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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/modules/pacing/packet_router.h"
12 
13 #include "webrtc/base/atomicops.h"
14 #include "webrtc/base/checks.h"
15 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
16 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
17 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
18 
19 namespace webrtc {
20 
PacketRouter()21 PacketRouter::PacketRouter() : transport_seq_(0) {
22 }
23 
~PacketRouter()24 PacketRouter::~PacketRouter() {
25   RTC_DCHECK(rtp_modules_.empty());
26 }
27 
AddRtpModule(RtpRtcp * rtp_module)28 void PacketRouter::AddRtpModule(RtpRtcp* rtp_module) {
29   rtc::CritScope cs(&modules_lock_);
30   RTC_DCHECK(std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module) ==
31              rtp_modules_.end());
32   rtp_modules_.push_back(rtp_module);
33 }
34 
RemoveRtpModule(RtpRtcp * rtp_module)35 void PacketRouter::RemoveRtpModule(RtpRtcp* rtp_module) {
36   rtc::CritScope cs(&modules_lock_);
37   auto it = std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module);
38   RTC_DCHECK(it != rtp_modules_.end());
39   rtp_modules_.erase(it);
40 }
41 
TimeToSendPacket(uint32_t ssrc,uint16_t sequence_number,int64_t capture_timestamp,bool retransmission)42 bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
43                                     uint16_t sequence_number,
44                                     int64_t capture_timestamp,
45                                     bool retransmission) {
46   rtc::CritScope cs(&modules_lock_);
47   for (auto* rtp_module : rtp_modules_) {
48     if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) {
49       return rtp_module->TimeToSendPacket(ssrc, sequence_number,
50                                           capture_timestamp, retransmission);
51     }
52   }
53   return true;
54 }
55 
TimeToSendPadding(size_t bytes_to_send)56 size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send) {
57   size_t total_bytes_sent = 0;
58   rtc::CritScope cs(&modules_lock_);
59   for (RtpRtcp* module : rtp_modules_) {
60     if (module->SendingMedia()) {
61       size_t bytes_sent =
62           module->TimeToSendPadding(bytes_to_send - total_bytes_sent);
63       total_bytes_sent += bytes_sent;
64       if (total_bytes_sent >= bytes_to_send)
65         break;
66     }
67   }
68   return total_bytes_sent;
69 }
70 
SetTransportWideSequenceNumber(uint16_t sequence_number)71 void PacketRouter::SetTransportWideSequenceNumber(uint16_t sequence_number) {
72   rtc::AtomicOps::ReleaseStore(&transport_seq_, sequence_number);
73 }
74 
AllocateSequenceNumber()75 uint16_t PacketRouter::AllocateSequenceNumber() {
76   int prev_seq = rtc::AtomicOps::AcquireLoad(&transport_seq_);
77   int desired_prev_seq;
78   int new_seq;
79   do {
80     desired_prev_seq = prev_seq;
81     new_seq = (desired_prev_seq + 1) & 0xFFFF;
82     // Note: CompareAndSwap returns the actual value of transport_seq at the
83     // time the CAS operation was executed. Thus, if prev_seq is returned, the
84     // operation was successful - otherwise we need to retry. Saving the
85     // return value saves us a load on retry.
86     prev_seq = rtc::AtomicOps::CompareAndSwap(&transport_seq_, desired_prev_seq,
87                                               new_seq);
88   } while (prev_seq != desired_prev_seq);
89 
90   return new_seq;
91 }
92 
SendFeedback(rtcp::TransportFeedback * packet)93 bool PacketRouter::SendFeedback(rtcp::TransportFeedback* packet) {
94   rtc::CritScope cs(&modules_lock_);
95   for (auto* rtp_module : rtp_modules_) {
96     packet->WithPacketSenderSsrc(rtp_module->SSRC());
97     if (rtp_module->SendFeedbackPacket(*packet))
98       return true;
99   }
100   return false;
101 }
102 
103 }  // namespace webrtc
104