Initial implementation of RtpTransportControllerReceive and related interfaces.

webrtc/call/rtp_transport_controller_receive.h

Index: webrtc/call/rtp_transport_controller_receive.h
diff --git a/webrtc/call/rtp_transport_controller_receive.h b/webrtc/call/rtp_transport_controller_receive.h
new file mode 100644
index 0000000000000000000000000000000000000000..961519374bbba59165af981d1a901d9f6ef9dcf7
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+++ b/webrtc/call/rtp_transport_controller_receive.h
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+/*
+ *  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_H_
+#define WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_H_
+
+#include <memory>
+
+#include "webrtc/base/array_view.h"
+
+// For DeliveryStatus.
+// TODO(nisse): This file ought to not depend on call.
+#include "webrtc/call/call.h"
+
+namespace webrtc {
+
+class ReceiveSideCongestionController;
+class RtpPacketReceived;
+
+// This class represents a receiver of already parsed RTP packets.
+class RtpPacketSinkInterface {
+ public:
+  virtual void OnRtpPacket(const RtpPacketReceived& packet) = 0;
+  virtual ~RtpPacketSinkInterface() {}

[danilchap, 2017/04/19 15:45:25]

in all 3 classes in this file prefer default order: Nested type, factory method,
destructor, other methods.
(https://google.github.io/styleguide/cppguide.html#Declaration_Order)

[nisse-webrtc, 2017/04/20 10:53:38]

Done.

+};
+
+// TODO(nisse): This additional interface may be a bit confusing, it's
+// needed because RTP extensions are currently associated with each
+// stream. We should be able to identify extensions earlier, since the
+// mapping is the same for all streams on the same transport. Then we
+// can delete this class, replacing it with RtpPacketSinkInterface.
+// There's also the subtlety that the per-transport negotiation may
+// differ from the extensions expected for a particular stream, and we
+// may have code that breaks if we identify extension which is not
+// consistent with the configuration of the stream.
+
+// This class represents a receiver of RTP packets, which has the
+// additional responsibility of identifying extension headers. I.e.,
+// the caller of OnRtpPacketReceive is expected to have called
+// packet->Parse, but not packet->IdentifyExtensions.
+class RtpPacketReceiverInterface {
+ public:
+  virtual bool OnRtpPacketReceive(RtpPacketReceived* packet) = 0;
+  virtual ~RtpPacketReceiverInterface() {}
+};
+
+// This class represents the RTP receive parsing and demuxing, for a
+// single transport. It isn't thread aware, leaving responsibility of
+// multithreading issues to the user of this class. TODO(nisse): Add
+// RTCP processing, we should aim to terminate RTCP and not leave any
+// RTCP processing to individual receive streams. TODO(nisse): Extract
+// parsing and demuxing logic into separate classes. A demuxer should
+// handle only a single transport.
+class RtpTransportControllerReceiveInterface {
+ public:
+  // Configuration needed for media-independent processing of received
+  // packets, in particular, feeding information to the congestion
+  // controller.
+
+  // TODO(nisse): It turns out this data is only used for the callback
+  // to the ReceiveSideCongestionController.
+  struct Config {

[danilchap, 2017/04/19 15:45:25]

is there plan to make it TransportController config? (then Create factory should
take the parameter) or just Receiver's config (then likely rename to something
like StreamConfig/ChannelConfig/SinkConfig)

+    // See draft-holmer-rmcat-transport-wide-cc-extensions for details.
+    bool use_send_side_bwe = false;
+  };
+
+  // Registers the receiver responsible for an ssrc.
+  virtual void AddReceiver(uint32_t ssrc,
+                           const Config& config,
+                           RtpPacketReceiverInterface* receiver) = 0;
+  // TODO(nisse): It's unclear what info is conveniently available at
+  // remove time. For now, we take only the |receiver| pointer and
+  // iterate over the mapping.
+  virtual void RemoveReceiver(const RtpPacketReceiverInterface* receiver) = 0;
+
+  // Used to represent auxillary sinks, currently used for FlexFec.
+  // The responsible receiver must be registered first.
+  virtual void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) = 0;
+  // TODO(nisse): It's unclear what info is conveniently available at
+  // remove time. For now, we take only the |receiver| pointer and
+  // iterate over the mapping.
+  virtual void RemoveSink(const RtpPacketSinkInterface* sink) = 0;
+
+#if 0

[danilchap, 2017/04/19 15:45:25]

better to remove for now and add when it will be used.
The interface might be different than current plan and it would be easier to
review which one is better when it will be in use.

[nisse-webrtc, 2017/04/20 10:53:38]

Done.

+  // TODO(nisse): Not yet implemented. We also need something similar to
+  // handle the mid extension.
+  // Incoming packets with unknown ssrcs represent unsignalled
+  // streams. We dispatch on payload type and media type. The receiver
+  // will typically create a new RtpReceiver to pass the packet to. A
+  // true return value from the callback means that we should retry
+  // lookup.
+  virtual void AddPayload(uint8_t payload_type,
+                          RtpPacketReceiverInterface *receiver) = 0;
+#endif
+  // Process raw incoming RTP packets.
+  // TODO(nisse): DeliveryStatus is needed for the current handling of
+  // unsignalled ssrcs. Change return type to bool or void, once we do
+  // that via AddPayload instead.
+  virtual PacketReceiver::DeliveryStatus OnRtpPacket(
+      int64_t arrival_time_ms,
+      rtc::ArrayView<const uint8_t> packet) = 0;
+
+  virtual ~RtpTransportControllerReceiveInterface() {}
+
+  // Creates the default implementation.
+  static std::unique_ptr<RtpTransportControllerReceiveInterface> Create(
+      ReceiveSideCongestionController* receive_side_cc,
+      bool enable_receive_side_bwe);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_H_