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openRtpServer接口增加单视频参数，加快单视频流注册速度 (#3342)

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only_audio -> only_track
This commit is contained in:
waken
2024-03-05 17:06:31 +08:00
committed by GitHub
parent ffdc13bfb9
commit 79b2aa6adc
10 changed files with 47 additions and 34 deletions
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18
server/WebApi.cpp
View File

@@ -415,7 +415,7 @@ 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, bool only_audio, bool multiplex) {
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) {
lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
if (s_rtpServerMap.find(stream_id) != s_rtpServerMap.end()) {
//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
@@ -423,7 +423,7 @@ uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mod
}
RtpServer::Ptr server = std::make_shared<RtpServer>();
server->start(local_port, stream_id, (RtpServer::TcpMode)tcp_mode, local_ip.c_str(), re_use_port, ssrc, only_audio, multiplex);
server->start(local_port, 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
lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
@@ -1198,12 +1198,17 @@ void installWebApi() {
//兼容老版本请求，新版本去除enable_tcp参数并新增tcp_mode参数
tcp_mode = 1;
}
auto only_track = allArgs["only_track"].as<int>();
if (allArgs["only_audio"].as<bool>()) {
// 兼容老版本请求，新版本去除only_audio参数并新增only_track参数
only_track = 1;
}
std::string local_ip = "::";
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>(),
allArgs["ssrc"].as<uint32_t>(), allArgs["only_audio"].as<bool>());
allArgs["ssrc"].as<uint32_t>(), only_track);
if (port == 0) {
throw InvalidArgsException("该stream_id已存在");
}
@@ -1220,11 +1225,16 @@ void installWebApi() {
// 兼容老版本请求，新版本去除enable_tcp参数并新增tcp_mode参数
tcp_mode = 1;
}
auto only_track = allArgs["only_track"].as<int>();
if (allArgs["only_audio"].as<bool>()) {
// 兼容老版本请求，新版本去除only_audio参数并新增only_track参数
only_track = 1;
}
std::string local_ip = "::";
if (!allArgs["local_ip"].empty()) {
local_ip = allArgs["local_ip"];
}
auto port = openRtpServer(allArgs["port"], stream_id, tcp_mode, local_ip, true, 0, allArgs["only_audio"].as<bool>(),true);
auto port = openRtpServer(allArgs["port"], stream_id, tcp_mode, local_ip, true, 0, only_track,true);
if (port == 0) {
throw InvalidArgsException("该stream_id已存在");
}

2
server/WebApi.h
View File

@@ -233,7 +233,7 @@ 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, bool only_audio, bool multiplex=false);
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);
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