live555在Ubuntu下的RTSP推流測試

       上一篇暫時驗證了RV1126交叉編譯live555的正確編譯,因爲本人想用RV1126做個一端推流另一端拉流的東西,但是尷尬的是手上只有一個RV1126開發板,想驗證成功了再向company申請買,因此出了這麼個想法,發送端暫時用ubuntu系統跑live555推流,這樣我可以在板子上驗證自己改寫的接受端DEMO.

       那麼開始

       新建個UbuntuTest文件夾把live555-latest.tar.gz和openssl-1.1.1v.tar.gz扔進去,隨後還是先搞openssl,解壓,進入openssl-1.1.1v,命令行,

./Configure linux-x86_64 --prefix=/home/alientek/UbuntuTest/openssl-build

       看了一眼我用的是Ubuntu 20.04.5 LTS,64位,那沒錯了,--prefix後接自己想安裝庫的文件位置,隨後成功生產個Makefile,接着

make

make test

make install

       openssl-build文件夾裏安裝好了,將他拷貝進live555解壓後的文件夾live裏,把文件夾名改了openssl,然後處理live

       修改config.linux文件

COMPILE_OPTS =		$(INCLUDES) -I/home/alientek/UbuntuTest/live/openssl/include/openssl -I. -O2 -DSOCKLEN_T=socklen_t -D_LARGEFILE_SOURCE=1 -D_FILE_OFFSET_BITS=64 -DNO_STD_LIB
C =			c
C_COMPILER =		cc
C_FLAGS =		$(COMPILE_OPTS) $(CPPFLAGS) $(CFLAGS)
CPP =			cpp
CPLUSPLUS_COMPILER =	c++
CPLUSPLUS_FLAGS =	$(COMPILE_OPTS) -Wall -DBSD=1 $(CPPFLAGS) $(CXXFLAGS)
OBJ =			o
LINK =			c++ -o
LINK_OPTS =		-L. $(LDFLAGS)
CONSOLE_LINK_OPTS =	$(LINK_OPTS)
LIBRARY_LINK =		ar cr 
LIBRARY_LINK_OPTS =	
LIB_SUFFIX =			a
LIBS_FOR_CONSOLE_APPLICATION =  -L/home/alientek/UbuntuTest/live/openssl/lib -lssl -lcrypto
LIBS_FOR_GUI_APPLICATION =
EXE =

COMPILE_OPTS和LIBS_FOR_CONSOLE_APPLICATION選項裏改寫自己的頭文件和lib的位置,COMPILE_OPTS最後加個-DNO_STD_LIB,然後命令行

./genMakefiles linux

        會生成個Makefile,繼續命令行

make

        編譯成功,找個測試程序測一下,testProgs裏有個testH264VideoStreamer

#include <liveMedia.hh>

#include <BasicUsageEnvironment.hh>
#include "announceURL.hh"
#include <GroupsockHelper.hh>

UsageEnvironment* env;
char const* inputFileName = "test.264";
H264VideoStreamFramer* videoSource;
RTPSink* videoSink;

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:
  struct sockaddr_storage destinationAddress;
  destinationAddress.ss_family = AF_INET;
  ((struct sockaddr_in&)destinationAddress).sin_addr.s_addr = chooseRandomIPv4SSMAddress(*env);
  // Note: This is a multicast address.  If you wish instead to stream
  // using unicast, then you should use the "testOnDemandRTSPServer"
  // test program - not this test program - as a model.

  const unsigned short rtpPortNum = 18888;
  const unsigned short rtcpPortNum = rtpPortNum+1;
  const unsigned char ttl = 255;

  const Port rtpPort(rtpPortNum);
  const Port rtcpPort(rtcpPortNum);

  Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
  rtpGroupsock.multicastSendOnly(); // we're a SSM source
  Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);
  rtcpGroupsock.multicastSendOnly(); // we're a SSM source

  // Create a 'H264 Video RTP' sink from the RTP 'groupsock':
  OutPacketBuffer::maxSize = 100000;
  videoSink = H264VideoRTPSink::createNew(*env, &rtpGroupsock, 96);

  // Create (and start) a 'RTCP instance' for this RTP sink:
  const unsigned estimatedSessionBandwidth = 500; // 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
  RTCPInstance* rtcp
  = RTCPInstance::createNew(*env, &rtcpGroupsock,
			    estimatedSessionBandwidth, CNAME,
			    videoSink, NULL /* we're a server */,
			    True /* we're a SSM source */);
  // Note: This starts RTCP running automatically

  RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554);
  if (rtspServer == NULL) {
    *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
    exit(1);
  }
  ServerMediaSession* sms
    = ServerMediaSession::createNew(*env, "testStream", inputFileName,
		   "Session streamed by \"testH264VideoStreamer\"",
					   True /*SSM*/);
  sms->addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, rtcp));
  rtspServer->addServerMediaSession(sms);
  announceURL(rtspServer, sms);

  // Start the streaming:
  *env << "Beginning streaming...\n";
  play();

  env->taskScheduler().doEventLoop(); // does not return

  return 0; // only to prevent compiler warning
}

void afterPlaying(void* /*clientData*/) {
  *env << "...done reading from file\n";
  videoSink->stopPlaying();
  Medium::close(videoSource);
  // Note that this also closes the input file that this source read from.

  // Start playing once again:
  play();
}

void play() {
  // Open the input file as a 'byte-stream file source':
  ByteStreamFileSource* fileSource
    = ByteStreamFileSource::createNew(*env, inputFileName);
  if (fileSource == NULL) {
    *env << "Unable to open file \"" << inputFileName
         << "\" as a byte-stream file source\n";
    exit(1);
  }

  FramedSource* videoES = fileSource;

  // Create a framer for the Video Elementary Stream:
  videoSource = H264VideoStreamFramer::createNew(*env, videoES);

  // Finally, start playing:
  *env << "Beginning to read from file...\n";
  videoSink->startPlaying(*videoSource, afterPlaying, videoSink);
}

         就他了,如果沒有test.264文件,去live555下載區下載,把testH264VideoStreamer執行程序和test.264放在一起,

./testH264VideoStreamer

         跑起來,本人非要再開個電腦搭個局域網,vlc播放器打開網絡串流

         接收到了.

         還有個testH265VideoStreamer.cpp,有空再試試.