mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 05:40:42 +01:00

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}
73 lines
2.5 KiB
C++
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
|