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6255a7f3a0
Bug: b/364184684 Change-Id: If03cd697fed05c24549b9ef80bbaf9f11b47d8bc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361640 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42959}
107 lines
3.4 KiB
C++
107 lines
3.4 KiB
C++
/*
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* Copyright (c) 2019 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/rtp_rtcp/source/source_tracker.h"
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#include <algorithm>
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#include <utility>
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#include "rtc_base/trace_event.h"
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namespace webrtc {
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SourceTracker::SourceTracker(Clock* clock) : clock_(clock) {
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RTC_DCHECK(clock_);
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}
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void SourceTracker::OnFrameDelivered(const RtpPacketInfos& packet_infos,
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Timestamp delivery_time) {
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TRACE_EVENT0("webrtc", "SourceTracker::OnFrameDelivered");
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if (packet_infos.empty()) {
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return;
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}
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if (delivery_time.IsInfinite()) {
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delivery_time = clock_->CurrentTime();
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}
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for (const RtpPacketInfo& packet_info : packet_infos) {
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for (uint32_t csrc : packet_info.csrcs()) {
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SourceKey key(RtpSourceType::CSRC, csrc);
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SourceEntry& entry = UpdateEntry(key);
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entry.timestamp = delivery_time;
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entry.audio_level = packet_info.audio_level();
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entry.absolute_capture_time = packet_info.absolute_capture_time();
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entry.local_capture_clock_offset =
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packet_info.local_capture_clock_offset();
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entry.rtp_timestamp = packet_info.rtp_timestamp();
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}
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SourceKey key(RtpSourceType::SSRC, packet_info.ssrc());
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SourceEntry& entry = UpdateEntry(key);
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entry.timestamp = delivery_time;
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entry.audio_level = packet_info.audio_level();
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entry.absolute_capture_time = packet_info.absolute_capture_time();
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entry.local_capture_clock_offset = packet_info.local_capture_clock_offset();
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entry.rtp_timestamp = packet_info.rtp_timestamp();
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}
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PruneEntries(delivery_time);
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}
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std::vector<RtpSource> SourceTracker::GetSources() const {
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PruneEntries(clock_->CurrentTime());
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std::vector<RtpSource> sources;
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for (const auto& pair : list_) {
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const SourceKey& key = pair.first;
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const SourceEntry& entry = pair.second;
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sources.emplace_back(
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entry.timestamp, key.source, key.source_type, entry.rtp_timestamp,
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RtpSource::Extensions{
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.audio_level = entry.audio_level,
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.absolute_capture_time = entry.absolute_capture_time,
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.local_capture_clock_offset = entry.local_capture_clock_offset});
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}
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return sources;
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}
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SourceTracker::SourceEntry& SourceTracker::UpdateEntry(const SourceKey& key) {
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// We intentionally do |find() + emplace()|, instead of checking the return
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// value of `emplace()`, for performance reasons. It's much more likely for
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// the key to already exist than for it not to.
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auto map_it = map_.find(key);
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if (map_it == map_.end()) {
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// Insert a new entry at the front of the list.
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list_.emplace_front(key, SourceEntry());
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map_.emplace(key, list_.begin());
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} else if (map_it->second != list_.begin()) {
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// Move the old entry to the front of the list.
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list_.splice(list_.begin(), list_, map_it->second);
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}
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return list_.front().second;
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}
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void SourceTracker::PruneEntries(Timestamp now) const {
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if (now < Timestamp::Zero() + kTimeout) {
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return;
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}
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Timestamp prune = now - kTimeout;
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while (!list_.empty() && list_.back().second.timestamp < prune) {
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map_.erase(list_.back().first);
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list_.pop_back();
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}
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}
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} // namespace webrtc
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