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/RTCAudioTrack.mm

@@ -0,0 +1,66 @@
+/*
+ *  Copyright 2015 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 "RTCAudioTrack+Private.h"
+
+#import "NSString+StdString.h"
+#import "RTCAudioSource+Private.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCPeerConnectionFactory+Private.h"
+
+#include "webrtc/rtc_base/checks.h"
+
+@implementation RTCAudioTrack
+
+@synthesize source = _source;
+
+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
+                         source:(RTCAudioSource *)source
+                        trackId:(NSString *)trackId {
+  RTC_DCHECK(factory);
+  RTC_DCHECK(source);
+  RTC_DCHECK(trackId.length);
+
+  std::string nativeId = [NSString stdStringForString:trackId];
+  rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
+      factory.nativeFactory->CreateAudioTrack(nativeId, source.nativeAudioSource);
+  if ([self initWithNativeTrack:track type:RTCMediaStreamTrackTypeAudio]) {
+    _source = source;
+  }
+  return self;
+}
+
+- (instancetype)initWithNativeTrack:
+    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+                               type:(RTCMediaStreamTrackType)type {
+  NSParameterAssert(nativeTrack);
+  NSParameterAssert(type == RTCMediaStreamTrackTypeAudio);
+  return [super initWithNativeTrack:nativeTrack type:type];
+}
+
+
+- (RTCAudioSource *)source {
+  if (!_source) {
+    rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
+        self.nativeAudioTrack->GetSource();
+    if (source) {
+      _source = [[RTCAudioSource alloc] initWithNativeAudioSource:source.get()];
+    }
+  }
+  return _source;
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::AudioTrackInterface>)nativeAudioTrack {
+  return static_cast<webrtc::AudioTrackInterface *>(self.nativeTrack.get());
+}
+
+@end