Erik Språng 696cea0843 Refactor some RtpSender-level tests into RtpRtcp-level tests
This prepares for ability to defer sequence number assignment to after
the pacing stage - a scenario where the RtpRtcp module rather than than
RTPSender class has responsibility for sequence numbering.

Bug: webrtc:11340
Change-Id: Ife88f60258b9b7cfd9dbd3326f02ac34da8f7603
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214967
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33702}
2021-04-13 08:37:14 +00:00
2018-10-05 14:40:21 +00:00
2021-04-12 14:03:57 +00:00
2021-04-12 22:24:56 +00:00
2021-04-12 22:24:56 +00:00
2021-04-12 07:21:03 +00:00
2021-01-20 15:01:07 +00:00
2020-07-13 11:42:07 +00:00
2021-03-22 11:57:23 +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.

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