[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
59e1464fcd
These fixes are automatically created by various analysis tools, but have been manually triggered to be applied. * the 'empty' method should be used to check for emptiness instead of 'size' (3 times) * using decl 'Return' is unused (4 times) * using decl '_' is unused (3 times) * using decl 'DoAll' is unused (2 times) * using decl 'SetArgPointee' is unused * using decl 'Dlrr' is unused * using decl 'IsEmpty' is unused * redundant get() call on smart pointer * using decl 'Invoke' is unused (2 times) * using decl 'SizeIs' is unused (3 times) * using decl 'make_tuple' is unused * using decl 'NiceMock' is unused * using decl 'SaveArg' is unused (2 times) * using decl 'AtLeast' is unused * using decl 'ElementsAre' is unused * using decl 'Gt' is unused Bug: None Change-Id: I97658fb0e94620b8319d7c3da29b15e27ec23188 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151133 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29056} |
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| api | ||
| audio | ||
| build_overrides | ||
| call | ||
| common_audio | ||
| common_video | ||
| data | ||
| examples | ||
| logging | ||
| media | ||
| modules | ||
| p2p | ||
| pc | ||
| resources | ||
| rtc_base | ||
| rtc_tools | ||
| sdk | ||
| stats | ||
| style-guide | ||
| system_wrappers | ||
| test | ||
| tools_webrtc | ||
| video | ||
| .clang-format | ||
| .git-blame-ignore-revs | ||
| .gitignore | ||
| .gn | ||
| .vpython | ||
| abseil-in-webrtc.md | ||
| AUTHORS | ||
| BUILD.gn | ||
| CODE_OF_CONDUCT.md | ||
| codereview.settings | ||
| common_types.h | ||
| DEPS | ||
| ENG_REVIEW_OWNERS | ||
| LICENSE | ||
| license_template.txt | ||
| native-api.md | ||
| OWNERS | ||
| PATENTS | ||
| presubmit_test_mocks.py | ||
| presubmit_test.py | ||
| PRESUBMIT.py | ||
| pylintrc | ||
| README.chromium | ||
| README.md | ||
| style-guide.md | ||
| WATCHLISTS | ||
| webrtc.gni | ||
| whitespace.txt | ||
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
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct