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
|
/*
* Copyright (c) 2015 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 "webrtc/video_engine/payload_router.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
PayloadRouter::PayloadRouter()
: crit_(CriticalSectionWrapper::CreateCriticalSection()),
active_(false) {}
PayloadRouter::~PayloadRouter() {}
size_t PayloadRouter::DefaultMaxPayloadLength() {
const size_t kIpUdpSrtpLength = 44;
return IP_PACKET_SIZE - kIpUdpSrtpLength;
}
void PayloadRouter::SetSendingRtpModules(
const std::list<RtpRtcp*>& rtp_modules) {
CriticalSectionScoped cs(crit_.get());
rtp_modules_.clear();
rtp_modules_.reserve(rtp_modules.size());
for (auto* rtp_module : rtp_modules) {
rtp_modules_.push_back(rtp_module);
}
}
void PayloadRouter::set_active(bool active) {
CriticalSectionScoped cs(crit_.get());
active_ = active;
}
bool PayloadRouter::active() {
CriticalSectionScoped cs(crit_.get());
return active_ && !rtp_modules_.empty();
}
bool PayloadRouter::RoutePayload(FrameType frame_type,
int8_t payload_type,
uint32_t time_stamp,
int64_t capture_time_ms,
const uint8_t* payload_data,
size_t payload_length,
const RTPFragmentationHeader* fragmentation,
const RTPVideoHeader* rtp_video_hdr) {
CriticalSectionScoped cs(crit_.get());
if (!active_ || rtp_modules_.empty())
return false;
// The simulcast index might actually be larger than the number of modules in
// case the encoder was processing a frame during a codec reconfig.
if (rtp_video_hdr != NULL &&
rtp_video_hdr->simulcastIdx >= rtp_modules_.size())
return false;
int stream_idx = 0;
if (rtp_video_hdr != NULL)
stream_idx = rtp_video_hdr->simulcastIdx;
return rtp_modules_[stream_idx]->SendOutgoingData(
frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
payload_length, fragmentation, rtp_video_hdr) == 0 ? true : false;
}
void PayloadRouter::SetTargetSendBitrates(
const std::vector<uint32_t>& stream_bitrates) {
CriticalSectionScoped cs(crit_.get());
if (stream_bitrates.size() < rtp_modules_.size()) {
// There can be a size mis-match during codec reconfiguration.
return;
}
int idx = 0;
for (auto* rtp_module : rtp_modules_) {
rtp_module->SetTargetSendBitrate(stream_bitrates[idx++]);
}
}
size_t PayloadRouter::MaxPayloadLength() const {
size_t min_payload_length = DefaultMaxPayloadLength();
CriticalSectionScoped cs(crit_.get());
for (auto* rtp_module : rtp_modules_) {
size_t module_payload_length = rtp_module->MaxDataPayloadLength();
if (module_payload_length < min_payload_length)
min_payload_length = module_payload_length;
}
return min_payload_length;
}
} // namespace webrtc
|
