| commit | 95eb1ba0db79d8fd134ae61b0a24648598684e8a | [log] [tgz] |
|---|---|---|
| author | ossu <ossu@webrtc.org> | Wed Jul 13 13:05:27 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Jul 13 13:05:32 2016 |
| tree | a9a455298fa37e163e15d880e8cb9f43b0cba18d | |
| parent | 7470eb7977c153d2a512f8376601b15c6266fa25 [diff] |
WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs. Changed WebRtcVoiceEngine to present receive codecs from the formats provided by its decoder factory. Added supported formats to BuiltinAudioDecoderFactory. Added helper functions for creating some simple decoder factories for mocking. Created a PayloadTypeMapper for assigning payload types to formats. I think we'll eventually want to use this further up, or possibly in both the audio and video sides. It would be best if the engines didn't have to talk payload types at all, but it might be more difficult to get around when payload types depend on each-other, like the RTX codecs for video. This CL also includes some changes to rtc::Optional. I've put them in a separate CL that should (or should not) land first, making these changes void. See: https://codereview.webrtc.org/2072713002/ BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2072753002 Cr-Commit-Position: refs/heads/master@{#13459}
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.