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0040b66ad3
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script from modules with parameters
'pacing video_coding congestion_controller remote_bitrate_estimator':
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118
Reviewed-on: https://webrtc-review.googlesource.com/83900
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23640}
67 lines
2.3 KiB
C++
67 lines
2.3 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
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#include <utility>
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/ptr_util.h"
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namespace webrtc {
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namespace {
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bool IsInSendTimeHistory(const PacketFeedback& packet) {
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return packet.send_time_ms != PacketFeedback::kNoSendTime;
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}
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} // namespace
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AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator()
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: AcknowledgedBitrateEstimator(rtc::MakeUnique<BitrateEstimator>()) {}
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AcknowledgedBitrateEstimator::~AcknowledgedBitrateEstimator() {}
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AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator(
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std::unique_ptr<BitrateEstimator> bitrate_estimator)
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: bitrate_estimator_(std::move(bitrate_estimator)) {}
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void AcknowledgedBitrateEstimator::IncomingPacketFeedbackVector(
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const std::vector<PacketFeedback>& packet_feedback_vector) {
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RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
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packet_feedback_vector.end(),
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PacketFeedbackComparator()));
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for (const auto& packet : packet_feedback_vector) {
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if (IsInSendTimeHistory(packet)) {
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MaybeExpectFastRateChange(packet.send_time_ms);
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bitrate_estimator_->Update(packet.arrival_time_ms,
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rtc::dchecked_cast<int>(packet.payload_size));
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}
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}
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}
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absl::optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
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return bitrate_estimator_->bitrate_bps();
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}
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void AcknowledgedBitrateEstimator::SetAlrEndedTimeMs(
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int64_t alr_ended_time_ms) {
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alr_ended_time_ms_.emplace(alr_ended_time_ms);
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}
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void AcknowledgedBitrateEstimator::MaybeExpectFastRateChange(
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int64_t packet_send_time_ms) {
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if (alr_ended_time_ms_ && packet_send_time_ms > *alr_ended_time_ms_) {
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bitrate_estimator_->ExpectFastRateChange();
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alr_ended_time_ms_.reset();
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}
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}
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} // namespace webrtc
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