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tests: rtp: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
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1 changed files with 2 additions and 2 deletions
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@ -2234,11 +2234,11 @@ GST_START_TEST (test_rtcp_compound_padding)
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rtcp_buffer = gst_rtcp_buffer_new (1400);
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fail_unless (gst_rtcp_buffer_map (rtcp_buffer, GST_MAP_READWRITE, &rtcp));
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rtcp_packet = g_slice_new0 (GstRTCPPacket);
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rtcp_packet = g_new0 (GstRTCPPacket, 1);
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fail_unless (gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_RR,
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rtcp_packet));
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gst_rtcp_packet_rr_set_ssrc (rtcp_packet, 1);
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g_slice_free (GstRTCPPacket, rtcp_packet);
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g_free (rtcp_packet);
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gst_rtcp_buffer_unmap (&rtcp);
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fail_unless (gst_rtcp_buffer_validate (rtcp_buffer));
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