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
|
/*
* 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.
*/
#ifndef MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_
#define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_
#include "api/array_view.h"
#include "api/rtp_headers.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/ntp_time.h"
namespace webrtc {
//
// Helper class for sending the `AbsoluteCaptureTime` header extension.
//
// Supports the "timestamp interpolation" optimization:
// A sender SHOULD save bandwidth by not sending abs-capture-time with every
// RTP packet. It SHOULD still send them at regular intervals (e.g. every
// second) to help mitigate the impact of clock drift and packet loss. Mixers
// SHOULD always send abs-capture-time with the first RTP packet after
// changing capture system.
//
// Timestamp interpolation works fine as long as there’s reasonably low
// NTP/RTP clock drift. This is not always true. Senders that detect “jumps”
// between its NTP and RTP clock mappings SHOULD send abs-capture-time with
// the first RTP packet after such a thing happening.
//
// See: https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
//
class AbsoluteCaptureTimeSender {
public:
static constexpr TimeDelta kInterpolationMaxInterval = TimeDelta::Seconds(1);
static constexpr TimeDelta kInterpolationMaxError = TimeDelta::Millis(1);
explicit AbsoluteCaptureTimeSender(Clock* clock);
// Returns the source (i.e. SSRC or CSRC) of the capture system.
static uint32_t GetSource(uint32_t ssrc,
rtc::ArrayView<const uint32_t> csrcs);
// Returns value to write into AbsoluteCaptureTime RTP header extension to be
// sent, or `absl::nullopt` if the header extension shouldn't be attached to
// the outgoing packet.
//
// - `source` - id of the capture system.
// - `rtp_timestamp` - capture time represented as rtp timestamp in the
// outgoing packet
// - `rtp_clock_frequency_hz` - description of the `rtp_timestamp` units -
// `rtp_timetamp` delta of `rtp_clock_freqnecy_hz` represents 1 second.
// - `absolute_capture_time` - time when a frame was captured by the capture
// system.
// - `estimated_capture_clock_offset` - estimated offset between capture
// system clock and local `clock` passed as the AbsoluteCaptureTimeSender
// construction paramter. Uses the same units as `absolute_capture_time`,
// i.e. delta of 2^32 represents 1 second. See AbsoluteCaptureTime type
// comments for more details.
// - `force` - when set to true, OnSendPacket is forced to return non-nullopt.
absl::optional<AbsoluteCaptureTime> OnSendPacket(
uint32_t source,
uint32_t rtp_timestamp,
int rtp_clock_frequency_hz,
NtpTime absolute_capture_time,
absl::optional<int64_t> estimated_capture_clock_offset,
bool force = false);
// Returns a header extension to be sent, or `absl::nullopt` if the header
// extension shouldn't be sent.
[[deprecated]] absl::optional<AbsoluteCaptureTime> 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);
private:
bool ShouldSendExtension(
Timestamp send_time,
uint32_t source,
uint32_t rtp_timestamp,
int rtp_clock_frequency_hz,
NtpTime absolute_capture_time,
absl::optional<int64_t> estimated_capture_clock_offset) const;
Clock* const clock_;
Timestamp last_send_time_ = Timestamp::MinusInfinity();
uint32_t last_source_;
uint32_t last_rtp_timestamp_;
int last_rtp_clock_frequency_hz_;
NtpTime last_absolute_capture_time_;
absl::optional<int64_t> last_estimated_capture_clock_offset_;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_
|
