Commit graph

3594 commits

Author SHA1 Message Date
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
Andy Wingo 13b122a106 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.
2005-08-22 15:11:31 +00:00
Thomas Vander Stichele 6d828d3b93 whitespace cleanup
Original commit message from CVS:
whitespace cleanup
2005-08-22 09:35:57 +00:00
Thomas Vander Stichele 9eb7c68f13 remove filter.func
Original commit message from CVS:
remove filter.func
2005-08-22 09:27:14 +00:00
Thomas Vander Stichele 6271b0c200 make sure registry is built properly
Original commit message from CVS:
make sure registry is built properly
2005-08-21 17:44:09 +00:00
Thomas Vander Stichele 2789040516 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.
2005-08-21 10:43:45 +00:00
Thomas Vander Stichele f8ad8cc4f6 only build if you have check
Original commit message from CVS:
only build if you have check
2005-08-20 20:55:58 +00:00
Thomas Vander Stichele 585493a9dd 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
2005-08-20 20:40:25 +00:00
Thomas Vander Stichele 7da31ee8b4 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
2005-08-20 18:07:10 +00:00
Andy Wingo b05796c9d9 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.
2005-08-19 16:13:54 +00:00
Christian Schaller 256c7c115e fix up spec to keep it working
Original commit message from CVS:
fix up spec to keep it working
2005-08-19 14:41:46 +00:00
Wim Taymans 7667a989d3 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.
2005-08-18 10:23:54 +00:00
Stefan Kost b5f1cf664d 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
2005-08-17 21:07:21 +00:00
Wim Taymans 4e3b19e5fb 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.
2005-08-16 15:53:59 +00:00
Andy Wingo 107da3e93c 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.
2005-08-16 14:35:52 +00:00
Thomas Vander Stichele 536acc96d9 mangled tmpl files
Original commit message from CVS:
mangled tmpl files
2005-08-15 16:15:32 +00:00
Thomas Vander Stichele e60ded44d7 add all plugin docs to the documentation
Original commit message from CVS:
add all plugin docs to the documentation
2005-08-15 14:52:08 +00:00
Thomas Vander Stichele 906b53b7dd begon
Original commit message from CVS:
begon
2005-08-15 14:50:57 +00:00
Thomas Vander Stichele 5b0a756a32 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
2005-08-15 14:49:58 +00:00
Thomas Vander Stichele feec34350b order by element name
Original commit message from CVS:
order by element name
2005-08-15 14:40:37 +00:00
Thomas Vander Stichele 1f40aa1676 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
2005-08-15 14:17:53 +00:00
Thomas Vander Stichele 3e589b0977 rename plugin
Original commit message from CVS:
rename plugin
2005-08-15 14:15:40 +00:00
Philippe Kalaf 96a0b1b9b9 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.
2005-08-12 13:34:56 +00:00
Philippe Kalaf b50d3fe5f6 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.
2005-08-10 20:52:37 +00:00
Tim-Philipp Müller b9b56ce7d3 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!)
2005-08-09 17:29:40 +00:00
Tim-Philipp Müller 822e77203a 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.
2005-08-09 16:59:21 +00:00
Edward Hervey b060089ac9 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
2005-08-08 16:58:29 +00:00
Andy Wingo 69d36f02ce 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.
2005-08-08 16:42:10 +00:00
Tim-Philipp Müller 074b6a3b64 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.
2005-08-08 14:13:59 +00:00