Due to recent Chrome infra changes in https://chromium-review.googlesource.com/c/472290/ tests running on swarming are now assumed to emit JSON results or will be marked as failing. This requires us to use our gtest-parallel wrapper for all our Swarming tests (or implement the --isolated-script-test-output flag, which normally only is implemented by the Chromium test launcher). The low_bandwidth_audio_test can actually run in parallel, so just change that. The webrtc_nonparallel_tests cannot, so this CL changes MB to pass --workers=1 flag to gtest-parallel, which makes the tests run in sequence. This adds a little confusion but the root problem is really that our gtest-parallel script [1] does a lot more than just running the tests in parallel these days, so it should probably be renamed. Also make sure gtest-parallel-wrapper.py [2] consumes the --isolated-script-test-chartjson-output flag (unused) so we don't pass it on to the test executable. [1]: https://chromium.googlesource.com/external/github.com/google/gtest-parallel/+/master/gtest-parallel [2]: https://chromium.googlesource.com/external/webrtc/+/master/tools-webrtc/gtest-parallel-wrapper.py BUG=709988 TBR=ehmaldonado@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2806373002 Cr-Commit-Position: refs/heads/master@{#17646}
Revert of CQ: Remove Linux ARM64 Debug trybot from default set. (patchset #1 id:1 of https://codereview.webrtc.org/2790263003/ )
Revert of Add first part of the network_tester functionality. (patchset #13 id:260001 of https://codereview.webrtc.org/2779233002/ )
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
Reland of PyLint fixes for tools-webrtc and webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2737233003/ )
Reland of Adding PRESUBMIT check on google::protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2791583002/ )
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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