Use SrsAsyncCallWorker in http hooks instead, to covert to async call. (#2542)

* Use SrsAsyncCallWorker in http hooks instead, to covert to async call.

* delete invalid function

trunk/src/app/srs_app_rtc_conn.cpp

@@ -351,6 +351,53 @@ srs_error_t SrsRtcPLIWorker::cycle()
 
     return err;
 }
+
+SrsRtcAsyncCallOnStop::SrsRtcAsyncCallOnStop(SrsContextId c, SrsRequest * r)
+{
+    cid = c;
+    req = r->copy();
+}
+
+SrsRtcAsyncCallOnStop::~SrsRtcAsyncCallOnStop()
+{
+    srs_freep(req);
+}
+
+srs_error_t SrsRtcAsyncCallOnStop::call()
+{
+    srs_error_t err = srs_success;
+
+    if (!_srs_config->get_vhost_http_hooks_enabled(req->vhost)) {
+        return err;
+    }
+
+    // the http hooks will cause context switch,
+    // so we must copy all hooks for the on_connect may freed.
+    // @see https://github.com/ossrs/srs/issues/475
+    vector<string> hooks;
+
+    if (true) {
+        SrsConfDirective* conf = _srs_config->get_vhost_on_stop(req->vhost);
+
+        if (!conf) {
+            return err;
+        }
+
+        hooks = conf->args;
+    }
+
+    for (int i = 0; i < (int)hooks.size(); i++) {
+        std::string url = hooks.at(i);
+        SrsHttpHooks::on_stop(url, req);
+    }
+
+    return err;
+}
+
+std::string SrsRtcAsyncCallOnStop::to_string()
+{
+    return std::string("");
+}
 
 SrsRtcPlayStream::SrsRtcPlayStream(SrsRtcConnection* s, const SrsContextId& cid)
 {
@@ -379,9 +426,8 @@ SrsRtcPlayStream::SrsRtcPlayStream(SrsRtcConnection* s, const SrsContextId& cid)
 
 SrsRtcPlayStream::~SrsRtcPlayStream()
 {
-    // TODO: FIXME: Use SrsAsyncCallWorker in http hooks instead, to covert to async call.
     if (req_) {
-        http_hooks_on_stop();		
+        session_->server_->exec_async_work(new SrsRtcAsyncCallOnStop(cid_, req_));
     }
 
     // TODO: FIXME: Should not do callback in de-constructor?
@@ -871,35 +917,6 @@ srs_error_t SrsRtcPlayStream::do_request_keyframe(uint32_t ssrc, SrsContextId ci
     return err;
 }
 
-void SrsRtcPlayStream::http_hooks_on_stop()
-{
-    if (!_srs_config->get_vhost_http_hooks_enabled(req_->vhost)) {
-        return;
-    }
-
-    // the http hooks will cause context switch,
-    // so we must copy all hooks for the on_connect may freed.
-    // @see https://github.com/ossrs/srs/issues/475
-    vector<string> hooks;
-
-    if (true) {
-        SrsConfDirective* conf = _srs_config->get_vhost_on_stop(req_->vhost);
-
-        if (!conf) {
-            return;
-        }
-
-        hooks = conf->args;
-    }
-
-    for (int i = 0; i < (int)hooks.size(); i++) {
-        std::string url = hooks.at(i);
-        SrsHttpHooks::on_stop(url, req_);
-    }
-
-    return;
-}
-
 SrsRtcPublishRtcpTimer::SrsRtcPublishRtcpTimer(SrsRtcPublishStream* p) : p_(p)
 {
     _srs_hybrid->timer1s()->subscribe(this);
@@ -979,6 +996,54 @@ srs_error_t SrsRtcPublishTwccTimer::on_timer(srs_utime_t interval)
     return err;
 }
 
+
+SrsRtcAsyncCallOnUnpublish::SrsRtcAsyncCallOnUnpublish(SrsContextId c, SrsRequest * r)
+{
+    cid = c;
+    req = r->copy();
+}
+
+SrsRtcAsyncCallOnUnpublish::~SrsRtcAsyncCallOnUnpublish()
+{
+    srs_freep(req);
+}
+
+srs_error_t SrsRtcAsyncCallOnUnpublish::call()
+{
+    srs_error_t err = srs_success;
+
+    if (!_srs_config->get_vhost_http_hooks_enabled(req->vhost)) {
+        return err;
+    }
+
+    // the http hooks will cause context switch,
+    // so we must copy all hooks for the on_connect may freed.
+    // @see https://github.com/ossrs/srs/issues/475
+    vector<string> hooks;
+
+    if (true) {
+        SrsConfDirective* conf = _srs_config->get_vhost_on_unpublish(req->vhost);
+
+        if (!conf) {
+            return err;
+        }
+
+        hooks = conf->args;
+    }
+
+    for (int i = 0; i < (int)hooks.size(); i++) {
+        std::string url = hooks.at(i);
+        SrsHttpHooks::on_unpublish(url, req);
+    }
+
+    return err;
+}
+
+std::string SrsRtcAsyncCallOnUnpublish::to_string()
+{
+    return std::string("");
+}
+
 SrsRtcPublishStream::SrsRtcPublishStream(SrsRtcConnection* session, const SrsContextId& cid)
 {
     cid_ = cid;
@@ -988,7 +1053,7 @@ SrsRtcPublishStream::SrsRtcPublishStream(SrsRtcConnection* session, const SrsCon
     pli_epp = new SrsErrorPithyPrint();
     twcc_epp_ = new SrsErrorPithyPrint(3.0);
 
-    req = NULL;
+    req_ = NULL;
     source = NULL;
     nn_simulate_nack_drop = 0;
     nack_enabled_ = false;
@@ -1009,8 +1074,8 @@ SrsRtcPublishStream::SrsRtcPublishStream(SrsRtcConnection* session, const SrsCon
 
 SrsRtcPublishStream::~SrsRtcPublishStream()
 {
-    if (req) {
+    if (req_) {
-        http_hooks_on_unpublish();
+        session_->server_->exec_async_work(new SrsRtcAsyncCallOnUnpublish(cid_, req_));
     }
 
     srs_freep(timer_rtcp_);
@@ -1027,7 +1092,7 @@ SrsRtcPublishStream::~SrsRtcPublishStream()
     // it must be called after source stream unpublish (set source stream is_created=false).
     // if not, it lead to republish failed.
     if (_srs_rtc_hijacker) {
-        _srs_rtc_hijacker->on_stop_publish(session_, this, req);
+        _srs_rtc_hijacker->on_stop_publish(session_, this, req_);
     }
 
     for (int i = 0; i < (int)video_tracks_.size(); ++i) {
@@ -1045,7 +1110,7 @@ SrsRtcPublishStream::~SrsRtcPublishStream()
     srs_freep(pli_worker_);
     srs_freep(twcc_epp_);
     srs_freep(pli_epp);
-    srs_freep(req);
+    srs_freep(req_);
 	
     // update the statistic when client coveried.
     SrsStatistic* stat = SrsStatistic::instance();
@@ -1056,7 +1121,7 @@ srs_error_t SrsRtcPublishStream::initialize(SrsRequest* r, SrsRtcSourceDescripti
 {
     srs_error_t err = srs_success;
 
-    req = r->copy();
+    req_ = r->copy();
 
     if (stream_desc->audio_track_desc_) {
         audio_tracks_.push_back(new SrsRtcAudioRecvTrack(session_, stream_desc->audio_track_desc_));
@@ -1083,10 +1148,10 @@ srs_error_t SrsRtcPublishStream::initialize(SrsRequest* r, SrsRtcSourceDescripti
         rtcp_twcc_.set_media_ssrc(media_ssrc);
     }
 
-    nack_enabled_ = _srs_config->get_rtc_nack_enabled(req->vhost);
+    nack_enabled_ = _srs_config->get_rtc_nack_enabled(req_->vhost);
-    nack_no_copy_ = _srs_config->get_rtc_nack_no_copy(req->vhost);
+    nack_no_copy_ = _srs_config->get_rtc_nack_no_copy(req_->vhost);
-    pt_to_drop_ = (uint16_t)_srs_config->get_rtc_drop_for_pt(req->vhost);
+    pt_to_drop_ = (uint16_t)_srs_config->get_rtc_drop_for_pt(req_->vhost);
-    twcc_enabled_ = _srs_config->get_rtc_twcc_enabled(req->vhost);
+    twcc_enabled_ = _srs_config->get_rtc_twcc_enabled(req_->vhost);
 
     // No TWCC when negotiate, disable it.
     if (twcc_id <= 0) {
@@ -1107,14 +1172,14 @@ srs_error_t SrsRtcPublishStream::initialize(SrsRequest* r, SrsRtcSourceDescripti
     }
 
     // Setup the publish stream in source to enable PLI as such.
-    if ((err = _srs_rtc_sources->fetch_or_create(req, &source)) != srs_success) {
+    if ((err = _srs_rtc_sources->fetch_or_create(req_, &source)) != srs_success) {
         return srs_error_wrap(err, "create source");
     }
     source->set_publish_stream(this);
 
     // Bridge to rtmp
 #if defined(SRS_RTC) && defined(SRS_FFMPEG_FIT)
-    bool rtc_to_rtmp = _srs_config->get_rtc_to_rtmp(req->vhost);
+    bool rtc_to_rtmp = _srs_config->get_rtc_to_rtmp(req_->vhost);
     if (rtc_to_rtmp) {
         SrsLiveSource *rtmp = NULL;
         if ((err = _srs_sources->fetch_or_create(r, _srs_hybrid->srs()->instance(), &rtmp)) != srs_success) {
@@ -1160,14 +1225,14 @@ srs_error_t SrsRtcPublishStream::start()
     }
 
     if (_srs_rtc_hijacker) {
-        if ((err = _srs_rtc_hijacker->on_start_publish(session_, this, req)) != srs_success) {
+        if ((err = _srs_rtc_hijacker->on_start_publish(session_, this, req_)) != srs_success) {
             return srs_error_wrap(err, "on start publish");
         }
     }
 	
     // update the statistic when client discoveried.
     SrsStatistic* stat = SrsStatistic::instance();
-    if ((err = stat->on_client(cid_.c_str(), req, session_, SrsRtcConnPublish)) != srs_success) {
+    if ((err = stat->on_client(cid_.c_str(), req_, session_, SrsRtcConnPublish)) != srs_success) {
         return srs_error_wrap(err, "rtc: stat client");
     }
 
@@ -1385,7 +1450,7 @@ srs_error_t SrsRtcPublishStream::do_on_rtp_plaintext(SrsRtpPacket*& pkt, SrsBuff
     }
 
     if (_srs_rtc_hijacker) {
-        if ((err = _srs_rtc_hijacker->on_rtp_packet(session_, this, req, pkt)) != srs_success) {
+        if ((err = _srs_rtc_hijacker->on_rtp_packet(session_, this, req_, pkt)) != srs_success) {
             return srs_error_wrap(err, "on rtp packet");
         }
     }
@@ -1677,33 +1742,6 @@ void SrsRtcPublishStream::update_send_report_time(uint32_t ssrc, const SrsNtp& n
     }
 }
 
-void SrsRtcPublishStream::http_hooks_on_unpublish()
-{
-    if (!_srs_config->get_vhost_http_hooks_enabled(req->vhost)) {
-        return;
-    }
-
-    // the http hooks will cause context switch,
-    // so we must copy all hooks for the on_connect may freed.
-    // @see https://github.com/ossrs/srs/issues/475
-    vector<string> hooks;
-
-    if (true) {
-        SrsConfDirective* conf = _srs_config->get_vhost_on_unpublish(req->vhost);
-
-        if (!conf) {
-            return;
-        }
-
-        hooks = conf->args;
-    }
-
-    for (int i = 0; i < (int)hooks.size(); i++) {
-        std::string url = hooks.at(i);
-        SrsHttpHooks::on_unpublish(url, req);
-    }
-}
-
 ISrsRtcConnectionHijacker::ISrsRtcConnectionHijacker()
 {
 }
@@ -1753,7 +1791,7 @@ srs_error_t SrsRtcConnectionNackTimer::on_timer(srs_utime_t interval)
 
 SrsRtcConnection::SrsRtcConnection(SrsRtcServer* s, const SrsContextId& cid)
 {
-    req = NULL;
+    req_ = NULL;
     cid_ = cid;
     hijacker_ = NULL;
 
@@ -1820,7 +1858,7 @@ SrsRtcConnection::~SrsRtcConnection()
     srs_freep(cache_buffer_);
 
     srs_freep(transport_);
-    srs_freep(req);
+    srs_freep(req_);
     srs_freep(pp_address_change);
     srs_freep(pli_epp);
 }
@@ -2019,7 +2057,7 @@ srs_error_t SrsRtcConnection::initialize(SrsRequest* r, bool dtls, bool srtp, st
     srs_error_t err = srs_success;
 
     username_ = username;
-    req = r->copy();
+    req_ = r->copy();
 
     if (!srtp) {
         srs_freep(transport_);
@@ -2036,10 +2074,10 @@ srs_error_t SrsRtcConnection::initialize(SrsRequest* r, bool dtls, bool srtp, st
     }
 
     // TODO: FIXME: Support reload.
-    session_timeout = _srs_config->get_rtc_stun_timeout(req->vhost);
+    session_timeout = _srs_config->get_rtc_stun_timeout(req_->vhost);
     last_stun_time = srs_get_system_time();
 
-    nack_enabled_ = _srs_config->get_rtc_nack_enabled(req->vhost);
+    nack_enabled_ = _srs_config->get_rtc_nack_enabled(req_->vhost);
 
     srs_trace("RTC init session, user=%s, url=%s, encrypt=%u/%u, DTLS(role=%s, version=%s), timeout=%dms, nack=%d",
         username.c_str(), r->get_stream_url().c_str(), dtls, srtp, cfg->dtls_role.c_str(), cfg->dtls_version.c_str(),
@@ -2691,7 +2729,7 @@ srs_error_t SrsRtcConnection::on_binding_request(SrsStunPacket* r)
 
     ++_srs_pps_sstuns->sugar;
 
-    bool strict_check = _srs_config->get_rtc_stun_strict_check(req->vhost);
+    bool strict_check = _srs_config->get_rtc_stun_strict_check(req_->vhost);
     if (strict_check && r->get_ice_controlled()) {
         // @see: https://tools.ietf.org/html/draft-ietf-ice-rfc5245bis-00#section-6.1.3.1
         // TODO: Send 487 (Role Conflict) error response.

trunk/src/app/srs_app_rtc_conn.hpp

@@ -23,6 +23,7 @@
 #include <srs_app_rtc_dtls.hpp>
 #include <srs_service_conn.hpp>
 #include <srs_app_conn.hpp>
+#include <srs_app_async_call.hpp>
 
 #include <string>
 #include <map>
@@ -195,6 +196,20 @@ public:
     virtual srs_error_t cycle();
 };
 
+// the rtc on_stop async call.
+class SrsRtcAsyncCallOnStop : public ISrsAsyncCallTask
+{
+private:
+    SrsContextId cid;
+    SrsRequest* req;
+public:
+    SrsRtcAsyncCallOnStop(SrsContextId c, SrsRequest* r);
+    virtual ~SrsRtcAsyncCallOnStop();
+public:
+    virtual srs_error_t call();
+    virtual std::string to_string();
+};
+
 // A RTC play stream, client pull and play stream from SRS.
 class SrsRtcPlayStream : public ISrsCoroutineHandler, public ISrsReloadHandler
     , public ISrsRtcPLIWorkerHandler, public ISrsRtcSourceChangeCallback
@@ -264,8 +279,6 @@ private:
 // Interface ISrsRtcPLIWorkerHandler
 public:
     virtual srs_error_t do_request_keyframe(uint32_t ssrc, SrsContextId cid);
-private:
-    virtual void http_hooks_on_stop();
 };
 
 // A fast timer for publish stream, for RTCP feedback.
@@ -294,6 +307,20 @@ private:
     srs_error_t on_timer(srs_utime_t interval);
 };
 
+// the rtc on_unpublish async call.
+class SrsRtcAsyncCallOnUnpublish : public ISrsAsyncCallTask
+{
+private:
+    SrsContextId cid;
+    SrsRequest* req;
+public:
+    SrsRtcAsyncCallOnUnpublish(SrsContextId c, SrsRequest* r);
+    virtual ~SrsRtcAsyncCallOnUnpublish();
+public:
+    virtual srs_error_t call();
+    virtual std::string to_string();
+};
+
 // A RTC publish stream, client push and publish stream to SRS.
 class SrsRtcPublishStream : public ISrsRtspPacketDecodeHandler
     , public ISrsRtcPublishStream, public ISrsRtcPLIWorkerHandler
@@ -319,7 +346,7 @@ private:
     bool request_keyframe_;
     SrsErrorPithyPrint* pli_epp;
 private:
-    SrsRequest* req;
+    SrsRequest* req_;
     SrsRtcSource* source;
     // Simulators.
     int nn_simulate_nack_drop;
@@ -377,8 +404,6 @@ private:
     SrsRtcVideoRecvTrack* get_video_track(uint32_t ssrc);
     void update_rtt(uint32_t ssrc, int rtt);
     void update_send_report_time(uint32_t ssrc, const SrsNtp& ntp, uint32_t rtp_time);
-private:
-    virtual void http_hooks_on_unpublish();
 };
 
 // Callback for RTC connection.
@@ -451,8 +476,7 @@ private:
 private:
     // For each RTC session, we use a specified cid for debugging logs.
     SrsContextId cid_;
-    // TODO: FIXME: Rename to req_.
+    SrsRequest* req_;
-    SrsRequest* req;
     SrsSdp remote_sdp;
     SrsSdp local_sdp;
 private:

trunk/src/app/srs_app_rtc_server.cpp

@@ -243,6 +243,7 @@ SrsRtcServer::SrsRtcServer()
 {
     handler = NULL;
     hijacker = NULL;
+    async = new SrsAsyncCallWorker();
 
     _srs_config->subscribe(this);
 }
@@ -258,6 +259,9 @@ SrsRtcServer::~SrsRtcServer()
             srs_freep(listener);
         }
     }
+
+    async->stop();
+    srs_freep(async);
 }
 
 srs_error_t SrsRtcServer::initialize()
@@ -273,6 +277,8 @@ srs_error_t SrsRtcServer::initialize()
         return srs_error_wrap(err, "black hole");
     }
 
+    async->start();
+
     return err;
 }
 
@@ -289,6 +295,11 @@ void SrsRtcServer::set_handler(ISrsRtcServerHandler* h)
 void SrsRtcServer::set_hijacker(ISrsRtcServerHijacker* h)
 {
     hijacker = h;
+}
+
+srs_error_t SrsRtcServer::exec_async_work(ISrsAsyncCallTask * t)
+{
+    return async->execute(t);
 }
 
 srs_error_t SrsRtcServer::listen_udp()

trunk/src/app/srs_app_rtc_server.hpp

@@ -15,6 +15,7 @@
 #include <srs_app_hourglass.hpp>
 #include <srs_app_hybrid.hpp>
 #include <srs_app_rtc_sdp.hpp>
+#include <srs_app_async_call.hpp>
 
 #include <string>
 
@@ -94,6 +95,7 @@ private:
     std::vector<SrsUdpMuxListener*> listeners;
     ISrsRtcServerHandler* handler;
     ISrsRtcServerHijacker* hijacker;
+    SrsAsyncCallWorker* async;
 public:
     SrsRtcServer();
     virtual ~SrsRtcServer();
@@ -106,6 +108,7 @@ public:
     // Set the handler for server events.
     void set_handler(ISrsRtcServerHandler* h);
     void set_hijacker(ISrsRtcServerHijacker* h);
+    srs_error_t exec_async_work(ISrsAsyncCallTask* t);
 public:
     // TODO: FIXME: Support gracefully quit.
     // TODO: FIXME: Support reload.