Danil Chapovalov fb1a0f0e1f Cleanup rtp utils in media/base
Remove unused functions GetRtpHeader/GetRtpHeaderLength
Replace usage of SetRtpHeader with webrtc::RtpPacket
Move SetRtpSsrc next to the only place it is used.

Bug: None
Change-Id: I3ecc244b1a2bdb2d68e0dbdb34dd60160a3101f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225547
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34447}
2021-07-09 17:48:26 +00:00
2018-10-05 14:40:21 +00:00
2021-07-09 17:48:26 +00:00
2021-06-21 22:21:04 +00:00
2021-07-09 13:50:29 +00:00
2021-07-09 07:49:43 +00:00
2021-01-20 15:01:07 +00:00
2021-04-26 16:39:07 +00:00
2020-07-13 11:42:07 +00:00

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.

Development

See here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info

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