57daeb7ac7
This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9. 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}
104 lines
3.5 KiB
C++
104 lines
3.5 KiB
C++
/*
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* Copyright (c) 2015 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/send_time_history.h"
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#include <algorithm>
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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/checks.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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SendTimeHistory::SendTimeHistory(const Clock* clock,
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int64_t packet_age_limit_ms)
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: clock_(clock), packet_age_limit_ms_(packet_age_limit_ms) {}
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SendTimeHistory::~SendTimeHistory() {}
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void SendTimeHistory::AddAndRemoveOld(const PacketFeedback& packet) {
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int64_t now_ms = clock_->TimeInMilliseconds();
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// Remove old.
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while (!history_.empty() &&
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now_ms - history_.begin()->second.creation_time_ms >
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packet_age_limit_ms_) {
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// TODO(sprang): Warn if erasing (too many) old items?
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history_.erase(history_.begin());
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}
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// Add new.
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int64_t unwrapped_seq_num = seq_num_unwrapper_.Unwrap(packet.sequence_number);
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history_.insert(std::make_pair(unwrapped_seq_num, packet));
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}
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bool SendTimeHistory::OnSentPacket(uint16_t sequence_number,
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int64_t send_time_ms) {
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int64_t unwrapped_seq_num = seq_num_unwrapper_.Unwrap(sequence_number);
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auto it = history_.find(unwrapped_seq_num);
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if (it == history_.end())
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return false;
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it->second.send_time_ms = send_time_ms;
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return true;
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}
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rtc::Optional<PacketFeedback> SendTimeHistory::GetPacket(
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uint16_t sequence_number) const {
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int64_t unwrapped_seq_num =
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seq_num_unwrapper_.UnwrapWithoutUpdate(sequence_number);
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rtc::Optional<PacketFeedback> optional_feedback;
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auto it = history_.find(unwrapped_seq_num);
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if (it != history_.end())
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optional_feedback.emplace(it->second);
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return optional_feedback;
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}
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bool SendTimeHistory::GetFeedback(PacketFeedback* packet_feedback,
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bool remove) {
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RTC_DCHECK(packet_feedback);
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int64_t unwrapped_seq_num =
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seq_num_unwrapper_.Unwrap(packet_feedback->sequence_number);
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latest_acked_seq_num_.emplace(
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std::max(unwrapped_seq_num, latest_acked_seq_num_.value_or(0)));
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RTC_DCHECK_GE(*latest_acked_seq_num_, 0);
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auto it = history_.find(unwrapped_seq_num);
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if (it == history_.end())
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return false;
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// Save arrival_time not to overwrite it.
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int64_t arrival_time_ms = packet_feedback->arrival_time_ms;
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*packet_feedback = it->second;
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packet_feedback->arrival_time_ms = arrival_time_ms;
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if (remove)
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history_.erase(it);
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return true;
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}
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size_t SendTimeHistory::GetOutstandingBytes(uint16_t local_net_id,
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uint16_t remote_net_id) const {
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size_t outstanding_bytes = 0;
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auto unacked_it = history_.begin();
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if (latest_acked_seq_num_) {
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unacked_it = history_.lower_bound(*latest_acked_seq_num_);
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}
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for (; unacked_it != history_.end(); ++unacked_it) {
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if (unacked_it->second.local_net_id == local_net_id &&
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unacked_it->second.remote_net_id == remote_net_id &&
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unacked_it->second.send_time_ms >= 0) {
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outstanding_bytes += unacked_it->second.payload_size;
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}
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}
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return outstanding_bytes;
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}
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} // namespace webrtc
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