From fee739c22489e991ebad03f887ee1f123d679258 Mon Sep 17 00:00:00 2001 From: "turaj@webrtc.org" Date: Wed, 12 Jun 2013 20:10:06 +0000 Subject: [PATCH] Risk of division by zero. bug=b9338699 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1634004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4223 4adac7df-926f-26a2-2b94-8c16560cd09d --- webrtc/modules/audio_coding/neteq/automode.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/webrtc/modules/audio_coding/neteq/automode.c b/webrtc/modules/audio_coding/neteq/automode.c index ea6fa8d762..646f164a8e 100644 --- a/webrtc/modules/audio_coding/neteq/automode.c +++ b/webrtc/modules/audio_coding/neteq/automode.c @@ -218,12 +218,12 @@ int WebRtcNetEQ_UpdateIatStatistics(AutomodeInst_t *inst, int maxBufLen, { int high_lim_delay; /* Convert the minimum delay from milliseconds to packets in Q8. - * |fsHz| is sampling rate in Hertz, and |inst->packetSpeechLenSamp| + * |fsHz| is sampling rate in Hertz, and |packetLenSamp| * is the number of samples per packet (according to the last * decoding). */ int32_t minimum_delay_q8 = ((inst->minimum_delay_ms * - (fsHz / 1000)) << 8) / inst->packetSpeechLenSamp; + (fsHz / 1000)) << 8) / packetLenSamp; inst->optBufLevel = tempvar; if (streamingMode != 0)