Bridger: Support RTC2RTMP bridger and shared FastTimer. 4.0.95

Author: winlinvip

README.md

@@ -157,6 +157,7 @@ Other important wiki:
 
 ## V4 changes
 
+* v5.0, 2021-04-20, Support RTC2RTMP bridger and shared FastTimer. 4.0.95
 * v5.0, 2021-04-20, Refine transcoder to support aac2opus and opus2aac. 4.0.94
 * v4.0, 2021-05-01, Timer: Extract and apply shared FastTimer. 4.0.93
 * v4.0, 2021-04-29, RTC: Support AV1 for Chrome M90. 4.0.91

trunk/conf/full.conf

@@ -571,6 +571,15 @@ vhost rtc.vhost.srs.com {
         #       discard     Discard aac audio packet.
         # default: transcode
         aac transcode;
+        ###############################################################
+        # For transmuxing RTC to RTMP.
+        # Whether trans-mux RTC to RTMP streaming.
+        # Default: off
+        rtc_to_rtmp off;
+        # The PLI interval in seconds, for RTC to RTMP.
+        # Note the available range is [0.5, 30]
+        # Default: 6.0
+        pli_for_rtmp 6.0;
     }
     ###############################################################
     # For transmuxing RTMP to RTC, it will impact the default values if RTC is on.

trunk/src/app/srs_app_config.cpp

@@ -3940,7 +3940,8 @@ srs_error_t SrsConfig::check_normal_config()
                 for (int j = 0; j < (int)conf->directives.size(); j++) {
                     string m = conf->at(j)->name;
                     if (m != "enabled" && m != "bframe" && m != "aac" && m != "stun_timeout" && m != "stun_strict_check"
-                        && m != "dtls_role" && m != "dtls_version" && m != "drop_for_pt") {
+                        && m != "dtls_role" && m != "dtls_version" && m != "drop_for_pt" && m != "rtc_to_rtmp"
+                        && m != "pli_for_rtmp") {
                         return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.rtc.%s of %s", m.c_str(), vhost->arg0().c_str());
                     }
                 }
@@ -5226,6 +5227,46 @@ int SrsConfig::get_rtc_drop_for_pt(string vhost)
     return ::atoi(conf->arg0().c_str());
 }
 
+bool SrsConfig::get_rtc_to_rtmp(string vhost)
+{
+    static bool DEFAULT = false;
+
+    SrsConfDirective* conf = get_rtc(vhost);
+    if (!conf) {
+        return DEFAULT;
+    }
+
+    conf = conf->get("rtc_to_rtmp");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+
+    return SRS_CONF_PERFER_FALSE(conf->arg0());
+}
+
+srs_utime_t SrsConfig::get_rtc_pli_for_rtmp(string vhost)
+{
+    static srs_utime_t DEFAULT = 6 * SRS_UTIME_SECONDS;
+
+    SrsConfDirective* conf = get_rtc(vhost);
+    if (!conf) {
+        return DEFAULT;
+    }
+
+    conf = conf->get("pli_for_rtmp");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+
+    srs_utime_t v = (srs_utime_t)(::atof(conf->arg0().c_str()) * SRS_UTIME_SECONDS);
+    if (v < 500 * SRS_UTIME_MILLISECONDS || v > 30 * SRS_UTIME_SECONDS) {
+        srs_warn("Reset pli %dms to %dms", srsu2msi(v), srsu2msi(DEFAULT));
+        return DEFAULT;
+    }
+
+    return v;
+}
+
 bool SrsConfig::get_rtc_nack_enabled(string vhost)
 {
     static bool DEFAULT = true;

trunk/src/app/srs_app_config.hpp

@@ -562,6 +562,8 @@ class SrsConfig
     std::string get_rtc_dtls_role(std::string vhost);
     std::string get_rtc_dtls_version(std::string vhost);
     int get_rtc_drop_for_pt(std::string vhost);
+    bool get_rtc_to_rtmp(std::string vhost);
+    srs_utime_t get_rtc_pli_for_rtmp(std::string vhost);
     bool get_rtc_nack_enabled(std::string vhost);
     bool get_rtc_nack_no_copy(std::string vhost);
     bool get_rtc_twcc_enabled(std::string vhost);

trunk/src/app/srs_app_listener.cpp

@@ -536,6 +536,10 @@ srs_error_t SrsUdpMuxListener::listen()
 
     srs_freep(trd);
     trd = new SrsSTCoroutine("udp", this, cid);
+
+    //change stack size to 256K, fix crash when call some 3rd-part api.
+    ((SrsSTCoroutine*)trd)->set_stack_size(1 << 18);
+
     if ((err = trd->start()) != srs_success) {
         return srs_error_wrap(err, "start thread");
     }

trunk/src/app/srs_app_rtc_conn.cpp

@@ -998,6 +998,28 @@ srs_error_t SrsRtcPublishStream::initialize(SrsRequest* r, SrsRtcStreamDescripti
     }
     source->set_publish_stream(this);
 
+    // Bridge to rtmp
+    bool rtc_to_rtmp = _srs_config->get_rtc_to_rtmp(req->vhost);
+    if (rtc_to_rtmp) {
+        SrsSource *rtmp = NULL;
+        if ((err = _srs_sources->fetch_or_create(r, _srs_hybrid->srs()->instance(), &rtmp)) != srs_success) {
+            return srs_error_wrap(err, "create source");
+        }
+
+        // TODO: FIMXE: Check it in SrsRtcConnection::add_publisher?
+        if (!rtmp->can_publish(false)) {
+            return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "rtmp stream %s busy", r->get_stream_url().c_str());
+        }
+
+        SrsRtmpFromRtcBridger *bridger = new SrsRtmpFromRtcBridger(rtmp);
+        if ((err = bridger->initialize(r)) != srs_success) {
+            srs_freep(bridger);
+            return srs_error_wrap(err, "create bridger");
+        }
+
+        source->set_bridger(bridger);
+    }
+
     return err;
 }