Refactor AVAudioSession intialization code.

webrtc/modules/audio_device/ios/objc/RTCAudioSession.h

 /*
  *  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 <AVFoundation/AVFoundation.h>
 #import <Foundation/Foundation.h>
 
 NS_ASSUME_NONNULL_BEGIN
 
 extern NSString * const kRTCAudioSessionErrorDomain;
 extern NSInteger const kRTCAudioSessionErrorLockRequired;
 
 @class RTCAudioSession;
+@class RTCAudioSessionConfiguration;
 
 // Surfaces AVAudioSession events. WebRTC will listen directly for notifications
 // from AVAudioSession and handle them before calling these delegate methods,
 // at which point applications can perform additional processing if required.
 @protocol RTCAudioSessionDelegate <NSObject>
 
 /** Called when AVAudioSession starts an interruption event. */
 - (void)audioSessionDidBeginInterruption:(RTCAudioSession *)session;
 
 /** Called when AVAudioSession ends an interruption event. */
 - (void)audioSessionDidEndInterruption:(RTCAudioSession *)session
                    shouldResumeSession:(BOOL)shouldResumeSession;
 
 /** Called when AVAudioSession changes the route. */
 - (void)audioSessionDidChangeRoute:(RTCAudioSession *)session
            reason:(AVAudioSessionRouteChangeReason)reason
     previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
 
 /** Called when AVAudioSession media server terminates. */
 - (void)audioSessionMediaServicesWereLost:(RTCAudioSession *)session;
 
 /** Called when AVAudioSession media server restarts. */
 - (void)audioSessionMediaServicesWereReset:(RTCAudioSession *)session;
 
+/** Called when WebRTC needs to take over audio. Applications should call
+ *  -[RTCAudioSession configure] to allow WebRTC to play and record audio.
+ *  TODO(tkchin): Implement this behavior in RTCAudioSession.
+ */
+- (void)audioSessionShouldConfigure:(RTCAudioSession *)session;
+
+/** Called when WebRTC no longer requires audio. Applications should restore
+ *  their audio state at this point.
+ *  TODO(tkchin): Implement this behavior in RTCAudioSession.
+ */
+- (void)audioSessionShouldUnconfigure:(RTCAudioSession *)session;
+
 // TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
 
 @end
 
 /** Proxy class for AVAudioSession that adds a locking mechanism similar to
  *  AVCaptureDevice. This is used to that interleaving configurations between
  *  WebRTC and the application layer are avoided. Only setter methods are
  *  currently proxied. Getters can be accessed directly off AVAudioSession.
  *
  *  RTCAudioSession also coordinates activation so that the audio session is
  *  activated only once. See |setActive:error:|.
  */
 @interface RTCAudioSession : NSObject
 
 /** Convenience property to access the AVAudioSession singleton. Callers should
  *  not call setters on AVAudioSession directly, but other method invocations
  *  are fine.
  */
 @property(nonatomic, readonly) AVAudioSession *session;
 
 /** Our best guess at whether the session is active based on results of calls to
  *  AVAudioSession.
  */
 @property(nonatomic, readonly) BOOL isActive;
 /** Whether RTCAudioSession is currently locked for configuration. */
 @property(nonatomic, readonly) BOOL isLocked;
 
+/** If YES, WebRTC will not initialize the audio unit automatically when an
+ *  audio track is ready for playout or recording. Instead, applications should
+ *  listen to the delegate method |audioSessionShouldConfigure| and configure
+ *  the session manually. This should be set before making WebRTC media calls.
+ *  TODO(tkchin): Implement behavior. Currently this just stores a BOOL.
+ */
+@property(nonatomic, assign) BOOL shouldDelayAudioConfiguration;
+
 // Proxy properties.
 @property(readonly) NSString *category;
 @property(readonly) AVAudioSessionCategoryOptions categoryOptions;
 @property(readonly) NSString *mode;
 @property(readonly) BOOL secondaryAudioShouldBeSilencedHint;
 @property(readonly) AVAudioSessionRouteDescription *currentRoute;
 @property(readonly) NSInteger maximumInputNumberOfChannels;
 @property(readonly) NSInteger maximumOutputNumberOfChannels;
 @property(readonly) float inputGain;
 @property(readonly) BOOL inputGainSettable;
@@ skipping 62 matching lines @@
                           error:(NSError **)outError;
 - (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort
                     error:(NSError **)outError;
 - (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
                      error:(NSError **)outError;
 - (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
                       error:(NSError **)outError;
 
 @end
 
+@interface RTCAudioSession (Configuration)
+
+/** Returns YES if the audio session is correctly configured for WebRTC.
+ *  |lockForConfiguration| must be called first.
+ */
+- (BOOL)isConfiguredForWebRTC;
+
+/** Applies the configuration to the current session. Attempts to set all
+ *  properties even if previous ones fail. Only the last error will be
+ *  returned.
+ *  Also calls setActive with |active|.
+ */
+- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
+                  active:(BOOL)active
+                   error:(NSError **)outError;
+
+/** Configure the audio session for WebRTC. On failure, we will attempt to
+ *  restore the previously used audio session configuration.
+ */
+- (BOOL)configureWebRTCSession:(NSError **)outError;
+
+@end
+
 NS_ASSUME_NONNULL_END