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}

call/rtp_transport_controller_send.cc

@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2017 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.
+ */
+
+#include "webrtc/call/rtp_transport_controller_send.h"
+
+namespace webrtc {
+
+RtpTransportControllerSend::RtpTransportControllerSend(
+    Clock* clock,
+    webrtc::RtcEventLog* event_log)
+    : pacer_(clock, &packet_router_, event_log),
+      send_side_cc_(clock, nullptr /* observer */, event_log, &pacer_) {}
+
+PacketRouter* RtpTransportControllerSend::packet_router() {
+  return &packet_router_;
+}
+
+PacedSender* RtpTransportControllerSend::pacer() {
+  return &pacer_;
+}
+
+SendSideCongestionController* RtpTransportControllerSend::send_side_cc() {
+  return &send_side_cc_;
+}
+
+TransportFeedbackObserver*
+RtpTransportControllerSend::transport_feedback_observer() {
+  return &send_side_cc_;
+}
+
+RtpPacketSender* RtpTransportControllerSend::packet_sender() {
+  return &pacer_;
+}
+
+const RtpKeepAliveConfig& RtpTransportControllerSend::keepalive_config() const {
+  return keepalive_;
+}
+
+void RtpTransportControllerSend::SetAllocatedSendBitrateLimits(
+    int min_send_bitrate_bps,
+    int max_padding_bitrate_bps) {
+  pacer_.SetSendBitrateLimits(min_send_bitrate_bps, max_padding_bitrate_bps);
+}
+
+void RtpTransportControllerSend::SetKeepAliveConfig(
+    const RtpKeepAliveConfig& config) {
+  keepalive_ = config;
+}
+
+}  // namespace webrtc