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/*
* 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 {
constexpr int64_t SourceTracker::kTimeoutMs;
SourceTracker::SourceTracker(Clock* clock) : clock_(clock) {}
void SourceTracker::OnFrameDelivered(const RtpPacketInfos& packet_infos) {
if (packet_infos.empty()) {
return;
}
TRACE_EVENT0("webrtc", "SourceTracker::OnFrameDelivered");
int64_t now_ms = clock_->TimeInMilliseconds();
MutexLock lock_scope(&lock_);
for (const RtpPacketInfo& packet_info : packet_infos) {
for (uint32_t csrc : packet_info.csrcs()) {
SourceKey key(RtpSourceType::CSRC, csrc);
SourceEntry& entry = UpdateEntry(key);
const auto packet_time = packet_info.receive_time().ms();
entry.timestamp_ms = packet_time ? packet_time : now_ms;
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_ms = now_ms;
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(now_ms);
}
std::vector<RtpSource> SourceTracker::GetSources() const {
std::vector<RtpSource> sources;
int64_t now_ms = clock_->TimeInMilliseconds();
MutexLock lock_scope(&lock_);
PruneEntries(now_ms);
for (const auto& pair : list_) {
const SourceKey& key = pair.first;
const SourceEntry& entry = pair.second;
sources.emplace_back(
entry.timestamp_ms, 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});
}
std::sort(sources.begin(), sources.end(), [](const auto &a, const auto &b){
return a.timestamp_ms() > b.timestamp_ms();
});
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(int64_t now_ms) const {
int64_t prune_ms = now_ms - kTimeoutMs;
while (!list_.empty() && list_.back().second.timestamp_ms < prune_ms) {
map_.erase(list_.back().first);
list_.pop_back();
}
}
} // namespace webrtc
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