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examples/sendrecv: Remove extra unref of webrtcbin

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The code now constructs webrtcbin with a floating ref and then
gives it to the pipeline. The extra unref is one too many.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3436>
This commit is contained in:
Jan Schmidt 2022-11-19 19:50:38 +11:00
parent b11169bd32
commit 8177588250
1 changed files with 1 additions and 2 deletions
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3
subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c
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@ -481,6 +481,7 @@ start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt)
g_assert_nonnull (webrtc1); g_assert_nonnull (webrtc1);
gst_util_set_object_arg (G_OBJECT (webrtc1), "bundle-policy", "max-bundle"); gst_util_set_object_arg (G_OBJECT (webrtc1), "bundle-policy", "max-bundle");
/* Takes ownership of each: */
gst_bin_add_many (GST_BIN (pipe1), audio_bin, video_bin, webrtc1, NULL); gst_bin_add_many (GST_BIN (pipe1), audio_bin, video_bin, webrtc1, NULL);
if (!gst_element_link (audio_bin, webrtc1)) { if (!gst_element_link (audio_bin, webrtc1)) {
@ -547,8 +548,6 @@ start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt)
/* Incoming streams will be exposed via this signal */ /* Incoming streams will be exposed via this signal */
g_signal_connect (webrtc1, "pad-added", G_CALLBACK (on_incoming_stream), g_signal_connect (webrtc1, "pad-added", G_CALLBACK (on_incoming_stream),
pipe1); pipe1);
/* Lifetime is the same as the pipeline itself */
gst_object_unref (webrtc1);
g_timeout_add (100, (GSourceFunc) webrtcbin_get_stats, webrtc1); g_timeout_add (100, (GSourceFunc) webrtcbin_get_stats, webrtc1);