Merge branch 'dev-28181' of https://github.com/xialixin/srs into feature/gb28281-2

trunk/src/app/srs_app_config.cpp

@@ -275,6 +275,11 @@ bool srs_stream_caster_is_flv(string caster)
     return caster == "flv";
 }
 
+bool srs_stream_caster_is_gb28181(string caster)
+{
+    return caster == "gb28181";
+}
+
 bool srs_config_apply_filter(SrsConfDirective* dvr_apply, SrsRequest* req)
 {
     static bool DEFAULT = true;
@@ -2137,7 +2142,26 @@ srs_error_t SrsConfig::global_to_json(SrsJsonObject* obj)
                     sobj->set(sdir->name, sdir->dumps_arg0_to_integer());
                 } else if (sdir->name == "rtp_port_max") {
                     sobj->set(sdir->name, sdir->dumps_arg0_to_integer());
+                } else if (sdir->name == "rtp_idle_timeout") {
+                    sobj->set(sdir->name, sdir->dumps_arg0_to_integer());
+                } else if (sdir->name == "ack_timeout") {
+                    sobj->set(sdir->name, sdir->dumps_arg0_to_integer());
+                } else if (sdir->name == "keepalive_timeout") {
+                    sobj->set(sdir->name, sdir->dumps_arg0_to_integer());
+                } else if (sdir->name == "audio_enable") {
+                    sobj->set(sdir->name, sdir->dumps_arg0_to_boolean());
+                } else if (sdir->name == "host") {
+                    sobj->set(sdir->name, sdir->dumps_arg0_to_str());
+                } else if (sdir->name == "serial") {
+                    sobj->set(sdir->name, sdir->dumps_arg0_to_str());
+                } else if (sdir->name == "realm") {
+                    sobj->set(sdir->name, sdir->dumps_arg0_to_str());
+                } else if (sdir->name == "wait_keyframe") {
+                    sobj->set(sdir->name, sdir->dumps_arg0_to_str());
+                } else if (sdir->name == "print_sip_message") {
+                    sobj->set(sdir->name, sdir->dumps_arg0_to_str());
                 }
+
             }
             obj->set(dir->name, sobj);
         } else {
@@ -3650,7 +3674,11 @@ srs_error_t SrsConfig::check_normal_config()
             SrsConfDirective* conf = stream_caster->at(i);
             string n = conf->name;
             if (n != "enabled" && n != "caster" && n != "output"
-                && n != "listen" && n != "rtp_port_min" && n != "rtp_port_max") {
+                && n != "listen" && n != "rtp_port_min" && n != "rtp_port_max"
+                && n != "rtp_idle_timeout" && n != "ack_timeout" && n != "keepalive_timeout"
+                && n != "host" && n != "serial" && n != "realm" 
+                && n != "audio_enable" && n != "wait_keyframe"
+                && n != "print_sip_message") {
                 return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal stream_caster.%s", n.c_str());
             }
         }
@@ -4266,6 +4294,150 @@ int SrsConfig::get_stream_caster_rtp_port_max(SrsConfDirective* conf)
     return ::atoi(conf->arg0().c_str());
 }
 
+int SrsConfig::get_stream_caster_gb28181_rtp_ide_timeout(SrsConfDirective* conf)
+{
+    static int DEFAULT = 30;
+    
+    if (!conf) {
+        return DEFAULT;
+    }
+    
+    conf = conf->get("rtp_ide_timeout");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    
+    return ::atoi(conf->arg0().c_str());
+}
+
+int SrsConfig::get_stream_caster_gb28181_ack_timeout(SrsConfDirective* conf)
+{
+    static int DEFAULT = 30;
+    
+    if (!conf) {
+        return DEFAULT;
+    }
+    
+    conf = conf->get("ack_timeout");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    
+    return ::atoi(conf->arg0().c_str());
+}
+
+int SrsConfig::get_stream_caster_gb28181_keepalive_timeout(SrsConfDirective* conf)
+{
+    static int DEFAULT = 30;
+    
+    if (!conf) {
+        return DEFAULT;
+    }
+    
+    conf = conf->get("keepalive_timeout");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    
+    return ::atoi(conf->arg0().c_str());
+}
+
+string SrsConfig::get_stream_caster_gb28181_host(SrsConfDirective* conf)
+{
+    static string DEFAULT = "127.0.0.1";
+    
+    if (!conf) {
+        return DEFAULT;
+    }
+    
+    conf = conf->get("host");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    
+    return conf->arg0();
+}
+
+string SrsConfig::get_stream_caster_gb28181_serial(SrsConfDirective* conf)
+{
+    static string DEFAULT = "";
+    
+    if (!conf) {
+        return DEFAULT;
+    }
+    
+    conf = conf->get("serial");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    
+    return conf->arg0();
+}
+
+string SrsConfig::get_stream_caster_gb28181_realm(SrsConfDirective* conf)
+{
+    static string DEFAULT = "";
+    
+    if (!conf) {
+        return DEFAULT;
+    }
+    
+    conf = conf->get("realm");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    
+    return conf->arg0();
+}
+
+bool SrsConfig::get_stream_caster_gb28181_audio_enable(SrsConfDirective* conf)
+{
+    static bool DEFAULT = true;
+    
+    if (!conf) {
+        return DEFAULT;
+    }
+    
+    conf = conf->get("audio_enable");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    
+    return SRS_CONF_PERFER_FALSE(conf->arg0());
+}
+
+bool SrsConfig::get_stream_caster_gb28181_print_sip_message(SrsConfDirective* conf)
+{
+    static bool DEFAULT = false;
+    
+    if (!conf) {
+        return DEFAULT;
+    }
+    
+    conf = conf->get("print_sip_message");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    
+    return SRS_CONF_PERFER_FALSE(conf->arg0());
+}
+
+bool SrsConfig::get_stream_caster_gb28181_wait_keyframe(SrsConfDirective* conf)
+{
+    static bool DEFAULT = false;
+    
+    if (!conf) {
+        return DEFAULT;
+    }
+    
+    conf = conf->get("wait_keyframe");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    
+    return SRS_CONF_PERFER_FALSE(conf->arg0());
+}
+
 SrsConfDirective* SrsConfig::get_vhost(string vhost, bool try_default_vhost)
 {
     srs_assert(root);

trunk/src/app/srs_app_config.hpp

@@ -119,6 +119,7 @@ extern bool srs_config_dvr_is_plan_session(std::string plan);
 extern bool srs_stream_caster_is_udp(std::string caster);
 extern bool srs_stream_caster_is_rtsp(std::string caster);
 extern bool srs_stream_caster_is_flv(std::string caster);
+extern bool srs_stream_caster_is_gb28181(std::string caster);
 // Whether the dvr_apply active the stream specified by req.
 extern bool srs_config_apply_filter(SrsConfDirective* dvr_apply, SrsRequest* req);
 
@@ -498,6 +499,17 @@ public:
     virtual int get_stream_caster_rtp_port_min(SrsConfDirective* conf);
     // Get the max udp port for rtp of stream caster rtsp.
     virtual int get_stream_caster_rtp_port_max(SrsConfDirective* conf);
+    
+    virtual int get_stream_caster_gb28181_rtp_ide_timeout(SrsConfDirective* conf);
+    virtual int get_stream_caster_gb28181_ack_timeout(SrsConfDirective* conf);
+    virtual int get_stream_caster_gb28181_keepalive_timeout(SrsConfDirective* conf);
+    virtual bool get_stream_caster_gb28181_audio_enable(SrsConfDirective* conf);
+    virtual std::string get_stream_caster_gb28181_host(SrsConfDirective* conf);
+    virtual std::string get_stream_caster_gb28181_serial(SrsConfDirective* conf);
+    virtual std::string get_stream_caster_gb28181_realm(SrsConfDirective* conf);
+    virtual bool get_stream_caster_gb28181_print_sip_message(SrsConfDirective* conf);
+    virtual bool get_stream_caster_gb28181_wait_keyframe(SrsConfDirective* conf);
+
 // vhost specified section
 public:
     // Get the vhost directive by vhost name.

trunk/src/app/srs_app_gb28181.hpp

@@ -0,0 +1,330 @@
+/**
+ * The MIT License (MIT)
+ *
+ * Copyright (c) 2013-2020 Winlin
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
+ * the Software, and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
+ * FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
+ * COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
+ * IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
+ * CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+#ifndef SRS_APP_GB28181_HPP
+#define SRS_APP_GB28181_HPP
+
+#include <srs_core.hpp>
+
+#include <arpa/inet.h>
+#include <string>
+#include <vector>
+#include <map>
+
+#include <srs_app_st.hpp>
+#include <srs_app_thread.hpp>
+#include <srs_app_listener.hpp>
+#include <srs_rtsp_stack.hpp>
+#include <srs_kernel_stream.hpp>
+#include <srs_app_log.hpp>
+#include <srs_kernel_file.hpp>
+
+class SrsStSocket;
+class SrsRtpConn;
+class SrsRtspConn;
+class SrsRtspStack;
+class SrsRtspCaster;
+class SrsConfDirective;
+class SrsRtpPacket;
+class SrsRequest;
+class SrsStSocket;
+class SrsRtmpClient;
+class SrsRawH264Stream;
+class SrsRawAacStream;
+struct SrsRawAacStreamCodec;
+class SrsSharedPtrMessage;
+class SrsAudioFrame;
+class SrsSimpleStream;
+class SrsPithyPrint;
+class SrsSimpleRtmpClient;
+class SrsSipStack;
+class SrsGb28181Caster;
+class SrsRtspJitter;
+class SrsRtspAudioCache;
+class SrsSipRequest;
+class SrsGb28181Conn;
+class SrsGb28281ClientInfo;
+
+/* gb28181 program stream struct define
+
+*/
+
+struct SrsPsPacketStartCode
+{
+	uint8_t start_code[3];
+	uint8_t stream_id[1];
+};
+
+struct SrsPsPacketHeader
+{
+	SrsPsPacketStartCode start;// 4
+	uint8_t info[9];
+	uint8_t stuffing_length;
+};
+
+struct SrsPsPacketBBHeader
+{
+	SrsPsPacketStartCode start;
+	uint16_t    length;
+};
+
+struct SrsPsePacket
+{
+	SrsPsPacketStartCode     start;
+	uint16_t    length;
+	uint8_t         info[2];
+	uint8_t         stuffing_length;
+};
+
+struct SrsPsMapPacket
+{
+	SrsPsPacketStartCode  start;
+	uint16_t length;
+};
+
+
+
+class SrsPsRtpPacket: public SrsRtpPacket
+{
+public:
+    SrsPsRtpPacket();
+    virtual ~SrsPsRtpPacket();
+public:
+    virtual srs_error_t decode(SrsBuffer* stream);
+};
+
+// A rtp connection which transport a stream.
+class SrsPsRtpConn: public ISrsUdpHandler
+{
+private:
+    SrsPithyPrint* pprint;
+    SrsUdpListener* listener;
+    SrsGb28181Conn* gb28181;
+    SrsPsRtpPacket* cache;
+    std::map<uint32_t, SrsSimpleStream*> cache_payload;
+    std::string session_id;
+    int _port;
+    uint32_t pre_timestamp;
+
+    SrsFileWriter ps_fw;
+    SrsFileWriter video_fw;
+    SrsFileWriter audio_fw;
+
+    bool first_keyframe_flag;
+    bool wait_first_keyframe;
+    bool audio_enable;
+  
+public:
+    SrsPsRtpConn(SrsGb28181Conn* r, int p, std::string sid, bool a, bool k);
+    virtual ~SrsPsRtpConn();
+
+private:
+   int64_t parse_ps_timestamp(const uint8_t* p);
+
+private:
+    bool can_send_ps_av_packet();
+    void dispose();
+public:
+    virtual int port();
+    virtual srs_error_t listen();
+// Interface ISrsUdpHandler
+public:
+    virtual srs_error_t on_udp_packet(const sockaddr* from, const int fromlen, char* buf, int nb_buf);
+    virtual srs_error_t on_ps_stream(char* ps_data, int ps_size, uint32_t timestamp);
+};
+
+class SrsGb28281ClientInfo {
+public:
+    SrsGb28281ClientInfo();
+    virtual ~SrsGb28281ClientInfo();
+
+public:
+    sockaddr* sock_from;
+    int sock_fromlen;
+    srs_netfd_t stfd;
+    SrsSipRequest *req;
+};
+
+enum Srs28181CtrlStatusType{
+     Srs28181Unkonw = 0,
+     Srs28181RegisterOk = 1,
+     Srs28181AliveOk = 2,
+     Srs28181InviteOk = 3,
+     Srs28181Trying = 4,
+     Srs28181Bye = 5,
+};
+
+class SrsGb28181Conn : public ISrsCoroutineHandler, public ISrsConnection
+{
+private:
+    std::string output_template;
+    SrsPithyPrint* pprint;
+public:
+    Srs28181CtrlStatusType register_status;
+    Srs28181CtrlStatusType alive_status;
+    Srs28181CtrlStatusType invite_status;
+    srs_utime_t register_time;
+    srs_utime_t alive_time;
+    srs_utime_t invite_time;
+    srs_utime_t recv_rtp_time;
+
+    std::string rtmp_url;
+    int reg_expires;
+
+private:
+    std::string session_id;
+    // video stream.
+    int video_id;
+    std::string video_codec;
+    SrsPsRtpConn* video_rtp;
+    // audio stream.
+    int audio_id;
+    std::string audio_codec;
+    int audio_sample_rate;
+    int audio_channel;
+    SrsPsRtpConn* audio_rtp;
+public:
+    SrsGb28281ClientInfo* info;
+private:
+    SrsStSocket* skt;
+    SrsSipStack* sip;
+    SrsGb28181Caster* caster;
+    SrsCoroutine* trd;
+private:
+    SrsSipRequest* req;
+    SrsSimpleRtmpClient* sdk;
+    SrsRtspJitter* vjitter;
+    SrsRtspJitter* ajitter;
+private:
+    SrsRawH264Stream* avc;
+    std::string h264_sps;
+    std::string h264_pps;
+    bool h264_sps_changed;
+    bool h264_pps_changed;
+    bool h264_sps_pps_sent;
+private:
+    SrsRawAacStream* aac;
+    std::string aac_specific_config;
+public:
+    SrsGb28181Conn(SrsGb28181Caster* c, std::string id);
+    virtual ~SrsGb28181Conn();
+public:
+    virtual srs_error_t serve();
+    virtual std::string remote_ip();
+    virtual void set_request_info(SrsSipRequest *req);
+    virtual std::string get_session_id();
+private:
+    virtual srs_error_t do_cycle();
+// internal methods
+public:
+    virtual srs_error_t start_rtp_listen(int port);
+    virtual srs_error_t stop_rtp_listen();
+// Interface ISrsOneCycleThreadHandler
+public:
+    virtual srs_error_t cycle();
+public:
+    virtual srs_error_t on_rtp_video(SrsSimpleStream* stream, int64_t dts, int keyframe);
+    virtual srs_error_t on_rtp_audio(SrsSimpleStream* stream, int64_t dts);
+private:
+    virtual srs_error_t write_h264_sps_pps(uint32_t dts, uint32_t pts);
+    virtual srs_error_t write_h264_ipb_frame(char* frame, int frame_size, uint32_t dts, uint32_t pts);
+    virtual srs_error_t write_audio_raw_frame(char* frame, int frame_size, SrsRawAacStreamCodec* codec, uint32_t dts);
+    virtual srs_error_t rtmp_write_packet(char type, uint32_t timestamp, char* data, int size);
+private:
+    // Connect to RTMP server.
+    virtual srs_error_t connect();
+    // Close the connection to RTMP server.
+    virtual void close();
+};
+
+class SrsGb28181Config
+{
+public:
+    std::string sip_host;
+    std::string sip_port;
+    std::string sip_serial;
+    std::string sip_realm;
+    int sip_ack_timeout;
+    int sip_keepalive_timeout;
+    int rtp_idle_timeout;
+    bool audio_enable;
+    std::string output;
+    int rtp_port_min;
+    int rtp_port_max;
+    int listen_port;
+    bool print_sip_message;
+    bool wait_keyframe;
+public:
+    SrsGb28181Config(SrsConfDirective* c);
+    virtual ~SrsGb28181Config();
+};
+
+//gb28181 conn manager
+class SrsGb28181Caster : public ISrsUdpHandler
+{
+private:
+    SrsGb28181Config *config;
+    // The key: port, value: whether used.
+    std::map<int, bool> used_ports;
+    SrsSipStack *sip;
+    srs_netfd_t lfd;
+private:
+    std::map<std::string, SrsGb28181Conn*> clients;
+    SrsCoroutineManager* manager;
+
+public:
+    SrsGb28181Caster(SrsConfDirective* c);
+    virtual ~SrsGb28181Caster();
+
+private:
+    srs_error_t fetch_or_create(SrsSipRequest* r, SrsGb28181Conn** gb28181);
+    virtual SrsGb28181Conn* fetch(const SrsSipRequest* r);
+    virtual void destroy();
+public:
+    // Alloc a rtp port from local ports pool.
+    // @param pport output the rtp port.
+    virtual srs_error_t alloc_port(int* pport);
+    // Free the alloced rtp port.
+    virtual void free_port(int lpmin, int lpmax);
+    virtual srs_error_t initialize();
+
+    virtual void set_stfd(srs_netfd_t fd);
+    virtual SrsGb28181Config GetGb28181Config();
+
+// Interface ISrsUdpHandler
+public:
+    virtual srs_error_t on_udp_packet(const sockaddr* from, const int fromlen, char* buf, int nb_buf);
+private:
+    virtual srs_error_t on_udp_bytes(std::string host, int port, char* buf, int nb_buf, sockaddr* from, int fromlen);
+// internal methods.
+public:
+    virtual srs_error_t send_message(sockaddr* f, int l, std::stringstream& ss);
+    virtual srs_error_t send_bye(SrsSipRequest *req, sockaddr *f, int l);
+    virtual srs_error_t send_ack(SrsSipRequest *req, sockaddr *f, int l);
+    virtual srs_error_t send_invite(SrsSipRequest *req, sockaddr *f, int l, int port);
+    virtual srs_error_t send_status(SrsSipRequest *req, sockaddr *f, int l);
+    virtual void remove(SrsGb28181Conn* conn);
+};
+
+#endif
+

trunk/src/app/srs_app_listener.cpp

@@ -59,6 +59,13 @@ srs_error_t ISrsUdpHandler::on_stfd_change(srs_netfd_t /*fd*/)
     return srs_success;
 }
 
+void ISrsUdpHandler::set_stfd(srs_netfd_t /*fd*/)
+{
+    
+}
+
+
+
 ISrsTcpHandler::ISrsTcpHandler()
 {
 }
@@ -104,6 +111,8 @@ srs_error_t SrsUdpListener::listen()
     if ((err = srs_udp_listen(ip, port, &lfd)) != srs_success) {
         return srs_error_wrap(err, "listen %s:%d", ip.c_str(), port);
     }
+
+    handler->set_stfd(lfd);
     
     srs_freep(trd);
     trd = new SrsSTCoroutine("udp", this);
@@ -206,4 +215,3 @@ srs_error_t SrsTcpListener::cycle()
     
     return err;
 }
-

trunk/src/app/srs_app_listener.hpp

@@ -43,6 +43,8 @@ public:
     // When fd changed, for instance, reload the listen port,
     // notify the handler and user can do something.
     virtual srs_error_t on_stfd_change(srs_netfd_t fd);
+    
+    virtual void set_stfd(srs_netfd_t fd);
 public:
     // When udp listener got a udp packet, notice server to process it.
     // @param type, the client type, used to create concrete connection,

trunk/src/app/srs_app_server.cpp

@@ -52,6 +52,7 @@ using namespace std;
 #include <srs_kernel_consts.hpp>
 #include <srs_app_thread.hpp>
 #include <srs_app_coworkers.hpp>
+#include <srs_app_gb28181.hpp>
 
 // system interval in srs_utime_t,
 // all resolution times should be times togother,
@@ -109,6 +110,8 @@ std::string srs_listener_type2string(SrsListenerType type)
             return "RTSP";
         case SrsListenerFlv:
             return "HTTP-FLV";
+        case SrsListenerGb28181:
+            return "GB28181-SIP over UDP";
         default:
             return "UNKONWN";
     }
@@ -301,7 +304,7 @@ srs_error_t SrsUdpStreamListener::listen(string i, int p)
     
     // the caller already ensure the type is ok,
     // we just assert here for unknown stream caster.
-    srs_assert(type == SrsListenerMpegTsOverUdp);
+    srs_assert(type == SrsListenerMpegTsOverUdp || type == SrsListenerGb28181);
     
     ip = i;
     port = p;
@@ -339,6 +342,22 @@ SrsUdpCasterListener::~SrsUdpCasterListener()
     srs_freep(caster);
 }
 
+
+SrsGb28181Listener::SrsGb28181Listener(SrsServer* svr, SrsListenerType t, SrsConfDirective* c) : SrsUdpStreamListener(svr, t, NULL)
+{
+    // the caller already ensure the type is ok,
+    // we just assert here for unknown stream caster.
+    srs_assert(type == SrsListenerGb28181);
+    if (type == SrsListenerGb28181) {
+        caster = new SrsGb28181Caster(c);
+    }
+}
+
+SrsGb28181Listener::~SrsGb28181Listener()
+{
+    srs_freep(caster);
+}
+
 SrsSignalManager* SrsSignalManager::instance = NULL;
 
 SrsSignalManager::SrsSignalManager(SrsServer* s)
@@ -1325,6 +1344,8 @@ srs_error_t SrsServer::listen_stream_caster()
             listener = new SrsRtspListener(this, SrsListenerRtsp, stream_caster);
         } else if (srs_stream_caster_is_flv(caster)) {
             listener = new SrsHttpFlvListener(this, SrsListenerFlv, stream_caster);
+        } else if (srs_stream_caster_is_gb28181(caster)) {
+            listener = new SrsGb28181Listener(this, SrsListenerGb28181, stream_caster);
         } else {
             return srs_error_new(ERROR_STREAM_CASTER_ENGINE, "invalid caster %s", caster.c_str());
         }

trunk/src/app/srs_app_server.hpp

@@ -69,6 +69,8 @@ enum SrsListenerType
     SrsListenerRtsp = 4,
     // TCP stream, FLV stream over HTTP.
     SrsListenerFlv = 5,
+    // UDP stream, gb28181 stream
+    SrsListenerGb28181 = 6,
 };
 
 // A common tcp listener, for RTMP/HTTP server.
@@ -156,6 +158,14 @@ public:
     virtual ~SrsUdpCasterListener();
 };
 
+// A UDP sip listener, for sip server.
+class SrsGb28181Listener :  public SrsUdpStreamListener
+{
+public:
+    SrsGb28181Listener(SrsServer* svr, SrsListenerType t, SrsConfDirective* c);
+    virtual ~SrsGb28181Listener();
+};
+
 // Convert signal to io,
 // @see: st-1.9/docs/notes.html
 class SrsSignalManager : public ISrsCoroutineHandler