X-Git-Url: http://review.tizen.org/git/?a=blobdiff_plain;f=src%2Fthird_party%2Fwebrtc%2Fvideo_engine%2Finclude%2Fvie_rtp_rtcp.h;h=5ef189234e85d0b875e04f0a902198c5af06e619;hb=ff3e2503a20db9193d323c1d19c38c68004dec4a;hp=9d899ad7f392df691fb24dbfb60a0f8d5d90545b;hpb=7338fba38ba696536d1cc9d389afd716a6ab2fe6;p=platform%2Fframework%2Fweb%2Fcrosswalk.git

diff --git a/src/third_party/webrtc/video_engine/include/vie_rtp_rtcp.h b/src/third_party/webrtc/video_engine/include/vie_rtp_rtcp.h
index 9d899ad..5ef1892 100644
--- a/src/third_party/webrtc/video_engine/include/vie_rtp_rtcp.h
+++ b/src/third_party/webrtc/video_engine/include/vie_rtp_rtcp.h
@@ -27,6 +27,7 @@
 namespace webrtc {
 
 class VideoEngine;
+struct ReceiveBandwidthEstimatorStats;
 
 // This enumerator sets the RTCP mode.
 enum ViERTCPMode {
@@ -389,6 +390,13 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
       const int video_channel,
       unsigned int* estimated_bandwidth) const = 0;
 
+  // This function gets the receive-side bandwidth esitmator statistics.
+  // TODO(jiayl): remove the default impl when libjingle's FakeWebRtcVideoEngine
+  // is updated.
+  virtual int GetReceiveBandwidthEstimatorStats(
+      const int video_channel,
+      ReceiveBandwidthEstimatorStats* output) const { return -1; }
+
   // This function enables capturing of RTP packets to a binary file on a
   // specific channel and for a given direction. The file can later be
   // replayed using e.g. RTP Tools rtpplay since the binary file format is