From e5be877476d76e5432484006ac5a770aec73621c Mon Sep 17 00:00:00 2001 From: "henrik.lundin@webrtc.org" Date: Wed, 19 Mar 2014 13:36:58 +0000 Subject: [PATCH] Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets BUG=2935 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5725 4adac7df-926f-26a2-2b94-8c16560cd09d --- webrtc/modules/audio_coding/neteq4/interface/neteq.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/webrtc/modules/audio_coding/neteq4/interface/neteq.h b/webrtc/modules/audio_coding/neteq4/interface/neteq.h index 617393093..466882a5f 100644 --- a/webrtc/modules/audio_coding/neteq4/interface/neteq.h +++ b/webrtc/modules/audio_coding/neteq4/interface/neteq.h @@ -102,7 +102,7 @@ class NetEq { kSyncPacketNotAccepted }; - static const int kMaxNumPacketsInBuffer = 240; // TODO(hlundin): Remove. + static const int kMaxNumPacketsInBuffer = 50; // TODO(hlundin): Remove. static const int kMaxBytesInBuffer = 113280; // TODO(hlundin): Remove. // Creates a new NetEq object, starting at the sample rate |sample_rate_hz|.