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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
kjellander1afca732016-02-07 20:46:45 -08002 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00003 *
kjellander1afca732016-02-07 20:46:45 -08004 * 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.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00009 */
10
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#ifndef MEDIA_BASE_FAKENETWORKINTERFACE_H_
12#define MEDIA_BASE_FAKENETWORKINTERFACE_H_
henrike@webrtc.org28e20752013-07-10 00:45:36 +000013
henrike@webrtc.org28e20752013-07-10 00:45:36 +000014#include <map>
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000015#include <vector>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000016
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020017#include "media/base/mediachannel.h"
18#include "media/base/rtputils.h"
19#include "rtc_base/byteorder.h"
20#include "rtc_base/copyonwritebuffer.h"
21#include "rtc_base/criticalsection.h"
22#include "rtc_base/dscp.h"
23#include "rtc_base/messagehandler.h"
24#include "rtc_base/messagequeue.h"
25#include "rtc_base/thread.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000026
27namespace cricket {
28
29// Fake NetworkInterface that sends/receives RTP/RTCP packets.
30class FakeNetworkInterface : public MediaChannel::NetworkInterface,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000031 public rtc::MessageHandler {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000032 public:
33 FakeNetworkInterface()
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000034 : thread_(rtc::Thread::Current()),
henrike@webrtc.org28e20752013-07-10 00:45:36 +000035 dest_(NULL),
36 conf_(false),
37 sendbuf_size_(-1),
wu@webrtc.orgde305012013-10-31 15:40:38 +000038 recvbuf_size_(-1),
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000039 dscp_(rtc::DSCP_NO_CHANGE) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000040 }
41
42 void SetDestination(MediaChannel* dest) { dest_ = dest; }
43
44 // Conference mode is a mode where instead of simply forwarding the packets,
45 // the transport will send multiple copies of the packet with the specified
46 // SSRCs. This allows us to simulate receiving media from multiple sources.
Peter Boström0c4e06b2015-10-07 12:23:21 +020047 void SetConferenceMode(bool conf, const std::vector<uint32_t>& ssrcs) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000048 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000049 conf_ = conf;
50 conf_sent_ssrcs_ = ssrcs;
51 }
52
53 int NumRtpBytes() {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000054 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000055 int bytes = 0;
56 for (size_t i = 0; i < rtp_packets_.size(); ++i) {
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +000057 bytes += static_cast<int>(rtp_packets_[i].size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +000058 }
59 return bytes;
60 }
61
Peter Boström0c4e06b2015-10-07 12:23:21 +020062 int NumRtpBytes(uint32_t ssrc) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000063 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000064 int bytes = 0;
65 GetNumRtpBytesAndPackets(ssrc, &bytes, NULL);
66 return bytes;
67 }
68
69 int NumRtpPackets() {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000070 rtc::CritScope cs(&crit_);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +000071 return static_cast<int>(rtp_packets_.size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +000072 }
73
Peter Boström0c4e06b2015-10-07 12:23:21 +020074 int NumRtpPackets(uint32_t ssrc) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000075 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000076 int packets = 0;
77 GetNumRtpBytesAndPackets(ssrc, NULL, &packets);
78 return packets;
79 }
80
81 int NumSentSsrcs() {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000082 rtc::CritScope cs(&crit_);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +000083 return static_cast<int>(sent_ssrcs_.size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +000084 }
85
86 // Note: callers are responsible for deleting the returned buffer.
jbaucheec21bd2016-03-20 06:15:43 -070087 const rtc::CopyOnWriteBuffer* GetRtpPacket(int index) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000088 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000089 if (index >= NumRtpPackets()) {
90 return NULL;
91 }
jbaucheec21bd2016-03-20 06:15:43 -070092 return new rtc::CopyOnWriteBuffer(rtp_packets_[index]);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000093 }
94
95 int NumRtcpPackets() {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000096 rtc::CritScope cs(&crit_);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +000097 return static_cast<int>(rtcp_packets_.size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +000098 }
99
100 // Note: callers are responsible for deleting the returned buffer.
jbaucheec21bd2016-03-20 06:15:43 -0700101 const rtc::CopyOnWriteBuffer* GetRtcpPacket(int index) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000102 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000103 if (index >= NumRtcpPackets()) {
104 return NULL;
105 }
jbaucheec21bd2016-03-20 06:15:43 -0700106 return new rtc::CopyOnWriteBuffer(rtcp_packets_[index]);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000107 }
108
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000109 int sendbuf_size() const { return sendbuf_size_; }
110 int recvbuf_size() const { return recvbuf_size_; }
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000111 rtc::DiffServCodePoint dscp() const { return dscp_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000112
113 protected:
jbaucheec21bd2016-03-20 06:15:43 -0700114 virtual bool SendPacket(rtc::CopyOnWriteBuffer* packet,
stefanc1aeaf02015-10-15 07:26:07 -0700115 const rtc::PacketOptions& options) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000116 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000117
Peter Boström0c4e06b2015-10-07 12:23:21 +0200118 uint32_t cur_ssrc = 0;
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +0000119 if (!GetRtpSsrc(packet->data(), packet->size(), &cur_ssrc)) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000120 return false;
121 }
122 sent_ssrcs_[cur_ssrc]++;
123
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000124 rtp_packets_.push_back(*packet);
125 if (conf_) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000126 for (size_t i = 0; i < conf_sent_ssrcs_.size(); ++i) {
jbaucheec21bd2016-03-20 06:15:43 -0700127 if (!SetRtpSsrc(packet->data(), packet->size(),
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000128 conf_sent_ssrcs_[i])) {
129 return false;
130 }
jbaucheec21bd2016-03-20 06:15:43 -0700131 PostMessage(ST_RTP, *packet);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000132 }
133 } else {
134 PostMessage(ST_RTP, *packet);
135 }
136 return true;
137 }
138
jbaucheec21bd2016-03-20 06:15:43 -0700139 virtual bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
stefanc1aeaf02015-10-15 07:26:07 -0700140 const rtc::PacketOptions& options) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000141 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000142 rtcp_packets_.push_back(*packet);
143 if (!conf_) {
144 // don't worry about RTCP in conf mode for now
145 PostMessage(ST_RTCP, *packet);
146 }
147 return true;
148 }
149
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000150 virtual int SetOption(SocketType type, rtc::Socket::Option opt,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000151 int option) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000152 if (opt == rtc::Socket::OPT_SNDBUF) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000153 sendbuf_size_ = option;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000154 } else if (opt == rtc::Socket::OPT_RCVBUF) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000155 recvbuf_size_ = option;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000156 } else if (opt == rtc::Socket::OPT_DSCP) {
157 dscp_ = static_cast<rtc::DiffServCodePoint>(option);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000158 }
159 return 0;
160 }
161
jbaucheec21bd2016-03-20 06:15:43 -0700162 void PostMessage(int id, const rtc::CopyOnWriteBuffer& packet) {
Taylor Brandstetter5d97a9a2016-06-10 14:17:27 -0700163 thread_->Post(RTC_FROM_HERE, this, id, rtc::WrapMessageData(packet));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000164 }
165
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000166 virtual void OnMessage(rtc::Message* msg) {
jbaucheec21bd2016-03-20 06:15:43 -0700167 rtc::TypedMessageData<rtc::CopyOnWriteBuffer>* msg_data =
168 static_cast<rtc::TypedMessageData<rtc::CopyOnWriteBuffer>*>(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000169 msg->pdata);
170 if (dest_) {
171 if (msg->message_id == ST_RTP) {
wu@webrtc.orga9890802013-12-13 00:21:03 +0000172 dest_->OnPacketReceived(&msg_data->data(),
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000173 rtc::CreatePacketTime(0));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000174 } else {
wu@webrtc.orga9890802013-12-13 00:21:03 +0000175 dest_->OnRtcpReceived(&msg_data->data(),
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000176 rtc::CreatePacketTime(0));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000177 }
178 }
179 delete msg_data;
180 }
181
182 private:
Peter Boström0c4e06b2015-10-07 12:23:21 +0200183 void GetNumRtpBytesAndPackets(uint32_t ssrc, int* bytes, int* packets) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000184 if (bytes) {
185 *bytes = 0;
186 }
187 if (packets) {
188 *packets = 0;
189 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200190 uint32_t cur_ssrc = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000191 for (size_t i = 0; i < rtp_packets_.size(); ++i) {
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +0000192 if (!GetRtpSsrc(rtp_packets_[i].data(), rtp_packets_[i].size(),
193 &cur_ssrc)) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000194 return;
195 }
196 if (ssrc == cur_ssrc) {
197 if (bytes) {
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +0000198 *bytes += static_cast<int>(rtp_packets_[i].size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000199 }
200 if (packets) {
201 ++(*packets);
202 }
203 }
204 }
205 }
206
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000207 rtc::Thread* thread_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000208 MediaChannel* dest_;
209 bool conf_;
210 // The ssrcs used in sending out packets in conference mode.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200211 std::vector<uint32_t> conf_sent_ssrcs_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000212 // Map to track counts of packets that have been sent per ssrc.
213 // This includes packets that are dropped.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200214 std::map<uint32_t, uint32_t> sent_ssrcs_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000215 // Map to track packet-number that needs to be dropped per ssrc.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200216 std::map<uint32_t, std::set<uint32_t> > drop_map_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000217 rtc::CriticalSection crit_;
jbaucheec21bd2016-03-20 06:15:43 -0700218 std::vector<rtc::CopyOnWriteBuffer> rtp_packets_;
219 std::vector<rtc::CopyOnWriteBuffer> rtcp_packets_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000220 int sendbuf_size_;
221 int recvbuf_size_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000222 rtc::DiffServCodePoint dscp_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000223};
224
225} // namespace cricket
226
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200227#endif // MEDIA_BASE_FAKENETWORKINTERFACE_H_