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}
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@ -77,14 +77,12 @@ NS_ASSUME_NONNULL_BEGIN
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- (NSError *)configurationErrorWithDescription:(NSString *)description;
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// Properties and methods for tests.
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@property(nonatomic, readonly)
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std::vector<__weak id<RTCAudioSessionDelegate> > delegates;
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@property(nonatomic, readonly) std::vector<__weak id<RTCAudioSessionDelegate> > delegates;
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- (void)notifyDidBeginInterruption;
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- (void)notifyDidEndInterruptionWithShouldResumeSession:
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(BOOL)shouldResumeSession;
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- (void)notifyDidEndInterruptionWithShouldResumeSession:(BOOL)shouldResumeSession;
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- (void)notifyDidChangeRouteWithReason:(AVAudioSessionRouteChangeReason)reason
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previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
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previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
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- (void)notifyMediaServicesWereLost;
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- (void)notifyMediaServicesWereReset;
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- (void)notifyDidChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
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@ -26,8 +26,7 @@ class AudioSessionObserver;
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/** |observer| is a raw pointer and should be kept alive
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* for this object's lifetime.
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*/
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- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer
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NS_DESIGNATED_INITIALIZER;
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- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer NS_DESIGNATED_INITIALIZER;
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@end
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