[原创]让android支持RTSP(live555分析)

如何让Android支持C++异常机制

Android不支持C++异常机制,如果需要用到的话,则需要在编译的时候加入比较完整的C++库.
Android支持的C++库可以在Android NDK中找到(解压后找到libsupc++.a放到代码环境中即可):
http://www.crystax.net/en/android/ndk/7
编译时加上参数:
-fexceptions -lstdc++
还需要将libsupc++.a链接上

 

移植live555到Android的例子

https://github.com/boltonli/ohbee/tree/master/android/streamer/jni

 

RTSP协议

参考: rfc2326, rfc3550, rfc3984

RTP Header结构[#0]
 

0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X|  CC   |M|     PT      |       sequence number         |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                           timestamp                           |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|           synchronization source (SSRC) identifier            |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
|            contributing source (CSRC) identifiers             |
|                             ....                              |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

 

H.264视频格式

参考: rfc3984, 『H.264中的NAL技术』, 『H.264 NAL层解析』

 

ACC音频格式

参考: ISO_IEC_13818-7.pdf

 

live555架构分析

0  总述

0.1  这里主要以H264+ACC为基础作介绍

0.2  live555中的demo说明,RTSP服务端为live555MediaServer,openRTSP为调试用客户端。

0.3  可以在live555中实现一个trace_bin的函数跟踪流媒体数据的处理过程。

    void trace_bin(const unsigned char *bytes_ptr, int bytes_num)
    {
        #define LOG_LINE_BYTES 16
        int i, j;

        for (i = 0; i <= bytes_num / LOG_LINE_BYTES; i++) {
            for (j = 0;
                 j < ( (i < (bytes_num / LOG_LINE_BYTES))
                       ? LOG_LINE_BYTES
                       : (bytes_num % LOG_LINE_BYTES) );
                 j++)
            {
                if (0 == j) printf("%04d   ", i * LOG_LINE_BYTES);
                if (LOG_LINE_BYTES/2 == j) printf("   ");
                printf(" %02x", bytes_ptr[i * LOG_LINE_BYTES + j]);
            }
            printf("\n");
        }
    }

 

1  宏观流程

1.1  对每个播放请求建立一个session,并对应音视频建立subsession,subsession则是具体处理流媒体的单位。

------------------------------------------------------[#1]--
session     <--->  client requst
   |
subsession  <--->  audio/video
------------------------------------------------------------


1.2  数据处理流程:

------------------------------------------------------[#2]--
source --> filter(source) ... --> sink
|                           |      |
+-------+-------------------+      |
        |                          v
        v                    subsession.createNewRTPSink()
subsession.createNewStreamSource()
------------------------------------------------------------


1.3  BasicTaskScheduler::SingleStep()
BasicTaskScheduler是live555的任务处理器,他的主要工作都是在SingleStep()中完成的.
在SingleStep()中主要完成下面三种工作:

    void BasicTaskScheduler::SingleStep(unsigned maxDelayTime) {
        //1. 处理io任务
        ...
        int selectResult = select(fMaxNumSockets, &readSet, &writeSet,
                                  &exceptionSet, &tv_timeToDelay);
        ...
        while ((handler = iter.next()) != NULL) {
            ...
            (*handler->handlerProc)(handler->clientData, resultConditionSet);
            break;
        }
        ...
        
        //2. handle any newly-triggered event
        ...
        
        //3. handle any delayed event
        fDelayQueue.handleAlarm();
    }


RTSP请求、链接建立、开始播放处理主要是在1中完成的,而视频播放主要是在3中完成。
以ACC播放为例:

    void ADTSAudioFileSource::doGetNextFrame() {
        // 读取数据并做一些简单处理
        ...
        int numBytesRead = fread(fTo, 1, numBytesToRead, fFid);
        ...

        // 将FramedSource::afterGetting加入fDelayQueue中
        // FramedSource::afterGetting会处理读取到数据,并又会调用
        // ADTSAudioFileSource::doGetNextFrame(),这样实现循环读取文件。
        nextTask() = envir().taskScheduler().scheduleDelayedTask(0,
                    (TaskFunc*)FramedSource::afterGetting, this);
    }


1.4  DelayQueue
fDelayQueue是一个需要处理的任务的队列,每次SingleStep()只会执行第一个任务head(),这里的任务对应DelayQueue的元素,DelayQueue的各个元素都会有自己的DelayTime,用来表示延时多久后执行。而队列中的元素便是按照DelayTime有小到大排列的,元素中fDeltaTimeRemaining记录的是该元素相对于它之前元素的延时。参照函数DelayQueue::addEntry()便可看出是如何入队列的。
例如([]中的数字便是相对延时(fDeltaTimeRemaining)):

    [0]->[1]->[3]->[2]->...->[1]->NULL
     ^
     |
    head()

在处理DelayQueue时往往都要先做一次计时同步操作synchronize(),因为DelayQueue中元素的延时都是相对的,所以一般只要处理首元素即可,不过如果同步之后延时有小于0的,便都会改为DELAY_ZERO(即表示需要立即执行的)。
执行任务:

    void DelayQueue::handleAlarm() {
        ...
        toRemove->handleTimeout();
    }

    void AlarmHandler::handleTimeout() {
        (*fProc)(fClientData);
        DelayQueueEntry::handleTimeout();
    }


任务在处理完成后便会被删除。

 

2  类关系

   * live555的流程分析主要就放在这个章节中,如果有需要参考函数关系或者对象关系的请参考3, 4两个章节。
2.1  涉及到的主要类的关系图:

------------------------------------------------------[#3]--
Medium
  +ServerMediaSubsession
  |  +OnDemandServerMediaSubsession
  |     +FileServerMediaSubsession
  |        +H264VideoFileServerMediaSubsession      //h264
  |        +ADTSAudioFileServerMediaSubsession      //aac
  |
  +MediaSink
  |  +RTPSink
  |     +MultiFramedRTPSink
  |        +VideoRTPSink
  |        |  +H264VideoRTPSink                     //h264
  |        +MPEG4GenericRTPSink                     //aac
  |
  +MediaSource
     +FramedSource      //+doGetNextFrame(); +fAfterGettingFunc;
        +FramedFilter
        |  +H264FUAFragmenter                       //h264
        |  +MPEGVideoStreamFramer
        |     +H264VideoStreamFramer                //h264
        +FramedFileSource
           +ByteStreamFileSource                    //h264
           +ADTSAudioFileSource                     //acc

StreamParser
  +MPEGVideoStreamParser
     +H264VideoStreamParser                         //h264
------------------------------------------------------------

我们看下FramedFilter和FramedFileSource相对于FramedSource增加了哪些成员:

    FramedFilter {
        FramedSource* fInputSource;
    }

    FramedFileSource {
        FILE* fFid;
    }

从两者的命名和增加的成员可以看出各自的作用。FramedFilter便是对应着[#2]中的filter,而FramedFileSource则是以本地文件为输入的source。

2.2  如何实现带有filter流程:
这便用到了FramedFilter中的fInputSource成员,以H264为例,

    H264VideoStreamFramer.fInputSource = ByteStreamFileSource; 
    H264FUAFragmenter.fInputSource = H264VideoStreamFramer;

将上游source赋值到下游filter的fInputSource即可,对于H264便可以得到下面的一个处理流程:

    ByteStreamFileSource -> H264VideoStreamFramer -> H264FUAFragmenter -> H264VideoRTPSink

在H264VideoStreamFramer的父类MPEGVideoStreamFramer中也有新增成员,

    MPEGVideoStreamFramer { 
        MPEGVideoStreamParser* fParser; 
    }

    MPEGVideoStreamFramer.fParser = H264VideoStreamParser;

H264VideoStreamParser是用来filter过程中处理视频数据的。

在MultiFramedRTPSink::buildAndSendPacket()中添加RTP头[#0]。

 

3  函数关系

3.1  H264函数调用关系

------------------------------------------------------[#4]-- 
RTSPServer::RTSPClientSession::handleCmd_SETUP() 
OnDemandServerMediaSubsession::getStreamParameters( 
    streamToken: new StreamState(
        fMediaSource: H264VideoFileServerMediaSubsession::createNewStreamSource() ) 
)

********** 
RTSPServer::RTSPClientSession::handleCmd_DESCRIBE() 
ServerMediaSession::generateSDPDescription() 
OnDemandServerMediaSubsession::sdpLines() 
H264VideoFileServerMediaSubsession::createNewStreamSource() 
H264VideoStreamFramer::createNew( fInputSource: ByteStreamFileSource::createNew(), 
                                  fParser: new H264VideoStreamParser(
                                      fInputSource: H264VideoStreamFramer.fInputSource) )

********** 
RTSPServer::RTSPClientSession::handleCmd_PLAY() 
H264VideoFileServerMediaSubsession::startStream() [OnDemandServerMediaSubsession::startStream()] 
StreamState::startPlaying() 
H264VideoRTPSink::startPlaying() [MediaSink::startPlaying(fMediaSource)]//got in handleCmd_SETUP()
H264VideoRTPSink::continuePlaying() 
    fSource, fOurFragmenter: H264FUAFragmenter(fInputSource: fMediaSource) 
MultiFramedRTPSink::continuePlaying() 
MultiFramedRTPSink::buildAndSendPacket() 
MultiFramedRTPSink::packFrame() 
H264FUAFragmenter::getNextFrame() [FramedSource::getNextFrame()] 
H264FUAFragmenter::doGetNextFrame() {1} 
1)=No NALU= 
  H264VideoStreamFramer::getNextFrame() [FramedSource::getNextFrame()] 
  MPEGVideoStreamFramer::doGetNextFrame() 
  H264VideoStreamParser::registerReadInterest() 
  MPEGVideoStreamFramer::continueReadProcessing() 
  H264VideoStreamParser::parse() 
  H264VideoStreamFramer::afterGetting() [FramedSource::afterGetting()] 
  H264FUAFragmenter::afterGettingFrame() 
  H264FUAFragmenter::afterGettingFrame1() 
  goto {1}  //Now we have got NALU 
2)=Has NALU= 
  FramedSource::afterGetting() 
  MultiFramedRTPSink::afterGettingFrame() 
  MultiFramedRTPSink::afterGettingFrame1() 
  MultiFramedRTPSink::sendPacketIfNecessary() 
------------------------------------------------------------

4  对象关系

4.1  H264对象关系图

------------------------------------------------------[#5]-- 
ServerMediaSession{#1}.fSubsessionsTail = H264VideoFileServerMediaSubsession{2}.fParentSession = {1} 
fStreamStates[] { 
  .subsession  = {2} 
  .streamToken = StreamState { 
                   .fMaster = {2} 
                   .fRTPSink = H264VideoRTPSink{5}.fSource/fOurFragmenter 
                             = H264FUAFragmenter{4} { 
                                 .fInputSource = H264VideoStreamFramer{3} 
                                 .fAfterGettingFunc = MultiFramedRTPSink::afterGettingFrame() 
                                 .fAfterGettingClientData = {5} 
                                 .fOnCloseFunc = MultiFramedRTPSink::ourHandleClosure() 
                                 .fOnCloseClientData = {5} 
                               } 
                   .fMediaSource = {3} { 
                                     .fParser = H264VideoStreamParser { 
                                                  .fInputSource = ByteStreamFileSource{6} 
                                                  .fTo = [{5}.]fOutBuf->curPtr() 
                                                } 
                                     .fInputSource = {6} 
                                     .fAfterGettingFunc = H264FUAFragmenter::afterGettingFrame() 
                                     .fAfterGettingClientData = {4} 
                                     .fOnCloseFunc = FramedSource::handleClosure() 
                                     .fOnCloseClientData = {4} 
                                   } 
                 } 
} 
------------------------------------------------------------


4.2  AAC对象关系图

------------------------------------------------------[#6]-- 
ServerMediaSession{1}.fSubsessionsTail = ADTSAudioFileServerMediaSubsession{2}.fParentSession = {1} 
fStreamStates[] { 
  .subsession  = {2} 
  .streamToken = StreamState { 
                   .fMaster = {2} 
                   .fRTPSink = MPEG4GenericRTPSink { 
                                 .fOutBuf = OutPacketBuffer 
                                 .fSource = ADTSAudioFileSource {3} 
                                 .fRTPInterface = RTPInterface.fGS = Groupsock 
                               } 
                   .fMediaSource = {3} 
                 } 
} 
------------------------------------------------------------

 

5  RTSP

5.1  RTSP命令和处理函数的对应关系:

    RTSP命令               live555中处理函数 
    --------------------------------------------- 
    OPTIONS        <--->  handleCmd_OPTIONS 
    DESCRIBE       <--->  handleCmd_DESCRIBE 
    SETUP          <--->  handleCmd_SETUP 
    PLAY           <--->  handleCmd_PLAY 
    PAUSE          <--->  handleCmd_PAUSE 
    TEARDOWN       <--->  handleCmd_TEARDOWN 
    GET_PARAMETER  <--->  handleCmd_GET_PARAMETER 
    SET_PARAMETER  <--->  handleCmd_SET_PARAMETER


5.2  RTSP播放交互示例(openRTSP)

-------------------------------------------------------------------------------- 
ubuntu$ ./openRTSP rtsp://192.168.43.1/grandma.264 
Opening connection to 192.168.43.1, port 554... 
...remote connection opened 
Sending request: OPTIONS rtsp://192.168.43.1/grandma.264 RTSP/1.0 
CSeq: 2 
User-Agent: ./openRTSP (LIVE555 Streaming Media v2012.02.29)


Received 152 new bytes of response data. 
Received a complete OPTIONS response: 
RTSP/1.0 200 OK 
CSeq: 2 
Date: Tue, Jan 25 2011 21:02:53 GMT 
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, GET_PARAMETER, SET_PARAMETER

-------------------------------------------------------------------------------- 
Sending request: DESCRIBE rtsp://192.168.43.1/grandma.264 RTSP/1.0 
CSeq: 3 
User-Agent: ./openRTSP (LIVE555 Streaming Media v2012.02.29) 
Accept: application/sdp


Received 682 new bytes of response data. 
Received a complete DESCRIBE response: 
RTSP/1.0 200 OK 
CSeq: 3 
Date: Tue, Jan 25 2011 21:02:53 GMT 
Content-Base: rtsp://192.168.43.1/grandma.264/ 
Content-Type: application/sdp 
Content-Length: 517

v=0 
o=- 1295989373493698 1 IN IP4 0.0.0.0 
s=H.264 Video, streamed by the LIVE555 Media Server 
i=grandma.264 
t=0 0 
a=tool:LIVE555 Streaming Media v2012.02.04 
a=type:broadcast 
a=control:* 
a=range:npt=0- 
a=x-qt-text-nam:H.264 Video, streamed by the LIVE555 Media Server 
a=x-qt-text-inf:grandma.264 
m=video 0 RTP/AVP 96 
c=IN IP4 0.0.0.0 
b=AS:500 
a=rtpmap:96 H264/90000 
a=fmtp:96 packetization-mode=1;profile-level-id=4D4033;
sprop-parameter-sets=Z01AM5p0FidCAAADAAIAAAMAZR4wZUA=,aO48gA== 
a=control:track1

Opened URL "rtsp://192.168.43.1/grandma.264", returning a SDP description: 
v=0 
o=- 1295989373493698 1 IN IP4 0.0.0.0 
s=H.264 Video, streamed by the LIVE555 Media Server 
i=grandma.264 
t=0 0 
a=tool:LIVE555 Streaming Media v2012.02.04 
a=type:broadcast 
a=control:* 
a=range:npt=0- 
a=x-qt-text-nam:H.264 Video, streamed by the LIVE555 Media Server 
a=x-qt-text-inf:grandma.264 
m=video 0 RTP/AVP 96 
c=IN IP4 0.0.0.0 
b=AS:500 
a=rtpmap:96 H264/90000 
a=fmtp:96 packetization-mode=1;profile-level-id=4D4033;
sprop-parameter-sets=Z01AM5p0FidCAAADAAIAAAMAZR4wZUA=,aO48gA== 
a=control:track1

Created receiver for "video/H264" subsession (client ports 56488-56489)

-------------------------------------------------------------------------------- 
Sending request: SETUP rtsp://192.168.43.1/grandma.264/track1 RTSP/1.0 
CSeq: 4 
User-Agent: ./openRTSP (LIVE555 Streaming Media v2012.02.29) 
Transport: RTP/AVP;unicast;client_port=56488-56489


Received 205 new bytes of response data. 
Received a complete SETUP response: 
RTSP/1.0 200 OK 
CSeq: 4 
Date: Tue, Jan 25 2011 21:02:53 GMT 
Transport: RTP/AVP;unicast;destination=192.168.43.244;source=192.168.43.1;
client_port=56488-56489;server_port=6970-6971 
Session: 7626020D


Setup "video/H264" subsession (client ports 56488-56489) 
Created output file: "video-H264-1"

-------------------------------------------------------------------------------- 
Sending request: PLAY rtsp://192.168.43.1/grandma.264/ RTSP/1.0 
CSeq: 5 
User-Agent: ./openRTSP (LIVE555 Streaming Media v2012.02.29) 
Session: 7626020D 
Range: npt=0.000-


Received 186 new bytes of response data. 
Received a complete PLAY response: 
RTSP/1.0 200 OK 
CSeq: 5 
Date: Tue, Jan 25 2011 21:02:53 GMT 
Range: npt=0.000- 
Session: 7626020D 
RTP-Info: url=rtsp://192.168.43.1/grandma.264/track1;seq=26490;rtptime=1809652062


Started playing session 
Receiving streamed data (signal with "kill -HUP 6297" or "kill -USR1 6297" to terminate)... 
Received RTCP "BYE" on "video/H264" subsession (after 35 seconds)

-------------------------------------------------------------------------------- 
Sending request: TEARDOWN rtsp://192.168.43.1/grandma.264/ RTSP/1.0 
CSeq: 6 
User-Agent: ./openRTSP (LIVE555 Streaming Media v2012.02.29) 
Session: 7626020D


Received 65 new bytes of response data. 
Received a complete TEARDOWN response: 
RTSP/1.0 200 OK 
CSeq: 6 
Date: Tue, Jan 25 2011 21:03:28 GMT 
--------------------------------------------------------------------------------
posted @ 2012-05-09 23:56  mr. dave  阅读(28753)  评论(3编辑  收藏  举报