1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
|
/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 2.1 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for
more details.
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
**********/
// Copyright (c) 1996-2000, Live Networks, Inc. All rights reserved
// A test program that streams GSM audio via RTP/RTCP
// main program
// NOTE: This program assumes the existence of a (currently nonexistent)
// function called "createNewGSMAudioSource()".
#include "liveMedia.hh"
#include "GroupsockHelper.hh"
#include "BasicUsageEnvironment.hh"
////////// Main program //////////
// To stream using "source-specific multicast" (SSM), uncomment the following:
//#define USE_SSM 1
#ifdef USE_SSM
Boolean const isSSM = True;
#else
Boolean const isSSM = False;
#endif
// To set up an internal RTSP server, uncomment the following:
//#define IMPLEMENT_RTSP_SERVER 1
// (Note that this RTSP server works for multicast only)
#ifdef IMPLEMENT_RTSP_SERVER
RTSPServer* rtspServer;
#endif
UsageEnvironment* env;
void afterPlaying(void* clientData); // forward
// A structure to hold the state of the current session.
// It is used in the "afterPlaying()" function to clean up the session.
struct sessionState_t {
FramedSource* source;
RTPSink* sink;
RTCPInstance* rtcpInstance;
Groupsock* rtpGroupsock;
Groupsock* rtcpGroupsock;
} sessionState;
void play(); // forward
int main(int argc, char** argv) {
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
// Create 'groupsocks' for RTP and RTCP:
char* destinationAddressStr
#ifdef USE_SSM
= "232.255.42.42";
#else
= "239.255.42.42";
// Note: This is a multicast address. If you wish to stream using
// unicast instead, then replace this string with the unicast address
// of the (single) destination. (You may also need to make a similar
// change to the receiver program.)
#endif
const unsigned short rtpPortNum = 6666;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 1; // low, in case routers don't admin scope
struct in_addr destinationAddress;
destinationAddress.s_addr = our_inet_addr(destinationAddressStr);
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
sessionState.rtpGroupsock
= new Groupsock(*env, destinationAddress, rtpPort, ttl);
sessionState.rtcpGroupsock
= new Groupsock(*env, destinationAddress, rtcpPort, ttl);
#ifdef USE_SSM
sessionState.rtpGroupsock->multicastSendOnly();
sessionState.rtcpGroupsock->multicastSendOnly();
#endif
// Create a 'GSM RTP' sink from the RTP 'groupsock':
sessionState.sink
= GSMAudioRTPSink::createNew(*env, sessionState.rtpGroupsock);
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = 160; // in kbps; for RTCP b/w share
const unsigned maxCNAMElen = 100;
unsigned char CNAME[maxCNAMElen+1];
gethostname((char*)CNAME, maxCNAMElen);
CNAME[maxCNAMElen] = '\0'; // just in case
sessionState.rtcpInstance
= RTCPInstance::createNew(*env, sessionState.rtcpGroupsock,
estimatedSessionBandwidth, CNAME,
sessionState.sink, NULL /* we're a server */,
isSSM);
// Note: This starts RTCP running automatically
#ifdef IMPLEMENT_RTSP_SERVER
rtspServer = RTSPServer::createNew(*env, 8554);
if (rtspServer == NULL) {
*env << "Failed to create RTSP server: " << env->getResultMsg() << "%s\n";
exit(1);
}
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, "testStream", "GSM input",
"Session streamed by \"testGSMStreamer\"", isSSM);
sms->addSubsession(PassiveServerMediaSubsession::createNew(*sessionState.sink, sessionState.rtcpInstance));
rtspServer->addServerMediaSession(sms);
char* url = rtspServer->rtspURL(sms);
*env << "Play this stream using the URL \"" << url << "\"\n";
delete[] url;
#endif
play();
env->taskScheduler().doEventLoop(); // does not return
return 0; // only to prevent compiler warning
}
void play() {
// Open the input source:
extern FramedSource* createNewGSMAudioSource(UsageEnvironment&);
sessionState.source = createNewGSMAudioSource(*env);
if (sessionState.source == NULL) {
*env << "Failed to create GSM source\n";
exit(1);
}
// Finally, start the streaming:
*env << "Beginning streaming...\n";
sessionState.sink->startPlaying(*sessionState.source, afterPlaying, NULL);
}
void afterPlaying(void* /*clientData*/) {
*env << "...done streaming\n";
// End this loop by closing the media:
#ifdef IMPLEMENT_RTSP_SERVER
Medium::close(rtspServer);
#endif
Medium::close(sessionState.rtcpInstance);
Medium::close(sessionState.sink);
delete sessionState.rtpGroupsock;
Medium::close(sessionState.source);
delete sessionState.rtcpGroupsock;
// And start another loop:
play();
}
|
