Refine SrsSource, bridge to RTC.

trunk/src/app/srs_app_rtc_source.cpp

@@ -152,6 +152,7 @@ SrsRtcSource::SrsRtcSource()
 
     _can_publish = true;
     rtc_publisher_ = NULL;
+    bridger_ = new SrsRtcFromRtmpBridger(this);
 }
 
 SrsRtcSource::~SrsRtcSource()
@@ -165,6 +166,7 @@ SrsRtcSource::~SrsRtcSource()
     srs_freep(rtc);
 
     srs_freep(req);
+    srs_freep(bridger_);
 }
 
 srs_error_t SrsRtcSource::initialize(SrsRequest* r)
@@ -225,6 +227,11 @@ int SrsRtcSource::pre_source_id()
     return _pre_source_id;
 }
 
+ISrsSourceBridger* SrsRtcSource::bridger()
+{
+    return bridger_;
+}
+
 srs_error_t SrsRtcSource::create_consumer(SrsConnection* conn, SrsRtcConsumer*& consumer)
 {
     srs_error_t err = srs_success;
@@ -382,6 +389,19 @@ srs_error_t SrsRtcSource::on_audio_imp(SrsSharedPtrMessage* msg)
         }
     }
 
+    // TODO: FIXME: Support parsing OPUS for RTC.
+    if ((err = format->on_audio(msg)) != srs_success) {
+        return srs_error_wrap(err, "format consume audio");
+    }
+
+    // Parse RTMP message to RTP packets, in FU-A if too large.
+    if ((err = rtc->on_audio(msg, format)) != srs_success) {
+        // TODO: We should support more strategies.
+        srs_warn("rtc: ignore audio error %s", srs_error_desc(err).c_str());
+        srs_error_reset(err);
+        rtc->on_unpublish();
+    }
+
     // copy to all consumer
     if (!drop_for_reduce) {
         for (int i = 0; i < (int)consumers.size(); i++) {
@@ -401,11 +421,6 @@ srs_error_t SrsRtcSource::on_audio_imp(SrsSharedPtrMessage* msg)
         }
     }
 
-    // when sequence header, donot push to gop cache and adjust the timestamp.
-    if (is_sequence_header) {
-        return err;
-    }
-
     // if atc, update the sequence header to abs time.
     if (meta->ash()) {
         meta->ash()->timestamp = msg->timestamp;
@@ -465,11 +480,6 @@ srs_error_t SrsRtcSource::on_video_imp(SrsSharedPtrMessage* msg)
         }
     }
 
-    // when sequence header, donot push to gop cache and adjust the timestamp.
-    if (is_sequence_header) {
-        return err;
-    }
-
     // if atc, update the sequence header to abs time.
     if (meta->vsh()) {
         meta->vsh()->timestamp = msg->timestamp;
@@ -551,3 +561,34 @@ SrsRtcSource* SrsRtcSourceManager::fetch(SrsRequest* r)
 
 SrsRtcSourceManager* _srs_rtc_sources = new SrsRtcSourceManager();
 
+SrsRtcFromRtmpBridger::SrsRtcFromRtmpBridger(SrsRtcSource* source)
+{
+    source_ = source;
+}
+
+SrsRtcFromRtmpBridger::~SrsRtcFromRtmpBridger()
+{
+}
+
+srs_error_t SrsRtcFromRtmpBridger::on_publish()
+{
+    // TODO: FIXME: Should sync with bridger?
+    return source_->on_publish();
+}
+
+srs_error_t SrsRtcFromRtmpBridger::on_audio(SrsSharedPtrMessage* audio)
+{
+    return source_->on_audio_imp(audio);
+}
+
+srs_error_t SrsRtcFromRtmpBridger::on_video(SrsSharedPtrMessage* video)
+{
+    return source_->on_video_imp(video);
+}
+
+void SrsRtcFromRtmpBridger::on_unpublish()
+{
+    // TODO: FIXME: Should sync with bridger?
+    source_->on_unpublish();
+}
+

trunk/src/app/srs_app_rtc_source.hpp

@@ -40,6 +40,7 @@ class SrsSharedPtrMessage;
 class SrsCommonMessage;
 class SrsMessageArray;
 class SrsRtcSource;
+class SrsRtcFromRtmpBridger;
 
 class SrsRtcConsumer : public ISrsConsumerQueue
 {
@@ -94,6 +95,8 @@ private:
     int _pre_source_id;
     SrsRequest* req;
     SrsRtcPublisher* rtc_publisher_;
+    // Transmux RTMP to RTC.
+    SrsRtcFromRtmpBridger* bridger_;
 private:
     // To delivery stream to clients.
     std::vector<SrsRtcConsumer*> consumers;
@@ -118,6 +121,8 @@ public:
     // Get current source id.
     virtual int source_id();
     virtual int pre_source_id();
+    // Get the bridger.
+    ISrsSourceBridger* bridger();
 public:
     // Create consumer
     // @param consumer, output the create consumer.
@@ -144,7 +149,6 @@ public:
     // TODO: FIXME: Merge with on_audio.
     srs_error_t on_rtc_audio(SrsSharedPtrMessage* audio);
     virtual srs_error_t on_video(SrsCommonMessage* video);
-private:
     virtual srs_error_t on_audio_imp(SrsSharedPtrMessage* audio);
     virtual srs_error_t on_video_imp(SrsSharedPtrMessage* video);
 };
@@ -171,5 +175,19 @@ private:
 // Global singleton instance.
 extern SrsRtcSourceManager* _srs_rtc_sources;
 
+class SrsRtcFromRtmpBridger : public ISrsSourceBridger
+{
+private:
+    SrsRtcSource* source_;
+public:
+    SrsRtcFromRtmpBridger(SrsRtcSource* source);
+    virtual ~SrsRtcFromRtmpBridger();
+public:
+    virtual srs_error_t on_publish();
+    virtual srs_error_t on_audio(SrsSharedPtrMessage* audio);
+    virtual srs_error_t on_video(SrsSharedPtrMessage* video);
+    virtual void on_unpublish();
+};
+
 #endif
 

trunk/src/app/srs_app_source.cpp

@@ -50,6 +50,7 @@ using namespace std;
 #include <srs_app_ng_exec.hpp>
 #include <srs_app_dash.hpp>
 #include <srs_protocol_format.hpp>
+#include <srs_app_rtc_source.hpp>
 
 #define CONST_MAX_JITTER_MS         250
 #define CONST_MAX_JITTER_MS_NEG         -250
@@ -1715,6 +1716,7 @@ srs_error_t SrsSourceManager::fetch_or_create(SrsRequest* r, ISrsSourceHandler*
 
     // Use lock to protect coroutine switch.
     // @bug https://github.com/ossrs/srs/issues/1230
+    // TODO: FIXME: Use smaller lock.
     SrsLocker(lock);
     
     SrsSource* source = NULL;
@@ -1729,17 +1731,41 @@ srs_error_t SrsSourceManager::fetch_or_create(SrsRequest* r, ISrsSourceHandler*
     // should always not exists for create a source.
     srs_assert (pool.find(stream_url) == pool.end());
 
+#ifdef SRS_RTC
+    bool rtc_server_enabled = _srs_config->get_rtc_server_enabled();
+    bool rtc_enabled = _srs_config->get_rtc_enabled(r->vhost);
+
+    // Get the RTC source and bridger.
+    SrsRtcSource* rtc = NULL;
+    if (rtc_server_enabled && rtc_enabled) {
+        if ((err = _srs_rtc_sources->fetch_or_create(r, &rtc)) != srs_success) {
+            err = srs_error_wrap(err, "init rtc %s", r->get_stream_url().c_str());
+            goto failed;
+        }
+    }
+#endif
     srs_trace("new source, stream_url=%s", stream_url.c_str());
 
     source = new SrsSource();
     if ((err = source->initialize(r, h)) != srs_success) {
-        return srs_error_wrap(err, "init source %s", r->get_stream_url().c_str());
+        err = srs_error_wrap(err, "init source %s", r->get_stream_url().c_str());
+        goto failed;
     }
 
+#ifdef SRS_RTC
+    // If rtc enabled, bridge RTMP source to RTC,
+    // all RTMP packets will be forwarded to RTC source.
+    if (source && rtc) {
+        source->bridge_to(rtc->bridger());
+    }
+#endif
+    
     pool[stream_url] = source;
-    
     *pps = source;
+    return err;
 
+failed:
+    srs_freep(source);
     return err;
 }
 
@@ -1832,6 +1858,14 @@ void SrsSourceManager::destroy()
     pool.clear();
 }
 
+ISrsSourceBridger::ISrsSourceBridger()
+{
+}
+
+ISrsSourceBridger::~ISrsSourceBridger()
+{
+}
+
 SrsSource::SrsSource()
 {
     req = NULL;
@@ -1843,6 +1877,9 @@ SrsSource::SrsSource()
     _pre_source_id = _source_id = -1;
     die_at = 0;
 
+    handler = NULL;
+    bridger = NULL;
+    
     play_edge = new SrsPlayEdge();
     publish_edge = new SrsPublishEdge();
     gop_cache = new SrsGopCache();
@@ -1948,6 +1985,11 @@ srs_error_t SrsSource::initialize(SrsRequest* r, ISrsSourceHandler* h)
     return err;
 }
 
+void SrsSource::set_bridger(ISrsSourceBridger* v)
+{
+    bridger = v;
+}
+
 srs_error_t SrsSource::on_reload_vhost_play(string vhost)
 {
     srs_error_t err = srs_success;
@@ -2198,6 +2240,11 @@ srs_error_t SrsSource::on_audio_imp(SrsSharedPtrMessage* msg)
         return srs_error_wrap(err, "consume audio");
     }
 
+    // For bridger to consume the message.
+    if (bridger && (err = bridger->on_audio(msg)) != srs_success) {
+        return srs_error_wrap(err, "bridger consume audio");
+    }
+
     // copy to all consumer
     if (!drop_for_reduce) {
         for (int i = 0; i < (int)consumers.size(); i++) {
@@ -2323,6 +2370,11 @@ srs_error_t SrsSource::on_video_imp(SrsSharedPtrMessage* msg)
         return srs_error_wrap(err, "hub consume video");
     }
 
+    // For bridger to consume the message.
+    if (bridger && (err = bridger->on_video(msg)) != srs_success) {
+        return srs_error_wrap(err, "bridger consume video");
+    }
+
     // copy to all consumer
     if (!drop_for_reduce) {
         for (int i = 0; i < (int)consumers.size(); i++) {
@@ -2482,6 +2534,11 @@ srs_error_t SrsSource::on_publish()
     if ((err = handler->on_publish(this, req)) != srs_success) {
         return srs_error_wrap(err, "handle publish");
     }
+
+    if (bridger && (err = bridger->on_publish()) != srs_success) {
+        return srs_error_wrap(err, "bridger publish");
+    }
+
     SrsStatistic* stat = SrsStatistic::instance();
     stat->on_stream_publish(req, _source_id);
     
@@ -2517,8 +2574,13 @@ void SrsSource::on_unpublish()
     srs_assert(handler);
     SrsStatistic* stat = SrsStatistic::instance();
     stat->on_stream_close(req);
+
     handler->on_unpublish(this, req);
 
+    if (bridger) {
+        bridger->on_unpublish();
+    }
+    
     // no consumer, stream is die.
     if (consumers.empty()) {
         die_at = srs_get_system_time();

trunk/src/app/srs_app_source.hpp

@@ -488,7 +488,7 @@ public:
 private:
     virtual srs_error_t do_cycle();
 public:
-    // when system exit, destroy the sources,
+    // when system exit, destroy th`e sources,
     // For gmc to analysis mem leaks.
     virtual void destroy();
 };
@@ -496,6 +496,19 @@ public:
 // Global singleton instance.
 extern SrsSourceManager* _srs_sources;
 
+// For two sources to bridge with each other.
+class ISrsSourceBridger
+{
+public:
+    ISrsSourceBridger();
+    virtual ~ISrsSourceBridger();
+public:
+    virtual srs_error_t on_publish() = 0;
+    virtual srs_error_t on_audio(SrsSharedPtrMessage* audio) = 0;
+    virtual srs_error_t on_video(SrsSharedPtrMessage* video) = 0;
+    virtual void on_unpublish() = 0;
+};
+
 // live streaming source.
 class SrsSource : public ISrsReloadHandler
 {
@@ -529,6 +542,8 @@ private:
     int64_t last_packet_time;
     // The event handler.
     ISrsSourceHandler* handler;
+    // The source bridger for other source.
+    ISrsSourceBridger* bridger;
     // The edge control service
     SrsPlayEdge* play_edge;
     SrsPublishEdge* publish_edge;
@@ -555,6 +570,8 @@ public:
 public:
     // Initialize the hls with handlers.
     virtual srs_error_t initialize(SrsRequest* r, ISrsSourceHandler* h);
+    // Bridge to other source, forward packets to it.
+    void bridge_to(ISrsSourceBridger* v);
 // Interface ISrsReloadHandler
 public:
     virtual srs_error_t on_reload_vhost_play(std::string vhost);