celtenc : default framesize is 480
setcaps : gets framesize and set caps
enc_chain : sets framesize in caps
Added frame-size to static caps of audio/x-celt
Replaced GST_DEBUG_OBJECT by GST_LOG_OBJECT in enc_chain
setcaps: get frame-size from int instead of string
setcaps: use default value for frame-size
There is unfortunately no G_*_FORMAT conversion specifier for differences of
pointers in glib, and we can't rely either on all platforms being 64bit.
So let's just cast the difference to a gint and be done with it.
There's not much point in using GST_DEBUG_FUNCPTR with GObject
virtual functions such as get_property, set_propery, finalize and
dispose, since they'll never be used by anyone anyway. Saves a
few bytes and possibly a sixteenth of a polar bear.
Merged from gst-plugins-base, 6f4c1ac583.
Replaced with "GStreamer maintainers
<gstreamer-devel@lists.sourceforge.net>" or just removed,
depending on the number of other authors.
Merged from gst-plugins-base, 0e9bc5125a.
Set the output caps on the srcpad before pushing the buffer because else core
will do a rather expensive check to see if we can actually accept those caps on
the srcpad.
Merged from gst-plugins-base, bdfb4b46d7.
Install a custom acceptcaps function instead of using the default expensive
check. We accept whatever downstream accepts so we pass along the acceptcaps
call to the downstream peer.
Merged from gst-plugins-base, 5b72f2adf9.
Clarify the ownership of the internal plugin feature list by making
a copy of any passed list. Avoids crashes when freeing a passed list,
or leaks caused by not freeing any internally built list.
When switching to a new seamless segment, use the new
gst_base_src_new_seamless_segment() function, because elements can't
send themselves seek events from the streaming thread.
Also remove GST_PLUGINS_BASE_LIBS from LIBADD since we don't
need to link against any of the -base libs (we just use a define
from the gstaudio headers).
When sending new-segment to a stream, ensure that there is either a valid
PCR, or else wait until there's a PTS on the stream (dropping packets if
needed) in order to avoid generating an invlaid new-segments event.
https://bugzilla.gnome.org/show_bug.cgi?id=595161
g_convert seems to add a single null terminating byte to
the end of the string, even when the output is UTF16, we
force the second 0 byte when copying to the output buffer.
This issue was causing random crashes because it was
assumed that the string resulting from g_convert had
2 extra bytes, but it has only one.
All the input pads for the resin stream selectors have the same source,
and receive/send the same segment info, so there's no need to send a
segment when switching and activating a different pad. Removing it makes
on-the-fly audio track switching work properly.
After switching audio track, mark the next buffer discont.
Avoid a potential deadlock that happens when a call to
gst_clock_id_wait_async() immediately calls the callback.
Improve previous chapter seeking slightly by avoiding 'segment
start != -1' assertions when there is no previous chapter and
the seek therefore fails.
