mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-18 07:47:17 +00:00
oggmux: Use GstAudioClippingMeta for Opus for accurate end clipping
... instead of relying on the segment. For the clipping at the start we assume a proper value in the OpusHead, as generated by opusparse or opusenc. Transmuxing in general is not guaranteed to produce the correct values, or even have a OpusHead (e.g. when having RTP input). https://bugzilla.gnome.org/show_bug.cgi?id=757153
This commit is contained in:
parent
a135868262
commit
0fa8d284c7
1 changed files with 31 additions and 14 deletions
|
@ -39,6 +39,7 @@
|
|||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/base/gstbytewriter.h>
|
||||
#include <gst/audio/audio.h>
|
||||
#include <gst/tag/tag.h>
|
||||
|
||||
#include "gstoggmux.h"
|
||||
|
@ -806,6 +807,7 @@ gst_ogg_mux_decorate_buffer (GstOggMux * ogg_mux, GstOggPadData * pad,
|
|||
GstClockTimeDiff diff;
|
||||
GstMapInfo map;
|
||||
ogg_packet packet;
|
||||
gboolean end_clip = TRUE;
|
||||
|
||||
/* ensure messing with metadata is ok */
|
||||
buf = gst_buffer_make_writable (buf);
|
||||
|
@ -853,20 +855,35 @@ gst_ogg_mux_decorate_buffer (GstOggMux * ogg_mux, GstOggPadData * pad,
|
|||
/* The last packet may have clipped samples. We need to test against
|
||||
* the segment to ensure we do not use a granpos that encompasses those.
|
||||
*/
|
||||
end_time =
|
||||
gst_ogg_stream_granule_to_time (&pad->map, pad->next_granule + duration);
|
||||
if (end_time > pad->segment.stop
|
||||
&& !GST_CLOCK_TIME_IS_VALID (gst_segment_to_running_time (&pad->segment,
|
||||
GST_FORMAT_TIME, pad->segment.start + end_time))) {
|
||||
gint64 actual_duration =
|
||||
gst_util_uint64_scale_round (pad->segment.stop - time,
|
||||
pad->map.granulerate_n,
|
||||
GST_SECOND * pad->map.granulerate_d);
|
||||
GST_INFO_OBJECT (ogg_mux,
|
||||
"Got clipped last packet of duration %" G_GINT64_FORMAT " (%"
|
||||
G_GINT64_FORMAT " clipped)", actual_duration,
|
||||
duration - actual_duration);
|
||||
duration = actual_duration;
|
||||
if (pad->map.audio_clipping) {
|
||||
GstAudioClippingMeta *cmeta = gst_buffer_get_audio_clipping_meta (buf);
|
||||
|
||||
g_assert (!cmeta || cmeta->format == GST_FORMAT_DEFAULT);
|
||||
if (cmeta && cmeta->end && cmeta->end < duration) {
|
||||
GST_DEBUG_OBJECT (pad->collect.pad,
|
||||
"Clipping %" G_GUINT64_FORMAT " samples at the end", cmeta->end);
|
||||
duration -= cmeta->end;
|
||||
end_clip = FALSE;
|
||||
}
|
||||
}
|
||||
|
||||
if (end_clip) {
|
||||
end_time =
|
||||
gst_ogg_stream_granule_to_time (&pad->map,
|
||||
pad->next_granule + duration);
|
||||
if (end_time > pad->segment.stop
|
||||
&& !GST_CLOCK_TIME_IS_VALID (gst_segment_to_running_time (&pad->segment,
|
||||
GST_FORMAT_TIME, pad->segment.start + end_time))) {
|
||||
gint64 actual_duration =
|
||||
gst_util_uint64_scale_round (pad->segment.stop - time,
|
||||
pad->map.granulerate_n,
|
||||
GST_SECOND * pad->map.granulerate_d);
|
||||
GST_INFO_OBJECT (ogg_mux,
|
||||
"Got clipped last packet of duration %" G_GINT64_FORMAT " (%"
|
||||
G_GINT64_FORMAT " clipped)", actual_duration,
|
||||
duration - actual_duration);
|
||||
duration = actual_duration;
|
||||
}
|
||||
}
|
||||
|
||||
GST_LOG_OBJECT (pad->collect.pad, "buffer ts %" GST_TIME_FORMAT
|
||||
|
|
Loading…
Reference in a new issue