From b73c3d1f5d5496f3a61a46a98905cdfd439e7e2b Mon Sep 17 00:00:00 2001 From: "pwestin@webrtc.org" Date: Tue, 24 Jan 2012 15:25:30 +0000 Subject: [PATCH] Bugfix android build. Review URL: https://webrtc-codereview.appspot.com/374003 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1532 4adac7df-926f-26a2-2b94-8c16560cd09d --- src/modules/rtp_rtcp/source/rtp_receiver.cc | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/src/modules/rtp_rtcp/source/rtp_receiver.cc b/src/modules/rtp_rtcp/source/rtp_receiver.cc index cf34041bf0..0bbcb4949a 100644 --- a/src/modules/rtp_rtcp/source/rtp_receiver.cc +++ b/src/modules/rtp_rtcp/source/rtp_receiver.cc @@ -992,12 +992,12 @@ bool RTPReceiver::RetransmitOfOldPacket( WebRtc_Word32 maxDelayMs = 0; _rtpRtcp.RTT(_SSRC, NULL, NULL, &minRTT, NULL); if (minRTT == 0) { - WebRtc_UWord32 jitter = _jitterQ4 >> 4; // Jitter variance in samples. + float jitter = _jitterQ4 >> 4; // Jitter variance in samples. // Jitter standard deviation in samples. - WebRtc_UWord32 jitterStd = sqrt(jitter); + float jitterStd = sqrt(jitter); // 2 times the std deviation => 95% confidence. // And transform to ms by dividing by the frequency in kHz. - maxDelayMs = (2 * jitterStd) / frequencyKHz; + maxDelayMs = static_cast((2 * jitterStd) / frequencyKHz); // Min maxDelayMs is 1. if (maxDelayMs == 0) {