Erik Språng 5cc9957062 Batch assign RTP seq# for all packets of a frame.
This avoids a potential race where other call sites could assign
sequence numbers while the video frame is mid packetization - resulting
in a non-contiguous video sequence.

Avoiding the tight lock-unlock within the loop also couldn't hurt from
a performance standpoint.

Bug: webrtc:12448
Change-Id: I6cc31c7743d2ca75caeaeffb98651a480dbe08e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207867
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33291}
2021-02-17 15:27:08 +00:00
2021-02-16 12:41:35 +00:00
2021-02-16 12:41:35 +00:00
2021-02-16 12:41:35 +00:00
2021-01-20 15:01:07 +00:00
2020-07-13 11:42:07 +00:00
2021-02-04 11:17:55 +00:00
2021-02-14 19:14:44 +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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