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Bugfix android build.

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Review URL: https://webrtc-codereview.appspot.com/374003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1532 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org 2012-01-24 15:25:30 +00:00
parent 96c39d1f73
commit b73c3d1f5d
1 changed files with 3 additions and 3 deletions
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6
src/modules/rtp_rtcp/source/rtp_receiver.cc
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@ -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<WebRtc_Word32>((2 * jitterStd) / frequencyKHz);
// Min maxDelayMs is 1.
if (maxDelayMs == 0) {