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

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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
This commit is contained in:
Marijn Suijten 2020-11-25 17:51:24 +01:00 committed by Sebastian Dröge
parent 552da8569b
commit 030b1b3fa5
1 changed files with 2 additions and 2 deletions
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4
tests/check/elements/rtp-payloading.c
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@ -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;