| commit | b86d4e4a8dec1eb1b801244a2a97cda66f561d8e | [log] [tgz] |
|---|---|---|
| author | Stefan Holmer <stefan@webrtc.org> | Mon Dec 07 09:26:18 2015 |
| committer | Stefan Holmer <stefan@webrtc.org> | Mon Dec 07 09:26:32 2015 |
| tree | 97d795f9ebdc3e90bf34ca439d250fcdac1c7a55 | |
| parent | 03f80ebb8310e5f04ced856f7ec8f14b94a0f47e [diff] |
Prepare the AudioSendStream to be hooked up to send-side BWE. This CL contains three changes as a preparation for adding audio send streams to the send-side BWE: 1. Audio packets are passed through the pacer with high priority. This is needed to be able to set transport sequence numbers on the packets. 2. A feedback observer is passed to the audio stream's rtcp receiver so that the BWE can get notified of any BWE feedback being received on the audio feedback channel. 3. Support for the transport sequence number header extension is added to audio send streams. BUG=webrtc:5263,webrtc:5307 R=mflodman@webrtc.org, solenberg@webrtc.org Review URL: https://codereview.webrtc.org/1479023002 . Cr-Commit-Position: refs/heads/master@{#10909}
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.