Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_dispose),
(gst_visual_getcaps), (gst_visual_src_setcaps),
(gst_visual_sink_setcaps), (get_buffer), (gst_visual_chain),
(gst_visual_change_state):
Finish fixing up libvisual plugin so that it runs.
Original commit message from CVS:
* check/pipelines/simple_launch_lines.c: (GST_START_TEST):
added another test that failes for me (test is not active by default)
Original commit message from CVS:
* configure.ac:
In the output at the end, don't show the first plugin on the same
line as "Core plug-ins, always built:".
Indent the output as for other plugin categories
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_create):
#define that can be used to not use peer buffer_alloc functions for
test purposes.
* sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
(gst_ximage_buffer_get_type), (gst_ximagesink_ximage_new),
(gst_ximagesink_show_frame):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_init),
(gst_xvimage_buffer_get_type), (gst_xvimagesink_setcaps),
(gst_xvimagesink_show_frame):
Error case handling fixes. gst-launch fakesrc ! x[v]imagesink now
fails gracefully instead of XError aborting or deadlocking.
Original commit message from CVS:
* configure.ac: Enable libvisual plugin.
* ext/libvisual/Makefile.am:
* ext/libvisual/visual.c: Fixes to make it compile.
Original commit message from CVS:
* common/gst-xmlinspect.py:
* common/gtk-doc-plugins.mak:
only inspect plugins for this given package
require gst-python 0.9
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Resync if the buffer timestamps drift more than a 10th
of a second.
Original commit message from CVS:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_set_property),
(gst_v4lsrc_get_property):
The 'timestamp-offset' property is registered as an int64, so
let's use g_value_{set|get}_int64() in our setter and getter
functions (makes it work and fixes warnings with gst-inspect).
Original commit message from CVS:
* examples/seeking/seek.c: (make_mp3_pipeline),
(make_mpeg_pipeline), (seek_cb), (start_seek), (stop_seek),
(play_cb), (pause_cb), (stop_cb):
update the example
Original commit message from CVS:
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_transform):
do not update controlled params, if buffer has no timestamp
Original commit message from CVS:
* configure.ac:
* gst/sine/Makefile.am:
* gst/volume/Makefile.am:
controllerized elements also need to link against controller-libs ;)
Original commit message from CVS:
reviewed by: <delete if not using a buddy>
* docs/libs/tmpl/gstcolorbalance.sgml:
* docs/libs/tmpl/gstgconf.sgml:
* docs/libs/tmpl/gstmixer.sgml:
* docs/libs/tmpl/gstringbuffer.sgml:
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_create):
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_transform):
controllerized two audio plugins
Original commit message from CVS:
2005-08-29 Andy Wingo <wingo@pobox.com>
* ext/vorbis/vorbisdec.c (vorbis_dec_convert, vorbis_dec_push)
(vorbis_handle_data_packet): Fix some int overflow errors.
Original commit message from CVS:
2005-08-29 Andy Wingo <wingo@pobox.com>
* ext/ogg/gstoggdemux.c (gst_ogg_demux_init): Init total_time to
-1.
(gst_ogg_demux_perform_seek): Clamp segment_stop only if it's
valid.
(gst_ogg_pad_submit_packet): Subtract the chain's begin_time only
if it's valid. Fixed streaming-mode playback.
Original commit message from CVS:
2005-08-29 Andy Wingo <wingo@pobox.com>
* check/elements/audioconvert.c: Convert from native endian, not
little endian.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(audio_convert_convert):
More elegant and working temp buffer selection algo.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(get_temp_buffer), (audio_convert_convert):
Use realloc else we lose our original data.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (ogg_find_peek):
Another from MikeS:
During typefinding, don't support negative offsets
(offsets from the end of the stream) in our typefind->peek() function
- nothing embedded in ogg ever needs them. However, we need to recognise
those requests and reject them, otherwise we return invalid pointers.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
(vorbisdec_finalize), (vorbis_handle_type_packet):
Big shout-out to MikeS for fixing this giant memory leak.
Huzzah!
Original commit message from CVS:
* check/Makefile.am:
* check/elements/audioconvert.c: (setup_audioconvert),
(cleanup_audioconvert), (get_int_caps), (verify_convert),
(GST_START_TEST), (audioconvert_suite), (main):
add a test for audioconvert
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
note that for buffers of 1/3 sec this means DURATION(c) is
one nanosecond more than for a and b
Original commit message from CVS:
* gst/audioresample/debug.c:
* gst/audioresample/gstaudioresample.c:
add room for extra overlap samples when asked to transform size
protect against possible mem corruption and check for discrepancies
between written size and outbuffer's size so we can warn for
potential problems
* gst/audioresample/resample.c: (resample_init),
(resample_get_output_size_for_input), (resample_get_output_size),
(resample_set_n_channels), (resample_set_format):
set debug level based on RESAMPLE_DEBUG env var
make sure that get_output_size* returns a whole number of
sample_size
set sample_size each time either channel or format is set
* gst/audioresample/resample_chunk.c: (resample_scale_chunk):
* gst/audioresample/resample_functable.c:
(resample_scale_functable):
* gst/audioresample/resample_ref.c: (resample_scale_ref):
remove r->sample_size, it's done in resample.c now
add some debugging to the ref implementation
make sure we only give back bytes that are wholes of the sample
size
Original commit message from CVS:
* check/Makefile.am:
* check/generic/states.c: (GST_START_TEST), (states_suite), (main):
add same test as to core, it bitches out on playbin atm.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_videoscale_init),
(gst_videoscale_prepare_size), (parse_caps),
(gst_videoscale_set_caps), (gst_videoscale_get_size),
(gst_videoscale_prepare_image), (gst_videoscale_transform_ip),
(gst_videoscale_transform):
* gst/videoscale/gstvideoscale.h:
Refactor, make use of BaseTranform really well.
Original commit message from CVS:
* check/Makefile.am:
Add CHECK_CFLAGS and LDFLAGS
* gst/playback/gstplaybasebin.c: (fill_buffer):
GST_MESSAGE_SRC became a GObject
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_set_sample),
(gst_ring_buffer_clear_all):
* gst-libs/gst/audio/gstringbuffer.h:
Added function to clear the ringbuffer.
Original commit message from CVS:
2005-08-24 Andy Wingo <wingo@pobox.com>
* sys/v4l/gstv4lelement.c (gst_v4lelement_start)
(gst_v4lelement_stop): Call _start and _stop for xoverlay instead
of _open and _close.
* sys/v4l/gstv4lxoverlay.h:
* sys/v4l/gstv4lxoverlay.c (gst_v4l_xoverlay_set_xwindow_id): Open
an Xv connection here, instead of all the time. Make Xv only be
loaded if you axe for it. Kindof a workaround for buggy behaviour
of Xv when using remote xservers (XvQueryExtension would block).
(gst_v4l_xoverlay_stop, gst_v4l_xoverlay_start): New functions,
replace the _open and _close public API. Only start the xv
connection if necessary.
(gst_v4l_xoverlay_open, gst_v4l_xoverlay_close): Made static.
Original commit message from CVS:
* examples/seeking/seek.c: (make_vorbis_pipeline),
(make_theora_pipeline), (make_vorbis_theora_pipeline), (do_seek):
Small seek updates.
Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosrc.c
(gst_base_audio_src_fixate): Only fixate endianness if it is
present in the caps.
Original commit message from CVS:
2005-08-22 Andy Wingo <wingo@pobox.com>
* ext/alsa/gstalsasink.c (gst_alsasink_get_property):
* ext/alsa/gstalsasrc.c (gst_alsasrc_get_property): Add a
device-name property.
Original commit message from CVS:
2005-08-22 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstaudiosrc.c: Implement open_device and
close_device in the ring buffer, like gstaudiosink.
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixer.c: Not a GObject any more. Include a nifty
macro to implement the interface without much code. Cleanups.
* ext/alsa/gstalsasrc.h:
* ext/alsa/gstalsasrc.c: Be a mixer. Open device and mixer in
READY.
* ext/alsa/Makefile.am: Add new files.
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsamixerelement.c: Split element code out from
mixer code so that alsasrc can be a mixer too.
Original commit message from CVS:
2005-08-19 Andy Wingo <wingo@pobox.com>
* ext/alsa/gstalsamixertrack.h:
* ext/alsa/gstalsamixertrack.c:
* ext/alsa/gstalsamixeroptions.h:
* ext/alsa/gstalsamixeroptions.c:
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixer.c: Port to 0.9.
* ext/alsa/Makefile.am: Build mixer, mixeroptions, mixertracks.
Remove gstalsa.c and alsaclock. No more cruft here.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_change_state):
Open and close device in READY<->NULL state change.
Original commit message from CVS:
2005-08-16 Andy Wingo <wingo@pobox.com>
* examples/seeking/Makefile.am: Don't compile non-compiling
compiled objects with the compiler.
* examples/seeking/seek.c (make_dv_pipeline): Update for new DV
elements.
Original commit message from CVS:
2005-08-12 Philippe Khalaf <burger@speedy.org>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Made a thread to release the queue.
Removed timestamp conversion for now.
Original commit message from CVS:
2005-08-10 Philippe Khalaf <burger@speedy.org>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Added rtp timestamp -> gst timestamp conversion.
Fixed several problems with queue.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
(gst_riff_parse_chunk):
Fix bug in debug message and add some more debug messages.
Original commit message from CVS:
2005-08-08 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_change_state): Open the device in NULL->READY
like good elements should. Close on READY->NULL too.
* gst-libs/gst/audio/gstaudiosink.c
(gst_audioringbuffer_open_device,
(gst_audioringbuffer_close_device, gst_audioringbuffer_acquire)
(gst_audioringbuffer_release): Updates for new ring buffer API,
hook into the new audio sink api.
* gst-libs/gst/audio/gstaudiosink.h (GstAudioSinkClass.open)
(GstAudioSinkClass.close): Just open and close the device -- no
resource allocation or configuration.
(GstAudioSinkClass.prepare, GstAudioSinkClass.unprepare): New
vmethods, handle device setup and resource allocation.
* ext/alsa/gstalsasink.c (gst_alsasink_open, gst_alsasink_close)
(gst_alsasink_prepare, gst_alsasink_unprepare): Update for new
base class API.
* gst-libs/gst/audio/gstringbuffer.h
(GstRingBufferClass.open_device, GstRingBufferClass.close_device):
New vmethods.
* gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_open_device)
(gst_ring_buffer_close_device, gst_ring_buffer_device_is_open):
New API functions. The device should be opened before acquiring
and closed after releasing.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (remove_sinks):
Remove visualization from parent explicitely; works around some
apparent refcount issue that I haven't tracked down yet.
Original commit message from CVS:
* common/gtk-doc-plugins.mak:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
renamed to actual element names, so much nicer to look at
* docs/plugins/tmpl/gstmultifdsink.sgml:
remove
* docs/plugins/tmpl/multifdsink.sgml:
* docs/plugins/tmpl/tcpserversink.sgml:
add
* ext/alsa/gstalsa.c:
* ext/cdparanoia/gstcdparanoia.c: (cdparanoia_get_property):
* ext/ogg/gstoggmux.c:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
* gst/playback/gstdecodebin.c:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init):
* gst/tcp/gsttcpserversink.c:
various fixes and documentation additions
Original commit message from CVS:
2005-08-04 Andy Wingo <wingo@pobox.com>
* gst/videoscale/gstvideoscale.c (gst_videoscale_get_size):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c
(gst_ffmpegcsp_get_size): Adapt to API changes.
* gst/videoscale/gstvideoscale.c (gst_videoscale_transform_ip):
Implement an in-place do-nothing transform.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
(gst_ximagesink_renegotiate_size):
Do not set new window sizes yet if we prepare a new buffer size
for upstream renegotiation (software scaling) at some point in the
future, because this new size waqs not actually accepted yet. Once
accepted, renegotiation later on will set the new sizes just fine.
Fixes a videotestsrc ! queue ! videoscale ! ximagesink xoverlay
embedding testcase.
Original commit message from CVS:
2005-08-03 Andy Wingo <wingo@pobox.com>
* sys/ximage/ximagesink.c (gst_ximagesink_renegotiate_size):
(gst_ximagesink_buffer_alloc):
Protect the height, width, and desired_caps with the pool_lock.
Fixes videotestsrc ! queue ! ximagesink.
Original commit message from CVS:
2005-08-02 Jan Schmidt <thaytan@mad.scientist.com>
* ext/ogg/gstoggmux.c: (gst_ogg_mux_change_state):
Stop collectpads before calling the parent state
change function on PAUSED->READY.
Original commit message from CVS:
* configure.ac:
When testing for X libs, use the X CFlags
* gst/adder/gstadder.c: (gst_adder_change_state):
Stop the collectpads before calling parent state change function
on PAUSED->READY, otherwise we deadlock deactivating pads.
Original commit message from CVS:
2005-07-29 Andy Wingo <wingo@pobox.com>
* ext/alsa/gstalsaplugin.c (plugin_init): We are primary audio
sinks.
* ext/alsa/gstalsasink.c (alsasink_sink_factory): Advertise our
support of both endiannesses.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element):
Switch to auto*sink elements as default sinks; add volume element
so that volume control in totem works.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gen_preroll_element):
* gst/playback/gstplaybin.c: (setup_sinks),
(gst_play_bin_change_state):
Refcount fix and more comments.
Original commit message from CVS:
2005-07-21 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* sys/ximage/Makefile.am:
* sys/ximage/ximage.c: (plugin_init):
* sys/ximage/ximagesink.c:
Prepare for adding ximagesrc, rename of plugin to ximage etc.
Original commit message from CVS:
2005-07-20 Andy Wingo <wingo@pobox.com>
* gst/videoscale/vs_image.c (vs_image_scale_nearest_YUYV): Typo
fix (?), fixes a seggie mcfalterson (#310894).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (fill_buffer):
Use _new_custom() so we can set custom message types for buffering
messages.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/gconf/.cvsignore:
* gst-libs/gst/gconf/Makefile.am:
* gst-libs/gst/gconf/test-gconf.c:
* pkgconfig/Makefile.am:
* pkgconfig/gstreamer-gconf-uninstalled.pc.in:
* pkgconfig/gstreamer-gconf.pc.in:
Remove gconf stuff, use gconf elements instead from now on.
Original commit message from CVS:
* examples/seeking/seek.c: (make_dv_pipeline),
(make_vorbis_theora_pipeline), (query_rates),
(query_positions_elems), (query_positions_pads), (do_seek):
Make correct DV pipeline.
Original commit message from CVS:
2005-07-18 Andy Wingo <wingo@pobox.com>
* configure.ac (DEFAULT_AUDIOSINK, DEFAULT_AUDIOSRC): Use alsa by
default. Also because it's the only thing that really works. (This
is used in the GConf elements).
Use AS_LIBTOOL_TAGS.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Align samples even if we have roundoff errors in the
timestamp conversion.
Original commit message from CVS:
* gst/tcp/gstfdset.c: (gst_fdset_free):
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_init),
(gst_multifdsink_add), (gst_multifdsink_remove),
(gst_multifdsink_clear), (gst_multifdsink_get_stats),
(gst_multifdsink_remove_client_link),
(gst_multifdsink_client_queue_data),
(gst_multifdsink_client_queue_caps),
(gst_multifdsink_client_queue_buffer),
(gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients),
(gst_multifdsink_stop):
* gst/tcp/gstmultifdsink.h:
0.8 backporting.
* sys/ximage/ximagesink.c: (gst_ximagesink_show_frame):
Also draw image when not from a pool.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered),
(mute_stream), (silence_stream):
Small debug additions.
Original commit message from CVS:
make GST_PLUGIN_LDFLAGS only be flags; GST_LIBS should be
added manually to each Makefile.am so we are sure it goes
*last* and doesn't add -L flags before linking in libs of our
own, like, say, internal .la libs, that then accidentally pick
up the installed copy.
Original commit message from CVS:
2005-07-13 Andy Wingo <wingo@pobox.com>
* sys/v4l/gstv4lsrc.c (gst_v4lsrc_stop): Fix a spurious warning.
(gst_v4lsrc_fixate): Fixate on format as well.
Original commit message from CVS:
2005-07-13 Andy Wingo <wingo@pobox.com>
* sys/xvimage/xvimagesink.c (gst_xvimage_buffer_destroy)
(gst_xvimagesink_xvimage_new): Ref the xvimagesink while the
buffer points to it.
(gst_xvimagesink_check_xshm_calls): Don't use our xvimage buffer,
rather just doing X calls ourselves. Also fixes a memleak.
Original commit message from CVS:
2005-07-12 Andy Wingo <wingo@pobox.com>
* sys/v4l/gstv4lsrc.c (gst_v4lsrc_get_property)
(gst_v4lsrc_set_property, gst_v4lsrc_class_init, gst_v4lsrc_init)
(gst_v4lsrc_create): Re-add the copy-mode property, default to
TRUE to avoid deadlocks if an element holds on to our buffers.
Original commit message from CVS:
2005-07-07 Andy Wingo <wingo@pobox.com>
* sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Also fixate the
framerate. Need to get a handle on when exactly this function is
called, tho.
Original commit message from CVS:
2005-07-07 Andy Wingo <wingo@pobox.com>
* sys/v4l/v4lsrc_calls.h:
* sys/v4l/v4lsrc_calls.c: Remove sync-related stuff.
(gst_v4lsrc_get_fps_list): Moved here from gstv4lsrc.c.
(gst_v4lsrc_buffer_new): Totally derive from GstBuffer.
* sys/v4l/v4l_calls.h: Cast to V4lElement.
* sys/v4l/v4l_calls.c: Header loc fixen, don't load mjpeg, all
v4lelements are sources.
* sys/v4l/gstv4lxoverlay.h:
* sys/v4l/gstv4lxoverlay.c:
* sys/v4l/gstv4ltuner.h:
* sys/v4l/gstv4ltuner.c: Header loc fixen.
* sys/v4l/gstv4lsrc.h:
* sys/v4l/gstv4lsrc.c: Crucial GPL update. Clean up a bit, port to
PushSrc/BaseSrc. Removed most sync-related properties, videorate
or something should handle that. Made a live source.
* sys/v4l/gstv4lelement.h:
* sys/v4l/gstv4lelement.c: Derive from GstPushSrc. No more
signals. Some cleanups.
* sys/v4l/gstv4lcolorbalance.h: Interface header update.
* sys/v4l/gstv4l.c: Don't register v4lelement, or the jpeg/mjpeg
stuff.
* sys/v4l/Makefile.am: Build everything except the jpeg/mjpeg
stuff.
* sys/Makefile.am (SUBDIRS): Hit the V4L crack pipe.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_src_getcaps),
(theora_dec_push), (theora_handle_data_packet):
Prepare for better timestamp fix later.
* gst/audioconvert/gstaudioconvert.c:
List most accurate caps first
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_loop):
Use proper pad task function.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_show_frame):
Fix deadlock when alloc failed.
Original commit message from CVS:
2005-07-04 Andy Wingo <wingo@pobox.com>
* gst/audioconvert/gstaudioconvert.c (gst_audio_convert_fixate):
No refcount leakage.
Original commit message from CVS:
2005-07-01 Andy Wingo <wingo@pobox.com>
* ext/theora/theoradec.c (theora_dec_src_getcaps): Implement a
getcaps to do explicit caps. Needs to be done in all decoders,
possibly via a base class.
* configure.ac (GST_PLUGIN_LDFLAGS): Add videoscale.
* ext/ogg/gstoggdemux.c (gst_ogg_pad_typefind): No need to set
caps on the sink pad, just rely on the pad template. Also, setting
ANY caps on a pad is not valid because the caps are not fixed.
* sys/ximage/ximagesink.c (gst_ximagesink_buffer_alloc): Set the
caps on the buffer, and get the width from the desired_caps if
they're set.
(gst_ximagesink_renegotiate_size): Implement via setting the
desired_caps on the ximagesink.
(gst_ximagesink_setcaps): Only reset the width of the player if it
wasn't already set. Not sure if this is right.
(gst_ximagesink_show_frame): Memcpy only for normal buffers.
* sys/ximage/ximagesink.h (desired_caps): New field, is the caps
that the user wants. NULL unless the window has been resized.
* gst/volume/gstvolume.c (volume_transform): Adapt to
basetransform refcount changes.
Original commit message from CVS:
2005-07-01 Andy Wingo <wingo@pobox.com>
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/gstvideoscale.h: Clean up, port to 0.9. Derives
from BaseTransform, implements a transform_caps. Removed dead code
including some PAR stuff that was never reached -- should probably
be added back somehow.
