Thomas Vander Stichele [Fri, 26 Aug 2005 17:35:28 +0000 (17:35 +0000)]
use base class' newsegment to properly timestamp
Original commit message from CVS:
use base class' newsegment to properly timestamp
Wim Taymans [Fri, 26 Aug 2005 17:30:41 +0000 (17:30 +0000)]
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.
Wim Taymans [Fri, 26 Aug 2005 15:43:56 +0000 (15:43 +0000)]
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.
Jan Schmidt [Fri, 26 Aug 2005 11:39:01 +0000 (11:39 +0000)]
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.
Jan Schmidt [Fri, 26 Aug 2005 10:50:56 +0000 (10:50 +0000)]
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!
Thomas Vander Stichele [Thu, 25 Aug 2005 18:27:24 +0000 (18:27 +0000)]
add more conversion tests
Original commit message from CVS:
add more conversion tests
Thomas Vander Stichele [Thu, 25 Aug 2005 18:03:48 +0000 (18:03 +0000)]
add more tests
Original commit message from CVS:
add more tests
Thomas Vander Stichele [Thu, 25 Aug 2005 17:32:34 +0000 (17:32 +0000)]
plug some leaks
Original commit message from CVS:
plug some leaks
Thomas Vander Stichele [Thu, 25 Aug 2005 17:20:02 +0000 (17:20 +0000)]
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
Thomas Vander Stichele [Thu, 25 Aug 2005 16:19:39 +0000 (16:19 +0000)]
some more testing for perfect streams
Original commit message from CVS:
some more testing for perfect streams
Thomas Vander Stichele [Thu, 25 Aug 2005 15:44:58 +0000 (15:44 +0000)]
add a check for audioresample
Original commit message from CVS:
add a check for audioresample
Thomas Vander Stichele [Thu, 25 Aug 2005 14:51:18 +0000 (14:51 +0000)]
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
Thomas Vander Stichele [Thu, 25 Aug 2005 12:31:31 +0000 (12:31 +0000)]
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
Jan Schmidt [Thu, 25 Aug 2005 10:50:44 +0000 (10:50 +0000)]
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.
Stefan Kost [Wed, 24 Aug 2005 21:32:59 +0000 (21:32 +0000)]
gst/volume/gstvolume.c: made set_caps function static
Original commit message from CVS:
* gst/volume/gstvolume.c:
made set_caps function static
Wim Taymans [Wed, 24 Aug 2005 21:03:32 +0000 (21:03 +0000)]
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.
Thomas Vander Stichele [Wed, 24 Aug 2005 18:40:27 +0000 (18:40 +0000)]
debugging fixes
Original commit message from CVS:
debugging fixes
Thomas Vander Stichele [Wed, 24 Aug 2005 18:13:15 +0000 (18:13 +0000)]
translate me baby
Original commit message from CVS:
translate me baby
Wim Taymans [Wed, 24 Aug 2005 18:04:45 +0000 (18:04 +0000)]
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.
Thomas Vander Stichele [Wed, 24 Aug 2005 18:03:12 +0000 (18:03 +0000)]
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
Christian Schaller [Wed, 24 Aug 2005 17:28:39 +0000 (17:28 +0000)]
add audioresample to spec file
Original commit message from CVS:
add audioresample to spec file
Christian Schaller [Wed, 24 Aug 2005 17:21:49 +0000 (17:21 +0000)]
fix broken header setup in Makefile.am
Original commit message from CVS:
fix broken header setup in Makefile.am
Thomas Vander Stichele [Wed, 24 Aug 2005 16:41:46 +0000 (16:41 +0000)]
dist more
Original commit message from CVS:
dist more
Thomas Vander Stichele [Wed, 24 Aug 2005 16:18:25 +0000 (16:18 +0000)]
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.
Wim Taymans [Wed, 24 Aug 2005 15:15:57 +0000 (15:15 +0000)]
configure.ac: Remove audioscale.
Original commit message from CVS:
* configure.ac:
Remove audioscale.
Wim Taymans [Wed, 24 Aug 2005 15:07:54 +0000 (15:07 +0000)]
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.
Thomas Vander Stichele [Wed, 24 Aug 2005 14:08:58 +0000 (14:08 +0000)]
port audioresample to basetransform
Original commit message from CVS:
port audioresample to basetransform
Thomas Vander Stichele [Wed, 24 Aug 2005 13:32:52 +0000 (13:32 +0000)]
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
Jan Schmidt [Wed, 24 Aug 2005 11:56:08 +0000 (11:56 +0000)]
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
Wim Taymans [Wed, 24 Aug 2005 11:29:10 +0000 (11:29 +0000)]
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.
Andy Wingo [Wed, 24 Aug 2005 11:13:54 +0000 (11:13 +0000)]
whoops
Original commit message from CVS:
whoops
Andy Wingo [Wed, 24 Aug 2005 11:07:51 +0000 (11:07 +0000)]
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.
David Schleef [Tue, 23 Aug 2005 19:29:38 +0000 (19:29 +0000)]
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:
Wim Taymans [Tue, 23 Aug 2005 18:30:07 +0000 (18:30 +0000)]
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.
Thomas Vander Stichele [Tue, 23 Aug 2005 18:19:10 +0000 (18:19 +0000)]
style fixes
Original commit message from CVS:
style fixes
Andy Wingo [Tue, 23 Aug 2005 13:29:17 +0000 (13:29 +0000)]
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.
Andy Wingo [Mon, 22 Aug 2005 16:50:59 +0000 (16:50 +0000)]
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.
Andy Wingo [Mon, 22 Aug 2005 15:11:31 +0000 (15:11 +0000)]
gst-libs/gst/audio/gstaudiosrc.*: Implement open_device and close_device in the ring buffer, like gstaudiosink.
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.
Thomas Vander Stichele [Mon, 22 Aug 2005 09:35:57 +0000 (09:35 +0000)]
whitespace cleanup
Original commit message from CVS:
whitespace cleanup
Thomas Vander Stichele [Mon, 22 Aug 2005 09:27:14 +0000 (09:27 +0000)]
remove filter.func
Original commit message from CVS:
remove filter.func
Thomas Vander Stichele [Sun, 21 Aug 2005 17:44:09 +0000 (17:44 +0000)]
make sure registry is built properly
Original commit message from CVS:
make sure registry is built properly
Thomas Vander Stichele [Sun, 21 Aug 2005 10:43:45 +0000 (10:43 +0000)]
use the setup/teardown methods to save code. save code is good.
Original commit message from CVS:
use the setup/teardown methods to save code. save code is good.
Thomas Vander Stichele [Sat, 20 Aug 2005 20:55:58 +0000 (20:55 +0000)]
only build if you have check
Original commit message from CVS:
only build if you have check
Thomas Vander Stichele [Sat, 20 Aug 2005 20:40:25 +0000 (20:40 +0000)]
yay, fix a segfault/security issue in vorbisdec gst-launch fakesrc ! vorbisdec wasn't happy add a test for vorbisdec
Original commit message from CVS:
yay, fix a segfault/security issue in vorbisdec
gst-launch fakesrc ! vorbisdec wasn't happy
add a test for vorbisdec
Thomas Vander Stichele [Sat, 20 Aug 2005 18:07:10 +0000 (18:07 +0000)]
add tests to gst-plugins-base add a volume element test clean up volume a little more for basetransform
Original commit message from CVS:
add tests to gst-plugins-base
add a volume element test
clean up volume a little more for basetransform
Andy Wingo [Fri, 19 Aug 2005 16:13:54 +0000 (16:13 +0000)]
ext/alsa/: Port to 0.9.
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.
Christian Schaller [Fri, 19 Aug 2005 14:41:46 +0000 (14:41 +0000)]
fix up spec to keep it working
Original commit message from CVS:
fix up spec to keep it working
Wim Taymans [Thu, 18 Aug 2005 10:23:54 +0000 (10:23 +0000)]
gst-libs/gst/rtp/gstbasertpdepayload.*: Fix for RTPBuffer changes.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_add_to_queue),
(gst_base_rtp_depayload_push),
(gst_base_rtp_depayload_queue_release):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Fix for RTPBuffer changes.
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data),
(gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data),
(gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len),
(gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len),
(gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data),
(gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len),
(gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version),
(gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding),
(gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to),
(gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension),
(gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc),
(gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc),
(gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker),
(gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type),
(gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq),
(gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp),
(gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len),
(gst_rtpbuffer_get_payload):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Don't subclass GstBuffer but add methods and helper functions
to construct and manipulate RTP packets in regular GstBuffers.
Stefan Kost [Wed, 17 Aug 2005 21:07:21 +0000 (21:07 +0000)]
gst/sine/gstsinesrc.c: moved statement below switch
Original commit message from CVS:
* gst/sine/gstsinesrc.c: (gst_sinesrc_src_query):
moved statement below switch
* gst/volume/gstvolume.c: (gst_volume_class_init):
added debug ptr
Wim Taymans [Tue, 16 Aug 2005 15:53:59 +0000 (15:53 +0000)]
gst-libs/gst/audio/gstbaseaudiosrc.c: Open and close device in READY<->NULL state change.
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.
Andy Wingo [Tue, 16 Aug 2005 14:35:52 +0000 (14:35 +0000)]
examples/seeking/Makefile.am: Don't compile non-compiling compiled objects with the compiler.
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.
Thomas Vander Stichele [Mon, 15 Aug 2005 16:15:32 +0000 (16:15 +0000)]
mangled tmpl files
Original commit message from CVS:
mangled tmpl files
Thomas Vander Stichele [Mon, 15 Aug 2005 14:52:08 +0000 (14:52 +0000)]
add all plugin docs to the documentation
Original commit message from CVS:
add all plugin docs to the documentation
Thomas Vander Stichele [Mon, 15 Aug 2005 14:50:57 +0000 (14:50 +0000)]
begon
Original commit message from CVS:
begon
Thomas Vander Stichele [Mon, 15 Aug 2005 14:49:58 +0000 (14:49 +0000)]
renamed these to make it clearer what we're documenting
Original commit message from CVS:
renamed these to make it clearer what we're documenting
Thomas Vander Stichele [Mon, 15 Aug 2005 14:40:37 +0000 (14:40 +0000)]
order by element name
Original commit message from CVS:
order by element name
Thomas Vander Stichele [Mon, 15 Aug 2005 14:17:53 +0000 (14:17 +0000)]
first stab at outputting xml descriptions of elements and plugins for doc build
Original commit message from CVS:
first stab at outputting xml descriptions of elements and plugins for doc build
Thomas Vander Stichele [Mon, 15 Aug 2005 14:15:40 +0000 (14:15 +0000)]
rename plugin
Original commit message from CVS:
rename plugin
Philippe Kalaf [Fri, 12 Aug 2005 13:34:56 +0000 (13:34 +0000)]
gst-libs/gst/rtp/gstbasertpdepayload.*: Made a thread to release the queue.
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.
Philippe Kalaf [Wed, 10 Aug 2005 20:52:37 +0000 (20:52 +0000)]
gst-libs/gst/rtp/gstbasertpdepayload.*: Added rtp timestamp -> gst timestamp conversion.
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.
Tim-Philipp Müller [Tue, 9 Aug 2005 17:29:40 +0000 (17:29 +0000)]
gst-libs/gst/: Add padding (you will need to rebuild gst-plugins-base, gst-plugins and all applications afterwards!)
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/net/gstnetbuffer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/rtp/gstrtpbuffer.h:
Add padding (you will need to rebuild gst-plugins-base,
gst-plugins and all applications afterwards!)
Tim-Philipp Müller [Tue, 9 Aug 2005 16:59:21 +0000 (16:59 +0000)]
gst-libs/gst/riff/riff-read.c: Fix bug in debug message and add some more debug messages.
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.
Edward Hervey [Mon, 8 Aug 2005 16:58:29 +0000 (16:58 +0000)]
gst-libs/gst/riff/riff-media.c: backported updates since branch
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
backported updates since branch
Andy Wingo [Mon, 8 Aug 2005 16:42:10 +0000 (16:42 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c
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.
Tim-Philipp Müller [Mon, 8 Aug 2005 14:13:59 +0000 (14:13 +0000)]
gst-libs/gst/interfaces/mixer.h: Reset padding to GST_PADDING.
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.h:
Reset padding to GST_PADDING.
Ronald S. Bultje [Mon, 8 Aug 2005 12:16:54 +0000 (12:16 +0000)]
gst/playback/gstplaybin.c: Remove visualization from parent explicitely; works around some apparent refcount issue th...
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.
Ronald S. Bultje [Mon, 8 Aug 2005 10:16:34 +0000 (10:16 +0000)]
ext/alsa/gstalsasink.c: Assign debug category, add negotiation debug msgs.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams):
Assign debug category, add negotiation debug msgs.
Ronald S. Bultje [Sun, 7 Aug 2005 14:21:06 +0000 (14:21 +0000)]
ext/gnomevfs/gstgnomevfssrc.c: Fix error code for file-not-found to NOT_FOUND.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_start):
Fix error code for file-not-found to NOT_FOUND.
Thomas Vander Stichele [Fri, 5 Aug 2005 18:51:29 +0000 (18:51 +0000)]
renamed to actual element names, so much nicer to look at
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
Thomas Vander Stichele [Fri, 5 Aug 2005 17:13:10 +0000 (17:13 +0000)]
first stab at documenting elements
Original commit message from CVS:
first stab at documenting elements
Ronald S. Bultje [Fri, 5 Aug 2005 15:53:25 +0000 (15:53 +0000)]
gst/playback/gstplaybin.c: Enable videoscale.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_video_element):
Enable videoscale.
Ronald S. Bultje [Fri, 5 Aug 2005 15:33:19 +0000 (15:33 +0000)]
gst-libs/gst/gconf/gconf.*: Fix some Andy Problem [tm].
Original commit message from CVS:
* gst-libs/gst/gconf/gconf.c:
* gst-libs/gst/gconf/gconf.h:
Fix some Andy Problem [tm].
Andy Wingo [Thu, 4 Aug 2005 19:52:32 +0000 (19:52 +0000)]
gst/videoscale/gstvideoscale.c (gst_videoscale_get_size): gst/ffmpegcolorspace/gstffmpegcolorspace.c
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.
Ronald S. Bultje [Thu, 4 Aug 2005 17:32:22 +0000 (17:32 +0000)]
sys/ximage/ximagesink.c: Do not set new window sizes yet if we prepare a new buffer size for upstream renegotiation (...
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.
Andy Wingo [Wed, 3 Aug 2005 16:44:18 +0000 (16:44 +0000)]
sys/ximage/ximagesink.c (gst_ximagesink_renegotiate_size): Protect the height, width, and desired_caps with the pool_...
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.
Stefan Kost [Tue, 2 Aug 2005 12:19:32 +0000 (12:19 +0000)]
removed another obsolete example
Original commit message from CVS:
removed another obsolete example
Stefan Kost [Tue, 2 Aug 2005 11:53:19 +0000 (11:53 +0000)]
removed obsolete include
Original commit message from CVS:
removed obsolete include
Edward Hervey [Tue, 2 Aug 2005 11:51:03 +0000 (11:51 +0000)]
gst/volume/gstvolume.c: include left from controller cleanup
Original commit message from CVS:
* gst/volume/gstvolume.c:
include left from controller cleanup
Jan Schmidt [Tue, 2 Aug 2005 09:41:05 +0000 (09:41 +0000)]
ext/ogg/gstoggmux.c: Stop collectpads before calling the parent state change function on PAUSED->READY.
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.
Jan Schmidt [Mon, 1 Aug 2005 21:14:22 +0000 (21:14 +0000)]
configure.ac: When testing for X libs, use the X CFlags
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.
Stefan Kost [Mon, 1 Aug 2005 16:20:33 +0000 (16:20 +0000)]
deactivate and remove dparams (libgstcontrol)
Original commit message from CVS:
deactivate and remove dparams (libgstcontrol)
Wim Taymans [Fri, 29 Jul 2005 17:07:39 +0000 (17:07 +0000)]
gst/audioconvert/gstaudioconvert.c: Convert me to BaseTransform!! help..
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link_src):
Convert me to BaseTransform!! help..
Andy Wingo [Fri, 29 Jul 2005 15:42:17 +0000 (15:42 +0000)]
ext/alsa/gstalsaplugin.c (plugin_init): We are primary audio sinks.
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.
Tim-Philipp Müller [Thu, 28 Jul 2005 19:21:19 +0000 (19:21 +0000)]
ext/vorbis/vorbisdec.c: Fix confusing debug message (s/event/query/)
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
Fix confusing debug message (s/event/query/)
Tim-Philipp Müller [Thu, 28 Jul 2005 18:42:54 +0000 (18:42 +0000)]
gst/videotestsrc/videotestsrc.h: Use "_stdint.h" instead of <stdint.h>
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.h:
Use "_stdint.h" instead of <stdint.h>
Tim-Philipp Müller [Thu, 28 Jul 2005 12:08:00 +0000 (12:08 +0000)]
gst/videotestsrc/videotestsrc.h: Use "_stdint.h" instead of <stdint.h>
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.h:
Use "_stdint.h" instead of <stdint.h>
Wim Taymans [Wed, 27 Jul 2005 19:19:20 +0000 (19:19 +0000)]
ext/vorbis/Makefile.am: Revert wrong commit.
Original commit message from CVS:
* ext/vorbis/Makefile.am:
Revert wrong commit.
Wim Taymans [Wed, 27 Jul 2005 19:13:27 +0000 (19:13 +0000)]
gst-libs/gst/audio/gstbaseaudiosrc.c: More compilation fixen.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_event):
More compilation fixen.
Wim Taymans [Wed, 27 Jul 2005 19:10:20 +0000 (19:10 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c: Fix compilation.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render),
(gst_base_audio_sink_create_ringbuffer),
(gst_base_audio_sink_change_state):
Fix compilation.
Wim Taymans [Wed, 27 Jul 2005 18:34:29 +0000 (18:34 +0000)]
examples/seeking/seek.c: Update seek example.
Original commit message from CVS:
* examples/seeking/seek.c: (setup_dynamic_link),
(make_dv_pipeline), (make_vorbis_theora_pipeline), (query_rates),
(query_positions_elems), (query_positions_pads), (do_seek):
Update seek example.
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_event),
(gst_ogg_pad_typefind), (gst_ogg_demux_chain_elem_pad),
(gst_ogg_demux_queue_data), (gst_ogg_demux_chain_peer),
(gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page),
(gst_ogg_demux_handle_event),
(gst_ogg_demux_deactivate_current_chain),
(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_collect_chain_info), (gst_ogg_demux_collect_info),
(gst_ogg_demux_chain), (gst_ogg_demux_send_event),
(gst_ogg_demux_loop):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
* ext/theora/theoradec.c: (theora_dec_src_event),
(theora_dec_src_getcaps), (theora_dec_sink_event),
(theora_dec_push), (theora_dec_chain):
* ext/vorbis/Makefile.am:
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_event),
(vorbis_dec_sink_event), (vorbis_dec_push),
(vorbis_handle_data_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_sink_event),
(gst_vorbisenc_chain):
* gst/playback/gststreaminfo.c: (cb_probe):
* gst/subparse/gstsubparse.c: (gst_subparse_src_event):
* gst/videorate/gstvideorate.c: (gst_videorate_event):
* gst/videoscale/gstvideoscale.c:
(gst_videoscale_handle_src_event):
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_event):
* sys/ximage/ximagesink.c: (gst_ximagesink_show_frame),
(gst_ximagesink_navigation_send_event):
* sys/xvimage/xvimagesink.c:
(gst_xvimagesink_navigation_send_event):
Various event updates and cleanups
Ronald S. Bultje [Wed, 27 Jul 2005 13:24:02 +0000 (13:24 +0000)]
gst/videoscale/gstvideoscale.c: Fix segfault for I420/YV12.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_videoscale_prepare_images):
Fix segfault for I420/YV12.
Ronald S. Bultje [Mon, 25 Jul 2005 15:31:51 +0000 (15:31 +0000)]
ext/vorbis/vorbisdec.c: Report bitrate.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
Report bitrate.
Thomas Vander Stichele [Mon, 25 Jul 2005 14:06:15 +0000 (14:06 +0000)]
forward port from 0.9 and enable videoflip now that it works
Original commit message from CVS:
forward port from 0.9 and enable videoflip now that it works
Ronald S. Bultje [Mon, 25 Jul 2005 09:21:01 +0000 (09:21 +0000)]
gst/playback/gstplaybin.c: Switch to auto*sink elements as default sinks; add volume element so that volume control i...
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.
Wim Taymans [Thu, 21 Jul 2005 20:01:46 +0000 (20:01 +0000)]
gst/playback/: Refcount fix and more comments.
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.
Zaheer Abbas Merali [Thu, 21 Jul 2005 18:07:38 +0000 (18:07 +0000)]
sys/ximage/: Prepare for adding ximagesrc, rename of plugin to ximage etc.
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.
Wim Taymans [Thu, 21 Jul 2005 17:25:40 +0000 (17:25 +0000)]
ext/ogg/gstoggdemux.c: Generate correct disconts for live chained oggs.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_event),
(gst_ogg_pad_internal_chain), (gst_ogg_pad_typefind),
(gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
(gst_ogg_pad_submit_page), (gst_ogg_chain_new),
(gst_ogg_demux_init), (gst_ogg_demux_activate_chain),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_chain_info),
(gst_ogg_demux_collect_info), (gst_ogg_demux_chain),
(gst_ogg_demux_send_event), (gst_ogg_demux_loop):
Generate correct disconts for live chained oggs.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render),
(gst_base_audio_sink_create_ringbuffer),
(gst_base_audio_sink_change_state):
Handle discont math correctly.
* gst/playback/gstplaybin.c: (add_sink):
Some small debug cleanup.
Wim Taymans [Thu, 21 Jul 2005 12:17:37 +0000 (12:17 +0000)]
ext/ogg/gstoggdemux.c: Reorganize code to send the right disconts when in streaming mode.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_init), (gst_ogg_pad_event),
(gst_ogg_pad_internal_chain), (gst_ogg_pad_typefind),
(gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
(gst_ogg_pad_submit_page), (gst_ogg_chain_new),
(gst_ogg_demux_init), (gst_ogg_demux_deactivate_current_chain),
(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_read_chain), (gst_ogg_demux_find_pad),
(gst_ogg_demux_collect_chain_info), (gst_ogg_demux_collect_info),
(gst_ogg_demux_find_chains), (gst_ogg_demux_chain),
(gst_ogg_demux_send_event), (gst_ogg_demux_loop),
(gst_ogg_demux_change_state), (gst_ogg_print):
Reorganize code to send the right disconts when in streaming
mode.
Christian Schaller [Thu, 21 Jul 2005 10:22:23 +0000 (10:22 +0000)]
remove gconf stuff from spec file
Original commit message from CVS:
remove gconf stuff from spec file
Andy Wingo [Wed, 20 Jul 2005 17:30:45 +0000 (17:30 +0000)]
gst/videoscale/vs_image.c (vs_image_scale_nearest_YUYV): Typo ixes a seggie mcfalterson (#310894).
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).
projects / platform / upstream / gst-plugins-base.git / log