Add a default RTT to CallStats and use different values for buffered/real-time mode.

BUG=1613 Review URL: https://webrtc-codereview.appspot.com/1326007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3888 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-23 15:58:23 +00:00
parent d25b602dc0
commit ccd4b2aec8
11 changed files with 60 additions and 43 deletions
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
@ -95,7 +95,8 @@ RTPReceiver::RTPReceiver(const int32_t id,
      max_reordering_threshold_(kDefaultMaxReorderingThreshold),
      rtx_(false),
      ssrc_rtx_(0),
-      payload_type_rtx_(-1) {
+      payload_type_rtx_(-1),
+      rtt_ms_(kInitialReceiveSideRtt) {
  assert(incoming_audio_messages_callback &&
         incoming_messages_callback &&
         incoming_payload_callback);