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
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
|
/*
* Copyright (c) 2022 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/audio_coding/neteq/packet_arrival_history.h"
#include <algorithm>
#include <cstdint>
#include "api/neteq/tick_timer.h"
#include "rtc_base/checks.h"
namespace webrtc {
PacketArrivalHistory::PacketArrivalHistory(const TickTimer* tick_timer,
int window_size_ms)
: tick_timer_(tick_timer), window_size_ms_(window_size_ms) {}
bool PacketArrivalHistory::Insert(uint32_t rtp_timestamp,
int packet_length_samples) {
int64_t arrival_timestamp =
tick_timer_->ticks() * tick_timer_->ms_per_tick() * sample_rate_khz_;
PacketArrival packet(timestamp_unwrapper_.Unwrap(rtp_timestamp),
arrival_timestamp, packet_length_samples);
if (IsObsolete(packet)) {
return false;
}
if (Contains(packet)) {
return false;
}
history_.emplace(packet.rtp_timestamp, packet);
if (packet != history_.rbegin()->second) {
// Packet was reordered.
return true;
}
// Remove old packets.
while (IsObsolete(history_.begin()->second)) {
if (history_.begin()->second == min_packet_arrivals_.front()) {
min_packet_arrivals_.pop_front();
}
if (history_.begin()->second == max_packet_arrivals_.front()) {
max_packet_arrivals_.pop_front();
}
history_.erase(history_.begin());
}
// Ensure ordering constraints.
while (!min_packet_arrivals_.empty() &&
packet <= min_packet_arrivals_.back()) {
min_packet_arrivals_.pop_back();
}
while (!max_packet_arrivals_.empty() &&
packet >= max_packet_arrivals_.back()) {
max_packet_arrivals_.pop_back();
}
min_packet_arrivals_.push_back(packet);
max_packet_arrivals_.push_back(packet);
return true;
}
void PacketArrivalHistory::Reset() {
history_.clear();
min_packet_arrivals_.clear();
max_packet_arrivals_.clear();
timestamp_unwrapper_.Reset();
}
int PacketArrivalHistory::GetDelayMs(uint32_t rtp_timestamp) const {
int64_t unwrapped_rtp_timestamp =
timestamp_unwrapper_.PeekUnwrap(rtp_timestamp);
int64_t current_timestamp =
tick_timer_->ticks() * tick_timer_->ms_per_tick() * sample_rate_khz_;
PacketArrival packet(unwrapped_rtp_timestamp, current_timestamp,
/*duration_ms=*/0);
return GetPacketArrivalDelayMs(packet);
}
int PacketArrivalHistory::GetMaxDelayMs() const {
if (max_packet_arrivals_.empty()) {
return 0;
}
return GetPacketArrivalDelayMs(max_packet_arrivals_.front());
}
bool PacketArrivalHistory::IsNewestRtpTimestamp(uint32_t rtp_timestamp) const {
if (history_.empty()) {
return true;
}
int64_t unwrapped_rtp_timestamp =
timestamp_unwrapper_.PeekUnwrap(rtp_timestamp);
return unwrapped_rtp_timestamp == history_.rbegin()->second.rtp_timestamp;
}
int PacketArrivalHistory::GetPacketArrivalDelayMs(
const PacketArrival& packet_arrival) const {
if (min_packet_arrivals_.empty()) {
return 0;
}
RTC_DCHECK_NE(sample_rate_khz_, 0);
// TODO(jakobi): Timestamps are first converted to millis for bit-exactness.
return std::max<int>(
packet_arrival.arrival_timestamp / sample_rate_khz_ -
min_packet_arrivals_.front().arrival_timestamp / sample_rate_khz_ -
(packet_arrival.rtp_timestamp / sample_rate_khz_ -
min_packet_arrivals_.front().rtp_timestamp / sample_rate_khz_),
0);
}
bool PacketArrivalHistory::IsObsolete(
const PacketArrival& packet_arrival) const {
if (history_.empty()) {
return false;
}
return packet_arrival.rtp_timestamp + window_size_ms_ * sample_rate_khz_ <
history_.rbegin()->second.rtp_timestamp;
}
bool PacketArrivalHistory::Contains(const PacketArrival& packet_arrival) const {
auto it = history_.upper_bound(packet_arrival.rtp_timestamp);
if (it == history_.begin()) {
return false;
}
--it;
return it->second.contains(packet_arrival);
}
} // namespace webrtc
|
