06013e99ac
This CL implements webrtc::AecDump, which is an interface defined in https://codereview.webrtc.org/2778783002. This AudioProcessing submodule writes audio and APM state to a file. The file writing is done by posting IO tasks (write_to_file_task.h) on an rtc::TaskQueue. There is an existing implementation for this through AudioProcessing::StartDebugRecording() and AudioProcessing::StopDebugRecording(). This implementation still works, and is used as the default until this dependent CL: https://codereview.webrtc.org/2896813002/. To be able to build webrtc without protobuf support, the interface is isolated from protobuf types. Audio data from AudioProcessing is passed to AecDumpImpl through the AecDump interface. There it is stored in protobuf objects, which are posted on the task queue. This functionality is verified correct by the CL https://codereview.webrtc.org/2864373002, which enables this recording submodule in APM tests. BUG=webrtc:7404 Review-Url: https://codereview.webrtc.org/2865113002 Cr-Commit-Position: refs/heads/master@{#18241}
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