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+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCAudioSession.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCAudioSessionConfiguration);
+
+@interface RTC_OBJC_TYPE (RTCAudioSession)
+()
+
+ /** Number of times setActive:YES has succeeded without a balanced call to
+ * setActive:NO.
+ */
+ @property(nonatomic, readonly) int activationCount;
+
+/** The number of times `beginWebRTCSession` was called without a balanced call
+ * to `endWebRTCSession`.
+ */
+@property(nonatomic, readonly) int webRTCSessionCount;
+
+/** Convenience BOOL that checks useManualAudio and isAudioEnebled. */
+@property(readonly) BOOL canPlayOrRecord;
+
+/** Tracks whether we have been sent an interruption event that hasn't been matched by either an
+ * interrupted end event or a foreground event.
+ */
+@property(nonatomic, assign) BOOL isInterrupted;
+
+/** Adds the delegate to the list of delegates, and places it at the front of
+ * the list. This delegate will be notified before other delegates of
+ * audio events.
+ */
+- (void)pushDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate;
+
+/** Signals RTCAudioSession that a WebRTC session is about to begin and
+ * audio configuration is needed. Will configure the audio session for WebRTC
+ * if not already configured and if configuration is not delayed.
+ * Successful calls must be balanced by a call to endWebRTCSession.
+ */
+- (BOOL)beginWebRTCSession:(NSError **)outError;
+
+/** Signals RTCAudioSession that a WebRTC session is about to end and audio
+ * unconfiguration is needed. Will unconfigure the audio session for WebRTC
+ * if this is the last unmatched call and if configuration is not delayed.
+ */
+- (BOOL)endWebRTCSession:(NSError **)outError;
+
+/** Configure the audio session for WebRTC. This call will fail if the session
+ * is already configured. On other failures, we will attempt to restore the
+ * previously used audio session configuration.
+ * `lockForConfiguration` must be called first.
+ * Successful calls to configureWebRTCSession must be matched by calls to
+ * `unconfigureWebRTCSession`.
+ */
+- (BOOL)configureWebRTCSession:(NSError **)outError;
+
+/** Unconfigures the session for WebRTC. This will attempt to restore the
+ * audio session to the settings used before `configureWebRTCSession` was
+ * called.
+ * `lockForConfiguration` must be called first.
+ */
+- (BOOL)unconfigureWebRTCSession:(NSError **)outError;
+
+/** Returns a configuration error with the given description. */
+- (NSError *)configurationErrorWithDescription:(NSString *)description;
+
+/** Notifies the receiver that a playout glitch was detected. */
+- (void)notifyDidDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
+
+/** Notifies the receiver that there was an error when starting an audio unit. */
+- (void)notifyAudioUnitStartFailedWithError:(OSStatus)error;
+
+// Properties and methods for tests.
+- (void)notifyDidBeginInterruption;
+- (void)notifyDidEndInterruptionWithShouldResumeSession:(BOOL)shouldResumeSession;
+- (void)notifyDidChangeRouteWithReason:(AVAudioSessionRouteChangeReason)reason
+ previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
+- (void)notifyMediaServicesWereLost;
+- (void)notifyMediaServicesWereReset;
+- (void)notifyDidChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
+- (void)notifyDidStartPlayOrRecord;
+- (void)notifyDidStopPlayOrRecord;
+
+@end
+
+NS_ASSUME_NONNULL_END