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1860 commits

Author SHA1 Message Date
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
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 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 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
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 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 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
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
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
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
Ronald S. Bultje c3c1f232c0 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.
2005-08-08 12:16:54 +00:00
Ronald S. Bultje be08afaabf 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.
2005-08-08 10:16:34 +00:00
Ronald S. Bultje ba32f38042 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.
2005-08-07 14:21:06 +00:00
Thomas Vander Stichele e571f069d1 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
2005-08-05 18:51:29 +00:00
Thomas Vander Stichele 29569738a6 first stab at documenting elements
Original commit message from CVS:
first stab at documenting elements
2005-08-05 17:13:10 +00:00
Ronald S. Bultje 70f589da21 gst/playback/gstplaybin.c: Enable videoscale.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_video_element):
Enable videoscale.
2005-08-05 15:53:25 +00:00
Ronald S. Bultje 6550429258 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].
2005-08-05 15:33:19 +00:00
Andy Wingo 7d10d86635 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.
2005-08-04 19:52:32 +00:00
Ronald S. Bultje 993a705188 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.
2005-08-04 17:32:22 +00:00
Andy Wingo 306ae5611c 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.
2005-08-03 16:44:18 +00:00
Edward Hervey a5af11190d gst/volume/gstvolume.c: include left from controller cleanup
Original commit message from CVS:
* gst/volume/gstvolume.c:
include left from controller cleanup
2005-08-02 11:51:03 +00:00
Jan Schmidt 49fc826134 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.
2005-08-02 09:41:05 +00:00
Jan Schmidt b34c0e7e53 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.
2005-08-01 21:14:22 +00:00
Stefan Kost 497b586076 deactivate and remove dparams (libgstcontrol)
Original commit message from CVS:
deactivate and remove dparams (libgstcontrol)
2005-08-01 16:20:33 +00:00
Wim Taymans a6d89f51fa 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..
2005-07-29 17:07:39 +00:00
Andy Wingo 708deec535 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.
2005-07-29 15:42:17 +00:00
Tim-Philipp Müller 7679fe45fb 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/)
2005-07-28 19:21:19 +00:00
Tim-Philipp Müller d9c8b9915c 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>
2005-07-28 12:08:00 +00:00