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

webrtc/call/rtp_packet_receiver.h

Index: webrtc/call/rtp_packet_receiver.h
diff --git a/webrtc/call/rtp_packet_receiver.h b/webrtc/call/rtp_packet_receiver.h
new file mode 100644
index 0000000000000000000000000000000000000000..0561ca35002d8728751f144345e1a05822310a49
--- /dev/null
+++ b/webrtc/call/rtp_packet_receiver.h
@@ -0,0 +1,49 @@
+/*
+ *  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 {
+ public:
+  struct RtpConfig {
+    RtpConfig() = default;  // Needed by std::map
+    RtpConfig(const std::vector<RtpExtension>& extensions,

[the sun, 2017/02/10 12:37:45]

nit: move whole impl of this ctor to .cc, then you can put UseSendSideBwe() in
an anonymous namespace in the .cc and avoid cluttering the interface.

+           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_