mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-15 06:40:43 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
No description
19773 commits 2 branches 91 tags 415 MiB
C++ 89.3%
Java 2.9%
C 2.8%
Objective-C++ 1.9%
Python 1.8%
Other 1%
Find a file
Autoroller 00a2c8dc9b Roll chromium_revision da2e37af11..f69055b78e (513169:513208) 
Change log: da2e37af11..f69055b78e
Full diff: da2e37af11..f69055b78e

Changed dependencies:
* src/ios: 39d5551d78..2ba066531a
* src/testing: 664e23d95d..7986d6d202
* src/third_party: 86359e278d..77782afb5d
* src/tools: 5d01fa2a2c..e5d5c39384
DEPS diff: da2e37af11..f69055b78e/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I3cbab0ee198008a91e9e4449e1301b1563c9e602
Reviewed-on: https://webrtc-review.googlesource.com/18020
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20541}
2017-11-01 20:12:46 +00:00
api Don't select audio codecs depending on GN vars build_with_{chromium|mozilla} 2017-11-01 18:59:27 +00:00
audio Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) 2017-11-01 11:04:26 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) 2017-11-01 11:04:26 +00:00
common_audio Replacing undefined left shifts with multiplication. 2017-10-31 09:43:02 +00:00
common_video Remove pbos@webrtc.org from all OWNERS. 2017-11-01 08:03:46 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Fix/suppress new warnings introduced in Chromium roll. 2017-10-30 16:10:29 +00:00
infra Link win_rel with LUCI win_rel. 2017-10-31 21:08:43 +00:00
logging Remove redundant unit tests for RtcEventLog. 2017-11-01 16:09:16 +00:00
media Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) 2017-11-01 11:04:26 +00:00
modules Don't select audio codecs depending on GN vars build_with_{chromium|mozilla} 2017-11-01 18:59:27 +00:00
ortc Disable flaky test OrtcFactoryIntegrationTest.BasicTwoWayAudioVideoRtpSendersAndReceivers. 2017-09-27 09:14:28 +00:00
p2p Enable the clang style plugin in primary p2p/ target 2017-11-01 00:19:05 +00:00
pc Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) 2017-11-01 11:04:26 +00:00
resources Fix of integer overflow in WebRtcAecm_ProcessBlock / ApmTest.Process 2017-10-23 14:25:37 +00:00
rtc_base Add MovingMedianFilter to rtc_base/numerics 2017-11-01 13:56:16 +00:00
rtc_tools Estimate RTP clock frequency and plot capture-send delay. 2017-10-26 08:42:54 +00:00
sdk Reland ObjC API for BWE allocation strategy 2017-11-01 15:21:06 +00:00
stats Added RTCMediaStreamTrackStats.jitterBufferDelay for audio 2017-10-02 10:47:00 +00:00
system_wrappers Remove references to and implementation of GetHistogramName(). 2017-10-30 19:20:49 +00:00
test Remove pbos@webrtc.org from all OWNERS. 2017-11-01 08:03:46 +00:00
tools_webrtc Remove pbos@webrtc.org from all OWNERS. 2017-11-01 08:03:46 +00:00
video Don't use old RTCP SR reports for remote clock estimation 2017-11-01 12:34:26 +00:00
voice_engine Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) 2017-11-01 11:04:26 +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 MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py. 2017-10-02 16:57:09 +00:00
.gn Remove remaining mentions of gflags 2017-09-25 15:34:41 +00:00
.vpython Add psutil to vpython dependencies (used on builder bots) 2017-09-04 08:04:16 +00:00
AUTHORS Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ ) 2017-09-25 13:37:12 +00:00
BUILD.gn Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." 2017-09-29 13:48:29 +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 Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
common_types.h Set RTPVideoHeader picture id in PayloadRouter if forced fallback for VP8 is enabled. 2017-10-06 13:41:14 +00:00
DEPS Roll chromium_revision da2e37af11..f69055b78e (513169:513208) 2017-11-01 20:12:46 +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 Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
OWNERS Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Enable cpplint in media/ 2017-10-31 17:46:42 +00:00
presubmit_test.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +00:00
presubmit_test_mocks.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +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 Update README.md and codereview.settings for new source location 2017-09-13 19:54:59 +00:00
style-guide.md Style guide: Attempt to make the L2 and L3 headings more visually distinct 2017-09-09 03:52:23 +00:00
typedefs.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
WATCHLISTS Add myself to the watchlist for webrtc/api/ and webrtc/base/ 2017-05-04 13:22:46 +00:00
webrtc.gni Don't select audio codecs depending on GN vars build_with_{chromium|mozilla} 2017-11-01 18:59:27 +00:00
whitespace.txt Moving src/webrtc into src/. 2017-09-15 04:25:06 +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.

More info