commit 47f0c4192df67999055b47f6a5d5314ce6a4ac4b
author stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Wed Dec 04 10:24:26 2013
committer stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Wed Dec 04 10:24:26 2013
tree 58cb58f5de2c2da75806d30c018d7a798480f865
parent 76c6ac4af070b485a0cbbeccbff34990d89e51b4

Adds support for sending redundant payloads over RTX.

TEST=trybots
BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5209 4adac7df-926f-26a2-2b94-8c16560cd09d

call.h

diff --git a/call.h b/call.h
index 186672d..7ee2592 100644
--- a/call.h
+++ b/call.h
@@ -38,12 +38,15 @@
  public:
   struct Config {
     explicit Config(newapi::Transport* send_transport)
-        : send_transport(send_transport),
+        : webrtc_config(NULL),
+          send_transport(send_transport),
           overuse_detection(false),
           voice_engine(NULL),
           trace_callback(NULL),
           trace_filter(kTraceDefault) {}
 
+    webrtc::Config* webrtc_config;
+
     newapi::Transport* send_transport;
     bool overuse_detection;
 
@@ -56,6 +59,9 @@
 
   static Call* Create(const Call::Config& config);
 
+  static Call* Create(const Call::Config& config,
+                      const webrtc::Config& webrtc_config);
+
   virtual std::vector<VideoCodec> GetVideoCodecs() = 0;
 
   virtual VideoSendStream::Config GetDefaultSendConfig() = 0;