Reformat the WebRTC code base
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}diff --git a/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h b/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h
index 8be8aad..f163b5a 100644
--- a/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h
+++ b/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h
@@ -77,14 +77,12 @@
- (NSError *)configurationErrorWithDescription:(NSString *)description;
// Properties and methods for tests.
-@property(nonatomic, readonly)
- std::vector<__weak id<RTCAudioSessionDelegate> > delegates;
+@property(nonatomic, readonly) std::vector<__weak id<RTCAudioSessionDelegate> > delegates;
- (void)notifyDidBeginInterruption;
-- (void)notifyDidEndInterruptionWithShouldResumeSession:
- (BOOL)shouldResumeSession;
+- (void)notifyDidEndInterruptionWithShouldResumeSession:(BOOL)shouldResumeSession;
- (void)notifyDidChangeRouteWithReason:(AVAudioSessionRouteChangeReason)reason
- previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
+ previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
- (void)notifyMediaServicesWereLost;
- (void)notifyMediaServicesWereReset;
- (void)notifyDidChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
diff --git a/sdk/objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.h b/sdk/objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.h
index 487bab1..7fb2184 100644
--- a/sdk/objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.h
+++ b/sdk/objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.h
@@ -26,8 +26,7 @@
/** |observer| is a raw pointer and should be kept alive
* for this object's lifetime.
*/
-- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer
- NS_DESIGNATED_INITIALIZER;
+- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer NS_DESIGNATED_INITIALIZER;
@end
diff --git a/sdk/objc/Framework/Classes/Common/NSString+StdString.h b/sdk/objc/Framework/Classes/Common/NSString+StdString.h
index 8bf6cc9..7f51a9f 100644
--- a/sdk/objc/Framework/Classes/Common/NSString+StdString.h
+++ b/sdk/objc/Framework/Classes/Common/NSString+StdString.h
@@ -19,7 +19,7 @@
@property(nonatomic, readonly) std::string stdString;
+ (std::string)stdStringForString:(NSString *)nsString;
-+ (NSString *)stringForStdString:(const std::string&)stdString;
++ (NSString *)stringForStdString:(const std::string &)stdString;
@end
diff --git a/sdk/objc/Framework/Classes/Common/helpers.h b/sdk/objc/Framework/Classes/Common/helpers.h
index ca56d51..7fd9b34 100644
--- a/sdk/objc/Framework/Classes/Common/helpers.h
+++ b/sdk/objc/Framework/Classes/Common/helpers.h
@@ -18,7 +18,7 @@
bool CheckAndLogError(BOOL success, NSError* error);
-NSString *NSStringFromStdString(const std::string& stdString);
+NSString* NSStringFromStdString(const std::string& stdString);
std::string StdStringFromNSString(NSString* nsString);
// Return thread ID as a string.
diff --git a/sdk/objc/Framework/Classes/Metal/RTCMTLRGBRenderer.h b/sdk/objc/Framework/Classes/Metal/RTCMTLRGBRenderer.h
index de27055..9db422c 100644
--- a/sdk/objc/Framework/Classes/Metal/RTCMTLRGBRenderer.h
+++ b/sdk/objc/Framework/Classes/Metal/RTCMTLRGBRenderer.h
@@ -13,7 +13,8 @@
#import "RTCMTLRenderer.h"
/** @abstract RGB/BGR renderer.
- * @discussion This renderer handles both kCVPixelFormatType_32BGRA and kCVPixelFormatType_32ARGB.
+ * @discussion This renderer handles both kCVPixelFormatType_32BGRA and
+ * kCVPixelFormatType_32ARGB.
*/
NS_AVAILABLE(10_11, 9_0)
@interface RTCMTLRGBRenderer : RTCMTLRenderer
diff --git a/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h b/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h
index bd1b6b7..f4a293b 100644
--- a/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h
+++ b/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h
@@ -21,7 +21,7 @@
/**
* Protocol defining ability to render RTCVideoFrame in Metal enabled views.
*/
-@protocol RTCMTLRenderer<NSObject>
+@protocol RTCMTLRenderer <NSObject>
/**
* Method to be implemented to perform actual rendering of the provided frame.
@@ -49,7 +49,7 @@
* Implementation of RTCMTLRenderer protocol.
*/
NS_AVAILABLE(10_11, 9_0)
-@interface RTCMTLRenderer : NSObject<RTCMTLRenderer>
+@interface RTCMTLRenderer : NSObject <RTCMTLRenderer>
/** @abstract A wrapped RTCVideoRotation, or nil.
@discussion When not nil, the rotation of the actual frame is ignored when rendering.
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource+Private.h
index 43aff02..752eb66 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource+Private.h
@@ -22,11 +22,11 @@
/** Initialize an RTCAudioSource from a native AudioSourceInterface. */
- (instancetype)initWithNativeAudioSource:
- (rtc::scoped_refptr<webrtc::AudioSourceInterface>)nativeAudioSource
+ (rtc::scoped_refptr<webrtc::AudioSourceInterface>)nativeAudioSource
NS_DESIGNATED_INITIALIZER;
- (instancetype)initWithNativeMediaSource:
- (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+ (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
type:(RTCMediaSourceType)type NS_UNAVAILABLE;
@end
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack+Private.h
index 176d03f..7a2f2c8 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack+Private.h
@@ -18,8 +18,7 @@
@interface RTCAudioTrack ()
/** 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:(RTCPeerConnectionFactory *)factory
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Native.h b/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Native.h
index f539d1c..64653e4 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Native.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Native.h
@@ -18,9 +18,10 @@
/** Optional TurnCustomizer.
* With this class one can modify outgoing TURN messages.
- * The object passed in must remain valid until PeerConnection::Close() is called.
+ * The object passed in must remain valid until PeerConnection::Close() is
+ * called.
*/
-@property(nonatomic, nullable) webrtc::TurnCustomizer *turnCustomizer;
+@property(nonatomic, nullable) webrtc::TurnCustomizer* turnCustomizer;
@end
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Private.h
index c572d63..ad57336 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Private.h
@@ -16,43 +16,43 @@
@interface RTCConfiguration ()
-+ (webrtc::PeerConnectionInterface::IceTransportsType)
- nativeTransportsTypeForTransportPolicy:(RTCIceTransportPolicy)policy;
++ (webrtc::PeerConnectionInterface::IceTransportsType)nativeTransportsTypeForTransportPolicy:
+ (RTCIceTransportPolicy)policy;
+ (RTCIceTransportPolicy)transportPolicyForTransportsType:
- (webrtc::PeerConnectionInterface::IceTransportsType)nativeType;
+ (webrtc::PeerConnectionInterface::IceTransportsType)nativeType;
+ (NSString *)stringForTransportPolicy:(RTCIceTransportPolicy)policy;
+ (webrtc::PeerConnectionInterface::BundlePolicy)nativeBundlePolicyForPolicy:
- (RTCBundlePolicy)policy;
+ (RTCBundlePolicy)policy;
+ (RTCBundlePolicy)bundlePolicyForNativePolicy:
- (webrtc::PeerConnectionInterface::BundlePolicy)nativePolicy;
+ (webrtc::PeerConnectionInterface::BundlePolicy)nativePolicy;
+ (NSString *)stringForBundlePolicy:(RTCBundlePolicy)policy;
+ (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativeRtcpMuxPolicyForPolicy:
- (RTCRtcpMuxPolicy)policy;
+ (RTCRtcpMuxPolicy)policy;
+ (RTCRtcpMuxPolicy)rtcpMuxPolicyForNativePolicy:
- (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativePolicy;
+ (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativePolicy;
+ (NSString *)stringForRtcpMuxPolicy:(RTCRtcpMuxPolicy)policy;
-+ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)
- nativeTcpCandidatePolicyForPolicy:(RTCTcpCandidatePolicy)policy;
++ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativeTcpCandidatePolicyForPolicy:
+ (RTCTcpCandidatePolicy)policy;
+ (RTCTcpCandidatePolicy)tcpCandidatePolicyForNativePolicy:
- (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativePolicy;
+ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativePolicy;
+ (NSString *)stringForTcpCandidatePolicy:(RTCTcpCandidatePolicy)policy;
-+ (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)
- nativeCandidateNetworkPolicyForPolicy:(RTCCandidateNetworkPolicy)policy;
++ (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativeCandidateNetworkPolicyForPolicy:
+ (RTCCandidateNetworkPolicy)policy;
+ (RTCCandidateNetworkPolicy)candidateNetworkPolicyForNativePolicy:
- (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativePolicy;
+ (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativePolicy;
+ (NSString *)stringForCandidateNetworkPolicy:(RTCCandidateNetworkPolicy)policy;
@@ -71,7 +71,7 @@
- (nullable webrtc::PeerConnectionInterface::RTCConfiguration *)createNativeConfiguration;
- (instancetype)initWithNativeConfiguration:
- (const webrtc::PeerConnectionInterface::RTCConfiguration &)config NS_DESIGNATED_INITIALIZER;
+ (const webrtc::PeerConnectionInterface::RTCConfiguration &)config NS_DESIGNATED_INITIALIZER;
@end
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCDataChannel+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCDataChannel+Private.h
index ca806ca..5ab308b 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCDataChannel+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCDataChannel+Private.h
@@ -24,23 +24,22 @@
@property(nonatomic, readonly) const webrtc::DataBuffer *nativeDataBuffer;
/** Initialize an RTCDataBuffer from a native DataBuffer. */
-- (instancetype)initWithNativeBuffer:(const webrtc::DataBuffer&)nativeBuffer;
+- (instancetype)initWithNativeBuffer:(const webrtc::DataBuffer &)nativeBuffer;
@end
-
@interface RTCDataChannel ()
/** Initialize an RTCDataChannel from a native DataChannelInterface. */
- (instancetype)initWithNativeDataChannel:
- (rtc::scoped_refptr<webrtc::DataChannelInterface>)nativeDataChannel
+ (rtc::scoped_refptr<webrtc::DataChannelInterface>)nativeDataChannel
NS_DESIGNATED_INITIALIZER;
-+ (webrtc::DataChannelInterface::DataState)
- nativeDataChannelStateForState:(RTCDataChannelState)state;
++ (webrtc::DataChannelInterface::DataState)nativeDataChannelStateForState:
+ (RTCDataChannelState)state;
+ (RTCDataChannelState)dataChannelStateForNativeState:
- (webrtc::DataChannelInterface::DataState)nativeState;
+ (webrtc::DataChannelInterface::DataState)nativeState;
+ (NSString *)stringForState:(RTCDataChannelState)state;
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCIceCandidate+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCIceCandidate+Private.h
index 6c47a39..47e934c 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCIceCandidate+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCIceCandidate+Private.h
@@ -22,15 +22,13 @@
* 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
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCIceServer+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCIceServer+Private.h
index 19a845b..0f6b7b7 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCIceServer+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCIceServer+Private.h
@@ -20,12 +20,10 @@
* 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
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCIntervalRange+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCIntervalRange+Private.h
index a80c99a..690daf3 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCIntervalRange+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCIntervalRange+Private.h
@@ -16,12 +16,10 @@
@interface RTCIntervalRange ()
-@property(nonatomic, readonly)
- std::unique_ptr<rtc::IntervalRange> nativeIntervalRange;
+@property(nonatomic, readonly) std::unique_ptr<rtc::IntervalRange> nativeIntervalRange;
- (instancetype)initWithNativeIntervalRange:(const rtc::IntervalRange &)config;
@end
NS_ASSUME_NONNULL_END
-
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints+Private.h
index 1955c8c..75cd79b 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints+Private.h
@@ -20,9 +20,8 @@
public:
virtual ~MediaConstraints();
MediaConstraints();
- MediaConstraints(
- const MediaConstraintsInterface::Constraints& mandatory,
- const MediaConstraintsInterface::Constraints& optional);
+ MediaConstraints(const MediaConstraintsInterface::Constraints& mandatory,
+ const MediaConstraintsInterface::Constraints& optional);
virtual const Constraints& GetMandatory() const;
virtual const Constraints& GetOptional() const;
@@ -33,7 +32,6 @@
} // namespace webrtc
-
NS_ASSUME_NONNULL_BEGIN
@interface RTCMediaConstraints ()
@@ -45,9 +43,8 @@
- (std::unique_ptr<webrtc::MediaConstraints>)nativeConstraints;
/** Return a native Constraints object representing these constraints */
-+ (webrtc::MediaConstraintsInterface::Constraints)
- nativeConstraintsForConstraints:
- (NSDictionary<NSString *, NSString *> *)constraints;
++ (webrtc::MediaConstraintsInterface::Constraints)nativeConstraintsForConstraints:
+ (NSDictionary<NSString*, NSString*>*)constraints;
@end
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource+Private.h
index 1656d16..f638249 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource+Private.h
@@ -21,19 +21,15 @@
@interface RTCMediaSource ()
-@property(nonatomic, readonly)
- rtc::scoped_refptr<webrtc::MediaSourceInterface> nativeMediaSource;
+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::MediaSourceInterface> nativeMediaSource;
- (instancetype)initWithNativeMediaSource:
- (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
- type:(RTCMediaSourceType)type
- NS_DESIGNATED_INITIALIZER;
+ (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;
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStream+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStream+Private.h
index 1d0847c..f95d15c 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStream+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStream+Private.h
@@ -20,16 +20,14 @@
* MediaStreamInterface representation of this RTCMediaStream object. This is
* needed to pass to the underlying C++ APIs.
*/
-@property(nonatomic, readonly)
- rtc::scoped_refptr<webrtc::MediaStreamInterface> nativeMediaStream;
+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::MediaStreamInterface> nativeMediaStream;
/** Initialize an RTCMediaStream with an id. */
-- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
- streamId:(NSString *)streamId;
+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory streamId:(NSString *)streamId;
/** Initialize an RTCMediaStream from a native MediaStreamInterface. */
- (instancetype)initWithNativeMediaStream:
- (rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream;
+ (rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream;
@end
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStreamTrack+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStreamTrack+Private.h
index bcf7a5f..bb24216 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStreamTrack+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStreamTrack+Private.h
@@ -25,27 +25,25 @@
* 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)initWithNativeTrack:
- (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
- type:(RTCMediaStreamTrackType)type
- NS_DESIGNATED_INITIALIZER;
+ (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+ type:(RTCMediaStreamTrackType)type NS_DESIGNATED_INITIALIZER;
- (instancetype)initWithNativeTrack:
- (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack;
+ (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack;
- (BOOL)isEqualToTrack:(RTCMediaStreamTrack *)track;
+ (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
- (RTCMediaStreamTrackState)state;
+ (RTCMediaStreamTrackState)state;
+ (RTCMediaStreamTrackState)trackStateForNativeState:
- (webrtc::MediaStreamTrackInterface::TrackState)nativeState;
+ (webrtc::MediaStreamTrackInterface::TrackState)nativeState;
+ (NSString *)stringForState:(RTCMediaStreamTrackState)state;
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCMetricsSampleInfo+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCMetricsSampleInfo+Private.h
index 48451df..ceaca8f 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCMetricsSampleInfo+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCMetricsSampleInfo+Private.h
@@ -20,8 +20,7 @@
@interface RTCMetricsSampleInfo ()
/** Initialize an RTCMetricsSampleInfo object from native SampleInfo. */
-- (instancetype)initWithNativeSampleInfo:
- (const webrtc::metrics::SampleInfo &)info;
+- (instancetype)initWithNativeSampleInfo:(const webrtc::metrics::SampleInfo &)info;
@end
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Private.h
index eae72fb..f533605 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Private.h
@@ -21,13 +21,11 @@
* id<RTCPeerConnectionDelegate> and call methods on that interface.
*/
class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
-
public:
PeerConnectionDelegateAdapter(RTCPeerConnection *peerConnection);
virtual ~PeerConnectionDelegateAdapter();
- void OnSignalingChange(
- PeerConnectionInterface::SignalingState new_state) override;
+ void OnSignalingChange(PeerConnectionInterface::SignalingState new_state) override;
void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) override;
@@ -35,77 +33,68 @@
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 OnIceGatheringChange(
- PeerConnectionInterface::IceGatheringState new_state) override;
+ void OnIceGatheringChange(PeerConnectionInterface::IceGatheringState new_state) override;
void OnIceCandidate(const IceCandidateInterface *candidate) override;
- void OnIceCandidatesRemoved(
- const std::vector<cricket::Candidate>& candidates) override;
+ void OnIceCandidatesRemoved(const std::vector<cricket::Candidate> &candidates) 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;
private:
__weak RTCPeerConnection *peer_connection_;
};
-} // namespace webrtc
-
+} // namespace webrtc
@interface RTCPeerConnection ()
/** The native PeerConnectionInterface created during construction. */
-@property(nonatomic, readonly)
- rtc::scoped_refptr<webrtc::PeerConnectionInterface> nativePeerConnection;
+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::PeerConnectionInterface>
+ nativePeerConnection;
/** Initialize an RTCPeerConnection with a configuration, constraints, and
* delegate.
*/
-- (instancetype)initWithFactory:
- (RTCPeerConnectionFactory *)factory
- configuration:
- (RTCConfiguration *)configuration
- constraints:
- (RTCMediaConstraints *)constraints
- delegate:
- (nullable id<RTCPeerConnectionDelegate>)delegate
+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
+ configuration:(RTCConfiguration *)configuration
+ constraints:(RTCMediaConstraints *)constraints
+ delegate:(nullable id<RTCPeerConnectionDelegate>)delegate
NS_DESIGNATED_INITIALIZER;
+ (webrtc::PeerConnectionInterface::SignalingState)nativeSignalingStateForState:
- (RTCSignalingState)state;
+ (RTCSignalingState)state;
+ (RTCSignalingState)signalingStateForNativeState:
- (webrtc::PeerConnectionInterface::SignalingState)nativeState;
+ (webrtc::PeerConnectionInterface::SignalingState)nativeState;
+ (NSString *)stringForSignalingState:(RTCSignalingState)state;
-+ (webrtc::PeerConnectionInterface::IceConnectionState)
- nativeIceConnectionStateForState:(RTCIceConnectionState)state;
++ (webrtc::PeerConnectionInterface::IceConnectionState)nativeIceConnectionStateForState:
+ (RTCIceConnectionState)state;
+ (RTCIceConnectionState)iceConnectionStateForNativeState:
- (webrtc::PeerConnectionInterface::IceConnectionState)nativeState;
+ (webrtc::PeerConnectionInterface::IceConnectionState)nativeState;
+ (NSString *)stringForIceConnectionState:(RTCIceConnectionState)state;
-+ (webrtc::PeerConnectionInterface::IceGatheringState)
- nativeIceGatheringStateForState:(RTCIceGatheringState)state;
++ (webrtc::PeerConnectionInterface::IceGatheringState)nativeIceGatheringStateForState:
+ (RTCIceGatheringState)state;
+ (RTCIceGatheringState)iceGatheringStateForNativeState:
- (webrtc::PeerConnectionInterface::IceGatheringState)nativeState;
+ (webrtc::PeerConnectionInterface::IceGatheringState)nativeState;
+ (NSString *)stringForIceGatheringState:(RTCIceGatheringState)state;
-+ (webrtc::PeerConnectionInterface::StatsOutputLevel)
- nativeStatsOutputLevelForLevel:(RTCStatsOutputLevel)level;
++ (webrtc::PeerConnectionInterface::StatsOutputLevel)nativeStatsOutputLevelForLevel:
+ (RTCStatsOutputLevel)level;
@end
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Private.h
index 1e5d179..50944ee 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Private.h
@@ -23,7 +23,8 @@
* C++ APIs.
*/
@property(nonatomic, readonly)
- rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> nativeFactory;
+ rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
+ nativeFactory;
@end
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters+Private.h
index f33a8cf..c6ca462 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters+Private.h
@@ -20,8 +20,7 @@
@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;
@end
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters+Private.h
index 673a485..c64501f 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters+Private.h
@@ -20,8 +20,7 @@
@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;
@end
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters+Private.h
index 4a0daaf..41d1b7c 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters+Private.h
@@ -20,8 +20,7 @@
@property(nonatomic, readonly) webrtc::RtpParameters nativeParameters;
/** Initialize the object with a native RtpParameters structure. */
-- (instancetype)initWithNativeParameters:
- (const webrtc::RtpParameters &)nativeParameters;
+- (instancetype)initWithNativeParameters:(const webrtc::RtpParameters &)nativeParameters;
@end
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCRtpReceiver+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCRtpReceiver+Private.h
index 63bdfa3..73f68f4 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCRtpReceiver+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCRtpReceiver+Private.h
@@ -30,12 +30,11 @@
@interface RTCRtpReceiver ()
-@property(nonatomic, readonly)
- rtc::scoped_refptr<webrtc::RtpReceiverInterface> nativeRtpReceiver;
+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::RtpReceiverInterface> nativeRtpReceiver;
/** Initialize an RTCRtpReceiver with a native RtpReceiverInterface. */
- (instancetype)initWithNativeRtpReceiver:
- (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver
+ (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver
NS_DESIGNATED_INITIALIZER;
+ (RTCRtpMediaType)mediaTypeForNativeMediaType:(cricket::MediaType)nativeMediaType;
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCRtpSender+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCRtpSender+Private.h
index e70be83..7164bca 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCRtpSender+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCRtpSender+Private.h
@@ -16,13 +16,11 @@
@interface 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)initWithNativeRtpSender:
- (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender
- NS_DESIGNATED_INITIALIZER;
+ (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender NS_DESIGNATED_INITIALIZER;
@end
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCSessionDescription+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCSessionDescription+Private.h
index 4c58fa7..08a6f55 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCSessionDescription+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCSessionDescription+Private.h
@@ -29,7 +29,7 @@
* description.
*/
- (instancetype)initWithNativeDescription:
- (const webrtc::SessionDescriptionInterface *)nativeDescription;
+ (const webrtc::SessionDescriptionInterface *)nativeDescription;
+ (std::string)stdStringForType:(RTCSdpType)type;
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter+Private.h
index bdec24d..80a2140 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter+Private.h
@@ -30,8 +30,7 @@
* 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<RTCVideoRenderer>)videoRenderer
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource+Private.h
index bd42484..fdb4522 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource+Private.h
@@ -22,17 +22,16 @@
* The VideoTrackSourceInterface object passed to this RTCVideoSource during
* construction.
*/
-@property(nonatomic, readonly)
- rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>
- nativeVideoSource;
+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>
+ nativeVideoSource;
/** Initialize an RTCVideoSource from a native VideoTrackSourceInterface. */
- (instancetype)initWithNativeVideoSource:
- (rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource
+ (rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource
NS_DESIGNATED_INITIALIZER;
- (instancetype)initWithNativeMediaSource:
- (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+ (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
type:(RTCMediaSourceType)type NS_UNAVAILABLE;
- (instancetype)initWithSignalingThread:(rtc::Thread *)signalingThread
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCVideoTrack+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCVideoTrack+Private.h
index 2cb6a13..14639be 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCVideoTrack+Private.h
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCVideoTrack+Private.h
@@ -17,8 +17,7 @@
@interface RTCVideoTrack ()
/** VideoTrackInterface created or passed in at construction. */
-@property(nonatomic, readonly)
- rtc::scoped_refptr<webrtc::VideoTrackInterface> nativeVideoTrack;
+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::VideoTrackInterface> nativeVideoTrack;
/** Initialize an RTCVideoTrack with its source and an id. */
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
diff --git a/sdk/objc/Framework/Classes/Video/RTCShader.h b/sdk/objc/Framework/Classes/Video/RTCShader.h
index 4d92d3d..67afabb 100644
--- a/sdk/objc/Framework/Classes/Video/RTCShader.h
+++ b/sdk/objc/Framework/Classes/Video/RTCShader.h
@@ -12,9 +12,10 @@
RTC_EXTERN const char kRTCVertexShaderSource[];
-RTC_EXTERN GLuint RTCCreateShader(GLenum type, const GLchar *source);
+RTC_EXTERN GLuint RTCCreateShader(GLenum type, const GLchar* source);
RTC_EXTERN GLuint RTCCreateProgram(GLuint vertexShader, GLuint fragmentShader);
-RTC_EXTERN GLuint RTCCreateProgramFromFragmentSource(const char fragmentShaderSource[]);
+RTC_EXTERN GLuint
+RTCCreateProgramFromFragmentSource(const char fragmentShaderSource[]);
RTC_EXTERN BOOL RTCCreateVertexBuffer(GLuint* vertexBuffer,
GLuint* vertexArray);
RTC_EXTERN void RTCSetVertexData(RTCVideoRotation rotation);
diff --git a/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.cc b/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.cc
index 625fc53..a373050 100644
--- a/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.cc
+++ b/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.cc
@@ -33,7 +33,7 @@
CMSampleBufferRef avcc_sample_buffer,
bool is_keyframe,
rtc::Buffer* annexb_buffer,
- std::unique_ptr<RTPFragmentationHeader> *out_header) {
+ std::unique_ptr<RTPFragmentationHeader>* out_header) {
RTC_DCHECK(avcc_sample_buffer);
RTC_DCHECK(out_header);
out_header->reset(nullptr);
diff --git a/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h b/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h
index cd81f12..0155882 100644
--- a/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h
+++ b/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h
@@ -33,7 +33,7 @@
CMSampleBufferRef avcc_sample_buffer,
bool is_keyframe,
rtc::Buffer* annexb_buffer,
- std::unique_ptr<RTPFragmentationHeader> *out_header);
+ std::unique_ptr<RTPFragmentationHeader>* out_header);
// Converts a buffer received from RTP into a sample buffer suitable for the
// VideoToolbox decoder. The RTP buffer is in annex b format whereas the sample
diff --git a/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter_unittest.cc b/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter_unittest.cc
index c8de64b..d907ef8 100644
--- a/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter_unittest.cc
+++ b/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter_unittest.cc
@@ -25,14 +25,11 @@
TEST(H264VideoToolboxNaluTest, TestCreateVideoFormatDescription) {
const uint8_t sps_pps_buffer[] = {
- // SPS nalu.
- 0x00, 0x00, 0x00, 0x01,
- 0x27, 0x42, 0x00, 0x1E, 0xAB, 0x40, 0xF0, 0x28, 0xD3, 0x70, 0x20, 0x20,
- 0x20, 0x20,
- // PPS nalu.
- 0x00, 0x00, 0x00, 0x01,
- 0x28, 0xCE, 0x3C, 0x30
- };
+ // SPS nalu.
+ 0x00, 0x00, 0x00, 0x01, 0x27, 0x42, 0x00, 0x1E, 0xAB, 0x40, 0xF0, 0x28,
+ 0xD3, 0x70, 0x20, 0x20, 0x20, 0x20,
+ // PPS nalu.
+ 0x00, 0x00, 0x00, 0x01, 0x28, 0xCE, 0x3C, 0x30};
CMVideoFormatDescriptionRef description =
CreateVideoFormatDescription(sps_pps_buffer, arraysize(sps_pps_buffer));
EXPECT_TRUE(description);
@@ -59,8 +56,8 @@
}
const uint8_t other_buffer[] = {0x00, 0x00, 0x00, 0x01, 0x28};
- EXPECT_FALSE(CreateVideoFormatDescription(other_buffer,
- arraysize(other_buffer)));
+ EXPECT_FALSE(
+ CreateVideoFormatDescription(other_buffer, arraysize(other_buffer)));
}
TEST(AnnexBBufferReaderTest, TestReadEmptyInput) {
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h b/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
index 6c4c96a..15b63d3 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
@@ -15,7 +15,7 @@
NS_ASSUME_NONNULL_BEGIN
-extern NSString * const kRTCAudioSessionErrorDomain;
+extern NSString *const kRTCAudioSessionErrorDomain;
/** Method that requires lock was called without lock. */
extern NSInteger const kRTCAudioSessionErrorLockRequired;
/** Unknown configuration error occurred. */
@@ -46,8 +46,8 @@
* route.
*/
- (void)audioSessionDidChangeRoute:(RTCAudioSession *)session
- reason:(AVAudioSessionRouteChangeReason)reason
- previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
+ reason:(AVAudioSessionRouteChangeReason)reason
+ previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
/** Called on a system notification thread when AVAudioSession media server
* terminates.
@@ -61,8 +61,7 @@
// TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
-- (void)audioSession:(RTCAudioSession *)session
- didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
+- (void)audioSession:(RTCAudioSession *)session didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
/** Called on a WebRTC thread when the audio device is notified to begin
* playback or recording.
@@ -75,8 +74,7 @@
- (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session;
/** Called when the AVAudioSession output volume value changes. */
-- (void)audioSession:(RTCAudioSession *)audioSession
- didChangeOutputVolume:(float)outputVolume;
+- (void)audioSession:(RTCAudioSession *)audioSession didChangeOutputVolume:(float)outputVolume;
/** Called when the audio device detects a playout glitch. The argument is the
* number of glitches detected so far in the current audio playout session.
@@ -170,14 +168,10 @@
@property(readonly) float inputGain;
@property(readonly) BOOL inputGainSettable;
@property(readonly) BOOL inputAvailable;
-@property(readonly, nullable)
- NSArray<AVAudioSessionDataSourceDescription *> * inputDataSources;
-@property(readonly, nullable)
- AVAudioSessionDataSourceDescription *inputDataSource;
-@property(readonly, nullable)
- NSArray<AVAudioSessionDataSourceDescription *> * outputDataSources;
-@property(readonly, nullable)
- AVAudioSessionDataSourceDescription *outputDataSource;
+@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *inputDataSources;
+@property(readonly, nullable) AVAudioSessionDataSourceDescription *inputDataSource;
+@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *outputDataSources;
+@property(readonly, nullable) AVAudioSessionDataSourceDescription *outputDataSource;
@property(readonly) double sampleRate;
@property(readonly) double preferredSampleRate;
@property(readonly) NSInteger inputNumberOfChannels;
@@ -211,8 +205,7 @@
* AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
* AVAudioSession.
*/
-- (BOOL)setActive:(BOOL)active
- error:(NSError **)outError;
+- (BOOL)setActive:(BOOL)active error:(NSError **)outError;
// The following methods are proxies for the associated methods on
// AVAudioSession. |lockForConfiguration| must be called before using them
@@ -224,16 +217,11 @@
- (BOOL)setMode:(NSString *)mode error:(NSError **)outError;
- (BOOL)setInputGain:(float)gain error:(NSError **)outError;
- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError;
-- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration
- error:(NSError **)outError;
-- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count
- error:(NSError **)outError;
-- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count
- error:(NSError **)outError;
-- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride
- error:(NSError **)outError;
-- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort
- error:(NSError **)outError;
+- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration error:(NSError **)outError;
+- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
+- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
+- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride error:(NSError **)outError;
+- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort error:(NSError **)outError;
- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
error:(NSError **)outError;
- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
@@ -247,8 +235,7 @@
* returned.
* |lockForConfiguration| must be called first.
*/
-- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
- error:(NSError **)outError;
+- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration error:(NSError **)outError;
/** Convenience method that calls both setConfiguration and setActive.
* |lockForConfiguration| must be called first.
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h b/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
index d62bb01..2995c9c 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
@@ -25,6 +25,6 @@
* is assigned to AVCaptureVideoPreviewLayer async in the same
* queue that the AVCaptureSession is started/stopped.
*/
-@property(nonatomic, strong) AVCaptureSession *captureSession;
+@property(nonatomic, strong) AVCaptureSession* captureSession;
@end
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h b/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h
index 3bfd60e..61868c5 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#import <Foundation/Foundation.h>
#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCapturer.h>
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h b/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
index b203b28..d274530 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
@@ -34,10 +34,7 @@
};
/** 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) {
@@ -88,8 +85,7 @@
@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;
/** By default, the PeerConnection will use a limited number of IPv6 network
* interfaces, in order to avoid too many ICE candidate pairs being created
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h b/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
index 893bd0a..4859e04 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
@@ -34,7 +34,6 @@
@end
-
@class RTCDataChannel;
RTC_EXPORT
@protocol RTCDataChannelDelegate <NSObject>
@@ -48,12 +47,10 @@
@optional
/** The data channel's |bufferedAmount| changed. */
-- (void)dataChannel:(RTCDataChannel *)dataChannel
- didChangeBufferedAmount:(uint64_t)amount;
+- (void)dataChannel:(RTCDataChannel *)dataChannel didChangeBufferedAmount:(uint64_t)amount;
@end
-
/** Represents the state of the data channel. */
typedef NS_ENUM(NSInteger, RTCDataChannelState) {
RTCDataChannelStateConnecting,
@@ -78,8 +75,7 @@
@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
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h b/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
index 65f5931..65ca8f8 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
@@ -45,7 +45,7 @@
@property(nonatomic, assign) int channelId;
/** Set by the application and opaque to the WebRTC implementation. */
-@property(nonatomic) NSString *protocol;
+@property(nonatomic) NSString* protocol;
@end
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h b/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
index 3dddead..0338747 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
@@ -34,8 +34,7 @@
* @param dispatchType The queue type to dispatch on.
* @param block The block to dispatch asynchronously.
*/
-+ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType
- block:(dispatch_block_t)block;
++ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType block:(dispatch_block_t)block;
/** Returns YES if run on queue for the dispatchType otherwise NO.
* Useful for asserting that a method is run on a correct queue.
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h b/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
index e6e5b5f..8fdd3c3 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
@@ -20,7 +20,7 @@
@class RTCEAGLVideoView;
RTC_EXPORT
-@protocol RTCEAGLVideoViewDelegate<RTCVideoViewDelegate>
+@protocol RTCEAGLVideoViewDelegate <RTCVideoViewDelegate>
@end
/**
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h b/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
index 1e8a591..ac8a910 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
@@ -52,13 +52,11 @@
- (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).
@@ -74,4 +72,3 @@
@end
NS_ASSUME_NONNULL_END
-
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h b/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
index 3b7f3e2..fb5064e 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
@@ -42,8 +42,7 @@
*/
- (instancetype)initWithSdp:(NSString *)sdp
sdpMLineIndex:(int)sdpMLineIndex
- sdpMid:(nullable NSString *)sdpMid
- NS_DESIGNATED_INITIALIZER;
+ sdpMid:(nullable NSString *)sdpMid NS_DESIGNATED_INITIALIZER;
@end
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h b/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
index 9384b15..00508eb 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
@@ -18,11 +18,8 @@
@property(nonatomic, readonly) NSInteger max;
- (instancetype)init;
-- (instancetype)initWithMin:(NSInteger)min
- max:(NSInteger)max
- NS_DESIGNATED_INITIALIZER;
+- (instancetype)initWithMin:(NSInteger)min max:(NSInteger)max NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
-
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCLogging.h b/sdk/objc/Framework/Headers/WebRTC/RTCLogging.h
index a65dcad..b758785 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCLogging.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCLogging.h
@@ -33,12 +33,9 @@
// Some convenience macros.
-#define RTCLogString(format, ...) \
- [NSString stringWithFormat:@"(%@:%d %s): " format, \
- RTCFileName(__FILE__), \
- __LINE__, \
- __FUNCTION__, \
- ##__VA_ARGS__]
+#define RTCLogString(format, ...) \
+ [NSString stringWithFormat:@"(%@:%d %s): " format, RTCFileName(__FILE__), \
+ __LINE__, __FUNCTION__, ##__VA_ARGS__]
#define RTCLogFormat(severity, format, ...) \
do { \
@@ -46,17 +43,17 @@
RTCLogEx(severity, log_string); \
} while (false)
-#define RTCLogVerbose(format, ...) \
- RTCLogFormat(RTCLoggingSeverityVerbose, format, ##__VA_ARGS__) \
+#define RTCLogVerbose(format, ...) \
+ RTCLogFormat(RTCLoggingSeverityVerbose, format, ##__VA_ARGS__)
-#define RTCLogInfo(format, ...) \
- RTCLogFormat(RTCLoggingSeverityInfo, format, ##__VA_ARGS__) \
+#define RTCLogInfo(format, ...) \
+ RTCLogFormat(RTCLoggingSeverityInfo, format, ##__VA_ARGS__)
-#define RTCLogWarning(format, ...) \
- RTCLogFormat(RTCLoggingSeverityWarning, format, ##__VA_ARGS__) \
+#define RTCLogWarning(format, ...) \
+ RTCLogFormat(RTCLoggingSeverityWarning, format, ##__VA_ARGS__)
-#define RTCLogError(format, ...) \
- RTCLogFormat(RTCLoggingSeverityError, format, ##__VA_ARGS__) \
+#define RTCLogError(format, ...) \
+ RTCLogFormat(RTCLoggingSeverityError, format, ##__VA_ARGS__)
#if !defined(NDEBUG)
#define RTCLogDebug(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h b/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h
index f82126c..d28d554 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h
@@ -12,7 +12,7 @@
#import "WebRTC/RTCVideoRenderer.h"
NS_AVAILABLE_MAC(10.11)
-@interface RTCMTLNSVideoView : NSView<RTCVideoRenderer>
+@interface RTCMTLNSVideoView : NSView <RTCVideoRenderer>
@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h b/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h
index da1797f..0615495 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h
@@ -32,7 +32,7 @@
NS_CLASS_AVAILABLE_IOS(9)
RTC_EXPORT
-@interface RTCMTLVideoView : UIView <RTCVideoRenderer>
+@interface RTCMTLVideoView : UIView<RTCVideoRenderer>
@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h b/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
index f93e5b5..fce0821 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
@@ -15,28 +15,28 @@
NS_ASSUME_NONNULL_BEGIN
/** Constraint keys for media sources. */
-RTC_EXTERN NSString * const kRTCMediaConstraintsMinAspectRatio;
-RTC_EXTERN NSString * const kRTCMediaConstraintsMaxAspectRatio;
-RTC_EXTERN NSString * const kRTCMediaConstraintsMaxWidth;
-RTC_EXTERN NSString * const kRTCMediaConstraintsMinWidth;
-RTC_EXTERN NSString * const kRTCMediaConstraintsMaxHeight;
-RTC_EXTERN NSString * const kRTCMediaConstraintsMinHeight;
-RTC_EXTERN NSString * const kRTCMediaConstraintsMaxFrameRate;
-RTC_EXTERN NSString * const kRTCMediaConstraintsMinFrameRate;
+RTC_EXTERN NSString *const kRTCMediaConstraintsMinAspectRatio;
+RTC_EXTERN NSString *const kRTCMediaConstraintsMaxAspectRatio;
+RTC_EXTERN NSString *const kRTCMediaConstraintsMaxWidth;
+RTC_EXTERN NSString *const kRTCMediaConstraintsMinWidth;
+RTC_EXTERN NSString *const kRTCMediaConstraintsMaxHeight;
+RTC_EXTERN NSString *const kRTCMediaConstraintsMinHeight;
+RTC_EXTERN NSString *const kRTCMediaConstraintsMaxFrameRate;
+RTC_EXTERN NSString *const kRTCMediaConstraintsMinFrameRate;
/** The value for this key should be a base64 encoded string containing
* the data from the serialized configuration proto.
*/
-RTC_EXTERN NSString * const kRTCMediaConstraintsAudioNetworkAdaptorConfig;
+RTC_EXTERN NSString *const kRTCMediaConstraintsAudioNetworkAdaptorConfig;
/** Constraint keys for generating offers and answers. */
-RTC_EXTERN NSString * const kRTCMediaConstraintsIceRestart;
-RTC_EXTERN NSString * const kRTCMediaConstraintsOfferToReceiveAudio;
-RTC_EXTERN NSString * const kRTCMediaConstraintsOfferToReceiveVideo;
-RTC_EXTERN NSString * const kRTCMediaConstraintsVoiceActivityDetection;
+RTC_EXTERN NSString *const kRTCMediaConstraintsIceRestart;
+RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveAudio;
+RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveVideo;
+RTC_EXTERN NSString *const kRTCMediaConstraintsVoiceActivityDetection;
/** Constraint values for Boolean parameters. */
-RTC_EXTERN NSString * const kRTCMediaConstraintsValueTrue;
-RTC_EXTERN NSString * const kRTCMediaConstraintsValueFalse;
+RTC_EXTERN NSString *const kRTCMediaConstraintsValueTrue;
+RTC_EXTERN NSString *const kRTCMediaConstraintsValueFalse;
RTC_EXPORT
@interface RTCMediaConstraints : NSObject
@@ -44,10 +44,9 @@
- (instancetype)init NS_UNAVAILABLE;
/** Initialize with mandatory and/or optional constraints. */
-- (instancetype)initWithMandatoryConstraints:
- (nullable NSDictionary<NSString *, NSString *> *)mandatory
- optionalConstraints:
- (nullable NSDictionary<NSString *, NSString *> *)optional
+- (instancetype)
+ initWithMandatoryConstraints:(nullable NSDictionary<NSString *, NSString *> *)mandatory
+ optionalConstraints:(nullable NSDictionary<NSString *, NSString *> *)optional
NS_DESIGNATED_INITIALIZER;
@end
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h b/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
index c42f0cf..aee9bdd 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
@@ -22,8 +22,8 @@
NS_ASSUME_NONNULL_BEGIN
-RTC_EXTERN NSString * const kRTCMediaStreamTrackKindAudio;
-RTC_EXTERN NSString * const kRTCMediaStreamTrackKindVideo;
+RTC_EXTERN NSString *const kRTCMediaStreamTrackKindAudio;
+RTC_EXTERN NSString *const kRTCMediaStreamTrackKindVideo;
RTC_EXPORT
@interface RTCMediaStreamTrack : NSObject
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCMetrics.h b/sdk/objc/Framework/Headers/WebRTC/RTCMetrics.h
index d22287e..cc89888 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCMetrics.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCMetrics.h
@@ -20,5 +20,4 @@
RTC_EXTERN void RTCEnableMetrics(void);
/** Gets and clears native histograms. */
-RTC_EXTERN NSArray<RTCMetricsSampleInfo *> *RTCGetAndResetMetrics(void);
-
+RTC_EXTERN NSArray<RTCMetricsSampleInfo*>* RTCGetAndResetMetrics(void);
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h b/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
index 62c7968..77e301f 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
@@ -21,7 +21,7 @@
@class RTCNSGLVideoView;
-@protocol RTCNSGLVideoViewDelegate<RTCVideoViewDelegate>
+@protocol RTCNSGLVideoViewDelegate <RTCVideoViewDelegate>
@end
@interface RTCNSGLVideoView : NSOpenGLView <RTCVideoRenderer>
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h b/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
index d9ddffb..94036cc 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
@@ -31,7 +31,7 @@
NS_ASSUME_NONNULL_BEGIN
-extern NSString * const kRTCPeerConnectionErrorDomain;
+extern NSString *const kRTCPeerConnectionErrorDomain;
extern int const kRTCSessionDescriptionErrorCode;
/** Represents the signaling state of the peer connection. */
@@ -80,14 +80,12 @@
didChangeSignalingState:(RTCSignalingState)stateChanged;
/** Called when media is received on a new stream from remote peer. */
-- (void)peerConnection:(RTCPeerConnection *)peerConnection
- didAddStream:(RTCMediaStream *)stream;
+- (void)peerConnection:(RTCPeerConnection *)peerConnection didAddStream:(RTCMediaStream *)stream;
/** Called when a remote peer closes a stream.
* This is not called when RTCSdpSemanticsUnifiedPlan is specified.
*/
-- (void)peerConnection:(RTCPeerConnection *)peerConnection
- didRemoveStream:(RTCMediaStream *)stream;
+- (void)peerConnection:(RTCPeerConnection *)peerConnection didRemoveStream:(RTCMediaStream *)stream;
/** Called when negotiation is needed, for example ICE has restarted. */
- (void)peerConnectionShouldNegotiate:(RTCPeerConnection *)peerConnection;
@@ -139,10 +137,8 @@
* |senders| instead.
*/
@property(nonatomic, readonly) NSArray<RTCMediaStream *> *localStreams;
-@property(nonatomic, readonly, nullable)
- RTCSessionDescription *localDescription;
-@property(nonatomic, readonly, nullable)
- RTCSessionDescription *remoteDescription;
+@property(nonatomic, readonly, nullable) RTCSessionDescription *localDescription;
+@property(nonatomic, readonly, nullable) RTCSessionDescription *remoteDescription;
@property(nonatomic, readonly) RTCSignalingState signalingState;
@property(nonatomic, readonly) RTCIceConnectionState iceConnectionState;
@property(nonatomic, readonly) RTCIceGatheringState iceGatheringState;
@@ -252,25 +248,21 @@
/** Generate an SDP offer. */
- (void)offerForConstraints:(RTCMediaConstraints *)constraints
- completionHandler:(nullable void (^)
- (RTCSessionDescription * _Nullable sdp,
- NSError * _Nullable error))completionHandler;
+ completionHandler:(nullable void (^)(RTCSessionDescription *_Nullable sdp,
+ NSError *_Nullable error))completionHandler;
/** Generate an SDP answer. */
- (void)answerForConstraints:(RTCMediaConstraints *)constraints
- completionHandler:(nullable void (^)
- (RTCSessionDescription * _Nullable sdp,
- NSError * _Nullable error))completionHandler;
+ completionHandler:(nullable void (^)(RTCSessionDescription *_Nullable sdp,
+ NSError *_Nullable error))completionHandler;
/** Apply the supplied RTCSessionDescription as the local description. */
- (void)setLocalDescription:(RTCSessionDescription *)sdp
- completionHandler:
- (nullable void (^)(NSError * _Nullable error))completionHandler;
+ completionHandler:(nullable void (^)(NSError *_Nullable error))completionHandler;
/** Apply the supplied RTCSessionDescription as the remote description. */
- (void)setRemoteDescription:(RTCSessionDescription *)sdp
- completionHandler:
- (nullable void (^)(NSError * _Nullable error))completionHandler;
+ completionHandler:(nullable void (^)(NSError *_Nullable error))completionHandler;
/** Limits the bandwidth allocated for all RTP streams sent by this
* PeerConnection. Nil parameters will be unchanged. Setting
@@ -282,8 +274,7 @@
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
@@ -312,11 +303,9 @@
/** Gather stats for the given RTCMediaStreamTrack. If |mediaStreamTrack| is nil
* statistics are gathered for all tracks.
*/
-- (void)statsForTrack:
- (nullable RTCMediaStreamTrack *)mediaStreamTrack
+- (void)statsForTrack:(nullable RTCMediaStreamTrack *)mediaStreamTrack
statsOutputLevel:(RTCStatsOutputLevel)statsOutputLevel
- completionHandler:
- (nullable void (^)(NSArray<RTCLegacyStatsReport *> *stats))completionHandler;
+ completionHandler:(nullable void (^)(NSArray<RTCLegacyStatsReport *> *stats))completionHandler;
@end
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h b/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
index d00f6f2..7f7b7e6 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
@@ -46,8 +46,7 @@
- (RTCAudioTrack *)audioTrackWithTrackId:(NSString *)trackId;
/** Initialize an RTCAudioTrack with a source and an id. */
-- (RTCAudioTrack *)audioTrackWithSource:(RTCAudioSource *)source
- trackId:(NSString *)trackId;
+- (RTCAudioTrack *)audioTrackWithSource:(RTCAudioSource *)source trackId:(NSString *)trackId;
/** Initialize a generic RTCVideoSource. The RTCVideoSource should be passed to a RTCVideoCapturer
* implementation, e.g. RTCCameraVideoCapturer, in order to produce frames.
@@ -55,8 +54,7 @@
- (RTCVideoSource *)videoSource;
/** Initialize an RTCVideoTrack with a source and an id. */
-- (RTCVideoTrack *)videoTrackWithSource:(RTCVideoSource *)source
- trackId:(NSString *)trackId;
+- (RTCVideoTrack *)videoTrackWithSource:(RTCVideoSource *)source trackId:(NSString *)trackId;
/** Initialize an RTCMediaStream with an id. */
- (RTCMediaStream *)mediaStreamWithStreamId:(NSString *)streamId;
@@ -64,19 +62,16 @@
/** Initialize an RTCPeerConnection with a configuration, constraints, and
* delegate.
*/
-- (RTCPeerConnection *)peerConnectionWithConfiguration:
- (RTCConfiguration *)configuration
- constraints:
- (RTCMediaConstraints *)constraints
+- (RTCPeerConnection *)peerConnectionWithConfiguration:(RTCConfiguration *)configuration
+ constraints:(RTCMediaConstraints *)constraints
delegate:
- (nullable id<RTCPeerConnectionDelegate>)delegate;
+ (nullable id<RTCPeerConnectionDelegate>)delegate;
/** Set the options to be used for subsequently created RTCPeerConnections */
- (void)setOptions:(nonnull RTCPeerConnectionFactoryOptions *)options;
/** 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;
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h b/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
index 4ba0364..bb9aa90 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
@@ -14,22 +14,22 @@
NS_ASSUME_NONNULL_BEGIN
-RTC_EXTERN const NSString * const kRTCRtxCodecName;
-RTC_EXTERN const NSString * const kRTCRedCodecName;
-RTC_EXTERN const NSString * const kRTCUlpfecCodecName;
-RTC_EXTERN const NSString * const kRTCFlexfecCodecName;
-RTC_EXTERN const NSString * const kRTCOpusCodecName;
-RTC_EXTERN const NSString * const kRTCIsacCodecName;
-RTC_EXTERN const NSString * const kRTCL16CodecName;
-RTC_EXTERN const NSString * const kRTCG722CodecName;
-RTC_EXTERN const NSString * const kRTCIlbcCodecName;
-RTC_EXTERN const NSString * const kRTCPcmuCodecName;
-RTC_EXTERN const NSString * const kRTCPcmaCodecName;
-RTC_EXTERN const NSString * const kRTCDtmfCodecName;
-RTC_EXTERN const NSString * const kRTCComfortNoiseCodecName;
-RTC_EXTERN const NSString * const kRTCVp8CodecName;
-RTC_EXTERN const NSString * const kRTCVp9CodecName;
-RTC_EXTERN const NSString * const kRTCH264CodecName;
+RTC_EXTERN const NSString *const kRTCRtxCodecName;
+RTC_EXTERN const NSString *const kRTCRedCodecName;
+RTC_EXTERN const NSString *const kRTCUlpfecCodecName;
+RTC_EXTERN const NSString *const kRTCFlexfecCodecName;
+RTC_EXTERN const NSString *const kRTCOpusCodecName;
+RTC_EXTERN const NSString *const kRTCIsacCodecName;
+RTC_EXTERN const NSString *const kRTCL16CodecName;
+RTC_EXTERN const NSString *const kRTCG722CodecName;
+RTC_EXTERN const NSString *const kRTCIlbcCodecName;
+RTC_EXTERN const NSString *const kRTCPcmuCodecName;
+RTC_EXTERN const NSString *const kRTCPcmaCodecName;
+RTC_EXTERN const NSString *const kRTCDtmfCodecName;
+RTC_EXTERN const NSString *const kRTCComfortNoiseCodecName;
+RTC_EXTERN const NSString *const kRTCVp8CodecName;
+RTC_EXTERN const NSString *const kRTCVp9CodecName;
+RTC_EXTERN const NSString *const kRTCH264CodecName;
/** Defined in http://w3c.github.io/webrtc-pc/#idl-def-RTCRtpCodecParameters */
RTC_EXPORT
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h b/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
index 4143980..7db53da 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
@@ -36,8 +36,7 @@
- (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;
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCTracing.h b/sdk/objc/Framework/Headers/WebRTC/RTCTracing.h
index 5e7e48f..899a570 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCTracing.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCTracing.h
@@ -16,6 +16,6 @@
/** Starts capture to specified file. Must be a valid writable path.
* Returns YES if capture starts.
*/
-RTC_EXTERN BOOL RTCStartInternalCapture(NSString *filePath);
+RTC_EXTERN BOOL RTCStartInternalCapture(NSString* filePath);
RTC_EXTERN void RTCStopInternalCapture(void);
RTC_EXTERN void RTCShutdownInternalTracer(void);
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h b/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
index c0d464e..5bc6d66 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
@@ -172,9 +172,9 @@
DEPRECATED_MSG_ATTRIBUTE("use startDecodeWithNumberOfCores: instead");
- (NSInteger)releaseDecoder;
- (NSInteger)decode:(RTCEncodedImage *)encodedImage
- missingFrames:(BOOL)missingFrames
- codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
- renderTimeMs:(int64_t)renderTimeMs;
+ missingFrames:(BOOL)missingFrames
+ codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
+ renderTimeMs:(int64_t)renderTimeMs;
- (NSString *)implementationName;
// TODO(andersc): Make non-optional when `startDecodeWithSettings:numberOfCores:` is removed.
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h b/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
index d18e96d..d4dbf88 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
@@ -69,7 +69,7 @@
/** Encoder. */
RTC_EXPORT
-@interface RTCVideoEncoderH264 : NSObject<RTCVideoEncoder>
+@interface RTCVideoEncoderH264 : NSObject <RTCVideoEncoder>
- (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo;
@@ -77,15 +77,15 @@
/** Decoder. */
RTC_EXPORT
-@interface RTCVideoDecoderH264 : NSObject<RTCVideoDecoder>
+@interface RTCVideoDecoderH264 : NSObject <RTCVideoDecoder>
@end
/** Encoder factory. */
RTC_EXPORT
-@interface RTCVideoEncoderFactoryH264 : NSObject<RTCVideoEncoderFactory>
+@interface RTCVideoEncoderFactoryH264 : NSObject <RTCVideoEncoderFactory>
@end
/** Decoder factory. */
RTC_EXPORT
-@interface RTCVideoDecoderFactoryH264 : NSObject<RTCVideoDecoderFactory>
+@interface RTCVideoDecoderFactoryH264 : NSObject <RTCVideoDecoderFactory>
@end
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h b/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
index dcc4c6e..cfccfa9 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
@@ -44,7 +44,7 @@
@property(nonatomic, readonly) id<RTCVideoFrameBuffer> buffer;
- (instancetype)init NS_UNAVAILABLE;
-- (instancetype)new NS_UNAVAILABLE;
+- (instancetype) new NS_UNAVAILABLE;
/** Initialize an RTCVideoFrame from a pixel buffer, rotation, and timestamp.
* Deprecated - initialize with a RTCCVPixelBuffer instead
diff --git a/sdk/objc/Framework/Native/api/audio_device_module.h b/sdk/objc/Framework/Native/api/audio_device_module.h
index 2c519ba..aaeb39b 100644
--- a/sdk/objc/Framework/Native/api/audio_device_module.h
+++ b/sdk/objc/Framework/Native/api/audio_device_module.h
@@ -21,4 +21,4 @@
} // namespace webrtc
-#endif // SDK_OBJC_FRAMEWORK_NATIVE_API_AUDIO_DEVICE_MODULE_H_
+#endif // SDK_OBJC_FRAMEWORK_NATIVE_API_AUDIO_DEVICE_MODULE_H_
diff --git a/sdk/objc/Framework/Native/src/audio/audio_device_ios.h b/sdk/objc/Framework/Native/src/audio/audio_device_ios.h
index 4001f49..56326c1 100644
--- a/sdk/objc/Framework/Native/src/audio/audio_device_ios.h
+++ b/sdk/objc/Framework/Native/src/audio/audio_device_ios.h
@@ -13,14 +13,14 @@
#include <memory>
-#include "sdk/objc/Framework/Headers/WebRTC/RTCMacros.h"
-#include "modules/audio_device/audio_device_generic.h"
#include "audio_session_observer.h"
-#include "voice_processing_audio_unit.h"
+#include "modules/audio_device/audio_device_generic.h"
#include "rtc_base/buffer.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/thread_checker.h"
+#include "sdk/objc/Framework/Headers/WebRTC/RTCMacros.h"
+#include "voice_processing_audio_unit.h"
RTC_FWD_DECL_OBJC_CLASS(RTCNativeAudioSessionDelegateAdapter);
@@ -155,7 +155,7 @@
AudioBufferList* io_data) override;
// Handles messages from posts.
- void OnMessage(rtc::Message *msg) override;
+ void OnMessage(rtc::Message* msg) override;
bool IsInterrupted();
diff --git a/sdk/objc/Framework/Native/src/audio/audio_device_module_ios.h b/sdk/objc/Framework/Native/src/audio/audio_device_module_ios.h
index 5f9bb6a..8e1ace4 100644
--- a/sdk/objc/Framework/Native/src/audio/audio_device_module_ios.h
+++ b/sdk/objc/Framework/Native/src/audio/audio_device_module_ios.h
@@ -20,124 +20,120 @@
#include "rtc_base/checks.h"
#include "rtc_base/criticalsection.h"
-
namespace webrtc {
class AudioDeviceGeneric;
namespace ios_adm {
- class AudioDeviceModuleIOS : public AudioDeviceModule {
+class AudioDeviceModuleIOS : public AudioDeviceModule {
+ public:
+ int32_t AttachAudioBuffer();
- public:
+ AudioDeviceModuleIOS();
+ ~AudioDeviceModuleIOS() override;
- int32_t AttachAudioBuffer();
+ // Retrieve the currently utilized audio layer
+ int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
- AudioDeviceModuleIOS();
- ~AudioDeviceModuleIOS() override;
+ // Full-duplex transportation of PCM audio
+ int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
- // Retrieve the currently utilized audio layer
- int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
+ // Main initializaton and termination
+ int32_t Init() override;
+ int32_t Terminate() override;
+ bool Initialized() const override;
- // Full-duplex transportation of PCM audio
- int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
-
- // Main initializaton and termination
- int32_t Init() override;
- int32_t Terminate() override;
- bool Initialized() const override;
-
- // Device enumeration
- int16_t PlayoutDevices() override;
- int16_t RecordingDevices() override;
- int32_t PlayoutDeviceName(uint16_t index,
+ // Device enumeration
+ int16_t PlayoutDevices() override;
+ int16_t RecordingDevices() override;
+ int32_t PlayoutDeviceName(uint16_t index,
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]) override;
+ int32_t RecordingDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) override;
- int32_t RecordingDeviceName(uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) override;
- // Device selection
- int32_t SetPlayoutDevice(uint16_t index) override;
- int32_t SetPlayoutDevice(WindowsDeviceType device) override;
- int32_t SetRecordingDevice(uint16_t index) override;
- int32_t SetRecordingDevice(WindowsDeviceType device) override;
+ // Device selection
+ int32_t SetPlayoutDevice(uint16_t index) override;
+ int32_t SetPlayoutDevice(WindowsDeviceType device) override;
+ int32_t SetRecordingDevice(uint16_t index) override;
+ int32_t SetRecordingDevice(WindowsDeviceType device) override;
- // Audio transport initialization
- int32_t PlayoutIsAvailable(bool* available) override;
- int32_t InitPlayout() override;
- bool PlayoutIsInitialized() const override;
- int32_t RecordingIsAvailable(bool* available) override;
- int32_t InitRecording() override;
- bool RecordingIsInitialized() const override;
+ // Audio transport initialization
+ int32_t PlayoutIsAvailable(bool* available) override;
+ int32_t InitPlayout() override;
+ bool PlayoutIsInitialized() const override;
+ int32_t RecordingIsAvailable(bool* available) override;
+ int32_t InitRecording() override;
+ bool RecordingIsInitialized() const override;
- // Audio transport control
- int32_t StartPlayout() override;
- int32_t StopPlayout() override;
- bool Playing() const override;
- int32_t StartRecording() override;
- int32_t StopRecording() override;
- bool Recording() const override;
+ // Audio transport control
+ int32_t StartPlayout() override;
+ int32_t StopPlayout() override;
+ bool Playing() const override;
+ int32_t StartRecording() override;
+ int32_t StopRecording() override;
+ bool Recording() const override;
- // Audio mixer initialization
- int32_t InitSpeaker() override;
- bool SpeakerIsInitialized() const override;
- int32_t InitMicrophone() override;
- bool MicrophoneIsInitialized() const override;
+ // Audio mixer initialization
+ int32_t InitSpeaker() override;
+ bool SpeakerIsInitialized() const override;
+ int32_t InitMicrophone() override;
+ bool MicrophoneIsInitialized() const override;
- // Speaker volume controls
- int32_t SpeakerVolumeIsAvailable(bool* available) override;
- int32_t SetSpeakerVolume(uint32_t volume) override;
- int32_t SpeakerVolume(uint32_t* volume) const override;
- int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override;
- int32_t MinSpeakerVolume(uint32_t* minVolume) const override;
+ // Speaker volume controls
+ int32_t SpeakerVolumeIsAvailable(bool* available) override;
+ int32_t SetSpeakerVolume(uint32_t volume) override;
+ int32_t SpeakerVolume(uint32_t* volume) const override;
+ int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override;
+ int32_t MinSpeakerVolume(uint32_t* minVolume) const override;
- // Microphone volume controls
- int32_t MicrophoneVolumeIsAvailable(bool* available) override;
- int32_t SetMicrophoneVolume(uint32_t volume) override;
- int32_t MicrophoneVolume(uint32_t* volume) const override;
- int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override;
- int32_t MinMicrophoneVolume(uint32_t* minVolume) const override;
+ // Microphone volume controls
+ int32_t MicrophoneVolumeIsAvailable(bool* available) override;
+ int32_t SetMicrophoneVolume(uint32_t volume) override;
+ int32_t MicrophoneVolume(uint32_t* volume) const override;
+ int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override;
+ int32_t MinMicrophoneVolume(uint32_t* minVolume) const override;
- // Speaker mute control
- int32_t SpeakerMuteIsAvailable(bool* available) override;
- int32_t SetSpeakerMute(bool enable) override;
- int32_t SpeakerMute(bool* enabled) const override;
+ // Speaker mute control
+ int32_t SpeakerMuteIsAvailable(bool* available) override;
+ int32_t SetSpeakerMute(bool enable) override;
+ int32_t SpeakerMute(bool* enabled) const override;
- // Microphone mute control
- int32_t MicrophoneMuteIsAvailable(bool* available) override;
- int32_t SetMicrophoneMute(bool enable) override;
- int32_t MicrophoneMute(bool* enabled) const override;
+ // Microphone mute control
+ int32_t MicrophoneMuteIsAvailable(bool* available) override;
+ int32_t SetMicrophoneMute(bool enable) override;
+ int32_t MicrophoneMute(bool* enabled) const override;
- // Stereo support
- int32_t StereoPlayoutIsAvailable(bool* available) const override;
- int32_t SetStereoPlayout(bool enable) override;
- int32_t StereoPlayout(bool* enabled) const override;
- int32_t StereoRecordingIsAvailable(bool* available) const override;
- int32_t SetStereoRecording(bool enable) override;
- int32_t StereoRecording(bool* enabled) const override;
+ // Stereo support
+ int32_t StereoPlayoutIsAvailable(bool* available) const override;
+ int32_t SetStereoPlayout(bool enable) override;
+ int32_t StereoPlayout(bool* enabled) const override;
+ int32_t StereoRecordingIsAvailable(bool* available) const override;
+ int32_t SetStereoRecording(bool enable) override;
+ int32_t StereoRecording(bool* enabled) const override;
- // Delay information and control
- int32_t PlayoutDelay(uint16_t* delayMS) const override;
+ // Delay information and control
+ int32_t PlayoutDelay(uint16_t* delayMS) const override;
- bool BuiltInAECIsAvailable() const override;
- int32_t EnableBuiltInAEC(bool enable) override;
- bool BuiltInAGCIsAvailable() const override;
- int32_t EnableBuiltInAGC(bool enable) override;
- bool BuiltInNSIsAvailable() const override;
- int32_t EnableBuiltInNS(bool enable) override;
+ bool BuiltInAECIsAvailable() const override;
+ int32_t EnableBuiltInAEC(bool enable) override;
+ bool BuiltInAGCIsAvailable() const override;
+ int32_t EnableBuiltInAGC(bool enable) override;
+ bool BuiltInNSIsAvailable() const override;
+ int32_t EnableBuiltInNS(bool enable) override;
#if defined(WEBRTC_IOS)
- int GetPlayoutAudioParameters(AudioParameters* params) const override;
- int GetRecordAudioParameters(AudioParameters* params) const override;
+ int GetPlayoutAudioParameters(AudioParameters* params) const override;
+ int GetRecordAudioParameters(AudioParameters* params) const override;
#endif // WEBRTC_IOS
- private:
- bool initialized_ = false;
- std::unique_ptr<AudioDeviceIOS> audio_device_;
- std::unique_ptr<AudioDeviceBuffer> audio_device_buffer_;
-
- };
-} // namespace ios_adm
-} // namespace webrtc
+ private:
+ bool initialized_ = false;
+ std::unique_ptr<AudioDeviceIOS> audio_device_;
+ std::unique_ptr<AudioDeviceBuffer> audio_device_buffer_;
+};
+} // namespace ios_adm
+} // namespace webrtc
#endif // SDK_IOS_NATIVE_API_AUDIO_DEVICE_MODULE_AUDIO_DEVICE_IOS_H_