全部博文（584）
分类： Android平台
2013-03-30 17:30:23
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):
编译时加上参数:
-fexceptions -lstdc++
还需要将libsupc++.a链接上
参考: rfc2326, rfc3550, rfc3984
RTP Header结构[#0]
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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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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synchronization source (SSRC) identifier |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
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contributing source (CSRC) identifiers |
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.... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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参考: rfc3984, 『H.264中的NAL技术』, 『H.264 NAL层解析』
参考: ISO_IEC_13818-7.pdf
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。
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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");
}
}
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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。
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------------------------------------------------------[#1]--
session
<---> client requst
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subsession
<---> audio/video
------------------------------------------------------------
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1.2 数据处理流程:
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------------------------------------------------------[#2]--
source
--> filter(source) ... --> sink
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+-------+-------------------+
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v
v
subsession.createNewRTPSink()
subsession.createNewStreamSource()
------------------------------------------------------------
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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:
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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();
}
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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：
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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);
}
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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))：
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[0]->[1]->[3]->[2]->...->[1]->NULL
^
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head()
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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：
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void DelayQueue::handleAlarm()
{
...
toRemove->handleTimeout();
}
void AlarmHandler::handleTimeout()
{
(*fProc)(fClientData);
DelayQueueEntry::handleTimeout();
}
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任务在处理完成后便会被删除。
* 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:
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------------------------------------------------------[#3]--
Medium
+ServerMediaSubsession
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+OnDemandServerMediaSubsession
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+FileServerMediaSubsession
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+H264VideoFileServerMediaSubsession //h264
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+ADTSAudioFileServerMediaSubsession //aac
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+MediaSink
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+RTPSink
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+MultiFramedRTPSink
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+VideoRTPSink
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| +H264VideoRTPSink //h264
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+MPEG4GenericRTPSink //aac
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+MediaSource
+FramedSource //+doGetNextFrame();
+fAfterGettingFunc;
+FramedFilter
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+H264FUAFragmenter //h264
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+MPEGVideoStreamFramer
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+H264VideoStreamFramer //h264
+FramedFileSource
+ByteStreamFileSource //h264
+ADTSAudioFileSource //acc
StreamParser
+MPEGVideoStreamParser
+H264VideoStreamParser //h264
------------------------------------------------------------
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我们看下FramedFilter和FramedFileSource相对于FramedSource增加了哪些成员:
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FramedFilter
{
FramedSource*
fInputSource;
}
FramedFileSource
{
FILE*
fFid;
}
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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，
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H264VideoStreamFramer.fInputSource
= ByteStreamFileSource;
H264FUAFragmenter.fInputSource
= H264VideoStreamFramer;
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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:
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ByteStreamFileSource
-> H264VideoStreamFramer -> H264FUAFragmenter -> H264VideoRTPSink
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在H264VideoStreamFramer的父类MPEGVideoStreamFramer中也有新增成员，
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MPEGVideoStreamFramer
{
MPEGVideoStreamParser*
fParser;
}
MPEGVideoStreamFramer.fParser
= H264VideoStreamParser;
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H264VideoStreamParser是用来filter过程中处理视频数据的。
在MultiFramedRTPSink::buildAndSendPacket()中添加RTP头[#0]。
3.1 H264函数调用关系
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------------------------------------------------------[#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()
------------------------------------------------------------
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4.1 H264对象关系图
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------------------------------------------------------[#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}
}
}
}
------------------------------------------------------------
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4.2 AAC对象关系图
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------------------------------------------------------[#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}
}
}
------------------------------------------------------------
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5.1 RTSP命令和处理函数的对应关系:
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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
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5.2 RTSP播放交互示例(openRTSP)
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--------------------------------------------------------------------------------
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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