This tester is an improved version of VideoProcessor and VideoCodecTestFixture and will eventually replace them. The tester provides better separation between codecs and testing logic. Its knowledge about codecs is limited to frame encode/decode calls and frame ready callbacks. Instantiation and configuration of codecs are the test responsibilities. Other differences: - Run encoding and decoding in separate threads - Run quality analysis in a separate thread - Reference frame buffering is moved into video source (which re-read frames from the file). - Make it possible to run decode-only tests This CL is MVP implementation: it adds only 1 test (video_codec_test.cc, ConstantRate/EncodeDecodeTest) and the test is disabled for now. Bug: b/261160916 Change-Id: Ida24a2fca1b1496237fa695c812084877c76379f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283525 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38901}
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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 here 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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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