Upstream version 5.34.104.0

[platform/framework/web/crosswalk.git] / src / third_party / webrtc / video_engine / include / vie_rtp_rtcp.h

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 (file)
--- 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