Add support for GCM cipher suites from RFC 7714.

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 140 matching lines @@
 
   // Made public for easier testing.
   void SetReadyToSend(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 SetCryptoOptions(const rtc::CryptoOptions& crypto_options);
+
  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);
 
   void set_transport_channel(TransportChannel* transport);
   void set_rtcp_transport_channel(TransportChannel* transport,
                                   bool update_writablity);
 
@@ skipping 105 matching lines @@
   bool SetRtcpMux_w(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 GetSrtpCryptoSuites(const rtc::CryptoOptions& crypto_options,
+      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);
   }
 
  private:
@@ skipping 17 matching lines @@
   bool enabled_;
   bool writable_;
   bool rtp_ready_to_send_;
   bool rtcp_ready_to_send_;
   bool was_ever_writable_;
   MediaContentDirection local_content_direction_;
   MediaContentDirection remote_content_direction_;
   bool has_received_packet_;
   bool dtls_keyed_;
   bool secure_required_;
+  rtc::CryptoOptions crypto_options_;

[mattdr, 2016/05/06 22:34:13]

Should we make this available as a public or protected getter, and then avoid
having to plumb crypto_options through the GetSrtpCryptoSuites functions?

[joachim, 2016/05/09 23:21:40]

Done.

   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,
                MediaEngineInterface* media_engine,
                VoiceMediaChannel* channel,
@@ skipping 70 matching lines @@
                                  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);
   bool GetStats_w(VoiceMediaInfo* stats);
 
   virtual void OnMessage(rtc::Message* pmsg);
-  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
+  void GetSrtpCryptoSuites(const rtc::CryptoOptions& crypto_options,
+      std::vector<int>* crypto_suites) const override;
   virtual void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
   virtual void OnMediaMonitorUpdate(
       VoiceMediaChannel* media_channel, const VoiceMediaInfo& info);
   void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info);
 
   static const int kEarlyMediaTimeout = 1000;
   MediaEngineInterface* media_engine_;
   bool received_media_;
   std::unique_ptr<VoiceMediaMonitor> media_monitor_;
@@ skipping 50 matching lines @@
                                  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);
 
   virtual void OnMessage(rtc::Message* pmsg);
-  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
+  void GetSrtpCryptoSuites(const rtc::CryptoOptions& crypto_options,
+      std::vector<int>* crypto_suites) const override;
   virtual void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
   virtual void OnMediaMonitorUpdate(
       VideoMediaChannel* media_channel, const VideoMediaInfo& info);
 
   std::unique_ptr<VideoMediaMonitor> media_monitor_;
 
   // Last VideoSendParameters sent down to the media_channel() via
   // SetSendParameters.
   VideoSendParameters last_send_params_;
@@ skipping 89 matching lines @@
   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);
 
   virtual void OnMessage(rtc::Message* pmsg);
-  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
+  void GetSrtpCryptoSuites(const rtc::CryptoOptions& crypto_options,
+      std::vector<int>* crypto_suites) const override;
   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);
   void OnDataChannelError(uint32_t ssrc, DataMediaChannel::Error error);
   void OnDataChannelReadyToSend(bool writable);
   void OnStreamClosedRemotely(uint32_t sid);
 
   std::unique_ptr<DataMediaMonitor> media_monitor_;
   // TODO(pthatcher): Make a separate SctpDataChannel and
   // RtpDataChannel instead of using this.
   DataChannelType data_channel_type_;
   bool ready_to_send_data_;
 
   // Last DataSendParameters sent down to the media_channel() via
   // 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_