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

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) {