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sakal ff0e72fd16 Add QP sum stats for received streams. 
This is not implemented yet in any of the decoders.

BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2649133005
Cr-Commit-Position: refs/heads/master@{#16475}
2017-02-07 15:15:17 +00:00
..
api
Reland of Drop frames until specified bitrate is achieved. (patchset #1 id:1 of https://codereview.webrtc.org/2666303002/ )
2017-02-07 15:02:22 +00:00
audio
Wire up audio packet loss to BWE.
2017-02-07 15:14:08 +00:00
base
Replace occurences of string by std::string.
2017-02-07 11:40:28 +00:00
call
Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.
2017-02-07 09:18:43 +00:00
common_audio
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
2017-01-24 14:58:22 +00:00
common_video
Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common.
2017-02-06 10:21:11 +00:00
examples
Add chromium-junit4 tag to instrumentation test AndroidManifests.
2017-02-07 12:12:30 +00:00
logging
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ )
2017-01-27 12:53:07 +00:00
media
Add QP sum stats for received streams.
2017-02-07 15:15:17 +00:00
modules
Add QP sum stats for received streams.
2017-02-07 15:15:17 +00:00
p2p
Fix Chromium FYI bot
2017-02-03 02:22:46 +00:00
pc
Add QP sum stats for received streams.
2017-02-07 15:15:17 +00:00
sdk
Add chromium-junit4 tag to instrumentation test AndroidManifests.
2017-02-07 12:12:30 +00:00
stats
RTCRTPStreamStats.mediaTrackId renamed to trackId.
2017-01-27 14:35:16 +00:00
system_wrappers
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
2017-01-24 14:58:22 +00:00
test
Wire up audio packet loss to BWE.
2017-02-07 15:14:08 +00:00
tools
Remove all occurrences of "using std::string".
2017-02-02 16:10:00 +00:00
video
Add QP sum stats for received streams.
2017-02-07 15:15:17 +00:00
voice_engine
Wire up audio packet loss to BWE.
2017-02-07 15:14:08 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common.
2017-02-06 10:21:11 +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
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +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
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +00:00
video_decoder.h
Add QP sum stats for received streams.
2017-02-07 15:15:17 +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
Add QP sum stats for received streams.
2017-02-07 15:15:17 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Adding build switch for Opus that supports 120ms ptime.
2017-02-02 01:31:11 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.