1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "webrtc/video/payload_router.h"
12
13 #include "webrtc/base/checks.h"
14 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
15 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
16 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
17
18 namespace webrtc {
19
PayloadRouter()20 PayloadRouter::PayloadRouter()
21 : crit_(CriticalSectionWrapper::CreateCriticalSection()),
22 active_(false) {}
23
~PayloadRouter()24 PayloadRouter::~PayloadRouter() {}
25
DefaultMaxPayloadLength()26 size_t PayloadRouter::DefaultMaxPayloadLength() {
27 const size_t kIpUdpSrtpLength = 44;
28 return IP_PACKET_SIZE - kIpUdpSrtpLength;
29 }
30
SetSendingRtpModules(const std::list<RtpRtcp * > & rtp_modules)31 void PayloadRouter::SetSendingRtpModules(
32 const std::list<RtpRtcp*>& rtp_modules) {
33 CriticalSectionScoped cs(crit_.get());
34 rtp_modules_.clear();
35 rtp_modules_.reserve(rtp_modules.size());
36 for (auto* rtp_module : rtp_modules) {
37 rtp_modules_.push_back(rtp_module);
38 }
39 }
40
set_active(bool active)41 void PayloadRouter::set_active(bool active) {
42 CriticalSectionScoped cs(crit_.get());
43 active_ = active;
44 }
45
active()46 bool PayloadRouter::active() {
47 CriticalSectionScoped cs(crit_.get());
48 return active_ && !rtp_modules_.empty();
49 }
50
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)51 bool PayloadRouter::RoutePayload(FrameType frame_type,
52 int8_t payload_type,
53 uint32_t time_stamp,
54 int64_t capture_time_ms,
55 const uint8_t* payload_data,
56 size_t payload_length,
57 const RTPFragmentationHeader* fragmentation,
58 const RTPVideoHeader* rtp_video_hdr) {
59 CriticalSectionScoped cs(crit_.get());
60 if (!active_ || rtp_modules_.empty())
61 return false;
62
63 // The simulcast index might actually be larger than the number of modules in
64 // case the encoder was processing a frame during a codec reconfig.
65 if (rtp_video_hdr != NULL &&
66 rtp_video_hdr->simulcastIdx >= rtp_modules_.size())
67 return false;
68
69 int stream_idx = 0;
70 if (rtp_video_hdr != NULL)
71 stream_idx = rtp_video_hdr->simulcastIdx;
72 return rtp_modules_[stream_idx]->SendOutgoingData(
73 frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
74 payload_length, fragmentation, rtp_video_hdr) == 0 ? true : false;
75 }
76
SetTargetSendBitrates(const std::vector<uint32_t> & stream_bitrates)77 void PayloadRouter::SetTargetSendBitrates(
78 const std::vector<uint32_t>& stream_bitrates) {
79 CriticalSectionScoped cs(crit_.get());
80 if (stream_bitrates.size() < rtp_modules_.size()) {
81 // There can be a size mis-match during codec reconfiguration.
82 return;
83 }
84 int idx = 0;
85 for (auto* rtp_module : rtp_modules_) {
86 rtp_module->SetTargetSendBitrate(stream_bitrates[idx++]);
87 }
88 }
89
MaxPayloadLength() const90 size_t PayloadRouter::MaxPayloadLength() const {
91 size_t min_payload_length = DefaultMaxPayloadLength();
92 CriticalSectionScoped cs(crit_.get());
93 for (auto* rtp_module : rtp_modules_) {
94 size_t module_payload_length = rtp_module->MaxDataPayloadLength();
95 if (module_payload_length < min_payload_length)
96 min_payload_length = module_payload_length;
97 }
98 return min_payload_length;
99 }
100
101 } // namespace webrtc
102