commit dbdb3a00797d28c2e21191efd859d01b2533c003
author Stefan Holmer <stefan@webrtc.org> Tue Jul 17 16:03:46 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Tue Jul 17 14:46:15 2018 +0000
tree 65dea86b28ed87f0c415e6b1240bc7a8e755ce87
parent e1d7b23915c87cf52d70ef0f53680bc613ddfa90

Refactoring PayloadRouter.

- 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}

video/BUILD.gn

diff --git a/video/BUILD.gn b/video/BUILD.gn
index 7bf7404..a135696 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -77,6 +77,7 @@
     "../modules/video_coding:packet",
     "../modules/video_coding:video_codec_interface",
     "../rtc_base:checks",
+    "../rtc_base:rate_limiter",
     "../rtc_base:stringutils",
     "../rtc_base/experiments:alr_experiment",
     "../rtc_base/experiments:quality_scaling_experiment",