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minyue 435ddf978d Add TransportFeedbackPacketLossTracker. 
This CL is to calculate packet loss metrics from TransportFeedback. The outcome of this will be passed down to audio encoder.

BUG=webrtc:6904

Review-Url: https://codereview.webrtc.org/2579613003
Cr-Commit-Position: refs/heads/master@{#16217}
2017-01-23 16:07:05 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Fixed public_deps for libjingle_peerconnection{,_api}
2017-01-23 15:37:43 +00:00
audio
Pass SdpAudioFormat through Channel, without converting to CodecInst
2017-01-19 15:03:59 +00:00
base
Delete or update left-over ASSERT use and comments.
2017-01-23 15:43:05 +00:00
build
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
call
Pass SdpAudioFormat through Channel, without converting to CodecInst
2017-01-19 15:03:59 +00:00
common_audio
Replace RTC_DCHECK(false) with RTC_NOTREACHED().
2017-01-12 10:24:27 +00:00
common_video
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
examples
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
logging
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false.
2017-01-17 10:37:34 +00:00
media
Delete or update left-over ASSERT use and comments.
2017-01-23 15:43:05 +00:00
modules
Delete or update left-over ASSERT use and comments.
2017-01-23 15:43:05 +00:00
p2p
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
pc
Fixed public_deps for libjingle_peerconnection{,_api}
2017-01-23 15:37:43 +00:00
sdk
Minor style change suggested by internal static analysis tool.
2017-01-23 14:09:35 +00:00
stats
RTCMediaStreamTrackStats.kind added and collected.
2017-01-17 10:53:23 +00:00
system_wrappers
Reland of Delete unused code from systeminfo. (patchset #1 id:1 of https://codereview.webrtc.org/2584563004/ )
2017-01-09 09:22:16 +00:00
test
Add TransportFeedbackPacketLossTracker.
2017-01-23 16:07:05 +00:00
tools
Script to start stubbed loopback video test with Espresso
2017-01-23 15:55:42 +00:00
video
RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector.
2017-01-23 15:21:55 +00:00
voice_engine
Add TransportFeedbackPacketLossTracker.
2017-01-23 16:07:05 +00:00
.gitignore
…
BUILD.gn
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
config.cc
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
config.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector.
2017-01-23 15:21:55 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.