webrtc_m130/webrtc/modules/video_coding/utility/video_coding_utility.gyp
minyue@webrtc.org 74aaf29a0f Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.
The filter is an exponential filter borrowed from video coding module.

The method is written in a new class called PacketLossProtector (not sure if the name is nice), which can be used in the future for more sophisticated logic.

BUG=
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6709 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 21:28:26 +00:00

27 lines
782 B
Python

# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../../../build/common.gypi',
],
'targets': [
{
'target_name': 'video_coding_utility',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
],
'sources': [
'include/frame_dropper.h',
'frame_dropper.cc',
],
},
], # targets
}