For #1659, #307, add switch to disable rtc

.circleci/config.yml

@@ -5,7 +5,7 @@ jobs:
       - image: ossrs/srs:dev
     steps:
       - checkout
-      - run: cd trunk && ./configure && make
+      - run: cd trunk && ./configure --without-rtc && make && ./configure && make
   test:
     docker:
       - image: ossrs/srs:dev

trunk/auto/auto_headers.sh

@@ -73,6 +73,12 @@ else
     srs_undefine_macro "SRS_AUTO_SRT" $SRS_AUTO_HEADERS_H
 fi
 
+if [ $SRS_RTC = YES ]; then
+    srs_define_macro "SRS_AUTO_RTC" $SRS_AUTO_HEADERS_H
+else
+    srs_undefine_macro "SRS_AUTO_RTC" $SRS_AUTO_HEADERS_H
+fi
+
 if [ $SRS_MEM_WATCH = YES ]; then
     srs_define_macro "SRS_AUTO_MEM_WATCH" $SRS_AUTO_HEADERS_H
 else

trunk/auto/depends.sh

@@ -399,7 +399,7 @@ fi
 #####################################################################################
 # libopus, for WebRTC to transcode AAC with Opus.
 #####################################################################################
-if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
+if [[ $SRS_EXPORT_LIBRTMP_PROJECT == NO && $SRS_RTC == YES ]]; then
     if [[ -f ${SRS_OBJS}/opus/lib/libopus.a ]]; then
         echo "The opus-1.3.1 is ok.";
     else
@@ -419,7 +419,7 @@ fi
 #####################################################################################
 # ffmpeg-fix, for WebRTC to transcode AAC with Opus.
 #####################################################################################
-if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
+if [[ $SRS_EXPORT_LIBRTMP_PROJECT == NO && $SRS_RTC == YES ]]; then
     if [[ -f ${SRS_OBJS}/ffmpeg/lib/libavcodec.a ]]; then
         echo "The ffmpeg-4.2-fit is ok.";
     else

trunk/auto/options.sh

@@ -17,6 +17,7 @@ help=no
 # feature options
 SRS_HDS=NO
 SRS_SRT=NO
+SRS_RTC=YES
 SRS_NGINX=NO
 SRS_FFMPEG_TOOL=NO
 SRS_LIBRTMP=NO
@@ -130,7 +131,8 @@ Features:
   --with-librtmp            Enable srs-librtmp, library for client.
   --with-research           Build the research tools.
   --with-utest              Build the utest for SRS.
-  --with-srt                Build the srt for SRS.
+  --with-srt                Build the SRT support for SRS.
+  --with-rtc                Build the WebRTC support for SRS.
 
   --without-ssl             Disable rtmp complex handshake.
   --without-hds             Disable hds, the adobe http dynamic streaming.
@@ -139,7 +141,8 @@ Features:
   --without-librtmp         Disable srs-librtmp, library for client.
   --without-research        Do not build the research tools.
   --without-utest           Do not build the utest for SRS.
-  --without-srt             Do not build the srt for SRS.
+  --without-srt             Do not build the SRT support for SRS.
+  --without-rtc             Do not build the WebRTC support for SRS.
 
   --prefix=<path>           The absolute installation path for srs. Default: $SRS_PREFIX
   --static                  Whether add '-static' to link options.
@@ -225,6 +228,7 @@ function parse_user_option() {
         --with-research)                SRS_RESEARCH=YES            ;;
         --with-utest)                   SRS_UTEST=YES               ;;
         --with-srt)                     SRS_SRT=YES                 ;;
+        --with-rtc)                     SRS_RTC=YES                 ;;
         --with-gperf)                   SRS_GPERF=YES               ;;
         --with-gmc)                     SRS_GPERF_MC=YES            ;;
         --with-gmd)                     SRS_GPERF_MD=YES            ;;
@@ -240,7 +244,8 @@ function parse_user_option() {
         --without-librtmp)              SRS_LIBRTMP=NO              ;;
         --without-research)             SRS_RESEARCH=NO             ;;
         --without-utest)                SRS_UTEST=NO                ;;
-        --without-srt)                  SRS_SRT=NO                 ;;
+        --without-srt)                  SRS_SRT=NO                  ;;
+        --without-rtc)                  SRS_RTC=NO                  ;;
         --without-gperf)                SRS_GPERF=NO                ;;
         --without-gmc)                  SRS_GPERF_MC=NO             ;;
         --without-gmd)                  SRS_GPERF_MD=NO             ;;
@@ -539,6 +544,7 @@ function regenerate_options() {
     if [ $SRS_RESEARCH = YES ]; then SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --with-research"; else SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --without-research"; fi
     if [ $SRS_UTEST = YES ]; then SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --with-utest"; else SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --without-utest"; fi
     if [ $SRS_SRT = YES ]; then SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --with-srt"; else SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --without-srt"; fi
+    if [ $SRS_RTC = YES ]; then SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --with-rtc"; else SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --without-rtc"; fi
     if [ $SRS_GPERF = YES ]; then SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --with-gperf"; else SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --without-gperf"; fi
     if [ $SRS_GPERF_MC = YES ]; then SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --with-gmc"; else SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --without-gmc"; fi
     if [ $SRS_GPERF_MD = YES ]; then SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --with-gmd"; else SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --without-gmd"; fi

trunk/configure

@@ -149,10 +149,14 @@ END
 LibSTRoot="${SRS_OBJS_DIR}/st"; LibSTfile="${LibSTRoot}/libst.a"
 if [[ $SRS_SHARED_ST == YES ]]; then LibSTfile="-lst"; fi
 # srtp
-LibSrtpRoot="${SRS_OBJS_DIR}/srtp2/include"; LibSrtpFile="${SRS_OBJS_DIR}/srtp2/lib/libsrtp2.a"
+if [[ $SRS_RTC == YES ]]; then
-# ffmpeg
+  LibSrtpRoot="${SRS_OBJS_DIR}/srtp2/include"; LibSrtpFile="${SRS_OBJS_DIR}/srtp2/lib/libsrtp2.a"
-LibFfmpegRoot="${SRS_OBJS_DIR}/ffmpeg/include"; LibFfmpegFile="${SRS_OBJS_DIR}/ffmpeg/lib/libavcodec.a ${SRS_OBJS_DIR}/ffmpeg/lib/libswresample.a ${SRS_OBJS_DIR}/ffmpeg/lib/libavutil.a -lpthread"
+fi
-LibFfmpegRoot="${LibFfmpegRoot} ${SRS_OBJS_DIR}/opus/include"; LibFfmpegFile="${LibFfmpegFile} ${SRS_OBJS_DIR}/opus/lib/libopus.a"
+# FFMPEG for WebRTC transcoding, such as aac to opus.
+if [[ $SRS_RTC == YES ]]; then
+  LibFfmpegRoot="${SRS_OBJS_DIR}/ffmpeg/include"; LibFfmpegFile="${SRS_OBJS_DIR}/ffmpeg/lib/libavcodec.a ${SRS_OBJS_DIR}/ffmpeg/lib/libswresample.a ${SRS_OBJS_DIR}/ffmpeg/lib/libavutil.a"
+  LibFfmpegRoot="${LibFfmpegRoot} ${SRS_OBJS_DIR}/opus/include"; LibFfmpegFile="${LibFfmpegFile} ${SRS_OBJS_DIR}/opus/lib/libopus.a"
+fi
 # openssl-1.1.0e, for the RTMP complex handshake.
 LibSSLRoot="";LibSSLfile=""
 if [[ $SRS_SSL == YES && $SRS_USE_SYS_SSL == NO ]]; then
@@ -173,7 +177,7 @@ if [[ $SRS_SRT == YES ]]; then
 fi
 # the link options, always use static link
 SrsLinkOptions="-ldl";
-if [[ $SRS_SRT == YES ]]; then
+if [[ $SRS_SRT == YES || $SRS_RTC == YES ]]; then
     SrsLinkOptions="${SrsLinkOptions} -lpthread";
 fi
 if [[ $SRS_SSL == YES && $SRS_USE_SYS_SSL == YES ]]; then
@@ -206,9 +210,12 @@ MODULE_ID="KERNEL"
 MODULE_DEPENDS=("CORE") 
 ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSSLRoot})
 MODULE_FILES=("srs_kernel_error" "srs_kernel_log" "srs_kernel_buffer"
-        "srs_kernel_utility" "srs_kernel_flv" "srs_kernel_rtp" "srs_kernel_codec" "srs_kernel_io"
+        "srs_kernel_utility" "srs_kernel_flv" "srs_kernel_codec" "srs_kernel_io"
         "srs_kernel_consts" "srs_kernel_aac" "srs_kernel_mp3" "srs_kernel_ts"
         "srs_kernel_stream" "srs_kernel_balance" "srs_kernel_mp4" "srs_kernel_file")
+if [[ $SRS_RTC == YES ]]; then
+    MODULE_FILES+=("srs_kernel_rtp")
+fi
 KERNEL_INCS="src/kernel"; MODULE_DIR=${KERNEL_INCS} . auto/modules.sh
 KERNEL_OBJS="${MODULE_OBJS[@]}"
 #
@@ -219,7 +226,10 @@ ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSSLRoot})
 MODULE_FILES=("srs_protocol_amf0" "srs_protocol_io" "srs_rtmp_stack"
         "srs_rtmp_handshake" "srs_protocol_utility" "srs_rtmp_msg_array" "srs_protocol_stream"
         "srs_raw_avc" "srs_rtsp_stack" "srs_http_stack" "srs_protocol_kbps" "srs_protocol_json"
-        "srs_stun_stack" "srs_protocol_format")
+        "srs_protocol_format")
+if [[ $SRS_RTC == YES ]]; then
+    MODULE_FILES+=("srs_stun_stack")
+fi
 PROTOCOL_INCS="src/protocol"; MODULE_DIR=${PROTOCOL_INCS} . auto/modules.sh
 PROTOCOL_OBJS="${MODULE_OBJS[@]}"
 #
@@ -238,7 +248,10 @@ fi
 if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
     MODULE_ID="SERVICE"
     MODULE_DEPENDS=("CORE" "KERNEL" "PROTOCOL")
-    ModuleLibIncs=(${LibSTRoot} ${LibSrtpRoot} ${LibFfmpegRoot} ${SRS_OBJS_DIR} ${LibSSLRoot})
+    ModuleLibIncs=(${LibSTRoot} ${SRS_OBJS_DIR} ${LibSSLRoot})
+    if [[ $SRS_RTC == YES ]]; then
+        ModuleLibIncs+=("${LibFfmpegRoot[*]}" ${LibSrtpRoot})
+    fi
     MODULE_FILES=("srs_service_log" "srs_service_st" "srs_service_http_client"
         "srs_service_http_conn" "srs_service_rtmp_conn" "srs_service_utility"
         "srs_service_conn")
@@ -251,7 +264,10 @@ fi
 if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
     MODULE_ID="APP" 
     MODULE_DEPENDS=("CORE" "KERNEL" "PROTOCOL" "SERVICE")
-    ModuleLibIncs=(${LibSTRoot} ${LibSrtpRoot} ${LibFfmpegRoot} ${SRS_OBJS_DIR} ${LibSSLRoot})
+    ModuleLibIncs=(${LibSTRoot} ${SRS_OBJS_DIR} ${LibSSLRoot})
+    if [[ $SRS_RTC == YES ]]; then
+        ModuleLibIncs+=("${LibFfmpegRoot[*]}" ${LibSrtpRoot})
+    fi
     MODULE_FILES=("srs_app_server" "srs_app_conn" "srs_app_rtmp_conn" "srs_app_source" 
             "srs_app_refer" "srs_app_hls" "srs_app_forward" "srs_app_encoder" "srs_app_http_stream"
             "srs_app_thread" "srs_app_bandwidth" "srs_app_st" "srs_app_log" "srs_app_config" 
@@ -259,10 +275,13 @@ if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
             "srs_app_ingest" "srs_app_ffmpeg" "srs_app_utility" "srs_app_edge"
             "srs_app_heartbeat" "srs_app_empty" "srs_app_http_client" "srs_app_http_static"
             "srs_app_recv_thread" "srs_app_security" "srs_app_statistic" "srs_app_hds"
-            "srs_app_mpegts_udp" "srs_app_rtc" "srs_app_rtc_conn" "srs_app_dtls" "srs_app_rtsp" "srs_app_listener" "srs_app_async_call"
+            "srs_app_mpegts_udp" "srs_app_rtsp" "srs_app_listener" "srs_app_async_call"
             "srs_app_caster_flv" "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_audio_recode")
+            "srs_app_coworkers" "srs_app_hybrid")
+    if [[ $SRS_RTC == YES ]]; then
+        MODULE_FILES+=("srs_app_rtc" "srs_app_rtc_conn" "srs_app_dtls" "srs_app_audio_recode")
+    fi
     DEFINES=""
     # add each modules for app
     for SRS_MODULE in ${SRS_MODULES[*]}; do
@@ -289,7 +308,10 @@ if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
     if [[ $SRS_SRT == YES ]]; then
         MODULE_DEPENDS+=("SRT")
     fi
-    ModuleLibIncs=(${LibSTRoot} ${LibSrtpRoot} ${LibFfmpegRoot} ${SRS_OBJS_DIR} ${LibGperfRoot} ${LibSSLRoot})
+    ModuleLibIncs=(${LibSTRoot} ${SRS_OBJS_DIR} ${LibGperfRoot} ${LibSSLRoot})
+    if [[ $SRS_RTC == YES ]]; then
+        ModuleLibIncs+=("${LibFfmpegRoot[*]}" ${LibSrtpRoot})
+    fi
     if [[ $SRS_SRT == YES ]]; then
         ModuleLibIncs+=("${LibSRTRoot[*]}")
     fi
@@ -302,7 +324,10 @@ fi
 if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
     MODULE_ID="MAIN"
     MODULE_DEPENDS=("CORE" "KERNEL" "PROTOCOL" "SERVICE")
-    ModuleLibIncs=(${LibSTRoot} ${LibSrtpRoot} ${LibFfmpegRoot} ${SRS_OBJS_DIR} ${LibGperfRoot} ${LibSSLRoot})
+    ModuleLibIncs=(${LibSTRoot} ${SRS_OBJS_DIR} ${LibGperfRoot} ${LibSSLRoot})
+    if [[ $SRS_RTC == YES ]]; then
+        ModuleLibIncs+=("${LibFfmpegRoot[*]}" ${LibSrtpRoot})
+    fi
     MODULE_FILES=()
     DEFINES=""
     # add each modules for main
@@ -329,13 +354,19 @@ if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
     done
     #
     # all depends libraries
-    ModuleLibFiles=(${LibSTfile} ${LibSrtpFile} ${LibFfmpegFile} ${LibSSLfile} ${LibGperfFile})
+    ModuleLibFiles=(${LibSTfile} ${LibSSLfile} ${LibGperfFile})
+    if [[ $SRS_RTC == YES ]]; then
+        ModuleLibFiles+=("${LibFfmpegFile[*]}" ${LibSrtpFile})
+    fi
     if [[ $SRS_SRT == YES ]]; then
         ModuleLibFiles+=("${LibSRTfile[*]}")
     fi
     # all depends objects
     MODULE_OBJS="${CORE_OBJS[@]} ${KERNEL_OBJS[@]} ${PROTOCOL_OBJS[@]} ${SERVICE_OBJS[@]} ${APP_OBJS[@]} ${SERVER_OBJS[@]}"
-    ModuleLibIncs=(${LibSTRoot} ${LibSrtpRoot} ${LibFfmpegRoot} ${SRS_OBJS_DIR} ${LibGperfRoot} ${LibSSLRoot})
+    ModuleLibIncs=(${LibSTRoot} ${SRS_OBJS_DIR} ${LibGperfRoot} ${LibSSLRoot})
+    if [[ $SRS_RTC == YES ]]; then
+        ModuleLibIncs+=("${LibFfmpegRoot[*]}" ${LibSrtpRoot})
+    fi
     if [[ $SRS_SRT == YES ]]; then
         MODULE_OBJS="${MODULE_OBJS} ${SRT_OBJS[@]}"
     fi
@@ -346,7 +377,10 @@ if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
     #
     # For modules, without the app module.
     MODULE_OBJS="${CORE_OBJS[@]} ${KERNEL_OBJS[@]} ${PROTOCOL_OBJS[@]} ${SERVICE_OBJS[@]} ${MAIN_OBJS[@]}"
-    ModuleLibFiles=(${LibSTfile} ${LibSrtpFile} ${LibFfmpegFile} ${LibSSLfile} ${LibGperfFile})
+    ModuleLibFiles=(${LibSTfile} ${LibSSLfile} ${LibGperfFile})
+    if [[ $SRS_RTC == YES ]]; then
+        ModuleLibFiles+=("${LibFfmpegFile[*]}" ${LibSrtpFile})
+    fi
     #
     for SRS_MODULE in ${SRS_MODULES[*]}; do
         . $SRS_MODULE/config
@@ -366,11 +400,17 @@ if [ $SRS_UTEST = YES ]; then
     MODULE_FILES=("srs_utest" "srs_utest_amf0" "srs_utest_protocol" "srs_utest_kernel" "srs_utest_core"
         "srs_utest_config" "srs_utest_rtmp" "srs_utest_http" "srs_utest_avc" "srs_utest_reload"
         "srs_utest_mp4" "srs_utest_service" "srs_utest_app")
-    ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSTRoot} ${LibSrtpRoot} ${LibFfmpegRoot} ${LibSSLRoot})
+    ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSTRoot} ${LibSSLRoot})
+    if [[ $SRS_RTC == YES ]]; then
+        ModuleLibIncs+=("${LibFfmpegRoot[*]}" ${LibSrtpRoot})
+    fi
     if [[ $SRS_SRT == YES ]]; then
         ModuleLibIncs+=("${LibSRTRoot[*]}")
     fi
-    ModuleLibFiles=(${LibSTfile} ${LibSrtpFile} ${LibFfmpegFile} ${LibSSLfile})
+    ModuleLibFiles=(${LibSTfile} ${LibSSLfile})
+    if [[ $SRS_RTC == YES ]]; then
+        ModuleLibFiles+=("${LibFfmpegFile[*]}" ${LibSrtpFile})
+    fi
     if [[ $SRS_SRT == YES ]]; then
         ModuleLibFiles+=("${LibSRTfile[*]}")
     fi

trunk/src/app/srs_app_http_api.cpp

@@ -46,7 +46,9 @@ using namespace std;
 #include <srs_protocol_amf0.hpp>
 #include <srs_protocol_utility.hpp>
 #include <srs_app_coworkers.hpp>
+#ifdef SRS_AUTO_RTC
 #include <srs_app_rtc_conn.hpp>
+#endif
 
 srs_error_t srs_api_response_jsonp(ISrsHttpResponseWriter* w, string callback, string data)
 {
@@ -781,6 +783,7 @@ srs_error_t SrsGoApiStreams::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessa
     return srs_api_response(w, r, obj->dumps());
 }
 
+#ifdef SRS_AUTO_RTC
 SrsGoApiSdp::SrsGoApiSdp(SrsRtcServer* rtc_svr)
 {
     rtc_server = rtc_svr;
@@ -907,6 +910,7 @@ srs_error_t SrsGoApiSdp::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessag
 
     return err;
 }
+#endif
 
 SrsGoApiClients::SrsGoApiClients()
 {

trunk/src/app/srs_app_http_api.hpp

@@ -166,6 +166,7 @@ public:
     virtual srs_error_t serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r);
 };
 
+#ifdef SRS_AUTO_RTC
 class SrsGoApiSdp : public ISrsHttpHandler
 {
 private:
@@ -178,6 +179,7 @@ public:
 private:
     virtual srs_error_t do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsJsonObject* res);
 };
+#endif
 
 class SrsGoApiClients : public ISrsHttpHandler
 {

trunk/src/app/srs_app_source.cpp

@@ -33,7 +33,6 @@ using namespace std;
 #include <srs_kernel_codec.hpp>
 #include <srs_kernel_rtp.hpp>
 #include <srs_app_hls.hpp>
-#include <srs_app_rtc.hpp>
 #include <srs_app_forward.hpp>
 #include <srs_app_config.hpp>
 #include <srs_app_encoder.hpp>
@@ -51,6 +50,9 @@ using namespace std;
 #include <srs_app_ng_exec.hpp>
 #include <srs_app_dash.hpp>
 #include <srs_protocol_format.hpp>
+#ifdef SRS_AUTO_RTC
+#include <srs_app_rtc.hpp>
+#endif
 
 #define CONST_MAX_JITTER_MS         250
 #define CONST_MAX_JITTER_MS_NEG         -250
@@ -816,6 +818,7 @@ SrsSharedPtrMessage* SrsMixQueue::pop()
     return msg;
 }
 
+#ifdef SRS_AUTO_RTC
 SrsRtpPacketQueue::SrsRtpPacketQueue()
 {
 }
@@ -864,6 +867,7 @@ SrsRtpSharedPacket* SrsRtpPacketQueue::find(const uint16_t& sequence)
 
     return pkt;
 }
+#endif
 
 SrsOriginHub::SrsOriginHub()
 {
@@ -875,7 +879,9 @@ SrsOriginHub::SrsOriginHub()
     dash = new SrsDash();
     dvr = new SrsDvr();
     encoder = new SrsEncoder();
+#ifdef SRS_AUTO_RTC
     rtc = new SrsRtc();
+#endif
 #ifdef SRS_AUTO_HDS
     hds = new SrsHds();
 #endif
@@ -919,10 +925,12 @@ srs_error_t SrsOriginHub::initialize(SrsSource* s, SrsRequest* r)
     if ((err = format->initialize()) != srs_success) {
         return srs_error_wrap(err, "format initialize");
     }
-    
+
+#ifdef SRS_AUTO_RTC
     if ((err = rtc->initialize(this, req)) != srs_success) {
         return srs_error_wrap(err, "rtc initialize");
     }
+#endif
     
     if ((err = hls->initialize(this, req)) != srs_success) {
         return srs_error_wrap(err, "hls initialize");
@@ -1022,11 +1030,13 @@ srs_error_t SrsOriginHub::on_audio(SrsSharedPtrMessage* shared_audio)
                   srs_flv_srates[c->sound_rate]);
     }
 
+#ifdef SRS_AUTO_RTC
     if ((err = rtc->on_audio(msg, format)) != srs_success) {
         srs_warn("rtc: ignore audio error %s", srs_error_desc(err).c_str());
         srs_error_reset(err);
         rtc->on_unpublish();
     }
+#endif
     
     if ((err = hls->on_audio(msg, format)) != srs_success) {
         // apply the error strategy for hls.
@@ -1121,6 +1131,7 @@ srs_error_t SrsOriginHub::on_video(SrsSharedPtrMessage* shared_video, bool is_se
         return err;
     }
 
+#ifdef SRS_AUTO_RTC
     // Parse RTMP message to RTP packets, in FU-A if too large.
     if ((err = rtc->on_video(msg, format)) != srs_success) {
         // TODO: We should support more strategies.
@@ -1132,6 +1143,7 @@ srs_error_t SrsOriginHub::on_video(SrsSharedPtrMessage* shared_video, bool is_se
     // TODO: FIXME: Refactor to move to rtp?
     // Save the RTP packets for find_rtp_packet() to rtx or restore it.
     source->rtp_queue->push(msg->rtp_packets);
+#endif
     
     if ((err = hls->on_video(msg, format)) != srs_success) {
         // TODO: We should support more strategies.
@@ -1200,10 +1212,12 @@ srs_error_t SrsOriginHub::on_publish()
     if ((err = encoder->on_publish(req)) != srs_success) {
         return srs_error_wrap(err, "encoder publish");
     }
-    
+
+#ifdef SRS_AUTO_RTC
     if ((err = rtc->on_publish()) != srs_success) {
         return srs_error_wrap(err, "rtc publish");
     }
+#endif
 
     if ((err = hls->on_publish()) != srs_success) {
         return srs_error_wrap(err, "hls publish");
@@ -1242,7 +1256,9 @@ void SrsOriginHub::on_unpublish()
     destroy_forwarders();
     
     encoder->on_unpublish();
+#ifdef SRS_AUTO_RTC
     rtc->on_unpublish();
+#endif
     hls->on_unpublish();
     dash->on_unpublish();
     dvr->on_unpublish();
@@ -1904,7 +1920,9 @@ SrsSource::SrsSource()
     jitter_algorithm = SrsRtmpJitterAlgorithmOFF;
     mix_correct = false;
     mix_queue = new SrsMixQueue();
+#ifdef SRS_AUTO_RTC
     rtp_queue = new SrsRtpPacketQueue();
+#endif
     
     _can_publish = true;
     _pre_source_id = _source_id = -1;
@@ -1934,7 +1952,9 @@ SrsSource::~SrsSource()
     srs_freep(hub);
     srs_freep(meta);
     srs_freep(mix_queue);
+#ifdef SRS_AUTO_RTC
     srs_freep(rtp_queue);
+#endif
     
     srs_freep(play_edge);
     srs_freep(publish_edge);
@@ -2692,7 +2712,9 @@ string SrsSource::get_curr_origin()
     return play_edge->get_curr_origin();
 }
 
+#ifdef SRS_AUTO_RTC
 SrsRtpSharedPacket* SrsSource::find_rtp_packet(const uint16_t& seq)
 {
     return rtp_queue->find(seq);
 }
+#endif

trunk/src/app/srs_app_source.hpp

@@ -325,6 +325,7 @@ public:
     virtual SrsSharedPtrMessage* pop();
 };
 
+#ifdef SRS_AUTO_RTC
 // To find the RTP packet for RTX or restore.
 class SrsRtpPacketQueue
 {
@@ -347,6 +348,7 @@ public:
     void insert(const uint16_t& sequence, SrsRtpSharedPacket* pkt);
     SrsRtpSharedPacket* find(const uint16_t& sequence);
 };
+#endif
 
 // The hub for origin is a collection of utilities for origin only,
 // For example, DVR, HLS, Forward and Transcode are only available for origin,
@@ -360,8 +362,10 @@ private:
 private:
     // The format, codec information.
     SrsRtmpFormat* format;
+#ifdef SRS_AUTO_RTC
     // rtc handler
     SrsRtc* rtc;
+#endif
     // hls handler.
     SrsHls* hls;
     // The DASH encoder.
@@ -534,8 +538,10 @@ private:
     bool mix_correct;
     // The mix queue to implements the mix correct algorithm.
     SrsMixQueue* mix_queue;
+#ifdef SRS_AUTO_RTC
     // rtp packet queue
     SrsRtpPacketQueue* rtp_queue;
+#endif
     // For play, whether enabled atc.
     // The atc(use absolute time and donot adjust time),
     // directly use msg time and donot adjust if atc is true,
@@ -625,8 +631,10 @@ public:
 public:
     virtual std::string get_curr_origin();
 public:
+#ifdef SRS_AUTO_RTC
     // Find rtp packet by sequence
     SrsRtpSharedPacket* find_rtp_packet(const uint16_t& seq);
+#endif
 };
 
 #endif

trunk/src/kernel/srs_kernel_flv.cpp

@@ -36,11 +36,13 @@ using namespace std;
 #include <srs_kernel_error.hpp>
 #include <srs_kernel_buffer.hpp>
 #include <srs_kernel_file.hpp>
-#include <srs_kernel_rtp.hpp>
 #include <srs_kernel_codec.hpp>
 #include <srs_kernel_utility.hpp>
 #include <srs_core_mem_watch.hpp>
 #include <srs_core_autofree.hpp>
+#ifdef SRS_AUTO_RTC
+#include <srs_kernel_rtp.hpp>
+#endif
 
 SrsMessageHeader::SrsMessageHeader()
 {
@@ -230,9 +232,11 @@ SrsSharedPtrMessage::~SrsSharedPtrMessage()
         }
     }
 
+#ifdef SRS_AUTO_RTC
     for (int i = 0; i < (int)rtp_packets.size(); ++i) {
         srs_freep(rtp_packets[i]);
     }
+#endif
 }
 
 srs_error_t SrsSharedPtrMessage::create(SrsCommonMessage* msg)
@@ -351,17 +355,21 @@ SrsSharedPtrMessage* SrsSharedPtrMessage::copy()
     copy->payload = ptr->payload;
     copy->size = ptr->size;
 
+#ifdef SRS_AUTO_RTC
     for (int i = 0; i < (int)rtp_packets.size(); ++i) {
         copy->rtp_packets.push_back(rtp_packets[i]->copy());
     }
+#endif
 
     return copy;
 }
 
+#ifdef SRS_AUTO_RTC
 void SrsSharedPtrMessage::set_rtp_packets(const std::vector<SrsRtpSharedPacket*>& pkts)
 {
     rtp_packets = pkts;
 }
+#endif
 
 SrsFlvTransmuxer::SrsFlvTransmuxer()
 {

trunk/src/kernel/srs_kernel_flv.hpp

@@ -288,7 +288,10 @@ public:
     //       video/audio packet use raw bytes, no video/audio packet.
     char* payload;
 
+#ifdef SRS_AUTO_RTC
     std::vector<SrsRtpSharedPacket*> rtp_packets;
+#endif
+
 private:
     class SrsSharedPtrPayload
     {
@@ -344,7 +347,9 @@ public:
     // @remark, assert object is created.
     virtual SrsSharedPtrMessage* copy();
 public:
+#ifdef SRS_AUTO_RTC
     virtual void set_rtp_packets(const std::vector<SrsRtpSharedPacket*>& pkts);
+#endif
 };
 
 // Transmux RTMP packets to FLV stream.

trunk/src/main/srs_main_server.cpp

@@ -50,7 +50,9 @@ using namespace std;
 #include <srs_core_autofree.hpp>
 #include <srs_kernel_file.hpp>
 #include <srs_app_hybrid.hpp>
+#ifdef SRS_AUTO_RTC
 #include <srs_app_rtc_conn.hpp>
+#endif
 
 #ifdef SRS_AUTO_SRT
 #include <srt_server.hpp>
@@ -449,7 +451,9 @@ srs_error_t run_hybrid_server()
     _srs_hybrid->register_server(new SrtServerAdapter());
 #endif
 
+#ifdef SRS_AUTO_RTC
     _srs_hybrid->register_server(new RtcServerAdapter());
+#endif
 
     // Do some system initialize.
     if ((err = _srs_hybrid->initialize()) != srs_success) {