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

webrtc/modules/audio_device/BUILD.gn

 # Copyright (c) 2014 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.gni")
 
@@ skipping 156 matching lines @@
           "AudioToolbox.framework",
           "CoreAudio.framework",
 
           # Needed for CGEventSourceKeyState in audio_device_mac.cc:
           "CoreGraphics.framework",
         ]
       }
       if (is_ios) {
         public_deps = [
           "../../base:rtc_base",
+          "../../sdk:rtc_sdk_audio_session",
           "../../sdk:rtc_sdk_common_objc",
         ]
         sources += [
           "ios/audio_device_ios.h",
           "ios/audio_device_ios.mm",
           "ios/audio_device_not_implemented_ios.mm",
           "ios/audio_session_observer.h",
-          "ios/objc/RTCAudioSession+Configuration.mm",
-          "ios/objc/RTCAudioSession+Private.h",
-          "ios/objc/RTCAudioSession.h",
-          "ios/objc/RTCAudioSession.mm",
-          "ios/objc/RTCAudioSessionConfiguration.h",
-          "ios/objc/RTCAudioSessionConfiguration.m",
           "ios/objc/RTCAudioSessionDelegateAdapter.h",
           "ios/objc/RTCAudioSessionDelegateAdapter.mm",
           "ios/voice_processing_audio_unit.h",
           "ios/voice_processing_audio_unit.mm",
         ]
         libs = [
           "AudioToolbox.framework",
           "AVFoundation.framework",
           "Foundation.framework",
           "UIKit.framework",
@@ skipping 81 matching lines @@
         "android/audio_manager_unittest.cc",
         "android/ensure_initialized.cc",
         "android/ensure_initialized.h",
       ]
       deps += [
         "../../../base",
         "//webrtc/sdk/android:libjingle_peerconnection_java",
       ]
     }
     if (is_ios) {
-      sources += [ "ios/objc/RTCAudioSessionTest.mm" ]
       if (target_cpu != "x64") {
         sources += [ "ios/audio_device_unittest_ios.cc" ]
       }
       deps += [ "//third_party/ocmock" ]
     }
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
@@ skipping 44 matching lines @@
       "android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java",
       "android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java",
       "android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java",
       "android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java",
     ]
     deps = [
       "//webrtc/base:base_java",
     ]
   }
 }