webrtc/call/call_config.cc
Henrik Boström cf2856b01c Add parameter to control the pacer's burst outside of field trials.
BurstyPacer is currently controlled via field trials. In order for
Chrome to be able to have burst without relying on a field trial, this
parameter is added.

When all burst experiments have concluded we may be able to have a
hardcoded constant instead, but for now the parameter is added to
RTCConfiguration.

NOTRY=True

Bug: chromium:1354491
Change-Id: I386c1651dbbcbf309c15ea3d3380cf8f632b5429
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283420
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38621}
2022-11-15 08:46:30 +00:00

41 lines
1.3 KiB
C++

/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "call/call_config.h"
#include "rtc_base/checks.h"
namespace webrtc {
CallConfig::CallConfig(RtcEventLog* event_log,
TaskQueueBase* network_task_queue /* = nullptr*/)
: event_log(event_log), network_task_queue_(network_task_queue) {
RTC_DCHECK(event_log);
}
CallConfig::CallConfig(const CallConfig& config) = default;
RtpTransportConfig CallConfig::ExtractTransportConfig() const {
RtpTransportConfig transportConfig;
transportConfig.bitrate_config = bitrate_config;
transportConfig.event_log = event_log;
transportConfig.network_controller_factory = network_controller_factory;
transportConfig.network_state_predictor_factory =
network_state_predictor_factory;
transportConfig.task_queue_factory = task_queue_factory;
transportConfig.trials = trials;
transportConfig.pacer_burst_interval = pacer_burst_interval;
return transportConfig;
}
CallConfig::~CallConfig() = default;
} // namespace webrtc