Set AES key and IV mask to CastSender

chrome/renderer/media/cast_rtp_stream.h

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CHROME_RENDERER_MEDIA_CAST_RTP_STREAM_H_
 #define CHROME_RENDERER_MEDIA_CAST_RTP_STREAM_H_
 
 #include <string>
 #include <vector>
 
@@ skipping 38 matching lines @@
   // Number of audio channels.
   int channels;
 
   // Width and height of the video content.
   int width;
   int height;
 
   // Name of the codec used.
   std::string codec_name;
 
+  // AES encryption key.
+  std::string aes_key;
+
+  // AES encryption IV mask.
+  std::string aes_iv_mask;
+
   // List of codec specific parameters.
   std::vector<CastCodecSpecificParams> codec_specific_params;
 
   CastRtpPayloadParams();
   ~CastRtpPayloadParams();
 };
 
 // Defines the capabilities of the transport.
 struct CastRtpCaps {
   // Defines a list of supported payloads.
   std::vector<CastRtpPayloadParams> payloads;
 
   // Names of supported RTCP features.
   std::vector<std::string> rtcp_features;
 
-  // Names of supported FEC (Forward Error Correction) mechanisms.
-  std::vector<std::string> fec_mechanisms;
-
   CastRtpCaps();
   ~CastRtpCaps();
 };
 
 typedef CastRtpCaps CastRtpParams;
 
 // This object represents a RTP stream that encodes and optionally
 // encrypt audio or video data from a WebMediaStreamTrack.
 // Note that this object does not actually output packets. It allows
 // configuration of encoding and RTP parameters and control such a logical
@@ skipping 25 matching lines @@
   blink::WebMediaStreamTrack track_;
   const scoped_refptr<CastSession> cast_session_;
   scoped_ptr<CastAudioSink> audio_sink_;
   scoped_ptr<CastVideoSink> video_sink_;
   CastRtpParams params_;
 
   DISALLOW_COPY_AND_ASSIGN(CastRtpStream);
 };
 
 #endif  // CHROME_RENDERER_MEDIA_CAST_RTP_STREAM_H_