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/PeerConnection/RTCRtpSender.mm

@@ -0,0 +1,94 @@
+/*
+ *  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 "RTCRtpSender+Private.h"
+
+#import "NSString+StdString.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCRtpParameters+Private.h"
+#import "WebRTC/RTCLogging.h"
+
+#include "webrtc/api/mediastreaminterface.h"
+
+@implementation RTCRtpSender {
+  rtc::scoped_refptr<webrtc::RtpSenderInterface> _nativeRtpSender;
+}
+
+- (NSString *)senderId {
+  return [NSString stringForStdString:_nativeRtpSender->id()];
+}
+
+- (RTCRtpParameters *)parameters {
+  return [[RTCRtpParameters alloc]
+      initWithNativeParameters:_nativeRtpSender->GetParameters()];
+}
+
+- (void)setParameters:(RTCRtpParameters *)parameters {
+  if (!_nativeRtpSender->SetParameters(parameters.nativeParameters)) {
+    RTCLogError(@"RTCRtpSender(%p): Failed to set parameters: %@", self,
+        parameters);
+  }
+}
+
+- (RTCMediaStreamTrack *)track {
+  rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack(
+    _nativeRtpSender->track());
+  if (nativeTrack) {
+    return [[RTCMediaStreamTrack alloc] initWithNativeTrack:nativeTrack];
+  }
+  return nil;
+}
+
+- (void)setTrack:(RTCMediaStreamTrack *)track {
+  if (!_nativeRtpSender->SetTrack(track.nativeTrack)) {
+    RTCLogError(@"RTCRtpSender(%p): Failed to set track %@", self, track);
+  }
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCRtpSender {\n  senderId: %@\n}",
+      self.senderId];
+}
+
+- (BOOL)isEqual:(id)object {
+  if (self == object) {
+    return YES;
+  }
+  if (object == nil) {
+    return NO;
+  }
+  if (![object isMemberOfClass:[self class]]) {
+    return NO;
+  }
+  RTCRtpSender *sender = (RTCRtpSender *)object;
+  return _nativeRtpSender == sender.nativeRtpSender;
+}
+
+- (NSUInteger)hash {
+  return (NSUInteger)_nativeRtpSender.get();
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
+  return _nativeRtpSender;
+}
+
+- (instancetype)initWithNativeRtpSender:
+    (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
+  NSParameterAssert(nativeRtpSender);
+  if (self = [super init]) {
+    _nativeRtpSender = nativeRtpSender;
+    RTCLogInfo(@"RTCRtpSender(%p): created sender: %@", self, self.description);
+  }
+  return self;
+}
+
+@end