Move RTCAudioSession* files modules/audio_device/ -> sdk/Framework.

webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSessionConfiguration.m

 /*
  *  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/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
+#import "WebRTC/RTCAudioSession.h"
+#import "WebRTC/RTCAudioSessionConfiguration.h"
 
 #import "WebRTC/RTCDispatcher.h"
 #import "WebRTC/UIDevice+RTCDevice.h"
 
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
 
 // Try to use mono to save resources. Also avoids channel format conversion
 // in the I/O audio unit. Initial tests have shown that it is possible to use
 // mono natively for built-in microphones and for BT headsets but not for
 // wired headsets. Wired headsets only support stereo as native channel format
 // but it is a low cost operation to do a format conversion to mono in the
 // audio unit. Hence, we will not hit a RTC_CHECK in
 // VerifyAudioParametersForActiveAudioSession() for a mismatch between the
 // preferred number of channels and the actual number of channels.
 const int kRTCAudioSessionPreferredNumberOfChannels = 1;
@@ skipping 98 matching lines @@
   }
 }
 
 + (void)setWebRTCConfiguration:(RTCAudioSessionConfiguration *)configuration {
   @synchronized(self) {
     gWebRTCConfiguration = configuration;
   }
 }
 
 @end