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
|
/*
* Copyright (c) 2018 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 CALL_CALL_CONFIG_H_
#define CALL_CALL_CONFIG_H_
#include <memory>
#include <optional>
#include "api/environment/environment.h"
#include "api/fec_controller.h"
#include "api/metronome/metronome.h"
#include "api/neteq/neteq_factory.h"
#include "api/network_state_predictor.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/task_queue_base.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/network_control.h"
#include "api/units/time_delta.h"
#include "call/audio_state.h"
#include "call/rtp_transport_config.h"
#include "call/rtp_transport_controller_send_factory_interface.h"
namespace webrtc {
class AudioProcessing;
struct CallConfig {
// If `network_task_queue` is set to nullptr, Call will assume that network
// related callbacks will be made on the same TQ as the Call instance was
// constructed on.
explicit CallConfig(const Environment& env,
TaskQueueBase* network_task_queue = nullptr);
// Move-only.
CallConfig(CallConfig&&) = default;
CallConfig& operator=(CallConfig&& other) = default;
~CallConfig();
RtpTransportConfig ExtractTransportConfig() const;
Environment env;
// Bitrate config used until valid bitrate estimates are calculated. Also
// used to cap total bitrate used. This comes from the remote connection.
BitrateConstraints bitrate_config;
// AudioState which is possibly shared between multiple calls.
scoped_refptr<AudioState> audio_state;
// Audio Processing Module to be used in this call.
AudioProcessing* audio_processing = nullptr;
// FecController to use for this call.
FecControllerFactoryInterface* fec_controller_factory = nullptr;
// NetworkStatePredictor to use for this call.
NetworkStatePredictorFactoryInterface* network_state_predictor_factory =
nullptr;
// Call-specific Network controller factory to use. If this is set, it
// takes precedence over network_controller_factory.
std::unique_ptr<NetworkControllerFactoryInterface>
per_call_network_controller_factory;
// Network controller factory to use for this call if
// per_call_network_controller_factory is null.
NetworkControllerFactoryInterface* network_controller_factory = nullptr;
// NetEq factory to use for this call.
NetEqFactory* neteq_factory = nullptr;
TaskQueueBase* network_task_queue_ = nullptr;
// RtpTransportControllerSend to use for this call.
RtpTransportControllerSendFactoryInterface*
rtp_transport_controller_send_factory = nullptr;
Metronome* decode_metronome = nullptr;
Metronome* encode_metronome = nullptr;
// The burst interval of the pacer, see TaskQueuePacedSender constructor.
std::optional<TimeDelta> pacer_burst_interval;
// Enables send packet batching from the egress RTP sender.
bool enable_send_packet_batching = false;
};
} // namespace webrtc
#endif // CALL_CALL_CONFIG_H_
|
