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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
This commit is contained in:
turaj@webrtc.org 2013-06-12 20:10:06 +00:00
parent dd97ef4e28
commit fee739c224
1 changed files with 2 additions and 2 deletions
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4
webrtc/modules/audio_coding/neteq/automode.c
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@ -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)