Commit graph

3722 commits

Author SHA1 Message Date
Stefan Kost
65799096bf gst/volume/gstvolume.c: do not update controlled params, if buffer has no 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
2005-08-29 20:20:42 +00:00
Stefan Kost
242ef1b05b controllerized elements also need to link against controller-libs ;)
Original commit message from CVS:
* configure.ac:
* gst/sine/Makefile.am:
* gst/volume/Makefile.am:
controllerized elements also need to link against controller-libs ;)
2005-08-29 19:52:52 +00:00
Stefan Kost
bef1be2e90 controllerized two audio plugins
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
2005-08-29 19:32:19 +00:00
Andy Wingo
9fbb72f41d ext/vorbis/vorbisdec.c (vorbis_dec_convert, vorbis_dec_push)
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.
2005-08-29 16:15:04 +00:00
Andy Wingo
13c10724db ext/ogg/gstoggdemux.c (gst_ogg_demux_init): Init total_time to
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.
2005-08-29 14:45:12 +00:00
Thomas Vander Stichele
ab1142d4a0 increase default timeout on tests for slow powerbooks
Original commit message from CVS:
increase default timeout on tests for slow powerbooks
2005-08-29 11:37:20 +00:00
Andy Wingo
af5663e170 check/elements/volume.c (cleanup_volume): Fix for running
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.
2005-08-29 11:18:29 +00:00
Andy Wingo
fd30c157b8 check/elements/audioconvert.c: Convert from native endian, not little endian.
Original commit message from CVS:
2005-08-29  Andy Wingo  <wingo@pobox.com>

* check/elements/audioconvert.c: Convert from native endian, not
little endian.
2005-08-29 11:01:06 +00:00
Michael Smith
d5a7ae1915 Add an ogg parser element
Original commit message from CVS:
Add an ogg parser element
2005-08-29 10:52:20 +00:00
Andy Wingo
c32721723b Updates for two-arg init from GST_BOILERPLATE_FULL.
Original commit message from CVS:
2005-08-28  Andy Wingo  <wingo@pobox.com>

* Updates for two-arg init from GST_BOILERPLATE_FULL.
2005-08-28 17:52:45 +00:00
Wim Taymans
b6c368ce67 gst/audioconvert/audioconvert.c: Cleanups.
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.
2005-08-26 18:57:30 +00:00
Wim Taymans
ddec57c089 gst/audioconvert/audioconvert.c: 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),
(audio_convert_convert):
More elegant and working temp buffer selection algo.
2005-08-26 18:43:02 +00:00
Wim Taymans
123aa7de1a gst/audioconvert/audioconvert.c: Use realloc else we lose our original data.
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.
2005-08-26 17:46:45 +00:00
Thomas Vander Stichele
f0f2b133dd use base class' newsegment to properly timestamp
Original commit message from CVS:

use base class' newsegment to properly timestamp
2005-08-26 17:35:28 +00:00
Wim Taymans
98fbd82d1c gst/audioconvert/: Oops, allocate enough space to perform the channel mix.
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.
2005-08-26 17:30:41 +00:00
Wim Taymans
ceb84de916 gst/audioconvert/: Cleanups, librarify a bit, optimize, better negotiation 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.
2005-08-26 15:43:56 +00:00
Jan Schmidt
ee2bc937be ext/ogg/gstoggdemux.c: Another from MikeS:
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.
2005-08-26 11:39:01 +00:00
Jan Schmidt
538eabd559 ext/: Big shout-out to MikeS for fixing this giant memory leak.
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!
2005-08-26 10:50:56 +00:00
Thomas Vander Stichele
3f478d73e9 add more conversion tests
Original commit message from CVS:
add more conversion tests
2005-08-25 18:27:24 +00:00
Thomas Vander Stichele
2042b4f2d9 add more tests
Original commit message from CVS:
add more tests
2005-08-25 18:03:48 +00:00
Thomas Vander Stichele
43332aed85 plug some leaks
Original commit message from CVS:
plug some leaks
2005-08-25 17:32:34 +00:00
Thomas Vander Stichele
6dff9c2cbd check/: add a test for audioconvert
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
2005-08-25 17:20:02 +00:00
Thomas Vander Stichele
8f3a11d6f2 some more testing for perfect streams
Original commit message from CVS:
some more testing for perfect streams
2005-08-25 16:19:39 +00:00
Thomas Vander Stichele
eae1250299 add a check for audioresample
Original commit message from CVS:
add a check for audioresample
2005-08-25 15:44:58 +00:00
Thomas Vander Stichele
f7cb2ba67a show some info on what's left in the queue
Original commit message from CVS:
show some info on what's left in the queue
2005-08-25 14:51:18 +00:00
Thomas Vander Stichele
7647f7fc4e gst/audioresample/: add room for extra overlap samples when asked to 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
2005-08-25 12:31:31 +00:00
Jan Schmidt
2a13ddfd65 gst/playback/gstplaybasebin.c: Revert unpopular change for GST_MESSAGE_SRC to GObject.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (fill_buffer):
Revert unpopular change for GST_MESSAGE_SRC to GObject.
2005-08-25 10:50:44 +00:00
Stefan Kost
be10c8f8ec gst/volume/gstvolume.c: made set_caps function static
Original commit message from CVS:
* gst/volume/gstvolume.c:
made set_caps function static
2005-08-24 21:32:59 +00:00
Wim Taymans
963941df57 ext/vorbis/vorbisenc.c: Stop leaking taglists.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_init),
(gst_vorbisenc_change_state):
Stop leaking taglists.
2005-08-24 21:03:32 +00:00
Thomas Vander Stichele
46e443bdd5 debugging fixes
Original commit message from CVS:
debugging fixes
2005-08-24 18:40:27 +00:00
Thomas Vander Stichele
ffc57169c1 translate me baby
Original commit message from CVS:
translate me baby
2005-08-24 18:13:15 +00:00
Wim Taymans
7824216cef ext/ogg/gstoggdemux.c: Parse seeking events better.
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.
2005-08-24 18:04:45 +00:00
Thomas Vander Stichele
2136419a0a c/: add core's plugins to the mix so that playbin works
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
2005-08-24 18:03:12 +00:00
Christian Schaller
e520824f28 add audioresample to spec file
Original commit message from CVS:
add audioresample to spec file
2005-08-24 17:28:39 +00:00
Christian Schaller
eeffbe7af4 fix broken header setup in Makefile.am
Original commit message from CVS:
fix broken header setup in Makefile.am
2005-08-24 17:21:49 +00:00
Thomas Vander Stichele
ebdf1ac224 dist more
Original commit message from CVS:
dist more
2005-08-24 16:41:46 +00:00
Thomas Vander Stichele
886b43679d check/: add same test as to core, it bitches out on playbin atm.
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.
2005-08-24 16:18:25 +00:00
Wim Taymans
f3ef56e841 configure.ac: Remove audioscale.
Original commit message from CVS:
* configure.ac:
Remove audioscale.
2005-08-24 15:15:57 +00:00
Wim Taymans
da25385ed2 gst/videoscale/gstvideoscale.*: Refactor, make use of BaseTranform really well.
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.
2005-08-24 15:07:54 +00:00
Thomas Vander Stichele
752a59192c port audioresample to basetransform
Original commit message from CVS:
port audioresample to basetransform
2005-08-24 14:08:58 +00:00
Thomas Vander Stichele
41a43b86a8 port audioconvert to basetransform fix ffmpegcsp and videoscale for basetransform changes
Original commit message from CVS:
port audioconvert to basetransform
fix ffmpegcsp and videoscale for basetransform changes
2005-08-24 13:32:52 +00:00
Jan Schmidt
80ad4cff17 check/Makefile.am: Add CHECK_CFLAGS and LDFLAGS
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
2005-08-24 11:56:08 +00:00
Wim Taymans
5ac2327f05 gst-libs/gst/audio/gstringbuffer.*: Added function to clear the ringbuffer.
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.
2005-08-24 11:29:10 +00:00
Andy Wingo
b45323de18 whoops
Original commit message from CVS:
whoops
2005-08-24 11:13:54 +00:00
Andy Wingo
7b9a366d6e sys/v4l/gstv4lelement.c (gst_v4lelement_start)
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.
2005-08-24 11:07:51 +00:00
David Schleef
ae8f41b658 gst/audioresample/Makefile.am: Leet audioresampling code
Original commit message from CVS:
* gst/audioresample/Makefile.am: Leet audioresampling code
* gst/audioresample/buffer.c:
* gst/audioresample/buffer.h:
* gst/audioresample/debug.c:
* gst/audioresample/debug.h:
* gst/audioresample/functable.c:
* gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample.h:
* gst/audioresample/resample_chunk.c:
* gst/audioresample/resample_functable.c:
* gst/audioresample/resample_ref.c:
2005-08-23 19:29:38 +00:00
Wim Taymans
84d0eb4f88 examples/seeking/seek.c: Small seek updates.
Original commit message from CVS:
* examples/seeking/seek.c: (make_vorbis_pipeline),
(make_theora_pipeline), (make_vorbis_theora_pipeline), (do_seek):
Small seek updates.
2005-08-23 18:30:07 +00:00
Thomas Vander Stichele
3cbcad1da3 style fixes
Original commit message from CVS:
style fixes
2005-08-23 18:19:10 +00:00
Andy Wingo
7afb104567 gst-libs/gst/audio/gstbaseaudiosrc.c
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.
2005-08-23 13:29:17 +00:00
Andy Wingo
1bbfa09389 ext/alsa/: Add a device-name property.
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.
2005-08-22 16:50:59 +00:00