| Age | Commit message (Collapse) | Author | Files | Lines |
|---|
|
Original commit message from CVS:
2005-09-02 Andy Wingo <wingo@pobox.com>
* All plugins updated for element state changes.
|
|
Original commit message from CVS:
2005-09-02 Andy Wingo <wingo@pobox.com>
* All plugins updated for element state changes.
|
|
Original commit message from CVS:
remove hook
|
|
Original commit message from CVS:
increase timeout a little
|
|
Original commit message from CVS:
update translations
|
|
more than a 10th of a second.
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.
|
|
int64, so let's use g_value_{set|get}_int64(...
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:
* check/elements/audioconvert.c: (setup_audioconvert):
* check/elements/audioresample.c: (setup_audioresample):
* check/elements/volume.c: (setup_volume):
Fix checks.
|
|
Original commit message from CVS:
* common/gtk-doc-plugins.mak:
* common/plugins.xsl:
* docs/plugins/Makefile.am:
make module a param
|
|
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
|
|
timestamp
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:
increase default timeout on tests for slow powerbooks
|
|
Original commit message from CVS:
2005-08-29 Andy Wingo <wingo@pobox.com>
* check/elements/volume.c (cleanup_volume): Fix for running
CK_FORK=no.
|
|
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:
Add an ogg parser element
|
|
Original commit message from CVS:
2005-08-28 Andy Wingo <wingo@pobox.com>
* Updates for two-arg init from GST_BOILERPLATE_FULL.
|
|
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):
Cleanups.
|
|
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),
(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:
use base class' newsegment to properly timestamp
|
|
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):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps), (gst_audio_convert_get_unit_size),
(gst_audio_convert_transform_caps),
(gst_audio_convert_fixate_caps), (gst_audio_convert_transform):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_mix):
Oops, allocate enough space to perform the channel mix.
|
|
and more.
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* 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):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_init),
(gst_audio_convert_dispose), (gst_audio_convert_parse_caps),
(gst_audio_convert_get_unit_size),
(gst_audio_convert_transform_caps),
(gst_audio_convert_fixate_caps), (gst_audio_convert_set_caps),
(gst_audio_convert_transform_ip), (gst_audio_convert_transform):
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_unset_matrix),
(gst_channel_mix_fill_identical),
(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
(gst_channel_mix_fill_normalize), (gst_channel_mix_fill_matrix),
(gst_channel_mix_setup_matrix), (gst_channel_mix_passthrough),
(gst_channel_mix_mix):
* gst/audioconvert/gstchannelmix.h:
Cleanups, librarify a bit, optimize, better negotiation and more.
|
|
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:
add more conversion tests
|
|
Original commit message from CVS:
add more tests
|
|
Original commit message from CVS:
plug some leaks
|
|
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:
some more testing for perfect streams
|
|
Original commit message from CVS:
add a check for audioresample
|
|
Original commit message from CVS:
show some info on what's left in the queue
|
|
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
|
|
to GObject.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (fill_buffer):
Revert unpopular change for GST_MESSAGE_SRC to GObject.
|
|
Original commit message from CVS:
* gst/volume/gstvolume.c:
made set_caps function static
|
|
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_init),
(gst_vorbisenc_change_state):
Stop leaking taglists.
|
|
Original commit message from CVS:
debugging fixes
|
|
Original commit message from CVS:
translate me baby
|
|
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
(gst_ogg_pad_event), (gst_ogg_demux_factory_filter),
(gst_ogg_pad_submit_packet), (gst_ogg_chain_new),
(gst_ogg_demux_init), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_collect_chain_info), (gst_ogg_demux_collect_info),
(gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
Parse seeking events better.
Unref static caps.
Generate correct newsegment events, fixes seeking in live oggs.
* ext/theora/theoradec.c: (theora_dec_src_query),
(theora_dec_src_event), (theora_dec_src_getcaps),
(theora_dec_sink_event), (theora_dec_push), (theora_dec_chain):
Use newsegment values to report correct play time.
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_src_event), (vorbis_dec_sink_event):
* ext/vorbis/vorbisdec.h:
Parse and use newsegment values to report correct play time.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
Clear ringbuffer on flush.
Use newsegment values to calculate playback time.
* sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times):
Basesink does newsegment calculations for us now.
|
|
Original commit message from CVS:
* check/Makefile.am:
* configure.ac:
add core's plugins to the mix so that playbin works
* check/generic/states.c: (GST_START_TEST):
set a 0 timeout on pipelines, so they don't force the next
state change
* gst/playback/gstplaybasebin.c: (setup_source), (prepare_output),
(gst_play_base_bin_change_state):
remove the crappy error handling and do GST error handling
|
|
Original commit message from CVS:
add audioresample to spec file
|
|
Original commit message from CVS:
fix broken header setup in Makefile.am
|
|
Original commit message from CVS:
dist more
|
|
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:
* configure.ac:
Remove audioscale.
|
|
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:
port audioresample to basetransform
|
