commit | 0d1305ee8873d4ae3d6a0401056cb3dab78fc390 | [log] [tgz] |
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author | frederik.riedel <frederik.riedel@frogg.io> | Thu Feb 23 21:57:00 2017 |
committer | Commit bot <commit-bot@chromium.org> | Thu Feb 23 21:57:00 2017 |
tree | 2ffa9da469ad107c048ccbfd98f3994453ddfbf4 | |
parent | 7aadbfa06e7296af0d0e7b4b793ae9f978d24fbc [diff] |
Added support for changing the volume of RTCAudioSource as discussed in BUG=webrtc:6533 This is a short term solution to change the volume of a RTCAudioTrack (which contains an RTCAudioSource property) until applyConstraints for RTCMediaStreamTracks has been implemented. This CL adds one new Objective-C method to AudioSourceInterface's wrapper: -(void)setVolume:(double)volume BUG=webrtc:6533, webrtc:6805 This is my first CL for Chromium/WebRTC, so please let me know if I did something wrong. Review-Url: https://codereview.webrtc.org/2534843002 Cr-Commit-Position: refs/heads/master@{#16809}
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.