https://webrtc-codereview.appspot.com/5309005/. R=mallinath@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
		
			
				
	
	
		
			260 lines
		
	
	
		
			8.2 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
			
		
		
	
	
			260 lines
		
	
	
		
			8.2 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
/*
 | 
						|
 * libjingle
 | 
						|
 * Copyright 2004 Google Inc.
 | 
						|
 *
 | 
						|
 * Redistribution and use in source and binary forms, with or without
 | 
						|
 * modification, are permitted provided that the following conditions are met:
 | 
						|
 *
 | 
						|
 *  1. Redistributions of source code must retain the above copyright notice,
 | 
						|
 *     this list of conditions and the following disclaimer.
 | 
						|
 *  2. Redistributions in binary form must reproduce the above copyright notice,
 | 
						|
 *     this list of conditions and the following disclaimer in the documentation
 | 
						|
 *     and/or other materials provided with the distribution.
 | 
						|
 *  3. The name of the author may not be used to endorse or promote products
 | 
						|
 *     derived from this software without specific prior written permission.
 | 
						|
 *
 | 
						|
 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
 | 
						|
 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
 | 
						|
 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
 | 
						|
 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
 | 
						|
 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
 | 
						|
 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
 | 
						|
 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
 | 
						|
 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
 | 
						|
 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
 | 
						|
 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 | 
						|
 */
 | 
						|
 | 
						|
#ifndef TALK_MEDIA_BASE_FAKENETWORKINTERFACE_H_
 | 
						|
#define TALK_MEDIA_BASE_FAKENETWORKINTERFACE_H_
 | 
						|
 | 
						|
#include <vector>
 | 
						|
#include <map>
 | 
						|
 | 
						|
#include "talk/base/buffer.h"
 | 
						|
#include "talk/base/byteorder.h"
 | 
						|
#include "talk/base/criticalsection.h"
 | 
						|
#include "talk/base/dscp.h"
 | 
						|
#include "talk/base/messagehandler.h"
 | 
						|
#include "talk/base/messagequeue.h"
 | 
						|
#include "talk/base/thread.h"
 | 
						|
#include "talk/media/base/mediachannel.h"
 | 
						|
#include "talk/media/base/rtputils.h"
 | 
						|
 | 
						|
namespace cricket {
 | 
						|
 | 
						|
// Fake NetworkInterface that sends/receives RTP/RTCP packets.
 | 
						|
class FakeNetworkInterface : public MediaChannel::NetworkInterface,
 | 
						|
                             public talk_base::MessageHandler {
 | 
						|
 public:
 | 
						|
  FakeNetworkInterface()
 | 
						|
      : thread_(talk_base::Thread::Current()),
 | 
						|
        dest_(NULL),
 | 
						|
        conf_(false),
 | 
						|
        sendbuf_size_(-1),
 | 
						|
        recvbuf_size_(-1),
 | 
						|
        dscp_(talk_base::DSCP_NO_CHANGE) {
 | 
						|
  }
 | 
						|
 | 
						|
  void SetDestination(MediaChannel* dest) { dest_ = dest; }
 | 
						|
 | 
						|
  // Conference mode is a mode where instead of simply forwarding the packets,
 | 
						|
  // the transport will send multiple copies of the packet with the specified
 | 
						|
  // SSRCs. This allows us to simulate receiving media from multiple sources.
 | 
						|
  void SetConferenceMode(bool conf, const std::vector<uint32>& ssrcs) {
 | 
						|
    talk_base::CritScope cs(&crit_);
 | 
						|
    conf_ = conf;
 | 
						|
    conf_sent_ssrcs_ = ssrcs;
 | 
						|
  }
 | 
						|
 | 
						|
  int NumRtpBytes() {
 | 
						|
    talk_base::CritScope cs(&crit_);
 | 
						|
    int bytes = 0;
 | 
						|
    for (size_t i = 0; i < rtp_packets_.size(); ++i) {
 | 
						|
      bytes += static_cast<int>(rtp_packets_[i].length());
 | 
						|
    }
 | 
						|
    return bytes;
 | 
						|
  }
 | 
						|
 | 
						|
  int NumRtpBytes(uint32 ssrc) {
 | 
						|
    talk_base::CritScope cs(&crit_);
 | 
						|
    int bytes = 0;
 | 
						|
    GetNumRtpBytesAndPackets(ssrc, &bytes, NULL);
 | 
						|
    return bytes;
 | 
						|
  }
 | 
						|
 | 
						|
  int NumRtpPackets() {
 | 
						|
    talk_base::CritScope cs(&crit_);
 | 
						|
    return static_cast<int>(rtp_packets_.size());
 | 
						|
  }
 | 
						|
 | 
						|
  int NumRtpPackets(uint32 ssrc) {
 | 
						|
    talk_base::CritScope cs(&crit_);
 | 
						|
    int packets = 0;
 | 
						|
    GetNumRtpBytesAndPackets(ssrc, NULL, &packets);
 | 
						|
    return packets;
 | 
						|
  }
 | 
						|
 | 
						|
  int NumSentSsrcs() {
 | 
						|
    talk_base::CritScope cs(&crit_);
 | 
						|
    return static_cast<int>(sent_ssrcs_.size());
 | 
						|
  }
 | 
						|
 | 
						|
  // Note: callers are responsible for deleting the returned buffer.
 | 
						|
  const talk_base::Buffer* GetRtpPacket(int index) {
 | 
						|
    talk_base::CritScope cs(&crit_);
 | 
						|
    if (index >= NumRtpPackets()) {
 | 
						|
      return NULL;
 | 
						|
    }
 | 
						|
    return new talk_base::Buffer(rtp_packets_[index]);
 | 
						|
  }
 | 
						|
 | 
						|
  int NumRtcpPackets() {
 | 
						|
    talk_base::CritScope cs(&crit_);
 | 
						|
    return static_cast<int>(rtcp_packets_.size());
 | 
						|
  }
 | 
						|
 | 
						|
  // Note: callers are responsible for deleting the returned buffer.
 | 
						|
  const talk_base::Buffer* GetRtcpPacket(int index) {
 | 
						|
    talk_base::CritScope cs(&crit_);
 | 
						|
    if (index >= NumRtcpPackets()) {
 | 
						|
      return NULL;
 | 
						|
    }
 | 
						|
    return new talk_base::Buffer(rtcp_packets_[index]);
 | 
						|
  }
 | 
						|
 | 
						|
  // Indicate that |n|'th packet for |ssrc| should be dropped.
 | 
						|
  void AddPacketDrop(uint32 ssrc, uint32 n) {
 | 
						|
    drop_map_[ssrc].insert(n);
 | 
						|
  }
 | 
						|
 | 
						|
  int sendbuf_size() const { return sendbuf_size_; }
 | 
						|
  int recvbuf_size() const { return recvbuf_size_; }
 | 
						|
  talk_base::DiffServCodePoint dscp() const { return dscp_; }
 | 
						|
 | 
						|
 protected:
 | 
						|
  virtual bool SendPacket(talk_base::Buffer* packet,
 | 
						|
                          talk_base::DiffServCodePoint dscp) {
 | 
						|
    talk_base::CritScope cs(&crit_);
 | 
						|
 | 
						|
    uint32 cur_ssrc = 0;
 | 
						|
    if (!GetRtpSsrc(packet->data(), packet->length(), &cur_ssrc)) {
 | 
						|
      return false;
 | 
						|
    }
 | 
						|
    sent_ssrcs_[cur_ssrc]++;
 | 
						|
 | 
						|
    // Check if we need to drop this packet.
 | 
						|
    std::map<uint32, std::set<uint32> >::iterator itr =
 | 
						|
      drop_map_.find(cur_ssrc);
 | 
						|
    if (itr != drop_map_.end() &&
 | 
						|
        itr->second.count(sent_ssrcs_[cur_ssrc]) > 0) {
 | 
						|
        // "Drop" the packet.
 | 
						|
        return true;
 | 
						|
    }
 | 
						|
 | 
						|
    rtp_packets_.push_back(*packet);
 | 
						|
    if (conf_) {
 | 
						|
      talk_base::Buffer buffer_copy(*packet);
 | 
						|
      for (size_t i = 0; i < conf_sent_ssrcs_.size(); ++i) {
 | 
						|
        if (!SetRtpSsrc(buffer_copy.data(), buffer_copy.length(),
 | 
						|
                        conf_sent_ssrcs_[i])) {
 | 
						|
          return false;
 | 
						|
        }
 | 
						|
        PostMessage(ST_RTP, buffer_copy);
 | 
						|
      }
 | 
						|
    } else {
 | 
						|
      PostMessage(ST_RTP, *packet);
 | 
						|
    }
 | 
						|
    return true;
 | 
						|
  }
 | 
						|
 | 
						|
  virtual bool SendRtcp(talk_base::Buffer* packet,
 | 
						|
                        talk_base::DiffServCodePoint dscp) {
 | 
						|
    talk_base::CritScope cs(&crit_);
 | 
						|
    rtcp_packets_.push_back(*packet);
 | 
						|
    if (!conf_) {
 | 
						|
      // don't worry about RTCP in conf mode for now
 | 
						|
      PostMessage(ST_RTCP, *packet);
 | 
						|
    }
 | 
						|
    return true;
 | 
						|
  }
 | 
						|
 | 
						|
  virtual int SetOption(SocketType type, talk_base::Socket::Option opt,
 | 
						|
                        int option) {
 | 
						|
    if (opt == talk_base::Socket::OPT_SNDBUF) {
 | 
						|
      sendbuf_size_ = option;
 | 
						|
    } else if (opt == talk_base::Socket::OPT_RCVBUF) {
 | 
						|
      recvbuf_size_ = option;
 | 
						|
    } else if (opt == talk_base::Socket::OPT_DSCP) {
 | 
						|
      dscp_ = static_cast<talk_base::DiffServCodePoint>(option);
 | 
						|
    }
 | 
						|
    return 0;
 | 
						|
  }
 | 
						|
 | 
						|
  void PostMessage(int id, const talk_base::Buffer& packet) {
 | 
						|
    thread_->Post(this, id, talk_base::WrapMessageData(packet));
 | 
						|
  }
 | 
						|
 | 
						|
  virtual void OnMessage(talk_base::Message* msg) {
 | 
						|
    talk_base::TypedMessageData<talk_base::Buffer>* msg_data =
 | 
						|
        static_cast<talk_base::TypedMessageData<talk_base::Buffer>*>(
 | 
						|
            msg->pdata);
 | 
						|
    if (dest_) {
 | 
						|
      if (msg->message_id == ST_RTP) {
 | 
						|
        dest_->OnPacketReceived(&msg_data->data(),
 | 
						|
                                talk_base::CreatePacketTime(0));
 | 
						|
      } else {
 | 
						|
        dest_->OnRtcpReceived(&msg_data->data(),
 | 
						|
                              talk_base::CreatePacketTime(0));
 | 
						|
      }
 | 
						|
    }
 | 
						|
    delete msg_data;
 | 
						|
  }
 | 
						|
 | 
						|
 private:
 | 
						|
  void GetNumRtpBytesAndPackets(uint32 ssrc, int* bytes, int* packets) {
 | 
						|
    if (bytes) {
 | 
						|
      *bytes = 0;
 | 
						|
    }
 | 
						|
    if (packets) {
 | 
						|
      *packets = 0;
 | 
						|
    }
 | 
						|
    uint32 cur_ssrc = 0;
 | 
						|
    for (size_t i = 0; i < rtp_packets_.size(); ++i) {
 | 
						|
      if (!GetRtpSsrc(rtp_packets_[i].data(),
 | 
						|
                      rtp_packets_[i].length(), &cur_ssrc)) {
 | 
						|
        return;
 | 
						|
      }
 | 
						|
      if (ssrc == cur_ssrc) {
 | 
						|
        if (bytes) {
 | 
						|
          *bytes += static_cast<int>(rtp_packets_[i].length());
 | 
						|
        }
 | 
						|
        if (packets) {
 | 
						|
          ++(*packets);
 | 
						|
        }
 | 
						|
      }
 | 
						|
    }
 | 
						|
  }
 | 
						|
 | 
						|
  talk_base::Thread* thread_;
 | 
						|
  MediaChannel* dest_;
 | 
						|
  bool conf_;
 | 
						|
  // The ssrcs used in sending out packets in conference mode.
 | 
						|
  std::vector<uint32> conf_sent_ssrcs_;
 | 
						|
  // Map to track counts of packets that have been sent per ssrc.
 | 
						|
  // This includes packets that are dropped.
 | 
						|
  std::map<uint32, uint32> sent_ssrcs_;
 | 
						|
  // Map to track packet-number that needs to be dropped per ssrc.
 | 
						|
  std::map<uint32, std::set<uint32> > drop_map_;
 | 
						|
  talk_base::CriticalSection crit_;
 | 
						|
  std::vector<talk_base::Buffer> rtp_packets_;
 | 
						|
  std::vector<talk_base::Buffer> rtcp_packets_;
 | 
						|
  int sendbuf_size_;
 | 
						|
  int recvbuf_size_;
 | 
						|
  talk_base::DiffServCodePoint dscp_;
 | 
						|
};
 | 
						|
 | 
						|
}  // namespace cricket
 | 
						|
 | 
						|
#endif  // TALK_MEDIA_BASE_FAKENETWORKINTERFACE_H_
 |