From 78f0db4710c7cf00830646c75d8782ab426d514c Mon Sep 17 00:00:00 2001 From: "turaj@webrtc.org" Date: Wed, 19 Feb 2014 23:07:31 +0000 Subject: [PATCH] Fix the break caused by r5579. 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 --- webrtc/modules/audio_coding/main/test/Channel.cc | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/webrtc/modules/audio_coding/main/test/Channel.cc b/webrtc/modules/audio_coding/main/test/Channel.cc index d32735e3ba..20ecf3a357 100644 --- a/webrtc/modules/audio_coding/main/test/Channel.cc +++ b/webrtc/modules/audio_coding/main/test/Channel.cc @@ -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)); } }