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