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deadbeef b465980fd7 In end-to-end PeerConnection tests, allow video to be downscaled. 
QualityScaler may scale down the resolution, so our tests shouldn't
expect the input resolution to exactly match the resolution received on
the other side. Instead, we now just check that the aspect ratio
matches.

BUG=webrtc:5907

Review-Url: https://codereview.webrtc.org/2547673002
Cr-Commit-Position: refs/heads/master@{#15373}
2016-12-02 00:23:36 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
In end-to-end PeerConnection tests, allow video to be downscaled.
2016-12-02 00:23:36 +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
Wire up BitrateAllocation to be sent as RTCP TargetBitrate
2016-12-01 14:34:18 +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
Cleanup RtpHeaderExtensionMap removing use of two legacy functions
2016-12-01 16:39:44 +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
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
tools
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac
2016-11-30 14:12:10 +00:00
video
Wire up BitrateAllocation to be sent as RTCP TargetBitrate
2016-12-01 14:34:18 +00:00
voice_engine
Prep to remove API-related #defines from voice_engine_configurations.h
2016-11-30 16:21:54 +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
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
RTC_[D]CHECK_op: Remove superfluous casts
2016-11-28 23:59:03 +00:00
common_types.h
Replace some asserts with DCHECKs
2016-11-29 13:30:47 +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
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
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 3 defines in voice_engine_configurations.h
2016-11-30 08:12:57 +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.