Age | Commit message (Collapse) | Author | Files | Lines |
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Original commit message from CVS:
fix spec file
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Original commit message from CVS:
back to head
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Original commit message from CVS:
release time
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Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
No need to take stream lock here.
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Original commit message from CVS:
fix build
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Original commit message from CVS:
some disting and build fixes
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Original commit message from CVS:
Gsmdec ported to 0.9. Tested with filesrc ! gsmdec ! alsasink and osssink.
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Original commit message from CVS:
* gst/audioresample/Makefile.am:
* gst/audioresample/debug.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/resample.c: Convert to using gst debugging
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Original commit message from CVS:
fix build and use of GST_LIBS
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Original commit message from CVS:
tta plugin ported to 0.9
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Original commit message from CVS:
Setting caps on the outgoing buffers.
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Original commit message from CVS:
fix Makefile.am ugh
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Original commit message from CVS:
Fixed configure.ac and ext/sdl/Makefile.am for sdl port to 0.9.
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Original commit message from CVS:
sdlvideosink ported to 0.9 and tested with filesrc ! mpeg2dec
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Original commit message from CVS:
sdlvideosink ported to 0.9 and tested with filesrc ! mpeg2dec
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Original commit message from CVS:
add check-valgrind target
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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.
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Original commit message from CVS:
Ported speed Plugin to GStreamer 0.9
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Original commit message from CVS:
back to HEAD
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Original commit message from CVS:
releasing 0.9.1
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Original commit message from CVS:
Fix up all the state change functions.
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Original commit message from CVS:
cleaning up bad
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Original commit message from CVS:
created gst-plugins-bad
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Original commit message from CVS:
* common/gtk-doc-plugins.mak:
* docs/plugins/Makefile.am:
fix distcheck
* gst/audioresample/resample.c:
fix wrong docstring
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Original commit message from CVS:
Ported GSM Encoder to GStreamer 0.9
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Original commit message from CVS:
2005-09-02 Andy Wingo <wingo@pobox.com>
* All plugins updated for element state changes.
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Original commit message from CVS:
2005-09-02 Andy Wingo <wingo@pobox.com>
* All plugins updated for element state changes.
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Original commit message from CVS:
update PORTED_09 file
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Original commit message from CVS:
Faac (AAC Encoder) ported for GStreamer 0.9
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Original commit message from CVS:
remove libdir
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Original commit message from CVS:
all these plugins are moved to gst-plugins-ugly
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Original commit message from CVS:
Port LPCM decoder to 0.9
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Original commit message from CVS:
* configure.ac:
Remove plugins that should have disappeared.
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Original commit message from CVS:
* autogen.sh:
* configure.ac:
Make autogen work again.
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Original commit message from CVS:
* check/elements/audioconvert.c: (setup_audioconvert):
* check/elements/audioresample.c: (setup_audioresample):
* check/elements/volume.c: (setup_volume):
Fix checks.
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Original commit message from CVS:
remove stuff
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Original commit message from CVS:
all these plugins are moved to gst-plugins-good
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Original commit message from CVS:
Ported to GStreamer 0.9. Need to fix performance issues.
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Original commit message from CVS:
2005-08-28 Andy Wingo <wingo@pobox.com>
* Updates for two-arg init from GST_BOILERPLATE.
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_init): Use
the second arg for the class, because G_OBJECT_GET_CLASS (self)
returns the wrong thing.
(gst_signal_processor_add_pad_from_template): Make pads of the
right type.
* ext/ladspa/gstladspa.c (gst_ladspa_class_get_param_spec): Make
writable param specs G_PARAM_CONSTRUCT so default values work.
(gst_ladspa_init): Use the second arg for the class.
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Original commit message from CVS:
2005-08-28 Andy Wingo <wingo@pobox.com>
* Updates for two-arg init from GST_BOILERPLATE_FULL.
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Original commit message from CVS:
use base class' newsegment to properly timestamp
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compile and register. I have more features tha...
Original commit message from CVS:
2005-08-26 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstladspa.c:
* ext/ladspa/gstladspa.h: Finish porting, still doesn't work but
it does compile and register. I have more features than you.
* ext/ladspa/gstsignalprocessor.h:
* ext/ladspa/gstsignalprocessor.c: Updates, bug fixen.
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Original commit message from CVS:
do proper cleanup/creation, fixes state changes
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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
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Original commit message from CVS:
some more testing for perfect streams
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Original commit message from CVS:
add a check for audioresample
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Original commit message from CVS:
show some info on what's left in the queue
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transform size protect against possible mem corr...
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
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Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_message_new):
Revert unpopular change for GST_MESSAGE_SRC to GObject.
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Original commit message from CVS:
add missing files
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