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_stream_receiver_controller_interface.h

@@ -0,0 +1,47 @@
+/*
+ *  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.
+ */
+#ifndef WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
+#define WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
+
+#include <memory>
+
+#include "webrtc/call/rtp_packet_sink_interface.h"
+
+namespace webrtc {
+
+// An RtpStreamReceiver is responsible for the rtp-specific but
+// media-independent state needed for receiving an RTP stream.
+// TODO(nisse): Currently, only owns the association between ssrc and
+// the stream's RtpPacketSinkInterface. Ownership of corresponding
+// objects from modules/rtp_rtcp/ should move to this class (or
+// rather, the corresponding implementation class). We should add
+// methods for getting rtp receive stats, and for sending RTCP
+// messages related to the receive stream.
+class RtpStreamReceiverInterface {
+ public:
+  virtual ~RtpStreamReceiverInterface() {}
+};
+
+// This class acts as a factory for RtpStreamReceiver objects.
+class RtpStreamReceiverControllerInterface {
+ public:
+  virtual ~RtpStreamReceiverControllerInterface() {}
+
+  virtual std::unique_ptr<RtpStreamReceiverInterface> CreateReceiver(
+      uint32_t ssrc,
+      RtpPacketSinkInterface* sink) = 0;
+  // For registering additional sinks, needed for FlexFEC.
+  virtual bool AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) = 0;
+  virtual size_t RemoveSink(const RtpPacketSinkInterface* sink) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_