Support encrypted RTP extensions (RFC 6904)

webrtc/pc/mediasession.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 446 matching lines @@
   const DataCodecs& data_codecs() const { return data_codecs_; }
   void set_data_codecs(const DataCodecs& codecs) { data_codecs_ = codecs; }
   SecurePolicy secure() const { return secure_; }
   void set_secure(SecurePolicy s) { secure_ = s; }
   // Decides if a StreamParams shall be added to the audio and video media
   // content in SessionDescription when CreateOffer and CreateAnswer is called
   // even if |options| don't include a Stream. This is needed to support legacy
   // applications. |add_legacy_| is true per default.
   void set_add_legacy_streams(bool add_legacy) { add_legacy_ = add_legacy; }
 
+  void set_enable_encrypted_rtp_header_extensions(bool enable) {
+    enable_encrypted_rtp_header_extensions_ = enable;
+  }
+
   SessionDescription* CreateOffer(
       const MediaSessionOptions& options,
       const SessionDescription* current_description) const;
   SessionDescription* CreateAnswer(
       const SessionDescription* offer,
       const MediaSessionOptions& options,
       const SessionDescription* current_description) const;
 
  private:
   const AudioCodecs& GetAudioCodecsForOffer(
@@ skipping 79 matching lines @@
 
   AudioCodecs audio_send_codecs_;
   AudioCodecs audio_recv_codecs_;
   AudioCodecs audio_sendrecv_codecs_;
   RtpHeaderExtensions audio_rtp_extensions_;
   VideoCodecs video_codecs_;
   RtpHeaderExtensions video_rtp_extensions_;
   DataCodecs data_codecs_;
   SecurePolicy secure_;
   bool add_legacy_;
+  bool enable_encrypted_rtp_header_extensions_ = false;
   std::string lang_;
   const TransportDescriptionFactory* transport_desc_factory_;
 };
 
 // Convenience functions.
 bool IsMediaContent(const ContentInfo* content);
 bool IsAudioContent(const ContentInfo* content);
 bool IsVideoContent(const ContentInfo* content);
 bool IsDataContent(const ContentInfo* content);
 const ContentInfo* GetFirstMediaContent(const ContentInfos& contents,
@@ skipping 39 matching lines @@
 void GetSupportedVideoCryptoSuiteNames(const rtc::CryptoOptions& crypto_options,
     std::vector<std::string>* crypto_suite_names);
 void GetSupportedDataCryptoSuiteNames(const rtc::CryptoOptions& crypto_options,
     std::vector<std::string>* crypto_suite_names);
 void GetDefaultSrtpCryptoSuiteNames(const rtc::CryptoOptions& crypto_options,
     std::vector<std::string>* crypto_suite_names);
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_MEDIASESSION_H_