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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;
- (BOOL)checkLock:(NSError **)outError;
/** 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;
// 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;
- (void)notifyDidDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
@end
NS_ASSUME_NONNULL_END