| commit | a8eb756a346682a4af66f9a9c0b87cfe1ed5b114 | [log] [tgz] |
|---|---|---|
| author | aleloi <aleloi@webrtc.org> | Mon Nov 28 15:02:13 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Nov 28 15:02:19 2016 |
| tree | 6c87d4226b7f04f1804575f15c232f74c283dc10 | |
| parent | 9abbf5ae4ec7d688a9b4aa03a405f3faadb74b90 [diff] |
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.
Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.
transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.
NOTRY=True
BUG=webrtc:5589, webrtc:5878, webrtc:6785
Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}
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.