Tim-Philipp Müller [Sat, 6 Oct 2007 16:13:14 +0000 (16:13 +0000)]
gst/id3demux/: Port ID3 tag demuxer over to the new GstTagDemux in -base (now would be a good time to test re-importi...
Original commit message from CVS:
* gst/id3demux/gstid3demux.c:
* gst/id3demux/gstid3demux.h:
* gst/id3demux/id3tags.c:
* gst/id3demux/id3tags.h:
* gst/id3demux/id3v2frames.c:
Port ID3 tag demuxer over to the new GstTagDemux in -base
(now would be a good time to test re-importing your music
collection).
Tim-Philipp Müller [Sat, 6 Oct 2007 15:13:09 +0000 (15:13 +0000)]
gst/apetag/: Port APE tag demuxer over to the new GstTagDemux in -base.
Original commit message from CVS:
* gst/apetag/Makefile.am:
* gst/apetag/gstapedemux.c:
* gst/apetag/gstapedemux.h:
* gst/apetag/gsttagdemux.c:
* gst/apetag/gsttagdemux.h:
Port APE tag demuxer over to the new GstTagDemux in -base.
Wim Taymans [Fri, 5 Oct 2007 13:18:19 +0000 (13:18 +0000)]
gst/rtsp/gstrtspsrc.c: Improve flushing behaviour.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
(gst_rtspsrc_perform_seek), (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_handle_internal_src_query),
(gst_rtspsrc_handle_src_query), (new_session_pad),
(gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_loop_send_cmd):
Improve flushing behaviour.
Set state of the udp sources to PAUSE/PLAYING correctly.
Handle events and queries for UDP and TCP transport now.
Stefan Kost [Thu, 4 Oct 2007 07:29:48 +0000 (07:29 +0000)]
gst/rtp/: Add log category.
Original commit message from CVS:
* gst/rtp/gstrtpgsmdepay.c:
* gst/rtp/gstrtpgsmpay.c:
Add log category.
Timo Hotti [Thu, 4 Oct 2007 07:24:02 +0000 (07:24 +0000)]
tests/check/: Add unit tests for payloaders/depayloaders.
Original commit message from CVS:
Patch by: Timo Hotti <Timo.Hotti@sysopendigia.com>
* tests/check/Makefile.am:
* tests/check/pipelines/simple-launch-lines.c:
Add unit tests for payloaders/depayloaders.
Stefan Kost [Tue, 2 Oct 2007 10:49:03 +0000 (10:49 +0000)]
gst/avi/gstavimux.*: Also save codec data for audio streams. Fixes #482495.
Original commit message from CVS:
* gst/avi/gstavimux.c:
* gst/avi/gstavimux.h:
Also save codec data for audio streams. Fixes #482495.
Stefan Kost [Tue, 2 Oct 2007 10:23:04 +0000 (10:23 +0000)]
gst/avi/gstavimux.c: Fix "Index entry has invalid stream nr 1".
Original commit message from CVS:
* gst/avi/gstavimux.c:
Fix "Index entry has invalid stream nr 1".
Add support for muxing aac - work in progress (see #482495).
Wim Taymans [Mon, 1 Oct 2007 16:34:56 +0000 (16:34 +0000)]
gst/rtsp/gstrtspsrc.*: Parse bandwidth modifiers, they are not yet configured in the session manager because we don't...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_bandwidth),
(gst_rtspsrc_collect_bandwidth), (gst_rtspsrc_create_stream),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
* gst/rtsp/gstrtspsrc.h:
Parse bandwidth modifiers, they are not yet configured in the session
manager because we don't have an API for that yet.
Wim Taymans [Mon, 1 Oct 2007 13:57:28 +0000 (13:57 +0000)]
gst/rtsp/gstrtspsrc.c: Use shiny new function in -base to get the default clock-rate.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
Use shiny new function in -base to get the default clock-rate.
Update some docs.
Sébastien Moutte [Sat, 29 Sep 2007 12:50:36 +0000 (12:50 +0000)]
win32/MANIFEST: Add files to win32 manifest.
Original commit message from CVS:
* win32/MANIFEST:
Add files to win32 manifest.
* win32/vs6/libgstaudiofx.dsp:
* win32/vs6/libgstqtdemux.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgstrtsp.dsp:
Update project files.
Wim Taymans [Fri, 28 Sep 2007 14:56:19 +0000 (14:56 +0000)]
gst/rtsp/gstrtspsrc.*: In TCP mode, only timestamp the first buffer. TCP is not real time and it does not make sense ...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
In TCP mode, only timestamp the first buffer. TCP is not real time and
it does not make sense to try to skew compensate, also some servers send
the first batch of data in a burst.
Tim-Philipp Müller [Thu, 27 Sep 2007 15:00:30 +0000 (15:00 +0000)]
gst/matroska/matroska-demux.c: Fix setting the discont flag on the first buffer pushed downstream for formats with pr...
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
Fix setting the discont flag on the first buffer
pushed downstream for formats with private codec
data that needs to be deserialised into buffers
(such as vorbis and FLAC when in a matroska container).
Antoine Tremblay [Thu, 27 Sep 2007 11:10:12 +0000 (11:10 +0000)]
gst/rtp/gstrtpmp4vpay.*: Free the config string. Fixes #480707.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init),
(gst_rtp_mp4v_pay_finalize), (gst_rtp_mp4v_pay_flush),
(gst_rtp_mp4v_pay_handle_buffer):
* gst/rtp/gstrtpmp4vpay.h:
Free the config string. Fixes #480707.
Clean up the timestamp code a little.
Wim Taymans [Wed, 26 Sep 2007 20:12:52 +0000 (20:12 +0000)]
gst/rtsp/gstrtspsrc.*: Set timestamps on RTP buffers in interleaved mode.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_handle_src_query), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_close):
* gst/rtsp/gstrtspsrc.h:
Set timestamps on RTP buffers in interleaved mode.
Mark first buffers with a DISCONT.
Remove flush hack now that sync for live sources has been figured out.
Wim Taymans [Wed, 26 Sep 2007 14:28:20 +0000 (14:28 +0000)]
gst/udp/gstudpsrc.c: Update documentation.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Update documentation.
Wim Taymans [Wed, 26 Sep 2007 14:26:39 +0000 (14:26 +0000)]
gst/qtdemux/gstrtpxqtdepay.*: Fail if we don't know the quicktime format.
Original commit message from CVS:
* gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process),
(gst_rtp_xqt_depay_change_state):
* gst/qtdemux/gstrtpxqtdepay.h:
Fail if we don't know the quicktime format.
Tim-Philipp Müller [Wed, 26 Sep 2007 13:19:17 +0000 (13:19 +0000)]
ext/flac/gstflacenc.*: Save the flow return from the last gst_pad_push() and make sure we pass the right flow return ...
Original commit message from CVS:
* ext/flac/gstflacenc.c:
* ext/flac/gstflacenc.h:
Save the flow return from the last gst_pad_push() and
make sure we pass the right flow return value upstream
in the case of failure; minor clean-ups.
Tim-Philipp Müller [Tue, 25 Sep 2007 19:09:33 +0000 (19:09 +0000)]
Add support for the new GST_TAG_COMPOSER (#459809).
Original commit message from CVS:
* ext/taglib/gstapev2mux.cc:
* ext/taglib/gstid3v2mux.cc:
* gst/apetag/gstapedemux.c:
Add support for the new GST_TAG_COMPOSER (#459809).
Tim-Philipp Müller [Tue, 25 Sep 2007 17:18:34 +0000 (17:18 +0000)]
gst/law/: Compulsive clean-ups: use boilerplate macros, add debug categories, fix up things to conform to symbol nome...
Original commit message from CVS:
* gst/law/alaw-decode.c:
* gst/law/alaw-decode.h:
* gst/law/alaw-encode.c:
* gst/law/alaw-encode.h:
* gst/law/alaw.c:
* gst/law/mulaw-conversion.h:
Compulsive clean-ups: use boilerplate macros, add debug
categories, fix up things to conform to symbol nomenklatura,
etc.
Laurent Glayal [Tue, 25 Sep 2007 16:05:29 +0000 (16:05 +0000)]
gst/law/: Use static tables for A-Law decoding and encoding; this makes
Original commit message from CVS:
Based on patch by: Laurent Glayal <spglegle yahoo fr>
* gst/law/alaw-decode.c:
* gst/law/alaw-encode.c:
Use static tables for A-Law decoding and encoding; this makes
A-Law decoding and encoding less CPU-intensive, but increases
the binary size a bit. Leaving old code around for now,
selectable by a define in the code. Fixes #435435.
Sebastian Dröge [Tue, 25 Sep 2007 08:51:36 +0000 (08:51 +0000)]
configure.ac: Use AG_GST_ARG_WITH_PLUGINS, AG_GST_ARG_ENABLE_EXTERNAL and
Original commit message from CVS:
* configure.ac:
Use AG_GST_ARG_WITH_PLUGINS, AG_GST_ARG_ENABLE_EXTERNAL and
AG_GST_ARG_ENABLE_EXPERIMENTAL instead of duplicating those macros
in configure.ac.
Sebastian Dröge [Tue, 25 Sep 2007 05:03:58 +0000 (05:03 +0000)]
gst/qtdemux/qtdemux.c: Add fourccs for MPEG2 HDV streams. Fixes #479960.
Original commit message from CVS:
Patch by: <j at bootlab dot org>
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
Add fourccs for MPEG2 HDV streams. Fixes #479960.
Stefan Kost [Mon, 24 Sep 2007 10:53:36 +0000 (10:53 +0000)]
Massive leak fixing, plus code cleanups.
Original commit message from CVS:
* ext/audioresample/gstaudioresample.c:
* ext/x264/gstx264enc.c:
* gst/dvdspu/gstdvdspu.c:
* gst/dvdspu/gstdvdspu.h:
* gst/festival/gstfestival.c:
* gst/h264parse/gsth264parse.c:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtsparse.c:
* gst/multifile/gstmultifilesink.c:
* gst/multifile/gstmultifilesrc.c:
* gst/nuvdemux/gstnuvdemux.c:
* sys/dshowsrcwrapper/gstdshowaudiosrc.c:
* sys/dshowsrcwrapper/gstdshowvideosrc.c:
* sys/vcd/vcdsrc.c:
Massive leak fixing, plus code cleanups.
Stefan Kost [Sun, 23 Sep 2007 18:57:14 +0000 (18:57 +0000)]
sys/oss/gstosshelper.c: Use GST_WARNING instead of a g_critical. This situation is not caused by the application.
Original commit message from CVS:
* sys/oss/gstosshelper.c:
Use GST_WARNING instead of a g_critical. This situation is not caused
by the application.
Thomas Vander Stichele [Sat, 22 Sep 2007 18:15:12 +0000 (18:15 +0000)]
po/: Updated translations.
Original commit message from CVS:
* po/LINGUAS:
* po/nl.po:
Updated translations.
Thomas Vander Stichele [Sat, 22 Sep 2007 18:13:58 +0000 (18:13 +0000)]
po/eu.po: Added Basque translation.
Original commit message from CVS:
translated by: Mikel Olasagasti <hey_neken@mundurat.net>
* po/eu.po:
Added Basque translation.
Thomas Vander Stichele [Sat, 22 Sep 2007 18:13:10 +0000 (18:13 +0000)]
po/: Added Chinese (traditional and Hong Kong) translation.
Original commit message from CVS:
translated by: Abel Cheung <abelcheung@gmail.com>
* po/zh_HK.po:
* po/zh_TW.po:
Added Chinese (traditional and Hong Kong) translation.
Thomas Vander Stichele [Sat, 22 Sep 2007 18:10:42 +0000 (18:10 +0000)]
po/pl.po: Added Polish translation.
Original commit message from CVS:
translated by: Jakub Bogusz <qboosh@pld-linux.org>
* po/pl.po:
Added Polish translation.
Thomas Vander Stichele [Sat, 22 Sep 2007 18:09:59 +0000 (18:09 +0000)]
po/fi.po: Added Finnish translation.
Original commit message from CVS:
translated by: Ilkka Tuohela <hile@iki.fi>
* po/fi.po:
Added Finnish translation.
Thomas Vander Stichele [Sat, 22 Sep 2007 18:09:09 +0000 (18:09 +0000)]
po/es.po: Added Spanish translation.
Original commit message from CVS:
translated by: Jorge González González <aloriel@gmail.com>
* po/es.po:
Added Spanish translation.
Thomas Vander Stichele [Sat, 22 Sep 2007 18:08:13 +0000 (18:08 +0000)]
po/da.po: Added Danish translation.
Original commit message from CVS:
translated by: Mogens Jaeger <mogens@jaeger.tf>
* po/da.po:
Added Danish translation.
Thomas Vander Stichele [Sat, 22 Sep 2007 18:06:55 +0000 (18:06 +0000)]
po/zh_CN.po: Added Chinese (simplified) translation.
Original commit message from CVS:
translated by: Funda Wang <fundawang@linux.net.cn>
* po/zh_CN.po:
Added Chinese (simplified) translation.
Thomas Vander Stichele [Sat, 22 Sep 2007 18:05:37 +0000 (18:05 +0000)]
po/bg.po: Added Bulgarian translation.
Original commit message from CVS:
translated by: Alexander Shopov <ash@contact.bg>
* po/bg.po:
Added Bulgarian translation.
Thomas Vander Stichele [Sat, 22 Sep 2007 08:12:57 +0000 (08:12 +0000)]
fix header and comments
Original commit message from CVS:
fix header and comments
Wim Taymans [Fri, 21 Sep 2007 11:34:34 +0000 (11:34 +0000)]
gst/rtp/gstrtpamrdepay.c: Set outgoing packet duration because we can. Fixes #478244 some more.
Original commit message from CVS:
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_process):
Set outgoing packet duration because we can. Fixes #478244 some more.
Stefan Kost [Thu, 20 Sep 2007 13:35:34 +0000 (13:35 +0000)]
ext/cairo/gsttextoverlay.c: Add info about static leak.
Original commit message from CVS:
* ext/cairo/gsttextoverlay.c:
Add info about static leak.
* tests/check/Makefile.am:
* tests/check/generic/states.c:
Improved state change unit test.
Stefan Kost [Wed, 19 Sep 2007 18:19:49 +0000 (18:19 +0000)]
Ignore registries in any format.
Original commit message from CVS:
* docs/plugins/.cvsignore:
* tests/check/.cvsignore:
Ignore registries in any format.
Wim Taymans [Wed, 19 Sep 2007 16:24:09 +0000 (16:24 +0000)]
gst/rtp/gstrtpL16pay.c: Removed some unused code.
Original commit message from CVS:
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_handle_buffer):
Removed some unused code.
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
* gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_handle_buffer):
* gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_handle_buffer):
* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_handle_buffer):
* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_init_packet),
(gst_rtp_theora_pay_flush_packet):
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_flush_packet):
Try to preserve the incomming buffer duration on the outgoing
packets. Fixes #478244.
Tim-Philipp Müller [Wed, 19 Sep 2007 10:22:40 +0000 (10:22 +0000)]
ext/taglib/: Work around compiler warnings with g++-4.2 when assigning a string constant to a gchar * (partially fixe...
Original commit message from CVS:
* ext/taglib/gstapev2mux.cc:
* ext/taglib/gstid3v2mux.cc:
Work around compiler warnings with g++-4.2 when assigning a
string constant to a gchar * (partially fixes #478092).
Tim-Philipp Müller [Tue, 18 Sep 2007 16:44:46 +0000 (16:44 +0000)]
configure.ac: We require core CVS now for gst_base_src_set_do_timestamp().
Original commit message from CVS:
* configure.ac:
We require core CVS now for gst_base_src_set_do_timestamp().
Stefan Kost [Tue, 18 Sep 2007 13:55:06 +0000 (13:55 +0000)]
gst/spectrum/: Handling window resize.
Original commit message from CVS:
* gst/spectrum/demo-audiotest.c:
* gst/spectrum/demo-osssrc.c:
Handling window resize.
Stefan Kost [Tue, 18 Sep 2007 11:45:06 +0000 (11:45 +0000)]
ChangeLog: Add missing newline.
Original commit message from CVS:
* ChangeLog:
Add missing newline.
* gst/librfb/rfbdecoder.c:
Fix the build (missing stdlib.h).
* gst/spectrum/gstspectrum.c:
* gst/spectrum/gstspectrum.h:
Use basetransform segment so that it is correctly managed on flushes
and start/stop. Report message timestamp as stream time, which is what
an application can understand. (Yes these are adapted from wim recent
level element changes)
Jan Schmidt [Mon, 17 Sep 2007 17:35:13 +0000 (17:35 +0000)]
gst/: Fix compiler warnings shown with Forte.
Original commit message from CVS:
* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_class_init):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(new_session_pad), (request_pt_map), (gst_rtspsrc_do_stream_eos),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
(gst_rtspsrc_handle_message):
Fix compiler warnings shown with Forte.
Wim Taymans [Mon, 17 Sep 2007 02:05:14 +0000 (02:05 +0000)]
gst/rtsp/gstrtspsrc.c: Give meaningfull error when all streams failed to configure for some reason.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams),
(gst_rtspsrc_dup_printf):
Give meaningfull error when all streams failed to configure for some
reason.
Wim Taymans [Sun, 16 Sep 2007 19:13:58 +0000 (19:13 +0000)]
gst/rtp/README: Update README with the design for synchronisation rules of RTP on sender and receiver.
Original commit message from CVS:
* gst/rtp/README:
Update README with the design for synchronisation rules of RTP on
sender and receiver.
Sebastian Dröge [Fri, 14 Sep 2007 09:40:49 +0000 (09:40 +0000)]
gst/wavparse/gstwavparse.c: Don't push EOS from the chain function, the element driving the pipeline is responsible f...
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_loop),
(gst_wavparse_chain):
Don't push EOS from the chain function, the element
driving the pipeline is responsible for this. The bug
this was meant to fix seems to be queue not forwarding
EOS in all cases (see #476514).
Wim Taymans [Thu, 13 Sep 2007 17:31:16 +0000 (17:31 +0000)]
gst/level/gstlevel.*: Use basetransform segment so that it is correctly managed on flushes and start/stop.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_class_init), (gst_level_start),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
Use basetransform segment so that it is correctly managed on flushes and
start/stop.
Report message timestamp as stream time, which is what an application
can understand.
Sebastian Dröge [Thu, 13 Sep 2007 15:04:15 +0000 (15:04 +0000)]
Update my mail address.
Original commit message from CVS:
* ext/taglib/gstapev2mux.cc:
* ext/taglib/gstapev2mux.h:
* ext/taglib/gsttaglibmux.c:
* tests/check/elements/apev2mux.c:
Update my mail address.
Sebastian Dröge [Thu, 13 Sep 2007 12:37:56 +0000 (12:37 +0000)]
gst/wavparse/gstwavparse.c: Add EOS logic for the push-based mode too. Fixes #476514.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_eos),
(gst_wavparse_loop), (gst_wavparse_chain):
Add EOS logic for the push-based mode too. Fixes #476514.
Wim Taymans [Wed, 12 Sep 2007 22:01:59 +0000 (22:01 +0000)]
gst/law/: Fix law encoder timestamps.
Original commit message from CVS:
* gst/law/alaw-encode.c: (gst_alawenc_init), (gst_alawenc_chain):
* gst/law/alaw-encode.h:
* gst/law/mulaw-encode.c: (gst_mulawenc_init),
(gst_mulawenc_chain):
* gst/law/mulaw-encode.h:
Fix law encoder timestamps.
Stefan Kost [Wed, 12 Sep 2007 09:13:39 +0000 (09:13 +0000)]
ext/gconf/gstgconfaudiosink.c: Fix warning when building without debug.
Original commit message from CVS:
* ext/gconf/gstgconfaudiosink.c:
Fix warning when building without debug.
* sys/oss/gstossmixertrack.c:
Use const like in alsamixertrack.c (fixes warnings).
Peter Kjellerstedt [Wed, 12 Sep 2007 08:38:21 +0000 (08:38 +0000)]
gst/: Printf format fixes (#476128).
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst-libs/gst/app/gstappsink.c:
* gst/flv/gstflvdemux.c:
* gst/flv/gstflvparse.c:
* gst/interleave/deinterleave.c:
* gst/switch/gstswitch.c:
Printf format fixes (#476128).
Wim Taymans [Tue, 11 Sep 2007 15:37:55 +0000 (15:37 +0000)]
sys/v4l2/v4l2src_calls.c: Fix framerate detection code some more.
Original commit message from CVS:
* sys/v4l2/v4l2src_calls.c:
(gst_v4l2src_probe_caps_for_format_and_size):
Fix framerate detection code some more.
Handle the case where there is a weird step in the stepwise framerates.
Don't overwrite the min interval with the framerate, use a temp variable
instead.
Use max in the Continuous framerate intervals instead of step, which is
1 according to the docs. Fixes #475424.
Wim Taymans [Mon, 10 Sep 2007 19:53:28 +0000 (19:53 +0000)]
gst/udp/gstudpsrc.c: Make udpsrc timestamp outgoing buffers based on when they were received.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create):
Make udpsrc timestamp outgoing buffers based on when they were received.
Also make it output a segment in time.
Stefan Kost [Mon, 10 Sep 2007 06:49:32 +0000 (06:49 +0000)]
gst/avi/gstavidemux.c: Plug a little leak. Little code cleanups.
Original commit message from CVS:
* gst/avi/gstavidemux.c:
Plug a little leak. Little code cleanups.
Tim-Philipp Müller [Sun, 9 Sep 2007 18:08:36 +0000 (18:08 +0000)]
configure.ac: Use AC_TRY_COMPILE instead of AC_TRY_RUN to check for old flac versions, 's good for cross-compilation ...
Original commit message from CVS:
* configure.ac:
Use AC_TRY_COMPILE instead of AC_TRY_RUN to check for old
flac versions, 's good for cross-compilation karma.
Haakon Sporsheim [Fri, 7 Sep 2007 18:04:41 +0000 (18:04 +0000)]
gst/rtp/gstrtph263pay.c: Fix up header structure so that compilers don't add padding between the structure fields, si...
Original commit message from CVS:
Patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
* gst/rtp/gstrtph263pay.c:
Fix up header structure so that compilers don't add padding
between the structure fields, since that would lead to us
sending RTP packets with broken headers (as is currently the
case when compiling with MSVC). Also see similar fixes in
libgstrtp in gst-plugins-base. (#474616; #471194)
Wim Taymans [Fri, 7 Sep 2007 16:04:14 +0000 (16:04 +0000)]
sys/v4l2/v4l2src_calls.c: Don't overwrite our GValue with 0 but instead use the previously computed value. Fixes #471...
Original commit message from CVS:
* sys/v4l2/v4l2src_calls.c:
(gst_v4l2src_probe_caps_for_format_and_size):
Don't overwrite our GValue with 0 but instead use the previously
computed value. Fixes #471823 some more.
Sebastian Dröge [Fri, 7 Sep 2007 15:54:38 +0000 (15:54 +0000)]
gst/spectrum/gstspectrum.c: Use the correct parameter order for the memset calls.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_start),
(gst_spectrum_transform_ip):
Use the correct parameter order for the memset calls.
Thanks to Christian Schaller for noticing.
Tim-Philipp Müller [Thu, 6 Sep 2007 12:00:36 +0000 (12:00 +0000)]
docs/plugins/gst-plugins-good-plugins.hierarchy: No tabs in this file please, or gtk-doc will end up documenting rath...
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
No tabs in this file please, or gtk-doc will end up documenting
rather absurd class hierarchies.
Tim-Philipp Müller [Thu, 6 Sep 2007 10:48:56 +0000 (10:48 +0000)]
ext/gconf/gstswitchsink.c: If the new kid element fails to change state for some reason forward the error message it ...
Original commit message from CVS:
* ext/gconf/gstswitchsink.c:
If the new kid element fails to change state for some reason
(e.g. esdsink not being able to connect to the sound server),
forward the error message it posted on the bus instead of just
posting a generic 'Internal state change error: please file a
bug' error message. Fixes #471364.
Sebastian Dröge [Thu, 6 Sep 2007 07:21:22 +0000 (07:21 +0000)]
Port GstSpectrum to GstAudioFilter and libgstfft, add support for int32, float and double, use floats for the message...
Original commit message from CVS:
* configure.ac:
* gst/spectrum/Makefile.am:
* gst/spectrum/demo-audiotest.c: (draw_spectrum),
(message_handler), (main):
* gst/spectrum/demo-osssrc.c: (draw_spectrum), (message_handler):
* gst/spectrum/gstspectrum.c: (gst_spectrum_base_init),
(gst_spectrum_class_init), (gst_spectrum_init),
(gst_spectrum_dispose), (gst_spectrum_set_property),
(gst_spectrum_get_property), (gst_spectrum_start),
(gst_spectrum_setup), (gst_spectrum_message_new),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Port GstSpectrum to GstAudioFilter and libgstfft, add support
for int32, float and double, use floats for the message contents,
average all FFTs done in one interval for better results, use
a better windowing function, allow posting the phase in the message
and actually do an FFT with the requested number of bands instead
of interpolating.
* tests/check/elements/spectrum.c: (GST_START_TEST),
(spectrum_suite):
Improve the units tests by checking for a 11025Hz sine wave
and add unit tests for all 4 supported sample types.
Tim-Philipp Müller [Wed, 5 Sep 2007 16:23:21 +0000 (16:23 +0000)]
gst/qtdemux/: Don't assume tags are encoded as UTF-8 (#473670).
Original commit message from CVS:
* gst/qtdemux/Makefile.am:
* gst/qtdemux/qtdemux.c:
Don't assume tags are encoded as UTF-8 (#473670).
Tim-Philipp Müller [Wed, 5 Sep 2007 14:43:16 +0000 (14:43 +0000)]
sys/v4l2/: Implement LATENCY queries in the crudest way possible so I don't have to use sync=false any longer when te...
Original commit message from CVS:
* sys/v4l2/gstv4l2src.c:
* sys/v4l2/gstv4l2src.h:
* sys/v4l2/v4l2src_calls.c:
Implement LATENCY queries in the crudest way possible so I don't
have to use sync=false any longer when testing with videosinks.
Tim-Philipp Müller [Wed, 5 Sep 2007 09:25:23 +0000 (09:25 +0000)]
configure.ac: Fix build.
Original commit message from CVS:
* configure.ac:
Fix build.
Wim Taymans [Wed, 5 Sep 2007 00:12:46 +0000 (00:12 +0000)]
sys/v4l2/v4l2src_calls.c: Add some more debugging in the framerate function.
Original commit message from CVS:
* sys/v4l2/v4l2src_calls.c:
(gst_v4l2src_probe_caps_for_format_and_size):
Add some more debugging in the framerate function.
Iterate stepwise framerate up to and _including_ the max and if nothing
was added to the list, add a dummy 0/1 to 100/1 framerate so that we
don't end up with an empty list.
Wim Taymans [Tue, 4 Sep 2007 22:42:21 +0000 (22:42 +0000)]
gst/udp/gstmultiudpsink.c: Add property do configure destination address/port pairs
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_set_clients_string),
(gst_multiudpsink_get_clients_string),
(gst_multiudpsink_set_property), (gst_multiudpsink_get_property),
(gst_multiudpsink_init_send), (gst_multiudpsink_add_internal),
(gst_multiudpsink_add), (gst_multiudpsink_clear_internal),
(gst_multiudpsink_clear):
Add property do configure destination address/port pairs
API:GstMultiUDPSink::clients
Wim Taymans [Tue, 4 Sep 2007 18:30:22 +0000 (18:30 +0000)]
tests/examples/: Added some RTP example scripts for sending and receiving RTP streams.
Original commit message from CVS:
* tests/examples/Makefile.am:
* tests/examples/rtp/Makefile.am:
* tests/examples/rtp/client-H263p-AMR.sh:
* tests/examples/rtp/client-H263p-PCMA.sdp:
* tests/examples/rtp/client-H263p-PCMA.sh:
* tests/examples/rtp/client-H264-PCMA.sdp:
* tests/examples/rtp/client-H264-PCMA.sh:
* tests/examples/rtp/client-PCMA.sh:
* tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh:
* tests/examples/rtp/server-alsasrc-PCMA.sh:
* tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
Added some RTP example scripts for sending and receiving RTP streams.
Wim Taymans [Tue, 4 Sep 2007 16:40:05 +0000 (16:40 +0000)]
sys/v4l2/gstv4l2src.c: Restructure the setcaps function so that we can also compute the expected GStreamer output siz...
Original commit message from CVS:
* sys/v4l2/gstv4l2src.c: (gst_v4l2_get_caps_info),
(gst_v4l2src_set_caps), (gst_v4l2src_get_mmap):
Restructure the setcaps function so that we can also compute the
expected GStreamer output size of the video frames.
Set frame_byte_size correctly so that read-based devices have a chance
of working correctly.
When grabbing a frame, discard frames that are not of the expected size.
Some cameras don't output the right framesize for the first buffer.
Try only a couple of times to get a valid frame, else error out.
* sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
(gst_v4l2_fill_lists), (gst_v4l2_get_input):
Add some more debug info when scanning the device.
* sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_new),
(gst_v4l2_buffer_pool_new), (gst_v4l2_buffer_pool_activate),
(gst_v4l2src_fill_format_list), (gst_v4l2src_grab_frame),
(gst_v4l2src_set_capture), (gst_v4l2src_capture_init):
Add some more debug info when dequeing a frame.
Stefan Kost [Tue, 4 Sep 2007 14:37:22 +0000 (14:37 +0000)]
gst/wavparse/gstwavparse.c: More code cleanups. Add some more comment and improve debugs logs.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
More code cleanups. Add some more comment and improve debugs logs.
Stefan Kost [Tue, 4 Sep 2007 07:58:36 +0000 (07:58 +0000)]
gst/wavparse/gstwavparse.*: Implement seek-query. Refactor duration calculations. Appropriate use of uint64_scale_int...
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
* gst/wavparse/gstwavparse.h:
Implement seek-query. Refactor duration calculations. Appropriate use
of uint64_scale_int and uint64_scale. Move repeadedly calculated stuff
out of loops.
Stefan Kost [Mon, 3 Sep 2007 07:44:34 +0000 (07:44 +0000)]
gst/avi/gstavidemux.c: Implement seek-query.
Original commit message from CVS:
* gst/avi/gstavidemux.c:
Implement seek-query.
Wim Taymans [Wed, 29 Aug 2007 21:43:08 +0000 (21:43 +0000)]
gst/rtsp/gstrtspsrc.c: Use new basesink async property to make sparse RTCP packet not wait for preroll.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_dup_printf):
Use new basesink async property to make sparse RTCP packet not wait for
preroll.
Jan Schmidt [Mon, 27 Aug 2007 14:44:19 +0000 (14:44 +0000)]
gst/audiofx/Makefile.am: Dist the right file.
Original commit message from CVS:
* gst/audiofx/Makefile.am:
Dist the right file.
Wim Taymans [Thu, 23 Aug 2007 16:27:36 +0000 (16:27 +0000)]
gst/rtsp/gstrtspsrc.c: Make sure we generate and parse floating point values in the POSIX locale instead of the curre...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_dup_printf),
(gst_rtspsrc_get_float), (gst_rtspsrc_play):
Make sure we generate and parse floating point values in the POSIX
locale instead of the current locale.
Wim Taymans [Wed, 22 Aug 2007 15:01:29 +0000 (15:01 +0000)]
gst/rtsp/gstrtspsrc.*: Fix method detection again.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
Fix method detection again.
Keep track of when we must send a Range header.
Use segment values for Range, Speed and Scale headers.
Parse Speed and Scale headers to update the segment values.
Mark Nauwelaerts [Wed, 22 Aug 2007 08:22:50 +0000 (08:22 +0000)]
sys/v4l2/v4l2src_calls.c: Handle optional v4l2 ioctls gracefully.
Original commit message from CVS:
patch by: Mark Nauwelaerts <manauw@skynet.be>
* sys/v4l2/v4l2src_calls.c:
Handle optional v4l2 ioctls gracefully.
Wim Taymans [Mon, 20 Aug 2007 16:52:03 +0000 (16:52 +0000)]
gst/rtp/: Added an H263 depayloader. Fixes #369392.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_base_init),
(gst_rtp_h263_depay_class_init), (gst_rtp_h263_depay_init),
(gst_rtp_h263_depay_finalize), (gst_rtp_h263_depay_setcaps),
(gst_rtp_h263_depay_process), (gst_rtp_h263_depay_set_property),
(gst_rtp_h263_depay_get_property),
(gst_rtp_h263_depay_change_state),
(gst_rtp_h263_depay_plugin_init):
* gst/rtp/gstrtph263depay.h:
Added an H263 depayloader. Fixes #369392.
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
(gst_rtp_h263p_depay_process):
* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
(gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_flush):
Make the H263+ pay/depayloader support H263-1998 and H263-2000
payloads.
Also alow plain H263 on the h263p payloaders. Fixes #465040.
Sebastian Dröge [Sun, 19 Aug 2007 19:16:33 +0000 (19:16 +0000)]
gst/filter/: Add small comparision with the chebyshev filters in the docs.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c:
* gst/filter/gstlpwsinc.c:
Add small comparision with the chebyshev filters in the docs.
Sebastian Dröge [Sun, 19 Aug 2007 19:11:04 +0000 (19:11 +0000)]
gst/audiofx/: Add small comparision with the windowed sinc filters in the docs.
Original commit message from CVS:
* gst/audiofx/audiochebyshevfreqband.c:
* gst/audiofx/audiochebyshevfreqlimit.c:
Add small comparision with the windowed sinc filters in the docs.
Sebastian Dröge [Sun, 19 Aug 2007 19:01:45 +0000 (19:01 +0000)]
tests/check/elements/: Also test everything in 32 bit float mode.
Original commit message from CVS:
* tests/check/elements/bpwsinc.c: (GST_START_TEST),
(bpwsinc_suite):
* tests/check/elements/lpwsinc.c: (GST_START_TEST),
(lpwsinc_suite):
Also test everything in 32 bit float mode.
Sebastian Dröge [Sun, 19 Aug 2007 18:47:19 +0000 (18:47 +0000)]
tests/check/elements/: Also test 32 bit float mode and the type 2 variants of the filters.
Original commit message from CVS:
* tests/check/elements/audiochebyshevfreqband.c: (GST_START_TEST),
(audiochebyshevfreqband_suite):
* tests/check/elements/audiochebyshevfreqlimit.c: (GST_START_TEST),
(audiochebyshevfreqlimit_suite):
Also test 32 bit float mode and the type 2 variants of the filters.
Wim Taymans [Sat, 18 Aug 2007 19:44:55 +0000 (19:44 +0000)]
gst/rtsp/gstrtspsrc.c: Refactor the udp and interleaved loop function a bit.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_loop):
Refactor the udp and interleaved loop function a bit.
Wim Taymans [Fri, 17 Aug 2007 17:08:11 +0000 (17:08 +0000)]
gst/rtsp/gstrtspsrc.*: Protect connection activity with a new lock, avoids deadlocks when going to PAUSED. Fixes #455...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_connection_send),
(gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_try_send), (gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Protect connection activity with a new lock, avoids deadlocks when going
to PAUSED. Fixes #455808.
Wim Taymans [Fri, 17 Aug 2007 15:30:39 +0000 (15:30 +0000)]
gst/debug/rndbuffersize.c: Fix debug statement.
Original commit message from CVS:
* gst/debug/rndbuffersize.c: (gst_rnd_buffer_size_loop):
Fix debug statement.
Wim Taymans [Fri, 17 Aug 2007 15:28:40 +0000 (15:28 +0000)]
gst/rtsp/gstrtspsrc.c: Fix stray %u in debug line as spotted by Saur on IRC.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_stream_eos):
Fix stray %u in debug line as spotted by Saur on IRC.
Sebastian Dröge [Fri, 17 Aug 2007 15:05:17 +0000 (15:05 +0000)]
Use generator macros for the process functions for the different sample types, add lower upper boundaries for the GOb...
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_set_property), (bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Use generator macros for the process functions for the different
sample types, add lower upper boundaries for the GObject properties
so automatically generated UIs can use sliders and change frequency
properties to floats to save a bit of memory, even ints would in
theory be enough. Also rename frequency to cutoff for consistency
reasons.
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
Regenerated for the above changes.
Sebastian Dröge [Fri, 17 Aug 2007 14:43:33 +0000 (14:43 +0000)]
gst/audiofx/: Use generator macros for the process functions for the different sample types, add lower upper boundari...
Original commit message from CVS:
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_class_init):
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_class_init):
Use generator macros for the process functions for the different
sample types, add lower upper boundaries for the GObject properties
so automatically generated UIs can use sliders and add a note about
the number of poles as a too high number of poles combined with
very low or very high frequencies will produce only noise.
* docs/plugins/gst-plugins-good-plugins.args:
Regenerated for the property changes.
Wim Taymans [Fri, 17 Aug 2007 14:15:19 +0000 (14:15 +0000)]
gst/rtsp/gstrtspsrc.*: Improve timeout handling.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_property),
(gst_rtspsrc_flush), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_parse_range),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Improve timeout handling.
Use the same socket for sending and receiving RTCP packets so that some
servers can track clients better.
Improve connection closed handling. Try to reconnect.
Don't overwrite our content base with NULL.
Improve debugging.
Improve range parsing and handling.
Remove flushing hack now that core does the right thing.
Wim Taymans [Fri, 17 Aug 2007 13:59:15 +0000 (13:59 +0000)]
gst/udp/gstmultiudpsink.*: Add support for getting and setting the socket to use.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_init), (gst_multiudpsink_set_property),
(gst_multiudpsink_get_property), (gst_multiudpsink_init_send),
(gst_multiudpsink_close), (gst_multiudpsink_add):
* gst/udp/gstmultiudpsink.h:
Add support for getting and setting the socket to use.
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
(gst_udpsrc_create), (gst_udpsrc_get_property):
Add support for getting the currently used socket.
Sebastian Dröge [Thu, 16 Aug 2007 19:22:48 +0000 (19:22 +0000)]
gst/filter/gstbpwsinc.*: Implement latency query and only forward those samples downstream that actually contain the ...
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (process_32), (process_64),
(bpwsinc_build_kernel), (bpwsinc_push_residue),
(bpwsinc_transform), (bpwsinc_start), (bpwsinc_query),
(bpwsinc_query_type), (bpwsinc_event), (bpwsinc_set_property):
* gst/filter/gstbpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/bpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
* gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
Reset residue length only when actually creating a residue.
Sebastian Dröge [Thu, 16 Aug 2007 17:02:07 +0000 (17:02 +0000)]
gst/audiofx/: Add Chebyshev lowpass/highpass and bandpass/bandreject elements.
Original commit message from CVS:
reviewed by: Stefan Kost <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_mode_get_type),
(gst_audio_chebyshev_freq_band_base_init),
(gst_audio_chebyshev_freq_band_dispose),
(gst_audio_chebyshev_freq_band_class_init),
(gst_audio_chebyshev_freq_band_init),
(generate_biquad_coefficients), (calculate_gain),
(generate_coefficients),
(gst_audio_chebyshev_freq_band_set_property),
(gst_audio_chebyshev_freq_band_get_property),
(gst_audio_chebyshev_freq_band_setup), (process), (process_64),
(process_32), (gst_audio_chebyshev_freq_band_transform_ip),
(gst_audio_chebyshev_freq_band_start):
* gst/audiofx/audiochebyshevfreqband.h:
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_mode_get_type),
(gst_audio_chebyshev_freq_limit_base_init),
(gst_audio_chebyshev_freq_limit_dispose),
(gst_audio_chebyshev_freq_limit_class_init),
(gst_audio_chebyshev_freq_limit_init),
(generate_biquad_coefficients), (calculate_gain),
(generate_coefficients),
(gst_audio_chebyshev_freq_limit_set_property),
(gst_audio_chebyshev_freq_limit_get_property),
(gst_audio_chebyshev_freq_limit_setup), (process), (process_64),
(process_32), (gst_audio_chebyshev_freq_limit_transform_ip),
(gst_audio_chebyshev_freq_limit_start):
* gst/audiofx/audiochebyshevfreqlimit.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add Chebyshev lowpass/highpass and bandpass/bandreject elements.
Fixes #464800.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/audiochebyshevfreqband.c:
(setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband),
(GST_START_TEST), (audiochebyshevfreqband_suite), (main):
* tests/check/elements/audiochebyshevfreqlimit.c:
(setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit),
(GST_START_TEST), (audiochebyshevfreqlimit_suite), (main):
Add unit tests for the chebyshev filters.
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-1394.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-jpeg.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-shout2send.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
And add docs for the chebyshev filters. While doing
that also run make update in docs/plugins.
Stefan Kost [Thu, 16 Aug 2007 12:15:06 +0000 (12:15 +0000)]
Make ro memory to share.
Original commit message from CVS:
* ext/annodex/gstcmmltag.c:
* gst/rtp/gstrtpvorbispay.c:
Make ro memory to share.
Wim Taymans [Thu, 16 Aug 2007 11:49:01 +0000 (11:49 +0000)]
gst/udp/gstudpsrc.c: Improve UDP performance by avoiding a select() when we have data available immediatly.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Improve UDP performance by avoiding a select() when we have data
available immediatly.
Wim Taymans [Thu, 16 Aug 2007 11:47:19 +0000 (11:47 +0000)]
gst/rtsp/gstrtpdec.*: Add (dummy) SSRC management signals.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT),
(gst_rtp_dec_class_init):
* gst/rtsp/gstrtpdec.h:
Add (dummy) SSRC management signals.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(find_stream), (gst_rtspsrc_create_stream), (new_session_pad),
(request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc),
(on_timeout), (gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add connection-speed property.
Add find_stream helper functions.
Handle stream EOS based on BYE messages or SSRC timeout.
Returns SUCCESS from the state change function as we hide our async
elements from the parent.
Sebastian Dröge [Thu, 16 Aug 2007 09:48:27 +0000 (09:48 +0000)]
gst/filter/gstlpwsinc.*: Implement latency query and only forward those samples downstream that actually contain the ...
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (process_32), (process_64),
(lpwsinc_build_kernel), (lpwsinc_push_residue),
(lpwsinc_transform), (lpwsinc_start), (lpwsinc_query),
(lpwsinc_query_type), (lpwsinc_event), (lpwsinc_set_property):
* gst/filter/gstlpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
Stefan Kost [Thu, 16 Aug 2007 07:40:48 +0000 (07:40 +0000)]
gst/debug/rndbuffersize.c: Fix da leak.
Original commit message from CVS:
* gst/debug/rndbuffersize.c:
Fix da leak.
Stefan Kost [Tue, 14 Aug 2007 13:50:43 +0000 (13:50 +0000)]
gst/debug/: Add new test element and clean-up the others a little.
Original commit message from CVS:
* gst/debug/Makefile.am:
* gst/debug/breakmydata.c:
* gst/debug/gstdebug.c:
* gst/debug/negotiation.c:
* gst/debug/progressreport.c:
* gst/debug/rndbuffersize.c:
* gst/debug/testplugin.c:
Add new test element and clean-up the others a little.
Sebastian Dröge [Mon, 13 Aug 2007 13:50:39 +0000 (13:50 +0000)]
Add docs for lpwsinc and bpwsinc and integrate them into the build system. While doing that also update all other doc...
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-xingheader.xml:
* docs/plugins/inspect/plugin-xvid.xml:
* gst/filter/gstbpwsinc.c:
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c:
* gst/filter/gstlpwsinc.h:
Add docs for lpwsinc and bpwsinc and integrate them
into the build system. While doing that also update
all other docs via make update in docs/plugins.
Sebastian Dröge [Sun, 12 Aug 2007 20:55:01 +0000 (20:55 +0000)]
tests/check/elements/bpwsinc.c: Make one test constraint a bit stricter.
Original commit message from CVS:
* tests/check/elements/bpwsinc.c: (GST_START_TEST):
Make one test constraint a bit stricter.
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