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examples: webrtc: fix unidirectional pipeline

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

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3550>
This commit is contained in:
Guillaume Desmottes 2022-12-09 13:49:44 +01:00
parent a874c9f2d4
commit cbab7ffefb
1 changed files with 1 additions and 1 deletions
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subprojects/gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c
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@ -243,7 +243,7 @@ create_receiver_entry (SoupWebsocketConnection * connection)
"rtph264pay config-interval=-1 name=payloader aggregate-mode=zero-latency ! " "rtph264pay config-interval=-1 name=payloader aggregate-mode=zero-latency ! "
"application/x-rtp,media=video,encoding-name=H264,payload=" "application/x-rtp,media=video,encoding-name=H264,payload="
RTP_PAYLOAD_TYPE " ! webrtcbin. " 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); RTP_AUDIO_PAYLOAD_TYPE " ! webrtcbin. ", &error);
if (error != NULL) { if (error != NULL) {
g_error ("Could not create WebRTC pipeline: %s\n", error->message); g_error ("Could not create WebRTC pipeline: %s\n", error->message);