_trial.h
  */
-RTC_EXTERN void RTCInitFieldTrialDictionary(NSDictionary<NSString *, NSString *> *fieldTrials);
+RTC_EXTERN void RTCInitFieldTrialDictionary(
+    NSDictionary<NSString *, NSString *> *fieldTrials);

sdk/objc/api/peerconnection/RTCFieldTrials.mm

@@ -16,29 +16,35 @@
 
 #include "system_wrappers/include/field_trial.h"
 
-NSString *const kRTCFieldTrialAudioForceABWENoTWCCKey = @"WebRTC-Audio-ABWENoTWCC";
+NSString *const kRTCFieldTrialAudioForceABWENoTWCCKey =
-NSString * const kRTCFieldTrialFlexFec03AdvertisedKey = @"WebRTC-FlexFEC-03-Advertised";
+    @"WebRTC-Audio-ABWENoTWCC";
-NSString * const kRTCFieldTrialFlexFec03Key = @"WebRTC-FlexFEC-03";
+NSString *const kRTCFieldTrialFlexFec03AdvertisedKey =
-NSString * const kRTCFieldTrialH264HighProfileKey = @"WebRTC-H264HighProfile";
+    @"WebRTC-FlexFEC-03-Advertised";
-NSString * const kRTCFieldTrialMinimizeResamplingOnMobileKey =
+NSString *const kRTCFieldTrialFlexFec03Key = @"WebRTC-FlexFEC-03";
+NSString *const kRTCFieldTrialH264HighProfileKey = @"WebRTC-H264HighProfile";
+NSString *const kRTCFieldTrialMinimizeResamplingOnMobileKey =
     @"WebRTC-Audio-MinimizeResamplingOnMobile";
-NSString *const kRTCFieldTrialUseNWPathMonitor = @"WebRTC-Network-UseNWPathMonitor";
+NSString *const kRTCFieldTrialUseNWPathMonitor =
-NSString * const kRTCFieldTrialEnabledValue = @"Enabled";
+    @"WebRTC-Network-UseNWPathMonitor";
+NSString *const kRTCFieldTrialEnabledValue = @"Enabled";
 
 // InitFieldTrialsFromString stores the char*, so the char array must outlive
 // the application.
 static char *gFieldTrialInitString = nullptr;
 
-void RTCInitFieldTrialDictionary(NSDictionary<NSString *, NSString *> *fieldTrials) {
+void RTCInitFieldTrialDictionary(
+    NSDictionary<NSString *, NSString *> *fieldTrials) {
   if (!fieldTrials) {
     RTCLogWarning(@"No fieldTrials provided.");
     return;
   }
   // Assemble the keys and values into the field trial string.
-  // We don't perform any extra format checking. That should be done by the underlying WebRTC calls.
+  // We don't perform any extra format checking. That should be done by the
+  // underlying WebRTC calls.
   NSMutableString *fieldTrialInitString = [NSMutableString string];
   for (NSString *key in fieldTrials) {
-    NSString *fieldTrialEntry = [NSString stringWithFormat:@"%@/%@/", key, fieldTrials[key]];
+    NSString *fieldTrialEntry =
+        [NSString stringWithFormat:@"%@/%@/", key, fieldTrials[key]];
     [fieldTrialInitString appendString:fieldTrialEntry];
   }
   size_t len = fieldTrialInitString.length + 1;

sdk/objc/api/peerconnection/RTCFileLogger.h

@@ -52,11 +52,13 @@ RTC_OBJC_EXPORT
 - (instancetype)init;
 
 // Create file logger with default rotation type.
-- (instancetype)initWithDirPath:(NSString *)dirPath maxFileSize:(NSUInteger)maxFileSize;
+- (instancetype)initWithDirPath:(NSString *)dirPath
+                    maxFileSize:(NSUInteger)maxFileSize;
 
 - (instancetype)initWithDirPath:(NSString *)dirPath
                     maxFileSize:(NSUInteger)maxFileSize
-                   rotationType:(RTCFileLoggerRotationType)rotationType NS_DESIGNATED_INITIALIZER;
+                   rotationType:(RTCFileLoggerRotationType)rotationType
+    NS_DESIGNATED_INITIALIZER;
 
 // Starts writing WebRTC logs to disk if not already started. Overwrites any
 // existing file(s).

sdk/objc/api/peerconnection/RTCFileLogger.mm

@@ -38,8 +38,7 @@ const char *kRTCFileLoggerRotatingLogPrefix = "rotating_log";
   NSString *documentsDirPath = [paths firstObject];
   NSString *defaultDirPath =
       [documentsDirPath stringByAppendingPathComponent:kDefaultLogDirName];
-  return [self initWithDirPath:defaultDirPath
+  return [self initWithDirPath:defaultDirPath maxFileSize:kDefaultMaxFileSize];
-                   maxFileSize:kDefaultMaxFileSize];
 }
 
 - (instancetype)initWithDirPath:(NSString *)dirPath
@@ -96,13 +95,13 @@ const char *kRTCFileLoggerRotatingLogPrefix = "rotating_log";
                                        _maxFileSize / 10));
       break;
     case RTCFileLoggerTypeCall:
-      _logSink.reset(
+      _logSink.reset(new rtc::CallSessionFileRotatingLogSink(
-          new rtc::CallSessionFileRotatingLogSink(_dirPath.UTF8String,
+          _dirPath.UTF8String, _maxFileSize));
-                                                  _maxFileSize));
       break;
   }
   if (!_logSink->Init()) {
-    RTC_LOG(LS_ERROR) << "Failed to open log files at path: " << _dirPath.UTF8String;
+    RTC_LOG(LS_ERROR) << "Failed to open log files at path: "
+                      << _dirPath.UTF8String;
     _logSink.reset();
     return;
   }
@@ -129,15 +128,16 @@ const char *kRTCFileLoggerRotatingLogPrefix = "rotating_log";
   if (_hasStarted) {
     return nil;
   }
-  NSMutableData* logData = [NSMutableData data];
+  NSMutableData *logData = [NSMutableData data];
   std::unique_ptr<rtc::FileRotatingStreamReader> stream;
-  switch(_rotationType) {
+  switch (_rotationType) {
     case RTCFileLoggerTypeApp:
-      stream = std::make_unique<rtc::FileRotatingStreamReader>(_dirPath.UTF8String,
+      stream = std::make_unique<rtc::FileRotatingStreamReader>(
-                                                               kRTCFileLoggerRotatingLogPrefix);
+          _dirPath.UTF8String, kRTCFileLoggerRotatingLogPrefix);
       break;
     case RTCFileLoggerTypeCall:
-      stream = std::make_unique<rtc::CallSessionFileRotatingStreamReader>(_dirPath.UTF8String);
+      stream = std::make_unique<rtc::CallSessionFileRotatingStreamReader>(
+          _dirPath.UTF8String);
       break;
   }
   size_t bufferSize = stream->GetSize();
@@ -146,7 +146,7 @@ const char *kRTCFileLoggerRotatingLogPrefix = "rotating_log";
   }
   // Allocate memory using malloc so we can pass it direcly to NSData without
   // copying.
-  std::unique_ptr<uint8_t[]> buffer(static_cast<uint8_t*>(malloc(bufferSize)));
+  std::unique_ptr<uint8_t[]> buffer(static_cast<uint8_t *>(malloc(bufferSize)));
   size_t read = stream->ReadAll(buffer.get(), bufferSize);
   logData = [[NSMutableData alloc] initWithBytesNoCopy:buffer.release()
                                                 length:read];

sdk/objc/api/peerconnection/RTCIceCandidate+Private.h

@@ -23,13 +23,15 @@ NS_ASSUME_NONNULL_BEGIN
      * The native IceCandidateInterface representation of this RTCIceCandidate
      * object. This is needed to pass to the underlying C++ APIs.
      */
-    @property(nonatomic, readonly) std::unique_ptr<webrtc::IceCandidateInterface> nativeCandidate;
+    @property(nonatomic, readonly)
+        std::unique_ptr<webrtc::IceCandidateInterface> nativeCandidate;
 
 /**
  * Initialize an RTCIceCandidate from a native IceCandidateInterface. No
  * ownership is taken of the native candidate.
  */
-- (instancetype)initWithNativeCandidate:(const webrtc::IceCandidateInterface *)candidate;
+- (instancetype)initWithNativeCandidate:
+    (const webrtc::IceCandidateInterface *)candidate;
 
 @end
 

sdk/objc/api/peerconnection/RTCIceCandidate.h

@@ -42,7 +42,8 @@ RTC_OBJC_EXPORT
  */
 - (instancetype)initWithSdp:(NSString *)sdp
               sdpMLineIndex:(int)sdpMLineIndex
-                     sdpMid:(nullable NSString *)sdpMid NS_DESIGNATED_INITIALIZER;
+                     sdpMid:(nullable NSString *)sdpMid
+    NS_DESIGNATED_INITIALIZER;
 
 @end
 

sdk/objc/api/peerconnection/RTCIceCandidate.mm

@@ -36,7 +36,8 @@
 }
 
 - (NSString *)description {
-  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCIceCandidate):\n%@\n%d\n%@\n%@",
+  return [NSString
+      stringWithFormat:@"RTC_OBJC_TYPE(RTCIceCandidate):\n%@\n%d\n%@\n%@",
                        _sdpMid,
                        _sdpMLineIndex,
                        _sdp,
@@ -55,7 +56,8 @@
       [self initWithSdp:[NSString stringForStdString:sdp]
           sdpMLineIndex:candidate->sdp_mline_index()
                  sdpMid:[NSString stringForStdString:candidate->sdp_mid()]];
-  rtcCandidate->_serverUrl = [NSString stringForStdString:candidate->server_url()];
+  rtcCandidate->_serverUrl =
+      [NSString stringForStdString:candidate->server_url()];
   return rtcCandidate;
 }
 

sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent+Private.h

@@ -17,9 +17,9 @@ NS_ASSUME_NONNULL_BEGIN
 @interface RTC_OBJC_TYPE (RTCIceCandidateErrorEvent)
 ()
 
-    - (instancetype)initWithAddress : (const std::string&)address port : (const int)port url
+    - (instancetype)initWithAddress : (const std::string&)address port
-    : (const std::string&)url errorCode : (const int)errorCode errorText
+    : (const int)port url : (const std::string&)url errorCode
-    : (const std::string&)errorText;
+    : (const int)errorCode errorText : (const std::string&)errorText;
 
 @end
 

sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent.h

@@ -23,16 +23,19 @@ RTC_OBJC_EXPORT
 /** The port used to communicate with the STUN or TURN server. */
 @property(nonatomic, readonly) int port;
 
-/** The STUN or TURN URL that identifies the STUN or TURN server for which the failure occurred. */
+/** The STUN or TURN URL that identifies the STUN or TURN server for which the
+ * failure occurred. */
 @property(nonatomic, readonly) NSString *url;
 
-/** The numeric STUN error code returned by the STUN or TURN server. If no host candidate can reach
+/** The numeric STUN error code returned by the STUN or TURN server. If no host
- * the server, errorCode will be set to the value 701 which is outside the STUN error code range.
+ * candidate can reach the server, errorCode will be set to the value 701 which
- * This error is only fired once per server URL while in the RTCIceGatheringState of "gathering". */
+ * is outside the STUN error code range. This error is only fired once per
+ * server URL while in the RTCIceGatheringState of "gathering". */
 @property(nonatomic, readonly) int errorCode;
 
-/** The STUN reason text returned by the STUN or TURN server. If the server could not be reached,
+/** The STUN reason text returned by the STUN or TURN server. If the server
- * errorText will be set to an implementation-specific value providing details about the error. */
+ * could not be reached, errorText will be set to an implementation-specific
+ * value providing details about the error. */
 @property(nonatomic, readonly) NSString *errorText;
 
 - (instancetype)init NS_DESIGNATED_INITIALIZER;

sdk/objc/api/peerconnection/RTCIceServer+Private.h

@@ -21,10 +21,12 @@ NS_ASSUME_NONNULL_BEGIN
      * IceServer struct representation of this RTCIceServer object's data.
      * This is needed to pass to the underlying C++ APIs.
      */
-    @property(nonatomic, readonly) webrtc::PeerConnectionInterface::IceServer nativeServer;
+    @property(nonatomic,
+              readonly) webrtc::PeerConnectionInterface::IceServer nativeServer;
 
 /** Initialize an RTCIceServer from a native IceServer. */
-- (instancetype)initWithNativeServer:(webrtc::PeerConnectionInterface::IceServer)nativeServer;
+- (instancetype)initWithNativeServer:
+    (webrtc::PeerConnectionInterface::IceServer)nativeServer;
 
 @end
 

sdk/objc/api/peerconnection/RTCIceServer.h

@@ -100,7 +100,8 @@ RTC_OBJC_EXPORT
  * optional credential, TLS cert policy, hostname, ALPN protocols and
  * elliptic curves.
  */
-- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+- (instancetype)
+    initWithURLStrings:(NSArray<NSString *> *)urlStrings
               username:(nullable NSString *)username
             credential:(nullable NSString *)credential
          tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy

sdk/objc/api/peerconnection/RTCIceServer.mm

@@ -23,9 +23,7 @@
 @synthesize tlsEllipticCurves = _tlsEllipticCurves;
 
 - (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings {
-  return [self initWithURLStrings:urlStrings
+  return [self initWithURLStrings:urlStrings username:nil credential:nil];
-                         username:nil
-                       credential:nil];
 }
 
 - (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
@@ -91,14 +89,18 @@
     _credential = [credential copy];
     _tlsCertPolicy = tlsCertPolicy;
     _hostname = [hostname copy];
-    _tlsAlpnProtocols = [[NSArray alloc] initWithArray:tlsAlpnProtocols copyItems:YES];
+    _tlsAlpnProtocols = [[NSArray alloc] initWithArray:tlsAlpnProtocols
-    _tlsEllipticCurves = [[NSArray alloc] initWithArray:tlsEllipticCurves copyItems:YES];
+                                             copyItems:YES];
+    _tlsEllipticCurves = [[NSArray alloc] initWithArray:tlsEllipticCurves
+                                              copyItems:YES];
   }
   return self;
 }
 
 - (NSString *)description {
-  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCIceServer):\n%@\n%@\n%@\n%@\n%@\n%@\n%@",
+  return
+      [NSString stringWithFormat:
+                    @"RTC_OBJC_TYPE(RTCIceServer):\n%@\n%@\n%@\n%@\n%@\n%@\n%@",
                     _urlStrings,
                     _username,
                     _credential,
@@ -126,17 +128,18 @@
   iceServer.password = [NSString stdStringForString:_credential];
   iceServer.hostname = [NSString stdStringForString:_hostname];
 
-  [_tlsAlpnProtocols enumerateObjectsUsingBlock:^(NSString *proto, NSUInteger idx, BOOL *stop) {
+  [_tlsAlpnProtocols enumerateObjectsUsingBlock:^(
+                         NSString *proto, NSUInteger idx, BOOL *stop) {
     iceServer.tls_alpn_protocols.push_back(proto.stdString);
   }];
 
-  [_tlsEllipticCurves enumerateObjectsUsingBlock:^(NSString *curve, NSUInteger idx, BOOL *stop) {
+  [_tlsEllipticCurves enumerateObjectsUsingBlock:^(
+                          NSString *curve, NSUInteger idx, BOOL *stop) {
     iceServer.tls_elliptic_curves.push_back(curve.stdString);
   }];
 
-  [_urlStrings enumerateObjectsUsingBlock:^(NSString *url,
+  [_urlStrings
-                                            NSUInteger idx,
+      enumerateObjectsUsingBlock:^(NSString *url, NSUInteger idx, BOOL *stop) {
-                                            BOOL *stop) {
         iceServer.urls.push_back(url.stdString);
       }];
 
@@ -168,8 +171,8 @@
   for (auto const &proto : nativeServer.tls_alpn_protocols) {
     [tlsAlpnProtocols addObject:[NSString stringForStdString:proto]];
   }
-  NSMutableArray *tlsEllipticCurves =
+  NSMutableArray *tlsEllipticCurves = [NSMutableArray
-      [NSMutableArray arrayWithCapacity:nativeServer.tls_elliptic_curves.size()];
+      arrayWithCapacity:nativeServer.tls_elliptic_curves.size()];
   for (auto const &curve : nativeServer.tls_elliptic_curves) {
     [tlsEllipticCurves addObject:[NSString stringForStdString:curve]];
   }

sdk/objc/api/peerconnection/RTCLegacyStatsReport+Private.h

@@ -18,7 +18,8 @@ NS_ASSUME_NONNULL_BEGIN
 ()
 
     /** Initialize an RTCLegacyStatsReport object from a native StatsReport. */
-    - (instancetype)initWithNativeReport : (const webrtc::StatsReport &)nativeReport;
+    - (instancetype)initWithNativeReport
+    : (const webrtc::StatsReport &)nativeReport;
 
 @end
 

sdk/objc/api/peerconnection/RTCLegacyStatsReport.mm

@@ -23,7 +23,8 @@
 @synthesize values = _values;
 
 - (NSString *)description {
-  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCLegacyStatsReport):\n%@\n%@\n%f\n%@",
+  return [NSString
+      stringWithFormat:@"RTC_OBJC_TYPE(RTCLegacyStatsReport):\n%@\n%@\n%f\n%@",
                        _reportId,
                        _type,
                        _timestamp,
@@ -37,17 +38,16 @@
   if (self) {
     _timestamp = nativeReport.timestamp();
     _type = [NSString stringForStdString:nativeReport.TypeToString()];
-    _reportId = [NSString stringForStdString:
+    _reportId = [NSString stringForStdString:nativeReport.id()->ToString()];
-        nativeReport.id()->ToString()];
 
     NSUInteger capacity = nativeReport.values().size();
     NSMutableDictionary *values =
         [NSMutableDictionary dictionaryWithCapacity:capacity];
     for (auto const &valuePair : nativeReport.values()) {
-      NSString *key = [NSString stringForStdString:
+      NSString *key =
-          valuePair.second->display_name()];
+          [NSString stringForStdString:valuePair.second->display_name()];
-      NSString *value = [NSString stringForStdString:
+      NSString *value =
-          valuePair.second->ToString()];
+          [NSString stringForStdString:valuePair.second->ToString()];
 
       // Not expecting duplicate keys.
       RTC_DCHECK(![values objectForKey:key]);

sdk/objc/api/peerconnection/RTCMediaConstraints+Private.h

@@ -20,8 +20,8 @@ NS_ASSUME_NONNULL_BEGIN
 ()
 
     /**
-     * A MediaConstraints representation of this RTCMediaConstraints object. This is
+     * A MediaConstraints representation of this RTCMediaConstraints object.
-     * needed to pass to the underlying C++ APIs.
+     * This is needed to pass to the underlying C++ APIs.
      */
     - (std::unique_ptr<webrtc::MediaConstraints>)nativeConstraints;
 

sdk/objc/api/peerconnection/RTCMediaConstraints.h

@@ -36,10 +36,11 @@ RTC_OBJC_EXPORT
 - (instancetype)init NS_UNAVAILABLE;
 
 /** Initialize with mandatory and/or optional constraints. */
-- (instancetype)
+- (instancetype)initWithMandatoryConstraints:
-    initWithMandatoryConstraints:(nullable NSDictionary<NSString *, NSString *> *)mandatory
+                    (nullable NSDictionary<NSString *, NSString *> *)mandatory
-             optionalConstraints:(nullable NSDictionary<NSString *, NSString *> *)optional
+                         optionalConstraints:
-    NS_DESIGNATED_INITIALIZER;
+                             (nullable NSDictionary<NSString *, NSString *> *)
+                                 optional NS_DESIGNATED_INITIALIZER;
 
 @end
 

sdk/objc/api/peerconnection/RTCMediaConstraints.mm

@@ -17,7 +17,8 @@
 NSString *const kRTCMediaConstraintsAudioNetworkAdaptorConfig =
     @(webrtc::MediaConstraints::kAudioNetworkAdaptorConfig);
 
-NSString *const kRTCMediaConstraintsIceRestart = @(webrtc::MediaConstraints::kIceRestart);
+NSString *const kRTCMediaConstraintsIceRestart =
+    @(webrtc::MediaConstraints::kIceRestart);
 NSString *const kRTCMediaConstraintsOfferToReceiveAudio =
     @(webrtc::MediaConstraints::kOfferToReceiveAudio);
 NSString *const kRTCMediaConstraintsOfferToReceiveVideo =
@@ -25,8 +26,10 @@ NSString *const kRTCMediaConstraintsOfferToReceiveVideo =
 NSString *const kRTCMediaConstraintsVoiceActivityDetection =
     @(webrtc::MediaConstraints::kVoiceActivityDetection);
 
-NSString *const kRTCMediaConstraintsValueTrue = @(webrtc::MediaConstraints::kValueTrue);
+NSString *const kRTCMediaConstraintsValueTrue =
-NSString *const kRTCMediaConstraintsValueFalse = @(webrtc::MediaConstraints::kValueFalse);
+    @(webrtc::MediaConstraints::kValueTrue);
+NSString *const kRTCMediaConstraintsValueFalse =
+    @(webrtc::MediaConstraints::kValueFalse);
 
 @implementation RTC_OBJC_TYPE (RTCMediaConstraints) {
   NSDictionary<NSString *, NSString *> *_mandatory;
@@ -48,8 +51,10 @@ NSString *const kRTCMediaConstraintsValueFalse = @(webrtc::MediaConstraints::kVa
 }
 
 - (NSString *)description {
-  return [NSString
+  return
-      stringWithFormat:@"RTC_OBJC_TYPE(RTCMediaConstraints):\n%@\n%@", _mandatory, _optional];
+      [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCMediaConstraints):\n%@\n%@",
+                                 _mandatory,
+                                 _optional];
 }
 
 #pragma mark - Private
@@ -69,17 +74,20 @@ NSString *const kRTCMediaConstraintsValueFalse = @(webrtc::MediaConstraints::kVa
     (NSDictionary<NSString *, NSString *> *)constraints {
   webrtc::MediaConstraints::Constraints nativeConstraints;
   for (NSString *key in constraints) {
-    NSAssert([key isKindOfClass:[NSString class]],
+    NSAssert(
-             @"%@ is not an NSString.", key);
+        [key isKindOfClass:[NSString class]], @"%@ is not an NSString.", key);
     NSString *value = [constraints objectForKey:key];
     NSAssert([value isKindOfClass:[NSString class]],
-             @"%@ is not an NSString.", value);
+             @"%@ is not an NSString.",
+             value);
     if ([kRTCMediaConstraintsAudioNetworkAdaptorConfig isEqualToString:key]) {
       // This value is base64 encoded.
-      NSData *charData = [[NSData alloc] initWithBase64EncodedString:value options:0];
+      NSData *charData = [[NSData alloc] initWithBase64EncodedString:value
-      std::string configValue =
+                                                             options:0];
-          std::string(reinterpret_cast<const char *>(charData.bytes), charData.length);
+      std::string configValue = std::string(
-      nativeConstraints.push_back(webrtc::MediaConstraints::Constraint(key.stdString, configValue));
+          reinterpret_cast<const char *>(charData.bytes), charData.length);
+      nativeConstraints.push_back(
+          webrtc::MediaConstraints::Constraint(key.stdString, configValue));
     } else {
       nativeConstraints.push_back(
           webrtc::MediaConstraints::Constraint(key.stdString, value.stdString));

sdk/objc/api/peerconnection/RTCMediaSource+Private.h

@@ -24,16 +24,20 @@ typedef NS_ENUM(NSInteger, RTCMediaSourceType) {
 @interface RTC_OBJC_TYPE (RTCMediaSource)
 ()
 
-    @property(nonatomic,
+    @property(nonatomic, readonly)
-              readonly) rtc::scoped_refptr<webrtc::MediaSourceInterface> nativeMediaSource;
+        rtc::scoped_refptr<webrtc::MediaSourceInterface> nativeMediaSource;
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
-              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+      initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+    nativeMediaSource:
+        (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
                  type:(RTCMediaSourceType)type NS_DESIGNATED_INITIALIZER;
 
-+ (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:(RTCSourceState)state;
++ (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:
+    (RTCSourceState)state;
 
-+ (RTCSourceState)sourceStateForNativeState:(webrtc::MediaSourceInterface::SourceState)nativeState;
++ (RTCSourceState)sourceStateForNativeState:
+    (webrtc::MediaSourceInterface::SourceState)nativeState;
 
 + (NSString *)stringForState:(RTCSourceState)state;
 

sdk/objc/api/peerconnection/RTCMediaSource.mm

@@ -19,8 +19,10 @@
 
 @synthesize nativeMediaSource = _nativeMediaSource;
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
-              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+      initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+    nativeMediaSource:
+        (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
                  type:(RTCMediaSourceType)type {
   RTC_DCHECK(factory);
   RTC_DCHECK(nativeMediaSource);

sdk/objc/api/peerconnection/RTCMediaStream+Private.h

@@ -18,19 +18,22 @@ NS_ASSUME_NONNULL_BEGIN
 ()
 
     /**
-     * MediaStreamInterface representation of this RTCMediaStream object. This is
+     * MediaStreamInterface representation of this RTCMediaStream object. This
-     * needed to pass to the underlying C++ APIs.
+     * is needed to pass to the underlying C++ APIs.
      */
-    @property(nonatomic,
+    @property(nonatomic, readonly)
-              readonly) rtc::scoped_refptr<webrtc::MediaStreamInterface> nativeMediaStream;
+        rtc::scoped_refptr<webrtc::MediaStreamInterface> nativeMediaStream;
 
 /** Initialize an RTCMediaStream with an id. */
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)initWithFactory:
+                    (RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
                        streamId:(NSString *)streamId;
 
 /** Initialize an RTCMediaStream from a native MediaStreamInterface. */
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
-              nativeMediaStream:(rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream;
+      initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+    nativeMediaStream:
+        (rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream;
 
 @end
 

sdk/objc/api/peerconnection/RTCMediaStream.h

@@ -25,7 +25,8 @@ RTC_OBJC_EXPORT
 @property(nonatomic, strong, readonly) NSArray<RTC_OBJC_TYPE(RTCAudioTrack) *> *audioTracks;
 
 /** The video tracks in this stream. */
-@property(nonatomic, strong, readonly) NSArray<RTC_OBJC_TYPE(RTCVideoTrack) *> *videoTracks;
+@property(nonatomic, strong, readonly)
+    NSArray<RTC_OBJC_TYPE(RTCVideoTrack) *> *videoTracks;
 
 /** An identifier for this media stream. */
 @property(nonatomic, readonly) NSString *streamId;

sdk/objc/api/peerconnection/RTCMediaStream.mm

@@ -24,7 +24,8 @@
   rtc::scoped_refptr<webrtc::MediaStreamInterface> _nativeMediaStream;
 }
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)initWithFactory:
+                    (RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
                        streamId:(NSString *)streamId {
   NSParameterAssert(factory);
   NSParameterAssert(streamId.length);
@@ -36,14 +37,16 @@
 
 - (NSArray<RTC_OBJC_TYPE(RTCAudioTrack) *> *)audioTracks {
   if (!_signalingThread->IsCurrent()) {
-    return _signalingThread->BlockingCall([self]() { return self.audioTracks; });
+    return _signalingThread->BlockingCall(
+        [self]() { return self.audioTracks; });
   }
   return [_audioTracks copy];
 }
 
 - (NSArray<RTC_OBJC_TYPE(RTCVideoTrack) *> *)videoTracks {
   if (!_signalingThread->IsCurrent()) {
-    return _signalingThread->BlockingCall([self]() { return self.videoTracks; });
+    return _signalingThread->BlockingCall(
+        [self]() { return self.videoTracks; });
   }
   return [_videoTracks copy];
 }
@@ -79,7 +82,8 @@
   }
   NSUInteger index = [_audioTracks indexOfObjectIdenticalTo:audioTrack];
   if (index == NSNotFound) {
-    RTC_LOG(LS_INFO) << "|removeAudioTrack| called on unexpected RTC_OBJC_TYPE(RTCAudioTrack)";
+    RTC_LOG(LS_INFO) << "|removeAudioTrack| called on unexpected "
+                        "RTC_OBJC_TYPE(RTCAudioTrack)";
     return;
   }
   if (_nativeMediaStream->RemoveTrack(audioTrack.nativeAudioTrack)) {
@@ -94,7 +98,8 @@
   }
   NSUInteger index = [_videoTracks indexOfObjectIdenticalTo:videoTrack];
   if (index == NSNotFound) {
-    RTC_LOG(LS_INFO) << "|removeVideoTrack| called on unexpected RTC_OBJC_TYPE(RTCVideoTrack)";
+    RTC_LOG(LS_INFO) << "|removeVideoTrack| called on unexpected "
+                        "RTC_OBJC_TYPE(RTCVideoTrack)";
     return;
   }
 
@@ -104,7 +109,8 @@
 }
 
 - (NSString *)description {
-  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCMediaStream):\n%@\nA=%lu\nV=%lu",
+  return [NSString
+      stringWithFormat:@"RTC_OBJC_TYPE(RTCMediaStream):\n%@\nA=%lu\nV=%lu",
                        self.streamId,
                        (unsigned long)self.audioTracks.count,
                        (unsigned long)self.videoTracks.count];
@@ -116,7 +122,8 @@
   return _nativeMediaStream;
 }
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
+      initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
     nativeMediaStream:
         (rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream {
   NSParameterAssert(nativeMediaStream);

sdk/objc/api/peerconnection/RTCMediaStreamTrack+Private.h

@@ -31,17 +31,23 @@ NS_ASSUME_NONNULL_BEGIN
  * The native MediaStreamTrackInterface passed in or created during
  * construction.
  */
-@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack;
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>
+        nativeTrack;
 
 /**
  * Initialize an RTCMediaStreamTrack from a native MediaStreamTrackInterface.
  */
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
-                    nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+    initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+        nativeTrack:
+            (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
                type:(RTCMediaStreamTrackType)type NS_DESIGNATED_INITIALIZER;
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
-                    nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack;
+    initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+        nativeTrack:
+            (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack;
 
 - (BOOL)isEqualToTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track;
 
@@ -54,7 +60,8 @@ NS_ASSUME_NONNULL_BEGIN
 + (NSString *)stringForState:(RTCMediaStreamTrackState)state;
 
 + (RTC_OBJC_TYPE(RTCMediaStreamTrack) *)
-    mediaTrackForNativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+    mediaTrackForNativeTrack:
+        (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
                      factory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory;
 
 @end

sdk/objc/api/peerconnection/RTCMediaStreamTrack.mm

@@ -14,9 +14,9 @@
 
 #import "helpers/NSString+StdString.h"
 
-NSString * const kRTCMediaStreamTrackKindAudio =
+NSString *const kRTCMediaStreamTrackKindAudio =
     @(webrtc::MediaStreamTrackInterface::kAudioKind);
-NSString * const kRTCMediaStreamTrackKindVideo =
+NSString *const kRTCMediaStreamTrackKindVideo =
     @(webrtc::MediaStreamTrackInterface::kVideoKind);
 
 @implementation RTC_OBJC_TYPE (RTCMediaStreamTrack) {
@@ -47,7 +47,8 @@ NSString * const kRTCMediaStreamTrackKindVideo =
 
 - (NSString *)description {
   NSString *readyState = [[self class] stringForState:self.readyState];
-  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCMediaStreamTrack):\n%@\n%@\n%@\n%@",
+  return [NSString
+      stringWithFormat:@"RTC_OBJC_TYPE(RTCMediaStreamTrack):\n%@\n%@\n%@\n%@",
                        self.kind,
                        self.trackId,
                        self.isEnabled ? @"enabled" : @"disabled",
@@ -76,8 +77,10 @@ NSString * const kRTCMediaStreamTrackKindVideo =
 
 @synthesize factory = _factory;
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
-                    nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+    initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+        nativeTrack:
+            (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
                type:(RTCMediaStreamTrackType)type {
   NSParameterAssert(nativeTrack);
   NSParameterAssert(factory);
@@ -90,16 +93,22 @@ NSString * const kRTCMediaStreamTrackKindVideo =
   return self;
 }
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
-                    nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
+    initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+        nativeTrack:
+            (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
   NSParameterAssert(nativeTrack);
   if (nativeTrack->kind() ==
       std::string(webrtc::MediaStreamTrackInterface::kAudioKind)) {
-    return [self initWithFactory:factory nativeTrack:nativeTrack type:RTCMediaStreamTrackTypeAudio];
+    return [self initWithFactory:factory
+                     nativeTrack:nativeTrack
+                            type:RTCMediaStreamTrackTypeAudio];
   }
   if (nativeTrack->kind() ==
       std::string(webrtc::MediaStreamTrackInterface::kVideoKind)) {
-    return [self initWithFactory:factory nativeTrack:nativeTrack type:RTCMediaStreamTrackTypeVideo];
+    return [self initWithFactory:factory
+                     nativeTrack:nativeTrack
+                            type:RTCMediaStreamTrackTypeVideo];
   }
   return nil;
 }
@@ -141,20 +150,26 @@ NSString * const kRTCMediaStreamTrackKindVideo =
 }
 
 + (RTC_OBJC_TYPE(RTCMediaStreamTrack) *)
-    mediaTrackForNativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+    mediaTrackForNativeTrack:
-                     factory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory {
+        (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+                     factory:
+                         (RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory {
   NSParameterAssert(nativeTrack);
   NSParameterAssert(factory);
   if (nativeTrack->kind() == webrtc::MediaStreamTrackInterface::kAudioKind) {
-    return [[RTC_OBJC_TYPE(RTCAudioTrack) alloc] initWithFactory:factory
+    return [[RTC_OBJC_TYPE(RTCAudioTrack) alloc]
+        initWithFactory:factory
             nativeTrack:nativeTrack
                    type:RTCMediaStreamTrackTypeAudio];
-  } else if (nativeTrack->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) {
+  } else if (nativeTrack->kind() ==
-    return [[RTC_OBJC_TYPE(RTCVideoTrack) alloc] initWithFactory:factory
+             webrtc::MediaStreamTrackInterface::kVideoKind) {
+    return [[RTC_OBJC_TYPE(RTCVideoTrack) alloc]
+        initWithFactory:factory
             nativeTrack:nativeTrack
                    type:RTCMediaStreamTrackTypeVideo];
   } else {
-    return [[RTC_OBJC_TYPE(RTCMediaStreamTrack) alloc] initWithFactory:factory
+    return [[RTC_OBJC_TYPE(RTCMediaStreamTrack) alloc]
+        initWithFactory:factory
             nativeTrack:nativeTrack];
   }
 }

sdk/objc/api/peerconnection/RTCMetrics.h

@@ -20,4 +20,5 @@
 RTC_EXTERN void RTCEnableMetrics(void);
 
 /** Gets and clears native histograms. */
-RTC_EXTERN NSArray<RTC_OBJC_TYPE(RTCMetricsSampleInfo) *>* RTCGetAndResetMetrics(void);
+RTC_EXTERN NSArray<RTC_OBJC_TYPE(RTCMetricsSampleInfo) *>*
+    RTCGetAndResetMetrics(void);

sdk/objc/api/peerconnection/RTCMetrics.mm

@@ -19,7 +19,9 @@ void RTCEnableMetrics(void) {
 }
 
 NSArray<RTC_OBJC_TYPE(RTCMetricsSampleInfo) *> *RTCGetAndResetMetrics(void) {
-  std::map<std::string, std::unique_ptr<webrtc::metrics::SampleInfo>, rtc::AbslStringViewCmp>
+  std::map<std::string,
+           std::unique_ptr<webrtc::metrics::SampleInfo>,
+           rtc::AbslStringViewCmp>
       histograms;
   webrtc::metrics::GetAndReset(&histograms);
 
@@ -27,7 +29,8 @@ NSArray<RTC_OBJC_TYPE(RTCMetricsSampleInfo) *> *RTCGetAndResetMetrics(void) {
       [NSMutableArray arrayWithCapacity:histograms.size()];
   for (auto const &histogram : histograms) {
     RTC_OBJC_TYPE(RTCMetricsSampleInfo) *metric =
-        [[RTC_OBJC_TYPE(RTCMetricsSampleInfo) alloc] initWithNativeSampleInfo:*histogram.second];
+        [[RTC_OBJC_TYPE(RTCMetricsSampleInfo) alloc]
+            initWithNativeSampleInfo:*histogram.second];
     [metrics addObject:metric];
   }
   return metrics;

sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h

@@ -18,7 +18,8 @@ NS_ASSUME_NONNULL_BEGIN
 ()
 
     /** Initialize an RTCMetricsSampleInfo object from native SampleInfo. */
-    - (instancetype)initWithNativeSampleInfo : (const webrtc::metrics::SampleInfo &)info;
+    - (instancetype)initWithNativeSampleInfo
+    : (const webrtc::metrics::SampleInfo &)info;
 
 @end
 

sdk/objc/api/peerconnection/RTCPeerConnection+DataChannel.mm

@@ -23,11 +23,13 @@
   std::string labelString = [NSString stdStringForString:label];
   const webrtc::DataChannelInit nativeInit =
       configuration.nativeDataChannelInit;
-  auto result = self.nativePeerConnection->CreateDataChannelOrError(labelString, &nativeInit);
+  auto result = self.nativePeerConnection->CreateDataChannelOrError(
+      labelString, &nativeInit);
   if (!result.ok()) {
     return nil;
   }
-  return [[RTC_OBJC_TYPE(RTCDataChannel) alloc] initWithFactory:self.factory
+  return [[RTC_OBJC_TYPE(RTCDataChannel) alloc]
+        initWithFactory:self.factory
       nativeDataChannel:result.MoveValue()];
 }
 

sdk/objc/api/peerconnection/RTCPeerConnection+Private.h

@@ -22,29 +22,36 @@ namespace webrtc {
  */
 class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
  public:
-  PeerConnectionDelegateAdapter(RTC_OBJC_TYPE(RTCPeerConnection) * peerConnection);
+  PeerConnectionDelegateAdapter(RTC_OBJC_TYPE(RTCPeerConnection) *
+                                peerConnection);
   ~PeerConnectionDelegateAdapter() override;
 
-  void OnSignalingChange(PeerConnectionInterface::SignalingState new_state) override;
+  void OnSignalingChange(
+      PeerConnectionInterface::SignalingState new_state) override;
 
   void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) override;
 
   void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) override;
 
-  void OnTrack(rtc::scoped_refptr<RtpTransceiverInterface> transceiver) override;
+  void OnTrack(
+      rtc::scoped_refptr<RtpTransceiverInterface> transceiver) override;
 
-  void OnDataChannel(rtc::scoped_refptr<DataChannelInterface> data_channel) override;
+  void OnDataChannel(
+      rtc::scoped_refptr<DataChannelInterface> data_channel) override;
 
   void OnRenegotiationNeeded() override;
 
-  void OnIceConnectionChange(PeerConnectionInterface::IceConnectionState new_state) override;
+  void OnIceConnectionChange(
+      PeerConnectionInterface::IceConnectionState new_state) override;
 
   void OnStandardizedIceConnectionChange(
       PeerConnectionInterface::IceConnectionState new_state) override;
 
-  void OnConnectionChange(PeerConnectionInterface::PeerConnectionState new_state) override;
+  void OnConnectionChange(
+      PeerConnectionInterface::PeerConnectionState new_state) override;
 
-  void OnIceGatheringChange(PeerConnectionInterface::IceGatheringState new_state) override;
+  void OnIceGatheringChange(
+      PeerConnectionInterface::IceGatheringState new_state) override;
 
   void OnIceCandidate(const IceCandidateInterface *candidate) override;
 
@@ -54,14 +61,18 @@ class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
                            int error_code,
                            const std::string &error_text) override;
 
-  void OnIceCandidatesRemoved(const std::vector<cricket::Candidate> &candidates) override;
+  void OnIceCandidatesRemoved(
+      const std::vector<cricket::Candidate> &candidates) override;
 
-  void OnIceSelectedCandidatePairChanged(const cricket::CandidatePairChangeEvent &event) override;
+  void OnIceSelectedCandidatePairChanged(
+      const cricket::CandidatePairChangeEvent &event) override;
 
   void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver,
-                  const std::vector<rtc::scoped_refptr<MediaStreamInterface>> &streams) override;
+                  const std::vector<rtc::scoped_refptr<MediaStreamInterface>>
+                      &streams) override;
 
-  void OnRemoveTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver) override;
+  void OnRemoveTrack(
+      rtc::scoped_refptr<RtpReceiverInterface> receiver) override;
 
  private:
   __weak RTC_OBJC_TYPE(RTCPeerConnection) * peer_connection_;
@@ -79,7 +90,8 @@ class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
     factory;
 
 /** The native PeerConnectionInterface created during construction. */
-@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::PeerConnectionInterface>
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::PeerConnectionInterface>
         nativePeerConnection;
 
 /** Initialize an RTCPeerConnection with a configuration, constraints, and
@@ -89,8 +101,10 @@ class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
         initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
           configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
             constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
-    certificateVerifier:(nullable id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)certificateVerifier
+    certificateVerifier:(nullable id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)
-               delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
+                            certificateVerifier
+               delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)
+                            delegate;
 
 /** Initialize an RTCPeerConnection with a configuration, constraints,
  *  delegate and PeerConnectionDependencies.
@@ -99,9 +113,10 @@ class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
     initWithDependencies:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
            configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
              constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
-            dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies
+            dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)
-                delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate
+                             dependencies
-    NS_DESIGNATED_INITIALIZER;
+                delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)
+                             delegate NS_DESIGNATED_INITIALIZER;
 
 + (webrtc::PeerConnectionInterface::SignalingState)nativeSignalingStateForState:
     (RTCSignalingState)state;
@@ -111,11 +126,11 @@ class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
 
 + (NSString *)stringForSignalingState:(RTCSignalingState)state;
 
-+ (webrtc::PeerConnectionInterface::IceConnectionState)nativeIceConnectionStateForState:
++ (webrtc::PeerConnectionInterface::IceConnectionState)
-    (RTCIceConnectionState)state;
+    nativeIceConnectionStateForState:(RTCIceConnectionState)state;
 
-+ (webrtc::PeerConnectionInterface::PeerConnectionState)nativeConnectionStateForState:
++ (webrtc::PeerConnectionInterface::PeerConnectionState)
-    (RTCPeerConnectionState)state;
+    nativeConnectionStateForState:(RTCPeerConnectionState)state;
 
 + (RTCIceConnectionState)iceConnectionStateForNativeState:
     (webrtc::PeerConnectionInterface::IceConnectionState)nativeState;
@@ -127,16 +142,16 @@ class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
 
 + (NSString *)stringForConnectionState:(RTCPeerConnectionState)state;
 
-+ (webrtc::PeerConnectionInterface::IceGatheringState)nativeIceGatheringStateForState:
++ (webrtc::PeerConnectionInterface::IceGatheringState)
-    (RTCIceGatheringState)state;
+    nativeIceGatheringStateForState:(RTCIceGatheringState)state;
 
 + (RTCIceGatheringState)iceGatheringStateForNativeState:
     (webrtc::PeerConnectionInterface::IceGatheringState)nativeState;
 
 + (NSString *)stringForIceGatheringState:(RTCIceGatheringState)state;
 
-+ (webrtc::PeerConnectionInterface::StatsOutputLevel)nativeStatsOutputLevelForLevel:
++ (webrtc::PeerConnectionInterface::StatsOutputLevel)
-    (RTCStatsOutputLevel)level;
+    nativeStatsOutputLevelForLevel:(RTCStatsOutputLevel)level;
 
 @end
 

sdk/objc/api/peerconnection/RTCPeerConnection+Stats.mm

@@ -23,10 +23,12 @@ namespace webrtc {
 
 class StatsCollectorCallbackAdapter : public RTCStatsCollectorCallback {
  public:
-  StatsCollectorCallbackAdapter(RTCStatisticsCompletionHandler completion_handler)
+  StatsCollectorCallbackAdapter(
+      RTCStatisticsCompletionHandler completion_handler)
       : completion_handler_(completion_handler) {}
 
-  void OnStatsDelivered(const rtc::scoped_refptr<const RTCStatsReport> &report) override {
+  void OnStatsDelivered(
+      const rtc::scoped_refptr<const RTCStatsReport> &report) override {
     RTC_DCHECK(completion_handler_);
     RTC_OBJC_TYPE(RTCStatisticsReport) *statisticsReport =
         [[RTC_OBJC_TYPE(RTCStatisticsReport) alloc] initWithReport:*report];
@@ -40,19 +42,20 @@ class StatsCollectorCallbackAdapter : public RTCStatsCollectorCallback {
 
 class StatsObserverAdapter : public StatsObserver {
  public:
-  StatsObserverAdapter(
+  StatsObserverAdapter(void (^completionHandler)(
-      void (^completionHandler)(NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats)) {
+      NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats)) {
     completion_handler_ = completionHandler;
   }
 
   ~StatsObserverAdapter() override { completion_handler_ = nil; }
 
-  void OnComplete(const StatsReports& reports) override {
+  void OnComplete(const StatsReports &reports) override {
     RTC_DCHECK(completion_handler_);
     NSMutableArray *stats = [NSMutableArray arrayWithCapacity:reports.size()];
-    for (const auto* report : reports) {
+    for (const auto *report : reports) {
       RTC_OBJC_TYPE(RTCLegacyStatsReport) *statsReport =
-          [[RTC_OBJC_TYPE(RTCLegacyStatsReport) alloc] initWithNativeReport:*report];
+          [[RTC_OBJC_TYPE(RTCLegacyStatsReport) alloc]
+              initWithNativeReport:*report];
       [stats addObject:statsReport];
     }
     completion_handler_(stats);
@@ -60,37 +63,45 @@ class StatsObserverAdapter : public StatsObserver {
   }
 
  private:
-  void (^completion_handler_)(NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats);
+  void (^completion_handler_)(
+      NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats);
 };
 }  // namespace webrtc
 
 @implementation RTC_OBJC_TYPE (RTCPeerConnection)
 (Stats)
 
-    - (void)statisticsForSender : (RTC_OBJC_TYPE(RTCRtpSender) *)sender completionHandler
+    - (void)statisticsForSender
+    : (RTC_OBJC_TYPE(RTCRtpSender) *)sender completionHandler
     : (RTCStatisticsCompletionHandler)completionHandler {
   rtc::scoped_refptr<webrtc::StatsCollectorCallbackAdapter> collector =
-      rtc::make_ref_counted<webrtc::StatsCollectorCallbackAdapter>(completionHandler);
+      rtc::make_ref_counted<webrtc::StatsCollectorCallbackAdapter>(
+          completionHandler);
   self.nativePeerConnection->GetStats(sender.nativeRtpSender, collector);
 }
 
 - (void)statisticsForReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)receiver
-            completionHandler:(RTCStatisticsCompletionHandler)completionHandler {
+            completionHandler:
+                (RTCStatisticsCompletionHandler)completionHandler {
   rtc::scoped_refptr<webrtc::StatsCollectorCallbackAdapter> collector =
-      rtc::make_ref_counted<webrtc::StatsCollectorCallbackAdapter>(completionHandler);
+      rtc::make_ref_counted<webrtc::StatsCollectorCallbackAdapter>(
+          completionHandler);
   self.nativePeerConnection->GetStats(receiver.nativeRtpReceiver, collector);
 }
 
-- (void)statisticsWithCompletionHandler:(RTCStatisticsCompletionHandler)completionHandler {
+- (void)statisticsWithCompletionHandler:
+    (RTCStatisticsCompletionHandler)completionHandler {
   rtc::scoped_refptr<webrtc::StatsCollectorCallbackAdapter> collector =
-      rtc::make_ref_counted<webrtc::StatsCollectorCallbackAdapter>(completionHandler);
+      rtc::make_ref_counted<webrtc::StatsCollectorCallbackAdapter>(
+          completionHandler);
   self.nativePeerConnection->GetStats(collector.get());
 }
 
 - (void)statsForTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)mediaStreamTrack
      statsOutputLevel:(RTCStatsOutputLevel)statsOutputLevel
     completionHandler:
-        (void (^)(NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats))completionHandler {
+        (void (^)(NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats))
+            completionHandler {
   rtc::scoped_refptr<webrtc::StatsObserverAdapter> observer =
       rtc::make_ref_counted<webrtc::StatsObserverAdapter>(completionHandler);
   webrtc::PeerConnectionInterface::StatsOutputLevel nativeOutputLevel =

sdk/objc/api/peerconnection/RTCPeerConnection.h

@@ -83,9 +83,11 @@ typedef NS_ENUM(NSInteger, RTCStatsOutputLevel) {
 };
 
 typedef void (^RTCCreateSessionDescriptionCompletionHandler)(
-    RTC_OBJC_TYPE(RTCSessionDescription) *_Nullable sdp, NSError *_Nullable error);
+    RTC_OBJC_TYPE(RTCSessionDescription) *_Nullable sdp,
+    NSError *_Nullable error);
 
-typedef void (^RTCSetSessionDescriptionCompletionHandler)(NSError *_Nullable error);
+typedef void (^RTCSetSessionDescriptionCompletionHandler)(
+    NSError *_Nullable error);
 
 @class RTC_OBJC_TYPE(RTCPeerConnection);
 
@@ -109,7 +111,8 @@ RTC_OBJC_EXPORT
        didRemoveStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
 
 /** Called when negotiation is needed, for example ICE has restarted. */
-- (void)peerConnectionShouldNegotiate:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection;
+- (void)peerConnectionShouldNegotiate:
+    (RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection;
 
 /** Called any time the IceConnectionState changes. */
 - (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
@@ -125,7 +128,8 @@ RTC_OBJC_EXPORT
 
 /** Called when a group of local Ice candidates have been removed. */
 - (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
-    didRemoveIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
+    didRemoveIceCandidates:
+        (NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
 
 /** New data channel has been opened. */
 - (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
@@ -146,7 +150,8 @@ RTC_OBJC_EXPORT
     didChangeConnectionState:(RTCPeerConnectionState)newState;
 
 - (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
-    didStartReceivingOnTransceiver:(RTC_OBJC_TYPE(RTCRtpTransceiver) *)transceiver;
+    didStartReceivingOnTransceiver:
+        (RTC_OBJC_TYPE(RTCRtpTransceiver) *)transceiver;
 
 /** Called when a receiver and its track are created. */
 - (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
@@ -166,7 +171,8 @@ RTC_OBJC_EXPORT
 
 /** Called when gathering of an ICE candidate failed. */
 - (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
-    didFailToGatherIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidateErrorEvent) *)event;
+    didFailToGatherIceCandidate:
+        (RTC_OBJC_TYPE(RTCIceCandidateErrorEvent) *)event;
 
 @end
 
@@ -180,14 +186,18 @@ RTC_OBJC_EXPORT
 /** This property is not available with RTCSdpSemanticsUnifiedPlan. Please use
  *  `senders` instead.
  */
-@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCMediaStream) *> *localStreams;
+@property(nonatomic, readonly)
-@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) * localDescription;
+    NSArray<RTC_OBJC_TYPE(RTCMediaStream) *> *localStreams;
-@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) * remoteDescription;
+@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) *
+    localDescription;
+@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) *
+    remoteDescription;
 @property(nonatomic, readonly) RTCSignalingState signalingState;
 @property(nonatomic, readonly) RTCIceConnectionState iceConnectionState;
 @property(nonatomic, readonly) RTCPeerConnectionState connectionState;
 @property(nonatomic, readonly) RTCIceGatheringState iceGatheringState;
-@property(nonatomic, readonly, copy) RTC_OBJC_TYPE(RTCConfiguration) * configuration;
+@property(nonatomic, readonly, copy) RTC_OBJC_TYPE(RTCConfiguration) *
+    configuration;
 
 /** Gets all RTCRtpSenders associated with this peer connection.
  *  Note: reading this property returns different instances of RTCRtpSender.
@@ -199,7 +209,8 @@ RTC_OBJC_EXPORT
  *  Note: reading this property returns different instances of RTCRtpReceiver.
  *  Use isEqual: instead of == to compare RTCRtpReceiver instances.
  */
-@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpReceiver) *> *receivers;
+@property(nonatomic, readonly)
+    NSArray<RTC_OBJC_TYPE(RTCRtpReceiver) *> *receivers;
 
 /** Gets all RTCRtpTransceivers associated with this peer connection.
  *  Note: reading this property returns different instances of
@@ -207,7 +218,8 @@ RTC_OBJC_EXPORT
  *  RTCRtpTransceiver instances. This is only available with
  * RTCSdpSemanticsUnifiedPlan specified.
  */
-@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpTransceiver) *> *transceivers;
+@property(nonatomic, readonly)
+    NSArray<RTC_OBJC_TYPE(RTCRtpTransceiver) *> *transceivers;
 
 - (instancetype)init NS_UNAVAILABLE;
 
@@ -224,14 +236,16 @@ RTC_OBJC_EXPORT
 
 /** Provide a remote candidate to the ICE Agent. */
 - (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate
-    DEPRECATED_MSG_ATTRIBUTE("Please use addIceCandidate:completionHandler: instead");
+    DEPRECATED_MSG_ATTRIBUTE(
+        "Please use addIceCandidate:completionHandler: instead");
 
 /** Provide a remote candidate to the ICE Agent. */
 - (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate
       completionHandler:(void (^)(NSError *_Nullable error))completionHandler;
 
 /** Remove a group of remote candidates from the ICE Agent. */
-- (void)removeIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
+- (void)removeIceCandidates:
+    (NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
 
 /** Add a new media stream to be sent on this peer connection.
  *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
@@ -253,7 +267,8 @@ RTC_OBJC_EXPORT
  *  - A sender already exists for the track.
  *  - The peer connection is closed.
  */
-- (nullable RTC_OBJC_TYPE(RTCRtpSender) *)addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
+- (nullable RTC_OBJC_TYPE(RTCRtpSender) *)
+     addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
     streamIds:(NSArray<NSString *> *)streamIds;
 
 /** With PlanB semantics, removes an RTCRtpSender from this peer connection.
@@ -295,28 +310,32 @@ RTC_OBJC_EXPORT
 /** Adds a transceiver with the given kind. Can either be RTCRtpMediaTypeAudio
  *  or RTCRtpMediaTypeVideo.
  */
-- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTCRtpMediaType)mediaType;
+- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:
+    (RTCRtpMediaType)mediaType;
 - (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)
     addTransceiverOfType:(RTCRtpMediaType)mediaType
                     init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init;
 
 /** Tells the PeerConnection that ICE should be restarted. This triggers a need
- * for negotiation and subsequent offerForConstraints:completionHandler call will act as if
+ * for negotiation and subsequent offerForConstraints:completionHandler call
- * RTCOfferAnswerOptions::ice_restart is true.
+ * will act as if RTCOfferAnswerOptions::ice_restart is true.
  */
 - (void)restartIce;
 
 /** Generate an SDP offer. */
 - (void)offerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
-          completionHandler:(RTCCreateSessionDescriptionCompletionHandler)completionHandler;
+          completionHandler:
+              (RTCCreateSessionDescriptionCompletionHandler)completionHandler;
 
 /** Generate an SDP answer. */
 - (void)answerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
-           completionHandler:(RTCCreateSessionDescriptionCompletionHandler)completionHandler;
+           completionHandler:
+               (RTCCreateSessionDescriptionCompletionHandler)completionHandler;
 
 /** Apply the supplied RTCSessionDescription as the local description. */
 - (void)setLocalDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
-          completionHandler:(RTCSetSessionDescriptionCompletionHandler)completionHandler;
+          completionHandler:
+              (RTCSetSessionDescriptionCompletionHandler)completionHandler;
 
 /** Creates an offer or answer (depending on current signaling state) and sets
  * it as the local session description. */
@@ -325,7 +344,8 @@ RTC_OBJC_EXPORT
 
 /** Apply the supplied RTCSessionDescription as the remote description. */
 - (void)setRemoteDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
-           completionHandler:(RTCSetSessionDescriptionCompletionHandler)completionHandler;
+           completionHandler:
+               (RTCSetSessionDescriptionCompletionHandler)completionHandler;
 
 /** Limits the bandwidth allocated for all RTP streams sent by this
  *  PeerConnection. Nil parameters will be unchanged. Setting
@@ -337,7 +357,8 @@ RTC_OBJC_EXPORT
               maxBitrateBps:(nullable NSNumber *)maxBitrateBps;
 
 /** Start or stop recording an Rtc EventLog. */
-- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
+- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath
+                      maxSizeInBytes:(int64_t)maxSizeInBytes;
 - (void)stopRtcEventLog;
 
 @end
@@ -360,25 +381,29 @@ RTC_OBJC_EXPORT
 
     /** Create a new data channel with the given label and configuration. */
     - (nullable RTC_OBJC_TYPE(RTCDataChannel) *)dataChannelForLabel
-    : (NSString *)label configuration : (RTC_OBJC_TYPE(RTCDataChannelConfiguration) *)configuration;
+    : (NSString *)label configuration
+    : (RTC_OBJC_TYPE(RTCDataChannelConfiguration) *)configuration;
 
 @end
 
-typedef void (^RTCStatisticsCompletionHandler)(RTC_OBJC_TYPE(RTCStatisticsReport) *);
+typedef void (^RTCStatisticsCompletionHandler)(
+    RTC_OBJC_TYPE(RTCStatisticsReport) *);
 
 @interface RTC_OBJC_TYPE (RTCPeerConnection)
 (Stats)
 
-    /** Gather stats for the given RTCMediaStreamTrack. If `mediaStreamTrack` is nil
+    /** Gather stats for the given RTCMediaStreamTrack. If `mediaStreamTrack` is
-     *  statistics are gathered for all tracks.
+     * nil statistics are gathered for all tracks.
      */
-    - (void)statsForTrack
+    - (void)statsForTrack : (nullable RTC_OBJC_TYPE(RTCMediaStreamTrack) *)
-    : (nullable RTC_OBJC_TYPE(RTCMediaStreamTrack) *)mediaStreamTrack statsOutputLevel
+                                mediaStreamTrack statsOutputLevel
     : (RTCStatsOutputLevel)statsOutputLevel completionHandler
-    : (nullable void (^)(NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats))completionHandler;
+    : (nullable void (^)(NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats))
+          completionHandler;
 
 /** Gather statistic through the v2 statistics API. */
-- (void)statisticsWithCompletionHandler:(RTCStatisticsCompletionHandler)completionHandler;
+- (void)statisticsWithCompletionHandler:
+    (RTCStatisticsCompletionHandler)completionHandler;
 
 /** Spec-compliant getStats() performing the stats selection algorithm with the
  *  sender.

sdk/objc/api/peerconnection/RTCPeerConnection.mm

@@ -36,15 +36,18 @@
 #include "rtc_base/numerics/safe_conversions.h"
 #include "sdk/objc/native/api/ssl_certificate_verifier.h"
 
-NSString *const kRTCPeerConnectionErrorDomain = @"org.webrtc.RTC_OBJC_TYPE(RTCPeerConnection)";
+NSString *const kRTCPeerConnectionErrorDomain =
+    @"org.webrtc.RTC_OBJC_TYPE(RTCPeerConnection)";
 int const kRTCPeerConnnectionSessionDescriptionError = -1;
 
 namespace {
 
-class SetSessionDescriptionObserver : public webrtc::SetLocalDescriptionObserverInterface,
+class SetSessionDescriptionObserver
+    : public webrtc::SetLocalDescriptionObserverInterface,
       public webrtc::SetRemoteDescriptionObserverInterface {
  public:
-  SetSessionDescriptionObserver(RTCSetSessionDescriptionCompletionHandler completionHandler) {
+  SetSessionDescriptionObserver(
+      RTCSetSessionDescriptionCompletionHandler completionHandler) {
     completion_handler_ = completionHandler;
   }
 
@@ -64,7 +67,8 @@ class SetSessionDescriptionObserver : public webrtc::SetLocalDescriptionObserver
     } else {
       // TODO(hta): Add handling of error.type()
       NSString *str = [NSString stringForStdString:error.message()];
-      NSError *err = [NSError errorWithDomain:kRTCPeerConnectionErrorDomain
+      NSError *err =
+          [NSError errorWithDomain:kRTCPeerConnectionErrorDomain
                               code:kRTCPeerConnnectionSessionDescriptionError
                           userInfo:@{NSLocalizedDescriptionKey : str}];
       completion_handler_(err);
@@ -82,18 +86,22 @@ class CreateSessionDescriptionObserverAdapter
     : public CreateSessionDescriptionObserver {
  public:
   CreateSessionDescriptionObserverAdapter(void (^completionHandler)(
-      RTC_OBJC_TYPE(RTCSessionDescription) * sessionDescription, NSError *error)) {
+      RTC_OBJC_TYPE(RTCSessionDescription) * sessionDescription,
+      NSError *error)) {
     completion_handler_ = completionHandler;
   }
 
-  ~CreateSessionDescriptionObserverAdapter() override { completion_handler_ = nil; }
+  ~CreateSessionDescriptionObserverAdapter() override {
+    completion_handler_ = nil;
+  }
 
   void OnSuccess(SessionDescriptionInterface *desc) override {
     RTC_DCHECK(completion_handler_);
     std::unique_ptr<webrtc::SessionDescriptionInterface> description =
         std::unique_ptr<webrtc::SessionDescriptionInterface>(desc);
     RTC_OBJC_TYPE(RTCSessionDescription) *session =
-        [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithNativeDescription:description.get()];
+        [[RTC_OBJC_TYPE(RTCSessionDescription) alloc]
+            initWithNativeDescription:description.get()];
     completion_handler_(session, nil);
     completion_handler_ = nil;
   }
@@ -102,21 +110,22 @@ class CreateSessionDescriptionObserverAdapter
     RTC_DCHECK(completion_handler_);
     // TODO(hta): Add handling of error.type()
     NSString *str = [NSString stringForStdString:error.message()];
-    NSError* err =
+    NSError *err =
         [NSError errorWithDomain:kRTCPeerConnectionErrorDomain
                             code:kRTCPeerConnnectionSessionDescriptionError
-                        userInfo:@{ NSLocalizedDescriptionKey : str }];
+                        userInfo:@{NSLocalizedDescriptionKey : str}];
     completion_handler_(nil, err);
     completion_handler_ = nil;
   }
 
  private:
-  void (^completion_handler_)(RTC_OBJC_TYPE(RTCSessionDescription) * sessionDescription,
+  void (^completion_handler_)(RTC_OBJC_TYPE(RTCSessionDescription) *
+                                  sessionDescription,
                               NSError *error);
 };
 
-PeerConnectionDelegateAdapter::PeerConnectionDelegateAdapter(RTC_OBJC_TYPE(RTCPeerConnection) *
+PeerConnectionDelegateAdapter::PeerConnectionDelegateAdapter(
-                                                             peerConnection) {
+    RTC_OBJC_TYPE(RTCPeerConnection) * peerConnection) {
   peer_connection_ = peerConnection;
 }
 
@@ -126,8 +135,8 @@ PeerConnectionDelegateAdapter::~PeerConnectionDelegateAdapter() {
 
 void PeerConnectionDelegateAdapter::OnSignalingChange(
     PeerConnectionInterface::SignalingState new_state) {
-  RTCSignalingState state =
+  RTCSignalingState state = [[RTC_OBJC_TYPE(RTCPeerConnection) class]
-      [[RTC_OBJC_TYPE(RTCPeerConnection) class] signalingStateForNativeState:new_state];
+      signalingStateForNativeState:new_state];
   RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
   [peer_connection.delegate peerConnection:peer_connection
                    didChangeSignalingState:state];
@@ -136,9 +145,8 @@ void PeerConnectionDelegateAdapter::OnSignalingChange(
 void PeerConnectionDelegateAdapter::OnAddStream(
     rtc::scoped_refptr<MediaStreamInterface> stream) {
   RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
-  RTC_OBJC_TYPE(RTCMediaStream) *mediaStream =
+  RTC_OBJC_TYPE(RTCMediaStream) *mediaStream = [[RTC_OBJC_TYPE(RTCMediaStream)
-      [[RTC_OBJC_TYPE(RTCMediaStream) alloc] initWithFactory:peer_connection.factory
+      alloc] initWithFactory:peer_connection.factory nativeMediaStream:stream];
-                                           nativeMediaStream:stream];
   [peer_connection.delegate peerConnection:peer_connection
                               didAddStream:mediaStream];
 }
@@ -146,9 +154,8 @@ void PeerConnectionDelegateAdapter::OnAddStream(
 void PeerConnectionDelegateAdapter::OnRemoveStream(
     rtc::scoped_refptr<MediaStreamInterface> stream) {
   RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
-  RTC_OBJC_TYPE(RTCMediaStream) *mediaStream =
+  RTC_OBJC_TYPE(RTCMediaStream) *mediaStream = [[RTC_OBJC_TYPE(RTCMediaStream)
-      [[RTC_OBJC_TYPE(RTCMediaStream) alloc] initWithFactory:peer_connection.factory
+      alloc] initWithFactory:peer_connection.factory nativeMediaStream:stream];
-                                           nativeMediaStream:stream];
 
   [peer_connection.delegate peerConnection:peer_connection
                            didRemoveStream:mediaStream];
@@ -158,10 +165,12 @@ void PeerConnectionDelegateAdapter::OnTrack(
     rtc::scoped_refptr<RtpTransceiverInterface> nativeTransceiver) {
   RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
   RTC_OBJC_TYPE(RTCRtpTransceiver) *transceiver =
-      [[RTC_OBJC_TYPE(RTCRtpTransceiver) alloc] initWithFactory:peer_connection.factory
+      [[RTC_OBJC_TYPE(RTCRtpTransceiver) alloc]
+               initWithFactory:peer_connection.factory
           nativeRtpTransceiver:nativeTransceiver];
   if ([peer_connection.delegate
-          respondsToSelector:@selector(peerConnection:didStartReceivingOnTransceiver:)]) {
+          respondsToSelector:@selector(peerConnection:
+                                 didStartReceivingOnTransceiver:)]) {
     [peer_connection.delegate peerConnection:peer_connection
               didStartReceivingOnTransceiver:transceiver];
   }
@@ -170,8 +179,8 @@ void PeerConnectionDelegateAdapter::OnTrack(
 void PeerConnectionDelegateAdapter::OnDataChannel(
     rtc::scoped_refptr<DataChannelInterface> data_channel) {
   RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
-  RTC_OBJC_TYPE(RTCDataChannel) *dataChannel =
+  RTC_OBJC_TYPE(RTCDataChannel) *dataChannel = [[RTC_OBJC_TYPE(RTCDataChannel)
-      [[RTC_OBJC_TYPE(RTCDataChannel) alloc] initWithFactory:peer_connection.factory
+      alloc] initWithFactory:peer_connection.factory
            nativeDataChannel:data_channel];
   [peer_connection.delegate peerConnection:peer_connection
                         didOpenDataChannel:dataChannel];
@@ -184,17 +193,19 @@ void PeerConnectionDelegateAdapter::OnRenegotiationNeeded() {
 
 void PeerConnectionDelegateAdapter::OnIceConnectionChange(
     PeerConnectionInterface::IceConnectionState new_state) {
-  RTCIceConnectionState state =
+  RTCIceConnectionState state = [RTC_OBJC_TYPE(RTCPeerConnection)
-      [RTC_OBJC_TYPE(RTCPeerConnection) iceConnectionStateForNativeState:new_state];
+      iceConnectionStateForNativeState:new_state];
-  [peer_connection_.delegate peerConnection:peer_connection_ didChangeIceConnectionState:state];
+  [peer_connection_.delegate peerConnection:peer_connection_
+                didChangeIceConnectionState:state];
 }
 
 void PeerConnectionDelegateAdapter::OnStandardizedIceConnectionChange(
     PeerConnectionInterface::IceConnectionState new_state) {
   if ([peer_connection_.delegate
-          respondsToSelector:@selector(peerConnection:didChangeStandardizedIceConnectionState:)]) {
+          respondsToSelector:@selector(peerConnection:
-    RTCIceConnectionState state =
+                                 didChangeStandardizedIceConnectionState:)]) {
-        [RTC_OBJC_TYPE(RTCPeerConnection) iceConnectionStateForNativeState:new_state];
+    RTCIceConnectionState state = [RTC_OBJC_TYPE(RTCPeerConnection)
+        iceConnectionStateForNativeState:new_state];
     [peer_connection_.delegate peerConnection:peer_connection_
         didChangeStandardizedIceConnectionState:state];
   }
@@ -202,18 +213,19 @@ void PeerConnectionDelegateAdapter::OnStandardizedIceConnectionChange(
 
 void PeerConnectionDelegateAdapter::OnConnectionChange(
     PeerConnectionInterface::PeerConnectionState new_state) {
-  if ([peer_connection_.delegate
+  if ([peer_connection_.delegate respondsToSelector:@selector
-          respondsToSelector:@selector(peerConnection:didChangeConnectionState:)]) {
+                                 (peerConnection:didChangeConnectionState:)]) {
-    RTCPeerConnectionState state =
+    RTCPeerConnectionState state = [RTC_OBJC_TYPE(RTCPeerConnection)
-        [RTC_OBJC_TYPE(RTCPeerConnection) connectionStateForNativeState:new_state];
+        connectionStateForNativeState:new_state];
-    [peer_connection_.delegate peerConnection:peer_connection_ didChangeConnectionState:state];
+    [peer_connection_.delegate peerConnection:peer_connection_
+                     didChangeConnectionState:state];
   }
 }
 
 void PeerConnectionDelegateAdapter::OnIceGatheringChange(
     PeerConnectionInterface::IceGatheringState new_state) {
-  RTCIceGatheringState state =
+  RTCIceGatheringState state = [[RTC_OBJC_TYPE(RTCPeerConnection) class]
-      [[RTC_OBJC_TYPE(RTCPeerConnection) class] iceGatheringStateForNativeState:new_state];
+      iceGatheringStateForNativeState:new_state];
   RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
   [peer_connection.delegate peerConnection:peer_connection
                 didChangeIceGatheringState:state];
@@ -222,38 +234,45 @@ void PeerConnectionDelegateAdapter::OnIceGatheringChange(
 void PeerConnectionDelegateAdapter::OnIceCandidate(
     const IceCandidateInterface *candidate) {
   RTC_OBJC_TYPE(RTCIceCandidate) *iceCandidate =
-      [[RTC_OBJC_TYPE(RTCIceCandidate) alloc] initWithNativeCandidate:candidate];
+      [[RTC_OBJC_TYPE(RTCIceCandidate) alloc]
+          initWithNativeCandidate:candidate];
   RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
   [peer_connection.delegate peerConnection:peer_connection
                    didGenerateIceCandidate:iceCandidate];
 }
 
-void PeerConnectionDelegateAdapter::OnIceCandidateError(const std::string &address,
+void PeerConnectionDelegateAdapter::OnIceCandidateError(
+    const std::string &address,
     int port,
     const std::string &url,
     int error_code,
     const std::string &error_text) {
   RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
   RTC_OBJC_TYPE(RTCIceCandidateErrorEvent) *event =
-      [[RTC_OBJC_TYPE(RTCIceCandidateErrorEvent) alloc] initWithAddress:address
+      [[RTC_OBJC_TYPE(RTCIceCandidateErrorEvent) alloc]
+          initWithAddress:address
                      port:port
                       url:url
                 errorCode:error_code
                 errorText:error_text];
-  if ([peer_connection.delegate respondsToSelector:@selector(peerConnection:
+  if ([peer_connection.delegate
+          respondsToSelector:@selector(peerConnection:
                                  didFailToGatherIceCandidate:)]) {
-    [peer_connection.delegate peerConnection:peer_connection didFailToGatherIceCandidate:event];
+    [peer_connection.delegate peerConnection:peer_connection
+                 didFailToGatherIceCandidate:event];
   }
 }
 
 void PeerConnectionDelegateAdapter::OnIceCandidatesRemoved(
-    const std::vector<cricket::Candidate>& candidates) {
+    const std::vector<cricket::Candidate> &candidates) {
-  NSMutableArray* ice_candidates =
+  NSMutableArray *ice_candidates =
       [NSMutableArray arrayWithCapacity:candidates.size()];
-  for (const auto& candidate : candidates) {
+  for (const auto &candidate : candidates) {
-    JsepIceCandidate candidate_wrapper(candidate.transport_name(), -1, candidate);
+    JsepIceCandidate candidate_wrapper(
+        candidate.transport_name(), -1, candidate);
     RTC_OBJC_TYPE(RTCIceCandidate) *ice_candidate =
-        [[RTC_OBJC_TYPE(RTCIceCandidate) alloc] initWithNativeCandidate:&candidate_wrapper];
+        [[RTC_OBJC_TYPE(RTCIceCandidate) alloc]
+            initWithNativeCandidate:&candidate_wrapper];
     [ice_candidates addObject:ice_candidate];
   }
   RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
@@ -265,18 +284,26 @@ void PeerConnectionDelegateAdapter::OnIceSelectedCandidatePairChanged(
     const cricket::CandidatePairChangeEvent &event) {
   const auto &selected_pair = event.selected_candidate_pair;
   JsepIceCandidate local_candidate_wrapper(
-      selected_pair.local_candidate().transport_name(), -1, selected_pair.local_candidate());
+      selected_pair.local_candidate().transport_name(),
+      -1,
+      selected_pair.local_candidate());
   RTC_OBJC_TYPE(RTCIceCandidate) *local_candidate =
-      [[RTC_OBJC_TYPE(RTCIceCandidate) alloc] initWithNativeCandidate:&local_candidate_wrapper];
+      [[RTC_OBJC_TYPE(RTCIceCandidate) alloc]
+          initWithNativeCandidate:&local_candidate_wrapper];
   JsepIceCandidate remote_candidate_wrapper(
-      selected_pair.remote_candidate().transport_name(), -1, selected_pair.remote_candidate());
+      selected_pair.remote_candidate().transport_name(),
+      -1,
+      selected_pair.remote_candidate());
   RTC_OBJC_TYPE(RTCIceCandidate) *remote_candidate =
-      [[RTC_OBJC_TYPE(RTCIceCandidate) alloc] initWithNativeCandidate:&remote_candidate_wrapper];
+      [[RTC_OBJC_TYPE(RTCIceCandidate) alloc]
+          initWithNativeCandidate:&remote_candidate_wrapper];
   RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
   NSString *nsstr_reason = [NSString stringForStdString:event.reason];
   if ([peer_connection.delegate
           respondsToSelector:@selector
-          (peerConnection:didChangeLocalCandidate:remoteCandidate:lastReceivedMs:changeReason:)]) {
+          (peerConnection:
+              didChangeLocalCandidate:remoteCandidate:lastReceivedMs
+                                     :changeReason:)]) {
     [peer_connection.delegate peerConnection:peer_connection
                      didChangeLocalCandidate:local_candidate
                              remoteCandidate:remote_candidate
@@ -289,17 +316,19 @@ void PeerConnectionDelegateAdapter::OnAddTrack(
     rtc::scoped_refptr<RtpReceiverInterface> receiver,
     const std::vector<rtc::scoped_refptr<MediaStreamInterface>> &streams) {
   RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
-  if ([peer_connection.delegate respondsToSelector:@selector(peerConnection:
+  if ([peer_connection.delegate respondsToSelector:@selector
-                                                             didAddReceiver:streams:)]) {
+                                (peerConnection:didAddReceiver:streams:)]) {
-    NSMutableArray *mediaStreams = [NSMutableArray arrayWithCapacity:streams.size()];
+    NSMutableArray *mediaStreams =
+        [NSMutableArray arrayWithCapacity:streams.size()];
     for (const auto &nativeStream : streams) {
       RTC_OBJC_TYPE(RTCMediaStream) *mediaStream =
-          [[RTC_OBJC_TYPE(RTCMediaStream) alloc] initWithFactory:peer_connection.factory
+          [[RTC_OBJC_TYPE(RTCMediaStream) alloc]
+                initWithFactory:peer_connection.factory
               nativeMediaStream:nativeStream];
       [mediaStreams addObject:mediaStream];
     }
-    RTC_OBJC_TYPE(RTCRtpReceiver) *rtpReceiver =
+    RTC_OBJC_TYPE(RTCRtpReceiver) *rtpReceiver = [[RTC_OBJC_TYPE(RTCRtpReceiver)
-        [[RTC_OBJC_TYPE(RTCRtpReceiver) alloc] initWithFactory:peer_connection.factory
+        alloc] initWithFactory:peer_connection.factory
              nativeRtpReceiver:receiver];
 
     [peer_connection.delegate peerConnection:peer_connection
@@ -311,11 +340,13 @@ void PeerConnectionDelegateAdapter::OnAddTrack(
 void PeerConnectionDelegateAdapter::OnRemoveTrack(
     rtc::scoped_refptr<RtpReceiverInterface> receiver) {
   RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
-  if ([peer_connection.delegate respondsToSelector:@selector(peerConnection:didRemoveReceiver:)]) {
+  if ([peer_connection.delegate
-    RTC_OBJC_TYPE(RTCRtpReceiver) *rtpReceiver =
+          respondsToSelector:@selector(peerConnection:didRemoveReceiver:)]) {
-        [[RTC_OBJC_TYPE(RTCRtpReceiver) alloc] initWithFactory:peer_connection.factory
+    RTC_OBJC_TYPE(RTCRtpReceiver) *rtpReceiver = [[RTC_OBJC_TYPE(RTCRtpReceiver)
+        alloc] initWithFactory:peer_connection.factory
              nativeRtpReceiver:receiver];
-    [peer_connection.delegate peerConnection:peer_connection didRemoveReceiver:rtpReceiver];
+    [peer_connection.delegate peerConnection:peer_connection
+                           didRemoveReceiver:rtpReceiver];
   }
 }
 
@@ -333,17 +364,19 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
 @synthesize delegate = _delegate;
 @synthesize factory = _factory;
 
-- (nullable instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (nullable instancetype)
+        initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
           configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
             constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
-                     certificateVerifier:
+    certificateVerifier:(nullable id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)
-                         (nullable id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)certificateVerifier
+                            certificateVerifier
                delegate:(id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
   NSParameterAssert(factory);
   std::unique_ptr<webrtc::PeerConnectionDependencies> dependencies =
       std::make_unique<webrtc::PeerConnectionDependencies>(nullptr);
   if (certificateVerifier != nil) {
-    dependencies->tls_cert_verifier = webrtc::ObjCToNativeCertificateVerifier(certificateVerifier);
+    dependencies->tls_cert_verifier =
+        webrtc::ObjCToNativeCertificateVerifier(certificateVerifier);
   }
   return [self initWithDependencies:factory
                       configuration:configuration
@@ -356,8 +389,10 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
     initWithDependencies:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
            configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
              constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
-            dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies
+            dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)
-                delegate:(id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
+                             dependencies
+                delegate:
+                    (id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
   NSParameterAssert(factory);
   NSParameterAssert(dependencies.get());
   std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration> config(
@@ -373,7 +408,8 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
 
     webrtc::PeerConnectionDependencies deps = std::move(*dependencies);
     deps.observer = _observer.get();
-    auto result = factory.nativeFactory->CreatePeerConnectionOrError(*config, std::move(deps));
+    auto result = factory.nativeFactory->CreatePeerConnectionOrError(
+        *config, std::move(deps));
 
     if (!result.ok()) {
       return nil;
@@ -391,21 +427,25 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
 }
 
 - (RTC_OBJC_TYPE(RTCSessionDescription) *)localDescription {
-  // It's only safe to operate on SessionDescriptionInterface on the signaling thread.
+  // It's only safe to operate on SessionDescriptionInterface on the signaling
+  // thread.
   return _peerConnection->signaling_thread()->BlockingCall([self] {
-    const webrtc::SessionDescriptionInterface *description = _peerConnection->local_description();
+    const webrtc::SessionDescriptionInterface *description =
-    return description ?
+        _peerConnection->local_description();
-        [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithNativeDescription:description] :
+    return description ? [[RTC_OBJC_TYPE(RTCSessionDescription) alloc]
+                             initWithNativeDescription:description] :
                          nil;
   });
 }
 
 - (RTC_OBJC_TYPE(RTCSessionDescription) *)remoteDescription {
-  // It's only safe to operate on SessionDescriptionInterface on the signaling thread.
+  // It's only safe to operate on SessionDescriptionInterface on the signaling
+  // thread.
   return _peerConnection->signaling_thread()->BlockingCall([self] {
-    const webrtc::SessionDescriptionInterface *description = _peerConnection->remote_description();
+    const webrtc::SessionDescriptionInterface *description =
-    return description ?
+        _peerConnection->remote_description();
-        [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithNativeDescription:description] :
+    return description ? [[RTC_OBJC_TYPE(RTCSessionDescription) alloc]
+                             initWithNativeDescription:description] :
                          nil;
   });
 }
@@ -416,17 +456,18 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
 }
 
 - (RTCIceConnectionState)iceConnectionState {
-  return [[self class] iceConnectionStateForNativeState:
+  return [[self class]
-      _peerConnection->ice_connection_state()];
+      iceConnectionStateForNativeState:_peerConnection->ice_connection_state()];
 }
 
 - (RTCPeerConnectionState)connectionState {
-  return [[self class] connectionStateForNativeState:_peerConnection->peer_connection_state()];
+  return [[self class]
+      connectionStateForNativeState:_peerConnection->peer_connection_state()];
 }
 
 - (RTCIceGatheringState)iceGatheringState {
-  return [[self class] iceGatheringStateForNativeState:
+  return [[self class]
-      _peerConnection->ice_gathering_state()];
+      iceGatheringStateForNativeState:_peerConnection->ice_gathering_state()];
 }
 
 - (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration {
@@ -435,15 +476,15 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
   if (!config) {
     return NO;
   }
-  CopyConstraintsIntoRtcConfiguration(_nativeConstraints.get(),
+  CopyConstraintsIntoRtcConfiguration(_nativeConstraints.get(), config.get());
-                                      config.get());
   return _peerConnection->SetConfiguration(*config).ok();
 }
 
 - (RTC_OBJC_TYPE(RTCConfiguration) *)configuration {
   webrtc::PeerConnectionInterface::RTCConfiguration config =
       _peerConnection->GetConfiguration();
-  return [[RTC_OBJC_TYPE(RTCConfiguration) alloc] initWithNativeConfiguration:config];
+  return [[RTC_OBJC_TYPE(RTCConfiguration) alloc]
+      initWithNativeConfiguration:config];
 }
 
 - (void)close {
@@ -464,14 +505,16 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
           completionHandler(nil);
         } else {
           NSString *str = [NSString stringForStdString:error.message()];
-          NSError *err = [NSError errorWithDomain:kRTCPeerConnectionErrorDomain
+          NSError *err =
+              [NSError errorWithDomain:kRTCPeerConnectionErrorDomain
                                   code:static_cast<NSInteger>(error.type())
                               userInfo:@{NSLocalizedDescriptionKey : str}];
           completionHandler(err);
         }
       });
 }
-- (void)removeIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)iceCandidates {
+- (void)removeIceCandidates:
+    (NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)iceCandidates {
   std::vector<cricket::Candidate> candidates;
   for (RTC_OBJC_TYPE(RTCIceCandidate) * iceCandidate in iceCandidates) {
     std::unique_ptr<const webrtc::IceCandidateInterface> candidate(
@@ -500,24 +543,30 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
   [_localStreams removeObject:stream];
 }
 
-- (nullable RTC_OBJC_TYPE(RTCRtpSender) *)addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
+- (nullable RTC_OBJC_TYPE(RTCRtpSender) *)
+     addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
     streamIds:(NSArray<NSString *> *)streamIds {
   std::vector<std::string> nativeStreamIds;
   for (NSString *streamId in streamIds) {
     nativeStreamIds.push_back([streamId UTF8String]);
   }
-  webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>> nativeSenderOrError =
+  webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>>
+      nativeSenderOrError =
           _peerConnection->AddTrack(track.nativeTrack, nativeStreamIds);
   if (!nativeSenderOrError.ok()) {
-    RTCLogError(@"Failed to add track %@: %s", track, nativeSenderOrError.error().message());
+    RTCLogError(@"Failed to add track %@: %s",
+                track,
+                nativeSenderOrError.error().message());
     return nil;
   }
-  return [[RTC_OBJC_TYPE(RTCRtpSender) alloc] initWithFactory:self.factory
+  return [[RTC_OBJC_TYPE(RTCRtpSender) alloc]
+      initWithFactory:self.factory
       nativeRtpSender:nativeSenderOrError.MoveValue()];
 }
 
 - (BOOL)removeTrack:(RTC_OBJC_TYPE(RTCRtpSender) *)sender {
-  bool result = _peerConnection->RemoveTrackOrError(sender.nativeRtpSender).ok();
+  bool result =
+      _peerConnection->RemoveTrackOrError(sender.nativeRtpSender).ok();
   if (!result) {
     RTCLogError(@"Failed to remote track %@", sender);
   }
@@ -527,17 +576,20 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
 - (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverWithTrack:
     (RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track {
   return [self addTransceiverWithTrack:track
-                                  init:[[RTC_OBJC_TYPE(RTCRtpTransceiverInit) alloc] init]];
+                                  init:[[RTC_OBJC_TYPE(RTCRtpTransceiverInit)
+                                           alloc] init]];
 }
 
 - (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)
     addTransceiverWithTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
                        init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init {
-  webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> nativeTransceiverOrError =
+  webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>>
+      nativeTransceiverOrError =
           _peerConnection->AddTransceiver(track.nativeTrack, init.nativeInit);
   if (!nativeTransceiverOrError.ok()) {
-    RTCLogError(
+    RTCLogError(@"Failed to add transceiver %@: %s",
-        @"Failed to add transceiver %@: %s", track, nativeTransceiverOrError.error().message());
+                track,
+                nativeTransceiverOrError.error().message());
     return nil;
   }
   return [[RTC_OBJC_TYPE(RTCRtpTransceiver) alloc]
@@ -545,17 +597,20 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
       nativeRtpTransceiver:nativeTransceiverOrError.MoveValue()];
 }
 
-- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTCRtpMediaType)mediaType {
+- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:
-  return [self addTransceiverOfType:mediaType
+    (RTCRtpMediaType)mediaType {
+  return [self
+      addTransceiverOfType:mediaType
                       init:[[RTC_OBJC_TYPE(RTCRtpTransceiverInit) alloc] init]];
 }
 
 - (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)
     addTransceiverOfType:(RTCRtpMediaType)mediaType
                     init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init {
-  webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> nativeTransceiverOrError =
+  webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>>
-      _peerConnection->AddTransceiver(
+      nativeTransceiverOrError = _peerConnection->AddTransceiver(
-          [RTC_OBJC_TYPE(RTCRtpReceiver) nativeMediaTypeForMediaType:mediaType], init.nativeInit);
+          [RTC_OBJC_TYPE(RTCRtpReceiver) nativeMediaTypeForMediaType:mediaType],
+          init.nativeInit);
   if (!nativeTransceiverOrError.ok()) {
     RTCLogError(@"Failed to add transceiver %@: %s",
                 [RTC_OBJC_TYPE(RTCRtpReceiver) stringForMediaType:mediaType],
@@ -572,29 +627,36 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
 }
 
 - (void)offerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
-          completionHandler:(RTCCreateSessionDescriptionCompletionHandler)completionHandler {
+          completionHandler:
+              (RTCCreateSessionDescriptionCompletionHandler)completionHandler {
   RTC_DCHECK(completionHandler != nil);
   rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter> observer =
-      rtc::make_ref_counted<webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler);
+      rtc::make_ref_counted<webrtc::CreateSessionDescriptionObserverAdapter>(
+          completionHandler);
   webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
-  CopyConstraintsIntoOfferAnswerOptions(constraints.nativeConstraints.get(), &options);
+  CopyConstraintsIntoOfferAnswerOptions(constraints.nativeConstraints.get(),
+                                        &options);
 
   _peerConnection->CreateOffer(observer.get(), options);
 }
 
 - (void)answerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
-           completionHandler:(RTCCreateSessionDescriptionCompletionHandler)completionHandler {
+           completionHandler:
+               (RTCCreateSessionDescriptionCompletionHandler)completionHandler {
   RTC_DCHECK(completionHandler != nil);
   rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter> observer =
-      rtc::make_ref_counted<webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler);
+      rtc::make_ref_counted<webrtc::CreateSessionDescriptionObserverAdapter>(
+          completionHandler);
   webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
-  CopyConstraintsIntoOfferAnswerOptions(constraints.nativeConstraints.get(), &options);
+  CopyConstraintsIntoOfferAnswerOptions(constraints.nativeConstraints.get(),
+                                        &options);
 
   _peerConnection->CreateAnswer(observer.get(), options);
 }
 
 - (void)setLocalDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
-          completionHandler:(RTCSetSessionDescriptionCompletionHandler)completionHandler {
+          completionHandler:
+              (RTCSetSessionDescriptionCompletionHandler)completionHandler {
   RTC_DCHECK(completionHandler != nil);
   rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface> observer =
       rtc::make_ref_counted<::SetSessionDescriptionObserver>(completionHandler);
@@ -610,7 +672,8 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
 }
 
 - (void)setRemoteDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
-           completionHandler:(RTCSetSessionDescriptionCompletionHandler)completionHandler {
+           completionHandler:
+               (RTCSetSessionDescriptionCompletionHandler)completionHandler {
   RTC_DCHECK(completionHandler != nil);
   rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface> observer =
       rtc::make_ref_counted<::SetSessionDescriptionObserver>(completionHandler);
@@ -648,7 +711,8 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
     return NO;
   }
   // TODO(eladalon): It would be better to not allow negative values into PC.
-  const size_t max_size = (maxSizeInBytes < 0) ? webrtc::RtcEventLog::kUnlimitedOutput :
+  const size_t max_size = (maxSizeInBytes < 0) ?
+      webrtc::RtcEventLog::kUnlimitedOutput :
       rtc::saturated_cast<size_t>(maxSizeInBytes);
 
   _hasStartedRtcEventLog = _peerConnection->StartRtcEventLog(
@@ -661,12 +725,14 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
   _hasStartedRtcEventLog = NO;
 }
 
-- (RTC_OBJC_TYPE(RTCRtpSender) *)senderWithKind:(NSString *)kind streamId:(NSString *)streamId {
+- (RTC_OBJC_TYPE(RTCRtpSender) *)senderWithKind:(NSString *)kind
+                                       streamId:(NSString *)streamId {
   std::string nativeKind = [NSString stdStringForString:kind];
   std::string nativeStreamId = [NSString stdStringForString:streamId];
   rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeSender(
       _peerConnection->CreateSender(nativeKind, nativeStreamId));
-  return nativeSender ? [[RTC_OBJC_TYPE(RTCRtpSender) alloc] initWithFactory:self.factory
+  return nativeSender ?
+      [[RTC_OBJC_TYPE(RTCRtpSender) alloc] initWithFactory:self.factory
                                            nativeRtpSender:nativeSender] :
       nil;
 }
@@ -698,12 +764,13 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
 }
 
 - (NSArray<RTC_OBJC_TYPE(RTCRtpTransceiver) *> *)transceivers {
-  std::vector<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> nativeTransceivers(
+  std::vector<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>>
-      _peerConnection->GetTransceivers());
+      nativeTransceivers(_peerConnection->GetTransceivers());
   NSMutableArray *transceivers = [[NSMutableArray alloc] init];
   for (const auto &nativeTransceiver : nativeTransceivers) {
     RTC_OBJC_TYPE(RTCRtpTransceiver) *transceiver =
-        [[RTC_OBJC_TYPE(RTCRtpTransceiver) alloc] initWithFactory:self.factory
+        [[RTC_OBJC_TYPE(RTCRtpTransceiver) alloc]
+                 initWithFactory:self.factory
             nativeRtpTransceiver:nativeTransceiver];
     [transceivers addObject:transceiver];
   }
@@ -765,8 +832,8 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
   }
 }
 
-+ (webrtc::PeerConnectionInterface::PeerConnectionState)nativeConnectionStateForState:
++ (webrtc::PeerConnectionInterface::PeerConnectionState)
-        (RTCPeerConnectionState)state {
+    nativeConnectionStateForState:(RTCPeerConnectionState)state {
   switch (state) {
     case RTCPeerConnectionStateNew:
       return webrtc::PeerConnectionInterface::PeerConnectionState::kNew;
@@ -777,7 +844,8 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
     case RTCPeerConnectionStateFailed:
       return webrtc::PeerConnectionInterface::PeerConnectionState::kFailed;
     case RTCPeerConnectionStateDisconnected:
-      return webrtc::PeerConnectionInterface::PeerConnectionState::kDisconnected;
+      return webrtc::PeerConnectionInterface::PeerConnectionState::
+          kDisconnected;
     case RTCPeerConnectionStateClosed:
       return webrtc::PeerConnectionInterface::PeerConnectionState::kClosed;
   }

sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h

@@ -23,59 +23,79 @@
 NS_ASSUME_NONNULL_BEGIN
 
 /**
- * This class extension exposes methods that work directly with injectable C++ components.
+ * This class extension exposes methods that work directly with injectable C++
+ * components.
  */
 @interface RTC_OBJC_TYPE (RTCPeerConnectionFactory)
 ()
 
-/* Initializer used when WebRTC is compiled with no media support */
+    /* Initializer used when WebRTC is compiled with no media support */
-- (instancetype)initWithNoMedia;
+    - (instancetype)initWithNoMedia;
 
 /* Initialize object with provided dependencies and with media support. */
 - (instancetype)initWithMediaAndDependencies:
     (webrtc::PeerConnectionFactoryDependencies)dependencies;
 
-/* Initialize object with injectable native audio/video encoder/decoder factories */
+/* Initialize object with injectable native audio/video encoder/decoder
-- (instancetype)initWithNativeAudioEncoderFactory:
+ * factories */
+- (instancetype)
+    initWithNativeAudioEncoderFactory:
         (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
             nativeAudioDecoderFactory:
-                            (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
+                (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)
+                    audioDecoderFactory
             nativeVideoEncoderFactory:
-                            (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
+                (std::unique_ptr<webrtc::VideoEncoderFactory>)
+                    videoEncoderFactory
             nativeVideoDecoderFactory:
-                            (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
+                (std::unique_ptr<webrtc::VideoDecoderFactory>)
+                    videoDecoderFactory
                     audioDeviceModule:
                         (nullable webrtc::AudioDeviceModule *)audioDeviceModule
                 audioProcessingModule:
-                                (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule;
+                    (rtc::scoped_refptr<webrtc::AudioProcessing>)
+                        audioProcessingModule;
 
 - (instancetype)
     initWithNativeAudioEncoderFactory:
         (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
             nativeAudioDecoderFactory:
-                (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
+                (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)
+                    audioDecoderFactory
             nativeVideoEncoderFactory:
-                (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
+                (std::unique_ptr<webrtc::VideoEncoderFactory>)
+                    videoEncoderFactory
             nativeVideoDecoderFactory:
-                (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
+                (std::unique_ptr<webrtc::VideoDecoderFactory>)
-                    audioDeviceModule:(nullable webrtc::AudioDeviceModule *)audioDeviceModule
+                    videoDecoderFactory
+                    audioDeviceModule:
+                        (nullable webrtc::AudioDeviceModule *)audioDeviceModule
                 audioProcessingModule:
-                    (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
+                    (rtc::scoped_refptr<webrtc::AudioProcessing>)
-             networkControllerFactory:(std::unique_ptr<webrtc::NetworkControllerFactoryInterface>)
+                        audioProcessingModule
+             networkControllerFactory:
+                 (std::unique_ptr<webrtc::NetworkControllerFactoryInterface>)
                      networkControllerFactory;
 
 - (instancetype)
-    initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+    initWithEncoderFactory:
-            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory;
+        (nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)
+                               decoderFactory;
 
 /** Initialize an RTCPeerConnection with a configuration, constraints, and
  *  dependencies.
  */
 - (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
-    peerConnectionWithDependencies:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+    peerConnectionWithDependencies:
-                       constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+        (RTC_OBJC_TYPE(RTCConfiguration) *)configuration
-                      dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies
+                       constraints:
-                          delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
+                           (RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+                      dependencies:
+                          (std::unique_ptr<webrtc::PeerConnectionDependencies>)
+                              dependencies
+                          delegate:(nullable id<RTC_OBJC_TYPE(
+                                        RTCPeerConnectionDelegate)>)delegate;
 
 @end
 

sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Private.h

@@ -24,8 +24,8 @@ NS_ASSUME_NONNULL_BEGIN
      * RTCPeerConnectionFactory object. This is needed to pass to the underlying
      * C++ APIs.
      */
-    @property(nonatomic,
+    @property(nonatomic, readonly) rtc::scoped_refptr<
-              readonly) rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> nativeFactory;
+        webrtc::PeerConnectionFactoryInterface> nativeFactory;
 
 @property(nonatomic, readonly) rtc::Thread* signalingThread;
 @property(nonatomic, readonly) rtc::Thread* workerThread;

sdk/objc/api/peerconnection/RTCPeerConnectionFactory.h

@@ -41,40 +41,50 @@ RTC_OBJC_EXPORT
 /* Initialize object with default H264 video encoder/decoder factories and default ADM */
 - (instancetype)init;
 
-/* Initialize object with injectable video encoder/decoder factories and default ADM */
+/* Initialize object with injectable video encoder/decoder factories and default
+ * ADM */
 - (instancetype)
-    initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+    initWithEncoderFactory:
-            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory;
+        (nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)
+                               decoderFactory;
 
-/* Initialize object with injectable video encoder/decoder factories and injectable ADM */
+/* Initialize object with injectable video encoder/decoder factories and
+ * injectable ADM */
 - (instancetype)
-    initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+    initWithEncoderFactory:
-            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory
+        (nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
-               audioDevice:(nullable id<RTC_OBJC_TYPE(RTCAudioDevice)>)audioDevice;
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)
+                               decoderFactory
+               audioDevice:
+                   (nullable id<RTC_OBJC_TYPE(RTCAudioDevice)>)audioDevice;
 
 /**
  * Valid kind values are kRTCMediaStreamTrackKindAudio and
  * kRTCMediaStreamTrackKindVideo.
  */
-- (RTC_OBJC_TYPE(RTCRtpCapabilities) *)rtpSenderCapabilitiesForKind:(NSString *)kind;
+- (RTC_OBJC_TYPE(RTCRtpCapabilities) *)rtpSenderCapabilitiesForKind:
+    (NSString *)kind;
 
 /**
  * Valid kind values are kRTCMediaStreamTrackKindAudio and
  * kRTCMediaStreamTrackKindVideo.
  */
-- (RTC_OBJC_TYPE(RTCRtpCapabilities) *)rtpReceiverCapabilitiesForKind:(NSString *)kind;
+- (RTC_OBJC_TYPE(RTCRtpCapabilities) *)rtpReceiverCapabilitiesForKind:
+    (NSString *)kind;
 
 /** Initialize an RTCAudioSource with constraints. */
 - (RTC_OBJC_TYPE(RTCAudioSource) *)audioSourceWithConstraints:
     (nullable RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints;
 
-/** Initialize an RTCAudioTrack with an id. Convenience ctor to use an audio source
+/** Initialize an RTCAudioTrack with an id. Convenience ctor to use an audio
- * with no constraints.
+ * source with no constraints.
  */
 - (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithTrackId:(NSString *)trackId;
 
 /** Initialize an RTCAudioTrack with a source and an id. */
-- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithSource:(RTC_OBJC_TYPE(RTCAudioSource) *)source
+- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithSource:
+                                      (RTC_OBJC_TYPE(RTCAudioSource) *)source
                                                trackId:(NSString *)trackId;
 
 /** Initialize a generic RTCVideoSource. The RTCVideoSource should be
@@ -91,7 +101,8 @@ RTC_OBJC_EXPORT
 - (RTC_OBJC_TYPE(RTCVideoSource) *)videoSourceForScreenCast:(BOOL)forScreenCast;
 
 /** Initialize an RTCVideoTrack with a source and an id. */
-- (RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrackWithSource:(RTC_OBJC_TYPE(RTCVideoSource) *)source
+- (RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrackWithSource:
+                                      (RTC_OBJC_TYPE(RTCVideoSource) *)source
                                                trackId:(NSString *)trackId;
 
 /** Initialize an RTCMediaStream with an id. */
@@ -101,22 +112,32 @@ RTC_OBJC_EXPORT
  *  delegate.
  */
 - (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
-    peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+    peerConnectionWithConfiguration:
-                        constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+        (RTC_OBJC_TYPE(RTCConfiguration) *)configuration
-                           delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
+                        constraints:
+                            (RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+                           delegate:(nullable id<RTC_OBJC_TYPE(
+                                         RTCPeerConnectionDelegate)>)delegate;
 
 - (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
-    peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+    peerConnectionWithConfiguration:
-                        constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+        (RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+                        constraints:
+                            (RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
                 certificateVerifier:
-                    (id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)certificateVerifier
+                    (id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)
-                           delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
+                        certificateVerifier
+                           delegate:(nullable id<RTC_OBJC_TYPE(
+                                         RTCPeerConnectionDelegate)>)delegate;
 
 /** Set the options to be used for subsequently created RTCPeerConnections */
-- (void)setOptions:(nonnull RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions) *)options;
+- (void)setOptions:
+    (nonnull RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions) *)options;
 
-/** Start an AecDump recording. This API call will likely change in the future. */
+/** Start an AecDump recording. This API call will likely change in the future.
-- (BOOL)startAecDumpWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
+ */
+- (BOOL)startAecDumpWithFilePath:(NSString *)filePath
+                  maxSizeInBytes:(int64_t)maxSizeInBytes;
 
 /* Stop an active AecDump recording */
 - (void)stopAecDump;

sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm

@@ -72,8 +72,10 @@
 
 - (instancetype)init {
   webrtc::PeerConnectionFactoryDependencies dependencies;
-  dependencies.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
+  dependencies.audio_encoder_factory =
-  dependencies.audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
+      webrtc::CreateBuiltinAudioEncoderFactory();
+  dependencies.audio_decoder_factory =
+      webrtc::CreateBuiltinAudioDecoderFactory();
   dependencies.video_encoder_factory = webrtc::ObjCToNativeVideoEncoderFactory(
       [[RTC_OBJC_TYPE(RTCVideoEncoderFactoryH264) alloc] init]);
   dependencies.video_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory(
@@ -83,26 +85,37 @@
 }
 
 - (instancetype)
-    initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+    initWithEncoderFactory:
-            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory {
+        (nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
-  return [self initWithEncoderFactory:encoderFactory decoderFactory:decoderFactory audioDevice:nil];
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)
+                               decoderFactory {
+  return [self initWithEncoderFactory:encoderFactory
+                       decoderFactory:decoderFactory
+                          audioDevice:nil];
 }
 
 - (instancetype)
-    initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+    initWithEncoderFactory:
-            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory
+        (nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
-               audioDevice:(nullable id<RTC_OBJC_TYPE(RTCAudioDevice)>)audioDevice {
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)
+                               decoderFactory
+               audioDevice:
+                   (nullable id<RTC_OBJC_TYPE(RTCAudioDevice)>)audioDevice {
 #ifdef HAVE_NO_MEDIA
   return [self initWithNoMedia];
 #else
   webrtc::PeerConnectionFactoryDependencies dependencies;
-  dependencies.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
+  dependencies.audio_encoder_factory =
-  dependencies.audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
+      webrtc::CreateBuiltinAudioEncoderFactory();
+  dependencies.audio_decoder_factory =
+      webrtc::CreateBuiltinAudioDecoderFactory();
   if (encoderFactory) {
-    dependencies.video_encoder_factory = webrtc::ObjCToNativeVideoEncoderFactory(encoderFactory);
+    dependencies.video_encoder_factory =
+        webrtc::ObjCToNativeVideoEncoderFactory(encoderFactory);
   }
   if (decoderFactory) {
-    dependencies.video_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory(decoderFactory);
+    dependencies.video_decoder_factory =
+        webrtc::ObjCToNativeVideoDecoderFactory(decoderFactory);
   }
   if (audioDevice) {
     dependencies.adm = webrtc::CreateAudioDeviceModule(audioDevice);
@@ -113,7 +126,8 @@
 #endif
 }
 
-- (instancetype)initWithNativeDependencies:(webrtc::PeerConnectionFactoryDependencies)dependencies {
+- (instancetype)initWithNativeDependencies:
+    (webrtc::PeerConnectionFactoryDependencies)dependencies {
   self = [super init];
   if (self) {
     _networkThread = rtc::Thread::CreateWithSocketServer();
@@ -131,7 +145,8 @@
     result = _signalingThread->Start();
     RTC_DCHECK(result) << "Failed to start signaling thread.";
 
-    // Set fields that are relevant both to 'no media' and 'with media' scenarios.
+    // Set fields that are relevant both to 'no media' and 'with media'
+    // scenarios.
     dependencies.network_thread = _networkThread.get();
     dependencies.worker_thread = _workerThread.get();
     dependencies.signaling_thread = _signalingThread.get();
@@ -140,30 +155,39 @@
     }
     if (dependencies.network_monitor_factory == nullptr &&
         dependencies.trials->IsEnabled("WebRTC-Network-UseNWPathMonitor")) {
-      dependencies.network_monitor_factory = webrtc::CreateNetworkMonitorFactory();
+      dependencies.network_monitor_factory =
+          webrtc::CreateNetworkMonitorFactory();
     }
 
-    _nativeFactory = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
+    _nativeFactory =
+        webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
     NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
   }
   return self;
 }
 
 - (instancetype)initWithNoMedia {
-  return [self initWithNativeDependencies:webrtc::PeerConnectionFactoryDependencies()];
+  return [self
+      initWithNativeDependencies:webrtc::PeerConnectionFactoryDependencies()];
 }
 
-- (instancetype)initWithNativeAudioEncoderFactory:
+- (instancetype)
+    initWithNativeAudioEncoderFactory:
         (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
             nativeAudioDecoderFactory:
-                            (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
+                (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)
+                    audioDecoderFactory
             nativeVideoEncoderFactory:
-                            (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
+                (std::unique_ptr<webrtc::VideoEncoderFactory>)
+                    videoEncoderFactory
             nativeVideoDecoderFactory:
-                            (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
+                (std::unique_ptr<webrtc::VideoDecoderFactory>)
-                                audioDeviceModule:(webrtc::AudioDeviceModule *)audioDeviceModule
+                    videoDecoderFactory
+                    audioDeviceModule:
+                        (webrtc::AudioDeviceModule *)audioDeviceModule
                 audioProcessingModule:
-                                (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule {
+                    (rtc::scoped_refptr<webrtc::AudioProcessing>)
+                        audioProcessingModule {
   webrtc::PeerConnectionFactoryDependencies dependencies;
   dependencies.audio_encoder_factory = std::move(audioEncoderFactory);
   dependencies.audio_decoder_factory = std::move(audioDecoderFactory);
@@ -171,22 +195,29 @@
   dependencies.video_decoder_factory = std::move(videoDecoderFactory);
   dependencies.adm = std::move(audioDeviceModule);
   if (audioProcessingModule != nullptr) {
-    dependencies.audio_processing_builder = CustomAudioProcessing(std::move(audioProcessingModule));
+    dependencies.audio_processing_builder =
+        CustomAudioProcessing(std::move(audioProcessingModule));
   }
   return [self initWithMediaAndDependencies:std::move(dependencies)];
 }
 
-- (instancetype)initWithNativeAudioEncoderFactory:
+- (instancetype)
+    initWithNativeAudioEncoderFactory:
         (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
             nativeAudioDecoderFactory:
-                            (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
+                (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)
+                    audioDecoderFactory
             nativeVideoEncoderFactory:
-                            (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
+                (std::unique_ptr<webrtc::VideoEncoderFactory>)
+                    videoEncoderFactory
             nativeVideoDecoderFactory:
-                            (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
+                (std::unique_ptr<webrtc::VideoDecoderFactory>)
-                                audioDeviceModule:(webrtc::AudioDeviceModule *)audioDeviceModule
+                    videoDecoderFactory
+                    audioDeviceModule:
+                        (webrtc::AudioDeviceModule *)audioDeviceModule
                 audioProcessingModule:
-                                (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
+                    (rtc::scoped_refptr<webrtc::AudioProcessing>)
+                        audioProcessingModule
              networkControllerFactory:
                  (std::unique_ptr<webrtc::NetworkControllerFactoryInterface>)
                      networkControllerFactory {
@@ -197,7 +228,8 @@
   dependencies.video_encoder_factory = std::move(videoEncoderFactory);
   dependencies.video_decoder_factory = std::move(videoDecoderFactory);
   if (audioProcessingModule != nullptr) {
-    dependencies.audio_processing_builder = CustomAudioProcessing(std::move(audioProcessingModule));
+    dependencies.audio_processing_builder =
+        CustomAudioProcessing(std::move(audioProcessingModule));
   }
   dependencies.network_controller_factory = std::move(networkControllerFactory);
   return [self initWithMediaAndDependencies:std::move(dependencies)];
@@ -218,24 +250,31 @@
   }
 #endif
   if (dependencies.event_log_factory == nullptr) {
-    dependencies.event_log_factory = std::make_unique<webrtc::RtcEventLogFactory>();
+    dependencies.event_log_factory =
+        std::make_unique<webrtc::RtcEventLogFactory>();
   }
   webrtc::EnableMedia(dependencies);
   return [self initWithNativeDependencies:std::move(dependencies)];
 }
 
-- (RTC_OBJC_TYPE(RTCRtpCapabilities) *)rtpSenderCapabilitiesForKind:(NSString *)kind {
+- (RTC_OBJC_TYPE(RTCRtpCapabilities) *)rtpSenderCapabilitiesForKind:
+    (NSString *)kind {
   cricket::MediaType mediaType = [[self class] mediaTypeForKind:kind];
 
-  webrtc::RtpCapabilities rtpCapabilities = _nativeFactory->GetRtpSenderCapabilities(mediaType);
+  webrtc::RtpCapabilities rtpCapabilities =
-  return [[RTC_OBJC_TYPE(RTCRtpCapabilities) alloc] initWithNativeRtpCapabilities:rtpCapabilities];
+      _nativeFactory->GetRtpSenderCapabilities(mediaType);
+  return [[RTC_OBJC_TYPE(RTCRtpCapabilities) alloc]
+      initWithNativeRtpCapabilities:rtpCapabilities];
 }
 
-- (RTC_OBJC_TYPE(RTCRtpCapabilities) *)rtpReceiverCapabilitiesForKind:(NSString *)kind {
+- (RTC_OBJC_TYPE(RTCRtpCapabilities) *)rtpReceiverCapabilitiesForKind:
+    (NSString *)kind {
   cricket::MediaType mediaType = [[self class] mediaTypeForKind:kind];
 
-  webrtc::RtpCapabilities rtpCapabilities = _nativeFactory->GetRtpReceiverCapabilities(mediaType);
+  webrtc::RtpCapabilities rtpCapabilities =
-  return [[RTC_OBJC_TYPE(RTCRtpCapabilities) alloc] initWithNativeRtpCapabilities:rtpCapabilities];
+      _nativeFactory->GetRtpReceiverCapabilities(mediaType);
+  return [[RTC_OBJC_TYPE(RTCRtpCapabilities) alloc]
+      initWithNativeRtpCapabilities:rtpCapabilities];
 }
 
 - (RTC_OBJC_TYPE(RTCAudioSource) *)audioSourceWithConstraints:
@@ -249,46 +288,61 @@
 
   rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
       _nativeFactory->CreateAudioSource(options);
-  return [[RTC_OBJC_TYPE(RTCAudioSource) alloc] initWithFactory:self nativeAudioSource:source];
+  return [[RTC_OBJC_TYPE(RTCAudioSource) alloc] initWithFactory:self
+                                              nativeAudioSource:source];
 }
 
 - (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithTrackId:(NSString *)trackId {
-  RTC_OBJC_TYPE(RTCAudioSource) *audioSource = [self audioSourceWithConstraints:nil];
+  RTC_OBJC_TYPE(RTCAudioSource) *audioSource =
+      [self audioSourceWithConstraints:nil];
   return [self audioTrackWithSource:audioSource trackId:trackId];
 }
 
-- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithSource:(RTC_OBJC_TYPE(RTCAudioSource) *)source
+- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithSource:
+                                      (RTC_OBJC_TYPE(RTCAudioSource) *)source
                                                trackId:(NSString *)trackId {
-  return [[RTC_OBJC_TYPE(RTCAudioTrack) alloc] initWithFactory:self source:source trackId:trackId];
+  return [[RTC_OBJC_TYPE(RTCAudioTrack) alloc] initWithFactory:self
+                                                        source:source
+                                                       trackId:trackId];
 }
 
 - (RTC_OBJC_TYPE(RTCVideoSource) *)videoSource {
-  return [[RTC_OBJC_TYPE(RTCVideoSource) alloc] initWithFactory:self
+  return [[RTC_OBJC_TYPE(RTCVideoSource) alloc]
+      initWithFactory:self
       signalingThread:_signalingThread.get()
          workerThread:_workerThread.get()];
 }
 
-- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSourceForScreenCast:(BOOL)forScreenCast {
+- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSourceForScreenCast:
-  return [[RTC_OBJC_TYPE(RTCVideoSource) alloc] initWithFactory:self
+    (BOOL)forScreenCast {
+  return [[RTC_OBJC_TYPE(RTCVideoSource) alloc]
+      initWithFactory:self
       signalingThread:_signalingThread.get()
          workerThread:_workerThread.get()
          isScreenCast:forScreenCast];
 }
 
-- (RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrackWithSource:(RTC_OBJC_TYPE(RTCVideoSource) *)source
+- (RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrackWithSource:
+                                      (RTC_OBJC_TYPE(RTCVideoSource) *)source
                                                trackId:(NSString *)trackId {
-  return [[RTC_OBJC_TYPE(RTCVideoTrack) alloc] initWithFactory:self source:source trackId:trackId];
+  return [[RTC_OBJC_TYPE(RTCVideoTrack) alloc] initWithFactory:self
+                                                        source:source
+                                                       trackId:trackId];
 }
 
-- (RTC_OBJC_TYPE(RTCMediaStream) *)mediaStreamWithStreamId:(NSString *)streamId {
+- (RTC_OBJC_TYPE(RTCMediaStream) *)mediaStreamWithStreamId:
-  return [[RTC_OBJC_TYPE(RTCMediaStream) alloc] initWithFactory:self streamId:streamId];
+    (NSString *)streamId {
+  return [[RTC_OBJC_TYPE(RTCMediaStream) alloc] initWithFactory:self
+                                                       streamId:streamId];
 }
 
 - (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
-    peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+    peerConnectionWithConfiguration:
-                        constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+        (RTC_OBJC_TYPE(RTCConfiguration) *)configuration
-                           delegate:
+                        constraints:
-                               (nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
+                            (RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+                           delegate:(nullable id<RTC_OBJC_TYPE(
+                                         RTCPeerConnectionDelegate)>)delegate {
   return [[RTC_OBJC_TYPE(RTCPeerConnection) alloc] initWithFactory:self
                                                      configuration:configuration
                                                        constraints:constraints
@@ -297,13 +351,17 @@
 }
 
 - (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
-    peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+    peerConnectionWithConfiguration:
-                        constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+        (RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+                        constraints:
+                            (RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
                 certificateVerifier:
-                    (id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)certificateVerifier
+                    (id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)
-                           delegate:
+                        certificateVerifier
-                               (nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
+                           delegate:(nullable id<RTC_OBJC_TYPE(
-  return [[RTC_OBJC_TYPE(RTCPeerConnection) alloc] initWithFactory:self
+                                         RTCPeerConnectionDelegate)>)delegate {
+  return [[RTC_OBJC_TYPE(RTCPeerConnection) alloc]
+          initWithFactory:self
             configuration:configuration
               constraints:constraints
       certificateVerifier:certificateVerifier
@@ -311,18 +369,26 @@
 }
 
 - (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
-    peerConnectionWithDependencies:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+    peerConnectionWithDependencies:
-                       constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+        (RTC_OBJC_TYPE(RTCConfiguration) *)configuration
-                      dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies
+                       constraints:
-                          delegate:(id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
+                           (RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
-  return [[RTC_OBJC_TYPE(RTCPeerConnection) alloc] initWithDependencies:self
+                      dependencies:
+                          (std::unique_ptr<webrtc::PeerConnectionDependencies>)
+                              dependencies
+                          delegate:
+                              (id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)
+                                  delegate {
+  return [[RTC_OBJC_TYPE(RTCPeerConnection) alloc]
+      initWithDependencies:self
              configuration:configuration
                constraints:constraints
               dependencies:std::move(dependencies)
                   delegate:delegate];
 }
 
-- (void)setOptions:(nonnull RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions) *)options {
+- (void)setOptions:
+    (nonnull RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions) *)options {
   RTC_DCHECK(options != nil);
   _nativeFactory->SetOptions(options.nativeOptions);
 }
@@ -338,7 +404,8 @@
   }
   FILE *f = fopen(filePath.UTF8String, "wb");
   if (!f) {
-    RTCLogError(@"Error opening file: %@. Error: %s", filePath, strerror(errno));
+    RTCLogError(
+        @"Error opening file: %@. Error: %s", filePath, strerror(errno));
     return NO;
   }
   _hasStartedAecDump = _nativeFactory->StartAecDump(f, maxSizeInBytes);

sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm

@@ -25,7 +25,8 @@
 @implementation RTCPeerConnectionFactoryBuilder (DefaultComponents)
 
 + (RTCPeerConnectionFactoryBuilder *)defaultBuilder {
-  RTCPeerConnectionFactoryBuilder *builder = [[RTCPeerConnectionFactoryBuilder alloc] init];
+  RTCPeerConnectionFactoryBuilder *builder =
+      [[RTCPeerConnectionFactoryBuilder alloc] init];
   auto audioEncoderFactory = webrtc::CreateBuiltinAudioEncoderFactory();
   [builder setAudioEncoderFactory:audioEncoderFactory];
 

sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.h

@@ -29,20 +29,27 @@ NS_ASSUME_NONNULL_BEGIN
 
 - (void)setFieldTrials:(std::unique_ptr<webrtc::FieldTrialsView>)fieldTrials;
 
-- (void)setVideoEncoderFactory:(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory;
+- (void)setVideoEncoderFactory:
+    (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory;
 
-- (void)setVideoDecoderFactory:(std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory;
+- (void)setVideoDecoderFactory:
+    (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory;
 
-- (void)setAudioEncoderFactory:(rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory;
+- (void)setAudioEncoderFactory:
+    (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory;
 
-- (void)setAudioDecoderFactory:(rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory;
+- (void)setAudioDecoderFactory:
+    (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory;
 
-- (void)setAudioDeviceModule:(rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule;
+- (void)setAudioDeviceModule:
+    (rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule;
 
-- (void)setAudioProcessingModule:(rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule;
+- (void)setAudioProcessingModule:
+    (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule;
 
 - (void)setAudioProcessingBuilder:
-    (std::unique_ptr<webrtc::AudioProcessingBuilderInterface>)audioProcessingBuilder;
+    (std::unique_ptr<webrtc::AudioProcessingBuilderInterface>)
+        audioProcessingBuilder;
 
 @end
 

sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm

@@ -35,11 +35,13 @@
   _dependencies.trials = std::move(fieldTrials);
 }
 
-- (void)setVideoEncoderFactory:(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory {
+- (void)setVideoEncoderFactory:
+    (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory {
   _dependencies.video_encoder_factory = std::move(videoEncoderFactory);
 }
 
-- (void)setVideoDecoderFactory:(std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory {
+- (void)setVideoDecoderFactory:
+    (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory {
   _dependencies.video_decoder_factory = std::move(videoDecoderFactory);
 }
 
@@ -53,17 +55,20 @@
   _dependencies.audio_decoder_factory = std::move(audioDecoderFactory);
 }
 
-- (void)setAudioDeviceModule:(rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule {
+- (void)setAudioDeviceModule:
+    (rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule {
   _dependencies.adm = std::move(audioDeviceModule);
 }
 
 - (void)setAudioProcessingModule:
     (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule {
-  _dependencies.audio_processing_builder = CustomAudioProcessing(std::move(audioProcessingModule));
+  _dependencies.audio_processing_builder =
+      CustomAudioProcessing(std::move(audioProcessingModule));
 }
 
 - (void)setAudioProcessingBuilder:
-    (std::unique_ptr<webrtc::AudioProcessingBuilderInterface>)audioProcessingBuilder {
+    (std::unique_ptr<webrtc::AudioProcessingBuilderInterface>)
+        audioProcessingBuilder {
   _dependencies.audio_processing_builder = std::move(audioProcessingBuilder);
 }
 

sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions+Private.h

@@ -19,7 +19,8 @@ NS_ASSUME_NONNULL_BEGIN
 
     /** Returns the equivalent native PeerConnectionFactoryInterface::Options
      * structure. */
-    @property(nonatomic, readonly) webrtc::PeerConnectionFactoryInterface::Options nativeOptions;
+    @property(nonatomic, readonly)
+        webrtc::PeerConnectionFactoryInterface::Options nativeOptions;
 
 @end
 

sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.mm

@@ -44,11 +44,15 @@ void setNetworkBit(webrtc::PeerConnectionFactoryInterface::Options* options,
   options.disable_encryption = self.disableEncryption;
   options.disable_network_monitor = self.disableNetworkMonitor;
 
-  setNetworkBit(&options, rtc::ADAPTER_TYPE_LOOPBACK, self.ignoreLoopbackNetworkAdapter);
+  setNetworkBit(
+      &options, rtc::ADAPTER_TYPE_LOOPBACK, self.ignoreLoopbackNetworkAdapter);
   setNetworkBit(&options, rtc::ADAPTER_TYPE_VPN, self.ignoreVPNNetworkAdapter);
-  setNetworkBit(&options, rtc::ADAPTER_TYPE_CELLULAR, self.ignoreCellularNetworkAdapter);
+  setNetworkBit(
-  setNetworkBit(&options, rtc::ADAPTER_TYPE_WIFI, self.ignoreWiFiNetworkAdapter);
+      &options, rtc::ADAPTER_TYPE_CELLULAR, self.ignoreCellularNetworkAdapter);
-  setNetworkBit(&options, rtc::ADAPTER_TYPE_ETHERNET, self.ignoreEthernetNetworkAdapter);
+  setNetworkBit(
+      &options, rtc::ADAPTER_TYPE_WIFI, self.ignoreWiFiNetworkAdapter);
+  setNetworkBit(
+      &options, rtc::ADAPTER_TYPE_ETHERNET, self.ignoreEthernetNetworkAdapter);
 
   return options;
 }

sdk/objc/api/peerconnection/RTCRtcpParameters+Private.h

@@ -21,8 +21,8 @@ NS_ASSUME_NONNULL_BEGIN
     @property(nonatomic, readonly) webrtc::RtcpParameters nativeParameters;
 
 /** Initialize the object with a native RtcpParameters structure. */
-- (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters
+- (instancetype)initWithNativeParameters:
-    NS_DESIGNATED_INITIALIZER;
+    (const webrtc::RtcpParameters &)nativeParameters NS_DESIGNATED_INITIALIZER;
 
 @end
 

sdk/objc/api/peerconnection/RTCRtcpParameters.mm

@@ -22,7 +22,8 @@
   return [self initWithNativeParameters:nativeParameters];
 }
 
-- (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters {
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtcpParameters &)nativeParameters {
   self = [super init];
   if (self) {
     _cname = [NSString stringForStdString:nativeParameters.cname];

sdk/objc/api/peerconnection/RTCRtpCapabilities+Private.h

@@ -14,18 +14,21 @@
 
 NS_ASSUME_NONNULL_BEGIN
 
-@interface RTC_OBJC_TYPE (RTCRtpCapabilities)()
+@interface RTC_OBJC_TYPE (RTCRtpCapabilities)
+()
 
-/**
+    /**
      * The native RtpCapabilities representation of this RTCRtpCapabilities
      * object. This is needed to pass to the underlying C++ APIs.
      */
-@property(nonatomic, readonly) webrtc::RtpCapabilities nativeRtpCapabilities;
+    @property(nonatomic,
+              readonly) webrtc::RtpCapabilities nativeRtpCapabilities;
 
 /**
  * Initialize an RTCRtpCapabilities from a native RtpCapabilities.
  */
-- (instancetype)initWithNativeRtpCapabilities:(const webrtc::RtpCapabilities &)rtpCapabilities;
+- (instancetype)initWithNativeRtpCapabilities:
+    (const webrtc::RtpCapabilities &)rtpCapabilities;
 
 @end
 

sdk/objc/api/peerconnection/RTCRtpCapabilities.mm

@@ -38,8 +38,11 @@
     _codecs = codecs;
 
     NSMutableArray *headerExtensions = [[NSMutableArray alloc] init];
-    for (const auto &headerExtension : nativeRtpCapabilities.header_extensions) {
+    for (const auto &headerExtension :
-      [headerExtensions addObject:[[RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) alloc]
+         nativeRtpCapabilities.header_extensions) {
+      [headerExtensions
+          addObject:
+              [[RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) alloc]
                   initWithNativeRtpHeaderExtensionCapability:headerExtension]];
     }
     _headerExtensions = headerExtensions;
@@ -52,8 +55,10 @@
   for (RTC_OBJC_TYPE(RTCRtpCodecCapability) * codec in _codecs) {
     rtpCapabilities.codecs.push_back(codec.nativeRtpCodecCapability);
   }
-  for (RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) * headerExtension in _headerExtensions) {
+  for (RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *
-    rtpCapabilities.header_extensions.push_back(headerExtension.nativeRtpHeaderExtensionCapability);
+       headerExtension in _headerExtensions) {
+    rtpCapabilities.header_extensions.push_back(
+        headerExtension.nativeRtpHeaderExtensionCapability);
   }
   return rtpCapabilities;
 }

sdk/objc/api/peerconnection/RTCRtpCodecCapability+Private.h

@@ -14,13 +14,16 @@
 
 NS_ASSUME_NONNULL_BEGIN
 
-@interface RTC_OBJC_TYPE (RTCRtpCodecCapability)()
+@interface RTC_OBJC_TYPE (RTCRtpCodecCapability)
+()
 
-/**
+    /**
- * The native RtpCodecCapability representation of this RTCRtpCodecCapability
+     * The native RtpCodecCapability representation of this
- * object. This is needed to pass to the underlying C++ APIs.
+     * RTCRtpCodecCapability object. This is needed to pass to the underlying
+     * C++ APIs.
      */
-@property(nonatomic, readonly) webrtc::RtpCodecCapability nativeRtpCodecCapability;
+    @property(nonatomic,
+              readonly) webrtc::RtpCodecCapability nativeRtpCodecCapability;
 
 /**
  * Initialize an RTCRtpCodecCapability from a native RtpCodecCapability.

sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm

@@ -36,8 +36,8 @@
   self = [super init];
   if (self) {
     if (nativeRtpCodecCapability.preferred_payload_type) {
-      _preferredPayloadType =
+      _preferredPayloadType = [NSNumber
-          [NSNumber numberWithInt:*nativeRtpCodecCapability.preferred_payload_type];
+          numberWithInt:*nativeRtpCodecCapability.preferred_payload_type];
     }
     _name = [NSString stringForStdString:nativeRtpCodecCapability.name];
     switch (nativeRtpCodecCapability.kind) {
@@ -55,10 +55,12 @@
         break;
     }
     if (nativeRtpCodecCapability.clock_rate) {
-      _clockRate = [NSNumber numberWithInt:*nativeRtpCodecCapability.clock_rate];
+      _clockRate =
+          [NSNumber numberWithInt:*nativeRtpCodecCapability.clock_rate];
     }
     if (nativeRtpCodecCapability.num_channels) {
-      _numChannels = [NSNumber numberWithInt:*nativeRtpCodecCapability.num_channels];
+      _numChannels =
+          [NSNumber numberWithInt:*nativeRtpCodecCapability.num_channels];
     }
     NSMutableDictionary *parameters = [NSMutableDictionary dictionary];
     for (const auto &parameter : nativeRtpCodecCapability.parameters) {
@@ -66,13 +68,15 @@
                      forKey:[NSString stringForStdString:parameter.first]];
     }
     _parameters = parameters;
-    _mimeType = [NSString stringForStdString:nativeRtpCodecCapability.mime_type()];
+    _mimeType =
+        [NSString stringForStdString:nativeRtpCodecCapability.mime_type()];
   }
   return self;
 }
 
 - (NSString *)description {
-  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpCodecCapability) {\n  "
+  return [NSString
+      stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpCodecCapability) {\n  "
                        @"preferredPayloadType: %@\n  name: %@\n  kind: %@\n  "
                        @"clockRate: %@\n  numChannels: %@\n  parameters: %@\n  "
                        @"mimeType: %@\n}",
@@ -88,7 +92,8 @@
 - (webrtc::RtpCodecCapability)nativeRtpCodecCapability {
   webrtc::RtpCodecCapability rtpCodecCapability;
   if (_preferredPayloadType != nil) {
-    rtpCodecCapability.preferred_payload_type = std::optional<int>(_preferredPayloadType.intValue);
+    rtpCodecCapability.preferred_payload_type =
+        std::optional<int>(_preferredPayloadType.intValue);
   }
   rtpCodecCapability.name = [NSString stdStringForString:_name];
   // NSString pointer comparison is safe here since "kind" is readonly and only

sdk/objc/api/peerconnection/RTCRtpCodecParameters+Private.h

@@ -21,7 +21,8 @@ NS_ASSUME_NONNULL_BEGIN
     @property(nonatomic, readonly) webrtc::RtpCodecParameters nativeParameters;
 
 /** Initialize the object with a native RtpCodecParameters structure. */
-- (instancetype)initWithNativeParameters:(const webrtc::RtpCodecParameters &)nativeParameters
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpCodecParameters &)nativeParameters
     NS_DESIGNATED_INITIALIZER;
 
 @end

sdk/objc/api/peerconnection/RTCRtpCodecParameters.mm

@@ -16,22 +16,22 @@
 #include "media/base/media_constants.h"
 #include "rtc_base/checks.h"
 
-const NSString * const kRTCRtxCodecName = @(cricket::kRtxCodecName);
+const NSString *const kRTCRtxCodecName = @(cricket::kRtxCodecName);
-const NSString * const kRTCRedCodecName = @(cricket::kRedCodecName);
+const NSString *const kRTCRedCodecName = @(cricket::kRedCodecName);
-const NSString * const kRTCUlpfecCodecName = @(cricket::kUlpfecCodecName);
+const NSString *const kRTCUlpfecCodecName = @(cricket::kUlpfecCodecName);
-const NSString * const kRTCFlexfecCodecName = @(cricket::kFlexfecCodecName);
+const NSString *const kRTCFlexfecCodecName = @(cricket::kFlexfecCodecName);
-const NSString * const kRTCOpusCodecName = @(cricket::kOpusCodecName);
+const NSString *const kRTCOpusCodecName = @(cricket::kOpusCodecName);
-const NSString * const kRTCL16CodecName  = @(cricket::kL16CodecName);
+const NSString *const kRTCL16CodecName = @(cricket::kL16CodecName);
-const NSString * const kRTCG722CodecName = @(cricket::kG722CodecName);
+const NSString *const kRTCG722CodecName = @(cricket::kG722CodecName);
-const NSString * const kRTCIlbcCodecName = @(cricket::kIlbcCodecName);
+const NSString *const kRTCIlbcCodecName = @(cricket::kIlbcCodecName);
-const NSString * const kRTCPcmuCodecName = @(cricket::kPcmuCodecName);
+const NSString *const kRTCPcmuCodecName = @(cricket::kPcmuCodecName);
-const NSString * const kRTCPcmaCodecName = @(cricket::kPcmaCodecName);
+const NSString *const kRTCPcmaCodecName = @(cricket::kPcmaCodecName);
-const NSString * const kRTCDtmfCodecName = @(cricket::kDtmfCodecName);
+const NSString *const kRTCDtmfCodecName = @(cricket::kDtmfCodecName);
-const NSString * const kRTCComfortNoiseCodecName =
+const NSString *const kRTCComfortNoiseCodecName =
     @(cricket::kComfortNoiseCodecName);
-const NSString * const kRTCVp8CodecName = @(cricket::kVp8CodecName);
+const NSString *const kRTCVp8CodecName = @(cricket::kVp8CodecName);
-const NSString * const kRTCVp9CodecName = @(cricket::kVp9CodecName);
+const NSString *const kRTCVp9CodecName = @(cricket::kVp9CodecName);
-const NSString * const kRTCH264CodecName = @(cricket::kH264CodecName);
+const NSString *const kRTCH264CodecName = @(cricket::kH264CodecName);
 
 @implementation RTC_OBJC_TYPE (RTCRtpCodecParameters)
 

sdk/objc/api/peerconnection/RTCRtpEncodingParameters+Private.h

@@ -18,10 +18,12 @@ NS_ASSUME_NONNULL_BEGIN
 ()
 
     /** Returns the equivalent native RtpEncodingParameters structure. */
-    @property(nonatomic, readonly) webrtc::RtpEncodingParameters nativeParameters;
+    @property(nonatomic,
+              readonly) webrtc::RtpEncodingParameters nativeParameters;
 
 /** Initialize the object with a native RtpEncodingParameters structure. */
-- (instancetype)initWithNativeParameters:(const webrtc::RtpEncodingParameters &)nativeParameters
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpEncodingParameters &)nativeParameters
     NS_DESIGNATED_INITIALIZER;
 
 @end

sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h

@@ -66,7 +66,8 @@ RTC_OBJC_EXPORT
 @property(nonatomic, assign) RTCPriority networkPriority;
 
 /** Allow dynamic frame length changes for audio:
- https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-adaptiveptime */
+ https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-adaptiveptime
+ */
 @property(nonatomic, assign) BOOL adaptiveAudioPacketTime;
 
 - (instancetype)init;

sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm

@@ -51,11 +51,12 @@
       _maxFramerate = [NSNumber numberWithInt:*nativeParameters.max_framerate];
     }
     if (nativeParameters.num_temporal_layers) {
-      _numTemporalLayers = [NSNumber numberWithInt:*nativeParameters.num_temporal_layers];
+      _numTemporalLayers =
+          [NSNumber numberWithInt:*nativeParameters.num_temporal_layers];
     }
     if (nativeParameters.scale_resolution_down_by) {
-      _scaleResolutionDownBy =
+      _scaleResolutionDownBy = [NSNumber
-          [NSNumber numberWithDouble:*nativeParameters.scale_resolution_down_by];
+          numberWithDouble:*nativeParameters.scale_resolution_down_by];
     }
     if (nativeParameters.ssrc) {
       _ssrc = [NSNumber numberWithUnsignedLong:*nativeParameters.ssrc];
@@ -84,17 +85,19 @@
     parameters.max_framerate = std::optional<int>(_maxFramerate.intValue);
   }
   if (_numTemporalLayers != nil) {
-    parameters.num_temporal_layers = std::optional<int>(_numTemporalLayers.intValue);
+    parameters.num_temporal_layers =
+        std::optional<int>(_numTemporalLayers.intValue);
   }
   if (_scaleResolutionDownBy != nil) {
-    parameters.scale_resolution_down_by = std::optional<double>(_scaleResolutionDownBy.doubleValue);
+    parameters.scale_resolution_down_by =
+        std::optional<double>(_scaleResolutionDownBy.doubleValue);
   }
   if (_ssrc != nil) {
     parameters.ssrc = std::optional<uint32_t>(_ssrc.unsignedLongValue);
   }
   parameters.bitrate_priority = _bitratePriority;
-  parameters.network_priority =
+  parameters.network_priority = [RTC_OBJC_TYPE(RTCRtpEncodingParameters)
-      [RTC_OBJC_TYPE(RTCRtpEncodingParameters) nativePriorityFromPriority:_networkPriority];
+      nativePriorityFromPriority:_networkPriority];
   parameters.adaptive_ptime = _adaptiveAudioPacketTime;
   return parameters;
 }

sdk/objc/api/peerconnection/RTCRtpHeaderExtension+Private.h

@@ -21,8 +21,8 @@ NS_ASSUME_NONNULL_BEGIN
     @property(nonatomic, readonly) webrtc::RtpExtension nativeParameters;
 
 /** Initialize the object with a native RtpExtension structure. */
-- (instancetype)initWithNativeParameters:(const webrtc::RtpExtension &)nativeParameters
+- (instancetype)initWithNativeParameters:
-    NS_DESIGNATED_INITIALIZER;
+    (const webrtc::RtpExtension &)nativeParameters NS_DESIGNATED_INITIALIZER;
 
 @end
 

sdk/objc/api/peerconnection/RTCRtpHeaderExtension.mm

@@ -23,7 +23,8 @@
   return [self initWithNativeParameters:nativeExtension];
 }
 
-- (instancetype)initWithNativeParameters:(const webrtc::RtpExtension &)nativeParameters {
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpExtension &)nativeParameters {
   self = [super init];
   if (self) {
     _uri = [NSString stringForStdString:nativeParameters.uri];

sdk/objc/api/peerconnection/RTCRtpHeaderExtensionCapability+Private.h

@@ -14,17 +14,20 @@
 
 NS_ASSUME_NONNULL_BEGIN
 
-@interface RTC_OBJC_TYPE (RTCRtpHeaderExtensionCapability)()
+@interface RTC_OBJC_TYPE (RTCRtpHeaderExtensionCapability)
+()
+
+    /**
+     * The native RtpHeaderExtensionCapability representation of this
+     * RTCRtpHeaderExtensionCapability object. This is needed to pass to the
+     * underlying C++ APIs.
+     */
+    @property(nonatomic, readonly) webrtc::RtpHeaderExtensionCapability
+    nativeRtpHeaderExtensionCapability;
 
 /**
-* The native RtpHeaderExtensionCapability representation of this
+ * Initialize an RTCRtpHeaderExtensionCapability from a native
-* RTCRtpHeaderExtensionCapability object. This is needed to pass to the underlying C++ APIs.
+ * RtpHeaderExtensionCapability.
-*/
-@property(nonatomic,
-          readonly) webrtc::RtpHeaderExtensionCapability nativeRtpHeaderExtensionCapability;
-
-/**
- * Initialize an RTCRtpHeaderExtensionCapability from a native RtpHeaderExtensionCapability.
  */
 - (instancetype)initWithNativeRtpHeaderExtensionCapability:
     (const webrtc::RtpHeaderExtensionCapability &)rtpHeaderExtensionCapability;

sdk/objc/api/peerconnection/RTCRtpHeaderExtensionCapability.h

@@ -26,7 +26,8 @@ RTC_OBJC_EXPORT
 @property(nonatomic, readonly, nullable) NSNumber* preferredId;
 
 /** Whether the header extension is encrypted or not. */
-@property(nonatomic, readonly, getter=isPreferredEncrypted) BOOL preferredEncrypted;
+@property(nonatomic, readonly, getter=isPreferredEncrypted)
+    BOOL preferredEncrypted;
 
 /** Direction of the header extension. */
 @property(nonatomic) RTCRtpTransceiverDirection direction;

sdk/objc/api/peerconnection/RTCRtpHeaderExtensionCapability.mm

@@ -22,26 +22,32 @@
 
 - (instancetype)init {
   webrtc::RtpHeaderExtensionCapability nativeRtpHeaderExtensionCapability;
-  return [self initWithNativeRtpHeaderExtensionCapability:nativeRtpHeaderExtensionCapability];
+  return [self initWithNativeRtpHeaderExtensionCapability:
+                   nativeRtpHeaderExtensionCapability];
 }
 
 - (instancetype)initWithNativeRtpHeaderExtensionCapability:
-    (const webrtc::RtpHeaderExtensionCapability &)nativeRtpHeaderExtensionCapability {
+    (const webrtc::RtpHeaderExtensionCapability &)
+        nativeRtpHeaderExtensionCapability {
   self = [super init];
   if (self) {
     _uri = [NSString stringForStdString:nativeRtpHeaderExtensionCapability.uri];
     if (nativeRtpHeaderExtensionCapability.preferred_id) {
-      _preferredId = [NSNumber numberWithInt:*nativeRtpHeaderExtensionCapability.preferred_id];
+      _preferredId = [NSNumber
+          numberWithInt:*nativeRtpHeaderExtensionCapability.preferred_id];
     }
     _preferredEncrypted = nativeRtpHeaderExtensionCapability.preferred_encrypt;
     _direction = [RTC_OBJC_TYPE(RTCRtpTransceiver)
-        rtpTransceiverDirectionFromNativeDirection:nativeRtpHeaderExtensionCapability.direction];
+        rtpTransceiverDirectionFromNativeDirection:
+            nativeRtpHeaderExtensionCapability.direction];
   }
   return self;
 }
 
 - (NSString *)description {
-  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) {\n  uri: "
+  return
+      [NSString stringWithFormat:
+                    @"RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) {\n  uri: "
                     @"%@\n  preferredId: %@\n  preferredEncrypted: %d\n}",
                     _uri,
                     _preferredId,
@@ -52,11 +58,12 @@
   webrtc::RtpHeaderExtensionCapability rtpHeaderExtensionCapability;
   rtpHeaderExtensionCapability.uri = [NSString stdStringForString:_uri];
   if (_preferredId != nil) {
-    rtpHeaderExtensionCapability.preferred_id = std::optional<int>(_preferredId.intValue);
+    rtpHeaderExtensionCapability.preferred_id =
+        std::optional<int>(_preferredId.intValue);
   }
   rtpHeaderExtensionCapability.preferred_encrypt = _preferredEncrypted;
-  rtpHeaderExtensionCapability.direction =
+  rtpHeaderExtensionCapability.direction = [RTC_OBJC_TYPE(RTCRtpTransceiver)
-      [RTC_OBJC_TYPE(RTCRtpTransceiver) nativeRtpTransceiverDirectionFromDirection:_direction];
+      nativeRtpTransceiverDirectionFromDirection:_direction];
   return rtpHeaderExtensionCapability;
 }
 

sdk/objc/api/peerconnection/RTCRtpParameters+Private.h

@@ -21,8 +21,8 @@ NS_ASSUME_NONNULL_BEGIN
     @property(nonatomic, readonly) webrtc::RtpParameters nativeParameters;
 
 /** Initialize the object with a native RtpParameters structure. */
-- (instancetype)initWithNativeParameters:(const webrtc::RtpParameters &)nativeParameters
+- (instancetype)initWithNativeParameters:
-    NS_DESIGNATED_INITIALIZER;
+    (const webrtc::RtpParameters &)nativeParameters NS_DESIGNATED_INITIALIZER;
 
 @end
 

sdk/objc/api/peerconnection/RTCRtpParameters.h

@@ -40,10 +40,12 @@ RTC_OBJC_EXPORT
     NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtension) *> *headerExtensions;
 
 /** The currently active encodings in the order of preference. */
-@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCRtpEncodingParameters) *> *encodings;
+@property(nonatomic, copy)
+    NSArray<RTC_OBJC_TYPE(RTCRtpEncodingParameters) *> *encodings;
 
 /** The negotiated set of send codecs in order of preference. */
-@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCRtpCodecParameters) *> *codecs;
+@property(nonatomic, copy)
+    NSArray<RTC_OBJC_TYPE(RTCRtpCodecParameters) *> *codecs;
 
 /**
  * Degradation preference in case of CPU adaptation or constrained bandwidth.

sdk/objc/api/peerconnection/RTCRtpParameters.mm

@@ -34,13 +34,15 @@
     (const webrtc::RtpParameters &)nativeParameters {
   self = [super init];
   if (self) {
-    _transactionId = [NSString stringForStdString:nativeParameters.transaction_id];
+    _transactionId =
-    _rtcp =
+        [NSString stringForStdString:nativeParameters.transaction_id];
-        [[RTC_OBJC_TYPE(RTCRtcpParameters) alloc] initWithNativeParameters:nativeParameters.rtcp];
+    _rtcp = [[RTC_OBJC_TYPE(RTCRtcpParameters) alloc]
+        initWithNativeParameters:nativeParameters.rtcp];
 
     NSMutableArray *headerExtensions = [[NSMutableArray alloc] init];
     for (const auto &headerExtension : nativeParameters.header_extensions) {
-      [headerExtensions addObject:[[RTC_OBJC_TYPE(RTCRtpHeaderExtension) alloc]
+      [headerExtensions
+          addObject:[[RTC_OBJC_TYPE(RTCRtpHeaderExtension) alloc]
                         initWithNativeParameters:headerExtension]];
     }
     _headerExtensions = headerExtensions;
@@ -54,14 +56,14 @@
 
     NSMutableArray *codecs = [[NSMutableArray alloc] init];
     for (const auto &codec : nativeParameters.codecs) {
-      [codecs
+      [codecs addObject:[[RTC_OBJC_TYPE(RTCRtpCodecParameters) alloc]
-          addObject:[[RTC_OBJC_TYPE(RTCRtpCodecParameters) alloc] initWithNativeParameters:codec]];
+                            initWithNativeParameters:codec]];
     }
     _codecs = codecs;
 
     _degradationPreference = [RTC_OBJC_TYPE(RTCRtpParameters)
-        degradationPreferenceFromNativeDegradationPreference:nativeParameters
+        degradationPreferenceFromNativeDegradationPreference:
-                                                                 .degradation_preference];
+            nativeParameters.degradation_preference];
   }
   return self;
 }
@@ -70,7 +72,8 @@
   webrtc::RtpParameters parameters;
   parameters.transaction_id = [NSString stdStringForString:_transactionId];
   parameters.rtcp = [_rtcp nativeParameters];
-  for (RTC_OBJC_TYPE(RTCRtpHeaderExtension) * headerExtension in _headerExtensions) {
+  for (RTC_OBJC_TYPE(RTCRtpHeaderExtension) *
+       headerExtension in _headerExtensions) {
     parameters.header_extensions.push_back(headerExtension.nativeParameters);
   }
   for (RTC_OBJC_TYPE(RTCRtpEncodingParameters) * encoding in _encodings) {
@@ -81,13 +84,14 @@
   }
   if (_degradationPreference) {
     parameters.degradation_preference = [RTC_OBJC_TYPE(RTCRtpParameters)
-        nativeDegradationPreferenceFromDegradationPreference:(RTCDegradationPreference)
+        nativeDegradationPreferenceFromDegradationPreference:
-                                                                 _degradationPreference.intValue];
+            (RTCDegradationPreference)_degradationPreference.intValue];
   }
   return parameters;
 }
 
-+ (webrtc::DegradationPreference)nativeDegradationPreferenceFromDegradationPreference:
++ (webrtc::DegradationPreference)
+    nativeDegradationPreferenceFromDegradationPreference:
         (RTCDegradationPreference)degradationPreference {
   switch (degradationPreference) {
     case RTCDegradationPreferenceDisabled:

sdk/objc/api/peerconnection/RTCRtpReceiver+Native.h

@@ -16,7 +16,8 @@
 NS_ASSUME_NONNULL_BEGIN
 
 /**
- * This class extension exposes methods that work directly with injectable C++ components.
+ * This class extension exposes methods that work directly with injectable C++
+ * components.
  */
 @interface RTC_OBJC_TYPE (RTCRtpReceiver)
 ()
@@ -25,7 +26,8 @@ NS_ASSUME_NONNULL_BEGIN
      * This will decrypt the entire frame using the user provided decryption
      * mechanism regardless of whether SRTP is enabled or not.
      */
-    - (void)setFrameDecryptor : (rtc::scoped_refptr<webrtc::FrameDecryptorInterface>)frameDecryptor;
+    - (void)setFrameDecryptor
+    : (rtc::scoped_refptr<webrtc::FrameDecryptorInterface>)frameDecryptor;
 
 @end
 

sdk/objc/api/peerconnection/RTCRtpReceiver+Private.h

@@ -33,15 +33,18 @@ class RtpReceiverDelegateAdapter : public RtpReceiverObserverInterface {
 @interface RTC_OBJC_TYPE (RTCRtpReceiver)
 ()
 
-    @property(nonatomic,
+    @property(nonatomic, readonly)
-              readonly) rtc::scoped_refptr<webrtc::RtpReceiverInterface> nativeRtpReceiver;
+        rtc::scoped_refptr<webrtc::RtpReceiverInterface> nativeRtpReceiver;
 
 /** Initialize an RTCRtpReceiver with a native RtpReceiverInterface. */
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
-              nativeRtpReceiver:(rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver
+      initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+    nativeRtpReceiver:
+        (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver
     NS_DESIGNATED_INITIALIZER;
 
-+ (RTCRtpMediaType)mediaTypeForNativeMediaType:(cricket::MediaType)nativeMediaType;
++ (RTCRtpMediaType)mediaTypeForNativeMediaType:
+    (cricket::MediaType)nativeMediaType;
 
 + (cricket::MediaType)nativeMediaTypeForMediaType:(RTCRtpMediaType)mediaType;
 

sdk/objc/api/peerconnection/RTCRtpReceiver.h

@@ -33,18 +33,19 @@ RTC_OBJC_EXPORT
 
     /** Called when the first RTP packet is received.
      *
-     *  Note: Currently if there are multiple RtpReceivers of the same media type,
+     *  Note: Currently if there are multiple RtpReceivers of the same media
-     *  they will all call OnFirstPacketReceived at once.
+     * type, they will all call OnFirstPacketReceived at once.
      *
-     *  For example, if we create three audio receivers, A/B/C, they will listen to
+     *  For example, if we create three audio receivers, A/B/C, they will listen
-     *  the same signal from the underneath network layer. Whenever the first audio packet
+     * to the same signal from the underneath network layer. Whenever the first
-     *  is received, the underneath signal will be fired. All the receivers A/B/C will be
+     * audio packet is received, the underneath signal will be fired. All the
-     *  notified and the callback of the receiver's delegate will be called.
+     * receivers A/B/C will be notified and the callback of the receiver's
+     * delegate will be called.
      *
      *  The process is the same for video receivers.
      */
-    - (void)rtpReceiver
+    - (void)rtpReceiver : (RTC_OBJC_TYPE(RTCRtpReceiver) *)
-    : (RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver didReceiveFirstPacketForMediaType
+                              rtpReceiver didReceiveFirstPacketForMediaType
     : (RTCRtpMediaType)mediaType;
 
 @end
@@ -70,12 +71,13 @@ RTC_OBJC_EXPORT
  *  RTCMediaStreamTrack. Use isEqual: instead of == to compare
  *  RTCMediaStreamTrack instances.
  */
-@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCMediaStreamTrack) * track;
+@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCMediaStreamTrack) *
+    track;
 
 /**
-Returns an array that contains an object for each unique SSRC (synchronization source) identifier
+Returns an array that contains an object for each unique SSRC (synchronization
-and for each unique CSRC (contributing source) received by the current RTCRtpReceiver in the last
+source) identifier and for each unique CSRC (contributing source) received by
-ten seconds.
+the current RTCRtpReceiver in the last ten seconds.
 */
 @property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpSource) *> *sources;
 

sdk/objc/api/peerconnection/RTCRtpReceiver.mm

@@ -21,7 +21,8 @@
 
 namespace webrtc {
 
-RtpReceiverDelegateAdapter::RtpReceiverDelegateAdapter(RTC_OBJC_TYPE(RTCRtpReceiver) * receiver) {
+RtpReceiverDelegateAdapter::RtpReceiverDelegateAdapter(
+    RTC_OBJC_TYPE(RTCRtpReceiver) * receiver) {
   RTC_CHECK(receiver);
   receiver_ = receiver;
 }
@@ -31,7 +32,8 @@ void RtpReceiverDelegateAdapter::OnFirstPacketReceived(
   RTCRtpMediaType packet_media_type =
       [RTC_OBJC_TYPE(RTCRtpReceiver) mediaTypeForNativeMediaType:media_type];
   RTC_OBJC_TYPE(RTCRtpReceiver) *receiver = receiver_;
-  [receiver.delegate rtpReceiver:receiver didReceiveFirstPacketForMediaType:packet_media_type];
+  [receiver.delegate rtpReceiver:receiver
+      didReceiveFirstPacketForMediaType:packet_media_type];
 }
 
 }  // namespace webrtc
@@ -57,7 +59,8 @@ void RtpReceiverDelegateAdapter::OnFirstPacketReceived(
   rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack(
       _nativeRtpReceiver->track());
   if (nativeTrack) {
-    return [RTC_OBJC_TYPE(RTCMediaStreamTrack) mediaTrackForNativeTrack:nativeTrack
+    return
+        [RTC_OBJC_TYPE(RTCMediaStreamTrack) mediaTrackForNativeTrack:nativeTrack
                                                              factory:_factory];
   }
   return nil;
@@ -65,15 +68,18 @@ void RtpReceiverDelegateAdapter::OnFirstPacketReceived(
 
 - (NSString *)description {
   return [NSString
-      stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpReceiver) {\n  receiverId: %@\n}", self.receiverId];
+      stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpReceiver) {\n  receiverId: %@\n}",
+                       self.receiverId];
 }
 
 - (NSArray<RTC_OBJC_TYPE(RTCRtpSource) *> *)sources {
-  std::vector<webrtc::RtpSource> nativeSources = _nativeRtpReceiver->GetSources();
+  std::vector<webrtc::RtpSource> nativeSources =
+      _nativeRtpReceiver->GetSources();
   NSMutableArray<RTC_OBJC_TYPE(RTCRtpSource) *> *result =
       [[NSMutableArray alloc] initWithCapacity:nativeSources.size()];
   for (auto nativeSource : nativeSources) {
-    [result addObject:[[RTC_OBJC_TYPE(RTCRtpSource) alloc] initWithNativeRtpSource:nativeSource]];
+    [result addObject:[[RTC_OBJC_TYPE(RTCRtpSource) alloc]
+                          initWithNativeRtpSource:nativeSource]];
   }
   return result;
 }
@@ -94,7 +100,8 @@ void RtpReceiverDelegateAdapter::OnFirstPacketReceived(
   if (![object isMemberOfClass:[self class]]) {
     return NO;
   }
-  RTC_OBJC_TYPE(RTCRtpReceiver) *receiver = (RTC_OBJC_TYPE(RTCRtpReceiver) *)object;
+  RTC_OBJC_TYPE(RTCRtpReceiver) *receiver =
+      (RTC_OBJC_TYPE(RTCRtpReceiver) *)object;
   return _nativeRtpReceiver == receiver.nativeRtpReceiver;
 }
 
@@ -104,7 +111,8 @@ void RtpReceiverDelegateAdapter::OnFirstPacketReceived(
 
 #pragma mark - Native
 
-- (void)setFrameDecryptor:(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>)frameDecryptor {
+- (void)setFrameDecryptor:
+    (rtc::scoped_refptr<webrtc::FrameDecryptorInterface>)frameDecryptor {
   _nativeRtpReceiver->SetFrameDecryptor(frameDecryptor);
 }
 
@@ -114,14 +122,17 @@ void RtpReceiverDelegateAdapter::OnFirstPacketReceived(
   return _nativeRtpReceiver;
 }
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
+      initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
     nativeRtpReceiver:
         (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver {
   self = [super init];
   if (self) {
     _factory = factory;
     _nativeRtpReceiver = nativeRtpReceiver;
-    RTCLogInfo(@"RTC_OBJC_TYPE(RTCRtpReceiver)(%p): created receiver: %@", self, self.description);
+    RTCLogInfo(@"RTC_OBJC_TYPE(RTCRtpReceiver)(%p): created receiver: %@",
+               self,
+               self.description);
     _observer.reset(new webrtc::RtpReceiverDelegateAdapter(self));
     _nativeRtpReceiver->SetObserver(_observer.get());
   }

sdk/objc/api/peerconnection/RTCRtpSender+Native.h

@@ -16,17 +16,19 @@
 NS_ASSUME_NONNULL_BEGIN
 
 /**
- * This class extension exposes methods that work directly with injectable C++ components.
+ * This class extension exposes methods that work directly with injectable C++
+ * components.
  */
 @interface RTC_OBJC_TYPE (RTCRtpSender)
 ()
 
     /** Sets a defined frame encryptor that will encrypt the entire frame
      * before it is sent across the network. This will encrypt the entire frame
-     * using the user provided encryption mechanism regardless of whether SRTP is
+     * using the user provided encryption mechanism regardless of whether SRTP
-     * enabled or not.
+     * is enabled or not.
      */
-    - (void)setFrameEncryptor : (rtc::scoped_refptr<webrtc::FrameEncryptorInterface>)frameEncryptor;
+    - (void)setFrameEncryptor
+    : (rtc::scoped_refptr<webrtc::FrameEncryptorInterface>)frameEncryptor;
 
 @end
 

sdk/objc/api/peerconnection/RTCRtpSender+Private.h

@@ -19,12 +19,14 @@ NS_ASSUME_NONNULL_BEGIN
 @interface RTC_OBJC_TYPE (RTCRtpSender)
 ()
 
-    @property(nonatomic, readonly) rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeRtpSender;
+    @property(nonatomic, readonly)
+        rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeRtpSender;
 
 /** Initialize an RTCRtpSender with a native RtpSenderInterface. */
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)initWithFactory:
-                nativeRtpSender:(rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender
+                    (RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
-    NS_DESIGNATED_INITIALIZER;
+                nativeRtpSender:(rtc::scoped_refptr<webrtc::RtpSenderInterface>)
+                                    nativeRtpSender NS_DESIGNATED_INITIALIZER;
 
 @end
 

sdk/objc/api/peerconnection/RTCRtpSender.h

@@ -40,7 +40,8 @@ RTC_OBJC_EXPORT
 @property(nonatomic, copy) NSArray<NSString *> *streamIds;
 
 /** The RTCDtmfSender accociated with the RTP sender. */
-@property(nonatomic, readonly, nullable) id<RTC_OBJC_TYPE(RTCDtmfSender)> dtmfSender;
+@property(nonatomic, readonly, nullable) id<RTC_OBJC_TYPE(RTCDtmfSender)>
+    dtmfSender;
 
 @end
 

sdk/objc/api/peerconnection/RTCRtpSender.mm

@@ -37,7 +37,10 @@
 
 - (void)setParameters:(RTC_OBJC_TYPE(RTCRtpParameters) *)parameters {
   if (!_nativeRtpSender->SetParameters(parameters.nativeParameters).ok()) {
-    RTCLogError(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): Failed to set parameters: %@", self, parameters);
+    RTCLogError(
+        @"RTC_OBJC_TYPE(RTCRtpSender)(%p): Failed to set parameters: %@",
+        self,
+        parameters);
   }
 }
 
@@ -45,7 +48,8 @@
   rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack(
       _nativeRtpSender->track());
   if (nativeTrack) {
-    return [RTC_OBJC_TYPE(RTCMediaStreamTrack) mediaTrackForNativeTrack:nativeTrack
+    return
+        [RTC_OBJC_TYPE(RTCMediaStreamTrack) mediaTrackForNativeTrack:nativeTrack
                                                              factory:_factory];
   }
   return nil;
@@ -53,13 +57,16 @@
 
 - (void)setTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track {
   if (!_nativeRtpSender->SetTrack(track.nativeTrack.get())) {
-    RTCLogError(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): Failed to set track %@", self, track);
+    RTCLogError(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): Failed to set track %@",
+                self,
+                track);
   }
 }
 
 - (NSArray<NSString *> *)streamIds {
   std::vector<std::string> nativeStreamIds = _nativeRtpSender->stream_ids();
-  NSMutableArray *streamIds = [NSMutableArray arrayWithCapacity:nativeStreamIds.size()];
+  NSMutableArray *streamIds =
+      [NSMutableArray arrayWithCapacity:nativeStreamIds.size()];
   for (const auto &s : nativeStreamIds) {
     [streamIds addObject:[NSString stringForStdString:s]];
   }
@@ -76,7 +83,8 @@
 
 - (NSString *)description {
   return [NSString
-      stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpSender) {\n  senderId: %@\n}", self.senderId];
+      stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpSender) {\n  senderId: %@\n}",
+                       self.senderId];
 }
 
 - (BOOL)isEqual:(id)object {
@@ -99,7 +107,8 @@
 
 #pragma mark - Native
 
-- (void)setFrameEncryptor:(rtc::scoped_refptr<webrtc::FrameEncryptorInterface>)frameEncryptor {
+- (void)setFrameEncryptor:
+    (rtc::scoped_refptr<webrtc::FrameEncryptorInterface>)frameEncryptor {
   _nativeRtpSender->SetFrameEncryptor(frameEncryptor);
 }
 
@@ -109,8 +118,10 @@
   return _nativeRtpSender;
 }
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
-                nativeRtpSender:(rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
+    initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+    nativeRtpSender:
+        (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
   NSParameterAssert(factory);
   NSParameterAssert(nativeRtpSender);
   self = [super init];
@@ -121,11 +132,13 @@
       rtc::scoped_refptr<webrtc::DtmfSenderInterface> nativeDtmfSender(
           _nativeRtpSender->GetDtmfSender());
       if (nativeDtmfSender) {
-        _dtmfSender =
+        _dtmfSender = [[RTC_OBJC_TYPE(RTCDtmfSender) alloc]
-            [[RTC_OBJC_TYPE(RTCDtmfSender) alloc] initWithNativeDtmfSender:nativeDtmfSender];
+            initWithNativeDtmfSender:nativeDtmfSender];
       }
     }
-    RTCLogInfo(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): created sender: %@", self, self.description);
+    RTCLogInfo(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): created sender: %@",
+               self,
+               self.description);
   }
   return self;
 }

sdk/objc/api/peerconnection/RTCRtpSource.h

@@ -27,29 +27,30 @@ RTC_OBJC_EXPORT
 (RTCRtpSource)<NSObject>
 
     /**
-    A positive integer value specifying the CSRC identifier of the contributing source or SSRC
+    A positive integer value specifying the CSRC identifier of the contributing
-    identifier of the synchronization source. This uniquely identifies the source of the particular
+    source or SSRC identifier of the synchronization source. This uniquely
-    stream RTP packets. */
+    identifies the source of the particular stream RTP packets. */
     @property(nonatomic, readonly) uint32_t sourceId;
 
 @property(nonatomic, readonly) RTCRtpSourceType sourceType;
 
 /**
-A floating-point value between 0.0 and 1.0 specifying the audio level contained in the last RTP
+A floating-point value between 0.0 and 1.0 specifying the audio level contained
-packet played from the contributing source.
+in the last RTP packet played from the contributing source.
 */
 @property(nonatomic, readonly, nullable) NSNumber *audioLevel;
 
 /**
-A timestamp indicating the most recent time at which a frame originating from this source was
+A timestamp indicating the most recent time at which a frame originating from
-delivered to the receiver's track
+this source was delivered to the receiver's track
 */
 @property(nonatomic, readonly) CFTimeInterval timestampUs;
 
 /**
-The RTP timestamp of the media. This source-generated timestamp indicates the time at which the
+The RTP timestamp of the media. This source-generated timestamp indicates the
-media in this packet, scheduled for play out at the time indicated by timestamp, was initially
+time at which the media in this packet, scheduled for play out at the time
-sampled or generated. It may be useful for sequencing and synchronization purposes.
+indicated by timestamp, was initially sampled or generated. It may be useful for
+sequencing and synchronization purposes.
 */
 @property(nonatomic, readonly) uint32_t rtpTimestamp;
 

sdk/objc/api/peerconnection/RTCRtpSource.mm

@@ -37,7 +37,8 @@
 
 - (RTCRtpSourceType)sourceType {
   return [RTC_OBJC_TYPE(RTCRtpSource)
-      rtpSourceTypeForNativeRtpSourceType:_nativeRtpSource.value().source_type()];
+      rtpSourceTypeForNativeRtpSourceType:_nativeRtpSource.value()
+                                              .source_type()];
 }
 
 - (NSNumber *)audioLevel {
@@ -59,12 +60,14 @@
 
 - (NSString *)description {
   return [NSString
-      stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpSource) {\n  sourceId: %d, sourceType: %@\n}",
+      stringWithFormat:
+          @"RTC_OBJC_TYPE(RTCRtpSource) {\n  sourceId: %d, sourceType: %@\n}",
           self.sourceId,
           [RTC_OBJC_TYPE(RTCRtpSource) stringForRtpSourceType:self.sourceType]];
 }
 
-- (instancetype)initWithNativeRtpSource:(const webrtc::RtpSource &)nativeRtpSource {
+- (instancetype)initWithNativeRtpSource:
+    (const webrtc::RtpSource &)nativeRtpSource {
   self = [super init];
   if (self) {
     _nativeRtpSource = nativeRtpSource;
@@ -72,7 +75,8 @@
   return self;
 }
 
-+ (RTCRtpSourceType)rtpSourceTypeForNativeRtpSourceType:(webrtc::RtpSourceType)nativeRtpSourceType {
++ (RTCRtpSourceType)rtpSourceTypeForNativeRtpSourceType:
+    (webrtc::RtpSourceType)nativeRtpSourceType {
   switch (nativeRtpSourceType) {
     case webrtc::RtpSourceType::SSRC:
       return RTCRtpSourceTypeSSRC;

sdk/objc/api/peerconnection/RTCRtpTransceiver+Private.h

@@ -26,14 +26,14 @@ NS_ASSUME_NONNULL_BEGIN
 @interface RTC_OBJC_TYPE (RTCRtpTransceiver)
 ()
 
-    @property(nonatomic,
+    @property(nonatomic, readonly) rtc::scoped_refptr<
-              readonly) rtc::scoped_refptr<webrtc::RtpTransceiverInterface> nativeRtpTransceiver;
+        webrtc::RtpTransceiverInterface> nativeRtpTransceiver;
 
 /** Initialize an RTCRtpTransceiver with a native RtpTransceiverInterface. */
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
-           nativeRtpTransceiver:
+         initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
-               (rtc::scoped_refptr<webrtc::RtpTransceiverInterface>)nativeRtpTransceiver
+    nativeRtpTransceiver:(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>)
-    NS_DESIGNATED_INITIALIZER;
+                             nativeRtpTransceiver NS_DESIGNATED_INITIALIZER;
 
 + (webrtc::RtpTransceiverDirection)nativeRtpTransceiverDirectionFromDirection:
     (RTCRtpTransceiverDirection)direction;

sdk/objc/api/peerconnection/RTCRtpTransceiver.h

@@ -41,7 +41,8 @@ RTC_OBJC_EXPORT
 @property(nonatomic) NSArray<NSString *> *streamIds;
 
 /** TODO(bugs.webrtc.org/7600): Not implemented. */
-@property(nonatomic) NSArray<RTC_OBJC_TYPE(RTCRtpEncodingParameters) *> *sendEncodings;
+@property(nonatomic)
+    NSArray<RTC_OBJC_TYPE(RTCRtpEncodingParameters) *> *sendEncodings;
 
 @end
 
@@ -50,10 +51,10 @@ RTC_OBJC_EXPORT
 @class RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability);
 
 /** The RTCRtpTransceiver maps to the RTCRtpTransceiver defined by the
- *  WebRTC specification. A transceiver represents a combination of an RTCRtpSender
+ *  WebRTC specification. A transceiver represents a combination of an
- *  and an RTCRtpReceiver that share a common mid. As defined in JSEP, an
+ * RTCRtpSender and an RTCRtpReceiver that share a common mid. As defined in
- *  RTCRtpTransceiver is said to be associated with a media description if its
+ * JSEP, an RTCRtpTransceiver is said to be associated with a media description
- *  mid property is non-nil; otherwise, it is said to be disassociated.
+ * if its mid property is non-nil; otherwise, it is said to be disassociated.
  *  JSEP: https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24
  *
  *  Note that RTCRtpTransceivers are only supported when using
@@ -66,8 +67,8 @@ RTC_OBJC_EXPORT
 @protocol RTC_OBJC_TYPE
 (RTCRtpTransceiver)<NSObject>
 
-    /** Media type of the transceiver. The sender and receiver will also have this
+    /** Media type of the transceiver. The sender and receiver will also have
-     *  type.
+     * this type.
      */
     @property(nonatomic, readonly) RTCRtpMediaType mediaType;
 
@@ -107,15 +108,18 @@ RTC_OBJC_EXPORT
 @property(nonatomic, readonly) RTCRtpTransceiverDirection direction;
 
 /** It will contain all the RTP header extensions that are supported.
- *  The direction attribute for all extensions that are mandatory to use MUST be initialized to an
+ *  The direction attribute for all extensions that are mandatory to use MUST be
- * appropriate value other than RTCRtpTransceiverDirectionStopped. The direction attribute for
+ * initialized to an appropriate value other than
- * extensions that will not be offered by default in an initial offer MUST be initialized to
+ * RTCRtpTransceiverDirectionStopped. The direction attribute for extensions
- * RTCRtpTransceiverDirectionStopped.
+ * that will not be offered by default in an initial offer MUST be initialized
+ * to RTCRtpTransceiverDirectionStopped.
  */
 @property(nonatomic, readonly, copy)
-    NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *> *headerExtensionsToNegotiate;
+    NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *>
+        *headerExtensionsToNegotiate;
 @property(nonatomic, readonly, copy)
-    NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *> *negotiatedHeaderExtensions;
+    NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *>
+        *negotiatedHeaderExtensions;
 
 /** The currentDirection attribute indicates the current direction negotiated
  *  for this transceiver. If this transceiver has never been represented in an
@@ -135,19 +139,22 @@ RTC_OBJC_EXPORT
  * by WebRTC for this transceiver.
  * https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-setcodecpreferences
  */
-- (BOOL)setCodecPreferences:(NSArray<RTC_OBJC_TYPE(RTCRtpCodecCapability) *> *_Nullable)codecs
+- (BOOL)setCodecPreferences:
+            (NSArray<RTC_OBJC_TYPE(RTCRtpCodecCapability) *> *_Nullable)codecs
                       error:(NSError **_Nullable)error;
 
 /** Deprecated version of [RTCRtpTransceiver setCodecPreferences:error:] */
-- (void)setCodecPreferences:(NSArray<RTC_OBJC_TYPE(RTCRtpCodecCapability) *> *_Nullable)codecs
+- (void)setCodecPreferences:
+    (NSArray<RTC_OBJC_TYPE(RTCRtpCodecCapability) *> *_Nullable)codecs
     RTC_OBJC_DEPRECATED("Use setCodecPreferences:error: instead.");
 
-/** The setHeaderExtensionsToNegotiate method overrides the default header extensions used
+/** The setHeaderExtensionsToNegotiate method overrides the default header
- *  by WebRTC for this transceiver.
+ * extensions used by WebRTC for this transceiver.
  *  https://w3c.github.io/webrtc-extensions/#ref-for-dom-rtcrtptransceiver-setheaderextensionstonegotiate
  */
 - (BOOL)setHeaderExtensionsToNegotiate:
-            (NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *> *)extensions
+            (NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *> *)
+                extensions
                                  error:(NSError **)error;
 
 /** An update of directionality does not take effect immediately. Instead,
@@ -155,7 +162,8 @@ RTC_OBJC_EXPORT
  *  descriptions as sendrecv, sendonly, recvonly, or inactive.
  *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
  */
-- (void)setDirection:(RTCRtpTransceiverDirection)direction error:(NSError **)error;
+- (void)setDirection:(RTCRtpTransceiverDirection)direction
+               error:(NSError **)error;
 
 @end
 

sdk/objc/api/peerconnection/RTCRtpTransceiver.mm

@@ -39,12 +39,13 @@ NSString *const kRTCRtpTransceiverErrorDomain = @"org.webrtc.RTCRtpTranceiver";
 
 - (webrtc::RtpTransceiverInit)nativeInit {
   webrtc::RtpTransceiverInit init;
-  init.direction =
+  init.direction = [RTC_OBJC_TYPE(RTCRtpTransceiver)
-      [RTC_OBJC_TYPE(RTCRtpTransceiver) nativeRtpTransceiverDirectionFromDirection:_direction];
+      nativeRtpTransceiverDirectionFromDirection:_direction];
   for (NSString *streamId in _streamIds) {
     init.stream_ids.push_back([streamId UTF8String]);
   }
-  for (RTC_OBJC_TYPE(RTCRtpEncodingParameters) * sendEncoding in _sendEncodings) {
+  for (RTC_OBJC_TYPE(RTCRtpEncodingParameters) *
+       sendEncoding in _sendEncodings) {
     init.send_encodings.push_back(sendEncoding.nativeParameters);
   }
   return init;
@@ -79,42 +80,52 @@ NSString *const kRTCRtpTransceiverErrorDomain = @"org.webrtc.RTCRtpTranceiver";
 
 - (RTCRtpTransceiverDirection)direction {
   return [RTC_OBJC_TYPE(RTCRtpTransceiver)
-      rtpTransceiverDirectionFromNativeDirection:_nativeRtpTransceiver->direction()];
+      rtpTransceiverDirectionFromNativeDirection:_nativeRtpTransceiver
+                                                     ->direction()];
 }
 
-- (NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *> *)headerExtensionsToNegotiate {
+- (NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *> *)
+    headerExtensionsToNegotiate {
   std::vector<webrtc::RtpHeaderExtensionCapability> nativeHeaderExtensions(
       _nativeRtpTransceiver->GetHeaderExtensionsToNegotiate());
 
   NSMutableArray *headerExtensions =
       [NSMutableArray arrayWithCapacity:nativeHeaderExtensions.size()];
   for (const auto &headerExtension : nativeHeaderExtensions) {
-    [headerExtensions addObject:[[RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) alloc]
+    [headerExtensions
+        addObject:
+            [[RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) alloc]
                 initWithNativeRtpHeaderExtensionCapability:headerExtension]];
   }
   return headerExtensions;
 }
 
-- (NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *> *)negotiatedHeaderExtensions {
+- (NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *> *)
+    negotiatedHeaderExtensions {
   std::vector<webrtc::RtpHeaderExtensionCapability> nativeHeaderExtensions(
       _nativeRtpTransceiver->GetNegotiatedHeaderExtensions());
 
   NSMutableArray *headerExtensions =
       [NSMutableArray arrayWithCapacity:nativeHeaderExtensions.size()];
   for (const auto &headerExtension : nativeHeaderExtensions) {
-    [headerExtensions addObject:[[RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) alloc]
+    [headerExtensions
+        addObject:
+            [[RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) alloc]
                 initWithNativeRtpHeaderExtensionCapability:headerExtension]];
   }
   return headerExtensions;
 }
 
-- (void)setDirection:(RTCRtpTransceiverDirection)direction error:(NSError **)error {
+- (void)setDirection:(RTCRtpTransceiverDirection)direction
+               error:(NSError **)error {
   webrtc::RTCError nativeError = _nativeRtpTransceiver->SetDirectionWithError(
-      [RTC_OBJC_TYPE(RTCRtpTransceiver) nativeRtpTransceiverDirectionFromDirection:direction]);
+      [RTC_OBJC_TYPE(RTCRtpTransceiver)
+          nativeRtpTransceiverDirectionFromDirection:direction]);
 
   if (!nativeError.ok() && error) {
     NSDictionary *userInfo = @{
-      NSLocalizedDescriptionKey : [NSString stringWithCString:nativeError.message()
+      NSLocalizedDescriptionKey :
+          [NSString stringWithCString:nativeError.message()
                              encoding:NSUTF8StringEncoding]
     };
     *error = [NSError errorWithDomain:kRTCRtpTransceiverErrorDomain
@@ -126,7 +137,8 @@ NSString *const kRTCRtpTransceiverErrorDomain = @"org.webrtc.RTCRtpTranceiver";
 - (BOOL)currentDirection:(RTCRtpTransceiverDirection *)currentDirectionOut {
   if (_nativeRtpTransceiver->current_direction()) {
     *currentDirectionOut = [RTC_OBJC_TYPE(RTCRtpTransceiver)
-        rtpTransceiverDirectionFromNativeDirection:*_nativeRtpTransceiver->current_direction()];
+        rtpTransceiverDirectionFromNativeDirection:*_nativeRtpTransceiver
+                                                        ->current_direction()];
     return YES;
   } else {
     return NO;
@@ -137,7 +149,8 @@ NSString *const kRTCRtpTransceiverErrorDomain = @"org.webrtc.RTCRtpTranceiver";
   _nativeRtpTransceiver->StopInternal();
 }
 
-- (BOOL)setCodecPreferences:(NSArray<RTC_OBJC_TYPE(RTCRtpCodecCapability) *> *)codecs
+- (BOOL)setCodecPreferences:
+            (NSArray<RTC_OBJC_TYPE(RTCRtpCodecCapability) *> *)codecs
                       error:(NSError **)error {
   std::vector<webrtc::RtpCodecCapability> codecCapabilities;
   if (codecs) {
@@ -145,33 +158,42 @@ NSString *const kRTCRtpTransceiverErrorDomain = @"org.webrtc.RTCRtpTranceiver";
       codecCapabilities.push_back(rtpCodecCapability.nativeRtpCodecCapability);
     }
   }
-  webrtc::RTCError nativeError = _nativeRtpTransceiver->SetCodecPreferences(codecCapabilities);
+  webrtc::RTCError nativeError =
+      _nativeRtpTransceiver->SetCodecPreferences(codecCapabilities);
   if (!nativeError.ok() && error) {
-    *error = [NSError
+    *error = [NSError errorWithDomain:kRTCRtpTransceiverErrorDomain
-        errorWithDomain:kRTCRtpTransceiverErrorDomain
                                  code:static_cast<int>(nativeError.type())
-               userInfo:@{@"message" : [NSString stringWithUTF8String:nativeError.message()]}];
+                             userInfo:@{
+                               @"message" : [NSString
+                                   stringWithUTF8String:nativeError.message()]
+                             }];
   }
   return nativeError.ok();
 }
 
-- (void)setCodecPreferences:(NSArray<RTC_OBJC_TYPE(RTCRtpCodecCapability) *> *)codecs {
+- (void)setCodecPreferences:
+    (NSArray<RTC_OBJC_TYPE(RTCRtpCodecCapability) *> *)codecs {
   [self setCodecPreferences:codecs error:nil];
 }
 
 - (BOOL)setHeaderExtensionsToNegotiate:
-            (NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *> *)extensions
+            (NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *> *)
+                extensions
                                  error:(NSError **)error {
   std::vector<webrtc::RtpHeaderExtensionCapability> headerExtensionCapabilities;
-  for (RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) * extension in extensions) {
+  for (RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *
-    headerExtensionCapabilities.push_back(extension.nativeRtpHeaderExtensionCapability);
+       extension in extensions) {
+    headerExtensionCapabilities.push_back(
+        extension.nativeRtpHeaderExtensionCapability);
   }
   webrtc::RTCError nativeError =
-      _nativeRtpTransceiver->SetHeaderExtensionsToNegotiate(headerExtensionCapabilities);
+      _nativeRtpTransceiver->SetHeaderExtensionsToNegotiate(
+          headerExtensionCapabilities);
   BOOL ok = nativeError.ok();
   if (!ok && error) {
     NSDictionary *userInfo = @{
-      NSLocalizedDescriptionKey : [NSString stringWithCString:nativeError.message()
+      NSLocalizedDescriptionKey :
+          [NSString stringWithCString:nativeError.message()
                              encoding:NSUTF8StringEncoding]
     };
     *error = [NSError errorWithDomain:kRTCRtpTransceiverErrorDomain
@@ -182,8 +204,8 @@ NSString *const kRTCRtpTransceiverErrorDomain = @"org.webrtc.RTCRtpTranceiver";
 }
 
 - (NSString *)description {
-  return [NSString
+  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpTransceiver) {\n  "
-      stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpTransceiver) {\n  sender: %@\n  receiver: %@\n}",
+                                    @"sender: %@\n  receiver: %@\n}",
                                     _sender,
                                     _receiver];
 }
@@ -198,7 +220,8 @@ NSString *const kRTCRtpTransceiverErrorDomain = @"org.webrtc.RTCRtpTranceiver";
   if (![object isMemberOfClass:[self class]]) {
     return NO;
   }
-  RTC_OBJC_TYPE(RTCRtpTransceiver) *transceiver = (RTC_OBJC_TYPE(RTCRtpTransceiver) *)object;
+  RTC_OBJC_TYPE(RTCRtpTransceiver) *transceiver =
+      (RTC_OBJC_TYPE(RTCRtpTransceiver) *)object;
   return _nativeRtpTransceiver == transceiver.nativeRtpTransceiver;
 }
 
@@ -212,22 +235,25 @@ NSString *const kRTCRtpTransceiverErrorDomain = @"org.webrtc.RTCRtpTranceiver";
   return _nativeRtpTransceiver;
 }
 
-- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+- (instancetype)
-           nativeRtpTransceiver:
+         initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
-               (rtc::scoped_refptr<webrtc::RtpTransceiverInterface>)nativeRtpTransceiver {
+    nativeRtpTransceiver:(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>)
+                             nativeRtpTransceiver {
   NSParameterAssert(factory);
   NSParameterAssert(nativeRtpTransceiver);
   self = [super init];
   if (self) {
     _factory = factory;
     _nativeRtpTransceiver = nativeRtpTransceiver;
-    _sender = [[RTC_OBJC_TYPE(RTCRtpSender) alloc] initWithFactory:_factory
+    _sender = [[RTC_OBJC_TYPE(RTCRtpSender) alloc]
+        initWithFactory:_factory
         nativeRtpSender:nativeRtpTransceiver->sender()];
-    _receiver =
+    _receiver = [[RTC_OBJC_TYPE(RTCRtpReceiver) alloc]
-        [[RTC_OBJC_TYPE(RTCRtpReceiver) alloc] initWithFactory:_factory
+          initWithFactory:_factory
         nativeRtpReceiver:nativeRtpTransceiver->receiver()];
-    RTCLogInfo(
+    RTCLogInfo(@"RTC_OBJC_TYPE(RTCRtpTransceiver)(%p): created transceiver: %@",
-        @"RTC_OBJC_TYPE(RTCRtpTransceiver)(%p): created transceiver: %@", self, self.description);
+               self,
+               self.description);
   }
   return self;
 }

sdk/objc/api/peerconnection/RTCSessionDescription+Private.h

@@ -19,11 +19,11 @@ NS_ASSUME_NONNULL_BEGIN
 
     /**
      * The native SessionDescriptionInterface representation of this
-     * RTCSessionDescription object. This is needed to pass to the underlying C++
+     * RTCSessionDescription object. This is needed to pass to the underlying
-     * APIs.
+     * C++ APIs.
      */
-    @property(nonatomic,
+    @property(nonatomic, readonly)
-              readonly) std::unique_ptr<webrtc::SessionDescriptionInterface> nativeDescription;
+        std::unique_ptr<webrtc::SessionDescriptionInterface> nativeDescription;
 
 /**
  * Initialize an RTCSessionDescription from a native

sdk/objc/api/peerconnection/RTCSessionDescription.h

@@ -37,7 +37,8 @@ RTC_OBJC_EXPORT
 - (instancetype)init NS_UNAVAILABLE;
 
 /** Initialize a session description with a type and SDP string. */
-- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp NS_DESIGNATED_INITIALIZER;
+- (instancetype)initWithType:(RTCSdpType)type
+                         sdp:(NSString *)sdp NS_DESIGNATED_INITIALIZER;
 
 + (NSString *)stringForType:(RTCSdpType)type;
 

sdk/objc/api/peerconnection/RTCSessionDescription.mm

@@ -41,7 +41,8 @@
 }
 
 - (NSString *)description {
-  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCSessionDescription):\n%@\n%@",
+  return [NSString
+      stringWithFormat:@"RTC_OBJC_TYPE(RTCSessionDescription):\n%@\n%@",
                        [[self class] stringForType:_type],
                        _sdp];
 }
@@ -51,7 +52,8 @@
 - (std::unique_ptr<webrtc::SessionDescriptionInterface>)nativeDescription {
   webrtc::SdpParseError error;
 
-  std::unique_ptr<webrtc::SessionDescriptionInterface> description(webrtc::CreateSessionDescription(
+  std::unique_ptr<webrtc::SessionDescriptionInterface> description(
+      webrtc::CreateSessionDescription(
           [[self class] nativeTypeForType:_type], _sdp.stdString, &error));
 
   if (!description) {
@@ -70,8 +72,7 @@
   nativeDescription->ToString(&sdp);
   RTCSdpType type = [[self class] typeForStdString:nativeDescription->type()];
 
-  return [self initWithType:type
+  return [self initWithType:type sdp:[NSString stringForStdString:sdp]];
-                        sdp:[NSString stringForStdString:sdp]];
 }
 
 + (std::string)stdStringForType:(RTCSdpType)type {