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Erik Språng 9526187dde Default enable abs send time bwe for CallTest 
Using the single stream bwe is really bad for the screenshare
test case in particular, but would probably help in other
cases as well so enabling it by default in CallTest setup.

BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43089004

Cr-Commit-Position: refs/heads/master@{#8971}
2015-04-10 09:58:51 +00:00
..
base
Cleanup: Remove unnecessary SHA1Transform() declaration.
2015-04-09 13:48:00 +00:00
build
Remove android_webview_build conditions.
2015-04-09 15:36:13 +00:00
common_audio
Remove android_webview_build conditions.
2015-04-09 15:36:13 +00:00
common_video
Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.""
2015-04-09 11:44:19 +00:00
examples/android
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
libjingle
Cleanup: Remove DISALLOW_EVIL_CONSTRUCTORS macro.
2015-04-04 23:56:56 +00:00
modules
Delays changing to COMMUNICATION mode until streaming starts.
2015-04-10 09:46:54 +00:00
overrides
Cleanup: Remove DISALLOW_EVIL_CONSTRUCTORS macro.
2015-04-04 23:56:56 +00:00
p2p
Remove Transport::Reset, which is never used, and only makes reading the code harder.
2015-04-10 00:45:10 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Remove android_webview_build conditions.
2015-04-09 15:36:13 +00:00
test
Default enable abs send time bwe for CallTest
2015-04-10 09:58:51 +00:00
tools
Exit after printing usage message.
2015-04-09 13:45:17 +00:00
video
Default enable abs send time bwe for CallTest
2015-04-10 09:58:51 +00:00
video_engine
Remove ViE interface usage from VideoReceiveStream.
2015-04-09 12:35:46 +00:00
voice_engine
talk/media/webrtc/webrtcvoiceengine: Delay Agnostic AEC should not override HW-AEC
2015-04-07 13:25:52 +00:00
.gitignore
…
BUILD.gn
Let Chromium declare the mips_dsp_rev build variable.
2015-03-04 09:51:17 +00:00
call.h
Remove ViE interface usage from VideoReceiveStream.
2015-04-09 12:35:46 +00:00
codereview.settings
…
common_types.cc
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common_types.h
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
…
config.cc
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
config.h
Add CVO support to Vie layer.
2015-03-12 20:51:50 +00:00
engine_configurations.h
Delete all codec-specific subclasses of ACMGenericCodec
2015-02-23 09:26:51 +00:00
experiments.h
…
frame_callback.h
…
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
supplement.gypi
…
transport.h
…
typedefs.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video_decoder.h
Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
2015-03-16 13:48:18 +00:00
video_encoder.h
…
video_engine_tests.isolate
…
video_frame.h
Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.""
2015-04-09 11:44:19 +00:00
video_receive_stream.h
Add decoder-timing stats to VideoReceiveStream.
2015-02-25 10:42:45 +00:00
video_renderer.h
…
video_send_stream.h
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
2015-03-18 09:51:44 +00:00
webrtc_examples.gyp
Remove video from WebRTC Android example.
2015-03-25 13:32:30 +00:00
webrtc_perf_tests.isolate
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
webrtc_tests.gypi
rtc_unittests on Android
2015-03-04 08:48:17 +00:00
webrtc.gyp
…

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.