Fix streamer example for safari

examples/streamer/client.js

@@ -54,12 +54,10 @@ function createPeerConnection() {
 
     // connect audio / video
     pc.addEventListener('track', function (evt) {
-        if (evt.track.kind == 'video') {
-            document.getElementById('media').style.display = 'block';
-            document.getElementById('video').srcObject = evt.streams[0];
-        } else {
-            document.getElementById('audio').srcObject = evt.streams[0];
-        }
+        document.getElementById('media').style.display = 'block';
+        const videoTag = document.getElementById('video');
+        videoTag.srcObject = evt.streams[0];
+        videoTag.play();
     });
 
     let time_start = null;

examples/streamer/index.html

@@ -52,7 +52,6 @@
 
 <div id="media" style="display: none">
     <h2>Media</h2>
-    <audio id="audio" autoplay></audio>
     <video id="video" autoplay playsinline></video>
 </div>
 

examples/streamer/main.cpp

@@ -214,7 +214,7 @@ int main(int argc, char **argv) try {
 shared_ptr<ClientTrackData> addVideo(const shared_ptr<PeerConnection> pc, const uint8_t payloadType, const uint32_t ssrc, const string cname, const string msid, const function<void (void)> onOpen) {
     auto video = Description::Video(cname);
     video.addH264Codec(payloadType);
-    video.addSSRC(ssrc, cname, msid);
+    video.addSSRC(ssrc, cname, msid, cname);
     auto track = pc->addTrack(video);
     // create RTP configuration
     auto rtpConfig = shared_ptr<RtpPacketizationConfig>(new RtpPacketizationConfig(ssrc, cname, payloadType, H264RtpPacketizer::defaultClockRate));
@@ -232,7 +232,7 @@ shared_ptr<ClientTrackData> addVideo(const shared_ptr<PeerConnection> pc, const
 shared_ptr<ClientTrackData> addAudio(const shared_ptr<PeerConnection> pc, const uint8_t payloadType, const uint32_t ssrc, const string cname, const string msid, const function<void (void)> onOpen) {
     auto audio = Description::Audio(cname);
     audio.addOpusCodec(payloadType);
-    audio.addSSRC(ssrc, cname, msid);
+    audio.addSSRC(ssrc, cname, msid, cname);
     auto track = pc->addTrack(audio);
     // create RTP configuration
     auto rtpConfig = shared_ptr<RtpPacketizationConfig>(new RtpPacketizationConfig(ssrc, cname, payloadType, OpusRtpPacketizer::defaultClockRate));