This is just general cleanup.
The assumed behavior is late decoding, and this function is not used to make any decision (except in the deprecated jitter buffer).
Bug: webrtc:12271
Change-Id: Ifb48186d55903f068f25e44c5f73e7a724f6f456
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200804
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32940}
Deletes all webrtc usage of this member. Next step is to delete
any downstream references, and when that's done, the member can be
deleted.
Bug: None
Change-Id: I3f3a94a063dccf56468a1069653efd3809875b01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181201
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31911}
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.
Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}
