webrtc/modules/congestion_controller/rtp/control_handler.cc
Danil Chapovalov dda037db07 Remove unused field trial DisablePacerEmergencyStop
This field trial was added 5 years ago in
https://webrtc-review.googlesource.com/c/src/+/111883
probably as a safe guard, but looks never used.

Bug: webrtc:11503
Change-Id: Ia9544b652b25fad4c614d66fe020f3d994c96505
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333380
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41490}
2024-01-09 15:03:34 +00:00

73 lines
2.5 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 "modules/congestion_controller/rtp/control_handler.h"
#include <algorithm>
#include <vector>
#include "api/units/data_rate.h"
#include "modules/pacing/pacing_controller.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/safe_minmax.h"
namespace webrtc {
void CongestionControlHandler::SetTargetRate(
TargetTransferRate new_target_rate) {
RTC_DCHECK_RUN_ON(&sequenced_checker_);
RTC_CHECK(new_target_rate.at_time.IsFinite());
last_incoming_ = new_target_rate;
}
void CongestionControlHandler::SetNetworkAvailability(bool network_available) {
RTC_DCHECK_RUN_ON(&sequenced_checker_);
network_available_ = network_available;
}
void CongestionControlHandler::SetPacerQueue(TimeDelta expected_queue_time) {
RTC_DCHECK_RUN_ON(&sequenced_checker_);
pacer_expected_queue_ms_ = expected_queue_time.ms();
}
absl::optional<TargetTransferRate> CongestionControlHandler::GetUpdate() {
RTC_DCHECK_RUN_ON(&sequenced_checker_);
if (!last_incoming_.has_value())
return absl::nullopt;
TargetTransferRate new_outgoing = *last_incoming_;
DataRate log_target_rate = new_outgoing.target_rate;
bool pause_encoding = false;
if (!network_available_) {
pause_encoding = true;
} else if (pacer_expected_queue_ms_ >
PacingController::kMaxExpectedQueueLength.ms()) {
pause_encoding = true;
}
if (pause_encoding)
new_outgoing.target_rate = DataRate::Zero();
if (!last_reported_ ||
last_reported_->target_rate != new_outgoing.target_rate ||
(!new_outgoing.target_rate.IsZero() &&
(last_reported_->network_estimate.loss_rate_ratio !=
new_outgoing.network_estimate.loss_rate_ratio ||
last_reported_->network_estimate.round_trip_time !=
new_outgoing.network_estimate.round_trip_time))) {
if (encoder_paused_in_last_report_ != pause_encoding)
RTC_LOG(LS_INFO) << "Bitrate estimate state changed, BWE: "
<< ToString(log_target_rate) << ".";
encoder_paused_in_last_report_ = pause_encoding;
last_reported_ = new_outgoing;
return new_outgoing;
}
return absl::nullopt;
}
} // namespace webrtc