android多媒体框架之流媒体具体流程篇3----base on jellybean(十三)

来源：互联网发布：股票价格统计软件编辑：程序博客网时间：2024/05/23 10:20

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上一篇我们讲到了从web server 中获取了sessiondescription，并解析出了media server的路径和一些基本的媒体信息。下面我们开始讲述如何跟mediaserver建立连接并控制服务器端和客户端以达到播放，暂停，停止的目的。

首先跟media server建立连接 SETUP：

具体的格式如下（UDP）：

C->A（audio）: SETUPrtsp://audio.com/twister/audio.en RTSP/1.0

CSeq: 1

Transport:RTP/AVP/UDP;unicast

;client_port=3056-3057

具体到代码的话，我们看myHandler.h中的setupTrack函数：

void setupTrack(size_t index) {

sp<APacketSource> source =

new APacketSource(mSessionDesc,index);

……………………….

AString url;

CHECK(mSessionDesc->findAttribute(index,"a=control", &url));

AString trackURL;

CHECK(MakeURL(mBaseURL.c_str(),url.c_str(), &trackURL));----检查session description中取出media server的URL是否正确

…………

AString request= "SETUP ";

request.append(trackURL);

request.append("RTSP/1.0\r\n");------拼接request字符

选择TCP连接还是ARTP连接，

if (mTryTCPInterleaving) {

size_t interleaveIndex = 2 *(mTracks.size() - 1);

info->mUsingInterleavedTCP =true;

info->mRTPSocket =interleaveIndex;

info->mRTCPSocket =interleaveIndex + 1;

request.append("Transport: RTP/AVP/TCP;interleaved=");

request.append(interleaveIndex);

request.append("-");

request.append(interleaveIndex + 1);

} else {

unsigned rtpPort;

ARTPConnection::MakePortPair(

&info->mRTPSocket,&info->mRTCPSocket, &rtpPort);

if (mUIDValid) {

HTTPBase::RegisterSocketUserTag(info->mRTPSocket, mUID,

(uint32_t)*(uint32_t*) "RTP_");

HTTPBase::RegisterSocketUserTag(info->mRTCPSocket, mUID,

(uint32_t)*(uint32_t*)"RTP_");

}

request.append("Transport:RTP/AVP/UDP;unicast;client_port=");

request.append(rtpPort);

request.append("-");

request.append(rtpPort+ 1);

}

request.append("\r\n");

if (index > 1) {

request.append("Session:");

request.append(mSessionID);

request.append("\r\n");

}

request.append("\r\n");

sp<AMessage> reply = newAMessage('setu', id());

reply->setSize("index",index);

reply->setSize("track-index", mTracks.size() - 1);

mConn->sendRequest(request.c_str(),reply);-----发送给服务器端，等待回复，返回的Amessage是“setu”

}

假设收到服务端的连接成功的消息，我们看看myHandler.h中onMessageReceived对应的”setu”如何处理，按道理应该回复回来的信息如下（UDP）：

A->C: RTSP/1.0200 OK

CSeq: 1

Session: 12345678

Transport:RTP/AVP/UDP;unicast

;client_port=3056-3057;

;server_port=5000-5001

virtualvoid onMessageReceived(const sp<AMessage> &msg) {

……

case 'setu':

{

……………………….

int32_t result;

CHECK(msg->findInt32("result", &result));

ALOGI("SETUP(%d) completedwith result %d (%s)",

index, result,strerror(-result));

if (result == OK) {

CHECK(track != NULL);

sp<RefBase> obj;

CHECK(msg->findObject("response",&obj));

sp<ARTSPResponse>response =

static_cast<ARTSPResponse *>(obj.get());

if(response->mStatusCode != 200) {

result = UNKNOWN_ERROR;

} else {

ssize_t i = response->mHeaders.indexOfKey("session");-------查找session id

CHECK_GE(i, 0);

mSessionID = response->mHeaders.valueAt(i);

………………………..

i =mSessionID.find(";");

if (i >= 0) {

// Remove options,i.e. ";timeout=90"

mSessionID.erase(i,mSessionID.size() - i);

}

i = response->mHeaders.indexOfKey("server");---server

if (i >= 0) {

AString server =response->mHeaders.valueAt(i);

if(server.startsWith("XenonStreamer")

||server.startsWith("XTream")) {

ALOGI("Usefake timestamps");

mUseSR = false;

}

sp<AMessage>notify = new AMessage('accu', id());

notify->setSize("track-index", trackIndex);

i =response->mHeaders.indexOfKey("transport");---transport

CHECK_GE(i, 0);

if(track->mRTPSocket != -1 && track->mRTCPSocket != -1) {

if(!track->mUsingInterleavedTCP) {

AStringtransport = response->mHeaders.valueAt(i);

……………….

++index;

if (result == OK &&index < mSessionDesc->countTracks()) {

setupTrack(index);----一般有两条track，先是audio track然后是videotrack

} else if(mSetupTracksSuccessful) {

建立完成后就可以“PLAY”了

++mKeepAliveGeneration;

postKeepAlive();

AStringrequest = "PLAY ";---------发送”PLAY”请求给服务器端

request.append(mControlURL);

request.append(" RTSP/1.0\r\n");

request.append("Session: ");

request.append(mSessionID);

request.append("\r\n");

sp<AMessage> reply = new AMessage('play', id());

mConn->sendRequest(request.c_str(), reply);

} else {

sp<AMessage> reply = newAMessage('disc', id());

mConn->disconnect(reply);

}

break;

}

完成“SETUP”阶段就可以“PLAY”了，发送给服务器端的格式如下：

C->V:PLAY rtsp://video.com/twister/video RTSP/1.0

CSeq: 2

Session:23456789

Range:smpte=0:10:00-

代码在myHandler.h中onMessageReceived对应的”setu”。

下面我们分析下服务器端返回后客户端如何处理“PLAY”。还是在myHandler.h中onMessageReceived函数：

case 'play':

{

………..

if (result == OK) {

sp<RefBase> obj;

CHECK(msg->findObject("response", &obj));

sp<ARTSPResponse>response =

static_cast<ARTSPResponse*>(obj.get());

if(response->mStatusCode != 200) {

result = UNKNOWN_ERROR;

} else {

parsePlayResponse(response);---解析response回来的数据

………………

}

if (result != OK) {

sp<AMessage> reply =new AMessage('disc', id());

mConn->disconnect(reply);

}

break;

}

response回来的格式一般如下：

V->C:RTSP/1.0 200 OK

CSeq: 2

Session:23456789

Range:smpte=0:10:00-0:20:00------------------播放从10分钟到20分钟时间段的视频

RTP-Info:url=rtsp://video.com/twister/video

;seq=12312232;rtptime=78712811

voidparsePlayResponse(const sp<ARTSPResponse> &response) {

if (mTracks.size() == 0) {

ALOGV("parsePlayResponse: latepackets ignored.");

return;

}

mPlayResponseReceived = true;

ssize_t i =response->mHeaders.indexOfKey("range");

…………

AString range = response->mHeaders.valueAt(i);

………………

i =response->mHeaders.indexOfKey("rtp-info");

CHECK_GE(i, 0);

AString rtpInfo =response->mHeaders.valueAt(i);

List<AString> streamInfos;

SplitString(rtpInfo, ",",&streamInfos);

int n = 1;

for (List<AString>::iterator it =streamInfos.begin();

it != streamInfos.end(); ++it) {

(*it).trim();

ALOGV("streamInfo[%d] =%s", n, (*it).c_str());

CHECK(GetAttribute((*it).c_str(),"url", &val));

size_t trackIndex = 0;

while (trackIndex <mTracks.size()) {

size_t startpos = 0;

if(mTracks.editItemAt(trackIndex).mURL.size() >= val.size()) {

startpos =mTracks.editItemAt(trackIndex).mURL.size() - val.size();

}

// Use AString::find in orderto allow the url in the RTP-Info to be a

// truncated variant (example:"url=trackID=1") of the complete SETUP url

if(mTracks.editItemAt(trackIndex).mURL.find(val.c_str(), startpos) == -1) {

++trackIndex;

} else {

// Found track

break;

}

CHECK_LT(trackIndex,mTracks.size());

char *end;

unsigned long seq = 0;

if (GetAttribute((*it).c_str(),"seq", &val)) {

seq = strtoul(val.c_str(),&end, 10);

} else {

CHECK(GetAttribute((*it).c_str(), "rtptime", &val));

}

TrackInfo *info = &mTracks.editItemAt(trackIndex);

info->mFirstSeqNumInSegment =seq;

info->mNewSegment = true;

uint32_t rtpTime = 0;

if (GetAttribute((*it).c_str(),"rtptime", &val)) {

rtpTime = strtoul(val.c_str(),&end, 10);

mReceivedRTPTime = true;

ALOGV("track #%d:rtpTime=%u <=> npt=%.2f", n, rtpTime, npt1);

} else {

ALOGV("no rtptime in playresponse: track #%d: rtpTime=%u <=> npt=%.2f", n,

rtpTime, npt1);

CHECK(GetAttribute((*it).c_str(), "seq", &val));

}

info->mRTPAnchor = rtpTime;

mLastMediaTimeUs = (int64_t)(npt1 *1E6);

mMediaAnchorUs = mLastMediaTimeUs;

// Removing packets with old RTPtimestamps

while (!info->mPackets.empty()){

sp<ABuffer> accessUnit =*info->mPackets.begin();

uint32_t firstRtpTime;

CHECK(accessUnit->meta()->findInt32("rtp-time", (int32_t*)&firstRtpTime));

if (firstRtpTime == rtpTime) {

break;

}

info->mPackets.erase(info->mPackets.begin());

}

++n;

}

至此video source 和audiosource就可以通过RTP不断的往客户端发送，客户端拿到这些数据就可以通过相应的解码器解析播放了。

我们的流媒体播放流程也讲得差不多了，如何关闭两端的流程就由大伙自己去看了。但是大家要注意一点有时候一些服务在关闭的时候没有发回“TEARDOWN”的response。

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