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Changing the buffer size (slots) to 1.5 seconds @ 30 ms packets

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This is a relanding of r5725, now with a fix for the failing tests.

BUG=2935
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10339005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5738 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
2014-03-20 15:09:38 +00:00
parent 979f1f8235
commit 3ab57c514c
2 changed files with 14 additions and 10 deletions
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2
webrtc/modules/audio_coding/neteq4/interface/neteq.h
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@ -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|.