RtpPacketReceiver base class and OnRtpPacket, with a pre-parsed RTP packet.

webrtc/call/rtp_packet_receiver.h

+/*
+ *  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_PACKET_RECEIVER_H_
+#define WEBRTC_CALL_RTP_PACKET_RECEIVER_H_
+
+#include <vector>
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
+
+namespace webrtc {
+
+class RtpPacketReceiver {

[Taylor Brandstetter, 2017/02/09 20:20:11]

How will this relate to RtpTransportReceiver?

[nisse-webrtc, 2017/02/10 08:09:23]

RtpTransportReceiver would keep any media-independent per-ssrc state needed in
the receive processing. While RtpPacketReceiver should be implemented by
media-dependent receive streams, and one object per stream rather than by ssrc.

If/when we introduce RtpTransportReceiver, we can move the things returned by
rtp_config below to that object, and it won't need much more than the list of
extensions and pointer to an RtpPacketReceiver. But I'm starting to think that
maybe we don't need any RtpTransportReceiver, it should be enough to let
RtpTransportController have a map from ssrc to stream/RtpPacketReceiver. 

Depends on details such as whether or not we want different treatment of the
multiple ssrcs which are used by the same stream. If it needs to be separated,
RtpTransportReceiver is the way to add one more level of indirection.

+ public:
+  struct RtpConfig {
+    RtpConfig() = default;  // Needed by std::map
+    RtpConfig(const std::vector<RtpExtension>& extensions,
+           bool transport_cc)
+        : extensions(extensions),
+          use_send_side_bwe(UseSendSideBwe(extensions, transport_cc)) {}
+    // Registered RTP header extensions for a stream. Note that RTP header
+    // extensions are negotiated per track ("m= line") in the SDP, but we have
+    // no notion of tracks at the Call level. We therefore store the RTP header
+    // extensions per SSRC instead, which leads to some storage overhead.
+    RtpHeaderExtensionMap extensions;
+    // Set if both RTP extension the RTCP feedback message needed for
+    // send side BWE are negotiated.
+    bool use_send_side_bwe = false;
+  };
+  virtual bool OnRtpPacket(const RtpPacketReceived& packet) = 0;
+  virtual const RtpConfig& rtp_config() const = 0;
+  virtual ~RtpPacketReceiver() {}
+
+ private:
+  static bool UseSendSideBwe(const std::vector<RtpExtension>& extensions,
+                             bool transport_cc);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTP_PACKET_RECEIVER_H_