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
This commit is contained in:
minyue@webrtc.org
2014-07-16 21:28:26 +00:00
parent 4c3e9917e7
commit 74aaf29a0f
20 changed files with 391 additions and 204 deletions

View File

@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_FRAME_DROPPER_H_
#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_FRAME_DROPPER_H_
#include "webrtc/modules/video_coding/utility/include/exp_filter.h"
#include "webrtc/base/exp_filter.h"
#include "webrtc/typedefs.h"
namespace webrtc
@ -72,23 +72,23 @@ private:
void UpdateRatio();
void CapAccumulator();
VCMExpFilter _keyFrameSizeAvgKbits;
VCMExpFilter _keyFrameRatio;
float _keyFrameSpreadFrames;
int32_t _keyFrameCount;
float _accumulator;
float _accumulatorMax;
float _targetBitRate;
bool _dropNext;
VCMExpFilter _dropRatio;
int32_t _dropCount;
float _windowSize;
float _incoming_frame_rate;
bool _wasBelowMax;
bool _enabled;
bool _fastMode;
float _cap_buffer_size;
float _max_time_drops;
rtc::ExpFilter _keyFrameSizeAvgKbits;
rtc::ExpFilter _keyFrameRatio;
float _keyFrameSpreadFrames;
int32_t _keyFrameCount;
float _accumulator;
float _accumulatorMax;
float _targetBitRate;
bool _dropNext;
rtc::ExpFilter _dropRatio;
int32_t _dropCount;
float _windowSize;
float _incoming_frame_rate;
bool _wasBelowMax;
bool _enabled;
bool _fastMode;
float _cap_buffer_size;
float _max_time_drops;
}; // end of VCMFrameDropper class
} // namespace webrtc