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ZLMediaKit
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补提rtp app名称自定义相关文件修改

This commit is contained in:
baiyfcu 2024-04-07 14:08:05 +08:00
parent d771530316
commit eab1dbc606
5 changed files with 20 additions and 11 deletions
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3
api/include/mk_rtp_server.h
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@ -20,10 +20,11 @@ typedef struct mk_rtp_server_t *mk_rtp_server;
* 创建GB28181 RTP 服务器
* @param port 监听端口，0则为随机
* @param tcp_mode tcp模式(0: 不监听端口 1: 监听端口 2: 主动连接到服务端)
* @param app_name 该端口绑定的app名称，为""使用默认配置 rtp
* @param stream_id 该端口绑定的流id
* @return
*/
API_EXPORT mk_rtp_server API_CALL mk_rtp_server_create(uint16_t port, int tcp_mode, const char *stream_id);
API_EXPORT mk_rtp_server API_CALL mk_rtp_server_create(uint16_t port, int tcp_mode, const char *app_name, const char *stream_id);
/**
* TCP 主动模式时连接到服务器是否成功的回调

4
api/source/mk_rtp_server.cpp
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@ -16,9 +16,9 @@ using namespace toolkit;
#include "Rtp/RtpServer.h"
using namespace mediakit;
API_EXPORT mk_rtp_server API_CALL mk_rtp_server_create(uint16_t port, int tcp_mode, const char *stream_id) {
API_EXPORT mk_rtp_server API_CALL mk_rtp_server_create(uint16_t port, int tcp_mode, const char *app_name, const char *stream_id) {
RtpServer::Ptr *server = new RtpServer::Ptr(new RtpServer);
(*server)->start(port, stream_id, (RtpServer::TcpMode)tcp_mode);
(*server)->start(port, app_name, stream_id, (RtpServer::TcpMode)tcp_mode);
return (mk_rtp_server)server;
}

18
server/WebApi.cpp
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@ -469,14 +469,14 @@ Value makeMediaSourceJson(MediaSource &media){
}
#if defined(ENABLE_RTPPROXY)
uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mode, const string &local_ip, bool re_use_port, uint32_t ssrc, int only_track, bool multiplex) {
uint16_t openRtpServer(uint16_t local_port, const string &app_name, const string &stream_id, int tcp_mode, const string &local_ip, bool re_use_port, uint32_t ssrc, int only_track, bool multiplex) {
if (s_rtp_server.find(stream_id)) {
//为了防止RtpProcess所有权限混乱的问题，不允许重复添加相同的stream_id
return 0;
}
auto server = s_rtp_server.makeWithAction(stream_id, [&](RtpServer::Ptr server) {
server->start(local_port, stream_id, (RtpServer::TcpMode)tcp_mode, local_ip.c_str(), re_use_port, ssrc, only_track, multiplex);
server->start(local_port, app_name, stream_id, (RtpServer::TcpMode)tcp_mode, local_ip.c_str(), re_use_port, ssrc, only_track, multiplex);
});
server->setOnDetach([stream_id]() {
//设置rtp超时移除事件
@ -1192,7 +1192,7 @@ void installWebApi() {
CHECK_SECRET();
CHECK_ARGS("stream_id");
auto process = RtpSelector::Instance().getProcess(allArgs["stream_id"], false);
auto process = RtpSelector::Instance().getProcess(allArgs["app_name"], allArgs["stream_id"], false);
if (!process) {
val["exist"] = false;
return;
@ -1204,6 +1204,7 @@ void installWebApi() {
api_regist("/index/api/openRtpServer",[](API_ARGS_MAP){
CHECK_SECRET();
CHECK_ARGS("port", "stream_id");
auto app_name = allArgs["app_name"]; // 默认rtp
auto stream_id = allArgs["stream_id"];
auto tcp_mode = allArgs["tcp_mode"].as<int>();
if (allArgs["enable_tcp"].as<int>() && !tcp_mode) {
@ -1219,7 +1220,8 @@ void installWebApi() {
if (!allArgs["local_ip"].empty()) {
local_ip = allArgs["local_ip"];
}
auto port = openRtpServer(allArgs["port"], stream_id, tcp_mode, local_ip, allArgs["re_use_port"].as<bool>(),
auto port = openRtpServer(
allArgs["port"], app_name, stream_id, tcp_mode, local_ip, allArgs["re_use_port"].as<bool>(),
allArgs["ssrc"].as<uint32_t>(), only_track);
if (port == 0) {
throw InvalidArgsException("该stream_id已存在");
@ -1231,6 +1233,7 @@ void installWebApi() {
api_regist("/index/api/openRtpServerMultiplex", [](API_ARGS_MAP) {
CHECK_SECRET();
CHECK_ARGS("port", "stream_id");
auto app_name = allArgs["app_name"];
auto stream_id = allArgs["stream_id"];
auto tcp_mode = allArgs["tcp_mode"].as<int>();
if (allArgs["enable_tcp"].as<int>() && !tcp_mode) {
@ -1246,7 +1249,7 @@ void installWebApi() {
if (!allArgs["local_ip"].empty()) {
local_ip = allArgs["local_ip"];
}
auto port = openRtpServer(allArgs["port"], stream_id, tcp_mode, local_ip, true, 0, only_track,true);
auto port = openRtpServer(allArgs["port"], app_name, stream_id, tcp_mode, local_ip, true, 0, only_track, true);
if (port == 0) {
throw InvalidArgsException("该stream_id已存在");
}
@ -1303,6 +1306,7 @@ void installWebApi() {
Value obj;
obj["stream_id"] = pr.first;
obj["port"] = pr.second->getPort();
obj["app_name"] = pr.second->getAppName();
val["data"].append(obj);
}
});
@ -1411,7 +1415,7 @@ void installWebApi() {
CHECK_SECRET();
CHECK_ARGS("stream_id");
//只是暂停流的检查，流媒体服务器做为流负载服务，收流就转发，RTSP/RTMP有自己暂停协议
auto rtp_process = RtpSelector::Instance().getProcess(allArgs["stream_id"], false);
auto rtp_process = RtpSelector::Instance().getProcess(allArgs["app"], allArgs["stream_id"], false);
if (rtp_process) {
rtp_process->setStopCheckRtp(true);
} else {
@ -1422,7 +1426,7 @@ void installWebApi() {
api_regist("/index/api/resumeRtpCheck", [](API_ARGS_MAP) {
CHECK_SECRET();
CHECK_ARGS("stream_id");
auto rtp_process = RtpSelector::Instance().getProcess(allArgs["stream_id"], false);
auto rtp_process = RtpSelector::Instance().getProcess(allArgs["app"], allArgs["stream_id"], false);
if (rtp_process) {
rtp_process->setStopCheckRtp(false);
} else {

4
server/WebApi.h
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@ -202,7 +202,9 @@ void installWebApi();
void unInstallWebApi();
#if defined(ENABLE_RTPPROXY)
uint16_t openRtpServer(uint16_t local_port, const std::string &stream_id, int tcp_mode, const std::string &local_ip, bool re_use_port, uint32_t ssrc, int only_track, bool multiplex=false);
uint16_t openRtpServer(uint16_t local_port, const std::string &app_name, const std::string &stream_id, int tcp_mode, const std::string &local_ip, bool re_use_port, uint32_t ssrc, int only_track, bool multiplex=false);
//void connectRtpServer(const std::string &stream_id, const std::string &dst_url, uint16_t dst_port, const std::function<void(const toolkit::SockException &ex)> &cb);
//bool closeRtpServer(const std::string &stream_id);
#endif
Json::Value makeMediaSourceJson(mediakit::MediaSource &media);

2
src/Common/config.cpp
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@ -262,6 +262,7 @@ const string kAudioMtuSize = RTP_FIELD "audioMtuSize";
const string kRtpMaxSize = RTP_FIELD "rtpMaxSize";
const string kLowLatency = RTP_FIELD "lowLatency";
const string kH264StapA = RTP_FIELD "h264_stap_a";
const string kRtpAppName = RTP_FIELD "rtpAppName";
static onceToken token([]() {
mINI::Instance()[kVideoMtuSize] = 1400;
@ -269,6 +270,7 @@ static onceToken token([]() {
mINI::Instance()[kRtpMaxSize] = 10;
mINI::Instance()[kLowLatency] = 0;
mINI::Instance()[kH264StapA] = 1;
mINI::Instance()[kRtpAppName] = "rtp";
});
} // namespace Rtp