| commit | 3059378f7d62d6e4f8f344fcc534edea3ce9a295 | [log] [tgz] |
|---|---|---|
| author | Anders Carlsson <andersc@webrtc.org> | Thu Jun 22 12:27:27 2017 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Jun 26 07:19:11 2017 |
| tree | 56503032691bc6e53e593cba54d0108fd2cb3d00 | |
| parent | 7941fc6e093a891e3afbc5e3c652e2afd752e5ca [diff] |
Always reset the audio session configuration after a call. After returning from the call the AVAudioSession was configured to use the receiver instead of the speaker for audio output. The configuration was only restored if the sound loop was previously playing, this change makes sure that the configuration is always reset so the sound can be played audibly after a call has been finished. Bug: webrtc:7792 Change-Id: Idabf6fadc8041b18722cb8f5e89e0c8c36b1b74d Reviewed-on: https://chromium-review.googlesource.com/544819 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18754}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.