1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
|
/*
* 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>
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;
}
int64_t now_ms = clock_->TimeInMilliseconds();
MutexLock lock_scope(&lock_);
for (const auto& 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.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.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{entry.audio_level, entry.absolute_capture_time});
}
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
|
