Remove BaseChannel's dependency on TransportController.

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 62 matching lines @@
 class BaseChannel
     : public rtc::MessageHandler, public sigslot::has_slots<>,
       public MediaChannel::NetworkInterface,
       public ConnectionStatsGetter {
  public:
   // |rtcp| represents whether or not this channel uses RTCP.
   // If |srtp_required| is true, the channel will not send or receive any
   // RTP/RTCP packets without using SRTP (either using SDES or DTLS-SRTP).
   BaseChannel(rtc::Thread* worker_thread,
               rtc::Thread* network_thread,
+              rtc::Thread* signaling_thread,
               MediaChannel* channel,
-              TransportController* transport_controller,
               const std::string& content_name,
               bool rtcp,
               bool srtp_required);
   virtual ~BaseChannel();
-  bool Init_w(const std::string* bundle_transport_name);
+  bool Init_w(TransportChannel* rtp_transport,
+              TransportChannel* rtcp_transport);
   // Deinit may be called multiple times and is simply ignored if it's already
   // done.
   void Deinit();
 
   rtc::Thread* worker_thread() const { return worker_thread_; }
   rtc::Thread* network_thread() const { return network_thread_; }
   const std::string& content_name() const { return content_name_; }
   const std::string& transport_name() const { return transport_name_; }
   bool enabled() const { return enabled_; }
 
   // 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_; }
 
   // Activate RTCP mux, regardless of the state so far.  Once
   // activated, it can not be deactivated, and if the remote
   // description doesn't support RTCP mux, setting the remote
   // description will fail.
   void ActivateRtcpMux();
-  bool SetTransport(const std::string& transport_name);
+  bool SetTransport(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 27 matching lines @@
   sigslot::signal2<BaseChannel*, bool> SignalDtlsSrtpSetupFailure;
   void SignalDtlsSrtpSetupFailure_n(bool rtcp);
   void SignalDtlsSrtpSetupFailure_s(bool rtcp);
 
   // Used for latency measurements.
   sigslot::signal1<BaseChannel*> SignalFirstPacketReceived;
 
   // Forward TransportChannel SignalSentPacket to worker thread.
   sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
 
-  // Only public for unit tests.  Otherwise, consider private.
-  TransportChannel* transport_channel() const { return transport_channel_; }
-  TransportChannel* rtcp_transport_channel() const {
-    return rtcp_transport_channel_;
-  }
+  // Emitted whenever the rtcp-mux is active and the rtcp-transport can be
+  // destroyed.

[Taylor Brandstetter, 2017/01/13 01:41:20]

Would be helpful to comment that this is fired on the network thread.

+  sigslot::signal1<const std::string&> SignalDestroyRtcpTransport;
+
+  TransportChannel* rtp_transport() const { return rtp_transport_; }
+  TransportChannel* rtcp_transport() const { return rtcp_transport_; }
+
+  bool NeedsRtcpTransport();
 
   // Made public for easier testing.
   //
   // Updates "ready to send" for an individual channel, and informs the media
   // channel that the transport is ready to send if each channel (in use) is
   // ready to send. This is more specific than just "writable"; it means the
   // last send didn't return ENOTCONN.
   //
   // This should be called whenever a channel's ready-to-send state changes,
   // or when RTCP muxing becomes active/inactive.
   void SetTransportChannelReadyToSend(bool rtcp, bool ready);
 
   // Only public for unit tests.  Otherwise, consider protected.
   int SetOption(SocketType type, rtc::Socket::Option o, int val)
       override;
   int SetOption_n(SocketType type, rtc::Socket::Option o, int val);
 
   SrtpFilter* srtp_filter() { return &srtp_filter_; }
 
   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 |transport_channel_| (and |rtcp_transport_channel_|, if
-  // |rtcp_enabled_| is true). Gets the transport channels from
-  // |transport_controller_|.
+  // 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(const std::string& transport_name);
+  bool SetTransport_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_channel);
+  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) {
     remote_content_direction_ = direction;
   }
   // These methods verify that:
   // * The required content description directions have been set.
   // * The channel is enabled.
   // * And for sending:
   //   - The SRTP filter is active if it's needed.
   //   - The transport has been writable before, meaning it should be at least
   //     possible to succeed in sending a packet.
   //
   // When any of these properties change, UpdateMediaSendRecvState_w should be
   // called.
   bool IsReadyToReceiveMedia_w() const;
   bool IsReadyToSendMedia_w() const;
-  rtc::Thread* signaling_thread() {
-    return transport_controller_->signaling_thread();
-  }
+  rtc::Thread* signaling_thread() { return signaling_thread_; }
 
   void ConnectToTransportChannel(TransportChannel* tc);
   void DisconnectFromTransportChannel(TransportChannel* tc);
 
   void FlushRtcpMessages_n();
 
   // NetworkInterface implementation, called by MediaEngine
   bool SendPacket(rtc::CopyOnWriteBuffer* packet,
                   const rtc::PacketOptions& options) override;
   bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
@@ skipping 114 matching lines @@
       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);
   }
 
  private:
-  bool InitNetwork_n(const std::string* bundle_transport_name);
+  bool InitNetwork_n(TransportChannel* rtp_transport,
+                     TransportChannel* rtcp_transport);
   void DisconnectTransportChannels_n();
-  void DestroyTransportChannels_n();
   void SignalSentPacket_n(rtc::PacketTransportInterface* transport,
                           const rtc::SentPacket& sent_packet);
   void SignalSentPacket_w(const rtc::SentPacket& sent_packet);
   bool IsReadyToSendMedia_n() const;
   void CacheRtpAbsSendTimeHeaderExtension_n(int rtp_abs_sendtime_extn_id);
   int GetTransportOverheadPerPacket() const;
   void UpdateTransportOverhead();
 
   rtc::Thread* const worker_thread_;
   rtc::Thread* const network_thread_;
+  rtc::Thread* const signaling_thread_;
   rtc::AsyncInvoker invoker_;
 
   const std::string content_name_;
   std::unique_ptr<ConnectionMonitor> connection_monitor_;
 
-  // Transport related members that should be accessed from network thread.
-  TransportController* const transport_controller_;
   std::string transport_name_;
   // Is RTCP used at all by this type of channel?
   // Expected to be true (as of typing this) for everything except data
   // channels.
   const bool rtcp_enabled_;
   // TODO(johan): Replace TransportChannel* with rtc::PacketTransportInterface*.
-  TransportChannel* transport_channel_ = nullptr;
+  TransportChannel* rtp_transport_ = nullptr;
   std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
-  TransportChannel* rtcp_transport_channel_ = nullptr;
+  TransportChannel* rtcp_transport_ = nullptr;
   std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
   SrtpFilter srtp_filter_;
   RtcpMuxFilter rtcp_mux_filter_;
   BundleFilter bundle_filter_;
   bool rtp_ready_to_send_ = false;
   bool rtcp_ready_to_send_ = false;
   bool writable_ = false;
   bool was_ever_writable_ = false;
   bool has_received_packet_ = false;
   bool dtls_keyed_ = false;
@@ skipping 14 matching lines @@
   MediaContentDirection remote_content_direction_ = MD_INACTIVE;
   CandidatePairInterface* selected_candidate_pair_;
 };
 
 // VoiceChannel is a specialization that adds support for early media, DTMF,
 // and input/output level monitoring.
 class VoiceChannel : public BaseChannel {
  public:
   VoiceChannel(rtc::Thread* worker_thread,
                rtc::Thread* network_thread,
+               rtc::Thread* signaling_thread,
                MediaEngineInterface* media_engine,
                VoiceMediaChannel* channel,
-               TransportController* transport_controller,
                const std::string& content_name,
                bool rtcp,
                bool srtp_required);
   ~VoiceChannel();
-  bool Init_w(const std::string* bundle_transport_name);
+  bool Init_w(TransportChannel* rtp_transport,
+              TransportChannel* rtcp_transport);
 
   // Configure sending media on the stream with SSRC |ssrc|
   // If there is only one sending stream SSRC 0 can be used.
   bool SetAudioSend(uint32_t ssrc,
                     bool enable,
                     const AudioOptions* options,
                     AudioSource* source);
 
   // downcasts a MediaChannel
   VoiceMediaChannel* media_channel() const override {
@@ skipping 90 matching lines @@
   AudioSendParameters last_send_params_;
   // Last AudioRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   AudioRecvParameters last_recv_params_;
 };
 
 // VideoChannel is a specialization for video.
 class VideoChannel : public BaseChannel {
  public:
   VideoChannel(rtc::Thread* worker_thread,
-               rtc::Thread* netwokr_thread,
+               rtc::Thread* network_thread,
+               rtc::Thread* signaling_thread,
                VideoMediaChannel* channel,
-               TransportController* transport_controller,
                const std::string& content_name,
                bool rtcp,
                bool srtp_required);
   ~VideoChannel();
-  bool Init_w(const std::string* bundle_transport_name);
+  bool Init_w(TransportChannel* rtp_transport,
+              TransportChannel* rtcp_transport);
 
   // downcasts a MediaChannel
   VideoMediaChannel* media_channel() const override {
     return static_cast<VideoMediaChannel*>(BaseChannel::media_channel());
   }
 
   bool SetSink(uint32_t ssrc,
                rtc::VideoSinkInterface<webrtc::VideoFrame>* sink);
   // Get statistics about the current media session.
   bool GetStats(VideoMediaInfo* stats);
@@ skipping 52 matching lines @@
   // Last VideoRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   VideoRecvParameters last_recv_params_;
 };
 
 // RtpDataChannel is a specialization for data.
 class RtpDataChannel : public BaseChannel {
  public:
   RtpDataChannel(rtc::Thread* worker_thread,
                  rtc::Thread* network_thread,
-                 DataMediaChannel* media_channel,
-                 TransportController* transport_controller,
+                 rtc::Thread* signaling_thread,
+                 DataMediaChannel* channel,
                  const std::string& content_name,
                  bool rtcp,
                  bool srtp_required);
   ~RtpDataChannel();
-  bool Init_w(const std::string* bundle_transport_name);
+  bool Init_w(TransportChannel* rtp_transport,
+              TransportChannel* rtcp_transport);
 
   virtual bool SendData(const SendDataParams& params,
                         const rtc::CopyOnWriteBuffer& payload,
                         SendDataResult* result);
 
   void StartMediaMonitor(int cms);
   void StopMediaMonitor();
 
   // Should be called on the signaling thread only.
   bool ready_to_send_data() const {
@@ skipping 82 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_