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}

modules/video_coding/jitter_buffer.cc

diff --git a/modules/video_coding/jitter_buffer.cc b/modules/video_coding/jitter_buffer.cc
index b98fd92..83f90e3 100644
--- a/modules/video_coding/jitter_buffer.cc
+++ b/modules/video_coding/jitter_buffer.cc
@@ -123,6 +123,9 @@
   }
 }
 
+Vp9SsMap::Vp9SsMap() {}
+Vp9SsMap::~Vp9SsMap() {}
+
 bool Vp9SsMap::Insert(const VCMPacket& packet) {
   if (!packet.video_header.vp9().ss_data_available)
     return false;