Introduce RtpStreamReceiver and RtpStreamReceiverControllerInterface.

webrtc/call/rtp_transport_controller_receive_interface.h

Index: webrtc/call/rtp_transport_controller_receive_interface.h
diff --git a/webrtc/call/rtp_transport_controller_receive_interface.h b/webrtc/call/rtp_transport_controller_receive_interface.h
new file mode 100644
index 0000000000000000000000000000000000000000..420b0e4f5b0f76ba2cb4b9e2ed021a96e304af3f
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+++ b/webrtc/call/rtp_transport_controller_receive_interface.h
@@ -0,0 +1,46 @@
+/*
+ *  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_TRANSPORT_CONTROLLER_RECEIVE_INTERFACE_H_
+#define WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_INTERFACE_H_
+
+#include <memory>
+
+#include "webrtc/call/rtp_packet_sink_interface.h"
+
+namespace webrtc {
+
+// An RtpTransportReceiver is responsible for the rtp-specific but
+// media-independent state needed for receiving a stream.

[pthatcher, 2017/06/02 22:44:17]

"stream" is ambiguous in the RTP world.  Needs to be "RTP stream" or "Encoded
Stream" or something else.

+// 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 implmentation class), and we should add
+// have methods for getting rtp receive stats.
+class RtpTransportReceiver {
+ public:
+  virtual ~RtpTransportReceiver() {}
+};
+
+// This class acts as a factory for RtpTransportReceive objects.
+class RtpTransportControllerReceiveInterface {
+ public:
+  virtual ~RtpTransportControllerReceiveInterface() {}
+
+  virtual std::unique_ptr<RtpTransportReceiver> CreateReceiver(
+      uint32_t ssrc,
+      RtpPacketSinkInterface* sink) = 0;
+  // For registering additional sinks, needed for FlexFEC.
+  virtual void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) = 0;
+  virtual size_t RemoveSink(const RtpPacketSinkInterface* sink) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_INTERFACE_H_