tests/rtp-payloading: Use new AudioFormatInfo::fill_silence function

The function is renamed to be properly associated with AudioFormatInfo
(its instance) instead of AudioFormat (an unrelated enum), see [1] for
the rename itself.

[1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/940

diff --git a/tests/check/elements/rtp-payloading.c b/tests/check/elements/rtp-payloading.c
index c7f4977..49db40f 100644 (file)
--- a/tests/check/elements/rtp-payloading.c
+++ b/tests/check/elements/rtp-payloading.c
@@ -1623,7 +1623,7 @@ GST_START_TEST (rtp_vorbis_renegotiate)
   gst_audio_info_from_caps (&info, caps);
   buffer = gst_buffer_new_and_alloc (44100 * info.bpf);
   gst_buffer_map (buffer, &map, GST_MAP_WRITE);
-  gst_audio_format_fill_silence (info.finfo, map.data, map.size);
+  gst_audio_format_info_fill_silence (info.finfo, map.data, map.size);
   gst_buffer_unmap (buffer, &map);
   GST_BUFFER_PTS (buffer) = 0;
   GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
@@ -1647,7 +1647,7 @@ GST_START_TEST (rtp_vorbis_renegotiate)
   gst_audio_info_from_caps (&info, caps);
   buffer = gst_buffer_new_and_alloc (48000 * info.bpf);
   gst_buffer_map (buffer, &map, GST_MAP_WRITE);
-  gst_audio_format_fill_silence (info.finfo, map.data, map.size);
+  gst_audio_format_info_fill_silence (info.finfo, map.data, map.size);
   gst_buffer_unmap (buffer, &map);
   GST_BUFFER_PTS (buffer) = 0;
   GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;