Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.

webrtc/pc/rtptransport.h

 /*
  *  Copyright 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_PC_RTPTRANSPORT_H_
 #define WEBRTC_PC_RTPTRANSPORT_H_
 
 #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
 
 namespace webrtc {
 
 class RtpTransport : public RtpTransportInterface, public sigslot::has_slots<> {
  public:
   RtpTransport(const RtpTransport&) = delete;
   RtpTransport& operator=(const RtpTransport&) = delete;
@@ skipping 26 matching lines @@
   // than just "writable"; it means the last send didn't return ENOTCONN.
   sigslot::signal1<bool> SignalReadyToSend;
 
   bool IsWritable(bool rtcp) const;
 
   bool SendPacket(bool rtcp,
                   const rtc::CopyOnWriteBuffer* packet,
                   const rtc::PacketOptions& options,
                   int flags);
 
+  bool HandlesPayloadType(int payload_type) const;
+
+  void AddHandledPayloadType(int payload_type);
+
+  // TODO(zstein): Consider having two signals - RtcPacketReceived and
+  // RtcpPacketReceived.
+  // The first argument is true for RTCP packets and false for RTP packets.
+  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
   // SignalReadyToSend.
   void SetReadyToSend(bool rtcp, bool ready);
 
   void MaybeSignalReadyToSend();
 
+  void OnReadPacket(rtc::PacketTransportInternal* transport,
+                    const char* data,
+                    size_t len,
+                    const rtc::PacketTime& packet_time,
+                    int flags);
+
+  bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
+
   bool rtcp_mux_enabled_;
 
   rtc::PacketTransportInternal* rtp_packet_transport_ = nullptr;
   rtc::PacketTransportInternal* rtcp_packet_transport_ = nullptr;
 
   bool ready_to_send_ = false;
   bool rtp_ready_to_send_ = false;
   bool rtcp_ready_to_send_ = false;
 
   RtcpParameters rtcp_parameters_;
+
+  cricket::BundleFilter bundle_filter_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_PC_RTPTRANSPORT_H_