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deadbeef c23efae6b2 Allow locally rendered video to be downscaled in end-to-end tests. 
A previous CL (https://codereview.webrtc.org/2547673002/) only did this
for end-to-end rendered video. But it appears locally rendered video is
downscaled too.

BUG=webrtc:6811

Review-Url: https://codereview.webrtc.org/2544003004
Cr-Commit-Position: refs/heads/master@{#15397}
2016-12-02 23:45:36 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Allow locally rendered video to be downscaled in end-to-end tests.
2016-12-02 23:45:36 +00:00
audio
Reland "Update rtt on audio only calls".
2016-11-30 15:51:19 +00:00
base
Templatize percentile_filter.h and move it to base/analytics.
2016-11-30 14:52:06 +00:00
build
Move webrtc/api/androidtests to webrtc/sdk/android/instrumentationtests
2016-12-01 09:45:35 +00:00
call
Reland "Update rtt on audio only calls".
2016-11-30 15:51:19 +00:00
common_audio
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac
2016-11-30 14:12:10 +00:00
common_video
Sanity check parsed QP values from H264 bitstream
2016-12-02 09:29:53 +00:00
examples
Reland of Allow custom metrics implementations on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2516403002/ )
2016-11-23 15:12:28 +00:00
libjingle
Remove all references to GYP
2016-11-16 19:11:38 +00:00
logging
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac
2016-11-30 14:12:10 +00:00
media
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
modules
Revert of Use different restrictions of acked bitrate lag depending on operating point. (patchset #3 id:40001 of https://codereview.webrtc.org/2542083003/ )
2016-12-02 19:29:41 +00:00
p2p
Remove all references to GYP
2016-11-16 19:11:38 +00:00
pc
Relaxing timeouts for TestMediaMonitor.
2016-12-01 20:49:25 +00:00
sdk
Move webrtc/api/androidtests to webrtc/sdk/android/instrumentationtests
2016-12-01 09:45:35 +00:00
stats
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
2016-11-30 09:50:36 +00:00
system_wrappers
Add linearly spaced counting histograms
2016-12-01 17:58:53 +00:00
test
Wire up rtcp xr target bitrate on receive side.
2016-12-02 15:29:48 +00:00
tools
Added tool for reference less video analysis (go/refless-video-analysis)
2016-12-02 13:00:10 +00:00
video
Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.
2016-12-02 16:58:02 +00:00
voice_engine
Deprecated SetAudioPacketSize from RTPSender and removed calls to it.
2016-12-02 10:40:12 +00:00
.gitignore
…
BUILD.gn
Move webrtc/api/androidtests to webrtc/sdk/android/instrumentationtests
2016-12-01 09:45:35 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
…
common_types.cc
RTC_[D]CHECK_op: Remove superfluous casts
2016-11-28 23:59:03 +00:00
common_types.h
Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.
2016-12-02 16:58:02 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +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
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
typedefs.h
…
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +00:00
voice_engine_configurations.h
Remove API-related #defines from voice_engine_configurations.h
2016-12-02 10:30:23 +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.