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hbos 820f578f2b RTCInboundRTPStreamStats's [fir/pli/nack]_count are collected for video. 
Previously this was only collected for RTCOutboundRTPStreamStats video,
with no comment saying it was missing for Inbound. (nack_count should be
collected vor audio as well but this is currently not available - there
is already an existing comment about this in rtcstats_objects.h.)

BUG=chromium:657855, chromium:657854, chromium:627816

Review-Url: https://codereview.webrtc.org/2515293002
Cr-Commit-Position: refs/heads/master@{#15185}
2016-11-22 11:16:56 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
RTCInboundRTPStreamStats's [fir/pli/nack]_count are collected for video.
2016-11-22 11:16:56 +00:00
audio
Expose RtpCodecParameters to VoiceMediaInfo stats.
2016-11-18 07:43:39 +00:00
base
Fix TimeCallback used by BoringSSL.
2016-11-21 22:34:02 +00:00
build
DEPS: Cleanup extra_gyp_flag and extra_gitignore.py
2016-11-22 10:10:09 +00:00
call
Passed AudioMixer to AudioState::Config.
2016-11-17 14:48:56 +00:00
common_audio
Revert of Move smoothing filter to common audio. (patchset #3 id:60001 of https://codereview.webrtc.org/2484153002/ )
2016-11-18 09:31:19 +00:00
common_video
Remove all references to GYP
2016-11-16 19:11:38 +00:00
examples
Set surface view surface size to minimum of the layout size and frame size.
2016-11-22 09:54:47 +00:00
libjingle
Remove all references to GYP
2016-11-16 19:11:38 +00:00
logging
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
media
Wire up FlexFEC in VideoEngine2.
2016-11-22 10:16:56 +00:00
modules
Let Opus increase complexity for low bitrates
2016-11-22 10:08:01 +00:00
p2p
Remove all references to GYP
2016-11-16 19:11:38 +00:00
pc
Remove all references to GYP
2016-11-16 19:11:38 +00:00
sdk
iOS: Add FlexFEC-03 field trial.
2016-11-22 07:43:27 +00:00
stats
RTCStats operator== bugfix
2016-11-21 17:17:51 +00:00
system_wrappers
Use NtpTime in RtcpMeasurement instead of uint sec/uint frac.
2016-11-17 10:27:20 +00:00
test
Changed the way we find the ProjectRootPath.
2016-11-18 15:06:47 +00:00
tools
Remove all references to GYP
2016-11-16 19:11:38 +00:00
video
Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/
2016-11-22 09:43:06 +00:00
voice_engine
Support multiple timestamp rates for sending DTMF.
2016-11-17 13:25:45 +00:00
.gitignore
…
BUILD.gn
Let Opus increase complexity for low bitrates
2016-11-22 10:08:01 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
…
common_types.cc
Reland of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2491613005/ )
2016-11-17 07:23:15 +00:00
common_types.h
Add overhead per packet observer to the rtp_sender.
2016-11-17 09:38:48 +00:00
config.cc
Wire up FlexFEC in VideoEngine2.
2016-11-22 10:16:56 +00:00
config.h
Wire up FlexFEC in VideoEngine2.
2016-11-22 10:16:56 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
Remove all references to GYP
2016-11-16 19:11:38 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
…
typedefs.h
…
video_decoder.h
Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/
2016-11-22 09:43:06 +00:00
video_encoder.h
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ )
2016-11-18 09:34:14 +00:00
video_frame.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
video_receive_stream.h
Remove RED/RTX workaround from sender/receiver and VideoEngine2.
2016-11-11 11:28:38 +00:00
video_send_stream.h
Wire up FlexfecSender in RTP module and VideoSendStream.
2016-11-15 13:25:44 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +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.