signalapp/webrtc
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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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cd7da92012
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Rasmus Brandt cd7da92012 Add MediaCodec VP tests for uncommon resolutions. 
Bug: None
Change-Id: Ibfc35af3635c3b3a50027c4cd828f78e7a438dcd
Reviewed-on: https://webrtc-review.googlesource.com/72342
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23020}
2018-04-25 11:31:13 +00:00
api Remove PeerConnectionInterface::UpdateIce 2018-04-24 16:15:51 +00:00
audio Uses config struct with bitrate allocator. 2018-04-19 09:15:33 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Using microseconds in FakeNetworkPipe. 2018-04-25 10:32:23 +00:00
common_audio Disabling openmax_dl 2018-04-24 13:08:56 +00:00
common_video Move BitrateAllocation to api/ and rename it VideoBitrateAllocation 2018-04-23 15:31:27 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Android: Only use Java VideoFrames internally 2018-04-24 13:51:11 +00:00
infra Revert "Disable internal lite bot." 2018-04-11 06:45:07 +00:00
logging Remove the limit on concurrent RtcEventLogImpl instances 2018-04-25 10:33:43 +00:00
media Move BitrateAllocation to api/ and rename it VideoBitrateAllocation 2018-04-23 15:31:27 +00:00
modules Add MediaCodec VP tests for uncommon resolutions. 2018-04-25 11:31:13 +00:00
ortc Disable flaky OrtcFactoryIntegrationTests on debug iOS 64 builds. 2018-04-24 13:17:06 +00:00
p2p Change the receiving state update rule. 2018-04-17 01:00:02 +00:00
pc Add IsClosed checks to various PeerConnection methods 2018-04-24 23:07:21 +00:00
resources Add MediaCodec VP tests for uncommon resolutions. 2018-04-25 11:31:13 +00:00
rtc_base Make quality scaler downscale faster. 2018-04-25 09:08:21 +00:00
rtc_tools Create new API for RtcEventLogParser. 2018-04-25 09:37:03 +00:00
sdk Move setting videoFrameSize into the main dispatch queue. 2018-04-24 16:23:59 +00:00
stats PeerConnectionInterface::GetStats() with selector argument added. 2018-03-26 12:08:20 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Clean up posix-only files. 2018-04-18 00:18:04 +00:00
test Moving demux from FakeNetworkPipe to DirectTransport. 2018-04-25 10:13:03 +00:00
tools_webrtc Building "all" with client.webrtc iOS bots. 2018-04-24 09:23:24 +00:00
video Make quality scaler downscale faster. 2018-04-25 09:08:21 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Create .git-blame-ignore-revs and add Java format CL to it. 2016-10-20 09:20:39 +00:00
.gitignore Reland "Make it possible to run video_quality_loopback_test in swarming." 2018-01-23 13:03:17 +00:00
.gn Opt out of "Migrate the Android Support Lib to android_deps". 2018-04-05 13:40:53 +00:00
.vpython vpython: Specify dependency on pywin32 2018-02-14 13:56:39 +00:00
AUTHORS ObjC: Add missing _lastDrawnFrame assignments 2018-04-11 12:51:06 +00:00
BUILD.gn Removing definition of _CRT_SECURE_NO_WARNINGS. 2018-04-24 08:30:04 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Make Gerrit the default for WebRTC changes 2017-09-29 01:38:07 +00:00
common_types.cc Move BitrateAllocation to api/ and rename it VideoBitrateAllocation 2018-04-23 15:31:27 +00:00
common_types.h Move BitrateAllocation to api/ and rename it VideoBitrateAllocation 2018-04-23 15:31:27 +00:00
DEPS Roll chromium_revision a5ade7264f..3a1317cc43 (553159:553262) 2018-04-24 21:22:31 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
LICENSE_THIRD_PARTY Remove custom MD5 / SHA-1 implementations. 2018-02-19 15:03:35 +00:00
native-api.md Remove legacy VoiceEngine. 2018-01-12 11:31:52 +00:00
OWNERS Add back OWNERS entris that went missing 2018-03-06 10:31:11 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +00:00
presubmit_test.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +00:00
PRESUBMIT.py Add ToString() methods to classes with << operators, preparing for deprecations. 2018-04-03 11:21:30 +00:00
pylintrc Removing invalid-name from disabled pylint checks. 2017-10-11 08:06:49 +00:00
README.chromium Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
README.md Tell users where they can find the native API headers 2017-11-14 10:36:46 +00:00
style-guide.md Add style guidance about forward declarations. 2018-03-28 20:58:27 +00:00
typedefs.h Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix 2018-02-05 11:24:59 +00:00
WATCHLISTS Fixing root_files WATCHLIST regex. 2018-04-19 06:52:18 +00:00
webrtc.gni Disabling openmax_dl 2018-04-24 13:08:56 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +00:00

README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info