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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>
Steve Antone78bcb92017-10-31 09:53:08 -070015#include <set>
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000016#include <vector>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000017
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020018#include "media/base/mediachannel.h"
19#include "media/base/rtputils.h"
20#include "rtc_base/byteorder.h"
21#include "rtc_base/copyonwritebuffer.h"
22#include "rtc_base/criticalsection.h"
23#include "rtc_base/dscp.h"
24#include "rtc_base/messagehandler.h"
25#include "rtc_base/messagequeue.h"
26#include "rtc_base/thread.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000027
28namespace cricket {
29
30// Fake NetworkInterface that sends/receives RTP/RTCP packets.
31class FakeNetworkInterface : public MediaChannel::NetworkInterface,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000032 public rtc::MessageHandler {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000033 public:
34 FakeNetworkInterface()
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000035 : thread_(rtc::Thread::Current()),
henrike@webrtc.org28e20752013-07-10 00:45:36 +000036 dest_(NULL),
37 conf_(false),
38 sendbuf_size_(-1),
wu@webrtc.orgde305012013-10-31 15:40:38 +000039 recvbuf_size_(-1),
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000040 dscp_(rtc::DSCP_NO_CHANGE) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000041 }
42
43 void SetDestination(MediaChannel* dest) { dest_ = dest; }
44
45 // Conference mode is a mode where instead of simply forwarding the packets,
46 // the transport will send multiple copies of the packet with the specified
47 // SSRCs. This allows us to simulate receiving media from multiple sources.
Peter Boström0c4e06b2015-10-07 12:23:21 +020048 void SetConferenceMode(bool conf, const std::vector<uint32_t>& ssrcs) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000049 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000050 conf_ = conf;
51 conf_sent_ssrcs_ = ssrcs;
52 }
53
54 int NumRtpBytes() {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000055 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000056 int bytes = 0;
57 for (size_t i = 0; i < rtp_packets_.size(); ++i) {
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +000058 bytes += static_cast<int>(rtp_packets_[i].size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +000059 }
60 return bytes;
61 }
62
Peter Boström0c4e06b2015-10-07 12:23:21 +020063 int NumRtpBytes(uint32_t ssrc) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000064 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000065 int bytes = 0;
66 GetNumRtpBytesAndPackets(ssrc, &bytes, NULL);
67 return bytes;
68 }
69
70 int NumRtpPackets() {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000071 rtc::CritScope cs(&crit_);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +000072 return static_cast<int>(rtp_packets_.size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +000073 }
74
Peter Boström0c4e06b2015-10-07 12:23:21 +020075 int NumRtpPackets(uint32_t ssrc) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000076 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000077 int packets = 0;
78 GetNumRtpBytesAndPackets(ssrc, NULL, &packets);
79 return packets;
80 }
81
82 int NumSentSsrcs() {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000083 rtc::CritScope cs(&crit_);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +000084 return static_cast<int>(sent_ssrcs_.size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +000085 }
86
87 // Note: callers are responsible for deleting the returned buffer.
jbaucheec21bd2016-03-20 06:15:43 -070088 const rtc::CopyOnWriteBuffer* GetRtpPacket(int index) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000089 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000090 if (index >= NumRtpPackets()) {
91 return NULL;
92 }
jbaucheec21bd2016-03-20 06:15:43 -070093 return new rtc::CopyOnWriteBuffer(rtp_packets_[index]);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000094 }
95
96 int NumRtcpPackets() {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000097 rtc::CritScope cs(&crit_);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +000098 return static_cast<int>(rtcp_packets_.size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +000099 }
100
101 // Note: callers are responsible for deleting the returned buffer.
jbaucheec21bd2016-03-20 06:15:43 -0700102 const rtc::CopyOnWriteBuffer* GetRtcpPacket(int index) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000103 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000104 if (index >= NumRtcpPackets()) {
105 return NULL;
106 }
jbaucheec21bd2016-03-20 06:15:43 -0700107 return new rtc::CopyOnWriteBuffer(rtcp_packets_[index]);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000108 }
109
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000110 int sendbuf_size() const { return sendbuf_size_; }
111 int recvbuf_size() const { return recvbuf_size_; }
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000112 rtc::DiffServCodePoint dscp() const { return dscp_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000113
114 protected:
jbaucheec21bd2016-03-20 06:15:43 -0700115 virtual bool SendPacket(rtc::CopyOnWriteBuffer* packet,
stefanc1aeaf02015-10-15 07:26:07 -0700116 const rtc::PacketOptions& options) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000117 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000118
Peter Boström0c4e06b2015-10-07 12:23:21 +0200119 uint32_t cur_ssrc = 0;
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +0000120 if (!GetRtpSsrc(packet->data(), packet->size(), &cur_ssrc)) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000121 return false;
122 }
123 sent_ssrcs_[cur_ssrc]++;
124
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000125 rtp_packets_.push_back(*packet);
126 if (conf_) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000127 for (size_t i = 0; i < conf_sent_ssrcs_.size(); ++i) {
jbaucheec21bd2016-03-20 06:15:43 -0700128 if (!SetRtpSsrc(packet->data(), packet->size(),
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000129 conf_sent_ssrcs_[i])) {
130 return false;
131 }
jbaucheec21bd2016-03-20 06:15:43 -0700132 PostMessage(ST_RTP, *packet);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000133 }
134 } else {
135 PostMessage(ST_RTP, *packet);
136 }
137 return true;
138 }
139
jbaucheec21bd2016-03-20 06:15:43 -0700140 virtual bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
stefanc1aeaf02015-10-15 07:26:07 -0700141 const rtc::PacketOptions& options) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000142 rtc::CritScope cs(&crit_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000143 rtcp_packets_.push_back(*packet);
144 if (!conf_) {
145 // don't worry about RTCP in conf mode for now
146 PostMessage(ST_RTCP, *packet);
147 }
148 return true;
149 }
150
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000151 virtual int SetOption(SocketType type, rtc::Socket::Option opt,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000152 int option) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000153 if (opt == rtc::Socket::OPT_SNDBUF) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000154 sendbuf_size_ = option;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000155 } else if (opt == rtc::Socket::OPT_RCVBUF) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000156 recvbuf_size_ = option;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000157 } else if (opt == rtc::Socket::OPT_DSCP) {
158 dscp_ = static_cast<rtc::DiffServCodePoint>(option);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000159 }
160 return 0;
161 }
162
jbaucheec21bd2016-03-20 06:15:43 -0700163 void PostMessage(int id, const rtc::CopyOnWriteBuffer& packet) {
Taylor Brandstetter5d97a9a2016-06-10 14:17:27 -0700164 thread_->Post(RTC_FROM_HERE, this, id, rtc::WrapMessageData(packet));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000165 }
166
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000167 virtual void OnMessage(rtc::Message* msg) {
jbaucheec21bd2016-03-20 06:15:43 -0700168 rtc::TypedMessageData<rtc::CopyOnWriteBuffer>* msg_data =
169 static_cast<rtc::TypedMessageData<rtc::CopyOnWriteBuffer>*>(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000170 msg->pdata);
171 if (dest_) {
172 if (msg->message_id == ST_RTP) {
wu@webrtc.orga9890802013-12-13 00:21:03 +0000173 dest_->OnPacketReceived(&msg_data->data(),
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000174 rtc::CreatePacketTime(0));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000175 } else {
wu@webrtc.orga9890802013-12-13 00:21:03 +0000176 dest_->OnRtcpReceived(&msg_data->data(),
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000177 rtc::CreatePacketTime(0));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000178 }
179 }
180 delete msg_data;
181 }
182
183 private:
Peter Boström0c4e06b2015-10-07 12:23:21 +0200184 void GetNumRtpBytesAndPackets(uint32_t ssrc, int* bytes, int* packets) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000185 if (bytes) {
186 *bytes = 0;
187 }
188 if (packets) {
189 *packets = 0;
190 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200191 uint32_t cur_ssrc = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000192 for (size_t i = 0; i < rtp_packets_.size(); ++i) {
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +0000193 if (!GetRtpSsrc(rtp_packets_[i].data(), rtp_packets_[i].size(),
194 &cur_ssrc)) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000195 return;
196 }
197 if (ssrc == cur_ssrc) {
198 if (bytes) {
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +0000199 *bytes += static_cast<int>(rtp_packets_[i].size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000200 }
201 if (packets) {
202 ++(*packets);
203 }
204 }
205 }
206 }
207
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000208 rtc::Thread* thread_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000209 MediaChannel* dest_;
210 bool conf_;
211 // The ssrcs used in sending out packets in conference mode.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200212 std::vector<uint32_t> conf_sent_ssrcs_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000213 // Map to track counts of packets that have been sent per ssrc.
214 // This includes packets that are dropped.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200215 std::map<uint32_t, uint32_t> sent_ssrcs_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000216 // Map to track packet-number that needs to be dropped per ssrc.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200217 std::map<uint32_t, std::set<uint32_t> > drop_map_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000218 rtc::CriticalSection crit_;
jbaucheec21bd2016-03-20 06:15:43 -0700219 std::vector<rtc::CopyOnWriteBuffer> rtp_packets_;
220 std::vector<rtc::CopyOnWriteBuffer> rtcp_packets_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000221 int sendbuf_size_;
222 int recvbuf_size_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000223 rtc::DiffServCodePoint dscp_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000224};
225
226} // namespace cricket
227
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200228#endif // MEDIA_BASE_FAKENETWORKINTERFACE_H_