Log SendSideCongestionController experiment
The SendSideCongestionController version is toggled by an experiment in
RtpTransportControllerSend. This CL adds a log statement of which
version is used, to make debugging easier.
Bug: webrtc:8415
Change-Id: I6201cf5f03e097cc07c6ae120dcff075c046c414
Reviewed-on: https://webrtc-review.googlesource.com/69808
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22858}diff --git a/call/rtp_transport_controller_send.cc b/call/rtp_transport_controller_send.cc
index a0c427a..23d8e0a 100644
--- a/call/rtp_transport_controller_send.cc
+++ b/call/rtp_transport_controller_send.cc
@@ -34,10 +34,12 @@
const BitrateConstraints& bitrate_config,
bool task_queue_controller) {
if (task_queue_controller) {
+ RTC_LOG(LS_INFO) << "Using TaskQueue based SSCC";
return rtc::MakeUnique<webrtc::webrtc_cc::SendSideCongestionController>(
clock, event_log, pacer, bitrate_config.start_bitrate_bps,
bitrate_config.min_bitrate_bps, bitrate_config.max_bitrate_bps);
}
+ RTC_LOG(LS_INFO) << "Using Legacy SSCC";
auto cc = rtc::MakeUnique<webrtc::SendSideCongestionController>(
clock, nullptr /* observer */, event_log, pacer);
cc->SignalNetworkState(kNetworkDown);