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This first counts ports (using the builtin slv2 functions to do so
rather than manually as before), then creates descriptors for each port
and finds all port groups. Only then are pad templates created (towards
using group information for creating multi-channel pads).
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- Separate gstsignalprocessor into a separate library (not sure if this
is in the right place, but it works for now anyway)
- Create LV2 element based on LADSPA element, port most discovery
functionality
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Register xmp schemas for photoshop and iptc. Map a few location tags there.
Add more dc tags. Fix reading xmp tag by iteration over known schemas. Add
some more debug logging.
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Coordinate conversion has been verified with exiv2 and some webpages.
Minutes wher totally off and seconds were not used at all.
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Replace port index arrays with a struct for holding more information
than just indices (e.g. groups).
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git://anongit.freedesktop.org/gstreamer/gst-plugins-bad into fdo
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Fix compile warnings on solaris from bzero not being declared (use
memset instead), and strict-aliasing warnings (use a temporary
variable).
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A CELT framesize can be between 64 and 512 samples.
Fix a typo in the bitrate property description.
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git://anongit.freedesktop.org/gstreamer/gst-plugins-bad into fdo
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Fixes compiler message about passing -Wdeclaration-after-statement
to the C++ compiler.
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git://anongit.freedesktop.org/gstreamer/gst-plugins-bad into fdo
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The gst_jack_audio_client_set_active() flags the port as deactivating and uses
a GCond to wait until the jack_process_cb() has run once more and cleared the
flag. This way the client zero's the buffer. This happens if one manyally go
to PAUSED and then to READY, while leting the mainloop run inbetween.
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Enable creation of the demuxer pads for all audio streams, even types we
don't yet support. This means that unsupported types (LPCM, DTS) are output,
but not linked to anything yet. If only unsupported streams are available,
the user hears silence instead of having the pipeline not pre-roll correctly.
This is a prerequisite for hooking up the automatic decoder switching.
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git://anongit.freedesktop.org/gstreamer/gst-plugins-bad into fdo
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Fix the same bug as the previous commit, but in resindvd's copy of
mpegdemux.
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Make sure we send events to all pads. Unmark the notlinked flag on
freshly selected pads to ensure they get data.
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Restore the old segment update behaviour, and instead extend the
close segment stop time if the SCR (last_stop) overruns the calculated
stop position.
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LV2 plugins, short of any surprises, should work with the same basic
functionality as LADSPA plugins.
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Add a property to support self-signed certificates in neonhttpsrc. This property
is FALSE by default.
Fixes #511097
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Fixes #583627
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Re-work the dtsdec element code to unify it with changes made it a52dec,
including support for reverse playback and dynamic channel negotiation
on the source pad.
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Make the debug statement correctly show the 'old' and 'new' button
coordinates, instead of the wrong way around.
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When updating a pad, send the update to half a second behind the SCR,
which avoids ever updating the start time for a pad to beyond the end of
the cell. Also, remember the last actual new-segment start time for each
pad, and use it when closing the segment.
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When the current button number is higher than the number of available
buttons, switch to the highest numbered button rather than the lowest.
Also, don't throw errors when we fail to retrieve some button info
from libdvdnav, just reset the highlight.
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Send the commands-changed navigation message when the set of available
DVD menu button actions changes, and handle the commands navigation
query so that (e.g.) Totem can know about the available navigation
commands.
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Protect pad exposure with a preroll lock to avoid situations
where no-more-pads is fired more than once, or fired just before
the last pad is actually added.
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Send segment updates to the audio and subpicture pads more frequently,
but less often to the video pad, where timestamps appear less often.
This helps with gap filling on some DVDs.
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When creating a filler audio buffer in rsnaudiomunge, generate
a bit more, as audio sinks don't seem to preroll otherwise. This
needs a better algorithm in general, to intelligently fill the
gap, rather than hard-coding a value.
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Make timed still frames work better by extending the current segment
when needed, and restarting the still sequence with the correct
remaining duration when the wait it interrupted by activation of a
highlight NAV packet.
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