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| author | Sebastian Dröge <sebastian@centricular.com> | 2020-12-16 18:32:25 +0200 |
|---|---|---|
| committer | Tim-Philipp Müller <tim@centricular.com> | 2021-01-13 00:25:21 +0000 |
| commit | 2afc46b5b724276d1e02eb91accdc6fa2badac38 (patch) | |
| tree | 67ee9f1828a4b1afc51a89a2b5fb658a90c28ae1 /sys | |
| parent | a35760806abf4196c6b1b58ff808ba6302c7fd61 (diff) | |
| download | gstreamer-plugins-bad-2afc46b5b724276d1e02eb91accdc6fa2badac38.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/1891>
Diffstat (limited to 'sys')
| -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 2fef934c8..1ec2dd045 100644 --- a/sys/decklink/gstdecklinkaudiosrc.cpp +++ b/sys/decklink/gstdecklinkaudiosrc.cpp @@ -632,6 +632,8 @@ 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, |