AI: Always enable SRT protocol. v7.0.69 (#4460)

Co-authored-by: OSSRS-AI <winlinam@gmail.com>

trunk/auto/auto_headers.sh

@@ -50,11 +50,7 @@ else
     srs_undefine_macro "SRS_HDS" $SRS_AUTO_HEADERS_H
 fi
 
-if [[ $SRS_SRT == YES ]]; then
+srs_define_macro "SRS_SRT" $SRS_AUTO_HEADERS_H
-    srs_define_macro "SRS_SRT" $SRS_AUTO_HEADERS_H
-else
-    srs_undefine_macro "SRS_SRT" $SRS_AUTO_HEADERS_H
-fi
 
 if [[ $SRS_CXX11 == YES ]]; then
     srs_define_macro "SRS_CXX11" $SRS_AUTO_HEADERS_H

trunk/auto/depends.sh

@@ -680,10 +680,10 @@ fi
 #####################################################################################
 # SRT module, https://github.com/ossrs/srs/issues/1147#issuecomment-577469119
 #####################################################################################
-if [[ $SRS_SRT == YES && $SRS_USE_SYS_SRT == YES ]]; then
+if [[ $SRS_USE_SYS_SRT == YES ]]; then
     echo "Warning: Use system libsrt, without compiling srt."
 fi
-if [[ $SRS_SRT == YES && $SRS_USE_SYS_SRT == NO ]]; then
+if [[ $SRS_USE_SYS_SRT == NO ]]; then
     # Always disable c++11 for libsrt, because only the srt-app requres it.
     LIBSRT_OPTIONS="--enable-apps=0  --enable-static=1 --enable-c++11=0"
     CMAKE_VERSION=$(cmake --version | head -n1 | cut -d' ' -f3)

trunk/auto/options.sh

@@ -575,6 +575,12 @@ function apply_auto_options() {
         SRS_RTC=YES
     fi
 
+    # Force enable SRT always - SRT support is required
+    if [[ $SRS_SRT != YES ]]; then
+        echo "Warning: SRT support is always enabled. Forcing SRT mode."
+        SRS_SRT=YES
+    fi
+
     if [[ $SRS_SRTP_ASM == YES && $SRS_NASM == NO ]]; then
         echo "Disable SRTP-ASM, because NASM is disabled."
         SRS_SRTP_ASM=NO

trunk/configure

@@ -178,16 +178,14 @@ if [[ $SRS_GPERF_MD == YES ]]; then
     LibGperfFile="${SRS_OBJS}/gperf/lib/libtcmalloc_debug.a";
 fi
 
-# srt code path
+# srt code path - always enabled
-if [[ $SRS_SRT == YES ]]; then
+LibSRTRoot="${SRS_OBJS}/srt/include"; LibSRTfile="${SRS_OBJS}/srt/lib/libsrt.a"
-    LibSRTRoot="${SRS_OBJS}/srt/include"; LibSRTfile="${SRS_OBJS}/srt/lib/libsrt.a"
+if [[ $SRS_SHARED_SRT == YES ]]; then LibSRTfile="-L${SRS_OBJS}/srt/lib -lsrt"; fi
-    if [[ $SRS_SHARED_SRT == YES ]]; then LibSRTfile="-L${SRS_OBJS}/srt/lib -lsrt"; fi
+if [[ $SRS_USE_SYS_SRT == YES ]]; then
-    if [[ $SRS_USE_SYS_SRT == YES ]]; then
+    LibSRTRoot=""; LibSRTfile="libsrt.a"
-        LibSRTRoot=""; LibSRTfile="libsrt.a"
+    if [[ $SRS_SHARED_SRT == YES ]]; then
-        if [[ $SRS_SHARED_SRT == YES ]]; then 
+        LibSRTfile="";
-            LibSRTfile=""; 
+        SrsLinkOptions="${SrsLinkOptions} -lsrt";
-            SrsLinkOptions="${SrsLinkOptions} -lsrt";
-        fi
     fi
 fi
 
@@ -266,10 +264,9 @@ MODULE_FILES=("srs_protocol_amf0" "srs_protocol_io" "srs_protocol_conn" "srs_pro
         "srs_protocol_raw_avc" "srs_protocol_http_stack" "srs_protocol_kbps" "srs_protocol_json"
         "srs_protocol_format" "srs_protocol_log" "srs_protocol_st" "srs_protocol_http_client"
         "srs_protocol_http_conn" "srs_protocol_rtmp_conn" "srs_protocol_protobuf")
-if [[ $SRS_SRT == YES ]]; then
+# Always include SRT protocol
-    MODULE_FILES+=("srs_protocol_srt")
+MODULE_FILES+=("srs_protocol_srt")
-    ModuleLibIncs+=(${LibSRTRoot})
+ModuleLibIncs+=(${LibSRTRoot})
-fi
 MODULE_FILES+=("srs_protocol_rtc_stun" "srs_protocol_rtp")
 if [[ $SRS_RTSP == YES ]]; then
     MODULE_FILES+=("srs_protocol_rtsp_stack")
@@ -300,9 +297,8 @@ MODULE_FILES=("srs_app_server" "srs_app_conn" "srs_app_rtmp_conn" "srs_app_sourc
         "srs_app_hourglass" "srs_app_dash" "srs_app_fragment" "srs_app_dvr"
         "srs_app_coworkers" "srs_app_circuit_breaker"
         "srs_app_stream_token")
-if [[ $SRS_SRT == YES ]]; then
+# Always include SRT app modules
-    MODULE_FILES+=("srs_app_srt_server" "srs_app_srt_listener" "srs_app_srt_conn" "srs_app_srt_utility" "srs_app_srt_source")
+MODULE_FILES+=("srs_app_srt_server" "srs_app_srt_listener" "srs_app_srt_conn" "srs_app_srt_utility" "srs_app_srt_source")
-fi
 MODULE_FILES+=("srs_app_rtc_conn" "srs_app_rtc_dtls" "srs_app_rtc_sdp" "srs_app_rtc_network"
     "srs_app_rtc_queue" "srs_app_rtc_server" "srs_app_rtc_source" "srs_app_rtc_api")
 if [[ $SRS_RTSP == YES ]]; then
@@ -344,19 +340,17 @@ ModuleLibFiles=(${LibSTfile} ${LibSSLfile} ${LibGperfFile} ${LibSrtpFile})
 if [[ $SRS_FFMPEG_FIT == YES ]]; then
     ModuleLibFiles+=("${LibFfmpegFile[*]}")
 fi
-if [[ $SRS_SRT == YES ]]; then
+# Always include SRT library files
-    ModuleLibFiles+=("${LibSRTfile[*]}")
+ModuleLibFiles+=("${LibSRTfile[*]}")
-fi
 # all depends objects
 MODULE_OBJS="${CORE_OBJS[@]} ${KERNEL_OBJS[@]} ${PROTOCOL_OBJS[@]} ${APP_OBJS[@]} ${SERVER_OBJS[@]}"
 ModuleLibIncs=(${SRS_OBJS} ${LibSTRoot} ${LibGperfRoot} ${LibSSLRoot} ${LibSrtpRoot})
 if [[ $SRS_FFMPEG_FIT == YES ]]; then
     ModuleLibIncs+=("${LibFfmpegRoot[*]}")
 fi
-if [[ $SRS_SRT == YES ]]; then
+# Always include SRT libraries
-    ModuleLibIncs+=(${LibSRTRoot})
+ModuleLibIncs+=(${LibSRTRoot})
-    MODULE_OBJS="${MODULE_OBJS} ${SRT_OBJS[@]}"
+MODULE_OBJS="${MODULE_OBJS} ${SRT_OBJS[@]}"
-fi
 LINK_OPTIONS="${LDFLAGS} ${SrsLinkOptions}${SrsGprofLink}${SrsGperfLink}"
 #
 # srs: srs(simple rtmp server) over st(state-threads)
@@ -368,9 +362,8 @@ ModuleLibFiles=(${LibSTfile} ${LibSSLfile} ${LibGperfFile} ${LibSrtpFile})
 if [[ $SRS_FFMPEG_FIT == YES ]]; then
     ModuleLibFiles+=("${LibFfmpegFile[*]}")
 fi
-if [[ $SRS_SRT == YES ]]; then
+# Always include SRT library files
-    ModuleLibFiles+=("${LibSRTfile[*]}")
+ModuleLibFiles+=("${LibSRTfile[*]}")
-fi
 # For utest on mac.
 # @see https://github.com/protocolbuffers/protobuf/issues/51#issuecomment-111044468
 if [[ $SRS_OSX == YES ]]; then
@@ -385,9 +378,8 @@ if [[ $SRS_UTEST == YES ]]; then
         "srs_utest_protocol" "srs_utest_protocol2" "srs_utest_kernel2" "srs_utest_protocol3"
         "srs_utest_st" "srs_utest_rtc2" "srs_utest_rtc3" "srs_utest_fmp4" "srs_utest_source_lock"
         "srs_utest_stream_token")
-    if [[ $SRS_SRT == YES ]]; then
+    # Always include SRT utest
-        MODULE_FILES+=("srs_utest_srt")
+    MODULE_FILES+=("srs_utest_srt")
-    fi
     if [[ $SRS_GB28181 == YES ]]; then
         MODULE_FILES+=("srs_utest_gb28181")
     fi
@@ -395,16 +387,14 @@ if [[ $SRS_UTEST == YES ]]; then
     if [[ $SRS_FFMPEG_FIT == YES ]]; then
         ModuleLibIncs+=("${LibFfmpegRoot[*]}")
     fi
-    if [[ $SRS_SRT == YES ]]; then
+    # Always include SRT library includes
-        ModuleLibIncs+=("${LibSRTRoot[*]}")
+    ModuleLibIncs+=("${LibSRTRoot[*]}")
-    fi
     ModuleLibFiles=(${LibSTfile} ${LibSSLfile} ${LibSrtpFile})
     if [[ $SRS_FFMPEG_FIT == YES ]]; then
         ModuleLibFiles+=("${LibFfmpegFile[*]}")
     fi
-    if [[ $SRS_SRT == YES ]]; then
+    # Always include SRT library files
-        ModuleLibFiles+=("${LibSRTfile[*]}")
+    ModuleLibFiles+=("${LibSRTfile[*]}")
-    fi
     MODULE_DEPENDS=("CORE" "KERNEL" "PROTOCOL" "APP")
     MODULE_OBJS="${CORE_OBJS[@]} ${KERNEL_OBJS[@]} ${PROTOCOL_OBJS[@]} ${APP_OBJS[@]} ${SRT_OBJS[@]}"
     LINK_OPTIONS="${LDFLAGS} -lpthread ${SrsLinkOptions}" MODULE_DIR="src/utest" APP_NAME="srs_utest" . $SRS_WORKDIR/auto/utest.sh
@@ -636,11 +626,7 @@ else
     echo -e "${GREEN}Warning: GB28181 is disabled.${BLACK}"
 fi
 echo -e "${GREEN}Experiment: HEVC/H.265 is enabled. https://github.com/ossrs/srs/issues/465${BLACK}"
-if [[ $SRS_SRT == YES ]]; then
+echo -e "${YELLOW}Experiment: SRT is enabled. https://github.com/ossrs/srs/issues/1147${BLACK}"
-    echo -e "${YELLOW}Experiment: SRT is enabled. https://github.com/ossrs/srs/issues/1147${BLACK}"
-else
-    echo -e "${GREEN}Warning: SRT is disabled.${BLACK}"
-fi
 echo -e "${YELLOW}Experiment: RTC is enabled. https://github.com/ossrs/srs/issues/307${BLACK}"
 if [[ $SRS_RTSP == YES ]]; then
     echo -e "${YELLOW}Experiment: RTSP is enabled (requires RTC).${BLACK}"

trunk/doc/CHANGELOG.md

@@ -7,6 +7,7 @@ The changelog for SRS.
 <a name="v7-changes"></a>
 
 ## SRS 7.0 Changelog
+* v7.0, 2025-08-31, Merge [#4460](https://github.com/ossrs/srs/pull/4460): AI: Always enable SRT protocol. v7.0.69 (#4460)
 * v7.0, 2025-08-31, Merge [#4459](https://github.com/ossrs/srs/pull/4459): AI: Merge SRT and RTC servers into unified SrsServer. v7.0.68 (#4459)
 * v7.0, 2025-08-29, Merge [#4457](https://github.com/ossrs/srs/pull/4457): Support IPv6 for all protocols: RTMP, HTTP/HTTPS, WebRTC, SRT, RTSP. v7.0.67 (#4457)
 * v7.0, 2025-08-28, Merge [#4456](https://github.com/ossrs/srs/pull/4456): AI: Remove cloud CLS and APM. v7.0.66 (#4456)

trunk/src/app/srs_app_pithy_print.cpp

@@ -213,12 +213,10 @@ 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()
 {
@@ -290,7 +288,6 @@ 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);
@@ -300,7 +297,6 @@ SrsPithyPrint *SrsPithyPrint::create_srt_publish()
 {
     return new SrsPithyPrint(SRS_CONSTS_STAGE_SRT_PUBLISH);
 }
-#endif
 
 SrsPithyPrint::~SrsPithyPrint()
 {

trunk/src/app/srs_app_pithy_print.hpp

@@ -139,10 +139,8 @@ public:
     // 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:

trunk/src/app/srs_app_rtc_conn.cpp

@@ -1189,7 +1189,6 @@ srs_error_t SrsRtcPublishStream::initialize(ISrsRequest *r, SrsRtcSourceDescript
     }
 
     // Check whether SRT stream is busy.
-#ifdef SRS_SRT
     bool srt_server_enabled = _srs_config->get_srt_enabled();
     bool srt_enabled = _srs_config->get_srt_enabled(r->vhost);
     if (srt_server_enabled && srt_enabled) {
@@ -1202,7 +1201,6 @@ srs_error_t SrsRtcPublishStream::initialize(ISrsRequest *r, SrsRtcSourceDescript
             return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "srt stream %s busy", r->get_stream_url().c_str());
         }
     }
-#endif
 
     // Bridge to rtmp
 #if defined(SRS_FFMPEG_FIT)

trunk/src/app/srs_app_rtmp_conn.cpp

@@ -1032,7 +1032,6 @@ srs_error_t SrsRtmpConn::acquire_publish(SrsSharedPtr<SrsLiveSource> source)
     }
 
     // Check whether SRT stream is busy.
-#ifdef SRS_SRT
     bool srt_server_enabled = _srs_config->get_srt_enabled();
     bool srt_enabled = _srs_config->get_srt_enabled(req->vhost);
     if (srt_server_enabled && srt_enabled && !info->edge) {
@@ -1045,7 +1044,6 @@ srs_error_t SrsRtmpConn::acquire_publish(SrsSharedPtr<SrsLiveSource> source)
             return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "srt stream %s busy", req->get_stream_url().c_str());
         }
     }
-#endif
 
 #ifdef SRS_RTSP
     // RTSP only support viewer, so we don't need to check it.

trunk/src/app/srs_app_server.cpp

@@ -53,11 +53,9 @@ using namespace std;
 #ifdef SRS_GB28181
 #include <srs_app_gb28181.hpp>
 #endif
-#ifdef SRS_SRT
 #include <srs_app_srt_conn.hpp>
 #include <srs_app_srt_server.hpp>
 #include <srs_app_srt_source.hpp>
-#endif
 #ifdef SRS_RTSP
 #include <srs_app_rtsp_conn.hpp>
 #include <srs_app_rtsp_source.hpp>
@@ -293,9 +291,7 @@ srs_error_t srs_global_initialize()
     _srs_circuit_breaker = new SrsCircuitBreaker();
     _srs_hooks = new SrsHttpHooks();
 
-#ifdef SRS_SRT
     _srs_srt_sources = new SrsSrtSourceManager();
-#endif
 
     _srs_rtc_sources = new SrsRtcSourceManager();
     _srs_blackhole = new SrsRtcBlackhole();
@@ -810,9 +806,7 @@ void SrsServer::destroy()
 #ifdef SRS_GB28181
     srs_freep(stream_caster_gb28181_);
 #endif
-#ifdef SRS_SRT
     close_srt_listeners();
-#endif
 
     // Cleanup WebRTC components
     if (true) {
@@ -851,9 +845,7 @@ void SrsServer::dispose()
 #ifdef SRS_GB28181
     stream_caster_gb28181_->close();
 #endif
-#ifdef SRS_SRT
     close_srt_listeners();
-#endif
 
     // Fast stop to notify FFMPEG to quit, wait for a while then fast kill.
     ingester_->dispose();
@@ -889,9 +881,7 @@ void SrsServer::gracefully_dispose()
 #ifdef SRS_GB28181
     stream_caster_gb28181_->close();
 #endif
-#ifdef SRS_SRT
     close_srt_listeners();
-#endif
     srs_trace("listeners closed");
 
     // Fast stop to notify FFMPEG to quit, wait for a while then fast kill.
@@ -929,7 +919,6 @@ srs_error_t SrsServer::initialize()
         return srs_error_wrap(err, "init server");
     }
 
-#ifdef SRS_SRT
     if ((err = srs_srt_log_initialize()) != srs_success) {
         return srs_error_wrap(err, "srt log initialize");
     }
@@ -943,7 +932,6 @@ srs_error_t SrsServer::initialize()
     if ((err = _srt_eventloop->start()) != srs_success) {
         return srs_error_wrap(err, "srt poller start");
     }
-#endif
 
     // Initialize WebRTC DTLS certificate
     if ((err = _srs_rtc_dtls_certificate->initialize()) != srs_success) {
@@ -1132,11 +1120,9 @@ srs_error_t SrsServer::run()
         return srs_error_wrap(err, "live sources");
     }
 
-#ifdef SRS_SRT
     if ((err = _srs_srt_sources->initialize()) != srs_success) {
         return srs_error_wrap(err, "srt sources");
     }
-#endif
 
     if ((err = _srs_rtc_sources->initialize()) != srs_success) {
         return srs_error_wrap(err, "rtc sources");
@@ -1319,12 +1305,10 @@ srs_error_t SrsServer::listen()
         }
     }
 
-#ifdef SRS_SRT
     // Listen MPEG-TS over SRT.
     if ((err = listen_srt_mpegts()) != srs_success) {
         return srs_error_wrap(err, "srt mpegts listen");
     }
-#endif
 
     // Listen WebRTC UDP.
     if ((err = listen_rtc_udp()) != srs_success) {
@@ -1794,13 +1778,11 @@ void SrsServer::resample_kbps()
             continue;
         }
 
-#ifdef SRS_SRT
         SrsMpegtsSrtConn *srt = dynamic_cast<SrsMpegtsSrtConn *>(c);
         if (srt) {
             stat->kbps_add_delta(c->get_id().c_str(), srt->delta());
             continue;
         }
-#endif
 
         SrsRtcConnection *rtc = dynamic_cast<SrsRtcConnection *>(c);
         if (rtc) {
@@ -1816,7 +1798,6 @@ void SrsServer::resample_kbps()
     stat->kbps_sample();
 }
 
-#ifdef SRS_SRT
 srs_error_t SrsServer::listen_srt_mpegts()
 {
     srs_error_t err = srs_success;
@@ -1913,7 +1894,6 @@ srs_error_t SrsServer::srt_fd_to_resource(srs_srt_t srt_fd, ISrsResource **pr)
 
     return err;
 }
-#endif
 
 srs_error_t SrsServer::listen_rtc_udp()
 {

trunk/src/app/srs_app_server.hpp

@@ -21,10 +21,8 @@
 #include <srs_app_st.hpp>
 #include <srs_protocol_st.hpp>
 
-#ifdef SRS_SRT
 #include <srs_app_srt_listener.hpp>
 #include <srs_protocol_srt.hpp>
-#endif
 
 class SrsAsyncCallWorker;
 class SrsUdpMuxListener;
@@ -194,10 +192,8 @@ private:
     // Stream Caster for GB28181.
     SrsGbListener *stream_caster_gb28181_;
 #endif
-#ifdef SRS_SRT
     // SRT acceptors for MPEG-TS over SRT.
     std::vector<SrsSrtAcceptor *> srt_acceptors_;
-#endif
     // WebRTC UDP listeners for RTC server functionality.
     std::vector<SrsUdpMuxListener *> rtc_listeners_;
     // WebRTC async call worker for non-blocking operations.
@@ -293,13 +289,11 @@ private:
     // Resample the server kbs.
     virtual void resample_kbps();
 
-#ifdef SRS_SRT
     // SRT-related methods
     virtual srs_error_t listen_srt_mpegts();
     virtual void close_srt_listeners();
     virtual srs_error_t accept_srt_client(srs_srt_t srt_fd);
     virtual srs_error_t srt_fd_to_resource(srs_srt_t srt_fd, ISrsResource **pr);
-#endif
 
     // WebRTC-related methods
     virtual srs_error_t listen_rtc_udp();

trunk/src/app/srs_app_srt_server.cpp

@@ -15,9 +15,7 @@ using namespace std;
 #include <srs_kernel_utility.hpp>
 #include <srs_protocol_log.hpp>
 
-#ifdef SRS_SRT
 SrsSrtEventLoop *_srt_eventloop = NULL;
-#endif
 
 SrsSrtAcceptor::SrsSrtAcceptor(ISrsSrtClientHandler *srt_handler)
 {

trunk/src/core/srs_core_version7.hpp

@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR 7
 #define VERSION_MINOR 0
-#define VERSION_REVISION 68
+#define VERSION_REVISION 69
 
 #endif

trunk/src/kernel/srs_kernel_kbps.cpp

@@ -111,3 +111,21 @@ srs_utime_t SrsWallClock::now()
 }
 
 SrsWallClock *_srs_clock = NULL;
+
+#if defined(SRS_DEBUG) && defined(SRS_DEBUG_STATS)
+SrsPps *_srs_pps_epoll = NULL;
+SrsPps *_srs_pps_epoll_zero = NULL;
+SrsPps *_srs_pps_epoll_shake = NULL;
+SrsPps *_srs_pps_epoll_spin = NULL;
+
+SrsPps *_srs_pps_sched_160ms = NULL;
+SrsPps *_srs_pps_sched_s = NULL;
+SrsPps *_srs_pps_sched_15ms = NULL;
+SrsPps *_srs_pps_sched_20ms = NULL;
+SrsPps *_srs_pps_sched_25ms = NULL;
+SrsPps *_srs_pps_sched_30ms = NULL;
+SrsPps *_srs_pps_sched_35ms = NULL;
+SrsPps *_srs_pps_sched_40ms = NULL;
+SrsPps *_srs_pps_sched_80ms = NULL;
+#endif
+

trunk/src/main/srs_main_server.cpp

@@ -34,20 +34,16 @@ using namespace std;
 #include <srs_app_circuit_breaker.hpp>
 #include <srs_app_config.hpp>
 #include <srs_app_log.hpp>
+#include <srs_app_rtc_conn.hpp>
 #include <srs_app_server.hpp>
+#include <srs_app_srt_server.hpp>
 #include <srs_app_utility.hpp>
 #include <srs_core_autofree.hpp>
 #include <srs_core_performance.hpp>
 #include <srs_kernel_error.hpp>
 #include <srs_kernel_file.hpp>
 #include <srs_kernel_utility.hpp>
-
-#include <srs_app_rtc_conn.hpp>
-
-#ifdef SRS_SRT
-#include <srs_app_srt_server.hpp>
 #include <srs_protocol_srt.hpp>
-#endif
 
 // pre-declare
 srs_error_t run_directly_or_daemon();
@@ -269,7 +265,7 @@ void show_macro_features()
         ss << ", dash:" << "on";
         ss << ", hls:" << srs_strconv_format_bool(true);
         ss << ", hds:" << srs_strconv_format_bool(SRS_HDS_BOOL);
-        ss << ", srt:" << srs_strconv_format_bool(SRS_SRT_BOOL);
+        ss << ", srt:" << srs_strconv_format_bool(true);
         // hc(http callback)
         ss << ", hc:" << srs_strconv_format_bool(true);
         // ha(http api)

trunk/src/utest/srs_utest.cpp

@@ -19,10 +19,8 @@ using namespace std;
 #include <sys/mman.h>
 #include <sys/types.h>
 
-#ifdef SRS_SRT
 #include <srs_app_srt_server.hpp>
 #include <srt/srt.h>
-#endif
 
 // Temporary disk config.
 std::string _srs_tmp_file_prefix = "/tmp/srs-utest-";
@@ -44,13 +42,11 @@ const char *_srs_binary = NULL;
 
 #include <srs_app_st.hpp>
 
-#ifdef SRS_SRT
 static void srs_srt_utest_null_log_handler(void *opaque, int level, const char *file, int line,
                                            const char *area, const char *message)
 {
     // srt null log handler, do no print any log.
 }
-#endif
 
 // Initialize global settings.
 srs_error_t prepare_main()
@@ -73,7 +69,6 @@ srs_error_t prepare_main()
     srs_freep(_srs_context);
     _srs_context = new SrsThreadContext();
 
-#ifdef SRS_SRT
     if ((err = srs_srt_log_initialize()) != srs_success) {
         return srs_error_wrap(err, "srt log initialize");
     }
@@ -88,7 +83,6 @@ srs_error_t prepare_main()
     if ((err = _srt_eventloop->start()) != srs_success) {
         return srs_error_wrap(err, "srt poller start");
     }
-#endif
 
     return err;
 }