More minor improvements to BaseChannel/transport code.

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 89 matching lines @@
 
   // This function returns true if we are using SRTP.
   bool secure() const { return srtp_filter_.IsActive(); }
   // The following function returns true if we are using
   // DTLS-based keying. If you turned off SRTP later, however
   // you could have secure() == false and dtls_secure() == true.
   bool secure_dtls() const { return dtls_keyed_; }
 
   bool writable() const { return writable_; }
 
-  bool SetTransport(TransportChannel* rtp_transport,
-                    TransportChannel* rtcp_transport);
+  // Set the transport(s), and update writability and "ready-to-send" state.
+  // |rtp_transport| must be non-null.
+  // |rtcp_transport| must be supplied if NeedsRtcpTransport() is true (meaning
+  // RTCP muxing is not fully active yet).
+  // |rtp_transport| and |rtcp_transport| must share the same transport name as
+  // well.
+  void SetTransports(TransportChannel* rtp_transport,
+                     TransportChannel* rtcp_transport);
   bool PushdownLocalDescription(const SessionDescription* local_desc,
                                 ContentAction action,
                                 std::string* error_desc);
   bool PushdownRemoteDescription(const SessionDescription* remote_desc,
                                 ContentAction action,
                                 std::string* error_desc);
   // Channel control
   bool SetLocalContent(const MediaContentDescription* content,
                        ContentAction action,
                        std::string* error_desc);
@@ skipping 65 matching lines @@
   virtual cricket::MediaType media_type() = 0;
 
   bool SetCryptoOptions(const rtc::CryptoOptions& crypto_options);
 
   // This function returns true if we require SRTP for call setup.
   bool srtp_required_for_testing() const { return srtp_required_; }
 
  protected:
   virtual MediaChannel* media_channel() const { return media_channel_; }
 
-  // Sets the |rtp_transport_| (and |rtcp_transport_|, if
-  // |rtcp_enabled_| is true).
-  // This method also updates writability and "ready-to-send" state.
-  bool SetTransport_n(TransportChannel* rtp_transport,
-                      TransportChannel* rtcp_transport);
+  void SetTransports_n(TransportChannel* rtp_transport,
+                       TransportChannel* rtcp_transport);
 
   // This does not update writability or "ready-to-send" state; it just
   // disconnects from the old channel and connects to the new one.
   void SetTransportChannel_n(bool rtcp, TransportChannel* new_transport);
 
   bool was_ever_writable() const { return was_ever_writable_; }
   void set_local_content_direction(MediaContentDirection direction) {
     local_content_direction_ = direction;
   }
   void set_remote_content_direction(MediaContentDirection direction) {
@@ skipping 509 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_