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让android支持RTSP及live555分析

2012-08-27 14:23 886 查看
#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} {

8000
.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
--------------------------------------------------------------------------------


*** END ***
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