Original commit message from CVS:
2005-09-15 Andy Wingo <wingo@pobox.com>
* configure.ac (plugindir): Remove the EOL matcher from the
regexp, as it causes me problems. Libtool? Make? Who knows?
Original commit message from CVS:
* check/generic/states.c:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
* gst/playback/gstdecodebin.c: (gst_decode_bin_init):
Fixes for changes in registry API.
* configure.ac: Only export gst_plugins_desc. Add -no-undefined
to GST_PLUGIN_LDFLAGS.
* ext/libvisual/visual.c: Make the library shut up.
* gst-libs/gst/audio/audio.c: Don't define a plugin in a library.
* gst-libs/gst/audio/gstaudiofilter.c: same
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
Audioconvert derives from GstBaseTransform and should
link to the library with our base elements to avoid
unresolved symbols. Makes things work with MinGW (#316160)
* gst/playback/test4.c: (main):
Fix MinGW build problem and use g_usleep() instead of
sleep() (#316162)
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
Cleanups, speedups, simplifications, added back support
for 24 bits.
Original commit message from CVS:
* check/Makefile.am:
* check/pipelines/simple_launch_lines.c: (setup_pipeline),
(run_pipeline), (GST_START_TEST), (simple_launch_lines_suite):
Add extra tests for basetransform based components.
Comment out the test_element_negotiation test until we decide
if it's testing correct behaviour.
* ext/libvisual/visual.c: (gst_visual_init), (get_buffer),
(gst_visual_chain), (gst_visual_change_state):
Slightly more correct but still bogus timestamping.
Fix state change function.
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_class_init):
* gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
(gst_videoscale_prepare_size), (gst_videoscale_set_caps),
(gst_videoscale_prepare_image):
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_transform_ip):
Basetransform updates. Enable passthrough modes.
* sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
(gst_ximagesink_renegotiate_size), (gst_ximagesink_xcontext_get),
(gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
Negotiation fix that allows the window to return to the original
size and renegotiate passthrough upstream. Extra debug output.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit):
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_transform):
fixing lost sync, some more debugging
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
(gst_ximagesink_check_xshm_calls), (gst_ximagesink_xcontext_get):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_init),
(gst_xvimagesink_check_xshm_calls):
Fix compilation when XShm is not available.
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
