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

webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm

 /*
  *  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/voice_processing_audio_unit.h"
 
 #include "webrtc/base/checks.h"
 
 #import "WebRTC/RTCLogging.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
+#import "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h"
 
 #if !defined(NDEBUG)
 static void LogStreamDescription(AudioStreamBasicDescription description) {
   char formatIdString[5];
   UInt32 formatId = CFSwapInt32HostToBig(description.mFormatID);
   bcopy(&formatId, formatIdString, 4);
   formatIdString[4] = '\0';
   RTCLog(@"AudioStreamBasicDescription: {\n"
           "  mSampleRate: %.2f\n"
           "  formatIDString: %s\n"
@@ skipping 357 matching lines @@
     OSStatus result = AudioComponentInstanceDispose(vpio_unit_);
     if (result != noErr) {
       RTCLogError(@"AudioComponentInstanceDispose failed. Error=%ld.",
                   (long)result);
     }
     vpio_unit_ = nullptr;
   }
 }
 
 }  // namespace webrtc