Refactor WebRTC self assignments in if clauses

This change refactors existing self-assignments within if clauses across
the WebRTC codebase.

*Why:*

- Bug Prevention: Assignments within conditionals are frequently
  unintended errors, often mistaken for equality checks.

- Clearer Code: Separating assignments from conditionals improves code
  readability and reduces the risk of misinterpretation.

Change-Id: I199dc26a35ceca109a2ac569b446811314dfdf0b
Bug: chromium:361594695
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360460
Reviewed-by: Chuck Hays <haysc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42850}

examples/objc/AppRTCMobile/ARDAppClient.m

@@ -84,7 +84,8 @@ static int const kKbpsMultiplier = 1000;
                          repeats:(BOOL)repeats
                     timerHandler:(void (^)(void))timerHandler {
   NSParameterAssert(timerHandler);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _timerHandler = timerHandler;
     _timer = [NSTimer scheduledTimerWithTimeInterval:interval
                                               target:self
@@ -140,7 +141,8 @@ static int const kKbpsMultiplier = 1000;
 }
 
 - (instancetype)initWithDelegate:(id<ARDAppClientDelegate>)delegate {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _roomServerClient = [[ARDAppEngineClient alloc] init];
     _delegate = delegate;
     NSURL *turnRequestURL = [NSURL URLWithString:kARDIceServerRequestUrl];
@@ -160,7 +162,8 @@ static int const kKbpsMultiplier = 1000;
   NSParameterAssert(rsClient);
   NSParameterAssert(channel);
   NSParameterAssert(turnClient);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _roomServerClient = rsClient;
     _channel = channel;
     _turnClient = turnClient;

examples/objc/AppRTCMobile/ARDCaptureController.m

@@ -24,12 +24,12 @@ const Float64 kFramerateLimit = 30.0;
 
 - (instancetype)initWithCapturer:(RTC_OBJC_TYPE(RTCCameraVideoCapturer) *)capturer
                         settings:(ARDSettingsModel *)settings {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _capturer = capturer;
     _settings = settings;
     _usingFrontCamera = YES;
   }
-
   return self;
 }
 

examples/objc/AppRTCMobile/ARDSignalingMessage.m

@@ -24,7 +24,8 @@ static NSString * const kARDTypeValueRemoveCandidates = @"remove-candidates";
 @synthesize type = _type;
 
 - (instancetype)initWithType:(ARDSignalingMessageType)type {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _type = type;
   }
   return self;
@@ -79,7 +80,8 @@ static NSString * const kARDTypeValueRemoveCandidates = @"remove-candidates";
 @synthesize candidate = _candidate;
 
 - (instancetype)initWithCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate {
-  if (self = [super initWithType:kARDSignalingMessageTypeCandidate]) {
+  self = [super initWithType:kARDSignalingMessageTypeCandidate];
+  if (self) {
     _candidate = candidate;
   }
   return self;
@@ -97,7 +99,8 @@ static NSString * const kARDTypeValueRemoveCandidates = @"remove-candidates";
 
 - (instancetype)initWithRemovedCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates {
   NSParameterAssert(candidates.count);
-  if (self = [super initWithType:kARDSignalingMessageTypeCandidateRemoval]) {
+  self = [super initWithType:kARDSignalingMessageTypeCandidateRemoval];
+  if (self) {
     _candidates = candidates;
   }
   return self;
@@ -130,7 +133,8 @@ static NSString * const kARDTypeValueRemoveCandidates = @"remove-candidates";
           NO, @"Unexpected type: %@", [RTC_OBJC_TYPE(RTCSessionDescription) stringForType:sdpType]);
       break;
   }
-  if (self = [super initWithType:messageType]) {
+  self = [super initWithType:messageType];
+  if (self) {
     _sessionDescription = description;
   }
   return self;

examples/objc/AppRTCMobile/ARDTURNClient.m

@@ -24,7 +24,8 @@ static NSInteger kARDTURNClientErrorBadResponse = -1;
 
 - (instancetype)initWithURL:(NSURL *)url {
   NSParameterAssert([url absoluteString].length);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _url = url;
   }
   return self;

examples/objc/AppRTCMobile/ARDWebSocketChannel.m

@@ -38,7 +38,8 @@ static NSString const *kARDWSSMessagePayloadKey = @"msg";
 - (instancetype)initWithURL:(NSURL *)url
                     restURL:(NSURL *)restURL
                    delegate:(id<ARDSignalingChannelDelegate>)delegate {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _url = url;
     _restURL = restURL;
     _delegate = delegate;

examples/objc/AppRTCMobile/ios/ARDFileCaptureController.m

@@ -22,7 +22,8 @@
 @synthesize fileCapturer = _fileCapturer;
 
 - (instancetype)initWithCapturer:(RTC_OBJC_TYPE(RTCFileVideoCapturer) *)capturer {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _fileCapturer = capturer;
   }
   return self;

examples/objc/AppRTCMobile/ios/ARDMainView.m

@@ -26,7 +26,8 @@ static CGFloat const kCallControlMargin = 8;
 }
 
 - (instancetype)initWithFrame:(CGRect)frame {
-  if (self = [super initWithFrame:frame]) {
+  self = [super initWithFrame:frame];
+  if (self) {
     _roomText = [[UITextField alloc] initWithFrame:CGRectZero];
     _roomText.borderStyle = UITextBorderStyleNone;
     _roomText.font = [UIFont systemFontOfSize:12];
@@ -82,7 +83,8 @@ static CGFloat const kCallControlMargin = 8;
 @synthesize isAudioLoopPlaying = _isAudioLoopPlaying;
 
 - (instancetype)initWithFrame:(CGRect)frame {
-  if (self = [super initWithFrame:frame]) {
+  self = [super initWithFrame:frame];
+  if (self) {
     _roomText = [[ARDRoomTextField alloc] initWithFrame:CGRectZero];
     [self addSubview:_roomText];
 

examples/objc/AppRTCMobile/ios/ARDStatsView.m

@@ -20,7 +20,8 @@
 }
 
 - (instancetype)initWithFrame:(CGRect)frame {
-  if (self = [super initWithFrame:frame]) {
+  self = [super initWithFrame:frame];
+  if (self) {
     _statsLabel = [[UILabel alloc] initWithFrame:CGRectZero];
     _statsLabel.numberOfLines = 0;
     _statsLabel.font = [UIFont fontWithName:@"Roboto" size:12];

examples/objc/AppRTCMobile/ios/ARDVideoCallView.m

@@ -39,8 +39,8 @@ static CGFloat const kStatusBarHeight = 20;
 @synthesize delegate = _delegate;
 
 - (instancetype)initWithFrame:(CGRect)frame {
-  if (self = [super initWithFrame:frame]) {
+  self = [super initWithFrame:frame];
-
+  if (self) {
     _remoteVideoView = [[RTC_OBJC_TYPE(RTCMTLVideoView) alloc] initWithFrame:CGRectZero];
 
     [self addSubview:_remoteVideoView];

examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m

@@ -45,7 +45,8 @@
 - (instancetype)initForRoom:(NSString *)room
                  isLoopback:(BOOL)isLoopback
                    delegate:(id<ARDVideoCallViewControllerDelegate>)delegate {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     ARDSettingsModel *settingsModel = [[ARDSettingsModel alloc] init];
     _delegate = delegate;
 

examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.m

@@ -26,7 +26,8 @@
 @synthesize capturer = _capturer;
 
 - (instancetype)init {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _callbackLogger = [[RTC_OBJC_TYPE(RTCCallbackLogger) alloc] init];
     os_log_t rtc_os_log = os_log_create("com.google.AppRTCMobile", "RTCLog");
     [_callbackLogger start:^(NSString *logMessage) {

examples/objc/AppRTCMobile/mac/APPRTCViewController.m

@@ -72,7 +72,8 @@ static NSUInteger const kBottomViewHeight = 200;
 #pragma mark - Private
 
 - (instancetype)initWithFrame:(NSRect)frame {
-  if (self = [super initWithFrame:frame]) {
+  self = [super initWithFrame:frame];
+  if (self) {
     [self setupViews];
   }
   return self;

modules/desktop_capture/mac/screen_capturer_sck.mm

@@ -370,7 +370,8 @@ std::unique_ptr<DesktopCapturer> CreateScreenCapturerSck(const DesktopCaptureOpt
 }
 
 - (instancetype)initWithCapturer:(webrtc::ScreenCapturerSck*)capturer {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _capturer = capturer;
   }
   return self;

sdk/objc/api/RTCVideoRendererAdapter.mm

@@ -53,7 +53,8 @@ class VideoRendererAdapter
 
 - (instancetype)initWithNativeRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)videoRenderer {
   NSParameterAssert(videoRenderer);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _videoRenderer = videoRenderer;
     _adapter.reset(new webrtc::VideoRendererAdapter(self));
   }

sdk/objc/api/peerconnection/RTCAudioSource.mm

@@ -24,9 +24,10 @@
   RTC_DCHECK(factory);
   RTC_DCHECK(nativeAudioSource);
 
-  if (self = [super initWithFactory:factory
+  self = [super initWithFactory:factory
-                  nativeMediaSource:nativeAudioSource
+              nativeMediaSource:nativeAudioSource
-                               type:RTCMediaSourceTypeAudio]) {
+                           type:RTCMediaSourceTypeAudio];
+  if (self) {
     _nativeAudioSource = nativeAudioSource;
   }
   return self;

sdk/objc/api/peerconnection/RTCCertificate.mm

@@ -28,7 +28,8 @@
 }
 
 - (instancetype)initWithPrivateKey:(NSString *)private_key certificate:(NSString *)certificate {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _private_key = [private_key copy];
     _certificate = [certificate copy];
   }

sdk/objc/api/peerconnection/RTCConfiguration.mm

@@ -72,7 +72,8 @@
 
 - (instancetype)initWithNativeConfiguration:
     (const webrtc::PeerConnectionInterface::RTCConfiguration &)config {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _enableDscp = config.dscp();
     NSMutableArray *iceServers = [NSMutableArray array];
     for (const webrtc::PeerConnectionInterface::IceServer& server : config.servers) {

sdk/objc/api/peerconnection/RTCCryptoOptions.mm

@@ -21,7 +21,8 @@
               srtpEnableAes128Sha1_32CryptoCipher:(BOOL)srtpEnableAes128Sha1_32CryptoCipher
            srtpEnableEncryptedRtpHeaderExtensions:(BOOL)srtpEnableEncryptedRtpHeaderExtensions
                      sframeRequireFrameEncryption:(BOOL)sframeRequireFrameEncryption {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _srtpEnableGcmCryptoSuites = srtpEnableGcmCryptoSuites;
     _srtpEnableAes128Sha1_32CryptoCipher = srtpEnableAes128Sha1_32CryptoCipher;
     _srtpEnableEncryptedRtpHeaderExtensions = srtpEnableEncryptedRtpHeaderExtensions;

sdk/objc/api/peerconnection/RTCDataChannel.mm

@@ -50,7 +50,8 @@ class DataChannelDelegateAdapter : public DataChannelObserver {
 
 - (instancetype)initWithData:(NSData *)data isBinary:(BOOL)isBinary {
   NSParameterAssert(data);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     rtc::CopyOnWriteBuffer buffer(
         reinterpret_cast<const uint8_t*>(data.bytes), data.length);
     _dataBuffer.reset(new webrtc::DataBuffer(buffer, isBinary));
@@ -70,7 +71,8 @@ class DataChannelDelegateAdapter : public DataChannelObserver {
 #pragma mark - Private
 
 - (instancetype)initWithNativeBuffer:(const webrtc::DataBuffer&)nativeBuffer {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _dataBuffer.reset(new webrtc::DataBuffer(nativeBuffer));
   }
   return self;
@@ -167,7 +169,8 @@ class DataChannelDelegateAdapter : public DataChannelObserver {
               nativeDataChannel:
                   (rtc::scoped_refptr<webrtc::DataChannelInterface>)nativeDataChannel {
   NSParameterAssert(nativeDataChannel);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _factory = factory;
     _nativeDataChannel = nativeDataChannel;
     _observer.reset(new webrtc::DataChannelDelegateAdapter(self));

sdk/objc/api/peerconnection/RTCDtmfSender.mm

@@ -64,7 +64,8 @@
 - (instancetype)initWithNativeDtmfSender:
         (rtc::scoped_refptr<webrtc::DtmfSenderInterface>)nativeDtmfSender {
   NSParameterAssert(nativeDtmfSender);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _nativeDtmfSender = nativeDtmfSender;
     RTCLogInfo(
         @"RTC_OBJC_TYPE(RTCDtmfSender)(%p): created DTMF sender: %@", self, self.description);

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

@@ -73,7 +73,8 @@ class ObjCEncodedImageBuffer : public webrtc::EncodedImageBufferInterface {
 }
 
 - (instancetype)initWithNativeEncodedImage:(const webrtc::EncodedImage &)encodedImage {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     // A reference to the encodedData must be stored so that it's kept alive as long
     // self.buffer references its underlying data.
     self.encodedData = encodedImage.GetEncodedData();

sdk/objc/api/peerconnection/RTCFileLogger.mm

@@ -54,7 +54,8 @@ const char *kRTCFileLoggerRotatingLogPrefix = "rotating_log";
                    rotationType:(RTCFileLoggerRotationType)rotationType {
   NSParameterAssert(dirPath.length);
   NSParameterAssert(maxFileSize);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     BOOL isDir = NO;
     NSFileManager *fileManager = [NSFileManager defaultManager];
     if ([fileManager fileExistsAtPath:dirPath isDirectory:&isDir]) {

sdk/objc/api/peerconnection/RTCIceCandidate.mm

@@ -26,7 +26,8 @@
               sdpMLineIndex:(int)sdpMLineIndex
                      sdpMid:(NSString *)sdpMid {
   NSParameterAssert(sdp.length);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _sdpMid = [sdpMid copy];
     _sdpMLineIndex = sdpMLineIndex;
     _sdp = [sdp copy];

sdk/objc/api/peerconnection/RTCIceServer.mm

@@ -84,7 +84,8 @@
                   tlsAlpnProtocols:(NSArray<NSString *> *)tlsAlpnProtocols
                  tlsEllipticCurves:(NSArray<NSString *> *)tlsEllipticCurves {
   NSParameterAssert(urlStrings.count);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _urlStrings = [[NSArray alloc] initWithArray:urlStrings copyItems:YES];
     _username = [username copy];
     _credential = [credential copy];

sdk/objc/api/peerconnection/RTCLegacyStatsReport.mm

@@ -33,7 +33,8 @@
 #pragma mark - Private
 
 - (instancetype)initWithNativeReport:(const webrtc::StatsReport &)nativeReport {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _timestamp = nativeReport.timestamp();
     _type = [NSString stringForStdString:nativeReport.TypeToString()];
     _reportId = [NSString stringForStdString:

sdk/objc/api/peerconnection/RTCMediaConstraints.mm

@@ -37,7 +37,8 @@ NSString *const kRTCMediaConstraintsValueFalse = @(webrtc::MediaConstraints::kVa
     (NSDictionary<NSString *, NSString *> *)mandatory
                          optionalConstraints:
     (NSDictionary<NSString *, NSString *> *)optional {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _mandatory = [[NSDictionary alloc] initWithDictionary:mandatory
                                                 copyItems:YES];
     _optional = [[NSDictionary alloc] initWithDictionary:optional

sdk/objc/api/peerconnection/RTCMediaSource.mm

@@ -24,7 +24,8 @@
                            type:(RTCMediaSourceType)type {
   RTC_DCHECK(factory);
   RTC_DCHECK(nativeMediaSource);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _factory = factory;
     _nativeMediaSource = nativeMediaSource;
     _type = type;

sdk/objc/api/peerconnection/RTCMediaStream.mm

@@ -120,7 +120,8 @@
               nativeMediaStream:
                   (rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream {
   NSParameterAssert(nativeMediaStream);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _factory = factory;
     _signalingThread = factory.signalingThread;
 

sdk/objc/api/peerconnection/RTCMediaStreamTrack.mm

@@ -81,7 +81,8 @@ NSString * const kRTCMediaStreamTrackKindVideo =
                            type:(RTCMediaStreamTrackType)type {
   NSParameterAssert(nativeTrack);
   NSParameterAssert(factory);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _factory = factory;
     _nativeTrack = nativeTrack;
     _type = type;

sdk/objc/api/peerconnection/RTCMetricsSampleInfo.mm

@@ -24,7 +24,8 @@
 
 - (instancetype)initWithNativeSampleInfo:
     (const webrtc::metrics::SampleInfo &)info {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _name = [NSString stringForStdString:info.name];
     _min = info.min;
     _max = info.max;

sdk/objc/api/peerconnection/RTCPeerConnection.mm

@@ -365,7 +365,8 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
   if (!config) {
     return nil;
   }
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _observer.reset(new webrtc::PeerConnectionDelegateAdapter(self));
     _nativeConstraints = constraints.nativeConstraints;
     CopyConstraintsIntoRtcConfiguration(_nativeConstraints.get(), config.get());

sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm

@@ -119,7 +119,8 @@
 }
 
 - (instancetype)initNative {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _networkThread = rtc::Thread::CreateWithSocketServer();
     _networkThread->SetName("network_thread", _networkThread.get());
     BOOL result = _networkThread->Start();

sdk/objc/api/peerconnection/RTCRtcpParameters.mm

@@ -23,7 +23,8 @@
 }
 
 - (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _cname = [NSString stringForStdString:nativeParameters.cname];
     _isReducedSize = nativeParameters.reduced_size;
   }

sdk/objc/api/peerconnection/RTCRtpCapabilities.mm

@@ -28,7 +28,8 @@
 
 - (instancetype)initWithNativeRtpCapabilities:
     (const webrtc::RtpCapabilities &)nativeRtpCapabilities {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     NSMutableArray *codecs = [[NSMutableArray alloc] init];
     for (const auto &codec : nativeRtpCapabilities.codecs) {
       [codecs addObject:[[RTC_OBJC_TYPE(RTCRtpCodecCapability) alloc]

sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm

@@ -33,7 +33,8 @@
 
 - (instancetype)initWithNativeRtpCodecCapability:
     (const webrtc::RtpCodecCapability &)nativeRtpCodecCapability {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     if (nativeRtpCodecCapability.preferred_payload_type) {
       _preferredPayloadType =
           [NSNumber numberWithInt:*nativeRtpCodecCapability.preferred_payload_type];

sdk/objc/api/peerconnection/RTCRtpCodecParameters.mm

@@ -49,7 +49,8 @@ const NSString * const kRTCH264CodecName = @(cricket::kH264CodecName);
 
 - (instancetype)initWithNativeParameters:
     (const webrtc::RtpCodecParameters &)nativeParameters {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _payloadType = nativeParameters.payload_type;
     _name = [NSString stringForStdString:nativeParameters.name];
     switch (nativeParameters.kind) {

sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm

@@ -33,7 +33,8 @@
 
 - (instancetype)initWithNativeParameters:
     (const webrtc::RtpEncodingParameters &)nativeParameters {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     if (!nativeParameters.rid.empty()) {
       _rid = [NSString stringForStdString:nativeParameters.rid];
     }

sdk/objc/api/peerconnection/RTCRtpHeaderExtension.mm

@@ -24,7 +24,8 @@
 }
 
 - (instancetype)initWithNativeParameters:(const webrtc::RtpExtension &)nativeParameters {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _uri = [NSString stringForStdString:nativeParameters.uri];
     _id = nativeParameters.id;
     _encrypted = nativeParameters.encrypt;

sdk/objc/api/peerconnection/RTCRtpHeaderExtensionCapability.mm

@@ -25,7 +25,8 @@
 
 - (instancetype)initWithNativeRtpHeaderExtensionCapability:
     (const webrtc::RtpHeaderExtensionCapability &)nativeRtpHeaderExtensionCapability {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _uri = [NSString stringForStdString:nativeRtpHeaderExtensionCapability.uri];
     if (nativeRtpHeaderExtensionCapability.preferred_id) {
       _preferredId = [NSNumber numberWithInt:*nativeRtpHeaderExtensionCapability.preferred_id];

sdk/objc/api/peerconnection/RTCRtpParameters.mm

@@ -32,7 +32,8 @@
 
 - (instancetype)initWithNativeParameters:
     (const webrtc::RtpParameters &)nativeParameters {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _transactionId = [NSString stringForStdString:nativeParameters.transaction_id];
     _rtcp =
         [[RTC_OBJC_TYPE(RTCRtcpParameters) alloc] initWithNativeParameters:nativeParameters.rtcp];

sdk/objc/api/peerconnection/RTCRtpReceiver.mm

@@ -117,7 +117,8 @@ void RtpReceiverDelegateAdapter::OnFirstPacketReceived(
 - (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
               nativeRtpReceiver:
                   (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _factory = factory;
     _nativeRtpReceiver = nativeRtpReceiver;
     RTCLogInfo(@"RTC_OBJC_TYPE(RTCRtpReceiver)(%p): created receiver: %@", self, self.description);

sdk/objc/api/peerconnection/RTCRtpSender.mm

@@ -113,7 +113,8 @@
                 nativeRtpSender:(rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
   NSParameterAssert(factory);
   NSParameterAssert(nativeRtpSender);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _factory = factory;
     _nativeRtpSender = nativeRtpSender;
     if (_nativeRtpSender->media_type() == cricket::MEDIA_TYPE_AUDIO) {

sdk/objc/api/peerconnection/RTCRtpSource.mm

@@ -65,7 +65,8 @@
 }
 
 - (instancetype)initWithNativeRtpSource:(const webrtc::RtpSource &)nativeRtpSource {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _nativeRtpSource = nativeRtpSource;
   }
   return self;

sdk/objc/api/peerconnection/RTCRtpTransceiver.mm

@@ -29,7 +29,8 @@ NSString *const kRTCRtpTransceiverErrorDomain = @"org.webrtc.RTCRtpTranceiver";
 @synthesize sendEncodings = _sendEncodings;
 
 - (instancetype)init {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _direction = RTCRtpTransceiverDirectionSendRecv;
   }
   return self;
@@ -166,7 +167,8 @@ NSString *const kRTCRtpTransceiverErrorDomain = @"org.webrtc.RTCRtpTranceiver";
                (rtc::scoped_refptr<webrtc::RtpTransceiverInterface>)nativeRtpTransceiver {
   NSParameterAssert(factory);
   NSParameterAssert(nativeRtpTransceiver);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _factory = factory;
     _nativeRtpTransceiver = nativeRtpTransceiver;
     _sender = [[RTC_OBJC_TYPE(RTCRtpSender) alloc] initWithFactory:_factory

sdk/objc/api/peerconnection/RTCSessionDescription.mm

@@ -31,7 +31,8 @@
 }
 
 - (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _type = type;
     _sdp = [sdp copy];
   }

sdk/objc/api/peerconnection/RTCStatisticsReport.mm

@@ -114,7 +114,8 @@ NSObject *ValueFromStatsAttribute(const Attribute &attribute) {
 @synthesize values = _values;
 
 - (instancetype)initWithStatistics:(const webrtc::RTCStats &)statistics {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _id = [NSString stringForStdString:statistics.id()];
     _timestamp_us = statistics.timestamp().us();
     _type = [NSString stringWithCString:statistics.type() encoding:NSUTF8StringEncoding];
@@ -161,7 +162,8 @@ NSObject *ValueFromStatsAttribute(const Attribute &attribute) {
 @implementation RTC_OBJC_TYPE (RTCStatisticsReport) (Private)
 
 - (instancetype)initWithReport : (const webrtc::RTCStatsReport &)report {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _timestamp_us = report.timestamp().us();
 
     NSMutableDictionary *statisticsById =

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

@@ -16,7 +16,8 @@
 (Private)
 
     - (instancetype)initWithNativeVideoCodec : (const webrtc::VideoCodec *)videoCodec {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     if (videoCodec) {
       const char *codecName = CodecTypeToPayloadString(videoCodec->codecType);
       self.name = [NSString stringWithUTF8String:codecName];
@@ -31,7 +32,6 @@
       self.mode = (RTCVideoCodecMode)videoCodec->mode;
     }
   }
-
   return self;
 }
 

sdk/objc/api/peerconnection/RTCVideoSource.mm

@@ -33,9 +33,10 @@ static webrtc::ObjCVideoTrackSource *getObjCVideoSource(
                   (rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource {
   RTC_DCHECK(factory);
   RTC_DCHECK(nativeVideoSource);
-  if (self = [super initWithFactory:factory
+  self = [super initWithFactory:factory
-                  nativeMediaSource:nativeVideoSource
+              nativeMediaSource:nativeVideoSource
-                               type:RTCMediaSourceTypeVideo]) {
+                           type:RTCMediaSourceTypeVideo];
+  if (self) {
     _nativeVideoSource = nativeVideoSource;
   }
   return self;

sdk/objc/api/peerconnection/RTCVideoTrack.mm

@@ -45,7 +45,8 @@
   NSParameterAssert(factory);
   NSParameterAssert(nativeMediaTrack);
   NSParameterAssert(type == RTCMediaStreamTrackTypeVideo);
-  if (self = [super initWithFactory:factory nativeTrack:nativeMediaTrack type:type]) {
+  self = [super initWithFactory:factory nativeTrack:nativeMediaTrack type:type];
+  if (self) {
     _adapters = [NSMutableArray array];
     _workerThread = factory.workerThread;
   }

sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.mm

@@ -20,10 +20,10 @@
 @implementation RTC_OBJC_TYPE (RTCI420Buffer)
 
 - (instancetype)initWithWidth:(int)width height:(int)height {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _i420Buffer = webrtc::I420Buffer::Create(width, height);
   }
-
   return self;
 }
 
@@ -32,7 +32,8 @@
                         dataY:(const uint8_t *)dataY
                         dataU:(const uint8_t *)dataU
                         dataV:(const uint8_t *)dataV {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _i420Buffer = webrtc::I420Buffer::Copy(
         width, height, dataY, width, dataU, (width + 1) / 2, dataV, (width + 1) / 2);
   }
@@ -44,18 +45,18 @@
                       strideY:(int)strideY
                       strideU:(int)strideU
                       strideV:(int)strideV {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _i420Buffer = webrtc::I420Buffer::Create(width, height, strideY, strideU, strideV);
   }
-
   return self;
 }
 
 - (instancetype)initWithFrameBuffer:(rtc::scoped_refptr<webrtc::I420BufferInterface>)i420Buffer {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _i420Buffer = i420Buffer;
   }
-
   return self;
 }
 

sdk/objc/base/RTCVideoCapturer.m

@@ -15,7 +15,8 @@
 @synthesize delegate = _delegate;
 
 - (instancetype)initWithDelegate:(id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>)delegate {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _delegate = delegate;
   }
   return self;

sdk/objc/base/RTCVideoCodecInfo.m

@@ -21,7 +21,8 @@
 
 - (instancetype)initWithName:(NSString *)name
                   parameters:(nullable NSDictionary<NSString *, NSString *> *)parameters {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _name = name;
     _parameters = (parameters ? parameters : @{});
   }

sdk/objc/base/RTCVideoEncoderQpThresholds.m

@@ -16,7 +16,8 @@
 @synthesize high = _high;
 
 - (instancetype)initWithThresholdsLow:(NSInteger)low high:(NSInteger)high {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _low = low;
     _high = high;
   }

sdk/objc/base/RTCVideoFrame.mm

@@ -66,12 +66,12 @@
 - (instancetype)initWithBuffer:(id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)>)buffer
                       rotation:(RTCVideoRotation)rotation
                    timeStampNs:(int64_t)timeStampNs {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _buffer = buffer;
     _rotation = rotation;
     _timeStampNs = timeStampNs;
   }
-
   return self;
 }
 

sdk/objc/components/audio/RTCAudioSession.mm

@@ -77,7 +77,8 @@ ABSL_CONST_INIT thread_local bool mutex_locked = false;
 
 /** This initializer provides a way for unit tests to inject a fake/mock audio session. */
 - (instancetype)initWithAudioSession:(id)audioSession {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _session = audioSession;
 
     NSNotificationCenter *center = [NSNotificationCenter defaultCenter];

sdk/objc/components/audio/RTCAudioSessionConfiguration.m

@@ -53,7 +53,8 @@ static RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *gWebRTCConfiguration = nil;
 @synthesize outputNumberOfChannels = _outputNumberOfChannels;
 
 - (instancetype)init {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     // Use a category which supports simultaneous recording and playback.
     // By default, using this category implies that our app’s audio is
     // nonmixable, hence activating the session will interrupt any other

sdk/objc/components/audio/RTCNativeAudioSessionDelegateAdapter.mm

@@ -20,7 +20,8 @@
 
 - (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer {
   RTC_DCHECK(observer);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _observer = observer;
   }
   return self;

sdk/objc/components/capturer/RTCCameraVideoCapturer.m

@@ -65,7 +65,8 @@ const int64_t kNanosecondsPerSecond = 1000000000;
 // This initializer is used for testing.
 - (instancetype)initWithDelegate:(__weak id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>)delegate
                   captureSession:(AVCaptureSession *)captureSession {
-  if (self = [super initWithDelegate:delegate]) {
+  self = [super initWithDelegate:delegate];
+  if (self) {
     // Create the capture session and all relevant inputs and outputs. We need
     // to do this in init because the application may want the capture session
     // before we start the capturer for e.g. AVCapturePreviewLayer. All objects

sdk/objc/components/network/RTCNetworkMonitor.mm

@@ -54,7 +54,8 @@ rtc::AdapterType AdapterTypeFromInterfaceType(nw_interface_type_t interfaceType)
 
 - (instancetype)initWithObserver:(webrtc::NetworkMonitorObserver *)observer {
   RTC_DCHECK(observer);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _observer = observer;
     if (@available(iOS 12, *)) {
       _pathMonitor = nw_path_monitor_create();

sdk/objc/components/renderer/metal/RTCMTLRenderer.mm

@@ -115,7 +115,8 @@ static const NSInteger kMaxInflightBuffers = 1;
 @synthesize rotationOverride = _rotationOverride;
 
 - (instancetype)init {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _inflight_semaphore = dispatch_semaphore_create(kMaxInflightBuffers);
   }
 

sdk/objc/components/renderer/opengl/RTCDisplayLinkTimer.m

@@ -19,7 +19,8 @@
 
 - (instancetype)initWithTimerHandler:(void (^)(void))timerHandler {
   NSParameterAssert(timerHandler);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _timerHandler = timerHandler;
     _displayLink =
         [CADisplayLink displayLinkWithTarget:self

sdk/objc/components/renderer/opengl/RTCEAGLVideoView.m

@@ -69,7 +69,8 @@
 }
 
 - (instancetype)initWithFrame:(CGRect)frame shader:(id<RTC_OBJC_TYPE(RTCVideoViewShading)>)shader {
-  if (self = [super initWithFrame:frame]) {
+  self = [super initWithFrame:frame];
+  if (self) {
     _shader = shader;
     if (![self configure]) {
       return nil;
@@ -80,7 +81,8 @@
 
 - (instancetype)initWithCoder:(NSCoder *)aDecoder
                        shader:(id<RTC_OBJC_TYPE(RTCVideoViewShading)>)shader {
-  if (self = [super initWithCoder:aDecoder]) {
+  self = [super initWithCoder:aDecoder];
+  if (self) {
     _shader = shader;
     if (![self configure]) {
       return nil;

sdk/objc/components/renderer/opengl/RTCI420TextureCache.mm

@@ -46,7 +46,8 @@ static const GLsizei kNumTextures = kNumTexturesPerSet * kNumTextureSets;
 }
 
 - (instancetype)initWithContext:(GlContextType *)context {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _hasUnpackRowLength = (context.API == kEAGLRenderingAPIOpenGLES3);
     glPixelStorei(GL_UNPACK_ALIGNMENT, 1);
 

sdk/objc/components/renderer/opengl/RTCNV12TextureCache.m

@@ -29,7 +29,8 @@
 }
 
 - (instancetype)initWithContext:(EAGLContext *)context {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     CVReturn ret = CVOpenGLESTextureCacheCreate(
         kCFAllocatorDefault, NULL,
 #if COREVIDEO_USE_EAGLCONTEXT_CLASS_IN_API

sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm

@@ -90,7 +90,8 @@ NSString *MaxSupportedProfileLevelConstrainedHigh() {
 @synthesize hexString = _hexString;
 
 - (instancetype)initWithHexString:(NSString *)hexString {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     self.hexString = hexString;
 
     absl::optional<webrtc::H264ProfileLevelId> profile_level_id =
@@ -104,7 +105,8 @@ NSString *MaxSupportedProfileLevelConstrainedHigh() {
 }
 
 - (instancetype)initWithProfile:(RTCH264Profile)profile level:(RTCH264Level)level {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     self.profile = profile;
     self.level = level;
 

sdk/objc/components/video_codec/RTCVideoEncoderH264.mm

@@ -344,7 +344,8 @@ NSUInteger GetMaxSampleRate(const webrtc::H264ProfileLevelId &profile_level_id)
 // conditions, 0.95 seems to give us better overall bitrate over long periods
 // of time.
 - (instancetype)initWithCodecInfo:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)codecInfo {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _codecInfo = codecInfo;
     _bitrateAdjuster.reset(new webrtc::BitrateAdjuster(.5, .95));
     _packetizationMode = RTCH264PacketizationModeNonInterleaved;

sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.mm

@@ -62,7 +62,8 @@
                          cropHeight:(int)cropHeight
                               cropX:(int)cropX
                               cropY:(int)cropY {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _width = adaptedWidth;
     _height = adaptedHeight;
     _pixelBuffer = pixelBuffer;

sdk/objc/native/src/objc_audio_device_delegate.mm

@@ -75,7 +75,8 @@ class AudioDeviceDelegateImpl final : public rtc::RefCountedNonVirtual<AudioDevi
                     (rtc::scoped_refptr<webrtc::objc_adm::ObjCAudioDeviceModule>)audioDeviceModule
                         audioDeviceThread:(rtc::Thread*)thread {
   RTC_DCHECK_RUN_ON(thread);
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     impl_ = rtc::make_ref_counted<AudioDeviceDelegateImpl>(audioDeviceModule, thread);
     preferredInputSampleRate_ = kPreferredInputSampleRate;
     preferredInputIOBufferDuration_ = kPeferredInputIOBufferDuration;

sdk/objc/unittests/RTCAudioSessionTest.mm

@@ -53,7 +53,8 @@
 @synthesize outputVolume = _outputVolume;
 
 - (instancetype)init {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _outputVolume = -1;
   }
   return self;
@@ -98,7 +99,8 @@
 @implementation RTCTestRemoveOnDeallocDelegate
 
 - (instancetype)init {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
     [session addDelegate:self];
   }

sdk/objc/unittests/RTCPeerConnectionFactory_xctest.m

@@ -42,7 +42,8 @@
 
 - (instancetype)initWithSupportedCodecs:
     (nonnull NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)supportedCodecs {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     _supportedCodecs = supportedCodecs;
   }
   return self;

sdk/objc/unittests/avformatmappertests.mm

@@ -53,7 +53,8 @@ static cricket::VideoFormat expectedFormat =
 - (instancetype)initWithMediaSubtype:(FourCharCode)subtype
                               minFps:(float)minFps
                               maxFps:(float)maxFps {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     CMVideoFormatDescriptionCreate(nil, subtype, kFormatWidth, kFormatHeight,
                                    nil, &_format);
     // We can use OCMock for the range.

test/mac/video_renderer_mac.mm

@@ -35,7 +35,8 @@
   static NSInteger nextYOrigin_;
 
 - (id)initWithTitle:(NSString *)title width:(int)width height:(int)height {
-  if (self = [super init]) {
+  self = [super init];
+  if (self) {
     title_ = title;
     width_ = width;
     height_ = height;