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Taylor Brandstetter 8fcf414846 Attempting to fix flaky tests that deal with STUN timeout. 
The STUN timeout is 9500ms, and the tests are waiting for 10000ms.

The 500ms margin of error is not enough for some bots (such as UBSan).

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2001093003 .

Cr-Commit-Position: refs/heads/master@{#12854}
2016-05-23 19:49:35 +00:00
..
api
Android: Drop old frame in SurfaceTextureHelper.startListening()
2016-05-23 14:26:59 +00:00
audio
Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate
2016-05-11 13:01:19 +00:00
base
Change initial DTLS retransmission timer from 1 second to 50ms.
2016-05-23 19:49:00 +00:00
build
base.gyp: Add conditions for NaCl builds.
2016-05-20 07:52:47 +00:00
call
Add a default implementation in metrics_default.cc of histograms methods in system_wrappers/interface/metrics.h.
2016-05-20 13:29:50 +00:00
common_audio
Fix an UBSan error (signed overflow) in saturating addition and subtraction
2016-05-23 11:07:10 +00:00
common_video
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1983583002/ )
2016-05-17 11:05:51 +00:00
examples
Pass around the candidate removals events in IOS clients
2016-05-23 18:53:26 +00:00
libjingle
Read recv timestamps from socket (posix only).
2016-05-23 16:19:37 +00:00
media
Don't do a thread jump for incoming frames.
2016-05-23 07:39:45 +00:00
modules
Fix UBSan errors (left shift of negative value)
2016-05-23 10:28:35 +00:00
p2p
Attempting to fix flaky tests that deal with STUN timeout.
2016-05-23 19:49:35 +00:00
pc
Do not create a temporary transport channel when using max-bundle
2016-05-18 00:49:58 +00:00
sdk
Pass around the candidate removals events in IOS clients
2016-05-23 18:53:26 +00:00
system_wrappers
Add a default implementation in metrics_default.cc of histograms methods in system_wrappers/interface/metrics.h.
2016-05-20 13:29:50 +00:00
test
Add a default implementation in metrics_default.cc of histograms methods in system_wrappers/interface/metrics.h.
2016-05-20 13:29:50 +00:00
tools
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
video
Add class which periodically computes statistics.
2016-05-23 13:08:05 +00:00
voice_engine
Enable muted state by default in VoE
2016-05-18 15:52:52 +00:00
.gitignore
…
audio_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_send_stream.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +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
…
BUILD.gn
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +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
…
common_types.h
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
2016-05-03 06:44:11 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +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
…
DEPS
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add root owners to webrtc/OWNERS
2016-05-11 12:47:43 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Disable libyuv jpeg support on Android
2016-05-17 15:39:15 +00:00
transport.h
…
typedefs.h
Remove runtime NEON detection
2016-05-17 17:56:48 +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
…
video_frame.h
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1983583002/ )
2016-05-17 11:05:51 +00:00
video_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
JUnit test framework for AppRTC Android demo.
2016-05-17 10:22:38 +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
…
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Add class which periodically computes statistics.
2016-05-23 13:08:05 +00:00
webrtc.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +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.