Introduce RtpStreamReceiver and RtpStreamReceiverControllerInterface.

webrtc/call/rtp_stream_receiver_controller.h

Index: webrtc/call/rtp_stream_receiver_controller.h
diff --git a/webrtc/call/rtp_stream_receiver_controller.h b/webrtc/call/rtp_stream_receiver_controller.h
new file mode 100644
index 0000000000000000000000000000000000000000..629f8a03de3ed59ed35cac366d203c3444ef8082
--- /dev/null
+++ b/webrtc/call/rtp_stream_receiver_controller.h
@@ -0,0 +1,67 @@
+/*
+ *  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/base/array_view.h"
+#include "webrtc/base/criticalsection.h"
+
+#include "webrtc/call/rtp_demuxer.h"
+#include "webrtc/call/rtp_stream_receiver_controller_interface.h"
+
+namespace webrtc {
+
+class ReceiveSideCongestionController;
+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:
+  // Implements RtpStreamReceiverControllerInterface.
+  std::unique_ptr<RtpStreamReceiver> CreateReceiver(
+      uint32_t ssrc,
+      RtpPacketSinkInterface* sink) override;
+
+  // Thread-safe wrappers for the corresponding RtpDemuxer methods.
+  void 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 RtpStreamReceiver {
+   public:
+    Receiver(RtpStreamReceiverController* controller,
+             uint32_t ssrc,
+             RtpPacketSinkInterface* sink);
+
+    ~Receiver() override;
+
+   private:
+    RtpStreamReceiverController* controller_;

[danilchap, 2017/06/13 15:58:32]

may be * const controller_; * const sink_
to stress they never change after creation.

[nisse-webrtc, 2017/06/14 06:31:06]

Done.

+    RtpPacketSinkInterface* sink_;
+  };
+
+  rtc::CriticalSection lock_;
+  RtpDemuxer demuxer_ GUARDED_BY(&lock_);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_