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.h

@@ -0,0 +1,72 @@
+/*
+ *  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_H_
+#define WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_
+
+#include <memory>
+
+#include "webrtc/call/rtp_demuxer.h"
+#include "webrtc/call/rtp_stream_receiver_controller_interface.h"
+#include "webrtc/rtc_base/criticalsection.h"
+
+namespace webrtc {
+
+class RtpPacketReceived;
+
+// This class represents the RTP receive parsing and demuxing, for a
+// single RTP session.
+// TODO(nisse): Add RTCP processing, we should aim to terminate RTCP
+// and not leave any RTCP processing to individual receive streams.
+// TODO(nisse): Extract per-packet processing, including parsing and
+// demuxing, into a separate class.
+class RtpStreamReceiverController
+    : public RtpStreamReceiverControllerInterface {
+ public:
+  RtpStreamReceiverController();
+  ~RtpStreamReceiverController() override;
+
+  // Implements RtpStreamReceiverControllerInterface.
+  std::unique_ptr<RtpStreamReceiverInterface> CreateReceiver(
+      uint32_t ssrc,
+      RtpPacketSinkInterface* sink) override;
+
+  // Thread-safe wrappers for the corresponding RtpDemuxer methods.
+  bool AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) override;
+  size_t RemoveSink(const RtpPacketSinkInterface* sink) override;
+
+  // TODO(nisse): Not yet responsible for parsing.
+  bool OnRtpPacket(const RtpPacketReceived& packet);
+
+ private:
+  class Receiver : public RtpStreamReceiverInterface {
+   public:
+    Receiver(RtpStreamReceiverController* controller,
+             uint32_t ssrc,
+             RtpPacketSinkInterface* sink);
+
+    ~Receiver() override;
+
+   private:
+    RtpStreamReceiverController* const controller_;
+    RtpPacketSinkInterface* const sink_;
+  };
+
+  // TODO(nisse): Move to a TaskQueue for synchronization. When used
+  // by Call, we expect construction and all methods but OnRtpPacket
+  // to be called on the same thread, and OnRtpPacket to be called
+  // by a single, but possibly distinct, thread. But applications not
+  // using Call may have use threads differently.
+  rtc::CriticalSection lock_;
+  RtpDemuxer demuxer_ RTC_GUARDED_BY(&lock_);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_