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
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
|
/*
* Copyright 2017 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 PC_RTP_TRANSPORT_INTERNAL_H_
#define PC_RTP_TRANSPORT_INTERNAL_H_
#include <cstdint>
#include <optional>
#include <string>
#include <utility>
#include "absl/functional/any_invocable.h"
#include "call/rtp_demuxer.h"
#include "pc/session_description.h"
#include "rtc_base/callback_list.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
#include "rtc_base/socket.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
namespace webrtc {
class CopyOnWriteBuffer;
// This class is an internal interface; it is not accessible to API consumers
// but is accessible to internal classes in order to send and receive RTP and
// RTCP packets belonging to a single RTP session. Additional convenience and
// configuration methods are also provided.
class RtpTransportInternal : public sigslot::has_slots<> {
public:
virtual ~RtpTransportInternal() = default;
virtual void SetRtcpMuxEnabled(bool enable) = 0;
virtual const std::string& transport_name() const = 0;
// Sets socket options on the underlying RTP or RTCP transports.
virtual int SetRtpOption(Socket::Option opt, int value) = 0;
virtual int SetRtcpOption(Socket::Option opt, int value) = 0;
virtual bool rtcp_mux_enabled() const = 0;
virtual bool IsReadyToSend() const = 0;
// Called whenever a transport's ready-to-send state changes. The argument
// is true if all used transports are ready to send. This is more specific
// than just "writable"; it means the last send didn't return ENOTCONN.
void SubscribeReadyToSend(const void* tag,
absl::AnyInvocable<void(bool)> callback) {
callback_list_ready_to_send_.AddReceiver(tag, std::move(callback));
}
void UnsubscribeReadyToSend(const void* tag) {
callback_list_ready_to_send_.RemoveReceivers(tag);
}
// Called whenever an RTCP packet is received. There is no equivalent signal
// for demuxable RTP packets because they would be forwarded to the
// BaseChannel through the RtpDemuxer callback.
void SubscribeRtcpPacketReceived(
const void* tag,
absl::AnyInvocable<void(webrtc::CopyOnWriteBuffer*, int64_t)> callback) {
callback_list_rtcp_packet_received_.AddReceiver(tag, std::move(callback));
}
// There doesn't seem to be a need to unsubscribe from this signal.
// Called whenever a RTP packet that can not be demuxed by the transport is
// received.
void SetUnDemuxableRtpPacketReceivedHandler(
absl::AnyInvocable<void(RtpPacketReceived&)> callback) {
callback_undemuxable_rtp_packet_received_ = std::move(callback);
}
// Called whenever the network route of the P2P layer transport changes.
// The argument is an optional network route.
void SubscribeNetworkRouteChanged(
const void* tag,
absl::AnyInvocable<void(std::optional<webrtc::NetworkRoute>)> callback) {
callback_list_network_route_changed_.AddReceiver(tag, std::move(callback));
}
void UnsubscribeNetworkRouteChanged(const void* tag) {
callback_list_network_route_changed_.RemoveReceivers(tag);
}
// Called whenever a transport's writable state might change. The argument is
// true if the transport is writable, otherwise it is false.
void SubscribeWritableState(const void* tag,
absl::AnyInvocable<void(bool)> callback) {
callback_list_writable_state_.AddReceiver(tag, std::move(callback));
}
void UnsubscribeWritableState(const void* tag) {
callback_list_writable_state_.RemoveReceivers(tag);
}
void SubscribeSentPacket(
const void* tag,
absl::AnyInvocable<void(const webrtc::SentPacketInfo&)> callback) {
callback_list_sent_packet_.AddReceiver(tag, std::move(callback));
}
void UnsubscribeSentPacket(const void* tag) {
callback_list_sent_packet_.RemoveReceivers(tag);
}
virtual bool IsWritable(bool rtcp) const = 0;
// TODO(zhihuang): Pass the `packet` by copy so that the original data
// wouldn't be modified.
virtual bool SendRtpPacket(CopyOnWriteBuffer* packet,
const AsyncSocketPacketOptions& options,
int flags) = 0;
virtual bool SendRtcpPacket(CopyOnWriteBuffer* packet,
const AsyncSocketPacketOptions& options,
int flags) = 0;
// This method updates the RTP header extension map so that the RTP transport
// can parse the received packets and identify the MID. This is called by the
// BaseChannel when setting the content description.
//
// TODO(zhihuang): Merging and replacing following methods handling header
// extensions with SetParameters:
// UpdateRtpHeaderExtensionMap,
// UpdateSendEncryptedHeaderExtensionIds,
// UpdateRecvEncryptedHeaderExtensionIds,
// CacheRtpAbsSendTimeHeaderExtension,
virtual void UpdateRtpHeaderExtensionMap(
const RtpHeaderExtensions& header_extensions) = 0;
virtual bool IsSrtpActive() const = 0;
virtual bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
RtpPacketSinkInterface* sink) = 0;
virtual bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) = 0;
protected:
void SendReadyToSend(bool arg) { callback_list_ready_to_send_.Send(arg); }
void SendRtcpPacketReceived(CopyOnWriteBuffer* buffer,
int64_t packet_time_us) {
callback_list_rtcp_packet_received_.Send(buffer, packet_time_us);
}
void NotifyUnDemuxableRtpPacketReceived(RtpPacketReceived& packet) {
callback_undemuxable_rtp_packet_received_(packet);
}
void SendNetworkRouteChanged(std::optional<NetworkRoute> route) {
callback_list_network_route_changed_.Send(route);
}
void SendWritableState(bool state) {
callback_list_writable_state_.Send(state);
}
void SendSentPacket(const SentPacketInfo& packet) {
callback_list_sent_packet_.Send(packet);
}
private:
CallbackList<bool> callback_list_ready_to_send_;
CallbackList<CopyOnWriteBuffer*, int64_t> callback_list_rtcp_packet_received_;
absl::AnyInvocable<void(RtpPacketReceived&)>
callback_undemuxable_rtp_packet_received_ =
[](RtpPacketReceived& packet) {};
CallbackList<std::optional<NetworkRoute>>
callback_list_network_route_changed_;
CallbackList<bool> callback_list_writable_state_;
CallbackList<const SentPacketInfo&> callback_list_sent_packet_;
};
} // namespace webrtc
#endif // PC_RTP_TRANSPORT_INTERNAL_H_
|
