让android支持RTSP及live555分析

#DATE 2012/05/09
#2012/08/27由cnblogs迁入

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 --------------------------------------------------------------------------------

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