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
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
|
/*
* Copyright (c) 2012 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_RTP_SENDER_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
#include <cstddef>
#include <cstdint>
#include <map>
#include <memory>
#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "api/environment/environment.h"
#include "api/rtp_packet_sender.h"
#include "modules/rtp_rtcp/include/flexfec_sender.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_header_extension_size.h"
#include "modules/rtp_rtcp/source/rtp_packet_history.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "rtc_base/random.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class FrameEncryptorInterface;
class RateLimiter;
class RtpPacketToSend;
// Maximum amount of padding in RFC 3550 is 255 bytes.
constexpr size_t kMaxPaddingLength = 255;
class RTPSender {
public:
RTPSender(const Environment& env,
const RtpRtcpInterface::Configuration& config,
RtpPacketHistory* packet_history,
RtpPacketSender* packet_sender);
RTPSender(const RTPSender&) = delete;
RTPSender& operator=(const RTPSender&) = delete;
~RTPSender();
void SetSendingMediaStatus(bool enabled) RTC_LOCKS_EXCLUDED(send_mutex_);
bool SendingMedia() const RTC_LOCKS_EXCLUDED(send_mutex_);
bool IsAudioConfigured() const RTC_LOCKS_EXCLUDED(send_mutex_);
uint32_t TimestampOffset() const RTC_LOCKS_EXCLUDED(send_mutex_);
void SetTimestampOffset(uint32_t timestamp) RTC_LOCKS_EXCLUDED(send_mutex_);
void SetMid(absl::string_view mid) RTC_LOCKS_EXCLUDED(send_mutex_);
uint16_t SequenceNumber() const RTC_LOCKS_EXCLUDED(send_mutex_);
void SetSequenceNumber(uint16_t seq) RTC_LOCKS_EXCLUDED(send_mutex_);
void SetMaxRtpPacketSize(size_t max_packet_size)
RTC_LOCKS_EXCLUDED(send_mutex_);
void SetExtmapAllowMixed(bool extmap_allow_mixed)
RTC_LOCKS_EXCLUDED(send_mutex_);
// RTP header extension
bool RegisterRtpHeaderExtension(absl::string_view uri, int id)
RTC_LOCKS_EXCLUDED(send_mutex_);
bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) const
RTC_LOCKS_EXCLUDED(send_mutex_);
void DeregisterRtpHeaderExtension(absl::string_view uri)
RTC_LOCKS_EXCLUDED(send_mutex_);
bool SupportsPadding() const RTC_LOCKS_EXCLUDED(send_mutex_);
bool SupportsRtxPayloadPadding() const RTC_LOCKS_EXCLUDED(send_mutex_);
std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
size_t target_size_bytes,
bool media_has_been_sent,
bool can_send_padding_on_media_ssrc) RTC_LOCKS_EXCLUDED(send_mutex_);
// NACK.
void OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers,
int64_t avg_rtt) RTC_LOCKS_EXCLUDED(send_mutex_);
int32_t ReSendPacket(uint16_t packet_id) RTC_LOCKS_EXCLUDED(send_mutex_);
// ACK.
void OnReceivedAckOnSsrc(int64_t extended_highest_sequence_number)
RTC_LOCKS_EXCLUDED(send_mutex_);
void OnReceivedAckOnRtxSsrc(int64_t extended_highest_sequence_number)
RTC_LOCKS_EXCLUDED(send_mutex_);
// RTX.
void SetRtxStatus(int mode) RTC_LOCKS_EXCLUDED(send_mutex_);
int RtxStatus() const RTC_LOCKS_EXCLUDED(send_mutex_);
std::optional<uint32_t> RtxSsrc() const RTC_LOCKS_EXCLUDED(send_mutex_) {
return rtx_ssrc_;
}
// Returns expected size difference between an RTX packet and media packet
// that RTX packet is created from. Returns 0 if RTX is disabled.
size_t RtxPacketOverhead() const;
void SetRtxPayloadType(int payload_type, int associated_payload_type)
RTC_LOCKS_EXCLUDED(send_mutex_);
// Size info for header extensions used by FEC packets.
static ArrayView<const RtpExtensionSize> FecExtensionSizes()
RTC_LOCKS_EXCLUDED(send_mutex_);
// Size info for header extensions used by video packets.
static ArrayView<const RtpExtensionSize> VideoExtensionSizes()
RTC_LOCKS_EXCLUDED(send_mutex_);
// Size info for header extensions used by audio packets.
static ArrayView<const RtpExtensionSize> AudioExtensionSizes()
RTC_LOCKS_EXCLUDED(send_mutex_);
// Create empty packet, fills ssrc, csrcs and reserve place for header
// extensions RtpSender updates before sending.
std::unique_ptr<RtpPacketToSend> AllocatePacket(
ArrayView<const uint32_t> csrcs = {}) RTC_LOCKS_EXCLUDED(send_mutex_);
// Maximum header overhead per fec/padding packet.
size_t FecOrPaddingPacketMaxRtpHeaderLength() const
RTC_LOCKS_EXCLUDED(send_mutex_);
// Expected header overhead per media packet.
size_t ExpectedPerPacketOverhead() const RTC_LOCKS_EXCLUDED(send_mutex_);
// Including RTP headers.
size_t MaxRtpPacketSize() const RTC_LOCKS_EXCLUDED(send_mutex_);
uint32_t SSRC() const RTC_LOCKS_EXCLUDED(send_mutex_) { return ssrc_; }
std::optional<uint32_t> FlexfecSsrc() const RTC_LOCKS_EXCLUDED(send_mutex_) {
return flexfec_ssrc_;
}
// Pass a set of packets to RtpPacketSender instance, for paced or immediate
// sending to the network.
void EnqueuePackets(std::vector<std::unique_ptr<RtpPacketToSend>> packets)
RTC_LOCKS_EXCLUDED(send_mutex_);
void SetRtpState(const RtpState& rtp_state) RTC_LOCKS_EXCLUDED(send_mutex_);
RtpState GetRtpState() const RTC_LOCKS_EXCLUDED(send_mutex_);
void SetRtxRtpState(const RtpState& rtp_state)
RTC_LOCKS_EXCLUDED(send_mutex_);
RtpState GetRtxRtpState() const RTC_LOCKS_EXCLUDED(send_mutex_);
private:
std::unique_ptr<RtpPacketToSend> BuildRtxPacket(
const RtpPacketToSend& packet);
bool IsFecPacket(const RtpPacketToSend& packet) const;
void UpdateHeaderSizes() RTC_EXCLUSIVE_LOCKS_REQUIRED(send_mutex_);
void UpdateLastPacketState(const RtpPacketToSend& packet)
RTC_EXCLUSIVE_LOCKS_REQUIRED(send_mutex_);
Clock* const clock_;
Random random_ RTC_GUARDED_BY(send_mutex_);
const bool audio_configured_;
const uint32_t ssrc_;
const std::optional<uint32_t> rtx_ssrc_;
const std::optional<uint32_t> flexfec_ssrc_;
RtpPacketHistory* const packet_history_;
RtpPacketSender* const paced_sender_;
mutable Mutex send_mutex_;
bool sending_media_ RTC_GUARDED_BY(send_mutex_);
size_t max_packet_size_;
RtpHeaderExtensionMap rtp_header_extension_map_ RTC_GUARDED_BY(send_mutex_);
size_t max_media_packet_header_ RTC_GUARDED_BY(send_mutex_);
size_t max_padding_fec_packet_header_ RTC_GUARDED_BY(send_mutex_);
// RTP variables
uint32_t timestamp_offset_ RTC_GUARDED_BY(send_mutex_);
// RID value to send in the RID or RepairedRID header extension.
const std::string rid_;
// MID value to send in the MID header extension.
std::string mid_ RTC_GUARDED_BY(send_mutex_);
// Should we send MID/RID even when ACKed? (see below).
const bool always_send_mid_and_rid_;
// Track if any ACK has been received on the SSRC and RTX SSRC to indicate
// when to stop sending the MID and RID header extensions.
bool ssrc_has_acked_ RTC_GUARDED_BY(send_mutex_);
bool rtx_ssrc_has_acked_ RTC_GUARDED_BY(send_mutex_);
// Maximum number of csrcs this sender is used with.
size_t max_num_csrcs_ RTC_GUARDED_BY(send_mutex_) = 0;
int rtx_ RTC_GUARDED_BY(send_mutex_);
// Mapping rtx_payload_type_map_[associated] = rtx.
std::map<int8_t, int8_t> rtx_payload_type_map_ RTC_GUARDED_BY(send_mutex_);
bool supports_bwe_extension_ RTC_GUARDED_BY(send_mutex_);
RateLimiter* const retransmission_rate_limiter_;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
|
