本文主要是介绍android多媒体框架之流媒体具体流程篇3----base on jellybean(十三)，希望对大家解决编程问题提供一定的参考价值，需要的开发者们随着小编来一起学习吧！
距离上一篇文章好久了，一直没更新上，在此深表歉意。
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");
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。这篇关于android多媒体框架之流媒体具体流程篇3----base on jellybean(十三)的文章就介绍到这儿，希望我们推荐的文章对编程师们有所帮助！
