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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
https://github.com/boltonli/ohbee/tree/master/android/streamer/jni
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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| contributing source (CSRC) identifiers |
| .... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
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"
);
}
}
|
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
|
subsession <---> audio/video
------------------------------------------------------------
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1.2 数据处理流程:
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------------------------------------------------------[#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:
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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();
}
|
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
);
}
|
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
^
|
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:
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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。
* 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
| +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:
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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,
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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;
|
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()
------------------------------------------------------------
|
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}
}
}
}
------------------------------------------------------------
|
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
|
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
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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