commit | 5de52fd38e8d80250d6e8ed79764ded1f39d4b5d | [log] [tgz] |
---|---|---|
author | aleloi <aleloi@webrtc.org> | Thu Nov 10 01:05:34 2016 -0800 |
committer | Commit bot <commit-bot@chromium.org> | Thu Nov 10 09:05:39 2016 +0000 |
tree | 249e027e9eb05a9fe1833e6cc048ca6e0d61d115 | |
parent | 37e4ad598d209fcec8abd677769e120b442007da [diff] |
Created a mocked AudioTransport. There are currently two nearly identical classes called MockAudioTransport defined in two unit tests: android/audio_transport_unittest.cc and /ios/audio_transport_unittest_ios.cc This change defines a common mocked AudioTransport. The two current mocks are rewritten to use the common one. A GN target is created for this mock and MockAudioDevice. This change will allow to provide a mocked AudioTransport to AudioState in a dependent CL https://codereview.webrtc.org/2454373002/ BUG=webrtc:6346 NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2493483002 Cr-Commit-Position: refs/heads/master@{#15010}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.