ApmTest.VerifyDebugDump(Int|Float) more insightful with failures
Instead of showing individual byte differences, this CL detects differences in the expected and actual byte streams of the evaluated AEC dump and, if detected, parses the `audioproc::Event` proto lite messages and calls EXPECT_EQ() for a subset of individual (sub-)fields. Note that messages are parsed only if the byte streams of each message pair do not match, so with no failures the test runs at no extra cost. Plus, the the added funcionality can only be enabled for local debugging by flipping the `kDumpWhenExpectMessageEqFails` flag - a code change cannot land if the flag is set to true. Note that `MessageDifferencer` (see [1]) could not be used because it is not implemented for `MessageLite` protos. [1] https://developers.google.com/protocol-buffers/docs/reference/cpp/google.protobuf.util.message_differencer Bug: b/241923537 Change-Id: I8e0eda3b1ecfe06945b6dad5ee8871f8200d76d7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270922 Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37765}
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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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