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Fix the break caused by r5579.

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TBR=tlegrand@google.com
BUG=

Review URL: https://webrtc-codereview.appspot.com/8939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5581 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
turaj@webrtc.org 2014-02-19 23:07:31 +00:00
parent 571df2dca9
commit 78f0db4710
1 changed files with 4 additions and 4 deletions
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8
webrtc/modules/audio_coding/main/test/Channel.cc
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@ -189,8 +189,8 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize) {
currentPayloadStr->lastPayloadLenByte = payloadSize;
currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
currentPayloadStr->payloadType = rtpInfo.header.payloadType;
memset(currentPayloadStr->frameSizeStats, 0,
sizeof(ACMTestPayloadStats::frameSizeStats));
memset(currentPayloadStr->frameSizeStats, 0, MAX_NUM_FRAMESIZES *
sizeof(ACMTestFrameSizeStats));
}
} else {
n = 0;
@ -202,8 +202,8 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize) {
_payloadStats[n].lastPayloadLenByte = payloadSize;
_payloadStats[n].lastTimestamp = rtpInfo.header.timestamp;
_payloadStats[n].payloadType = rtpInfo.header.payloadType;
memset(_payloadStats[n].frameSizeStats, 0,
sizeof(ACMTestPayloadStats::frameSizeStats));
memset(_payloadStats[n].frameSizeStats, 0, MAX_NUM_FRAMESIZES *
sizeof(ACMTestFrameSizeStats));
}
}