commit | 309aafe3513af2c53439cb0ea105a39156547acb | [log] [tgz] |
---|---|---|
author | Piotr (Peter) Slatala <psla@webrtc.org> | Tue Jan 15 14:24:34 2019 -0800 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Jan 16 15:23:17 2019 +0000 |
tree | 3293d25dffe72dfde3949bb2f5fe375354c48e52 | |
parent | e7d08df83764208982a61f6128be544e7b590442 [diff] |
Add 'AudioPacket' notification to media transport interface. So far, base channel was only notifying about 'first audio packet' when RTP was used, and it never notified about it when media_transport interface was used. This change adds a sigslot to notify about a new media packet to the media transport interface. Bug: webrtc:9719 Change-Id: Ie9230c407f35b1aaa71ba71008ac34ba8869e2d4 Reviewed-on: https://webrtc-review.googlesource.com/c/117249 Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26282}
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.
Authoritative list of directories that contain the native API header files.