New class RtpDemuxer and RtpPacketSinkInterface, use in Call.

webrtc/call/rtp_demuxer.h

Index: webrtc/call/rtp_demuxer.h
diff --git a/webrtc/call/rtp_demuxer.h b/webrtc/call/rtp_demuxer.h
new file mode 100644
index 0000000000000000000000000000000000000000..859cb3d2792194d72af4546f424c1b66a5f9e3cb
--- /dev/null
+++ b/webrtc/call/rtp_demuxer.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_DEMUXER_H_
+#define WEBRTC_CALL_RTP_DEMUXER_H_
+
+#include <map>
+
+namespace webrtc {
+
+class RtpPacketReceived;
+
+// This class represents a receiver of an already parsed RTP packets.
+class RtpPacketSinkInterface {
+ public:
+  virtual ~RtpPacketSinkInterface() {}
+  virtual void OnRtpPacket(const RtpPacketReceived& packet) = 0;
+};
+
+// This class represents the RTP demuxing, for a single transport. It
+// isn't thread aware, leaving responsibility of multithreading issues
+// to the user of this class.

[Taylor Brandstetter, 2017/05/10 20:59:43]

Could you mention in a comment that this should also eventually handle payload
type and MID-based demuxing? Payload type based demuxing is currently handled in
BaseChannel, which we're refactoring out into RtpTransport. Which hopefully in
the future will deliver packets directly to RtpDemuxer.

[nisse-webrtc, 2017/05/12 08:50:08]

I'm adding a TODO. Have I got this right, that payload type and MID demuxing is
relevant only for unsignalled ssrcs?

[Taylor Brandstetter, 2017/05/12 16:36:17]

Correct. Though JSEP no longer requires SSRC signaling, so this is something
we'll likely have to deal with soon.

+class RtpDemuxer {
+ public:
+  RtpDemuxer();
+  ~RtpDemuxer();
+
+  // Registers a sink. The same sink can be registered for multiple ssrcs.
+  void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink);
+  // Removes a sink. Returns deletion count (a sink may be registered
+  // for multiple ssrcs).
+  size_t RemoveSink(const RtpPacketSinkInterface* sink);
+
+  // Returns true if at least one matching sink was found, otherwise false.
+  bool OnRtpPacket(const RtpPacketReceived& packet);
+
+ private:
+  std::multimap<uint32_t, RtpPacketSinkInterface*> sinks_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTP_DEMUXER_H_