This patch prevents timestamp like "1 1:00:00", which would have been seen
as hour 101 by our parser, and allow single digit hour, minute and seconds
as it's already supported by the parser, and also by other implementation
like in mplayer. This fixes bug 657872.
https://bugzilla.gnome.org/show_bug.cgi?id=657872
This will cause not-linked errors that usually don't happen
because normal decoders/parsers will set srcpad caps before
allocating buffers from downstream.
g_value_get_object() does not give us our own ref.
Fixes "Trying to dispose object "flacparse", but it still has a parent "registry0".
You need to let the parent manage the object instead of unreffing the object directly."
and similar warnings.
https://bugzilla.gnome.org/show_bug.cgi?id=658416
If both quality and bitrate are set, libtheora will try to meet
both constraints, causing it to prefer emitting a smaller number
of good frames, to emitting the full number of frames that would
not meet the requested quality. This causes a slideshow effect
when the bitrate is low and the quality is high. And the default
theoraenc is high (48/63).
So only set quality when it is requested, and leave it unset
otherwise.
https://bugzilla.gnome.org/show_bug.cgi?id=658443
Merge in doc updates for audio enums from 0.10, and get rid
of the #if #else in the enum list, since that confuses gtk-doc.
Conflicts:
gst-libs/gst/audio/audio.c
gst-libs/gst/audio/audio.h
This is done by adding a capsfilter after every parser/converter that contains
all possible caps supported by downstream elements. A capsfilter is necessary
here because the decoder is only selected after the parser selected a format
and the parser can't know what downstream would support otherwise.
Remove the _ in front of the endianness prefix.
Remove the _3 postfix for the 24 bits formats.
Add a _32 postfix after the formats that occupy extra space beyond their
natural size.
The result is that the GST_AUDIO_NE() macro can simply append the endianness
after all formats and that we only specify a different sample width when it is
different from the natural size of the sample. This makes things more consistent
and follows the pulseaudio conventions instead of the alsa ones.
Sort muxers based on their caps and ranking before iterating to
find one that fits the profile.
Sorting is done by putting the elements that have a pad template
that can produce the exact caps that is on the profile. For example:
when asking for "video/quicktime, variant=iso", muxers that
have this exact caps on their pad templates will be put first on
the list than ones that have only "video/quicktime".
https://bugzilla.gnome.org/show_bug.cgi?id=651496
This reverts commit 105814e2c7.
The general consensus seems to be that we should revert this for
now. If such behaviour is desired, we should probably enable it
via a flag. And maybe use the scaletempo plugin instead.
Adds a Lanczos-derived scaling method, which is rather slow, but very
high quality. Adds a few properties that can be used to tune various
scaling properties: sharpness, sharpen, envelope, dither. Not currently
Orcified, but was designed with that in mind.
The average_period_set variable can be accessed in different threads, so
always lock it when reading. Furthermore when switching to averaging
mode we should make sure we don't have cached buffers that aren't used
in that mode. And any modeswitch will cause the latency to change, so we
should post a NewLatency message
