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
|
/*
* 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/absolute_capture_time_sender.h"
#include <limits>
#include "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h"
#include "system_wrappers/include/ntp_time.h"
namespace webrtc {
namespace {
constexpr Timestamp kInvalidLastSendTime = Timestamp::MinusInfinity();
} // namespace
constexpr TimeDelta AbsoluteCaptureTimeSender::kInterpolationMaxInterval;
constexpr TimeDelta AbsoluteCaptureTimeSender::kInterpolationMaxError;
static_assert(
AbsoluteCaptureTimeInterpolator::kInterpolationMaxInterval >=
AbsoluteCaptureTimeSender::kInterpolationMaxInterval,
"Receivers should be as willing to interpolate timestamps as senders.");
AbsoluteCaptureTimeSender::AbsoluteCaptureTimeSender(Clock* clock)
: clock_(clock), last_send_time_(kInvalidLastSendTime) {}
uint32_t AbsoluteCaptureTimeSender::GetSource(
uint32_t ssrc,
rtc::ArrayView<const uint32_t> csrcs) {
return AbsoluteCaptureTimeInterpolator::GetSource(ssrc, csrcs);
}
absl::optional<AbsoluteCaptureTime> AbsoluteCaptureTimeSender::OnSendPacket(
uint32_t source,
uint32_t rtp_timestamp,
uint32_t rtp_clock_frequency,
uint64_t absolute_capture_timestamp,
absl::optional<int64_t> estimated_capture_clock_offset) {
const Timestamp send_time = clock_->CurrentTime();
MutexLock lock(&mutex_);
if (!ShouldSendExtension(send_time, source, rtp_timestamp,
rtp_clock_frequency, absolute_capture_timestamp,
estimated_capture_clock_offset)) {
return absl::nullopt;
}
last_source_ = source;
last_rtp_timestamp_ = rtp_timestamp;
last_rtp_clock_frequency_ = rtp_clock_frequency;
last_absolute_capture_timestamp_ = absolute_capture_timestamp;
last_estimated_capture_clock_offset_ = estimated_capture_clock_offset;
last_send_time_ = send_time;
AbsoluteCaptureTime extension;
extension.absolute_capture_timestamp = absolute_capture_timestamp;
extension.estimated_capture_clock_offset = estimated_capture_clock_offset;
return extension;
}
bool AbsoluteCaptureTimeSender::ShouldSendExtension(
Timestamp send_time,
uint32_t source,
uint32_t rtp_timestamp,
uint32_t rtp_clock_frequency,
uint64_t absolute_capture_timestamp,
absl::optional<int64_t> estimated_capture_clock_offset) const {
// Should if we've never sent anything before.
if (last_send_time_ == kInvalidLastSendTime) {
return true;
}
// Should if the last sent extension is too old.
if ((send_time - last_send_time_) > kInterpolationMaxInterval) {
return true;
}
// Should if the source has changed.
if (last_source_ != source) {
return true;
}
// Should if the RTP clock frequency has changed.
if (last_rtp_clock_frequency_ != rtp_clock_frequency) {
return true;
}
// Should if the RTP clock frequency is invalid.
if (rtp_clock_frequency <= 0) {
return true;
}
// Should if the estimated capture clock offset has changed.
if (last_estimated_capture_clock_offset_ != estimated_capture_clock_offset) {
return true;
}
// Should if interpolation would introduce too much error.
const uint64_t interpolated_absolute_capture_timestamp =
AbsoluteCaptureTimeInterpolator::InterpolateAbsoluteCaptureTimestamp(
rtp_timestamp, rtp_clock_frequency, last_rtp_timestamp_,
last_absolute_capture_timestamp_);
const int64_t interpolation_error_ms = UQ32x32ToInt64Ms(std::min(
interpolated_absolute_capture_timestamp - absolute_capture_timestamp,
absolute_capture_timestamp - interpolated_absolute_capture_timestamp));
if (interpolation_error_ms > kInterpolationMaxError.ms()) {
return true;
}
return false;
}
} // namespace webrtc
|
