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mikescarlett 70035cae4d Fix QuicSession to unbuffer data when the QuicTransportChannel reconnects 
The QuicWriteBlockedList needs to register outgoing QUIC
streams so that when the QuicTransportChannel becomes
unwritable and QUIC streams have buffered data, they can
send data once the QuicTransportChannel becomes writable.

Otherwise the QUIC streams will remain write blocked
after the QuicTransportChannel is writable.

BUG=

Review-Url: https://codereview.webrtc.org/1888903002
Cr-Commit-Position: refs/heads/master@{#12573}
2016-04-30 01:14:47 +00:00
..
api
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
audio
Switch voice transport to use Call and Stream instead of VoENetwork.
2016-04-29 07:57:21 +00:00
base
Re-add a (dummy) webrtc/base/buffer.cc to hopefully unbreak the Chromium build
2016-04-29 18:18:55 +00:00
build
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
call
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
common_audio
Revert of Avoiding overflow in cross correlation in NetEq. (patchset #6 id:180001 of https://codereview.webrtc.org/1908623002/ )
2016-04-28 09:16:54 +00:00
common_video
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
examples
Added the API to create an RTCRtpSender to the Objective C wrapper.
2016-04-29 22:30:24 +00:00
libjingle
Remove the rtc_relative_path GYP variable and similar defines
2016-04-27 17:13:28 +00:00
media
Remove usage of VoENetwork from VoEWrapper and FakeWebRtcVoiceEngine.
2016-04-29 16:05:35 +00:00
modules
Revert of Remove VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1917083003/ )
2016-04-29 23:10:36 +00:00
p2p
Fix QuicSession to unbuffer data when the QuicTransportChannel reconnects
2016-04-30 01:14:47 +00:00
pc
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
sdk
Added the API to create an RTCRtpSender to the Objective C wrapper.
2016-04-29 22:30:24 +00:00
system_wrappers
Replace scoped_ptr with unique_ptr in webrtc/base/
2016-04-26 10:13:28 +00:00
test
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
tools
Adding default_exe_manifest to some gn binaries
2016-04-28 07:21:14 +00:00
video
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
voice_engine
Switch voice transport to use Call and Stream instead of VoENetwork.
2016-04-29 07:57:21 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Switch voice transport to use Call and Stream instead of VoENetwork.
2016-04-29 07:57:21 +00:00
audio_send_stream.h
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
2016-03-11 11:06:48 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
call.h
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Tune BWE to be a bit less sensitive to spurious delay events.
2016-03-10 13:13:26 +00:00
common.gyp
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Change transport sequence number extension strings to specify what revision is implemented.
2016-03-01 15:27:27 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
DEPS
Build dynamic iOS SDK.
2016-04-27 08:54:27 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
Make pbos owner of additional video files.
2016-02-26 12:29:17 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Remove use_openssl from webrtc
2016-04-18 18:13:17 +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
Update attributes to match gclibc's ansidecl.h
2016-01-08 12:51:47 +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
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
2016-04-29 09:39:33 +00:00
video_receive_stream.h
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
video_send_stream.h
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
webrtc_examples.gyp
Build dynamic iOS SDK.
2016-04-27 08:54:27 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Reland of Delete video_render module. (patchset #1 id:1 of https://codereview.webrtc.org/1923613003/ )
2016-04-29 07:58:48 +00:00
webrtc.gyp
Reland of Delete video_render module. (patchset #1 id:1 of https://codereview.webrtc.org/1923613003/ )
2016-04-29 07:58:48 +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.