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Java 2.9%
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Steve Anton e78bcb97c3 Enable cpplint in media/ 
Bug: webrtc:5584
Change-Id: I2fd1395d35596d9002e19cc90fcda3a5d4cde9e7
Reviewed-on: https://webrtc-review.googlesource.com/16564
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20504}
2017-10-31 17:46:42 +00:00
api Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API 2017-10-31 12:35:42 +00:00
audio Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API 2017-10-31 12:35:42 +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 2017-10-31 12:35:42 +00:00
common_audio Replacing undefined left shifts with multiplication. 2017-10-31 09:43:02 +00:00
common_video Use $rtc_libyuv_dir in common_video/BUILD.gn, not hard-encoded "libyuv" 2017-10-16 16:05:37 +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 Temporarily remove linux_ubsan from commit queue 2017-10-27 09:49:17 +00:00
logging Fix AudioLevel print-out in rtc_event_log2text 2017-10-31 14:42:03 +00:00
media Enable cpplint in media/ 2017-10-31 17:46:42 +00:00
modules New PacedSender constructor with injected PacketQueue 2017-10-31 11:39:22 +00:00
ortc Disable flaky test OrtcFactoryIntegrationTest.BasicTwoWayAudioVideoRtpSendersAndReceivers. 2017-09-27 09:14:28 +00:00
p2p Fix clang style warnings in p2p/base/portallocator files 2017-10-31 16:58:22 +00:00
pc Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API 2017-10-31 12:35:42 +00:00
resources Fix of integer overflow in WebRtcAecm_ProcessBlock / ApmTest.Process 2017-10-23 14:25:37 +00:00
rtc_base Adding RTC_ prefixed LOG macros. 2017-10-27 08:24:37 +00:00
rtc_tools Estimate RTP clock frequency and plot capture-send delay. 2017-10-26 08:42:54 +00:00
sdk Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API 2017-10-31 12:35:42 +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 Allow injection of custom network models in place of FakeNetworkPipe. 2017-10-26 11:11:25 +00:00
tools_webrtc MB: Add Android Perf (swarming) 2017-10-23 13:02:07 +00:00
video Fix typo in VideoSendTiming header extension structure 2017-10-31 11:20:22 +00:00
voice_engine Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API 2017-10-31 12:35:42 +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 de566f6b16..07493032b5 (512797:512803) 2017-10-31 13:13:02 +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 Remove QUIC transport/data channel 2017-10-24 16:14:18 +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