From b88fc18abafa32dcddbec6822cfa96ea25b95cb7 Mon Sep 17 00:00:00 2001 From: "stefan@webrtc.org" Date: Thu, 5 Dec 2013 11:36:46 +0000 Subject: [PATCH] Fix bug where fraction_lost is always set to 0 when getting received RTCP statistics. R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5039004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5221 4adac7df-926f-26a2-2b94-8c16560cd09d --- webrtc/video_engine/vie_channel.cc | 2 -- 1 file changed, 2 deletions(-) diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc index 0c595fc82..eaa60447a 100644 --- a/webrtc/video_engine/vie_channel.cc +++ b/webrtc/video_engine/vie_channel.cc @@ -1293,7 +1293,6 @@ int32_t ViEChannel::GetReceivedRtcpStatistics(uint16_t* fraction_lost, "%s", __FUNCTION__); uint32_t remote_ssrc = vie_receiver_.GetRemoteSsrc(); - uint8_t frac_lost = 0; StreamStatistician* statistician = vie_receiver_.GetReceiveStatistics()->GetStatistician(remote_ssrc); StreamStatistician::Statistics receive_stats; @@ -1307,7 +1306,6 @@ int32_t ViEChannel::GetReceivedRtcpStatistics(uint16_t* fraction_lost, *cumulative_lost = receive_stats.cumulative_lost; *extended_max = receive_stats.extended_max_sequence_number; *jitter_samples = receive_stats.jitter; - *fraction_lost = frac_lost; uint16_t dummy = 0; uint16_t rtt = 0;