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danilchap 4a0c76402e Add rtcp::TransportFeedback::GetReceivedPackets() 
It combines and simplify use of GetStatusVector and GetReceiveDeltas accesors.

Replace use of all GetStatusVector/GetReceiveDeltasUs

BUG=None

Review-Url: https://codereview.webrtc.org/2633923003
Cr-Commit-Position: refs/heads/master@{#16139}
2017-01-18 10:40:30 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
More minor improvements to BaseChannel/transport code.
2017-01-18 02:32:35 +00:00
audio
Fix for bwe with overhead on audio only calls.
2017-01-17 07:55:07 +00:00
base
Remove dependency to Chromium code from WebRTC Java code.
2017-01-17 14:46:34 +00:00
build
Replace default locale with US locale on Android.
2017-01-16 12:57:32 +00:00
call
Move congestion controller processing to the pacer thread.
2017-01-18 08:10:31 +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
Android: AppRTCMobile: Don't leak CallActivity.
2017-01-17 15:25:53 +00:00
logging
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false.
2017-01-17 10:37:34 +00:00
media
Add experimental simulcast screen content mode
2017-01-18 08:36:31 +00:00
modules
Add rtcp::TransportFeedback::GetReceivedPackets()
2017-01-18 10:40:30 +00:00
p2p
Remove the IceTransportInternal2.
2017-01-18 03:40:19 +00:00
pc
More minor improvements to BaseChannel/transport code.
2017-01-18 02:32:35 +00:00
sdk
Increase bitrate adjustment values for VP8 Exynos encoder
2017-01-17 23:32:02 +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
Let FlexfecReceiveStreamImpl send RTCP RRs.
2017-01-17 09:33:54 +00:00
tools
Comparison of videos with reference frame not starting from zero
2017-01-17 11:24:57 +00:00
video
Refactor OveruseFrameDetector to use timing in us units
2017-01-18 10:16:20 +00:00
voice_engine
UdpTransport:IsIpAddressValid: Added extra :: check for ipv6
2017-01-17 10:31:37 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Refactor OveruseFrameDetector to use timing in us units
2017-01-18 10:16:20 +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
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +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.