examples: webrtc: fix unidirectional pipeline

'autoaudiosrc' does not have a 'is-live' property.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3550>

diff --git a/subprojects/gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c b/subprojects/gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c
index 593d861..ef52f41 100644 (file)
--- a/subprojects/gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c
+++ b/subprojects/gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c
@@ -243,7 +243,7 @@ create_receiver_entry (SoupWebsocketConnection * connection)
       "rtph264pay config-interval=-1 name=payloader aggregate-mode=zero-latency ! "
       "application/x-rtp,media=video,encoding-name=H264,payload="
       RTP_PAYLOAD_TYPE " ! webrtcbin. "
-      "autoaudiosrc is-live=1 ! queue max-size-buffers=1 leaky=downstream ! audioconvert ! audioresample ! opusenc ! rtpopuspay pt="
+      "autoaudiosrc ! queue max-size-buffers=1 leaky=downstream ! audioconvert ! audioresample ! opusenc ! rtpopuspay pt="
       RTP_AUDIO_PAYLOAD_TYPE " ! webrtcbin. ", &error);
   if (error != NULL) {
     g_error ("Could not create WebRTC pipeline: %s\n", error->message);