Refine SRT code, with StateThread adpater

trunk/auto/options.sh

@@ -432,12 +432,6 @@ function apply_auto_options() {
       SRS_TOOL_LD=$SRS_TOOL_CC
     fi
 
-    # The SRT code in SRS requires c++11, although we build libsrt without c++11.
-    # TODO: FIXME: Remove c++11 code in SRT of SRS.
-    if [[ $SRS_SRT == YES ]]; then
-        SRS_CXX11=YES
-    fi
-
     # Enable FFmpeg fit for RTC to transcode audio from AAC to OPUS, if user enabled it.
     if [[ $SRS_RTC == YES && $SRS_FFMPEG_FIT == RESERVED ]]; then
         SRS_FFMPEG_FIT=YES

trunk/conf/srt.conf

@@ -18,6 +18,11 @@ http_server {
 
 srt_server {
     enabled on;
+    tsbpdmode off;
+    tlpktdrop off;
+    latency 0;
+    sendbuf 2000000;
+    recvbuf 2000000;
     listen 10080;
     maxbw 1000000000;
     connect_timeout 4000;

trunk/configure

@@ -228,6 +228,9 @@ MODULE_FILES=("srs_protocol_amf0" "srs_protocol_io" "srs_rtmp_stack"
         "srs_raw_avc" "srs_rtsp_stack" "srs_http_stack" "srs_protocol_kbps" "srs_protocol_json"
         "srs_protocol_format" "srs_service_log" "srs_service_st" "srs_service_http_client" "srs_service_http_conn"
         "srs_service_rtmp_conn" "srs_service_utility" "srs_service_conn")
+if [[ $SRS_SRT == YES ]]; then
+    MODULE_FILES+=("srs_service_st_srt")
+fi
 if [[ $SRS_RTC == YES ]]; then
     MODULE_FILES+=("srs_rtc_stun_stack")
     ModuleLibIncs+=(${LibSrtpRoot})
@@ -237,16 +240,6 @@ if [[ $SRS_FFMPEG_FIT == YES ]]; then
 fi
 PROTOCOL_INCS="src/protocol"; MODULE_DIR=${PROTOCOL_INCS} . auto/modules.sh
 PROTOCOL_OBJS="${MODULE_OBJS[@]}"
-#
-#srt protocol features.
-if [[ $SRS_SRT == YES ]]; then
-    MODULE_ID="SRT"
-    MODULE_DEPENDS=("CORE" "KERNEL" "PROTOCOL" "APP")
-    ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSSLRoot} ${LibSRTRoot})
-    MODULE_FILES=("srt_server" "srt_handle" "srt_conn" "srt_to_rtmp" "ts_demux" "srt_data" "srt_log")
-    SRT_INCS=(${LibSRTRoot} ${SrsSRTRoot}); MODULE_DIR=${SrsSRTRoot} . auto/modules.sh
-    SRT_OBJS="${MODULE_OBJS[@]}"
-fi
 
 #
 #App Module, for SRS server only.
@@ -273,6 +266,9 @@ MODULE_FILES=("srs_app_server" "srs_app_conn" "srs_app_rtmp_conn" "srs_app_sourc
         "srs_app_caster_flv" "srs_app_latest_version" "srs_app_uuid" "srs_app_process" "srs_app_ng_exec"
         "srs_app_hourglass" "srs_app_dash" "srs_app_fragment" "srs_app_dvr"
         "srs_app_coworkers" "srs_app_hybrid" "srs_app_threads")
+if [[ $SRS_SRT == YES ]]; then
+    MODULE_FILES+=("srs_app_srt_server" "srs_app_srt_listener" "srs_app_srt_conn" "srs_app_srt_utility" "srs_app_srt_source")
+fi
 if [[ $SRS_RTC == YES ]]; then
     MODULE_FILES+=("srs_app_rtc_conn" "srs_app_rtc_dtls" "srs_app_rtc_sdp"
         "srs_app_rtc_queue" "srs_app_rtc_server" "srs_app_rtc_source" "srs_app_rtc_api")

trunk/src/app/srs_app_config.cpp

@@ -2546,8 +2546,9 @@ srs_error_t SrsConfig::check_normal_config()
                 && n != "mss" && n != "latency" && n != "recvlatency"
                 && n != "peerlatency" && n != "tlpkdrop" && n != "connect_timeout"
                 && n != "sendbuf" && n != "recvbuf" && n != "payloadsize"
-                && n != "default_app" && n != "mix_correct" && n != "sei_filter") {
-                return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal srt_stream.%s", n.c_str());
+                && n != "default_app" && n != "mix_correct" && n != "sei_filter"
+                && n != "tlpktdrop" && n != "tsbpdmode") {
+                return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal srt_server.%s", n.c_str());
             }
         }
     }
@@ -6798,6 +6799,20 @@ int SrsConfig::get_srto_mss() {
     return atoi(conf->arg0().c_str());
 }
 
+bool SrsConfig::get_srto_tsbpdmode() {
+    static bool DEFAULT = false;
+    SrsConfDirective* conf = root->get("srt_server");
+    if (!conf) {
+        return DEFAULT;
+    }
+
+    conf = conf->get("tsbpdmode");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    return SRS_CONF_PERFER_TRUE(conf->arg0());
+}
+
 int SrsConfig::get_srto_latency() {
     static int DEFAULT = 120;
     SrsConfDirective* conf = root->get("srt_server");
@@ -6854,14 +6869,18 @@ bool SrsConfig::get_srt_sei_filter() {
     return SRS_CONF_PERFER_TRUE(conf->arg0());
 }
 
-bool SrsConfig::get_srto_tlpkdrop() {
+bool SrsConfig::get_srto_tlpktdrop() {
     static bool DEFAULT = true;
-    SrsConfDirective* conf = root->get("srt_server");
-    if (!conf) {
+    SrsConfDirective* srt_server_conf = root->get("srt_server");
+    if (!srt_server_conf) {
         return DEFAULT;
     }
 
-    conf = conf->get("tlpkdrop");
+    SrsConfDirective* conf = srt_server_conf->get("tlpkdrop");
+    if (! conf) {
+        // make it compatible tlpkdrop and tlpktdrop opt.
+        conf = srt_server_conf->get("tlpktdrop");
+    }
     if (!conf || conf->arg0().empty()) {
         return DEFAULT;
     }
@@ -6882,6 +6901,20 @@ int SrsConfig::get_srto_conntimeout() {
     return atoi(conf->arg0().c_str());
 }
 
+int SrsConfig::get_srto_peeridletimeout() {
+    static int DEFAULT = 10000;
+    SrsConfDirective* conf = root->get("srt_server");
+    if (!conf) {
+        return DEFAULT;
+    }
+
+    conf = conf->get("peer_idle_timeout");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    return atoi(conf->arg0().c_str());
+}
+
 int SrsConfig::get_srto_sendbuf() {
     static int64_t DEFAULT = 8192 * (1500-28);
     SrsConfDirective* conf = root->get("srt_server");

trunk/src/app/srs_app_config.hpp

@@ -636,6 +636,8 @@ class SrsConfig
     virtual int get_srto_maxbw();
     // Get the srt SRTO_MSS, Maximum Segment Size, default is 1500.
     virtual int get_srto_mss();
+    // Get the srt SRTO_TSBPDMODE, timestamp base packet delivery mode, default is false.
+    virtual bool get_srto_tsbpdmode();
     // Get the srt SRTO_LATENCY, latency, default is 0 which means peer/recv latency is 120ms.
     virtual int get_srto_latency();
     // Get the srt SRTO_RCVLATENCY, recv latency, default is 120ms.
@@ -644,10 +646,12 @@ class SrsConfig
     virtual int get_srto_peer_latency();
     // Get the srt h264 sei filter, default is on, it will drop h264 sei packet.
     virtual bool get_srt_sei_filter();
-    // Get the srt SRTO_TLPKDROP, Too-late Packet Drop, default is true.
-    virtual bool get_srto_tlpkdrop();
+    // Get the srt SRTO_TLPKTDROP, Too-late Packet Drop, default is true.
+    virtual bool get_srto_tlpktdrop();
     // Get the srt SRTO_CONNTIMEO, connection timeout, default is 3000ms.
     virtual int get_srto_conntimeout();
+    // Get the srt SRTO_PEERIDLETIMEO, peer idle timeout, default is 10000ms.
+    virtual int get_srto_peeridletimeout();
     // Get the srt SRTO_SNDBUF, send buffer, default is 8192 × (1500-28).
     virtual int get_srto_sendbuf();
     // Get the srt SRTO_RCVBUF, recv buffer, default is 8192 × (1500-28).

trunk/src/app/srs_app_pithy_print.cpp

@@ -219,6 +219,13 @@ SrsPithyPrint::SrsPithyPrint(int _stage_id)
 // for the rtc recv
 #define SRS_CONSTS_STAGE_RTC_RECV 14
 
+#ifdef SRS_SRT
+// the pithy stage for srt play clients.
+#define SRS_CONSTS_STAGE_SRT_PLAY 15
+// the pithy stage for srt publish clients.
+#define SRS_CONSTS_STAGE_SRT_PUBLISH 16
+#endif
+
 SrsPithyPrint* SrsPithyPrint::create_rtmp_play()
 {
     return new SrsPithyPrint(SRS_CONSTS_STAGE_PLAY_USER);
@@ -289,6 +296,18 @@ SrsPithyPrint* SrsPithyPrint::create_rtc_recv(int fd)
     return new SrsPithyPrint(fd<<16 | SRS_CONSTS_STAGE_RTC_RECV);
 }
 
+#ifdef SRS_SRT
+SrsPithyPrint* SrsPithyPrint::create_srt_play()
+{
+    return new SrsPithyPrint(SRS_CONSTS_STAGE_SRT_PLAY);
+}
+
+SrsPithyPrint* SrsPithyPrint::create_srt_publish()
+{
+    return new SrsPithyPrint(SRS_CONSTS_STAGE_SRT_PUBLISH);
+}
+#endif
+
 SrsPithyPrint::~SrsPithyPrint()
 {
     leave_stage();

trunk/src/app/srs_app_pithy_print.hpp

@@ -130,6 +130,10 @@ class SrsPithyPrint
     // For RTC sender and receiver, we create printer for each fd.
     static SrsPithyPrint* create_rtc_send(int fd);
     static SrsPithyPrint* create_rtc_recv(int fd);
+#ifdef SRS_SRT
+    static SrsPithyPrint* create_srt_play();
+    static SrsPithyPrint* create_srt_publish();
+#endif
     virtual ~SrsPithyPrint();
 private:
     // Enter the specified stage, return the client id.