mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-18 16:17:50 +01:00
ea86bb74fc
This reverts commit65792c5a5c. Reason for revert: <INSERT REASONING HERE> Original change's description: > Revert "Revert "Reland "Moved congestion controller to task queue.""" > > This reverts commit4e849f6925. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Reland "Moved congestion controller to task queue."" > > > > This reverts commit57daeb7ac7. > > > > Reason for revert: Cause increased congestion and deadlocks in downstream project > > > > Original change's description: > > > Reland "Moved congestion controller to task queue." > > > > > > This is a reland of0cbcba7ea0. > > > > > > Original change's description: > > > > Moved congestion controller to task queue. > > > > > > > > The goal of this work is to make it easier to experiment with the > > > > bandwidth estimation implementation. For this reason network control > > > > functionality is moved from SendSideCongestionController(SSCC), > > > > PacedSender and BitrateController to the newly created > > > > GoogCcNetworkController which implements the newly created > > > > NetworkControllerInterface. This allows the implementation to be > > > > replaced at runtime in the future. > > > > > > > > This is the first part of a split of a larger CL, see: > > > > https://webrtc-review.googlesource.com/c/src/+/39788/8 > > > > For further explanations. > > > > > > > > Bug: webrtc:8415 > > > > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3 > > > > Reviewed-on: https://webrtc-review.googlesource.com/43840 > > > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > > > Reviewed-by: Björn Terelius <terelius@webrtc.org> > > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#21868} > > > > > > Bug: webrtc:8415 > > > Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da > > > Reviewed-on: https://webrtc-review.googlesource.com/48000 > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21899} > > > > TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:8415 > > Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83 > > Reviewed-on: https://webrtc-review.googlesource.com/52980 > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22017} > > TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > Change-Id: I3393b74370c4f4d0955f50728005b2b925be169b > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8415 > Reviewed-on: https://webrtc-review.googlesource.com/53262 > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22023} TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org Change-Id: Id68ad986ee51142b7be3381d0793709b4392fe2c No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8415 Reviewed-on: https://webrtc-review.googlesource.com/53360 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22024}
65 lines
2.2 KiB
C++
65 lines
2.2 KiB
C++
/*
|
|
* Copyright (c) 2017 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/acknowledged_bitrate_estimator.h"
|
|
|
|
#include <utility>
|
|
|
|
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
|
#include "rtc_base/numerics/safe_conversions.h"
|
|
#include "rtc_base/ptr_util.h"
|
|
|
|
namespace webrtc {
|
|
|
|
namespace {
|
|
bool IsInSendTimeHistory(const PacketFeedback& packet) {
|
|
return packet.send_time_ms != PacketFeedback::kNoSendTime;
|
|
}
|
|
} // namespace
|
|
|
|
AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator()
|
|
: AcknowledgedBitrateEstimator(rtc::MakeUnique<BitrateEstimator>()) {}
|
|
|
|
AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator(
|
|
std::unique_ptr<BitrateEstimator> bitrate_estimator)
|
|
: bitrate_estimator_(std::move(bitrate_estimator)) {}
|
|
|
|
void AcknowledgedBitrateEstimator::IncomingPacketFeedbackVector(
|
|
const std::vector<PacketFeedback>& packet_feedback_vector) {
|
|
RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
|
|
packet_feedback_vector.end(),
|
|
PacketFeedbackComparator()));
|
|
for (const auto& packet : packet_feedback_vector) {
|
|
if (IsInSendTimeHistory(packet)) {
|
|
MaybeExpectFastRateChange(packet.send_time_ms);
|
|
bitrate_estimator_->Update(packet.arrival_time_ms,
|
|
rtc::dchecked_cast<int>(packet.payload_size));
|
|
}
|
|
}
|
|
}
|
|
|
|
rtc::Optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
|
|
return bitrate_estimator_->bitrate_bps();
|
|
}
|
|
|
|
void AcknowledgedBitrateEstimator::SetAlrEndedTimeMs(
|
|
int64_t alr_ended_time_ms) {
|
|
alr_ended_time_ms_.emplace(alr_ended_time_ms);
|
|
}
|
|
|
|
void AcknowledgedBitrateEstimator::MaybeExpectFastRateChange(
|
|
int64_t packet_send_time_ms) {
|
|
if (alr_ended_time_ms_ && packet_send_time_ms > *alr_ended_time_ms_) {
|
|
bitrate_estimator_->ExpectFastRateChange();
|
|
alr_ended_time_ms_.reset();
|
|
}
|
|
}
|
|
|
|
} // namespace webrtc
|