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Tim Na b8c775aeaf Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api 
Bug: webrtc:11251
Change-Id: Id3b6ff1814931d8250c4aaac59e494521fbe93ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164560
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30238}
2020-01-13 18:31:30 +00:00
api Revert "Reland "Reland "Reland "Distinguish between send and receive video codecs"""" 2020-01-13 09:03:37 +00:00
audio Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api 2020-01-13 18:31:30 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api 2020-01-13 18:31:30 +00:00
common_audio Remove potential real-time reallocation in PushResampler 2019-12-11 13:16:37 +00:00
common_video Add ability to resize buffers pool in decoder and use it in IVF generator 2019-12-16 14:51:16 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Update Linux documentation links 2020-01-10 07:51:46 +00:00
examples Cleanup: Replace MessageQueue pointers with Thread pointers. 2020-01-10 19:03:12 +00:00
logging Prefix ENABLE_RTC_EVENT_LOG with WEBRTC_. 2019-11-29 09:45:50 +00:00
media Add saza@ and peah@ to OWNERS of some audio files 2020-01-13 12:31:21 +00:00
modules Delete RtpGenericDepacketizer as no longer used 2020-01-13 13:45:37 +00:00
p2p Avoid [[nodiscard]] warning C4834 with MSVC 2019 2020-01-13 10:07:35 +00:00
pc Revert "Reland "Reland "Reland "Distinguish between send and receive video codecs"""" 2020-01-13 09:03:37 +00:00
resources Make the high-pass filter operate in full-band 2019-12-18 16:01:24 +00:00
rtc_base Cleanup: Merges Thread and MessageQueue. 2020-01-13 13:53:20 +00:00
rtc_tools [Android] Replace java_files with sources 2020-01-02 20:08:20 +00:00
sdk [iOS] Reset VT session when H264 decoder malfunction error happen 2020-01-13 14:57:36 +00:00
stats Implement RTCOutboundRtpStreamStats::remoteId. 2020-01-07 17:26:01 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Add directive to make webrtc metrics optional. 2019-12-09 13:55:50 +00:00
test Delete RtpGenericDepacketizer as no longer used 2020-01-13 13:45:37 +00:00
tools_webrtc [UBSan] Remove suppression for opus. 2019-12-10 08:59:30 +00:00
video Move DegradationPreference logic out of VideoSourceSinkController. 2020-01-13 17:24:48 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore new format cleanup. 2019-07-11 16:18:51 +00:00
.gitignore Add .clangd to .gitignore 2019-10-28 12:27:50 +00:00
.gn Switch to compiling WebRTC -std=c++14 by default 2019-09-09 19:24:16 +00:00
.vpython Add source-side perf upload script for WebRTC. 2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md Fix typo in abseil-in-webrtc.md. 2019-12-18 14:27:34 +00:00
AUTHORS Update Android camera switch API to allow specifying a name 2020-01-09 16:04:09 +00:00
BUILD.gn [Android] Replace java_files with sources 2020-01-02 20:08:20 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit 2018-10-25 08:19:53 +00:00
common_types.h Format almost everything. 2019-07-08 13:45:15 +00:00
DEPS Roll chromium_revision d794106d9d..b581de5b1b (730346:730447) 2020-01-12 14:38:41 +00:00
ENG_REVIEW_OWNERS Enforce LGTM from owners of depends-on paths in DEPS via presubmit. 2018-09-28 12:49:54 +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
native-api.md Delete unused I420 "codec" 2018-12-18 12:30:58 +00:00
OWNERS Add #COMPONENT to WebRTC. 2019-10-08 12:20:39 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Make RTCAudioSession accessible to Swift. 2020-01-08 09:15:25 +00:00
presubmit_test.py Use source_sets in component builds and static_library in release builds. 2019-10-17 21:17:18 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
pylintrc Fixing py lint errors 2018-07-23 15:28:48 +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 guidance to style guide how to reference a bug in a TODO 2019-12-11 11:55:52 +00:00
WATCHLISTS Add saza to audio watchlists 2019-09-03 14:55:43 +00:00
webrtc.gni Do not disable metrics by default. 2019-12-11 08:08:58 +00:00
webrtc_lib_link_test.cc Rewrite the lib link test to just be a binary. 2019-10-18 07:42:20 +00:00
whitespace.txt Revert "Whitespace change" 2019-11-11 14:58:20 +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