28654cbc22
TBR=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1848004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4380 4adac7df-926f-26a2-2b94-8c16560cd09d
250 lines
7.6 KiB
C++
250 lines
7.6 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/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) {
|
|
}
|
|
|
|
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_; }
|
|
|
|
protected:
|
|
virtual bool SendPacket(talk_base::Buffer* packet) {
|
|
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::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;
|
|
}
|
|
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());
|
|
} else {
|
|
dest_->OnRtcpReceived(&msg_data->data());
|
|
}
|
|
}
|
|
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_;
|
|
};
|
|
|
|
} // namespace cricket
|
|
|
|
#endif // TALK_MEDIA_BASE_FAKENETWORKINTERFACE_H_
|