admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
Stefan Holmer 52200d0b7f Stop increasing loss-based BWE if no feedback is received. 
This includes if RTCP is received, but the number of packets received by the
other end hasn't increased.

Further, if no RTCP is received for more than 3 feedback intervals (3 seconds)
we start reducing the estimate by 20%. This is put under an experiment.

BUG=webrtc:6238
R=terelius@webrtc.org

Review URL: https://codereview.webrtc.org/2262213002 .

Cr-Commit-Position: refs/heads/master@{#14306}
2016-09-20 12:14:52 +00:00
..
androidjunit
Make android_junit_tests use LocalRobolectricTestRunner
2016-09-18 19:28:15 +00:00
api
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
audio
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
base
Revert of Allow the DTLS fingerprint verification to occur after the handshake. (patchset #11 id:200001 of https://codereview.webrtc.org/2163683003/ )
2016-09-20 00:21:00 +00:00
build
MB: Add Android swarming bots.
2016-09-19 21:01:23 +00:00
call
Stop increasing loss-based BWE if no feedback is received.
2016-09-20 12:14:52 +00:00
common_audio
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
common_video
Reland of Update test code to use I420Buffer when writing pixel data. (patchset #1 id:1 of https://codereview.webrtc.org/2343083002/ )
2016-09-19 07:34:52 +00:00
examples
Merge AppRTCDemoJUnitTest into android_junit_tests target
2016-09-19 15:53:30 +00:00
libjingle
GN: Change group deps to public_deps.
2016-09-15 07:53:34 +00:00
media
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
modules
Stop increasing loss-based BWE if no feedback is received.
2016-09-20 12:14:52 +00:00
p2p
Remove dead code branches from P2PtransportChannel unittest.
2016-09-20 07:23:33 +00:00
pc
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
sdk
GN: Enable rtc_common_config for more targets.
2016-09-18 15:12:36 +00:00
stats
RTCTestStats moved into webrtc/stats/test (rtc_stats_test_utils)
2016-09-19 13:06:05 +00:00
system_wrappers
Revert of Roll chromium_revision cf9457edb7..cede888c27 (416297:419407) (patchset #3 id:40001 of https://codereview.webrtc.org/2348133003/ )
2016-09-19 08:55:21 +00:00
test
Reland of Update test code to use I420Buffer when writing pixel data. (patchset #1 id:1 of https://codereview.webrtc.org/2343083002/ )
2016-09-19 07:34:52 +00:00
tools
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
video
Do not update stream synchronization if no new video packet has been received since last update (e.g. video muted).
2016-09-17 08:00:04 +00:00
voice_engine
Add voe_cmd_test to voice_engine/BUILD.gn (and remove it from voice_engine.gyp, together with the channel_transport gyp target)
2016-09-15 20:12:10 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Reenabled the RtcEventLog unittests
2016-09-19 21:42:10 +00:00
call.h
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
2016-09-01 08:17:43 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
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
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
common.gyp
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
DEPS
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
engine_configurations.h
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +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
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Add codec name to CodecSpecificInfo and get the codec name stats from there instead.
2016-09-01 07:21:18 +00:00
video_frame.h
Introduce webrtc::VideoFrame::timestamp_us, and corresponding constructor.
2016-09-05 07:51:25 +00:00
video_receive_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
video_send_stream.h
Replace VideoCapturerInput with VideoSinkInterface.
2016-09-16 14:53:48 +00:00
webrtc_examples.gyp
Merge AppRTCDemoJUnitTest into android_junit_tests target
2016-09-19 15:53:30 +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
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +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.