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Jonas Olsson addc380168 Change some SSL logging to use DLOG 
Bug: webrtc:8529
Change-Id: I0242ff201c5c7ac00169444a346e462157703ac6
Reviewed-on: https://webrtc-review.googlesource.com/46260
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21836}
2018-02-01 09:53:51 +00:00
api Removing useless dependencies on //testing/gmock. 2018-01-26 13:34:12 +00:00
audio Revert "Break up rtc_event_log_api to solve circular dependencies." 2018-01-31 09:39:44 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Revert "Break up rtc_event_log_api to solve circular dependencies." 2018-01-31 09:39:44 +00:00
common_audio Removing useless dependencies on //testing/gmock. 2018-01-26 13:34:12 +00:00
common_video Reland "Rename stereo video codec to multiplex" 2018-01-29 20:37:59 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Ignore Chromium-specific flags on video_quality_loopback_test. 2018-01-31 13:50:09 +00:00
infra Make new Chromium compile trybots mandatory. 2018-01-12 09:41:51 +00:00
logging Revert "Break up rtc_event_log_api to solve circular dependencies." 2018-01-31 09:39:44 +00:00
media Revert "Break up rtc_event_log_api to solve circular dependencies." 2018-01-31 09:39:44 +00:00
modules Revert "Updates tests for turning simulcast streams on/off." 2018-01-31 22:11:17 +00:00
ortc Revert "Break up rtc_event_log_api to solve circular dependencies." 2018-01-31 09:39:44 +00:00
p2p Removing henrika from p2p/OWNERS and rtc_base/OWNERS 2018-01-30 10:16:19 +00:00
pc Revert "Break up rtc_event_log_api to solve circular dependencies." 2018-01-31 09:39:44 +00:00
resources Add a new NetEq decoding unit test for Opus with DTX 2017-11-28 10:45:38 +00:00
rtc_base Change some SSL logging to use DLOG 2018-02-01 09:53:51 +00:00
rtc_tools Revert "Break up rtc_event_log_api to solve circular dependencies." 2018-01-31 09:39:44 +00:00
sdk Revert "Break up rtc_event_log_api to solve circular dependencies." 2018-01-31 09:39:44 +00:00
stats Removing useless dependencies on //testing/gmock. 2018-01-26 13:34:12 +00:00
system_wrappers Reland Use runtime enabled features API to enable dual stream mode 2018-01-18 12:22:49 +00:00
test Revert "Break up rtc_event_log_api to solve circular dependencies." 2018-01-31 09:39:44 +00:00
tools_webrtc Parameterize PeerConnection end to end tests for Unified Plan 2018-01-25 04:03:18 +00:00
video Fix spelling of (internal) method name UpdateChannelParameters. 2018-01-31 14:26:49 +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 Pinning NDK api levels. 2018-01-18 16:55:58 +00:00
.vpython Add psutil to vpython dependencies (used on builder bots) 2017-09-04 08:04:16 +00:00
AUTHORS Add missing stdio.h header in files using scanf/sscanf function. 2018-01-26 13:15:52 +00:00
BUILD.gn Enable building WebRTC without built-in software codecs 2018-01-31 08:33:59 +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 Add flexfec payload name to string-type conversions 2018-01-31 08:58:39 +00:00
common_types.h Reland "Rename stereo video codec to multiplex" 2018-01-29 20:37:59 +00:00
DEPS Roll chromium_revision ebad4703ef..1bf6c49e7a (533453:533562) 2018-02-01 05:37:32 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt
LICENSE_THIRD_PARTY Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
native-api.md Remove legacy VoiceEngine. 2018-01-12 11:31:52 +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 Delete unused MediaFile module. 2018-01-29 11:18:18 +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 Tell users where they can find the native API headers 2017-11-14 10:36:46 +00:00
style-guide.md Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" 2018-01-10 15:55:04 +00:00
typedefs.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
WATCHLISTS Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
webrtc.gni Enable building WebRTC without built-in software codecs 2018-01-31 08:33:59 +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.

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

More info