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Java 2.9%
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Mirko Bonadei fb59a6aa3f Return const char* from ToString(RTCErrorType error). 
Returning absl::string_view causes problems to the Chromium/WebRTC
component build because absl::operator<< needs to be exported.

This CL switches to `const char*` which should be enough to avoid
to generate temporaries.

Bug: webrtc:9419
Change-Id: If169a6f95c7efd21ac8ce108c7f2f80a76ff2313
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153842
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29250}
2019-09-20 08:27:52 +00:00
api Return const char* from ToString(RTCErrorType error). 2019-09-20 08:27:52 +00:00
audio Propagating TargetRate struct to BitrateAllocator. 2019-09-19 14:03:04 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Propagating TargetRate struct to BitrateAllocator. 2019-09-19 14:03:04 +00:00
common_audio Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
common_video Cleaning up C++14 move into lambda TODOs. 2019-09-17 19:18:26 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Fixing some typos. 2019-09-10 10:03:50 +00:00
examples New build target api:media_interface 2019-09-19 09:32:27 +00:00
logging Adds remote estimates to rtc event log. 2019-09-19 09:22:37 +00:00
media New build target api:media_interface 2019-09-19 09:32:27 +00:00
modules Delete dead code inside #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED 2019-09-20 07:40:10 +00:00
p2p Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
pc New build target api:media_interface 2019-09-19 09:32:27 +00:00
resources Use the AEC3 high-pass filter for the whole APM 2019-08-23 20:04:10 +00:00
rtc_base Improve field trial error message. 2019-09-19 09:38:49 +00:00
rtc_tools Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
sdk Adds logging of audio sessions status on the recording side in ADM for Android. 2019-09-19 11:35:10 +00:00
stats Add qualityLimitationResolutionChanges stat 2019-09-09 15:22:57 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Move code related to VideoCodingModule to its own build target 2019-09-10 12:34:38 +00:00
test Add frame receive to frame rendered metric to video_quality_analyzer 2019-09-19 14:43:04 +00:00
tools_webrtc Stop explicitly setting use_prebuilt_instrumented_libraries on msan bots. 2019-09-12 18:21:38 +00:00
video Make GetBitstream non-virtual since it is no longer needed for testing. 2019-09-19 14:04:09 +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 Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH. 2019-09-04 18:49:28 +00:00
.gn Switch to compiling WebRTC -std=c++14 by default 2019-09-09 19:24:16 +00:00
.vpython Add vpython dependencies needed to run presubmit tests on LUCI 2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md Update style guide for absl::make_unique. 2019-09-18 06:10:58 +00:00
AUTHORS Revert "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5." 2019-09-06 05:36:23 +00:00
BUILD.gn Introduce api/crypto/BUILD.gn. 2019-09-13 17:21:47 +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 a536fa4a4a..303c57cf17 (698214:698351) 2019-09-20 04:41:10 +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 juberti@ to webrtc root owners 2019-05-17 18:11:58 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py absl::make_unique presubmit check. 2019-09-17 17:47:31 +00:00
presubmit_test.py Fixing py lint errors 2018-07-23 15:28:48 +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 Update WebRTC's C++ style guide to reflect the switch to C++14. 2019-09-16 11:45:35 +00:00
WATCHLISTS Add saza to audio watchlists 2019-09-03 14:55:43 +00:00
webrtc.gni Remove rtc_use_lto GN arg. 2019-08-20 14:00:49 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +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