Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h

@@ -0,0 +1,97 @@
+/*
+ *  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/sdk/objc/Framework/Headers/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