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| author | Sebastian Dröge <sebastian@centricular.com> | 2020-12-16 18:32:25 +0200 |
|---|---|---|
| committer | GStreamer Merge Bot <gitlab-merge-bot@gstreamer-foundation.org> | 2020-12-17 09:00:24 +0000 |
| commit | 70facfa8d32317678078f8b88fcc26b899df4c44 (patch) | |
| tree | 9e0217a77edf33173090b00ec8315e3c284f6dc6 /sys/decklink | |
| parent | 8af91be222c3c9cb7ed430cfe29b29ea148f290a (diff) | |
| download | gstreamer-plugins-bad-70facfa8d32317678078f8b88fcc26b899df4c44.tar.gz | |
decklinkaudiosrc: Fix duration of the first audio frame after each discont
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1886>
Diffstat (limited to 'sys/decklink')
| -rw-r--r-- | sys/decklink/gstdecklinkaudiosrc.cpp | 2 |
1 files changed, 2 insertions, 0 deletions
diff --git a/sys/decklink/gstdecklinkaudiosrc.cpp b/sys/decklink/gstdecklinkaudiosrc.cpp index 5824f9087..7ebf0106c 100644 --- a/sys/decklink/gstdecklinkaudiosrc.cpp +++ b/sys/decklink/gstdecklinkaudiosrc.cpp @@ -654,6 +654,8 @@ retry: // Convert to the sample numbers start_offset = gst_util_uint64_scale (start_time, self->info.rate, GST_SECOND); + // Convert back to round down to a sample multiple and get rid of rounding errors + start_time = gst_util_uint64_scale (start_offset, GST_SECOND, self->info.rate); end_offset = start_offset + sample_count; end_time = gst_util_uint64_scale_int (end_offset, GST_SECOND, |