text: /*
   *  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 "WebRTC/RTCAudioSession.h"
  
  #include <vector>
  
  NS_ASSUME_NONNULL_BEGIN
  
  @class RTCAudioSessionConfiguration;
  
  @interface 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<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.
  @property(nonatomic, readonly)
      std::vector<__weak id<RTCAudioSessionDelegate> > delegates;
  
  - (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
  