Removing access to send side cc in rtp controller.
This CL removes direct access to SendSideCongestionController (SSCC) via
the RtpTransportControllerSend interface and replaces all usages with
calls on RtpTransportControllerSend which will in turn calls SSCC. This
prepares for later refactor of RtpTransportControllerSend.
Bug: webrtc:8415
Change-Id: I68363a3ab0203b95579f747402a1e7f58a5eeeb5
Reviewed-on: https://webrtc-review.googlesource.com/53860
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22044}diff --git a/rtc_tools/event_log_visualizer/analyzer.cc b/rtc_tools/event_log_visualizer/analyzer.cc
index c32c676..0249ec4 100644
--- a/rtc_tools/event_log_visualizer/analyzer.cc
+++ b/rtc_tools/event_log_visualizer/analyzer.cc
@@ -671,7 +671,7 @@
<< " (LOG_START, LOG_END) segments in log.";
}
-class BitrateObserver : public SendSideCongestionController::Observer,
+class BitrateObserver : public NetworkChangedObserver,
public RemoteBitrateObserver {
public:
BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {}