| /* |
| * Copyright (c) 2019 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/rtp_rtcp/source/source_tracker.h" |
| |
| #include <algorithm> |
| #include <utility> |
| |
| #include "rtc_base/trace_event.h" |
| |
| namespace webrtc { |
| |
| SourceTracker::SourceTracker(Clock* clock) : clock_(clock) { |
| RTC_DCHECK(clock_); |
| } |
| |
| void SourceTracker::OnFrameDelivered(const RtpPacketInfos& packet_infos, |
| Timestamp delivery_time) { |
| TRACE_EVENT0("webrtc", "SourceTracker::OnFrameDelivered"); |
| if (packet_infos.empty()) { |
| return; |
| } |
| if (delivery_time.IsInfinite()) { |
| delivery_time = clock_->CurrentTime(); |
| } |
| |
| for (const RtpPacketInfo& packet_info : packet_infos) { |
| for (uint32_t csrc : packet_info.csrcs()) { |
| SourceKey key(RtpSourceType::CSRC, csrc); |
| SourceEntry& entry = UpdateEntry(key); |
| |
| entry.timestamp = delivery_time; |
| entry.audio_level = packet_info.audio_level(); |
| entry.absolute_capture_time = packet_info.absolute_capture_time(); |
| entry.local_capture_clock_offset = |
| packet_info.local_capture_clock_offset(); |
| entry.rtp_timestamp = packet_info.rtp_timestamp(); |
| } |
| |
| SourceKey key(RtpSourceType::SSRC, packet_info.ssrc()); |
| SourceEntry& entry = UpdateEntry(key); |
| |
| entry.timestamp = delivery_time; |
| entry.audio_level = packet_info.audio_level(); |
| entry.absolute_capture_time = packet_info.absolute_capture_time(); |
| entry.local_capture_clock_offset = packet_info.local_capture_clock_offset(); |
| entry.rtp_timestamp = packet_info.rtp_timestamp(); |
| } |
| |
| PruneEntries(delivery_time); |
| } |
| |
| std::vector<RtpSource> SourceTracker::GetSources() const { |
| PruneEntries(clock_->CurrentTime()); |
| |
| std::vector<RtpSource> sources; |
| for (const auto& pair : list_) { |
| const SourceKey& key = pair.first; |
| const SourceEntry& entry = pair.second; |
| |
| sources.emplace_back( |
| entry.timestamp, key.source, key.source_type, entry.rtp_timestamp, |
| RtpSource::Extensions{ |
| .audio_level = entry.audio_level, |
| .absolute_capture_time = entry.absolute_capture_time, |
| .local_capture_clock_offset = entry.local_capture_clock_offset}); |
| } |
| |
| return sources; |
| } |
| |
| SourceTracker::SourceEntry& SourceTracker::UpdateEntry(const SourceKey& key) { |
| // We intentionally do |find() + emplace()|, instead of checking the return |
| // value of `emplace()`, for performance reasons. It's much more likely for |
| // the key to already exist than for it not to. |
| auto map_it = map_.find(key); |
| if (map_it == map_.end()) { |
| // Insert a new entry at the front of the list. |
| list_.emplace_front(key, SourceEntry()); |
| map_.emplace(key, list_.begin()); |
| } else if (map_it->second != list_.begin()) { |
| // Move the old entry to the front of the list. |
| list_.splice(list_.begin(), list_, map_it->second); |
| } |
| |
| return list_.front().second; |
| } |
| |
| void SourceTracker::PruneEntries(Timestamp now) const { |
| if (now < Timestamp::Zero() + kTimeout) { |
| return; |
| } |
| Timestamp prune = now - kTimeout; |
| while (!list_.empty() && list_.back().second.timestamp < prune) { |
| map_.erase(list_.back().first); |
| list_.pop_back(); |
| } |
| } |
| |
| } // namespace webrtc |