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Irfan Sheriff 1eb12934e7 Handle BW drop in ALR region and initiate probing 
Original change by isheriff@chromium.org: http://crrev.com/2387463002#ps40001

BUG=webrtc:6332
TBR=philipel@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2432633002 .

Patch from Irfan Sheriff <isheriff@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#14673}
2016-10-19 00:04:35 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
RTCDataChannelStats[1] added, supporting all stats members.
2016-10-18 19:48:37 +00:00
audio
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
base
Add rtc::Optional::emplace
2016-10-18 11:07:25 +00:00
build
Removed suppressions for the data race inside the APM that is now fixed.
2016-10-14 21:44:54 +00:00
call
Revert of Add path for recovered packets from internal::Call to RtpStreamReceiver. (patchset #2 id:60001 of https://codereview.webrtc.org/2390823009/ )
2016-10-17 16:42:38 +00:00
common_audio
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
common_video
Add unittest for I420Buffer::Rotate.
2016-10-18 13:01:38 +00:00
examples
Second try to get "Support for video file instead of camera and output video out to file" accepted
2016-10-18 15:47:59 +00:00
libjingle
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
logging
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
media
Set min and max rate on caller and on callee side.
2016-10-18 16:39:28 +00:00
modules
Handle BW drop in ALR region and initiate probing
2016-10-19 00:04:35 +00:00
p2p
Prepare for introduction of rtc::PacketTransportInterface.
2016-10-18 22:38:43 +00:00
pc
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
sdk
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
stats
RTCDataChannelStats[1] added, supporting all stats members.
2016-10-18 19:48:37 +00:00
system_wrappers
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
test
Add a fuzzer for FlexfecReceiver.
2016-10-18 08:18:12 +00:00
tools
During a fix of an unrelated issue, a bug was introduced in the rtp analyzer tool: when the number of data points was divisible by RTPStatitstics.PLOT_RESOLUTION_MS (which is 50), pyplot.plot was called with arrays of different lengths. One of the arrays could be one element larger.
2016-10-18 08:44:50 +00:00
video
Fix for flaky test: EndToEndTest.VerifyHistogramStatsWithRtx
2016-10-18 18:50:57 +00:00
voice_engine
Call OnTransportFeedback just when feedback_observer exist.
2016-10-18 16:40:38 +00:00
.gitignore
…
BUILD.gn
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
call.h
Refactoring: move ownership of RtcEventLog from Call to PeerConnection
2016-10-07 18:53:15 +00:00
codereview.settings
…
common_types.cc
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
common_types.h
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
config.h
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +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
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
…
transport.h
…
typedefs.h
…
video_decoder.h
…
video_encoder.h
Make MediaCodecEncoder fallback to a software encoder on failure.
2016-10-06 12:55:19 +00:00
video_frame.h
Revert of Delete webrtc::VideoFrame::CopyFrame. (patchset #2 id:20001 of https://codereview.webrtc.org/2371363003/ )
2016-10-06 13:29:34 +00:00
video_receive_stream.h
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
video_send_stream.h
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
webrtc_examples.gyp
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +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.