Add RTC stream merging demo by FFmpeg. 4.0.97

README.md

@@ -176,6 +176,7 @@ The ports used by SRS:
 
 ## V4 changes
 
+* v4.0, 2021-05-03, Add RTC stream merging demo by FFmpeg. 4.0.97
 * v4.0, 2021-05-02, Add one to one demo. 4.0.96
 * v4.0, 2021-04-20, Support RTC2RTMP bridger and shared FastTimer. 4.0.95
 * v4.0, 2021-04-20, Refine transcoder to support aac2opus and opus2aac. 4.0.94

trunk/3rdparty/signaling/www/demos/js/srs.sig.js

@@ -100,7 +100,6 @@ function SrsRtcSignalingParse(location) {
 
     let wsPort = location.href.split('wsp=')[1];
     wsPort = wsPort? wsPort.split('&')[0] : location.host.split(':')[1];
-    wsHost = wsPort? wsHost.split(':')[0] + ':' + wsPort : wsHost;
 
     let host = location.href.split('host=')[1];
     host = host? host.split('&')[0] : location.hostname;
@@ -114,6 +113,29 @@ function SrsRtcSignalingParse(location) {
     let autostart = location.href.split('autostart=')[1];
     autostart = autostart && autostart.split('&')[0] === 'true';
 
+    // Remove data in query.
+    if (query) {
+        query = query.replace('wss=' + wsSchema, '');
+        query = query.replace('wsh=' + wsHost, '');
+        query = query.replace('wsp=' + wsPort, '');
+        if (room) {
+            query = query.replace('room=' + room, '');
+        }
+        query = query.replace('display=' + display, '');
+        query = query.replace('autostart=' + autostart, '');
+
+        while (query.indexOf('&&') >= 0) {
+            query = query.replace('&&', '&');
+        }
+        query = query.replace('?&', '?');
+        if (query.lastIndexOf('?') == query.length - 1) {
+            query = query.substr(0, query.length - 1);
+        }
+    }
+
+    // Regenerate the host of websocket.
+    wsHost = wsPort? wsHost.split(':')[0] + ':' + wsPort : wsHost;
+
     return {
         query: query, wsSchema: wsSchema, wsHost: wsHost, host: host,
         room: room, display: display, autostart: autostart,

trunk/3rdparty/signaling/www/demos/one2one.html

@@ -63,6 +63,25 @@
             <a href="javascript:control_alert_peer()">Alert</a>
         </div>
     </div>
+
+    <div class="accordion">
+        <div class="accordion-group">
+            <div class="accordion-heading">
+                <a href="#" id="ff_preview" target="_blank" class="accordion-toggle">
+                    FFmpeg合流转直播预览：rtmp://<span class="ff_host"></span>/<span class="ff_app"></span>/merge
+                </a>
+            </div>
+            <div class="accordion-body collapse in">
+                <div class="accordion-inner">
+                    ffmpeg -f flv -i rtmp://<span class="ff_host"></span>/<span class="ff_app"></span>/<span class="ff_first"></span> -f flv -i rtmp://<span class="ff_host"></span>/<span class="ff_app"></span>/<span class="ff_second"></span> \ <br/>
+                    &nbsp;&nbsp;&nbsp;&nbsp; -filter_complex "[1:v]scale=w=96:h=72[ckout];[0:v][ckout]overlay=x=W-w-10:y=H-h-10[out]" -map "[out]" \ <br/>
+                    &nbsp;&nbsp;&nbsp;&nbsp; -c:v libx264 -profile:v high -preset medium \ <br/>
+                    &nbsp;&nbsp;&nbsp;&nbsp; -filter_complex amix -c:a aac \ <br/>
+                    &nbsp;&nbsp;&nbsp;&nbsp; -f flv -y rtmp://<span class="ff_host"></span>/<span class="ff_app"></span>/merge
+                </div>
+            </div>
+        </div>
+    </div>
 </div>
 <script type="text/javascript">
     var sig = null;
@@ -139,6 +158,10 @@
         };
 
         var startPublish = function (host, room, display) {
+            $(".ff_first").each(function(i,e) {
+                $(e).text(display);
+            });
+
             var url = 'webrtc://' + host + '/' + room + '/' + display + conf.query;
             $('#rtc_media_publisher').show();
             $('#publisher').show();
@@ -153,7 +176,7 @@
             };
 
             return publisher.publish(url).then(function(session){
-                $('#self').text('Self: ' + display);
+                $('#self').text('Self: ' + url);
             }).catch(function (reason) {
                 publisher.close();
                 $('#rtc_media_publisher').hide();
@@ -162,6 +185,10 @@
         };
 
         var startPlay = function (host, room, display) {
+            $(".ff_second").each(function(i,e) {
+                $(e).text(display);
+            });
+
             var url = 'webrtc://' + host + '/' + room + '/' + display + conf.query;
             $('#rtc_media_player').show();
             $('#player').show();
@@ -192,6 +219,14 @@
         conf.room && $('#txt_room').val(conf.room);
         $('#txt_display').val(conf.display);
 
+        $(".ff_host").each(function(i,e) {
+            $(e).text(conf.host);
+        });
+        $(".ff_app").each(function(i,e) {
+            $(e).text($('#txt_room').val());
+        });
+        $('#ff_preview').attr('href', 'http://ossrs.net/players/srs_player.html?app=' + $('#txt_room').val() + '&stream=merge.flv&server=' + conf.host + '&vhost=' + conf.host + '&autostart=true');
+
         $("#btn_start").click(startDemo);
         if (conf.autostart) {
             startDemo();

trunk/src/app/srs_app_rtc_conn.cpp

@@ -1012,6 +1012,10 @@ srs_error_t SrsRtcPublishStream::initialize(SrsRequest* r, SrsRtcStreamDescripti
             return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "rtmp stream %s busy", r->get_stream_url().c_str());
         }
 
+        // Disable GOP cache for RTC2RTMP bridger, to keep the streams in sync,
+        // especially for stream merging.
+        rtmp->set_cache(false);
+
         SrsRtmpFromRtcBridger *bridger = new SrsRtmpFromRtcBridger(rtmp);
         if ((err = bridger->initialize(r)) != srs_success) {
             srs_freep(bridger);

trunk/src/core/srs_core_version4.hpp

@@ -26,6 +26,6 @@
 
 #define VERSION_MAJOR       4
 #define VERSION_MINOR       0
-#define VERSION_REVISION    96
+#define VERSION_REVISION    97
 
 #endif