Network thread

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 29 matching lines @@
 
 namespace webrtc {
 class AudioSinkInterface;
 }  // namespace webrtc
 
 namespace cricket {
 
 struct CryptoParams;
 class MediaContentDescription;
 
-enum SinkType {
-  SINK_PRE_CRYPTO,  // Sink packets before encryption or after decryption.
-  SINK_POST_CRYPTO  // Sink packets after encryption or before decryption.
-};
-
 // BaseChannel contains logic common to voice and video, including
 // enable, marshaling calls to a worker thread, and
 // connection and media monitors.
 //
 // WARNING! SUBCLASSES MUST CALL Deinit() IN THEIR DESTRUCTORS!
 // This is required to avoid a data race between the destructor modifying the
 // vtable, and the media channel's thread using BaseChannel as the
 // NetworkInterface.
 
 class BaseChannel
     : public rtc::MessageHandler, public sigslot::has_slots<>,
       public MediaChannel::NetworkInterface,
       public ConnectionStatsGetter {
  public:
-  BaseChannel(rtc::Thread* thread,
+  BaseChannel(rtc::Thread* worker_thread,
+              rtc::Thread* network_thread,
               MediaChannel* channel,
               TransportController* transport_controller,
               const std::string& content_name,
               bool rtcp);
   virtual ~BaseChannel();
   bool Init();
   // Deinit may be called multiple times and is simply ignored if it's alreay
   // 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_; }
   TransportChannel* transport_channel() const {
     return transport_channel_;
   }
   TransportChannel* rtcp_transport_channel() const {
     return rtcp_transport_channel_;
   }
   bool enabled() const { return enabled_; }
 
@@ skipping 45 matching lines @@
   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> SignalDtlsSetupFailure;
-  void SignalDtlsSetupFailure_w(bool rtcp);
+  void SignalDtlsSetupFailure_n(bool rtcp);
   void SignalDtlsSetupFailure_s(bool rtcp);
 
   // Used for latency measurements.
   sigslot::signal1<BaseChannel*> SignalFirstPacketReceived;
 
   // Made public for easier testing.
-  void SetReadyToSend(bool rtcp, bool ready);
+  void SetReadyToSend_n(bool rtcp, bool ready);
 
   // Only public for unit tests.  Otherwise, consider protected.
   int SetOption(SocketType type, rtc::Socket::Option o, int val)
       override;
 
   SrtpFilter* srtp_filter() { return &srtp_filter_; }
 
+  bool SetRtpTransportParameters_n(const MediaContentDescription* content,
+                                   ContentAction action,
+                                   ContentSource src,
+                                   std::string* error_desc);
+
  protected:
   virtual MediaChannel* media_channel() const { return media_channel_; }
   // Sets the |transport_channel_| (and |rtcp_transport_channel_|, if |rtcp_| is
   // true). Gets the transport channels from |transport_controller_|.
-  bool SetTransport_w(const std::string& transport_name);
+  bool SetTransport_n(const std::string& transport_name);
 
-  void set_transport_channel(TransportChannel* transport);
-  void set_rtcp_transport_channel(TransportChannel* transport,
+  void set_transport_channel(const std::string& transport_name);
+  void set_rtcp_transport_channel(const std::string* transport_name,
                                   bool update_writablity);
 
   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;
   }
   void set_secure_required(bool secure_required) {
@@ skipping 31 matching lines @@
   void OnSelectedCandidatePairChanged(
       TransportChannel* channel,
       CandidatePairInterface* selected_candidate_pair,
       int last_sent_packet_id);
 
   bool PacketIsRtcp(const TransportChannel* channel, const char* data,
                     size_t len);
   bool SendPacket(bool rtcp,
                   rtc::CopyOnWriteBuffer* packet,
                   const rtc::PacketOptions& options);
+
   virtual bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
-  void HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet,
+  void HandlePacket(bool rtcp,
+                    rtc::CopyOnWriteBuffer* packet,
                     const rtc::PacketTime& packet_time);
 
   void EnableMedia_w();
   void DisableMedia_w();
-  void UpdateWritableState_w();
-  void ChannelWritable_w();
-  void ChannelNotWritable_w();
+  void UpdateWritableState_n();
+  void ChannelWritable_n();
+  void ChannelNotWritable_n();
   bool AddRecvStream_w(const StreamParams& sp);
   bool RemoveRecvStream_w(uint32_t ssrc);
   bool AddSendStream_w(const StreamParams& sp);
   bool RemoveSendStream_w(uint32_t ssrc);
   virtual bool ShouldSetupDtlsSrtp() const;
   // Do the DTLS key expansion and impose it on the SRTP/SRTCP filters.
   // |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
-  bool SetupDtlsSrtp(bool rtcp_channel);
-  void MaybeSetupDtlsSrtp_w();
+  bool SetupDtlsSrtp_n(bool rtcp_channel);
+  void MaybeSetupDtlsSrtp_n();
   // Set the DTLS-SRTP cipher policy on this channel as appropriate.
   bool SetDtlsSrtpCryptoSuites(TransportChannel* tc, bool rtcp);
 
-  virtual void ChangeState() = 0;
+  void ChangeState_n();
+  virtual void ChangeState_w() = 0;
 
   // Gets the content info appropriate to the channel (audio or video).
   virtual const ContentInfo* GetFirstContent(
       const SessionDescription* sdesc) = 0;
   bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
                             ContentAction action,
                             std::string* error_desc);
   bool UpdateRemoteStreams_w(const std::vector<StreamParams>& streams,
                              ContentAction action,
                              std::string* error_desc);
   virtual bool SetLocalContent_w(const MediaContentDescription* content,
                                  ContentAction action,
                                  std::string* error_desc) = 0;
   virtual bool SetRemoteContent_w(const MediaContentDescription* content,
                                   ContentAction action,
                                   std::string* error_desc) = 0;
-  bool SetRtpTransportParameters_w(const MediaContentDescription* content,
-                                   ContentAction action,
-                                   ContentSource src,
-                                   std::string* error_desc);
 
   // Helper method to get RTP Absoulute SendTime extension header id if
   // present in remote supported extensions list.
   void MaybeCacheRtpAbsSendTimeHeaderExtension(
       const std::vector<RtpHeaderExtension>& extensions);
 
   bool CheckSrtpConfig(const std::vector<CryptoParams>& cryptos,
                        bool* dtls,
                        std::string* error_desc);
-  bool SetSrtp_w(const std::vector<CryptoParams>& params,
+  bool SetSrtp_n(const std::vector<CryptoParams>& params,
                  ContentAction action,
                  ContentSource src,
                  std::string* error_desc);
-  void ActivateRtcpMux_w();
-  bool SetRtcpMux_w(bool enable,
+  void ActivateRtcpMux_n();
+  bool SetRtcpMux_n(bool enable,
                     ContentAction action,
                     ContentSource src,
                     std::string* error_desc);
 
   // From MessageHandler
   void OnMessage(rtc::Message* pmsg) override;
 
   // Handled in derived classes
   // Get the SRTP crypto suites to use for RTP media
   virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const = 0;
   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 FunctorT& functor) {
     return worker_thread_->Invoke<bool>(functor);
   }
+  template <class FunctorT>
+  bool InvokeOnNetwork(const FunctorT& functor) {
+    return network_thread_->Invoke<bool>(functor);
+  }
 
  private:
   rtc::Thread* worker_thread_;
+  rtc::Thread* network_thread_;
   TransportController* transport_controller_;
   MediaChannel* media_channel_;
   std::vector<StreamParams> local_streams_;
   std::vector<StreamParams> remote_streams_;
 
   const std::string content_name_;
   std::string transport_name_;
   bool rtcp_transport_enabled_;
   TransportChannel* transport_channel_;
   std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
@@ skipping 13 matching lines @@
   bool has_received_packet_;
   bool dtls_keyed_;
   bool secure_required_;
   int rtp_abs_sendtime_extn_id_;
 };
 
 // VoiceChannel is a specialization that adds support for early media, DTMF,
 // and input/output level monitoring.
 class VoiceChannel : public BaseChannel {
  public:
-  VoiceChannel(rtc::Thread* thread,
+  VoiceChannel(rtc::Thread* worker_thread,
+               rtc::Thread* network_thread,
                MediaEngineInterface* media_engine,
                VoiceMediaChannel* channel,
                TransportController* transport_controller,
                const std::string& content_name,
                bool rtcp);
   ~VoiceChannel();
   bool Init();
 
   // Configure sending media on the stream with SSRC |ssrc|
   // If there is only one sending stream SSRC 0 can be used.
@@ skipping 48 matching lines @@
   void GetActiveStreams_w(AudioInfo::StreamList* actives);
   webrtc::RtpParameters GetRtpParameters_w(uint32_t ssrc) const;
   bool SetRtpParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);
 
  private:
   // overrides from BaseChannel
   virtual void OnChannelRead(TransportChannel* channel,
                              const char* data, size_t len,
                              const rtc::PacketTime& packet_time,
                              int flags);
-  virtual void ChangeState();
+  void ChangeState_w() override;
   virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
   virtual bool SetLocalContent_w(const MediaContentDescription* content,
                                  ContentAction action,
                                  std::string* error_desc);
   virtual bool SetRemoteContent_w(const MediaContentDescription* content,
                                   ContentAction action,
                                   std::string* error_desc);
   void HandleEarlyMediaTimeout();
   bool InsertDtmf_w(uint32_t ssrc, int event, int duration);
   bool SetOutputVolume_w(uint32_t ssrc, double volume);
@@ skipping 17 matching lines @@
   // SetSendParameters.
   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* thread,
+  VideoChannel(rtc::Thread* worker_thread,
+               rtc::Thread* netwokr_thread,
                VideoMediaChannel* channel,
                TransportController* transport_controller,
                const std::string& content_name,
                bool rtcp);
   ~VideoChannel();
   bool Init();
 
   // downcasts a MediaChannel
   virtual VideoMediaChannel* media_channel() const {
     return static_cast<VideoMediaChannel*>(BaseChannel::media_channel());
@@ skipping 13 matching lines @@
   void StartMediaMonitor(int cms);
   void StopMediaMonitor();
   sigslot::signal2<VideoChannel*, const VideoMediaInfo&> SignalMediaMonitor;
 
   bool SetVideoSend(uint32_t ssrc, bool enable, const VideoOptions* options);
   webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const;
   bool SetRtpParameters(uint32_t ssrc, const webrtc::RtpParameters& parameters);
 
  private:
   // overrides from BaseChannel
-  virtual void ChangeState();
+  void ChangeState_w() override;
   virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
   virtual bool SetLocalContent_w(const MediaContentDescription* content,
                                  ContentAction action,
                                  std::string* error_desc);
   virtual bool SetRemoteContent_w(const MediaContentDescription* content,
                                   ContentAction action,
                                   std::string* error_desc);
   bool GetStats_w(VideoMediaInfo* stats);
   webrtc::RtpParameters GetRtpParameters_w(uint32_t ssrc) const;
   bool SetRtpParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);
@@ skipping 11 matching lines @@
   // SetSendParameters.
   VideoSendParameters last_send_params_;
   // Last VideoRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   VideoRecvParameters last_recv_params_;
 };
 
 // DataChannel is a specialization for data.
 class DataChannel : public BaseChannel {
  public:
-  DataChannel(rtc::Thread* thread,
+  DataChannel(rtc::Thread* worker_thread,
+              rtc::Thread* network_thread,
               DataMediaChannel* media_channel,
               TransportController* transport_controller,
               const std::string& content_name,
               bool rtcp);
   ~DataChannel();
   bool Init();
 
   virtual bool SendData(const SendDataParams& params,
                         const rtc::CopyOnWriteBuffer& payload,
                         SendDataResult* result);
@@ skipping 66 matching lines @@
   // Same as SetDataChannelType, but extracts the type from the
   // DataContentDescription.
   bool SetDataChannelTypeFromContent(const DataContentDescription* content,
                                      std::string* error_desc);
   virtual bool SetLocalContent_w(const MediaContentDescription* content,
                                  ContentAction action,
                                  std::string* error_desc);
   virtual bool SetRemoteContent_w(const MediaContentDescription* content,
                                   ContentAction action,
                                   std::string* error_desc);
-  virtual void ChangeState();
-  virtual bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
+  void ChangeState_w() override;
+  bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet) override;
 
   virtual void OnMessage(rtc::Message* pmsg);
   virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
   virtual void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
   virtual void OnMediaMonitorUpdate(
       DataMediaChannel* media_channel, const DataMediaInfo& info);
   virtual bool ShouldSetupDtlsSrtp() const;
   void OnDataReceived(
       const ReceiveDataParams& params, const char* data, size_t len);
@@ skipping 11 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_