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ehmaldonado ebb0b8ec9a Increase the threshold for RunPlayoutAndRecordingInFullDuplex. 
RunPlayoutAndRecordingInFullDuplex fails sometimes on Android swarming
bots, presumably because the timing is hardware dependent.

This test ensures that audio starts pumping. The exact performance is
not that important.

R=kjellander@webrtc.org, henrika@webrtc.org
BUG=webrtc:6464
NOTRY=True

Review-Url: https://codereview.webrtc.org/2391563002
Cr-Commit-Position: refs/heads/master@{#14492}
2016-10-04 08:59:05 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
RTCCertificateStats[1] added.
2016-10-03 21:17:03 +00:00
audio
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
base
rtc::FunctionView improvements: accept function pointers and nullptr
2016-10-03 19:40:41 +00:00
build
Roll chromium_revision f86fb54ec3..dd442d4812 (420104:421425)
2016-09-28 07:45:12 +00:00
call
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
common_audio
Revert of Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere (patchset #2 id:20001 of https://codereview.webrtc.org/2384693002/ )
2016-10-03 15:32:36 +00:00
common_video
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
examples
Disable TCPChannelClientTest.testConnectIPv6
2016-09-30 07:20:12 +00:00
libjingle
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
logging
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
media
Revert of Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere (patchset #2 id:20001 of https://codereview.webrtc.org/2384693002/ )
2016-10-03 15:32:36 +00:00
modules
Increase the threshold for RunPlayoutAndRecordingInFullDuplex.
2016-10-04 08:59:05 +00:00
p2p
Emit SignalReadyToSend even for "presumed writable" connections.
2016-09-30 22:16:57 +00:00
pc
Update WebRTC to build against libsrtp 2.0
2016-09-28 21:08:53 +00:00
sdk
Remove Crit::Scope lock by using atomic bool property.
2016-09-30 07:21:28 +00:00
stats
RTCCertificateStats[1] added.
2016-10-03 21:17:03 +00:00
system_wrappers
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
test
Simplify public interface of ProducerFec.
2016-10-04 08:57:47 +00:00
tools
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
video
Fixed flaky VideoSendStreamTests after ViEEncoder changes
2016-10-04 08:47:05 +00:00
voice_engine
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
.gitignore
…
BUILD.gn
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +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
…
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
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
config.cc
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
config.h
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
DEPS
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +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
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
…
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
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
video_receive_stream.h
Add logging statements to places where the frame might be dropped in WebRTC pipeline.
2016-09-30 13:19:12 +00:00
video_send_stream.h
Add VideoSendStream::Stats::prefered_media_bitrate_bps
2016-09-29 09:49:01 +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.