RTC: Cleanup code, remove RTC from SrsSource

trunk/src/app/srs_app_rtc_source.cpp

@@ -159,6 +159,7 @@ SrsRtcSource::SrsRtcSource()
     rtc = new SrsRtc();
 
     _can_publish = true;
+    rtc_publisher_ = NULL;
 }
 
 SrsRtcSource::~SrsRtcSource()

trunk/src/app/srs_app_source.cpp

@@ -50,10 +50,6 @@ using namespace std;
 #include <srs_app_ng_exec.hpp>
 #include <srs_app_dash.hpp>
 #include <srs_protocol_format.hpp>
-#ifdef SRS_RTC
-#include <srs_app_rtc.hpp>
-#include <srs_app_rtc_conn.hpp>
-#endif
 
 #define CONST_MAX_JITTER_MS         250
 #define CONST_MAX_JITTER_MS_NEG         -250
@@ -867,9 +863,6 @@ SrsOriginHub::SrsOriginHub()
     dash = new SrsDash();
     dvr = new SrsDvr();
     encoder = new SrsEncoder();
-#ifdef SRS_RTC
-    rtc = new SrsRtc();
-#endif
 #ifdef SRS_HDS
     hds = new SrsHds();
 #endif
@@ -913,12 +906,6 @@ srs_error_t SrsOriginHub::initialize(SrsSource* s, SrsRequest* r)
     if ((err = format->initialize()) != srs_success) {
         return srs_error_wrap(err, "format initialize");
     }
-
-#ifdef SRS_RTC
-    if ((err = rtc->initialize(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");
@@ -1018,15 +1005,6 @@ srs_error_t SrsOriginHub::on_audio(SrsSharedPtrMessage* shared_audio)
                   flv_sample_sizes[c->sound_size], flv_sound_types[c->sound_type],
                   srs_flv_srates[c->sound_rate]);
     }
-
-#ifdef SRS_RTC
-    // TODO: FIXME: Support parsing OPUS for 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.
@@ -1120,16 +1098,6 @@ srs_error_t SrsOriginHub::on_video(SrsSharedPtrMessage* shared_video, bool is_se
     if (format->vcodec && !format->vcodec->is_avc_codec_ok()) {
         return err;
     }
-
-#ifdef SRS_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.
-        srs_warn("rtc: ignore video error %s", srs_error_desc(err).c_str());
-        srs_error_reset(err);
-        rtc->on_unpublish();
-    }
-#endif
     
     if ((err = hls->on_video(msg, format)) != srs_success) {
         // TODO: We should support more strategies.
@@ -1199,12 +1167,6 @@ srs_error_t SrsOriginHub::on_publish()
         return srs_error_wrap(err, "encoder publish");
     }
 
-#ifdef SRS_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");
     }
@@ -1242,9 +1204,6 @@ void SrsOriginHub::on_unpublish()
     destroy_forwarders();
     
     encoder->on_unpublish();
-#ifdef SRS_RTC
-    rtc->on_unpublish();
-#endif
     hls->on_unpublish();
     dash->on_unpublish();
     dvr->on_unpublish();
@@ -1922,10 +1881,6 @@ SrsSource::SrsSource()
     
     _srs_config->subscribe(this);
     atc = false;
-
-#ifdef SRS_RTC
-    rtc_publisher_ = NULL;
-#endif
 }
 
 SrsSource::~SrsSource()
@@ -2703,26 +2658,3 @@ string SrsSource::get_curr_origin()
     return play_edge->get_curr_origin();
 }
 
-#ifdef SRS_RTC
-SrsMetaCache* SrsSource::cached_meta()
-{
-    return meta;
-}
-
-SrsRtcPublisher* SrsSource::rtc_publisher()
-{
-    return rtc_publisher_;
-}
-
-void SrsSource::set_rtc_publisher(SrsRtcPublisher* v)
-{
-    rtc_publisher_ = v;
-}
-
-srs_error_t SrsSource::on_rtc_audio(SrsSharedPtrMessage* audio)
-{
-    // TODO: FIXME: Merge with on_audio.
-    // TODO: FIXME: Print key information.
-    return on_audio_imp(audio);
-}
-#endif

trunk/src/app/srs_app_source.hpp

@@ -62,9 +62,6 @@ class SrsBuffer;
 #ifdef SRS_HDS
 class SrsHds;
 #endif
-#ifdef SRS_RTC
-class SrsRtcPublisher;
-#endif
 
 // The time jitter algorithm:
 // 1. full, to ensure stream start at zero, and ensure stream monotonically increasing.
@@ -358,10 +355,6 @@ private:
 private:
     // The format, codec information.
     SrsRtmpFormat* format;
-#ifdef SRS_RTC
-    // rtc handler
-    SrsRtc* rtc;
-#endif
     // hls handler.
     SrsHls* hls;
     // The DASH encoder.
@@ -560,10 +553,6 @@ private:
     // The last die time, when all consumers quit and no publisher,
     // We will remove the source when source die.
     srs_utime_t die_at;
-#ifdef SRS_RTC
-private:
-    SrsRtcPublisher* rtc_publisher_;
-#endif
 public:
     SrsSource();
     virtual ~SrsSource();
@@ -630,17 +619,6 @@ public:
     virtual void on_edge_proxy_unpublish();
 public:
     virtual std::string get_curr_origin();
-#ifdef SRS_RTC
-public:
-    // For RTC, we need to package SPS/PPS(in cached meta) before each IDR.
-    SrsMetaCache* cached_meta();
-    // Get and set the publisher, passed to consumer to process requests such as PLI.
-    SrsRtcPublisher* rtc_publisher();
-    void set_rtc_publisher(SrsRtcPublisher* v);
-    // When got RTC audio message, which is encoded in opus.
-    // TODO: FIXME: Merge with on_audio.
-    srs_error_t on_rtc_audio(SrsSharedPtrMessage* audio);
-#endif
 };
 
 #endif