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sprang affa39cb39 Remove time constraint on first retransmit of a packet. 
We don't allow more than one retransmission within one RTT, but the RTT
estimate might be off. Reasonably, the remote end will not send a NACK
until the packet after has been received - so always resend on first
request.

Review URL: https://codereview.webrtc.org/1414563003

Cr-Commit-Position: refs/heads/master@{#10362}
2015-10-21 20:46:42 +00:00
..
audio
Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
2015-10-20 13:41:06 +00:00
base
Introduce rtc::Maybe<T>, which either contains a T or not.
2015-10-21 10:44:17 +00:00
build
Suppress libyuv::TestCpuFlag races.
2015-10-20 11:37:41 +00:00
call
Rename ChannelGroup to CongestionController and move to webrtc/call/.
2015-10-21 13:52:33 +00:00
common_audio
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
common_video
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
examples
Create network change notifier and pass the event to NetworkManager
2015-10-19 16:39:38 +00:00
libjingle
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
modules
Remove time constraint on first retransmit of a packet.
2015-10-21 20:46:42 +00:00
p2p
Revert of Add experiment on weak ping delay during call set up time (patchset #4 id:60001 of https://codereview.webrtc.org/1411883002/ )
2015-10-21 08:07:32 +00:00
sound
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
system_wrappers
Add stats for average QP per frame for VP8 (for received video streams):
2015-10-21 06:55:32 +00:00
test
Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
2015-10-20 13:41:06 +00:00
tools
Fix minor GYP error in webrtc/tools/internal_tools.gyp
2015-10-07 12:51:40 +00:00
video
Remove MockVideoCapturer
2015-10-21 16:56:14 +00:00
video_engine
Rename ChannelGroup to CongestionController and move to webrtc/call/.
2015-10-21 13:52:33 +00:00
voice_engine
Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
2015-10-20 13:41:06 +00:00
.gitignore
…
audio_receive_stream.h
Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
2015-10-20 13:41:06 +00:00
audio_send_stream.h
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
2015-10-16 21:35:11 +00:00
BUILD.gn
Update build files to use webrtc_overrides in Chromium instead of overrides.
2015-09-30 10:49:11 +00:00
call.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
codereview.settings
Update lower-level codereview.settings files.
2015-10-06 21:19:34 +00:00
common_types.cc
…
common_types.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
common.gyp
…
common.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
config.h
Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ )
2015-10-12 13:33:24 +00:00
engine_configurations.h
Don't link with audio codecs that we don't use
2015-10-01 04:54:29 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
libjingle_examples.gyp
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
…
supplement.gypi
…
transport.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
typedefs.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
video_decoder.h
…
video_encoder.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
video_engine_tests.isolate
…
video_frame.h
Add stats for average QP per frame for VP8 (for received video streams):
2015-10-21 06:55:32 +00:00
video_receive_stream.h
Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
2015-10-14 18:29:56 +00:00
video_renderer.h
…
video_send_stream.h
Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
2015-10-07 09:43:25 +00:00
webrtc_examples.gyp
…
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
2015-10-16 21:35:11 +00:00
webrtc.gyp
Remove references to libpeerconnection.
2015-10-08 12:41:05 +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.