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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
3 * Copyright 2004 Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#ifndef TALK_MEDIA_BASE_FAKENETWORKINTERFACE_H_
29#define TALK_MEDIA_BASE_FAKENETWORKINTERFACE_H_
30
31#include <vector>
32#include <map>
33
34#include "talk/base/buffer.h"
35#include "talk/base/byteorder.h"
36#include "talk/base/criticalsection.h"
wu@webrtc.orgde305012013-10-31 15:40:38 +000037#include "talk/base/dscp.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000038#include "talk/base/messagehandler.h"
39#include "talk/base/messagequeue.h"
40#include "talk/base/thread.h"
41#include "talk/media/base/mediachannel.h"
42#include "talk/media/base/rtputils.h"
43
44namespace cricket {
45
46// Fake NetworkInterface that sends/receives RTP/RTCP packets.
47class FakeNetworkInterface : public MediaChannel::NetworkInterface,
48 public talk_base::MessageHandler {
49 public:
50 FakeNetworkInterface()
51 : thread_(talk_base::Thread::Current()),
52 dest_(NULL),
53 conf_(false),
54 sendbuf_size_(-1),
wu@webrtc.orgde305012013-10-31 15:40:38 +000055 recvbuf_size_(-1),
56 dscp_(talk_base::DSCP_NO_CHANGE) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000057 }
58
59 void SetDestination(MediaChannel* dest) { dest_ = dest; }
60
61 // Conference mode is a mode where instead of simply forwarding the packets,
62 // the transport will send multiple copies of the packet with the specified
63 // SSRCs. This allows us to simulate receiving media from multiple sources.
64 void SetConferenceMode(bool conf, const std::vector<uint32>& ssrcs) {
65 talk_base::CritScope cs(&crit_);
66 conf_ = conf;
67 conf_sent_ssrcs_ = ssrcs;
68 }
69
70 int NumRtpBytes() {
71 talk_base::CritScope cs(&crit_);
72 int bytes = 0;
73 for (size_t i = 0; i < rtp_packets_.size(); ++i) {
henrike@webrtc.org28654cb2013-07-22 21:07:49 +000074 bytes += static_cast<int>(rtp_packets_[i].length());
henrike@webrtc.org28e20752013-07-10 00:45:36 +000075 }
76 return bytes;
77 }
78
79 int NumRtpBytes(uint32 ssrc) {
80 talk_base::CritScope cs(&crit_);
81 int bytes = 0;
82 GetNumRtpBytesAndPackets(ssrc, &bytes, NULL);
83 return bytes;
84 }
85
86 int NumRtpPackets() {
87 talk_base::CritScope cs(&crit_);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +000088 return static_cast<int>(rtp_packets_.size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +000089 }
90
91 int NumRtpPackets(uint32 ssrc) {
92 talk_base::CritScope cs(&crit_);
93 int packets = 0;
94 GetNumRtpBytesAndPackets(ssrc, NULL, &packets);
95 return packets;
96 }
97
98 int NumSentSsrcs() {
99 talk_base::CritScope cs(&crit_);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000100 return static_cast<int>(sent_ssrcs_.size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000101 }
102
103 // Note: callers are responsible for deleting the returned buffer.
104 const talk_base::Buffer* GetRtpPacket(int index) {
105 talk_base::CritScope cs(&crit_);
106 if (index >= NumRtpPackets()) {
107 return NULL;
108 }
109 return new talk_base::Buffer(rtp_packets_[index]);
110 }
111
112 int NumRtcpPackets() {
113 talk_base::CritScope cs(&crit_);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000114 return static_cast<int>(rtcp_packets_.size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000115 }
116
117 // Note: callers are responsible for deleting the returned buffer.
118 const talk_base::Buffer* GetRtcpPacket(int index) {
119 talk_base::CritScope cs(&crit_);
120 if (index >= NumRtcpPackets()) {
121 return NULL;
122 }
123 return new talk_base::Buffer(rtcp_packets_[index]);
124 }
125
126 // Indicate that |n|'th packet for |ssrc| should be dropped.
127 void AddPacketDrop(uint32 ssrc, uint32 n) {
128 drop_map_[ssrc].insert(n);
129 }
130
131 int sendbuf_size() const { return sendbuf_size_; }
132 int recvbuf_size() const { return recvbuf_size_; }
wu@webrtc.orgde305012013-10-31 15:40:38 +0000133 talk_base::DiffServCodePoint dscp() const { return dscp_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000134
135 protected:
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000136 virtual bool SendPacket(talk_base::Buffer* packet,
137 talk_base::DiffServCodePoint dscp) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000138 talk_base::CritScope cs(&crit_);
139
140 uint32 cur_ssrc = 0;
141 if (!GetRtpSsrc(packet->data(), packet->length(), &cur_ssrc)) {
142 return false;
143 }
144 sent_ssrcs_[cur_ssrc]++;
145
146 // Check if we need to drop this packet.
147 std::map<uint32, std::set<uint32> >::iterator itr =
148 drop_map_.find(cur_ssrc);
149 if (itr != drop_map_.end() &&
150 itr->second.count(sent_ssrcs_[cur_ssrc]) > 0) {
151 // "Drop" the packet.
152 return true;
153 }
154
155 rtp_packets_.push_back(*packet);
156 if (conf_) {
157 talk_base::Buffer buffer_copy(*packet);
158 for (size_t i = 0; i < conf_sent_ssrcs_.size(); ++i) {
159 if (!SetRtpSsrc(buffer_copy.data(), buffer_copy.length(),
160 conf_sent_ssrcs_[i])) {
161 return false;
162 }
163 PostMessage(ST_RTP, buffer_copy);
164 }
165 } else {
166 PostMessage(ST_RTP, *packet);
167 }
168 return true;
169 }
170
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000171 virtual bool SendRtcp(talk_base::Buffer* packet,
172 talk_base::DiffServCodePoint dscp) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000173 talk_base::CritScope cs(&crit_);
174 rtcp_packets_.push_back(*packet);
175 if (!conf_) {
176 // don't worry about RTCP in conf mode for now
177 PostMessage(ST_RTCP, *packet);
178 }
179 return true;
180 }
181
182 virtual int SetOption(SocketType type, talk_base::Socket::Option opt,
183 int option) {
184 if (opt == talk_base::Socket::OPT_SNDBUF) {
185 sendbuf_size_ = option;
186 } else if (opt == talk_base::Socket::OPT_RCVBUF) {
187 recvbuf_size_ = option;
wu@webrtc.orgde305012013-10-31 15:40:38 +0000188 } else if (opt == talk_base::Socket::OPT_DSCP) {
189 dscp_ = static_cast<talk_base::DiffServCodePoint>(option);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000190 }
191 return 0;
192 }
193
194 void PostMessage(int id, const talk_base::Buffer& packet) {
195 thread_->Post(this, id, talk_base::WrapMessageData(packet));
196 }
197
198 virtual void OnMessage(talk_base::Message* msg) {
199 talk_base::TypedMessageData<talk_base::Buffer>* msg_data =
200 static_cast<talk_base::TypedMessageData<talk_base::Buffer>*>(
201 msg->pdata);
202 if (dest_) {
203 if (msg->message_id == ST_RTP) {
204 dest_->OnPacketReceived(&msg_data->data());
205 } else {
206 dest_->OnRtcpReceived(&msg_data->data());
207 }
208 }
209 delete msg_data;
210 }
211
212 private:
213 void GetNumRtpBytesAndPackets(uint32 ssrc, int* bytes, int* packets) {
214 if (bytes) {
215 *bytes = 0;
216 }
217 if (packets) {
218 *packets = 0;
219 }
220 uint32 cur_ssrc = 0;
221 for (size_t i = 0; i < rtp_packets_.size(); ++i) {
222 if (!GetRtpSsrc(rtp_packets_[i].data(),
223 rtp_packets_[i].length(), &cur_ssrc)) {
224 return;
225 }
226 if (ssrc == cur_ssrc) {
227 if (bytes) {
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000228 *bytes += static_cast<int>(rtp_packets_[i].length());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000229 }
230 if (packets) {
231 ++(*packets);
232 }
233 }
234 }
235 }
236
237 talk_base::Thread* thread_;
238 MediaChannel* dest_;
239 bool conf_;
240 // The ssrcs used in sending out packets in conference mode.
241 std::vector<uint32> conf_sent_ssrcs_;
242 // Map to track counts of packets that have been sent per ssrc.
243 // This includes packets that are dropped.
244 std::map<uint32, uint32> sent_ssrcs_;
245 // Map to track packet-number that needs to be dropped per ssrc.
246 std::map<uint32, std::set<uint32> > drop_map_;
247 talk_base::CriticalSection crit_;
248 std::vector<talk_base::Buffer> rtp_packets_;
249 std::vector<talk_base::Buffer> rtcp_packets_;
250 int sendbuf_size_;
251 int recvbuf_size_;
wu@webrtc.orgde305012013-10-31 15:40:38 +0000252 talk_base::DiffServCodePoint dscp_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000253};
254
255} // namespace cricket
256
257#endif // TALK_MEDIA_BASE_FAKENETWORKINTERFACE_H_