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      1 /*
      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 <map>
     32 #include <vector>
     33 
     34 #include "talk/media/base/mediachannel.h"
     35 #include "talk/media/base/rtputils.h"
     36 #include "webrtc/base/buffer.h"
     37 #include "webrtc/base/byteorder.h"
     38 #include "webrtc/base/criticalsection.h"
     39 #include "webrtc/base/dscp.h"
     40 #include "webrtc/base/messagehandler.h"
     41 #include "webrtc/base/messagequeue.h"
     42 #include "webrtc/base/thread.h"
     43 
     44 namespace cricket {
     45 
     46 // Fake NetworkInterface that sends/receives RTP/RTCP packets.
     47 class FakeNetworkInterface : public MediaChannel::NetworkInterface,
     48                              public rtc::MessageHandler {
     49  public:
     50   FakeNetworkInterface()
     51       : thread_(rtc::Thread::Current()),
     52         dest_(NULL),
     53         conf_(false),
     54         sendbuf_size_(-1),
     55         recvbuf_size_(-1),
     56         dscp_(rtc::DSCP_NO_CHANGE) {
     57   }
     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_t>& ssrcs) {
     65     rtc::CritScope cs(&crit_);
     66     conf_ = conf;
     67     conf_sent_ssrcs_ = ssrcs;
     68   }
     69 
     70   int NumRtpBytes() {
     71     rtc::CritScope cs(&crit_);
     72     int bytes = 0;
     73     for (size_t i = 0; i < rtp_packets_.size(); ++i) {
     74       bytes += static_cast<int>(rtp_packets_[i].size());
     75     }
     76     return bytes;
     77   }
     78 
     79   int NumRtpBytes(uint32_t ssrc) {
     80     rtc::CritScope cs(&crit_);
     81     int bytes = 0;
     82     GetNumRtpBytesAndPackets(ssrc, &bytes, NULL);
     83     return bytes;
     84   }
     85 
     86   int NumRtpPackets() {
     87     rtc::CritScope cs(&crit_);
     88     return static_cast<int>(rtp_packets_.size());
     89   }
     90 
     91   int NumRtpPackets(uint32_t ssrc) {
     92     rtc::CritScope cs(&crit_);
     93     int packets = 0;
     94     GetNumRtpBytesAndPackets(ssrc, NULL, &packets);
     95     return packets;
     96   }
     97 
     98   int NumSentSsrcs() {
     99     rtc::CritScope cs(&crit_);
    100     return static_cast<int>(sent_ssrcs_.size());
    101   }
    102 
    103   // Note: callers are responsible for deleting the returned buffer.
    104   const rtc::Buffer* GetRtpPacket(int index) {
    105     rtc::CritScope cs(&crit_);
    106     if (index >= NumRtpPackets()) {
    107       return NULL;
    108     }
    109     return new rtc::Buffer(rtp_packets_[index]);
    110   }
    111 
    112   int NumRtcpPackets() {
    113     rtc::CritScope cs(&crit_);
    114     return static_cast<int>(rtcp_packets_.size());
    115   }
    116 
    117   // Note: callers are responsible for deleting the returned buffer.
    118   const rtc::Buffer* GetRtcpPacket(int index) {
    119     rtc::CritScope cs(&crit_);
    120     if (index >= NumRtcpPackets()) {
    121       return NULL;
    122     }
    123     return new rtc::Buffer(rtcp_packets_[index]);
    124   }
    125 
    126   int sendbuf_size() const { return sendbuf_size_; }
    127   int recvbuf_size() const { return recvbuf_size_; }
    128   rtc::DiffServCodePoint dscp() const { return dscp_; }
    129 
    130  protected:
    131   virtual bool SendPacket(rtc::Buffer* packet,
    132                           const rtc::PacketOptions& options) {
    133     rtc::CritScope cs(&crit_);
    134 
    135     uint32_t cur_ssrc = 0;
    136     if (!GetRtpSsrc(packet->data(), packet->size(), &cur_ssrc)) {
    137       return false;
    138     }
    139     sent_ssrcs_[cur_ssrc]++;
    140 
    141     rtp_packets_.push_back(*packet);
    142     if (conf_) {
    143       rtc::Buffer buffer_copy(*packet);
    144       for (size_t i = 0; i < conf_sent_ssrcs_.size(); ++i) {
    145         if (!SetRtpSsrc(buffer_copy.data(), buffer_copy.size(),
    146                         conf_sent_ssrcs_[i])) {
    147           return false;
    148         }
    149         PostMessage(ST_RTP, buffer_copy);
    150       }
    151     } else {
    152       PostMessage(ST_RTP, *packet);
    153     }
    154     return true;
    155   }
    156 
    157   virtual bool SendRtcp(rtc::Buffer* packet,
    158                         const rtc::PacketOptions& options) {
    159     rtc::CritScope cs(&crit_);
    160     rtcp_packets_.push_back(*packet);
    161     if (!conf_) {
    162       // don't worry about RTCP in conf mode for now
    163       PostMessage(ST_RTCP, *packet);
    164     }
    165     return true;
    166   }
    167 
    168   virtual int SetOption(SocketType type, rtc::Socket::Option opt,
    169                         int option) {
    170     if (opt == rtc::Socket::OPT_SNDBUF) {
    171       sendbuf_size_ = option;
    172     } else if (opt == rtc::Socket::OPT_RCVBUF) {
    173       recvbuf_size_ = option;
    174     } else if (opt == rtc::Socket::OPT_DSCP) {
    175       dscp_ = static_cast<rtc::DiffServCodePoint>(option);
    176     }
    177     return 0;
    178   }
    179 
    180   void PostMessage(int id, const rtc::Buffer& packet) {
    181     thread_->Post(this, id, rtc::WrapMessageData(packet));
    182   }
    183 
    184   virtual void OnMessage(rtc::Message* msg) {
    185     rtc::TypedMessageData<rtc::Buffer>* msg_data =
    186         static_cast<rtc::TypedMessageData<rtc::Buffer>*>(
    187             msg->pdata);
    188     if (dest_) {
    189       if (msg->message_id == ST_RTP) {
    190         dest_->OnPacketReceived(&msg_data->data(),
    191                                 rtc::CreatePacketTime(0));
    192       } else {
    193         dest_->OnRtcpReceived(&msg_data->data(),
    194                               rtc::CreatePacketTime(0));
    195       }
    196     }
    197     delete msg_data;
    198   }
    199 
    200  private:
    201   void GetNumRtpBytesAndPackets(uint32_t ssrc, int* bytes, int* packets) {
    202     if (bytes) {
    203       *bytes = 0;
    204     }
    205     if (packets) {
    206       *packets = 0;
    207     }
    208     uint32_t cur_ssrc = 0;
    209     for (size_t i = 0; i < rtp_packets_.size(); ++i) {
    210       if (!GetRtpSsrc(rtp_packets_[i].data(), rtp_packets_[i].size(),
    211                       &cur_ssrc)) {
    212         return;
    213       }
    214       if (ssrc == cur_ssrc) {
    215         if (bytes) {
    216           *bytes += static_cast<int>(rtp_packets_[i].size());
    217         }
    218         if (packets) {
    219           ++(*packets);
    220         }
    221       }
    222     }
    223   }
    224 
    225   rtc::Thread* thread_;
    226   MediaChannel* dest_;
    227   bool conf_;
    228   // The ssrcs used in sending out packets in conference mode.
    229   std::vector<uint32_t> conf_sent_ssrcs_;
    230   // Map to track counts of packets that have been sent per ssrc.
    231   // This includes packets that are dropped.
    232   std::map<uint32_t, uint32_t> sent_ssrcs_;
    233   // Map to track packet-number that needs to be dropped per ssrc.
    234   std::map<uint32_t, std::set<uint32_t> > drop_map_;
    235   rtc::CriticalSection crit_;
    236   std::vector<rtc::Buffer> rtp_packets_;
    237   std::vector<rtc::Buffer> rtcp_packets_;
    238   int sendbuf_size_;
    239   int recvbuf_size_;
    240   rtc::DiffServCodePoint dscp_;
    241 };
    242 
    243 }  // namespace cricket
    244 
    245 #endif  // TALK_MEDIA_BASE_FAKENETWORKINTERFACE_H_
    246