webrtc/call/test
Marina Ciocea e77912ba8c Insert frame transformer between Encoded and Packetizer.
Add a new API in RTPSenderInterface, to be called from the browser side
to insert a frame transformer between the Encoded and the Packetizer.

The frame transformer is passed from RTPSenderInterface through the
library to be eventually set in RTPSenderVideo, where the frame
transformation will occur in the follow-up CL
https://webrtc-review.googlesource.com/c/src/+/169128.

Insertable Streams Web API explainer:
https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
Change-Id: I46cd0d8a798c2736c837e90cbf90d8901c7d27fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169127
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30642}
2020-02-28 07:43:13 +00:00
..
mock_audio_send_stream.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mock_bitrate_allocator.h Declares BitrateAllocator methods const. 2018-10-24 15:26:36 +00:00
mock_rtp_packet_sink_interface.h Format almost everything. 2019-07-08 13:45:15 +00:00
mock_rtp_transport_controller_send.h Insert frame transformer between Encoded and Packetizer. 2020-02-28 07:43:13 +00:00