Edgard Lima [Tue, 20 Sep 2005 18:49:05 +0000 (18:49 +0000)]
Fixed configure.ac and ext/sdl/Makefile.am for sdl port to 0.9.
Original commit message from CVS:
Fixed configure.ac and ext/sdl/Makefile.am for sdl port to 0.9.
Edgard Lima [Mon, 19 Sep 2005 21:51:02 +0000 (21:51 +0000)]
sdlvideosink ported to 0.9 and tested with filesrc ! mpeg2dec
Original commit message from CVS:
sdlvideosink ported to 0.9 and tested with filesrc ! mpeg2dec
Edgard Lima [Mon, 19 Sep 2005 21:47:54 +0000 (21:47 +0000)]
sdlvideosink ported to 0.9 and tested with filesrc ! mpeg2dec
Original commit message from CVS:
sdlvideosink ported to 0.9 and tested with filesrc ! mpeg2dec
Thomas Vander Stichele [Mon, 19 Sep 2005 16:44:37 +0000 (16:44 +0000)]
add check-valgrind target
Original commit message from CVS:
add check-valgrind target
Jan Schmidt [Fri, 9 Sep 2005 17:53:47 +0000 (17:53 +0000)]
check/: Add extra tests for basetransform based components.
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.
Flavio Oliveira [Thu, 8 Sep 2005 15:56:38 +0000 (15:56 +0000)]
Ported speed Plugin to GStreamer 0.9
Original commit message from CVS:
Ported speed Plugin to GStreamer 0.9
Thomas Vander Stichele [Tue, 6 Sep 2005 14:06:55 +0000 (14:06 +0000)]
back to HEAD
Original commit message from CVS:
back to HEAD
Thomas Vander Stichele [Tue, 6 Sep 2005 14:05:50 +0000 (14:05 +0000)]
releasing 0.9.1
Original commit message from CVS:
releasing 0.9.1
Jan Schmidt [Mon, 5 Sep 2005 17:20:29 +0000 (17:20 +0000)]
Fix up all the state change functions.
Original commit message from CVS:
Fix up all the state change functions.
Thomas Vander Stichele [Mon, 5 Sep 2005 15:10:18 +0000 (15:10 +0000)]
cleaning up bad
Original commit message from CVS:
cleaning up bad
Thomas Vander Stichele [Mon, 5 Sep 2005 13:49:41 +0000 (13:49 +0000)]
created gst-plugins-bad
Original commit message from CVS:
created gst-plugins-bad
Thomas Vander Stichele [Sun, 4 Sep 2005 10:38:45 +0000 (10:38 +0000)]
fix distcheck
Original commit message from CVS:
* common/gtk-doc-plugins.mak:
* docs/plugins/Makefile.am:
fix distcheck
* gst/audioresample/resample.c:
fix wrong docstring
Flavio Oliveira [Fri, 2 Sep 2005 20:21:17 +0000 (20:21 +0000)]
Ported GSM Encoder to GStreamer 0.9
Original commit message from CVS:
Ported GSM Encoder to GStreamer 0.9
Andy Wingo [Fri, 2 Sep 2005 15:44:44 +0000 (15:44 +0000)]
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.
Andy Wingo [Fri, 2 Sep 2005 15:43:14 +0000 (15:43 +0000)]
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.
Flavio Oliveira [Thu, 1 Sep 2005 19:53:12 +0000 (19:53 +0000)]
update PORTED_09 file
Original commit message from CVS:
update PORTED_09 file
Flavio Oliveira [Thu, 1 Sep 2005 19:12:44 +0000 (19:12 +0000)]
Faac (AAC Encoder) ported for GStreamer 0.9
Original commit message from CVS:
Faac (AAC Encoder) ported for GStreamer 0.9
Thomas Vander Stichele [Thu, 1 Sep 2005 16:58:00 +0000 (16:58 +0000)]
remove libdir
Original commit message from CVS:
remove libdir
Thomas Vander Stichele [Wed, 31 Aug 2005 18:07:29 +0000 (18:07 +0000)]
all these plugins are moved to gst-plugins-ugly
Original commit message from CVS:
all these plugins are moved to gst-plugins-ugly
Michael Smith [Wed, 31 Aug 2005 11:53:35 +0000 (11:53 +0000)]
Port LPCM decoder to 0.9
Original commit message from CVS:
Port LPCM decoder to 0.9
Jan Schmidt [Tue, 30 Aug 2005 20:51:28 +0000 (20:51 +0000)]
configure.ac: Remove plugins that should have disappeared.
Original commit message from CVS:
* configure.ac:
Remove plugins that should have disappeared.
Jan Schmidt [Tue, 30 Aug 2005 20:42:39 +0000 (20:42 +0000)]
Make autogen work again.
Original commit message from CVS:
* autogen.sh:
* configure.ac:
Make autogen work again.
Wim Taymans [Tue, 30 Aug 2005 19:54:35 +0000 (19:54 +0000)]
check/elements/: Fix checks.
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.
Thomas Vander Stichele [Tue, 30 Aug 2005 18:06:52 +0000 (18:06 +0000)]
remove stuff
Original commit message from CVS:
remove stuff
Thomas Vander Stichele [Tue, 30 Aug 2005 18:00:06 +0000 (18:00 +0000)]
all these plugins are moved to gst-plugins-good
Original commit message from CVS:
all these plugins are moved to gst-plugins-good
Flavio Oliveira [Tue, 30 Aug 2005 12:59:30 +0000 (12:59 +0000)]
Ported to GStreamer 0.9. Need to fix performance issues.
Original commit message from CVS:
Ported to GStreamer 0.9. Need to fix performance issues.
Andy Wingo [Sun, 28 Aug 2005 17:59:20 +0000 (17:59 +0000)]
Updates for two-arg init from GST_BOILERPLATE.
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.
Andy Wingo [Sun, 28 Aug 2005 17:52:45 +0000 (17:52 +0000)]
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.
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
Andy Wingo [Fri, 26 Aug 2005 11:05:13 +0000 (11:05 +0000)]
ext/ladspa/gstladspa.*: Finish porting, still doesn't work but it does 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.
Thomas Vander Stichele [Fri, 26 Aug 2005 00:42:57 +0000 (00:42 +0000)]
do proper cleanup/creation, fixes state changes
Original commit message from CVS:
do proper cleanup/creation, fixes state changes
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 10:53:49 +0000 (10:53 +0000)]
gst/level/gstlevel.c: Revert unpopular change for GST_MESSAGE_SRC to GObject.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_message_new):
Revert unpopular change for GST_MESSAGE_SRC to GObject.
Andy Wingo [Thu, 25 Aug 2005 09:32:22 +0000 (09:32 +0000)]
add missing files
Original commit message from CVS:
add missing files
Andy Wingo [Thu, 25 Aug 2005 09:30:24 +0000 (09:30 +0000)]
ext/ladspa/gstladspa.*: Halfway-ported. Doesn't compile yet.
Original commit message from CVS:
2005-08-25 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstladspa.h:
* ext/ladspa/gstladspa.c: Halfway-ported. Doesn't compile yet.
* ext/ladspa/gstsignalprocessor.h:
* ext/ladspa/gstsignalprocessor.c: New files, the start of a base
class for DSP elements.
* configure.ac: Sort the external libs checks, add a ladspa check,
output the ladspa makefile.
Owen Fraser-Green [Thu, 25 Aug 2005 00:32:21 +0000 (00:32 +0000)]
Fixed EOS and improved robustness for malformed indices.
Original commit message from CVS:
Fixed EOS and improved robustness for malformed indices.
Christian Schaller [Wed, 24 Aug 2005 17:40:17 +0000 (17:40 +0000)]
add lame
Original commit message from CVS:
add lame
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
Andy Wingo [Wed, 24 Aug 2005 15:51:48 +0000 (15:51 +0000)]
ext/dv/gstdvdemux.c (gst_dvdemux_demux_frame): Send out valid segment end timestamps.
Original commit message from CVS:
2005-08-24 Andy Wingo <wingo@pobox.com>
* ext/dv/gstdvdemux.c (gst_dvdemux_demux_frame): Send out valid
segment end timestamps.
(Also commit an old changelog entry)
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 13:34:05 +0000 (13:34 +0000)]
enable more; update for basetransform
Original commit message from CVS:
enable more; update for basetransform
Jan Schmidt [Wed, 24 Aug 2005 11:58:59 +0000 (11:58 +0000)]
gst/level/gstlevel.c: GST_MESSAGE_SRC became a GObject
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_message_new):
GST_MESSAGE_SRC became a GObject
Christian Schaller [Wed, 24 Aug 2005 11:42:58 +0000 (11:42 +0000)]
fix speex entry
Original commit message from CVS:
fix speex entry
Christian Schaller [Wed, 24 Aug 2005 10:04:35 +0000 (10:04 +0000)]
add speex to spec file and remove gstosslement from POTFILES.in
Original commit message from CVS:
add speex to spec file and remove gstosslement from POTFILES.in
Stefan Kost [Tue, 23 Aug 2005 19:40:11 +0000 (19:40 +0000)]
ext/speex/gstspeexenc.h: Fixed include path of adapter
Original commit message from CVS:
* ext/speex/gstspeexenc.h:
Fixed include path of adapter
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 18:24:36 +0000 (18:24 +0000)]
ext/speex/: Fix property warning.
Original commit message from CVS:
* ext/speex/gstspeexdec.c: (gst_speex_dec_class_init):
* ext/speex/gstspeexenc.c: (gst_speexenc_class_init):
Fix property warning.
Wim Taymans [Tue, 23 Aug 2005 16:54:38 +0000 (16:54 +0000)]
gst/rtp/: Small updates, RFC reference to payload encoders.
Original commit message from CVS:
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
(gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain):
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain):
* gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init),
(gst_rtph263penc_flush), (gst_rtph263penc_chain):
Small updates, RFC reference to payload encoders.
Ronald S. Bultje [Tue, 23 Aug 2005 15:14:01 +0000 (15:14 +0000)]
Port speexdec. Leads to some unfamiliar warnings on console, but works otherwise.
Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/speex/Makefile.am:
* ext/speex/gstspeex.c: (plugin_init):
* ext/speex/gstspeexdec.c: (speex_get_query_types),
(gst_speex_dec_init), (speex_dec_src_query), (speex_dec_src_event),
(speex_dec_event), (speex_dec_chain):
Port speexdec. Leads to some unfamiliar warnings on console,
but works otherwise.
Andy Wingo [Tue, 23 Aug 2005 14:10:04 +0000 (14:10 +0000)]
sys/oss/gstosssrc.c (gst_oss_src_open): Set the device-name property after opening the mixer.
Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* sys/oss/gstosssrc.c (gst_oss_src_open): Set the device-name
property after opening the mixer.
Andy Wingo [Tue, 23 Aug 2005 13:59:22 +0000 (13:59 +0000)]
sys/oss/gstosssrc.*: Easy to implement a mixer, eh...
Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* sys/oss/gstosssrc.c:
* sys/oss/gstosssrc.h: Easy to implement a mixer, eh...
Andy Wingo [Tue, 23 Aug 2005 13:53:17 +0000 (13:53 +0000)]
sys/oss/gstossmixerelement.*: Added mixer element like alsamixer.
Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* sys/oss/gstossmixerelement.h:
* sys/oss/gstossmixerelement.c: Added mixer element like
alsamixer.
* sys/oss/Makefile.am:
* sys/oss/gstossaudio.c: Register the ossmixer element.
Andy Wingo [Tue, 23 Aug 2005 13:27:30 +0000 (13:27 +0000)]
changelog
Original commit message from CVS:
changelog
Andy Wingo [Tue, 23 Aug 2005 09:46:29 +0000 (09:46 +0000)]
sys/oss/gstosssrc.*: Totally ported, dude.
Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* sys/oss/gstosssrc.h:
* sys/oss/gstosssrc.c: Totally ported, dude.
* sys/oss/Makefile.am:
* sys/oss/gstossaudio.c: Add osssrc.
* sys/oss/gstosssink.c: We do native byte order.
Owen Fraser-Green [Tue, 23 Aug 2005 07:54:31 +0000 (07:54 +0000)]
Fixed mishandling events and incorrect audio skipping after seek.
Original commit message from CVS:
Fixed mishandling events and incorrect audio skipping after seek.
Jan Schmidt [Mon, 22 Aug 2005 19:58:57 +0000 (19:58 +0000)]
ext/mad/gstid3tag.c: Works a bit better now, but still needs a rewrite to use get_range instead of this seeking nasti...
Original commit message from CVS:
* ext/mad/gstid3tag.c: (gst_id3_tag_init),
(gst_id3_tag_sink_event), (gst_id3_tag_do_caps_nego),
(gst_id3_tag_chain), (gst_id3_tag_change_state), (plugin_init):
Works a bit better now, but still needs a rewrite to use
get_range instead of this seeking nastiness.
Ronald S. Bultje [Mon, 22 Aug 2005 11:20:18 +0000 (11:20 +0000)]
Port flacdec (seeking is still slow'ish).
Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/flac/Makefile.am:
* ext/flac/gstflac.c: (plugin_init):
* ext/flac/gstflacdec.c: (flacdec_get_type), (gst_flacdec_init),
(gst_flacdec_update_metadata), (gst_flacdec_seek),
(gst_flacdec_tell), (gst_flacdec_length), (gst_flacdec_read),
(gst_flacdec_write), (gst_flacdec_loop),
(gst_flacdec_get_src_query_types), (gst_flacdec_src_query),
(gst_flacdec_src_event), (gst_flacdec_sink_activate),
(gst_flacdec_sink_activate_pull), (gst_flacdec_change_state):
* ext/flac/gstflacdec.h:
Port flacdec (seeking is still slow'ish).
Owen Fraser-Green [Mon, 22 Aug 2005 07:32:59 +0000 (07:32 +0000)]
Fixed some seeking issues
Original commit message from CVS:
Fixed some seeking issues
Christian Schaller [Fri, 19 Aug 2005 16:25:07 +0000 (16:25 +0000)]
add mpegaudioparse to spec file
Original commit message from CVS:
add mpegaudioparse to spec file
Wim Taymans [Fri, 19 Aug 2005 16:22:41 +0000 (16:22 +0000)]
gst/udp/gstmultiudpsink.c: Remove get_time code that is both wrong and unneeded.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init):
Remove get_time code that is both wrong and unneeded.
Wim Taymans [Fri, 19 Aug 2005 15:01:05 +0000 (15:01 +0000)]
gst/rtp/gstrtph263penc.*: Added configurable pt and ssrc, to be merged in the caps or a base class...
Original commit message from CVS:
* gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init),
(gst_rtph263penc_flush), (gst_rtph263penc_chain),
(gst_rtph263penc_set_property), (gst_rtph263penc_get_property):
* gst/rtp/gstrtph263penc.h:
Added configurable pt and ssrc, to be merged in the caps or
a base class...
Wim Taymans [Fri, 19 Aug 2005 14:51:20 +0000 (14:51 +0000)]
gst/rtp/: Some cleanups in the h263p (de)payloaders.
Original commit message from CVS:
* gst/rtp/gstrtph263pdec.c: (gst_rtph263pdec_init),
(gst_rtph263pdec_chain):
* gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init),
(gst_rtph263penc_flush), (gst_rtph263penc_chain):
Some cleanups in the h263p (de)payloaders.
Christian Schaller [Fri, 19 Aug 2005 14:49:11 +0000 (14:49 +0000)]
add mpegaudioparse to configure.ac
Original commit message from CVS:
add mpegaudioparse to configure.ac
Christian Schaller [Fri, 19 Aug 2005 14:44:09 +0000 (14:44 +0000)]
latest makefile and spec file fixes
Original commit message from CVS:
latest makefile and spec file fixes
Wim Taymans [Fri, 19 Aug 2005 12:44:32 +0000 (12:44 +0000)]
ext/amrnb/: Update caps with audio/AMR.
Original commit message from CVS:
* ext/amrnb/amrnbdec.c:
* ext/amrnb/amrnbenc.c: (gst_amrnbenc_setcaps):
* ext/amrnb/amrnbparse.c:
Update caps with audio/AMR.
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
(gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain),
(gst_rtpamrdec_change_state):
* gst/rtp/gstrtpamrdec.h:
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain):
Dont set FT headers twice, it was already in the encoded
bitstream.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play):
* gst/rtsp/rtspconnection.c: (parse_line):
Cleanups
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_create), (gst_udpsrc_set_property),
(gst_udpsrc_get_property):
* gst/udp/gstudpsrc.h:
Added caps property, we need this soon to type the buffers.
Wim Taymans [Thu, 18 Aug 2005 20:14:44 +0000 (20:14 +0000)]
gst/rtp/gstrtpamrdec.c: Fix up amr depayloader a bit.
Original commit message from CVS:
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
(gst_rtpamrdec_chain):
Fix up amr depayloader a bit.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play):
Look for options result in Public and Allow header fields..
spec says Allow but some servers return Public...
Wim Taymans [Thu, 18 Aug 2005 19:47:38 +0000 (19:47 +0000)]
gst/rtp/gstrtpamrenc.*: Added payload_type and ssrc properties to the payloader.
Original commit message from CVS:
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain),
(gst_rtpamrenc_set_property), (gst_rtpamrenc_get_property):
* gst/rtp/gstrtpamrenc.h:
Added payload_type and ssrc properties to the payloader.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play):
Options need to be stripped and are in the Public header field.
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
Fix url / parsing...
Wim Taymans [Thu, 18 Aug 2005 17:39:10 +0000 (17:39 +0000)]
gst/rtp/: Added very simplistic amr payloader. depayloader does not work yet.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_get_type),
(gst_rtpamrdec_base_init), (gst_rtpamrdec_class_init),
(gst_rtpamrdec_init), (gst_rtpamrdec_chain),
(gst_rtpamrdec_set_property), (gst_rtpamrdec_get_property),
(gst_rtpamrdec_change_state), (gst_rtpamrdec_plugin_init):
* gst/rtp/gstrtpamrdec.h:
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_get_type),
(gst_rtpamrenc_base_init), (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain),
(gst_rtpamrenc_set_property), (gst_rtpamrenc_get_property),
(gst_rtpamrenc_change_state), (gst_rtpamrenc_plugin_init):
* gst/rtp/gstrtpamrenc.h:
* gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_class_init),
(gst_rtpmpaenc_flush), (gst_rtpmpaenc_chain):
Added very simplistic amr payloader. depayloader does not
work yet.
Wim Taymans [Thu, 18 Aug 2005 16:53:12 +0000 (16:53 +0000)]
gst/rtsp/: Handle RTSP defaults better.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_method_as_text), (rtsp_find_method):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_parse):
Handle RTSP defaults better.
Issue OPTIONS request to figure out what we are allowed to do.
Make the methods a bitfield so we can easily collect supported
options.
Fix rtsp_find_method.
Do proper RTSP connection shutdown.
Wim Taymans [Thu, 18 Aug 2005 10:33:13 +0000 (10:33 +0000)]
gst/rtp/: Remove old code that is now in gst-libs/gst/rtp/.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp-common.h:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpL16enc.h:
* gst/rtp/gstrtpdec.c: (gst_rtpdec_get_type),
(gst_rtpdec_class_init), (gst_rtpdec_chain_rtp),
(gst_rtpdec_chain_rtcp), (gst_rtpdec_change_state),
(gst_rtpdec_plugin_init):
* gst/rtp/gstrtph263pdec.c: (gst_rtph263pdec_get_type),
(gst_rtph263pdec_base_init), (gst_rtph263pdec_class_init),
(gst_rtph263pdec_init), (gst_rtph263pdec_chain),
(gst_rtph263pdec_set_property), (gst_rtph263pdec_get_property),
(gst_rtph263pdec_change_state), (gst_rtph263pdec_plugin_init):
* gst/rtp/gstrtph263pdec.h:
* gst/rtp/gstrtph263penc.c: (gst_rtph263penc_get_type),
(gst_rtph263penc_base_init), (gst_rtph263penc_class_init),
(gst_rtph263penc_init), (gst_rtph263penc_flush),
(gst_rtph263penc_chain), (gst_rtph263penc_set_property),
(gst_rtph263penc_get_property), (gst_rtph263penc_change_state),
(gst_rtph263penc_plugin_init):
* gst/rtp/gstrtph263penc.h:
* gst/rtp/gstrtpmpadec.c: (gst_rtpmpadec_get_type),
(gst_rtpmpadec_base_init), (gst_rtpmpadec_class_init),
(gst_rtpmpadec_init), (gst_rtpmpadec_chain),
(gst_rtpmpadec_set_property), (gst_rtpmpadec_get_property),
(gst_rtpmpadec_change_state), (gst_rtpmpadec_plugin_init):
* gst/rtp/gstrtpmpadec.h:
* gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_get_type),
(gst_rtpmpaenc_base_init), (gst_rtpmpaenc_class_init),
(gst_rtpmpaenc_init), (gst_rtpmpaenc_flush), (gst_rtpmpaenc_chain),
(gst_rtpmpaenc_set_property), (gst_rtpmpaenc_get_property),
(gst_rtpmpaenc_change_state), (gst_rtpmpaenc_plugin_init):
* gst/rtp/gstrtpmpaenc.h:
* gst/rtp/rtp-packet.c:
* gst/rtp/rtp-packet.h:
Remove old code that is now in gst-libs/gst/rtp/.
Added some payload/depayloaders.
* gst/udp/gstudpsink.c: (gst_udpsink_class_init):
Fix port number range.
Wim Taymans [Wed, 17 Aug 2005 19:05:51 +0000 (19:05 +0000)]
configure.ac: Added mpegaudioparse
Original commit message from CVS:
* configure.ac:
Added mpegaudioparse
* ext/lame/gstlame.c: (gst_lame_src_getcaps),
(gst_lame_src_setcaps), (gst_lame_sink_setcaps),
(gst_lame_sink_event), (gst_lame_chain):
Some cleanups.
Fix memleak.
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_class_init), (gst_mp3parse_init),
(gst_mp3parse_chain), (gst_mp3parse_change_state):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Ported mpegaudioparse
Christian Schaller [Wed, 17 Aug 2005 11:30:19 +0000 (11:30 +0000)]
removing README from Makefile.am as its gone from CVS
Original commit message from CVS:
removing README from Makefile.am as its gone from CVS
Wim Taymans [Wed, 17 Aug 2005 10:30:55 +0000 (10:30 +0000)]
gst/rtsp/gstrtspsrc.c: Support absolute control urls too.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open), (gst_rtspsrc_play):
Support absolute control urls too.
Ronald S. Bultje [Tue, 16 Aug 2005 16:30:23 +0000 (16:30 +0000)]
gst/: Uncomment metadata and codec-name handling.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_stream_header):
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_add_stream), (qtdemux_parse_tree):
Uncomment metadata and codec-name handling.
Wim Taymans [Tue, 16 Aug 2005 16:12:14 +0000 (16:12 +0000)]
Fix compile warning.
Original commit message from CVS:
* configure.ac:
* ext/amrnb/amrnbparse.c: (gst_amrnbparse_read_header):
Fix compile warning.
* ext/lame/gstlame.c: (gst_lame_class_init),
(gst_lame_src_getcaps), (gst_lame_src_setcaps),
(gst_lame_sink_setcaps), (gst_lame_init), (gst_lame_sink_event),
(gst_lame_chain), (gst_lame_change_state):
* ext/lame/gstlame.h:
Port lame plugin
Andy Wingo [Tue, 16 Aug 2005 14:04:38 +0000 (14:04 +0000)]
ext/raw1394/gstdv1394src.c (gst_dv1394src_iso_receive): Note license info in the source code -- was only in the commi...
Original commit message from CVS:
2005-08-16 Andy Wingo <wingo@pobox.com>
* ext/raw1394/gstdv1394src.c (gst_dv1394src_iso_receive): Note
license info in the source code -- was only in the commit log
before.
* ext/dv/gstdvdec.h:
* ext/dv/gstdvdec.c: Only decodes systemstream=FALSE dv video --
old pipelines using dvdec should probably have a dvdemux first.
* ext/dv/gstdvdemux.h:
* ext/dv/gstdvdemux.c: Split out from dvdec, chunks the incoming
systemstream=TRUE data into frames, sets caps data, and spits out
PCM audio in addition to systemstream=FALSE video frames. Operates
in chain mode only for now; should make a getrange version as
well.
* ext/dv/gstdv.c: New file, registers the libgstdv plugin.
* ext/dv/Makefile.am: Library name changed to libgstdv. Split
dvdec into dvdemux and dvdec.
Thomas Vander Stichele [Tue, 16 Aug 2005 13:37:47 +0000 (13:37 +0000)]
remove seeking example, they're in gst-plugins-base
Original commit message from CVS:
remove seeking example, they're in gst-plugins-base
Ronald S. Bultje [Tue, 16 Aug 2005 09:54:10 +0000 (09:54 +0000)]
Handle _push() return values.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_event), (gst_faad_chain):
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
Handle _push() return values.
Ronald S. Bultje [Mon, 15 Aug 2005 15:58:49 +0000 (15:58 +0000)]
Fix debug.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_event):
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
Fix debug.
Ronald S. Bultje [Mon, 15 Aug 2005 13:44:50 +0000 (13:44 +0000)]
gst/qtdemux/qtdemux.c: Forwardport from 0.8 to implement RLE.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak),
(qtdemux_video_caps):
Forwardport from 0.8 to implement RLE.
Wim Taymans [Mon, 15 Aug 2005 13:29:40 +0000 (13:29 +0000)]
gst/rtsp/README: Added rtsp server implementation docs.
Original commit message from CVS:
* gst/rtsp/README:
Added rtsp server implementation docs.
Thomas Vander Stichele [Sun, 14 Aug 2005 16:24:49 +0000 (16:24 +0000)]
rename
Original commit message from CVS:
rename
Thomas Vander Stichele [Sun, 14 Aug 2005 16:21:00 +0000 (16:21 +0000)]
pound some sense in the colorspace elements
Original commit message from CVS:
pound some sense in the colorspace elements
Thomas Vander Stichele [Sun, 14 Aug 2005 16:10:55 +0000 (16:10 +0000)]
licensing, name and description changes
Original commit message from CVS:
licensing, name and description changes
Thomas Vander Stichele [Sat, 13 Aug 2005 15:30:32 +0000 (15:30 +0000)]
conform
Original commit message from CVS:
conform
Thomas Vander Stichele [Sat, 13 Aug 2005 15:06:56 +0000 (15:06 +0000)]
conform
Original commit message from CVS:
conform
Tim-Philipp Müller [Sat, 13 Aug 2005 11:58:02 +0000 (11:58 +0000)]
Use -lgstfoo-@GST_MAJORMINOR@ instead of -lgstfoo-0.9
Original commit message from CVS:
* ext/mad/Makefile.am:
* gst/avi/Makefile.am:
* gst/effectv/Makefile.am:
* gst/udp/Makefile.am:
* gst/wavparse/Makefile.am:
Use -lgstfoo-@GST_MAJORMINOR@ instead of -lgstfoo-0.9
Thomas Vander Stichele [Sat, 13 Aug 2005 11:43:57 +0000 (11:43 +0000)]
removed from HEAD
Original commit message from CVS:
removed from HEAD
Tim-Philipp Müller [Fri, 12 Aug 2005 19:33:30 +0000 (19:33 +0000)]
ext/jpeg/gstjpegdec.c: Fix decoding of pictures with certain uneven or unaligned widths where jpeglib needs more hori...
Original commit message from CVS:
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_decode_indirect),
(gst_jpeg_dec_decode_direct), (gst_jpeg_dec_chain):
Fix decoding of pictures with certain uneven or unaligned
widths where jpeglib needs more horizontal padding than our
I420 buffers provide, resulting in blocky artifacts at the
left side of the picture (#164176).
Also make use of our shiny new GST_ROUND_N() macros.
Tim-Philipp Müller [Thu, 11 Aug 2005 15:02:37 +0000 (15:02 +0000)]
ext/jpeg/gstjpegdec.*: Fix crashes/invalid memory access for pictures that have a height that is not a multiple of 16...
Original commit message from CVS:
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init), (gst_jpeg_dec_chain),
(gst_jpeg_dec_change_state):
* ext/jpeg/gstjpegdec.h:
Fix crashes/invalid memory access for pictures that have a height
that is not a multiple of 16 (or rather: v_samp_factor * DCTSIZE).
Also fix the state change function for downwards state changes
(need to chain up to parent before destroying our resources, to
make sure pads get deactivated and our chain function isn't
running and using those very same resources in another thread).
The jpeg line buffer only needs to be v_samp_factor*DCTSIZE lines
per plane, not picture_height lines; allocate that on the stack.
Tim-Philipp Müller [Wed, 10 Aug 2005 12:12:31 +0000 (12:12 +0000)]
gst/wavparse/gstwavparse.c: Add some fixes from 0.8 branch: allow 24/32bps songs and blockalign samples to the header...
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers),
(gst_wavparse_stream_data):
Add some fixes from 0.8 branch: allow 24/32bps songs and
blockalign samples to the header-specified size, if any
(#311070); error out on channels==0 or bitrate==0
(#309043, #304588).
Thomas Vander Stichele [Wed, 10 Aug 2005 10:54:02 +0000 (10:54 +0000)]
port fixes from 0.8 to level
Original commit message from CVS:
port fixes from 0.8 to level
Ronald S. Bultje [Wed, 10 Aug 2005 09:33:58 +0000 (09:33 +0000)]
ext/faad/gstfaad.c: Add debug category, remove Close() call that made it crash whenever reusing, renegotiating or any...
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_class_init), (gst_faad_setcaps):
Add debug category, remove Close() call that made it crash
whenever reusing, renegotiating or anything; Close() actually
free()s the handle and should only be called on READY->NULL.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
Actually set caps on buffer (in addition to pad), also.
Owen Fraser-Green [Wed, 10 Aug 2005 08:33:18 +0000 (08:33 +0000)]
Implemented push-pull and seeking in rmdemux
Original commit message from CVS:
Implemented push-pull and seeking in rmdemux
Ronald S. Bultje [Tue, 9 Aug 2005 14:43:48 +0000 (14:43 +0000)]
ext/faad/gstfaad.c: Sign/unsign mismatch.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_event):
Sign/unsign mismatch.
* configure.ac:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_class_init),
(gst_qtdemux_init), (gst_qtdemux_get_src_query_types),
(gst_qtdemux_handle_src_query), (gst_qtdemux_handle_src_event),
(plugin_init), (gst_qtdemux_handle_sink_event),
(gst_qtdemux_change_state), (gst_qtdemux_loop_header),
(qtdemux_sink_activate), (qtdemux_sink_activate_pull),
(gst_qtdemux_add_stream), (qtdemux_parse), (qtdemux_parse_tree),
(qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps):
* gst/qtdemux/qtdemux.h:
Half-assed port (hey, it works).
projects / platform / upstream / gst-plugins-bad.git / log