Move bundle filter from BaseChannel to RtpTransport.

webrtc/pc/channel.h

 /*
  *  Copyright 2004 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.
  */
 
@@ skipping 131 matching lines @@
   bool RemoveRecvStream(uint32_t ssrc);
   bool AddSendStream(const StreamParams& sp);
   bool RemoveSendStream(uint32_t ssrc);
 
   // Monitoring
   void StartConnectionMonitor(int cms);
   void StopConnectionMonitor();
   // For ConnectionStatsGetter, used by ConnectionMonitor
   bool GetConnectionStats(ConnectionInfos* infos) override;
 
-  BundleFilter* bundle_filter() { return &bundle_filter_; }
-
   const std::vector<StreamParams>& local_streams() const {
     return local_streams_;
   }
   const std::vector<StreamParams>& remote_streams() const {
     return remote_streams_;
   }
 
   sigslot::signal2<BaseChannel*, bool> SignalDtlsSrtpSetupFailure;
   void SignalDtlsSrtpSetupFailure_n(bool rtcp);
   void SignalDtlsSrtpSetupFailure_s(bool rtcp);
@@ skipping 27 matching lines @@
       override;
   int SetOption_n(SocketType type, rtc::Socket::Option o, int val);
 
   SrtpFilter* srtp_filter() { return &srtp_filter_; }
 
   virtual cricket::MediaType media_type() = 0;
 
   // This function returns true if we require SRTP for call setup.
   bool srtp_required_for_testing() const { return srtp_required_; }
 
+  // Public for testing.
+  bool HandlesPayloadType(int payload_type);

[Taylor Brandstetter, 2017/05/17 02:12:42]

const?

[Zach Stein, 2017/05/30 19:08:38]

Done.

+
  protected:
   virtual MediaChannel* media_channel() const { return media_channel_; }
 
   void SetTransports_n(DtlsTransportInternal* rtp_dtls_transport,
                        DtlsTransportInternal* rtcp_dtls_transport,
                        rtc::PacketTransportInternal* rtp_packet_transport,
                        rtc::PacketTransportInternal* rtcp_packet_transport);
 
   // This does not update writability or "ready-to-send" state; it just
   // disconnects from the old channel and connects to the new one.
@@ skipping 139 matching lines @@
   virtual void OnConnectionMonitorUpdate(ConnectionMonitor* monitor,
       const std::vector<ConnectionInfo>& infos) = 0;
 
   // Helper function for invoking bool-returning methods on the worker thread.
   template <class FunctorT>
   bool InvokeOnWorker(const rtc::Location& posted_from,
                       const FunctorT& functor) {
     return worker_thread_->Invoke<bool>(posted_from, functor);
   }
 
+  void AddHandledPayloadType(int payload_type);
+
  private:
   bool InitNetwork_n(DtlsTransportInternal* rtp_dtls_transport,
                      DtlsTransportInternal* rtcp_dtls_transport,
                      rtc::PacketTransportInternal* rtp_packet_transport,
                      rtc::PacketTransportInternal* rtcp_packet_transport);
   void DisconnectTransportChannels_n();
   void SignalSentPacket_n(rtc::PacketTransportInternal* transport,
                           const rtc::SentPacket& sent_packet);
   void SignalSentPacket_w(const rtc::SentPacket& sent_packet);
   bool IsReadyToSendMedia_n() const;
@@ skipping 17 matching lines @@
   // Separate DTLS/non-DTLS pointers to support using BaseChannel without DTLS.
   // Temporary measure until more refactoring is done.
   // If non-null, "X_dtls_transport_" will always equal "X_packet_transport_".
   DtlsTransportInternal* rtp_dtls_transport_ = nullptr;
   DtlsTransportInternal* rtcp_dtls_transport_ = nullptr;
   webrtc::RtpTransport rtp_transport_;
   std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
   std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
   SrtpFilter srtp_filter_;
   RtcpMuxFilter rtcp_mux_filter_;
-  BundleFilter bundle_filter_;
   bool writable_ = false;
   bool was_ever_writable_ = false;
   bool has_received_packet_ = false;
   bool dtls_keyed_ = false;
   const bool srtp_required_ = true;
   int rtp_abs_sendtime_extn_id_ = -1;
 
   // MediaChannel related members that should be accessed from the worker
   // thread.
   MediaChannel* const media_channel_;
@@ skipping 313 matching lines @@
   // SetSendParameters.
   DataSendParameters last_send_params_;
   // Last DataRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   DataRecvParameters last_recv_params_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_CHANNEL_H_