live555 基本套路

2017/06/05 | live555 | justchen | 暂无评论 | 9338 views

点播模式

TaskScheduler* scheduler = BasicTaskScheduler::createNew();
BasicUsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
RTSPServer* rtspServer = RTSPServer::createNew(*env);
ServerMediaSession* sms = ServerMediaSession::createNew(*env);
sms->addSubsession(H264VideoFileServerMediaSubsession::createNew(*env, "test.264",false));
sms->addSubsession(ADTSAudioFileServerMediaSubsession::createNew(*env, "test.aac",false));
rtspServer->addServerMediaSession(sms);

1

2

3

4

5

6

7

8

TaskScheduler* scheduler = BasicTaskScheduler::createNew();

BasicUsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);

RTSPServer* rtspServer = RTSPServer::createNew(*env);

ServerMediaSession* sms = ServerMediaSession::createNew(*env);

sms->addSubsession(H264VideoFileServerMediaSubsession::createNew(*env, "test.264",false));

sms->addSubsession(ADTSAudioFileServerMediaSubsession::createNew(*env, "test.aac",false));

rtspServer->addServerMediaSession(sms);

组播模式

OutPacketBuffer::maxSize = xxx;
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
BasicUsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);

Groupsock *rtpGroupsock = new Groupsock(*env, destinationAddress, rtpPort, ttl);
Groupsock *rtcpGroupsock = new Groupsock(*env, destinationAddress, rtcpPort, ttl);
rtpGroupsock->multicastSendOnly();
rtcpGroupsock->multicastSendOnly();

RTPSink* rtpSink = H264VideoRTPSink::createNew(*env, rtpGroupsock, 96);   // 针对H264
RTCPInstance* rtcp = RTCPInstance::createNew(*env, &rtcpGroupsock,500/*码率 Kbps*/, CNAME/*HOST NAME*/,rtpSink, NULL /* we're a server */,True /* we're a SSM source */);
RTSPServer* rtspServer = RTSPServer::createNew(*env, 端口号,authDB(没有则NULL));
        if (rtspServer == NULL) {
            *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
            exit(1);
        }
sms = ServerMediaSession::createNew(*env,"流名称", NULL, "流描述", True /*SSM*/);
sms->addSubsession(PassiveServerMediaSubsession::createNew(rtpSink, rtcp));
rtspServer->addServerMediaSession(sms);

// === 最主要的一步,数据源获取
FramedSource* liveSource = JappByteStreamLiveSource::createNew(*env, NULL, 0,this);
H264VideoStreamDiscreteFramer* source = H264VideoStreamDiscreteFramer::createNew(*env, liveSource);

rtpSink->startPlaying(*source, NULL, rtpSink);

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

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

OutPacketBuffer::maxSize = xxx;

TaskScheduler* scheduler = BasicTaskScheduler::createNew();

BasicUsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);

Groupsock *rtpGroupsock = new Groupsock(*env, destinationAddress, rtpPort, ttl);

Groupsock *rtcpGroupsock = new Groupsock(*env, destinationAddress, rtcpPort, ttl);

rtpGroupsock->multicastSendOnly();

rtcpGroupsock->multicastSendOnly();

RTPSink* rtpSink = H264VideoRTPSink::createNew(*env, rtpGroupsock, 96); // 针对H264

RTCPInstance* rtcp = RTCPInstance::createNew(*env, &rtcpGroupsock,500/*码率 Kbps*/, CNAME/*HOST NAME*/,rtpSink, NULL /* we're a server */,True /* we're a SSM source */);

RTSPServer* rtspServer = RTSPServer::createNew(*env, 端口号,authDB(没有则NULL));

if (rtspServer == NULL) {

*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";

exit(1);

}

sms = ServerMediaSession::createNew(*env,"流名称", NULL, "流描述", True /*SSM*/);

sms->addSubsession(PassiveServerMediaSubsession::createNew(rtpSink, rtcp));

rtspServer->addServerMediaSession(sms);

// === 最主要的一步,数据源获取

FramedSource* liveSource = JappByteStreamLiveSource::createNew(*env, NULL, 0,this);

H264VideoStreamDiscreteFramer* source = H264VideoStreamDiscreteFramer::createNew(*env, liveSource);

rtpSink->startPlaying(*source, NULL, rtpSink);

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

live555 socket buffer修改

2017/06/01 | live555,嵌入式笔记,流媒体 | justchen | 暂无评论 | 12529 views

点播 LIVE SERVER 虽然默认 50K按理说是够的,但实际上遇到大码流时依然会丢包,并不是网络问题,而是socket buff不够大. 从网上得到的信息为修改increaseSendBufferTo 它的参数, 不过有很多地方调用它,主要是修改liveMedia\RTPInterface.cpp中的increaseSendBuff……

ITU709 ITU601 简单说明

2017/01/11 | 流媒体 | justchen | 暂无评论 | 13522 views

如果数据位宽都以8位来说. ITU709:允许 0~255之间所有数据 ITU601:只允许 16~235之间数据,因为 0 FF是用来标记同步的

组播之SSM ASM

2016/10/20 | live555,流媒体 | justchen | 暂无评论 | 24669 views

在live555中有这样一段代码:

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

1

2

3

4

5

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

这里的SSM是指Source-Specific Multicast，指定信源组播. 除SSM之外,组播类型还有ASM（Any-Source Multicast，任意信源组播）模型。其中，ASM模型包括PIM-DM和PIM-SM两种组播业务……

修改testRTSPClient.cpp 为保存h264文件

2016/02/26 | live555 | justchen | 暂无评论 | 13255 views

live555源码中,testRTSPClient.cpp默认只打印下接收的信息就收工了. 需要修改DummySink::afterGettingFrame,在其中增加00 00 00 01 以及获取sps pps添加到相应位置

unsigned char *pH264 = NULL;    // 数据buffer

void DummySink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
      struct timeval presentationTime, unsigned /*durationInMicroseconds*/) {
 printf("__FUNCTION__  = %s\n", __FUNCTION__ );
 if(0 == strncmp(fSubsession.codecName(), "H264", 16))
 {
  unsigned char nalu_header[4] = { 0, 0, 0, 1 };
  unsigned char extraData[256];
  unsigned int num = 0;

  pH264 = (unsigned char*)malloc(256*1024);

  SPropRecord *pSPropRecord;
  pSPropRecord = parseSPropParameterSets(fSubsession.fmtp_spropparametersets(), num);

  unsigned int extraLen;
  extraLen = 0;

  //p_record[0] is sps
  //p+record[1] is pps
  for(unsigned int i = 0; i < num; i++){
   memcpy(&extraData[extraLen], &nalu_header[0], 4);
   extraLen += 4;
   memcpy(&extraData[extraLen], pSPropRecord[i].sPropBytes, pSPropRecord[i].sPropLength);
   extraLen += pSPropRecord[i].sPropLength;
  }/*for i*/

  memcpy(&extraData[extraLen], &nalu_header[0], 4);
  extraLen += 4;

  delete[] pSPropRecord ;

  memcpy(pH264, &extraData[0], extraLen);
  memcpy(pH264 + extraLen, fReceiveBuffer, frameSize);
  
  int totalSize;
  totalSize = extraLen + frameSize;

  static FILE *fp = fopen("saved.h264", "wb");

  fwrite(pH264, 1,  totalSize, fp);
  fflush(fp);
  printf("\tsaved %d bytes\n", totalSize);
  free(pH264);
 }/*if 0 == strncmp(fSubsession.codecName(), "H264", 16)*/
  // Then continue, to request the next frame of data:
  continuePlaying();
}

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

unsigned char *pH264 = NULL; // 数据buffer

void DummySink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,

struct timeval presentationTime, unsigned /*durationInMicroseconds*/) {

printf("__FUNCTION__ = %s\n", __FUNCTION__ );

if(0 == strncmp(fSubsession.codecName(), "H264", 16))

{

unsigned char nalu_header[4] = { 0, 0, 0, 1 };

unsigned char extraData[256];

unsigned int num = 0;

pH264 = (unsigned char*)malloc(256*1024);

SPropRecord *pSPropRecord;

pSPropRecord = parseSPropParameterSets(fSubsession.fmtp_spropparametersets(), num);

unsigned int extraLen;

extraLen = 0;

//p_record[0] is sps

//p+record[1] is pps

for(unsigned int i = 0; i < num; i++){

memcpy(&extraData[extraLen], &nalu_header[0], 4);

extraLen += 4;

memcpy(&extraData[extraLen], pSPropRecord[i].sPropBytes, pSPropRecord[i].sPropLength);

extraLen += pSPropRecord[i].sPropLength;

}/*for i*/

memcpy(&extraData[extraLen], &nalu_header[0], 4);

extraLen += 4;

delete[] pSPropRecord ;

memcpy(pH264, &extraData[0], extraLen);

memcpy(pH264 + extraLen, fReceiveBuffer, frameSize);

int totalSize;

totalSize = extraLen + frameSize;

static FILE *fp = fopen("saved.h264", "wb");

fwrite(pH264, 1, totalSize, fp);

fflush(fp);

printf("\tsaved %d bytes\n", totalSize);

free(pH264);

}/*if 0 == strncmp(fSubsession.codecName(), "H264", 16)*/

// Then continue, to request the next frame of data:

continuePlaying();

}

live555海思编译参数

2016/02/23 | live555,嵌入式笔记 | justchen | 暂无评论 | 10399 views

创建并编辑config.hisiv100文件如下:

CROSS_COMPILE?=     arm-hisiv100nptl-linux-
COMPILE_OPTS =      $(INCLUDES) -I. -O2 -DSOCKLEN_T=socklen_t -DNO_SSTREAM=1 -D_LARGEFILE_SOURCE=1 -D_FILE_OFFSET_BITS=64 -DLOCALE_NOT_USED
C =         c
C_COMPILER =        $(CROSS_COMPILE)gcc
C_FLAGS =       $(COMPILE_OPTS)
CPP =           cpp
CPLUSPLUS_COMPILER =    $(CROSS_COMPILE)g++
CPLUSPLUS_FLAGS =   $(COMPILE_OPTS) -Wall -DBSD=1
OBJ =           o
LINK =          $(CROSS_COMPILE)g++ -o
LINK_OPTS =     
CONSOLE_LINK_OPTS = $(LINK_OPTS)
LIBRARY_LINK =      $(CROSS_COMPILE)ar cr 
LIBRARY_LINK_OPTS = $(LINK_OPTS)
LIB_SUFFIX =            a
LIBS_FOR_CONSOLE_APPLICATION =
LIBS_FOR_GUI_APPLICATION =
EXE =

1

2

3

4

5

6

7

8

9

10

11

12

13

14

15

16

17

18

19

CROSS_COMPILE?= arm-hisiv100nptl-linux-

COMPILE_OPTS = $(INCLUDES) -I. -O2 -DSOCKLEN_T=socklen_t -DNO_SSTREAM=1 -D_LARGEFILE_SOURCE=1 -D_FILE_OFFSET_BITS=64 -DLOCALE_NOT_USED

C = c

C_COMPILER = $(CROSS_COMPILE)gcc

C_FLAGS = $(COMPILE_OPTS)

CPP = cpp

CPLUSPLUS_COMPILER = $(CROSS_COMPILE)g++

CPLUSPLUS_FLAGS = $(COMPILE_OPTS) -Wall -DBSD=1

OBJ = o

LINK = $(CROSS_COMPILE)g++ -o

LINK_OPTS =

CONSOLE_LINK_OPTS = $(LINK_OPTS)

LIBRARY_LINK = $(CROSS_COMPILE)ar cr

LIBRARY_LINK_OPTS = $(LINK_OPTS)

LIB_SUFFIX = a

LIBS_FOR_CONSOLE_APPLICATION =

LIBS_FOR_GUI_APPLICATION =

EXE =

在修改config文件中我是直接拷贝config.amelinux那个文件,编译会出现include/Locale.hh:47: fatal error: xlocale.h: No such file or direc……

网网'Blog

分类目录：流媒体