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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
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waken 2024-03-05 17:06:31 +08:00 committed by GitHub
parent ffdc13bfb9
commit 79b2aa6adc
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10 changed files with 47 additions and 34 deletions
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12
postman/ZLMediaKit.postman_collection.json
View File

@ -1,11 +1,11 @@
{
"info": {
"_postman_id": "08e3bc35-5318-4949-81bb-90d854706194",
"_postman_id": "08c77fc3-7670-428c-bde4-80c8cc9f389f",
"name": "ZLMediaKit",
"description": "媒体服务器",
"schema": "https://schema.getpostman.com/json/collection/v2.1.0/collection.json",
"_exporter_id": "29185956",
"_collection_link": "https://lively-station-598157.postman.co/workspace/%E6%B5%81%E5%AA%92%E4%BD%93%E6%9C%8D%E5%8A%A1~1e119172-45b0-4ed6-b1fc-8a15d0e2d5f8/collection/29185956-08e3bc35-5318-4949-81bb-90d854706194?action=share&source=collection_link&creator=29185956"
"_collection_link": "https://lively-station-598157.postman.co/workspace/%E6%B5%81%E5%AA%92%E4%BD%93%E6%9C%8D%E5%8A%A1~1e119172-45b0-4ed6-b1fc-8a15d0e2d5f8/collection/29185956-08c77fc3-7670-428c-bde4-80c8cc9f389f?action=share&source=collection_link&creator=29185956"
},
"item": [
{
@ -1470,9 +1470,9 @@
"disabled": true
},
{
"key": "only_audio",
"key": "only_track",
"value": "1",
"description": "是否为单音频track，用于语音对讲",
"description": "是否为单音频/单视频track，0：不设置，1：单音频，2：单视频",
"disabled": true
},
{
@ -1523,9 +1523,9 @@
"description": "该端口绑定的流id\n"
},
{
"key": "only_audio",
"key": "only_track",
"value": "0",
"description": "是否为单音频track，用于语音对讲",
"description": "是否为单音频/单视频track，0：不设置，1：单音频，2：单视频",
"disabled": true
},
{

18
server/WebApi.cpp
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@ -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

2
src/Common/MediaSink.cpp
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@ -133,7 +133,7 @@ void MediaSink::checkTrackIfReady() {
}
GET_CONFIG(uint32_t, kMaxAddTrackMS, General::kWaitAddTrackMS);
if (_track_map.size() == 1 && _ticker.elapsedTime() > kMaxAddTrackMS) {
if (_track_map.size() == 1 && (_ticker.elapsedTime() > kMaxAddTrackMS || !_enable_audio)) {
// 如果只有一个Track，那么在该Track添加后，我们最多还等待若干时间(可能后面还会添加Track)
emitAllTrackReady();
return;

10
src/Rtp/RtpProcess.cpp
View File

@ -199,8 +199,8 @@ void RtpProcess::setStopCheckRtp(bool is_check){
}
}
void RtpProcess::setOnlyAudio(bool only_audio){
_only_audio = only_audio;
void RtpProcess::setOnlyTrack(OnlyTrack only_track) {
_only_track = only_track;
}
void RtpProcess::onDetach() {
@ -259,8 +259,10 @@ void RtpProcess::emitOnPublish() {
if (!option.stream_replace.empty()) {
RtpSelector::Instance().addStreamReplace(strong_self->_media_info.stream, option.stream_replace);
}
if (strong_self->_only_audio) {
strong_self->_muxer->setOnlyAudio();
switch (strong_self->_only_track) {
case kOnlyAudio: strong_self->_muxer->setOnlyAudio(); break;
case kOnlyVideo: strong_self->_muxer->enableAudio(false); break;
default: break;
}
strong_self->_muxer->setMediaListener(strong_self);
strong_self->doCachedFunc();

7
src/Rtp/RtpProcess.h
View File

@ -24,6 +24,7 @@ public:
friend class RtpProcessHelper;
RtpProcess(const std::string &stream_id);
~RtpProcess();
enum OnlyTrack { kAll = 0, kOnlyAudio = 1, kOnlyVideo = 2 };
/**
* 输入rtp
@ -58,10 +59,10 @@ public:
void setStopCheckRtp(bool is_check=false);
/**
* 设置为单track，单音频时可以加快媒体注册速度
* 设置为单track，单音频/单视频时可以加快媒体注册速度
* 请在inputRtp前调用此方法，否则可能会是空操作
*/
void setOnlyAudio(bool only_audio);
void setOnlyTrack(OnlyTrack only_track);
/**
* flush输出缓存
@ -93,7 +94,7 @@ private:
void doCachedFunc();
private:
bool _only_audio = false;
OnlyTrack _only_track = kAll;
std::string _auth_err;
uint64_t _dts = 0;
uint64_t _total_bytes = 0;

16
src/Rtp/RtpServer.cpp
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@ -42,12 +42,12 @@ public:
}
}
void setRtpServerInfo(uint16_t local_port,RtpServer::TcpMode mode,bool re_use_port,uint32_t ssrc, bool only_audio) {
void setRtpServerInfo(uint16_t local_port, RtpServer::TcpMode mode, bool re_use_port, uint32_t ssrc, int only_track) {
_local_port = local_port;
_tcp_mode = mode;
_re_use_port = re_use_port;
_ssrc = ssrc;
_only_audio = only_audio;
_only_track = only_track;
}
void setOnDetach(function<void()> cb) {
@ -61,7 +61,7 @@ public:
void onRecvRtp(const Socket::Ptr &sock, const Buffer::Ptr &buf, struct sockaddr *addr) {
if (!_process) {
_process = RtpSelector::Instance().getProcess(_stream_id, true);
_process->setOnlyAudio(_only_audio);
_process->setOnlyTrack((RtpProcess::OnlyTrack)_only_track);
_process->setOnDetach(std::move(_on_detach));
cancelDelayTask();
}
@ -142,7 +142,7 @@ private:
private:
bool _re_use_port = false;
bool _only_audio = false;
int _only_track = 0;
uint16_t _local_port = 0;
uint32_t _ssrc = 0;
RtpServer::TcpMode _tcp_mode = RtpServer::NONE;
@ -156,7 +156,7 @@ private:
EventPoller::DelayTask::Ptr _delay_task;
};
void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_mode, const char *local_ip, bool re_use_port, uint32_t ssrc, bool only_audio, bool multiplex) {
void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_mode, const char *local_ip, bool re_use_port, uint32_t ssrc, int only_track, bool multiplex) {
//创建udp服务器
Socket::Ptr rtp_socket = Socket::createSocket(nullptr, true);
Socket::Ptr rtcp_socket = Socket::createSocket(nullptr, true);
@ -184,7 +184,7 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
tcp_server = std::make_shared<TcpServer>(rtp_socket->getPoller());
(*tcp_server)[RtpSession::kStreamID] = stream_id;
(*tcp_server)[RtpSession::kSSRC] = ssrc;
(*tcp_server)[RtpSession::kOnlyAudio] = only_audio;
(*tcp_server)[RtpSession::kOnlyTrack] = only_track;
if (tcp_mode == PASSIVE) {
tcp_server->start<RtpSession>(local_port, local_ip);
} else if (stream_id.empty()) {
@ -201,7 +201,7 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
//指定了流id，那么一个端口一个流(不管是否包含多个ssrc的多个流，绑定rtp源后，会筛选掉ip端口不匹配的流)
helper = std::make_shared<RtcpHelper>(std::move(rtcp_socket), stream_id);
helper->startRtcp();
helper->setRtpServerInfo(local_port, tcp_mode, re_use_port, ssrc, only_audio);
helper->setRtpServerInfo(local_port, tcp_mode, re_use_port, ssrc, only_track);
bool bind_peer_addr = false;
auto ssrc_ptr = std::make_shared<uint32_t>(ssrc);
_ssrc = ssrc_ptr;
@ -223,7 +223,7 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
} else {
//单端口多线程接收多个流，根据ssrc区分流
udp_server = std::make_shared<UdpServer>(rtp_socket->getPoller());
(*udp_server)[RtpSession::kOnlyAudio] = only_audio;
(*udp_server)[RtpSession::kOnlyTrack] = only_track;
(*udp_server)[RtpSession::kUdpRecvBuffer] = udpRecvSocketBuffer;
udp_server->start<RtpSession>(local_port, local_ip);
rtp_socket = nullptr;

4
src/Rtp/RtpServer.h
View File

@ -44,7 +44,7 @@ public:
* @param multiplex 多路复用
*/
void start(uint16_t local_port, const std::string &stream_id = "", TcpMode tcp_mode = PASSIVE,
const char *local_ip = "::", bool re_use_port = true, uint32_t ssrc = 0, bool only_audio = false, bool multiplex = false);
const char *local_ip = "::", bool re_use_port = true, uint32_t ssrc = 0, int only_track = 0, bool multiplex = false);
/**
* 连接到tcp服务(tcp主动模式)
@ -81,7 +81,7 @@ protected:
std::shared_ptr<RtcpHelper> _rtcp_helper;
std::function<void()> _on_cleanup;
bool _only_audio = false;
int _only_track = 0;
//用于tcp主动模式
TcpMode _tcp_mode = NONE;
};

6
src/Rtp/RtpSession.cpp
View File

@ -23,7 +23,7 @@ namespace mediakit{
const string RtpSession::kStreamID = "stream_id";
const string RtpSession::kSSRC = "ssrc";
const string RtpSession::kOnlyAudio = "only_audio";
const string RtpSession::kOnlyTrack = "only_track";
const string RtpSession::kUdpRecvBuffer = "udp_recv_socket_buffer";
void RtpSession::attachServer(const Server &server) {
@ -33,7 +33,7 @@ void RtpSession::attachServer(const Server &server) {
void RtpSession::setParams(mINI &ini) {
_stream_id = ini[kStreamID];
_ssrc = ini[kSSRC];
_only_audio = ini[kOnlyAudio];
_only_track = ini[kOnlyTrack];
int udp_socket_buffer = ini[kUdpRecvBuffer];
if (_is_udp) {
// 设置udp socket读缓存
@ -125,7 +125,7 @@ void RtpSession::onRtpPacket(const char *data, size_t len) {
_delay_close = true;
return;
}
_process->setOnlyAudio(_only_audio);
_process->setOnlyTrack((RtpProcess::OnlyTrack)_only_track);
_process->setDelegate(static_pointer_cast<RtpSession>(shared_from_this()));
}
try {

4
src/Rtp/RtpSession.h
View File

@ -24,7 +24,7 @@ class RtpSession : public toolkit::Session, public RtpSplitter, public MediaSour
public:
static const std::string kStreamID;
static const std::string kSSRC;
static const std::string kOnlyAudio;
static const std::string kOnlyTrack;
static const std::string kUdpRecvBuffer;
RtpSession(const toolkit::Socket::Ptr &sock);
@ -52,7 +52,7 @@ private:
bool _is_udp = false;
bool _search_rtp = false;
bool _search_rtp_finished = false;
bool _only_audio = false;
int _only_track = 0;
uint32_t _ssrc = 0;
toolkit::Ticker _ticker;
std::string _stream_id;