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/Headers/WebRTC/RTCRtpReceiver.h

@@ -0,0 +1,82 @@
+/*
+ *  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 <Foundation/Foundation.h>
+
+#import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCMediaStreamTrack.h>
+#import <WebRTC/RTCRtpParameters.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Represents the media type of the RtpReceiver. */
+typedef NS_ENUM(NSInteger, RTCRtpMediaType) {
+  RTCRtpMediaTypeAudio,
+  RTCRtpMediaTypeVideo,
+  RTCRtpMediaTypeData,
+};
+
+@class RTCRtpReceiver;
+
+RTC_EXPORT
+@protocol RTCRtpReceiverDelegate <NSObject>
+
+/** Called when the first RTP packet is received.
+ *
+ *  Note: Currently if there are multiple RtpReceivers of the same media type,
+ *  they will all call OnFirstPacketReceived at once.
+ *
+ *  For example, if we create three audio receivers, A/B/C, they will listen to
+ *  the same signal from the underneath network layer. Whenever the first audio packet
+ *  is received, the underneath signal will be fired. All the receivers A/B/C will be
+ *  notified and the callback of the receiver's delegate will be called.
+ *
+ *  The process is the same for video receivers.
+ */
+- (void)rtpReceiver:(RTCRtpReceiver *)rtpReceiver
+    didReceiveFirstPacketForMediaType:(RTCRtpMediaType)mediaType;
+
+@end
+
+RTC_EXPORT
+@protocol RTCRtpReceiver <NSObject>
+
+/** A unique identifier for this receiver. */
+@property(nonatomic, readonly) NSString *receiverId;
+
+/** The currently active RTCRtpParameters, as defined in
+ *  https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
+ *
+ *  The WebRTC specification only defines RTCRtpParameters in terms of senders,
+ *  but this API also applies them to receivers, similar to ORTC:
+ *  http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
+ */
+@property(nonatomic, readonly) RTCRtpParameters *parameters;
+
+/** The RTCMediaStreamTrack associated with the receiver.
+ *  Note: reading this property returns a new instance of
+ *  RTCMediaStreamTrack. Use isEqual: instead of == to compare
+ *  RTCMediaStreamTrack instances.
+ */
+@property(nonatomic, readonly) RTCMediaStreamTrack *track;
+
+/** The delegate for this RtpReceiver. */
+@property(nonatomic, weak) id<RTCRtpReceiverDelegate> delegate;
+
+@end
+
+RTC_EXPORT
+@interface RTCRtpReceiver : NSObject <RTCRtpReceiver>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END