e8a77e3309
This change is a major refactoring of the neteq_rtpplay tool. It consists of the following parts: - NetEqTest class: Breaks out the main simulation loop from neteq_rtpplay into a separate class with well defined inputs and outputs. - NetEqInput: Interface class for the input to NetEqTest. - NetEqPacketSourceInput: Implementation of NetEqInput that provides a PacketSource objects with a NetEqInput interface. This has two subclasses; one for RtpFileSource and one for RtcEventLogSource. - NetEqReplacementInput: An object that modifies the packets provided by another NetEqInput object, and replaces the packet payloads with meta data readable by a FakeDecodeFromFile decoder. - FakeDecodeFromFile: An AudioDecoder implementation that produces "decoded" data by reading from an audio file. BUG=webrtc:2692, webrtc:5447 Review-Url: https://codereview.webrtc.org/2020363003 Cr-Commit-Position: refs/heads/master@{#13252}
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. This page is maintained by the Google Chrome team.
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