From 81775882509c6db43e29aa63fcd2ad86f877ff94 Mon Sep 17 00:00:00 2001 From: Jan Schmidt <jan@centricular.com> Date: Sat, 19 Nov 2022 19:50:38 +1100 Subject: [PATCH] examples/sendrecv: Remove extra unref of webrtcbin 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> --- subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c index e71fe3bbac..7e1c88d905 100644 --- a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c +++ b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c @@ -481,6 +481,7 @@ start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt) g_assert_nonnull (webrtc1); 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); 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 */ g_signal_connect (webrtc1, "pad-added", G_CALLBACK (on_incoming_stream), pipe1); - /* Lifetime is the same as the pipeline itself */ - gst_object_unref (webrtc1); g_timeout_add (100, (GSourceFunc) webrtcbin_get_stats, webrtc1);