Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.

webrtc/pc/rtptransport.h

Index: webrtc/pc/rtptransport.h
diff --git a/webrtc/pc/rtptransport.h b/webrtc/pc/rtptransport.h
index f9bee1b6cc7da03d6ad5a93680b0a70f004bf8cd..900865a0eae0975e15bdc603ccb2652f94308bca 100644
--- a/webrtc/pc/rtptransport.h
+++ b/webrtc/pc/rtptransport.h
@@ -13,11 +13,13 @@
 
 #include "webrtc/api/ortc/rtptransportinterface.h"
 #include "webrtc/base/sigslot.h"
+#include "webrtc/pc/bundlefilter.h"
 
 namespace rtc {
 
 class CopyOnWriteBuffer;
 struct PacketOptions;
+struct PacketTime;
 class PacketTransportInternal;
 
 }  // namespace rtc
@@ -64,11 +66,24 @@ class RtpTransport : public RtpTransportInterface, public sigslot::has_slots<> {
                   const rtc::PacketOptions& options,
                   int flags);
 
+  bool HandlesPayloadType(int payload_type) const;
+
+  void AddHandledPayloadType(int payload_type);
+
+  bool received_media() const { return received_media_; }
+
+  sigslot::signal0<> SignalFirstPacketReceived;
+
+  sigslot::signal3<bool, rtc::CopyOnWriteBuffer&, const rtc::PacketTime&>
+      SignalPacketReceived;
+
  protected:
   // TODO(zstein): Remove this when we remove RtpTransportAdapter.
   RtpTransportAdapter* GetInternal() override;
 
  private:
+  bool HandlesPacket(const uint8_t* data, size_t len);
+
   void OnReadyToSend(rtc::PacketTransportInternal* transport);
 
   // Updates "ready to send" for an individual channel and fires
@@ -77,6 +92,18 @@ class RtpTransport : public RtpTransportInterface, public sigslot::has_slots<> {
 
   void MaybeSignalReadyToSend();
 
+  void OnReadPacket(rtc::PacketTransportInternal* transport,
+                    const char* data,
+                    size_t len,
+                    const rtc::PacketTime& packet_time,
+                    int flags);
+
+  bool PacketIsRtcp(const rtc::PacketTransportInternal* transport,
+                    const char* data,
+                    size_t len);
+
+  bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
+
   bool rtcp_mux_enabled_;
 
   rtc::PacketTransportInternal* rtp_packet_transport_ = nullptr;
@@ -87,6 +114,11 @@ class RtpTransport : public RtpTransportInterface, public sigslot::has_slots<> {
   bool rtcp_ready_to_send_ = false;
 
   RtcpParameters rtcp_parameters_;
+
+  cricket::BundleFilter bundle_filter_;
+
+  bool has_received_packet_ = false;
+  bool received_media_ = false;
 };
 
 }  // namespace webrtc