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Fredrik Solenberg cf73c96a79 Add AudioDeviceModule to AudioState::Config. 
This is to prepare client code for landing https://webrtc-review.googlesource.com/c/src/+/26681.

Bug: webrtc:4690
Change-Id: I82b24d876f9345ca7f59bfd6fc7ab26ba694b0d8
Reviewed-on: https://webrtc-review.googlesource.com/28320
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21043}
2017-12-04 15:18:59 +00:00
api Stop using public_deps to depend on libyuv. 2017-12-04 14:16:08 +00:00
audio Use the new AudioProcessing statistics everywhere. 2017-11-24 18:17:39 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Add AudioDeviceModule to AudioState::Config. 2017-12-04 15:18:59 +00:00
common_audio Optional: Use nullopt and implicit construction in /common_audio 2017-11-28 15:46:48 +00:00
common_video Stop using public_deps to depend on libyuv. 2017-12-04 14:16:08 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Roll Chromium + Fix Android lint suppressions 2017-11-30 16:59:50 +00:00
infra Adding win_asan to CQ. 2017-11-22 09:12:18 +00:00
logging Stop using ByteSize (deprecated) to get the size of a proto message. 2017-11-30 14:27:50 +00:00
media Stop using public_deps to depend on libyuv. 2017-12-04 14:16:08 +00:00
modules Stop using public_deps to depend on libyuv. 2017-12-04 14:16:08 +00:00
ortc Stop using public_deps in ortc. 2017-12-04 14:57:08 +00:00
p2p Fix cpplint errors in port/port_unittest 2017-11-29 19:57:09 +00:00
pc Call SrtpTransport::EnableExternalAuth when enabling SDES. 2017-11-30 23:47:30 +00:00
resources Add a new NetEq decoding unit test for Opus with DTX 2017-11-28 10:45:38 +00:00
rtc_base Remove dependency on rtc::Event from rtc::Thread. 2017-12-04 15:15:08 +00:00
rtc_tools Stop using public_deps to depend on libyuv. 2017-12-04 14:16:08 +00:00
sdk Stop using public_deps to depend on libyuv. 2017-12-04 14:16:08 +00:00
stats Add network_type to local RTCIceCandidateStats 2017-11-21 19:58:37 +00:00
system_wrappers Reland of "Use moving median filters in RemoteNtpTimeEstimator, RtpToNtpEstimator"" 2017-11-16 12:10:04 +00:00
test Stop using public_deps to depend on libyuv. 2017-12-04 14:16:08 +00:00
tools_webrtc Roll Chromium + Fix Android lint suppressions 2017-11-30 16:59:50 +00:00
video PictureIdTest: Add tests for temporal layer 0 picture index, tl0_pic_idx. 2017-12-04 11:03:48 +00:00
voice_engine Remove voice_engine_defines.h 2017-11-30 10:22:10 +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 Roll chromium_revision f93b8b19f2..adf969a7cb (513366:514871) and more. 2017-11-08 21:42:48 +00:00
.gn Roll chromium_revision 5df4e1bfe7..5bd5874cbf (518692:519731) + iOS fix 2017-11-28 18:42:38 +00:00
.vpython Add psutil to vpython dependencies (used on builder bots) 2017-09-04 08:04:16 +00:00
AUTHORS Delete wrapper API ConvertToI420 for YUV conversion to I420 2017-11-15 11:10:20 +00:00
BUILD.gn Remove webrtc_tests. 2017-12-04 10:15:18 +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 a new function to BitrateAllocation: HasBitrate. 2017-11-23 15:00:08 +00:00
common_types.h Reland "Add stereo codec header and pass it through RTP" 2017-11-30 01:44:19 +00:00
DEPS Roll chromium_revision 0eb1e0ff9e..7d467b79bf (521321:521327) 2017-12-04 14:17:48 +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
native-api.md Tell users where they can find the native API headers 2017-11-14 10:36:46 +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 Tell users where they can find the native API headers 2017-11-14 10:36:46 +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 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 Removing assert in rtc_test template. 2017-11-22 08:26:37 +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