configuration, inputs and outputs over a period of time. It is activated by AudioProcessing::StartRecording. The data is stored in binary protobuf format in a specified file. The file IO is, as of this CL, done from the real-time audio thread. This CL contains an interface for AecDump, a new APM submodule that will handle the recordings. Calls to the new interface from the AudioProcessingModule are added. These calls have no effect, and for a short while, audio_processing_impl.cc will contain two copies of recording calls. The original calls are guarded by the WEBRTC_AUDIOPROC_DEBUG_DUMP preprocessor define. They still have an effect, while the new ones do not. In the following CLs, the old recording calls will be removed, and an implementation of AecDump added. The reasons for the refactoring is to move file IO operations from the real-time audio thread, to add a top-level low-priority task queue for logging tasks like this, to simplify and modularize audio_processing_impl.cc and remove some of the preprocessor directives. These goals will be archived by the upcoming CLs. The implementation is in https://codereview.webrtc.org/2865113002. BUG=webrtc:7404 Review-Url: https://codereview.webrtc.org/2778783002 Cr-Commit-Position: refs/heads/master@{#18233}
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
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/ )
…
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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