合并代码

This commit is contained in:
xiongziliang 2021-05-22 09:26:49 +08:00
commit 8cbec43ee8
4 changed files with 69 additions and 55 deletions

@ -1 +1 @@
Subproject commit d5de1d4f0c86139fe8f0b6d956e45dee3b151783
Subproject commit b04d132139399637cbb201c2832dd98dc2786e82

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@ -512,6 +512,12 @@
"value": null,
"description": "rtsp拉流时拉流方式0tcp1udp2组播",
"disabled": true
},
{
"key": "timeout_sec",
"value": "10",
"description": "拉流超时时间单位秒float类型",
"disabled": true
}
]
}

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@ -600,6 +600,7 @@ void installWebApi() {
bool enable_hls,
bool enable_mp4,
int rtp_type,
float timeoutSec,
const function<void(const SockException &ex,const string &key)> &cb){
auto key = getProxyKey(vhost,app,stream);
lock_guard<recursive_mutex> lck(s_proxyMapMtx);
@ -614,6 +615,12 @@ void installWebApi() {
//指定RTP over TCP(播放rtsp时有效)
(*player)[kRtpType] = rtp_type;
if (timeoutSec > 0.1) {
//播放握手超时时间
(*player)[kTimeoutMS] = timeoutSec * 1000;
}
//开始播放,如果播放失败或者播放中止,将会自动重试若干次,默认一直重试
player->setPlayCallbackOnce([cb,key](const SockException &ex){
if(ex){
@ -643,6 +650,7 @@ void installWebApi() {
allArgs["enable_hls"],/* 是否hls转发 */
allArgs["enable_mp4"],/* 是否MP4录制 */
allArgs["rtp_type"],
allArgs["timeout_sec"],
[invoker,val,headerOut](const SockException &ex,const string &key) mutable{
if (ex) {
val["code"] = API::OtherFailed;
@ -1133,6 +1141,7 @@ void installWebApi() {
true,/* 开启hls转发 */
false,/* 禁用MP4录制 */
0,//rtp over tcp方式拉流
10,//10秒超时
[invoker,val,headerOut](const SockException &ex,const string &key) mutable{
if(ex){
val["code"] = API::OtherFailed;

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@ -227,31 +227,7 @@ void RtspSession::handleReq_ANNOUNCE(const Parser &parser) {
throw SockException(Err_shutdown, StrPrinter << err << ":" << full_url);
}
SdpParser sdpParser(parser.Content());
_sessionid = makeRandStr(12);
_sdp_track = sdpParser.getAvailableTrack();
if (_sdp_track.empty()) {
//sdp无效
static constexpr auto err = "sdp中无有效track";
sendRtspResponse("403 Forbidden", {"Content-Type", "text/plain"}, err);
throw SockException(Err_shutdown,StrPrinter << err << ":" << full_url);
}
_rtcp_context.clear();
for (auto &track : _sdp_track) {
_rtcp_context.emplace_back(std::make_shared<RtcpContext>(track->_samplerate, true));
}
_push_src = std::make_shared<RtspMediaSourceImp>(_media_info._vhost, _media_info._app, _media_info._streamid);
_push_src->setListener(dynamic_pointer_cast<MediaSourceEvent>(shared_from_this()));
_push_src->setSdp(sdpParser.toString());
sendRtspResponse("200 OK");
}
void RtspSession::handleReq_RECORD(const Parser &parser){
if (_sdp_track.empty() || parser["Session"] != _sessionid) {
send_SessionNotFound();
throw SockException(Err_shutdown, _sdp_track.empty() ? "can not find any availabe track when record" : "session not found when record");
}
auto onRes = [this](const string &err, bool enableHls, bool enableMP4){
auto onRes = [this, parser, full_url](const string &err, bool enableHls, bool enableMP4){
bool authSuccess = err.empty();
if (!authSuccess) {
sendRtspResponse("401 Unauthorized", {"Content-Type", "text/plain"}, err);
@ -259,25 +235,25 @@ void RtspSession::handleReq_RECORD(const Parser &parser){
return;
}
//设置转协议
_push_src->setProtocolTranslation(enableHls, enableMP4);
_StrPrinter rtp_info;
for(auto &track : _sdp_track){
if (track->_inited == false) {
//还有track没有setup
shutdown(SockException(Err_shutdown,"track not setuped"));
SdpParser sdpParser(parser.Content());
_sessionid = makeRandStr(12);
_sdp_track = sdpParser.getAvailableTrack();
if (_sdp_track.empty()) {
//sdp无效
static constexpr auto err = "sdp中无有效track";
sendRtspResponse("403 Forbidden", {"Content-Type", "text/plain"}, err);
shutdown(SockException(Err_shutdown, StrPrinter << err << ":" << full_url));
return;
}
rtp_info << "url=" << track->getControlUrl(_content_base) << ",";
}
rtp_info.pop_back();
sendRtspResponse("200 OK", {"RTP-Info",rtp_info});
if(_rtp_type == Rtsp::RTP_TCP){
//如果是rtsp推流服务器并且是TCP推流设置socket flags,,这样能提升接收性能
setSocketFlags();
_rtcp_context.clear();
for (auto &track : _sdp_track) {
_rtcp_context.emplace_back(std::make_shared<RtcpContext>(track->_samplerate, true));
}
_push_src = std::make_shared<RtspMediaSourceImp>(_media_info._vhost, _media_info._app, _media_info._streamid);
_push_src->setListener(dynamic_pointer_cast<MediaSourceEvent>(shared_from_this()));
_push_src->setProtocolTranslation(enableHls, enableMP4);
_push_src->setSdp(sdpParser.toString());
sendRtspResponse("200 OK");
};
weak_ptr<RtspSession> weakSelf = dynamic_pointer_cast<RtspSession>(shared_from_this());
@ -305,6 +281,29 @@ void RtspSession::handleReq_RECORD(const Parser &parser){
}
}
void RtspSession::handleReq_RECORD(const Parser &parser){
if (_sdp_track.empty() || parser["Session"] != _sessionid) {
send_SessionNotFound();
throw SockException(Err_shutdown, _sdp_track.empty() ? "can not find any availabe track when record" : "session not found when record");
}
_StrPrinter rtp_info;
for (auto &track : _sdp_track) {
if (track->_inited == false) {
//还有track没有setup
shutdown(SockException(Err_shutdown, "track not setuped"));
return;
}
rtp_info << "url=" << track->getControlUrl(_content_base) << ",";
}
rtp_info.pop_back();
sendRtspResponse("200 OK", {"RTP-Info", rtp_info});
if (_rtp_type == Rtsp::RTP_TCP) {
//如果是rtsp推流服务器并且是TCP推流设置socket flags,,这样能提升接收性能
setSocketFlags();
}
}
void RtspSession::emitOnPlay(){
weak_ptr<RtspSession> weakSelf = dynamic_pointer_cast<RtspSession>(shared_from_this());
//url鉴权回调