Commit 2f9b3e18 by 夏楚 Committed by GitHub

Merge pull request #1984 from Dw9/master

c api support srt server
parents c363b303 43bf7c79
......@@ -162,11 +162,18 @@ API_EXPORT uint16_t API_CALL mk_rtp_server_start(uint16_t port);
/**
* 创建rtc服务器
* @param port rtp监听端口
* @param port rtc监听端口
* @return 0:失败,非0:端口号
*/
API_EXPORT uint16_t API_CALL mk_rtc_server_start(uint16_t port);
/**
* 创建srt服务器
* @param port srt监听端口
* @return 0:失败,非0:端口号
*/
API_EXPORT uint16_t API_CALL mk_srt_server_start(uint16_t port);
/**
* 创建shell服务器
......
......@@ -40,6 +40,11 @@ static std::shared_ptr<RtpServer> rtpServer;
static std::shared_ptr<UdpServer> rtcServer;
#endif
#if defined(ENABLE_SRT)
#include "../srt/SrtSession.hpp"
static std::shared_ptr<UdpServer> srtServer;
#endif
//////////////////////////environment init///////////////////////////
API_EXPORT void API_CALL mk_env_init(const mk_config *cfg) {
......@@ -62,9 +67,16 @@ API_EXPORT void API_CALL mk_stop_all_server(){
CLEAR_ARR(rtsp_server);
CLEAR_ARR(rtmp_server);
CLEAR_ARR(http_server);
shell_server = nullptr;
#ifdef ENABLE_RTPPROXY
rtpServer = nullptr;
#endif
#ifdef ENABLE_WEBRTC
rtcServer = nullptr;
#endif
#ifdef ENABLE_SRT
srtServer = nullptr;
#endif
stopAllTcpServer();
}
......@@ -254,6 +266,36 @@ API_EXPORT uint16_t API_CALL mk_rtc_server_start(uint16_t port) {
#endif
}
API_EXPORT uint16_t API_CALL mk_srt_server_start(uint16_t port) {
#ifdef ENABLE_SRT
try {
srtServer = std::make_shared<UdpServer>();
srtServer->setOnCreateSocket([](const EventPoller::Ptr &poller, const Buffer::Ptr &buf, struct sockaddr *, int) {
if (!buf) {
return Socket::createSocket(poller, false);
}
auto new_poller = SRT::SrtSession::queryPoller(buf);
if (!new_poller) {
//握手第一阶段
return Socket::createSocket(poller, false);
}
return Socket::createSocket(new_poller, false);
});
srtServer->start<SRT::SrtSession>(port);
return srtServer->getPort();
} catch (std::exception &ex) {
srtServer.reset();
WarnL << ex.what();
return 0;
}
#else
WarnL << "未启用该功能!";
return 0;
#endif
}
API_EXPORT uint16_t API_CALL mk_shell_server_start(uint16_t port){
try {
shell_server = std::make_shared<TcpServer>();
......
......@@ -416,6 +416,7 @@ int main(int argc, char *argv[]) {
mk_shell_server_start(9000);
mk_rtp_server_start(10000);
mk_rtc_server_start(8000);
mk_srt_server_start(9000);
mk_events events = {
.on_mk_media_changed = on_mk_media_changed,
......
......@@ -336,12 +336,9 @@ int start_main(int argc,char *argv[]) {
if (rtcPort) { rtcSrv->start<WebRtcSession>(rtcPort); }
#endif//defined(ENABLE_WEBRTC)
#if defined(ENABLE_SRT)
// srt udp服务器
if(srtPort){
srtSrv->start<SRT::SrtSession>(srtPort);
}
if(srtPort) { srtSrv->start<SRT::SrtSession>(srtPort); }
#endif//defined(ENABLE_SRT)
} catch (std::exception &ex) {
......
#include "Util/onceToken.h"
#include "Util/mini.h"
#include <iterator>
#include <stdlib.h>
......@@ -15,6 +17,13 @@ const std::string kPort = SRT_FIELD "port";
const std::string kLatencyMul = SRT_FIELD "latencyMul";
const std::string kPktBufSize = SRT_FIELD "pktBufSize";
static onceToken token([]() {
mINI::Instance()[kTimeOutSec] = 5;
mINI::Instance()[kPort] = 9000;
mINI::Instance()[kLatencyMul] = 4;
mINI::Instance()[kPktBufSize] = 8192;
});
static std::atomic<uint32_t> s_srt_socket_id_generate { 125 };
//////////// SrtTransport //////////////////////////
SrtTransport::SrtTransport(const EventPoller::Ptr &poller)
......
......@@ -46,14 +46,14 @@
</p>
<p>
<label for="methond">methond(play or push or echo):</label>
<input type="radio" name="methond" value="echo" >echo
<input type="radio" name="methond" value="push" >push
<input type="radio" name="methond" value="play" checked = true>play
<label for="method">method(play or push or echo):</label>
<input type="radio" name="method" value="echo" >echo
<input type="radio" name="method" value="push" >push
<input type="radio" name="method" value="play" checked = true>play
</p>
<p>
<label for="resilution">resolution:</label>
<select id="resilution">
<label for="resolution">resolution:</label>
<select id="resolution">
</select>
</p>
<p>
......@@ -98,7 +98,7 @@
url = "http://127.0.0.1"+"/index/api/webrtc?app=live&stream=test&type=play"
}
document.getElementById('streamUrl').value = url
document.getElementsByName("methond").forEach((el,idx)=>{
document.getElementsByName("method").forEach((el,idx)=>{
el.onclick=function(e){
let url = new URL(document.getElementById('streamUrl').value);
url.searchParams.set("type",el.value)
......@@ -118,14 +118,14 @@
opt = document.createElement('option');
opt.text = r.label +"("+r.width+"x"+r.height+")";
opt.value = r;
document.getElementById("resilution").add(opt,null)
document.getElementById("resolution").add(opt,null)
//console.log(opt.text.match(/\d+/g))
})
function start_play(){
let elr = document.getElementById("resilution");
let elr = document.getElementById("resolution");
let res = elr.options[elr.selectedIndex].text.match(/\d+/g);
let h = parseInt(res.pop());
let w = parseInt(res.pop());
......@@ -204,7 +204,7 @@
function start()
{
stop();
let elr = document.getElementById("resilution");
let elr = document.getElementById("resolution");
let res = elr.options[elr.selectedIndex].text.match(/\d+/g);
let h = parseInt(res.pop());
let w = parseInt(res.pop());
......
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