Thomas Vander Stichele [Fri, 8 Jun 2007 20:20:34 +0000 (20:20 +0000)]
Move wavpack to good.
Original commit message from CVS:
* configure.ac:
* 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.hierarchy:
* docs/plugins/inspect/plugin-wavpack.xml:
* ext/Makefile.am:
* ext/wavpack/Makefile.am:
* ext/wavpack/gstwavpack.c:
* ext/wavpack/gstwavpackcommon.c:
* ext/wavpack/gstwavpackcommon.h:
* ext/wavpack/gstwavpackdec.c:
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackenc.c:
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.c:
* ext/wavpack/gstwavpackparse.h:
* ext/wavpack/gstwavpackstreamreader.c:
* ext/wavpack/gstwavpackstreamreader.h:
* ext/wavpack/md5.c:
* ext/wavpack/md5.h:
* tests/check/Makefile.am:
* tests/check/elements/wavpackdec.c:
* tests/check/elements/wavpackenc.c:
* tests/check/elements/wavpackparse.c:
Move wavpack to good.
Jan Schmidt [Fri, 8 Jun 2007 17:36:46 +0000 (17:36 +0000)]
Remove DirectDraw & DirectSound plugins, as they've moved to Good
Original commit message from CVS:
* configure.ac:
* 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/inspect/plugin-directdraw.xml:
* docs/plugins/inspect/plugin-directsound.xml:
* sys/Makefile.am:
* sys/directdraw/Makefile.am:
* sys/directdraw/gstdirectdrawplugin.c:
* sys/directdraw/gstdirectdrawsink.c:
* sys/directdraw/gstdirectdrawsink.h:
* sys/directsound/Makefile.am:
* sys/directsound/gstdirectsoundplugin.c:
* sys/directsound/gstdirectsoundsink.c:
* sys/directsound/gstdirectsoundsink.h:
* win32/MANIFEST:
* win32/gst.sln:
* win32/vs6/libgstdirectdraw.dsp:
* win32/vs6/libgstdirectsound.dsp:
* win32/vs7/libgstdirectdraw.vcproj:
* win32/vs7/libgstdirectsound.vcproj:
* win32/vs8/libgstdirectdraw.vcproj:
* win32/vs8/libgstdirectsound.vcproj:
Remove DirectDraw & DirectSound plugins, as they've moved to Good
Tim-Philipp Müller [Fri, 8 Jun 2007 16:49:41 +0000 (16:49 +0000)]
When driving the pipeline, also post an error when we get a not-linked flow return from downstream.
Original commit message from CVS:
* ext/timidity/gsttimidity.c: (gst_timidity_loop):
* ext/timidity/gstwildmidi.c: (gst_wildmidi_loop):
* gst/tta/gstttaparse.c: (gst_tta_parse_loop):
When driving the pipeline, also post an error when we get a
not-linked flow return from downstream.
Jan Schmidt [Fri, 8 Jun 2007 16:31:15 +0000 (16:31 +0000)]
Rename the keep-aspect-ratio property to force-aspect-ratio to make it consistent with xvimagesink and ximagesink.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
* sys/directdraw/gstdirectdrawsink.c:
(gst_directdraw_sink_class_init):
Rename the keep-aspect-ratio property to force-aspect-ratio to make
it consistent with xvimagesink and ximagesink.
Tim-Philipp Müller [Fri, 8 Jun 2007 08:12:43 +0000 (08:12 +0000)]
tests/icles/videocrop-test.c: Default to xvimagesink instead of autovideosink while autovideosink/ghostpads/whatever ...
Original commit message from CVS:
* tests/icles/videocrop-test.c: (main):
Default to xvimagesink instead of autovideosink while
autovideosink/ghostpads/whatever don't handle the way we use it in
the way we expect it to.
Tim-Philipp Müller [Wed, 6 Jun 2007 18:30:22 +0000 (18:30 +0000)]
configure.ac: Bump requirements to released versions of core and -base, and remove special-casing for equalizer and r...
Original commit message from CVS:
* configure.ac:
Bump requirements to released versions of core and -base, and remove
special-casing for equalizer and rtpmanager as it's not needed any
longer.
Jan Schmidt [Wed, 6 Jun 2007 15:58:31 +0000 (15:58 +0000)]
sys/glsink/glimagesink.c: Sprinkle in some XSync calls to avoid raciness with broken drivers (ATI) when re-using a si...
Original commit message from CVS:
* sys/glsink/glimagesink.c: (gst_glimage_sink_stop),
(gst_glimage_sink_create_window), (gst_glimage_sink_init_display):
Sprinkle in some XSync calls to avoid raciness with broken
drivers (ATI) when re-using a single glimagesink.
Jan Schmidt [Wed, 6 Jun 2007 11:03:24 +0000 (11:03 +0000)]
gst/mpegvideoparse/: Fix some silly bugs with calculating the guard sizes.
Original commit message from CVS:
* gst/mpegvideoparse/mpegpacketiser.c:
(mpeg_util_parse_extension_packet), (mpeg_util_parse_sequence_hdr),
(mpeg_util_parse_picture_hdr):
* gst/mpegvideoparse/mpegvideoparse.c:
(mpegvideoparse_handle_sequence), (mpegvideoparse_drain_avail):
Fix some silly bugs with calculating the guard sizes.
Properly compare the old sequence header structure with the new one.
Don't error out on an invalid sequence - just ignore it.
Tim-Philipp Müller [Wed, 6 Jun 2007 09:11:31 +0000 (09:11 +0000)]
gst/real/gstrealvideodec.c: Printf fix in debug statement; also print the right number there.
Original commit message from CVS:
* gst/real/gstrealvideodec.c: (gst_real_video_dec_decode):
Printf fix in debug statement; also print the right number there.
Tim-Philipp Müller [Wed, 6 Jun 2007 08:06:44 +0000 (08:06 +0000)]
ext/sdl/: Add GST_CFLAGS, which apparently somehow fixes the build somewhere (#444499); initialise variable to avoid ...
Original commit message from CVS:
* ext/sdl/Makefile.am:
* ext/sdl/sdlvideosink.c: (gst_sdlv_process_events):
Add GST_CFLAGS, which apparently somehow fixes the build somewhere
(#444499); initialise variable to avoid false compiler warning.
Christian Schaller [Tue, 5 Jun 2007 14:17:36 +0000 (14:17 +0000)]
fix up spec file
Original commit message from CVS:
fix up spec file
Wim Taymans [Fri, 1 Jun 2007 14:36:04 +0000 (14:36 +0000)]
Add plugin to generate a pattern detectable by videodetect.
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/inspect/plugin-videosignal.xml:
* gst/videosignal/Makefile.am:
* gst/videosignal/gstvideomark.c: (gst_video_mark_set_caps),
(gst_video_mark_draw_box), (gst_video_mark_420),
(gst_video_mark_transform_ip), (gst_video_mark_set_property),
(gst_video_mark_get_property), (gst_video_mark_base_init),
(gst_video_mark_class_init), (gst_video_mark_init),
(gst_video_mark_get_type):
* gst/videosignal/gstvideomark.h:
* gst/videosignal/gstvideosignal.c: (plugin_init):
Add plugin to generate a pattern detectable by videodetect.
Tim-Philipp Müller [Fri, 1 Jun 2007 11:59:52 +0000 (11:59 +0000)]
ext/neon/gstneonhttpsrc.c: Don't leak buffer when returning an error.
Original commit message from CVS:
* ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_create):
Don't leak buffer when returning an error.
René Stadler [Fri, 1 Jun 2007 09:58:01 +0000 (09:58 +0000)]
ext/neon/gstneonhttpsrc.*: Deprecated "uri" property. Clean up property descriptions.
Original commit message from CVS:
Patch by René Stadler <mail at renestadler dot de>:
* ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_class_init),
(gst_neonhttp_src_init), (gst_neonhttp_src_dispose),
(gst_neonhttp_src_set_property), (gst_neonhttp_src_get_property),
(gst_neonhttp_src_start), (gst_neonhttp_src_do_seek),
(gst_neonhttp_src_set_location),
(gst_neonhttp_src_send_request_and_redirect),
(gst_neonhttp_src_uri_get_uri), (gst_neonhttp_src_uri_set_uri):
* ext/neon/gstneonhttpsrc.h:
Deprecated "uri" property. Clean up property descriptions.
Change default User-Agent to the slightly more descriptive
"GStreamer neonhttpsrc".
Various other small cleanups, mostly property related.
Tim-Philipp Müller [Thu, 31 May 2007 19:12:37 +0000 (19:12 +0000)]
ext/libmms/gstmms.h: No reason to use gpointers instead of typed pointes here as far as I can see.
Original commit message from CVS:
* ext/libmms/gstmms.h:
No reason to use gpointers instead of typed pointes here as far as I
can see.
* ext/mythtv/gstmythtvsrc.c:
* ext/neon/gstneonhttpsrc.c:
* gst/switch/gstswitch.c:
Don't use gtk-doc magic markers for things that aren't meant to be
parsed by gtk-doc. Makes gtk-doc complain a bit less.
Wim Taymans [Wed, 30 May 2007 14:52:07 +0000 (14:52 +0000)]
Added videosignal plugin with two plugins to analyse video frames.
Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/inspect/plugin-videosignal.xml:
* gst/videosignal/Makefile.am:
* gst/videosignal/gstvideoanalyse.c: (gst_video_analyse_set_caps),
(gst_video_analyse_post_message), (gst_video_analyse_420),
(gst_video_analyse_transform_ip), (gst_video_analyse_set_property),
(gst_video_analyse_get_property), (gst_video_analyse_base_init),
(gst_video_analyse_class_init), (gst_video_analyse_init),
(gst_video_analyse_get_type):
* gst/videosignal/gstvideoanalyse.h:
* gst/videosignal/gstvideodetect.c: (gst_video_detect_set_caps),
(gst_video_detect_post_message),
(gst_video_detect_calc_brightness), (gst_video_detect_420),
(gst_video_detect_transform_ip), (gst_video_detect_set_property),
(gst_video_detect_get_property), (gst_video_detect_base_init),
(gst_video_detect_class_init), (gst_video_detect_init),
(gst_video_detect_get_type):
* gst/videosignal/gstvideodetect.h:
* gst/videosignal/gstvideosignal.c: (plugin_init):
* gst/videosignal/gstvideosignal.h:
Added videosignal plugin with two plugins to analyse video frames.
Added videoanalyse to report about brightness and variance in video
frames.
Added videodetect to detect predefined patterns in a video signal.
Tim-Philipp Müller [Wed, 30 May 2007 09:30:52 +0000 (09:30 +0000)]
docs/plugins/gst-plugins-bad-plugins.*: More updates.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.prerequisites:
* docs/plugins/gst-plugins-bad-plugins.signals:
More updates.
Tim-Philipp Müller [Wed, 30 May 2007 09:08:24 +0000 (09:08 +0000)]
docs/plugins/: Updates; update inspect info for rtpmanager => gstrtpmanager rename, hopefully that makes the build bo...
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-rtpmanager.xml:
* docs/plugins/inspect/plugin-sdl.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-swfdec.xml:
Updates; update inspect info for rtpmanager => gstrtpmanager rename,
hopefully that makes the build bots happy again.
Wim Taymans [Mon, 28 May 2007 16:37:47 +0000 (16:37 +0000)]
Rename elements to avoid conflict with farsight elements with the same name. Fixes #430664.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst/rtpmanager/gstrtpbin.c: (create_session), (create_stream),
(gst_rtp_bin_class_init), (create_recv_rtp), (create_recv_rtcp),
(create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpclient.c: (create_stream),
(gst_rtp_client_request_new_pad):
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpssrcdemux.c:
Rename elements to avoid conflict with farsight elements with the same
name. Fixes #430664.
Vincent Torri [Thu, 24 May 2007 08:35:23 +0000 (08:35 +0000)]
sys/directdraw/gstdirectdrawsink.*: Fix more warnings when compiling with MingW (#439914).
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry fr>
* sys/directdraw/gstdirectdrawsink.c:
(gst_directdraw_sink_buffer_alloc),
(gst_directdraw_sink_show_frame),
(gst_directdraw_sink_check_primary_surface),
(gst_directdraw_sink_check_offscreen_surface),
(EnumModesCallback2), (gst_directdraw_sink_get_ddrawcaps),
(gst_directdraw_sink_surface_create):
* sys/directdraw/gstdirectdrawsink.h:
Fix more warnings when compiling with MingW (#439914).
Sébastien Moutte [Wed, 23 May 2007 22:44:12 +0000 (22:44 +0000)]
docs/plugins/gst-plugins-bad-plugins.args: Remove directsoundsink property doc as this sink use the mixer interface now.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
Remove directsoundsink property doc as this sink use the mixer
interface now.
* docs/plugins/gst-plugins-bad-plugins.interfaces:
Add interfaces implemented by Windows sinks.
* sys/directsound/gstdirectsoundsink.c:
* sys/directsound/gstdirectsoundsink.h:
Remove directsoundsink property and implement the mixer interface.
* win32/vs6/gst_plugins_bad.dsw:
* win32/vs6/libgstdirectsound.dsp:
Update project files.
* gst-libs/gst/dshow/gstdshow.cpp:
* gst-libs/gst/dshow/gstdshow.h:
* gst-libs/gst/dshow/gstdshowfakesink.cpp:
* gst-libs/gst/dshow/gstdshowfakesink.h:
* gst-libs/gst/dshow/gstdshowfakesrc.cpp:
* gst-libs/gst/dshow/gstdshowfakesrc.h:
* gst-libs/gst/dshow/gstdshowinterface.cpp:
* gst-libs/gst/dshow/gstdshowinterface.h:
* win32/common/libgstdshow.def:
* win32/vs6/libgstdshow.dsp:
Add a new gst library which allow to create internal Direct Show
graph (pipelines) to wrap Windows sources, decoders or encoders.
It includes a DirectShow fake source and sink and utility functions.
* sys/dshowsrcwrapper/gstdshowaudiosrc.c:
* sys/dshowsrcwrapper/gstdshowaudiosrc.h:
* sys/dshowsrcwrapper/gstdshowsrcwrapper.c:
* sys/dshowsrcwrapper/gstdshowsrcwrapper.h:
* sys/dshowsrcwrapper/gstdshowvideosrc.c:
* sys/dshowsrcwrapper/gstdshowvideosrc.h:
* win32/vs6/libdshowsrcwrapper.dsp:
Add a new plugin to wrap DirectShow sources on Windows.
It gets data from any webcam, dv cam, micro. We could add
tv tunner card later.
René Stadler [Wed, 23 May 2007 18:00:15 +0000 (18:00 +0000)]
ext/sdl/sdlvideosink.c: Separate the authors by newlines instead of nothing. Fixes #440774.
Original commit message from CVS:
Patch by René Stadler <mail at renestadler dot de>:
* ext/sdl/sdlvideosink.c:
Separate the authors by newlines instead of nothing. Fixes #440774.
Wim Taymans [Wed, 23 May 2007 16:14:13 +0000 (16:14 +0000)]
docs/plugins/inspect/plugin-rtpmanager.xml: Add doc xml file.
Original commit message from CVS:
* docs/plugins/inspect/plugin-rtpmanager.xml:
Add doc xml file.
Wim Taymans [Wed, 23 May 2007 13:08:52 +0000 (13:08 +0000)]
Document stuff.
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:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
(gst_rtp_pt_demux_clear_pt_map):
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(rtcp_thread), (gst_rtp_session_clear_pt_map):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_class_init):
Document stuff.
Add clear-pt-map action signal where needed.
Stefan Kost [Tue, 22 May 2007 11:14:13 +0000 (11:14 +0000)]
configure.ac: Depend on gstreamer-0.10.12.1. gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _Gs...
Original commit message from CVS:
* configure.ac:
Depend on gstreamer-0.10.12.1.
* gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN,
_GstIirEqualizerBand, object, _GstIirEqualizerBandClass,
parent_class, gst_iir_equalizer_band_set_property,
gst_iir_equalizer_band_class_init, gst_iir_equalizer_band_get_type,
gst_iir_equalizer_child_proxy_get_child_by_index,
gst_iir_equalizer_child_proxy_get_children_count,
gst_iir_equalizer_child_proxy_interface_init, setup_filter,
gst_iir_equalizer_compute_frequencies,
gst_iir_equalizer_set_property, gst_iir_equalizer_get_property,
plugin_init):
* gst/equalizer/gstiirequalizer.h (audiofilter):
* gst/equalizer/gstiirequalizernbands.c (ARG_NUM_BANDS,
gst_iir_equalizer_nbands_base_init, gst_iir_equalizer_nbands_init,
gst_iir_equalizer_nbands_set_property):
Use new locking macros.
* gst/filter/gstbpwsinc.c (bpwsinc_set_caps):
Add fixme.
* gst/spectrum/gstspectrum.c (SPECTRUM_WINDOW_BASE,
SPECTRUM_WINDOW_LEN, gst_spectrum_init, gst_spectrum_set_property,
gst_spectrum_event, gst_spectrum_transform_ip):
Use new locking macros. Turn two fixed values into #defines.
Edward Hervey [Tue, 22 May 2007 11:03:30 +0000 (11:03 +0000)]
docs/plugins/Makefile.am: Also look for .m (objectivec) files.
Original commit message from CVS:
* docs/plugins/Makefile.am:
Also look for .m (objectivec) files.
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* sys/osxvideo/osxvideosink.m:
Add documentation for element and properties.
Stefan Kost [Mon, 21 May 2007 14:01:16 +0000 (14:01 +0000)]
ChangeLog: ChangeLog surgery. gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _GstIirEqualizerBa...
Original commit message from CVS:
* ChangeLog:
ChangeLog surgery.
* gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN,
_GstIirEqualizerBand, object, _GstIirEqualizerBandClass,
parent_class, gst_iir_equalizer_band_set_property,
gst_iir_equalizer_band_class_init, gst_iir_equalizer_band_get_type,
gst_iir_equalizer_child_proxy_get_child_by_index,
gst_iir_equalizer_child_proxy_get_children_count,
gst_iir_equalizer_child_proxy_interface_init, setup_filter,
gst_iir_equalizer_compute_frequencies, plugin_init):
* tests/icles/equalizer-test.c:
Add fixme and comment for example.
Stefan Kost [Mon, 21 May 2007 12:43:37 +0000 (12:43 +0000)]
gst/spectrum/gstspectrum.c (gst_spectrum_set_property, gst_spectrum_event, gst_spectrum_transform_ip):
Original commit message from CVS:
* gst/spectrum/gstspectrum.c (gst_spectrum_set_property,
gst_spectrum_event, gst_spectrum_transform_ip):
Use lock to protect from concurrent access.
Sebastian Dröge [Mon, 21 May 2007 11:37:16 +0000 (11:37 +0000)]
ext/wavpack/gstwavpackenc.c: Specify and use properties as unsigned int that are an unsigned int.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
(gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property):
Specify and use properties as unsigned int that are an unsigned int.
Sebastian Dröge [Mon, 21 May 2007 11:17:21 +0000 (11:17 +0000)]
ext/wavpack/gstwavpackenc.*: Fixup docs, make the bitrate property an int as it should be and allow to set the differ...
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
(gst_wavpack_enc_init), (gst_wavpack_enc_set_wp_config),
(gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property):
* ext/wavpack/gstwavpackenc.h:
Fixup docs, make the bitrate property an int as it should be and
allow to set the different extra processing modes instead of only
allowing none and the default one.
Sebastian Dröge [Sun, 20 May 2007 21:31:58 +0000 (21:31 +0000)]
ext/wavpack/gstwavpackenc.c: Add missing audioconverts in the example pipelines of wavpackenc. As the wavpack stuff n...
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c:
Add missing audioconverts in the example pipelines of wavpackenc. As
the wavpack stuff now needs input with 32 bit width (and random depth)
this is needed now. The example pipelines for the parser and decoder
are still fine.
Thomas Vander Stichele [Sun, 20 May 2007 15:12:31 +0000 (15:12 +0000)]
docs/plugins/: Fix docs build and hierarchy.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
Fix docs build and hierarchy.
Tim-Philipp Müller [Sun, 20 May 2007 14:59:46 +0000 (14:59 +0000)]
sys/directdraw/gstdirectdrawsink.c: Bunch of small fixes: remove static function that doesn't exist; declare another ...
Original commit message from CVS:
* sys/directdraw/gstdirectdrawsink.c: (gst_ddrawsurface_finalize),
(gst_directdraw_sink_buffer_alloc),
(gst_directdraw_sink_get_ddrawcaps),
(gst_directdraw_sink_surface_create):
Bunch of small fixes: remove static function that doesn't exist;
declare another one that does; printf format fix; use right macro
when specifying debug category; remove a bunch of unused variables;
#if 0 out an unused chunk of code (partially fixes #439914).
Tim-Philipp Müller [Sun, 20 May 2007 14:14:49 +0000 (14:14 +0000)]
gst/: Printf format fixes (#439910, #439911).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample):
* gst/switch/gstswitch.c: (gst_switch_chain):
Printf format fixes (#439910, #439911).
Tim-Philipp Müller [Sat, 19 May 2007 10:06:02 +0000 (10:06 +0000)]
tests/check/Makefile.am: Remove bits for deinterleave check which isn't in CVS yet.
Original commit message from CVS:
* tests/check/Makefile.am:
Remove bits for deinterleave check which isn't in CVS yet.
René Stadler [Sat, 19 May 2007 10:01:45 +0000 (10:01 +0000)]
Add replaygain playback elements (#412710).
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/inspect/plugin-replaygain.xml:
* gst/replaygain/Makefile.am:
* gst/replaygain/gstrganalysis.c: (gst_rg_analysis_class_init),
(gst_rg_analysis_start), (gst_rg_analysis_set_caps),
(gst_rg_analysis_transform_ip), (gst_rg_analysis_event),
(gst_rg_analysis_stop), (gst_rg_analysis_handle_tags),
(gst_rg_analysis_handle_eos), (gst_rg_analysis_track_result),
(gst_rg_analysis_album_result):
* gst/replaygain/gstrganalysis.h:
* gst/replaygain/gstrglimiter.c: (gst_rg_limiter_base_init),
(gst_rg_limiter_class_init), (gst_rg_limiter_init),
(gst_rg_limiter_set_property), (gst_rg_limiter_get_property),
(gst_rg_limiter_transform_ip):
* gst/replaygain/gstrglimiter.h:
* gst/replaygain/gstrgvolume.c: (gst_rg_volume_base_init),
(gst_rg_volume_class_init), (gst_rg_volume_init),
(gst_rg_volume_set_property), (gst_rg_volume_get_property),
(gst_rg_volume_dispose), (gst_rg_volume_change_state),
(gst_rg_volume_sink_event), (gst_rg_volume_tag_event),
(gst_rg_volume_reset), (gst_rg_volume_update_gain),
(gst_rg_volume_determine_gain):
* gst/replaygain/gstrgvolume.h:
* gst/replaygain/replaygain.c: (plugin_init):
* gst/replaygain/replaygain.h:
* gst/replaygain/rganalysis.h:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/rganalysis.c: (send_eos_event),
(GST_START_TEST):
* tests/check/elements/rglimiter.c: (setup_rglimiter),
(cleanup_rglimiter), (set_playing_state), (create_test_buffer),
(verify_test_buffer), (GST_START_TEST), (rglimiter_suite), (main):
* tests/check/elements/rgvolume.c: (event_func), (setup_rgvolume),
(cleanup_rgvolume), (set_playing_state), (set_null_state),
(send_eos_event), (send_tag_event), (test_buffer_new),
(fail_unless_target_gain), (fail_unless_result_gain),
(fail_unless_gain), (GST_START_TEST), (rgvolume_suite), (main):
Add replaygain playback elements (#412710).
Christian Schaller [Fri, 18 May 2007 16:45:54 +0000 (16:45 +0000)]
update
Original commit message from CVS:
update
Jan Schmidt [Fri, 18 May 2007 12:39:49 +0000 (12:39 +0000)]
sys/glsink/glimagesink.c: Update the cached caps after opening the display so that we report only the supported caps ...
Original commit message from CVS:
* sys/glsink/glimagesink.c: (gst_glimage_sink_init_display):
Update the cached caps after opening the display so that we report
only the supported caps formats, not just the template caps.
Fixes: #439405
Stefan Kost [Fri, 18 May 2007 09:35:28 +0000 (09:35 +0000)]
ext/amrwb/gstamrwbdec.c (gst_amrwbdec_debug, GST_CAT_DEFAULT, _do_init, gst_amrwbdec_base_init, gst_amrwbdec_class_in...
Original commit message from CVS:
* ext/amrwb/gstamrwbdec.c (gst_amrwbdec_debug, GST_CAT_DEFAULT,
_do_init, gst_amrwbdec_base_init, gst_amrwbdec_class_init):
* ext/amrwb/gstamrwbenc.c (gst_amrwbenc_debug, GST_CAT_DEFAULT,
_do_init, gst_amrwbenc_base_init, gst_amrwbenc_class_init):
* ext/amrwb/gstamrwbparse.c (gst_amrwbparse_debug, GST_CAT_DEFAULT,
_do_init, gst_amrwbparse_base_init, gst_amrwbparse_class_init):
First round of cleanups, that use GST_BOILERPLATE, GST_ELEMENT_DETAILS,
GST_DEBUG_FUNCPTR and add log-category.
Edward Hervey [Thu, 17 May 2007 10:59:00 +0000 (10:59 +0000)]
sys/osxvideo/osxvideosink.*: Remove the event-loop-in-separate-thread modifications, because MacOSX is $#@(*%$# ! For...
Original commit message from CVS:
* sys/osxvideo/osxvideosink.h:
* sys/osxvideo/osxvideosink.m:
Remove the event-loop-in-separate-thread modifications, because MacOSX
is $#@(*%$# ! For those wondering, the event handling needs to be done
in the main thread after all..
Zaheer Abbas Merali [Thu, 17 May 2007 10:08:24 +0000 (10:08 +0000)]
gst/switch/gstswitch.c (ARG_0, ARG_NB_SOURCES, ARG_ACTIVE_SOURCE,
Original commit message from CVS:
* gst/switch/gstswitch.c (ARG_0, ARG_NB_SOURCES, ARG_ACTIVE_SOURCE,
ARG_START_VALUE, ARG_STOP_VALUE, ARG_LAST_TS, ARG_QUEUE_BUFFERS,
parent_class, gst_switch_release_pad, gst_switch_request_new_pad,
gst_switch_chain, gst_switch_event, gst_switch_set_property,
gst_switch_get_property, gst_switch_getcaps, gst_switch_dispose,
unref_buffer, unref_buffers_and_destroy_list, gst_switch_init,
gst_switch_base_init, gst_switch_class_init):
* gst/switch/gstswitch.h (need_to_send_newsegment, queue_buffers,
stop_value, start_value, current_start, last_ts, stored_buffers):
Add handling of application provided stop and start values, allowing
A/V sync across 2 switch elements.
Edward Hervey [Thu, 17 May 2007 09:41:48 +0000 (09:41 +0000)]
sys/osxvideo/osxvideosink.*: Fix a stupid #if vs #ifdef bug. Should use the proper colorspace now.
Original commit message from CVS:
* sys/osxvideo/osxvideosink.h:
* sys/osxvideo/osxvideosink.m:
Fix a stupid #if vs #ifdef bug. Should use the proper colorspace now.
Use a separate thread/task for the cocoa event_loop, else it wouldn't
stop.
Tim-Philipp Müller [Wed, 16 May 2007 17:14:43 +0000 (17:14 +0000)]
gst/real/: Don't crash when we get a buffer and our input caps haven't been set yet; also, don't leak all the input b...
Original commit message from CVS:
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_chain),
(gst_real_audio_dec_setcaps):
* gst/real/gstrealvideodec.c: (gst_real_video_dec_chain):
Don't crash when we get a buffer and our input caps haven't been set
yet; also, don't leak all the input buffers (realaudiodec only).
Stanislav Brabec [Wed, 16 May 2007 13:50:51 +0000 (13:50 +0000)]
Use external shared libamrwb. Fixes #423741 (with lots of cleanup).
Original commit message from CVS:
patch by: Stanislav Brabec <sbrabec@suse.cz>
* configure.ac:
* ext/amrwb/Makefile.am:
* ext/amrwb/amrwb-code/Makefile.am:
* ext/amrwb/amrwb-code/amrwb/Makefile.am:
* ext/amrwb/amrwb-code/amrwb/README:
* ext/amrwb/gstamrwbdec.c:
* ext/amrwb/gstamrwbdec.h (__GST_AMRWBDEC_H__):
* ext/amrwb/gstamrwbenc.h (__GST_AMRWBENC_H__):
* ext/amrwb/gstamrwbparse.c:
* ext/amrwb/gstamrwbparse.h (__GST_AMRWBPARSE_H__):
* gst-libs/Makefile.am:
* gst-libs/ext/Makefile.am:
* gst-libs/ext/amrwb/Makefile.am:
* gst-libs/ext/amrwb/README:
Use external shared libamrwb. Fixes #423741 (with lots of cleanup).
Stefan Kost [Tue, 15 May 2007 21:23:53 +0000 (21:23 +0000)]
ext/x264/gstx264enc.c (gst_x264_enc_init_encoder): This needs a version check.
Original commit message from CVS:
* ext/x264/gstx264enc.c (gst_x264_enc_init_encoder):
This needs a version check.
* gst/bayer/Makefile.am:
Fix the build.
Tim-Philipp Müller [Tue, 15 May 2007 17:22:58 +0000 (17:22 +0000)]
Add DIRECTDRAW_CFLAGS and DIRECTSOUND_CFLAGS to Makefile.am; save and restore the various flags in the directdraw/dir...
Original commit message from CVS:
* configure.ac:
* sys/directdraw/Makefile.am:
* sys/directsound/Makefile.am:
Add DIRECTDRAW_CFLAGS and DIRECTSOUND_CFLAGS to Makefile.am; save
and restore the various flags in the directdraw/directsound
detection section. Apparently improves cross-compiling for win32
with mingw32 under some circumstances (#437539).
Wim Taymans [Tue, 15 May 2007 13:29:53 +0000 (13:29 +0000)]
gst/rtpmanager/gstrtpptdemux.c: We always use fixed caps.
Original commit message from CVS:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
We always use fixed caps.
David Schleef [Tue, 15 May 2007 03:45:45 +0000 (03:45 +0000)]
gst/rtpmanager/gstrtpbin.c: g_hash_table_remove_all() only exists in 2.12. Work around.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
g_hash_table_remove_all() only exists in 2.12. Work around.
David Schleef [Tue, 15 May 2007 02:54:52 +0000 (02:54 +0000)]
Add a Bayer-to-RGB converter. You know you want one, uh-huh.
Original commit message from CVS:
* configure.ac:
* gst/bayer/Makefile.am:
* gst/bayer/gstbayer.c:
* gst/bayer/gstbayer2rgb.c:
Add a Bayer-to-RGB converter. You know you want one, uh-huh.
Partial fix for #314160.
Zaheer Abbas Merali [Mon, 14 May 2007 18:03:58 +0000 (18:03 +0000)]
gst/switch/gstswitch.c (ARG_ACTIVE_SOURCE, ARG_STOP_VALUE,
Original commit message from CVS:
* gst/switch/gstswitch.c (ARG_ACTIVE_SOURCE, ARG_STOP_VALUE,
ARG_LAST_TS, parent_class, gst_switch_release_pad,
gst_switch_request_new_pad, gst_switch_chain, gst_switch_event,
gst_switch_set_property, gst_switch_get_property, gst_switch_getcaps,
gst_switch_dispose, gst_switch_init, gst_switch_class_init):
* gst/switch/gstswitch.h (previous_sinkpad, nb_sinkpads, stop_value,
current_start, last_ts):
Allow application to provide a stop timestamp, so a new segment
update can be sent before switching.
Wim Taymans [Mon, 14 May 2007 15:28:36 +0000 (15:28 +0000)]
gst/rtpmanager/async_jitter_queue.c: Fix leak when flushing.
Original commit message from CVS:
* gst/rtpmanager/async_jitter_queue.c:
(async_jitter_queue_set_flushing_unlocked):
Fix leak when flushing.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
Add clear-pt-map signal.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_loop):
Init clock-rate to -1 to mark unknow clock rate.
Fix flushing.
David Schleef [Sun, 13 May 2007 19:57:45 +0000 (19:57 +0000)]
gst/replaygain/rganalysis.c: Fix wrong ifdef for visual C++. Fixes: #437403.
Original commit message from CVS:
* gst/replaygain/rganalysis.c:
Fix wrong ifdef for visual C++. Fixes: #437403.
By Ali Sabil <ali.sabil@gmail.com>.
David Schleef [Sun, 13 May 2007 07:08:06 +0000 (07:08 +0000)]
ext/neon/gstneonhttpsrc.c: Make redirection the default behavior. Fixes #413818.
Original commit message from CVS:
* ext/neon/gstneonhttpsrc.c:
Make redirection the default behavior. Fixes #413818.
Christian Schaller [Fri, 11 May 2007 11:20:33 +0000 (11:20 +0000)]
add latest plugin
Original commit message from CVS:
add latest plugin
Stefan Kost [Thu, 10 May 2007 14:02:07 +0000 (14:02 +0000)]
gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtde...
Original commit message from CVS:
* gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream,
gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment,
gst_qtdemux_prepare_current_sample, gst_qtdemux_combine_flows,
gst_qtdemux_loop_state_movie, gst_qtdemux_loop,
qtdemux_parse_segments, qtdemux_parse_trak):
* gst/rtpmanager/rtpsession.c (rtp_session_get_bandwidth,
rtp_session_get_rtcp_bandwidth, rtp_session_get_cname,
rtp_session_get_name, rtp_session_get_email, rtp_session_get_phone,
rtp_session_get_location, rtp_session_get_tool,
rtp_session_process_bye, session_report_blocks):
* gst/rtpmanager/rtpsource.c (rtp_source_process_rtp,
rtp_source_send_rtp, rtp_source_process_sr, rtp_source_process_rb):
More format arg fixing (spotted by Ali Sabil <ali.sabil@gmail.com>).
* gst/switch/Makefile.am:
Add require libraries(spotted by Ali Sabil <ali.sabil@gmail.com>).
Stefan Kost [Thu, 10 May 2007 12:38:49 +0000 (12:38 +0000)]
gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration, async_jitter_queue_ref, async_jitter_queue_ref_unlocked, a...
Original commit message from CVS:
* gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration,
async_jitter_queue_ref, async_jitter_queue_ref_unlocked,
async_jitter_queue_set_low_threshold,
async_jitter_queue_length_ts_units_unlocked,
async_jitter_queue_unref_and_unlock, async_jitter_queue_unref,
async_jitter_queue_lock, async_jitter_queue_push,
async_jitter_queue_push_unlocked, async_jitter_queue_push_sorted,
async_jitter_queue_pop_intern_unlocked, async_jitter_queue_pop,
async_jitter_queue_pop_unlocked, async_jitter_queue_length_unlocked,
async_jitter_queue_set_flushing_unlocked,
async_jitter_queue_unset_flushing_unlocked):
Format arg fix (spotted by Ali Sabil <ali.sabil@gmail.com>)
Stefan Kost [Thu, 10 May 2007 07:41:47 +0000 (07:41 +0000)]
common/m4/gst-x11.m4: Restore CFLAGS and LIBS.
Original commit message from CVS:
* common/m4/gst-x11.m4:
Restore CFLAGS and LIBS.
* configure.ac:
Revert previous patch.
Ali Sabil [Thu, 10 May 2007 06:35:31 +0000 (06:35 +0000)]
configure.ac: Save and restore CFLAGS for OpenGL check. Fixes #437260.
Original commit message from CVS:
Patch by: Ali Sabil <ali.sabil@gmail.com>
* configure.ac:
Save and restore CFLAGS for OpenGL check. Fixes #437260.
Sébastien Moutte [Wed, 9 May 2007 20:01:21 +0000 (20:01 +0000)]
docs/plugins/gst-plugins-bad-plugins.args: Add directraw and directsound sinks properties.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
Add directraw and directsound sinks properties.
Tim-Philipp Müller [Wed, 9 May 2007 17:58:07 +0000 (17:58 +0000)]
configure.ac: Fix --disable-external (hopefully).
Original commit message from CVS:
* configure.ac:
Fix --disable-external (hopefully).
Wim Taymans [Wed, 9 May 2007 11:24:22 +0000 (11:24 +0000)]
gst/rtpmanager/gstrtpjitterbuffer.c: Pass queries upstream.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_query):
Pass queries upstream.
Tim-Philipp Müller [Sun, 6 May 2007 15:25:05 +0000 (15:25 +0000)]
gst/real/: Use GModule instead of using dlsym() directly. Fixes #430598.
Original commit message from CVS:
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps),
(gst_real_audio_dec_finalize):
* gst/real/gstrealaudiodec.h:
* gst/real/gstrealvideodec.c: (open_library), (close_library):
* gst/real/gstrealvideodec.h:
Use GModule instead of using dlsym() directly. Fixes #430598.
Sébastien Moutte [Fri, 4 May 2007 21:02:58 +0000 (21:02 +0000)]
docs/plugins/: Add docs for Windows sinks.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
Add docs for Windows sinks.
Tim-Philipp Müller [Fri, 4 May 2007 17:20:31 +0000 (17:20 +0000)]
gst/speed/gstspeed.c: Fix event handling a bit by replacing completely dubious code written by someone else with comp...
Original commit message from CVS:
* gst/speed/gstspeed.c: (speed_src_event), (speed_sink_event),
(speed_chain), (speed_change_state):
Fix event handling a bit by replacing completely dubious code
written by someone else with completely dubious code written
by me. Should at least fix #412077 though.
Tim-Philipp Müller [Fri, 4 May 2007 16:11:46 +0000 (16:11 +0000)]
gst/speed/gstspeed.c: Add debug category; use gst_pad_query_peer_*() utility functions; use gst_util_scale*(); add gt...
Original commit message from CVS:
* gst/speed/gstspeed.c: (speed_src_query), (speed_chain),
(plugin_init):
Add debug category; use gst_pad_query_peer_*() utility functions;
use gst_util_scale*(); add gtk-doc blurb.
Wim Taymans [Fri, 4 May 2007 12:32:27 +0000 (12:32 +0000)]
gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug info.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_query):
Add some debug info.
* gst/rtpmanager/rtpsession.c: (rtp_session_init),
(rtp_session_send_rtp):
Store real user name in the session.
Zaheer Abbas Merali [Thu, 3 May 2007 16:49:05 +0000 (16:49 +0000)]
examples/switch/switcher.c (loop, my_bus_callback, switch_timer, last_message_received, main): gst/switch/gstswitch.c...
Original commit message from CVS:
* configure.ac:
* examples/Makefile.am:
* examples/switch/switcher.c (loop, my_bus_callback, switch_timer,
last_message_received, main):
* gst/switch/gstswitch.c (GST_CAT_DEFAULT, gst_switch_details,
gst_switch_src_factory, parent_class, gst_switch_release_pad,
gst_switch_request_new_pad, gst_switch_chain, gst_switch_event,
gst_switch_set_property, gst_switch_get_property,
gst_switch_get_linked_pad, gst_switch_getcaps,
gst_switch_bufferalloc, gst_switch_get_linked_pads,
gst_switch_dispose, gst_switch_init, gst_switch_base_init,
gst_switch_class_init):
* gst/switch/gstswitch.h (GstSwitch, GstSwitchClass, _GstSwitch,
element, active_sinkpad, srcpad, nb_sinkpads, newsegment_events,
need_to_send_newsegment):
Port switch element and example program to 0.10.
Sebastian Dröge [Wed, 2 May 2007 18:31:16 +0000 (18:31 +0000)]
ext/wavpack/gstwavpack.c: Call bindtextdomain() to get localized strings.
Original commit message from CVS:
* ext/wavpack/gstwavpack.c: (plugin_init):
Call bindtextdomain() to get localized strings.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
(gst_wavpack_parse_handle_seek_event),
(gst_wavpack_parse_push_buffer), (gst_wavpack_parse_chain):
* ext/wavpack/gstwavpackparse.h:
Handle DISCONT buffers by correctly setting the DISCONT flag
on outgoing buffers when necessary.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_handle_seek_event)
Send newsegment from the streaming thread.
Sebastian Dröge [Wed, 2 May 2007 16:58:06 +0000 (16:58 +0000)]
ext/wavpack/gstwavpackparse.c: Remove old workaround that was needed when seeking after the last sample. With the fix...
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_handle_seek_event):
Remove old workaround that was needed when seeking after the last
sample. With the fixed error handling this works now as expected
without pushing the last sample although it wasn't requested.
Sebastian Dröge [Wed, 2 May 2007 16:45:43 +0000 (16:45 +0000)]
ext/wavpack/gstwavpackparse.c: Handle segment seeks in the seek event handler, correctly work with stop position == -...
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_handle_seek_event):
Handle segment seeks in the seek event handler, correctly work with
stop position == -1 and instead of stopping the task on seek just
pause it.
Sebastian Dröge [Wed, 2 May 2007 16:19:58 +0000 (16:19 +0000)]
ext/wavpack/gstwavpackparse.c: Add handling for segment seeks.
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_loop):
Add handling for segment seeks.
Sebastian Dröge [Wed, 2 May 2007 15:13:04 +0000 (15:13 +0000)]
ext/wavpack/gstwavpackparse.c: Correctly handle errors, especially in the loop function. Before it was easy to get th...
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_pull_buffer),
(gst_wavpack_parse_create_src_pad),
(gst_wavpack_parse_resync_loop), (gst_wavpack_parse_loop),
(gst_wavpack_parse_chain):
Correctly handle errors, especially in the loop function. Before it
was easy to get the task paused but no error being posted on the bus.
Christian Schaller [Mon, 30 Apr 2007 15:39:09 +0000 (15:39 +0000)]
update spec
Original commit message from CVS:
update spec
Wim Taymans [Mon, 30 Apr 2007 13:41:30 +0000 (13:41 +0000)]
gst/rtpmanager/async_jitter_queue.c: Fix the case where the buffer underruns and does not block.
Original commit message from CVS:
* gst/rtpmanager/async_jitter_queue.c: (signal_waiting_threads),
(async_jitter_queue_pop_intern_unlocked):
Fix the case where the buffer underruns and does not block.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
(create_recv_rtcp), (create_send_rtp), (create_rtcp),
(gst_rtp_bin_request_new_pad):
Rename RTCP send pad, like in the session manager.
Allow getting an RTCP pad for receiving even if we don't receive RTP.
fix handling of send_rtp_src pad.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
When no pt map could be found, fall back to the sinkpad caps.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink),
(create_send_rtcp_src):
Fix pad names.
* gst/rtpmanager/rtpsession.c: (source_push_rtp),
(rtp_session_create_source), (rtp_session_process_sr),
(rtp_session_send_rtp), (session_start_rtcp):
* gst/rtpmanager/rtpsession.h:
Unlock session when performing a callback.
Add callbacks for the internal session object.
Fix sending of RTP packets.
first attempt at adding NTP times in the SR packets.
Small debug and doc improvements.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
Update stats for SR reports.
Wim Taymans [Sun, 29 Apr 2007 14:46:27 +0000 (14:46 +0000)]
gst/rtpmanager/gstrtpsession.c: Remove debug.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp):
Remove debug.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_sr),
(rtp_session_process_sdes), (calculate_rtcp_interval),
(rtp_session_next_timeout), (session_report_blocks):
* gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
Improve debugging
Fix interval for BYE/RTCP packets.
Thomas Vander Stichele [Sun, 29 Apr 2007 14:01:05 +0000 (14:01 +0000)]
docs/plugins/gst-plugins-bad-plugins.*: Commit result of running scanobj-update
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
Commit result of running scanobj-update
Thomas Vander Stichele [Sun, 29 Apr 2007 13:56:18 +0000 (13:56 +0000)]
80 char police
Original commit message from CVS:
80 char police
Thomas Vander Stichele [Sun, 29 Apr 2007 13:53:17 +0000 (13:53 +0000)]
autogen.sh: Require automake 1.7
Original commit message from CVS:
* autogen.sh:
Require automake 1.7
* ext/alsaspdif/Makefile.am:
* ext/divx/Makefile.am:
* ext/ivorbis/Makefile.am:
* ext/musicbrainz/Makefile.am:
* ext/neon/Makefile.am:
* ext/sdl/Makefile.am:
* ext/swfdec/Makefile.am:
* ext/theora/Makefile.am:
* ext/wavpack/Makefile.am:
* ext/xvid/Makefile.am:
* gst/modplug/Makefile.am:
Fix up Makefile.am accordingly.
Thomas Vander Stichele [Sun, 29 Apr 2007 13:49:02 +0000 (13:49 +0000)]
docs/plugins/inspect/: Add jack and update.
Original commit message from CVS:
* docs/plugins/inspect/plugin-alsaspdif.xml:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dfbvideosink.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-glimagesink.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-mms.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-neon.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-sdl.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-videocrop.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
* docs/plugins/inspect/plugin-xingheader.xml:
* docs/plugins/inspect/plugin-xvid.xml:
Add jack and update.
Tim-Philipp Müller [Sat, 28 Apr 2007 12:46:47 +0000 (12:46 +0000)]
configure.ac: Don't build equalizer unless we have core from CVS (it won't work with earlier versions due to GstChild...
Original commit message from CVS:
* configure.ac:
Don't build equalizer unless we have core from CVS (it won't
work with earlier versions due to GstChildProxy brokeness).
Also up requirements to last released core/base.
Julien Moutte [Fri, 27 Apr 2007 15:33:46 +0000 (15:33 +0000)]
ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888).
Original commit message from CVS:
2007-04-27 Julien MOUTTE <julien@moutte.net>
* ext/theora/theoradec.c: (_theora_granule_time),
(theora_dec_push_forward), (theora_handle_data_packet),
(theora_dec_decode_buffer): Calculate buffer duration correctly
to generate a perfect stream (#433888).
* gst/audioresample/gstaudioresample.c:
(audioresample_check_discont): Glib provides ABS.
Wim Taymans [Fri, 27 Apr 2007 15:09:12 +0000 (15:09 +0000)]
gst/rtpmanager/gstrtpsession.c: Move reconsideration code to the rtpsession object.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(gst_rtp_session_send_rtcp), (gst_rtp_session_reconsider):
Move reconsideration code to the rtpsession object.
Simplify timout handling and add reconsideration.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize), (on_bye_ssrc),
(on_bye_timeout), (on_timeout), (rtp_session_set_callbacks),
(obtain_source), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_bye), (rtp_session_process_rtcp),
(calculate_rtcp_interval), (rtp_session_send_bye),
(rtp_session_next_timeout), (session_start_rtcp),
(session_report_blocks), (session_cleanup), (session_sdes),
(session_bye), (is_rtcp_time), (rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Handle timeout of inactive sources and senders.
Implement BYE scheduling.
* gst/rtpmanager/rtpsource.c: (calculate_jitter),
(rtp_source_process_sr), (rtp_source_get_last_sr),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
Add members to check for timeouts.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter),
(rtp_stats_calculate_bye_interval):
* gst/rtpmanager/rtpstats.h:
Use RFC algorithm for calculating the reporting interval.
Edward Hervey [Thu, 26 Apr 2007 14:31:32 +0000 (14:31 +0000)]
docs/plugins/: Add documentation for osxvideo
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/inspect/plugin-osxvideo.xml:
Add documentation for osxvideo
Wim Taymans [Wed, 25 Apr 2007 16:38:03 +0000 (16:38 +0000)]
gst/rtpmanager/gstrtpsession.c: Implement forward and reverse reconsideration.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
Implement forward and reverse reconsideration.
* gst/rtpmanager/rtpsession.c: (rtp_session_get_num_sources),
(rtp_session_get_num_active_sources), (rtp_session_process_sr),
(session_report_blocks):
* gst/rtpmanager/rtpsession.h:
Small cleanups.
Wim Taymans [Wed, 25 Apr 2007 15:48:46 +0000 (15:48 +0000)]
gst/rtpmanager/gstrtpbin.*: Make default jitterbuffer latency configurable.
Original commit message from CVS:
reviewed by: <delete if not using a buddy>
* gst/rtpmanager/gstrtpbin.c: (create_stream),
(gst_rtp_bin_class_init), (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property):
* gst/rtpmanager/gstrtpbin.h:
Make default jitterbuffer latency configurable.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Debuging cleanups.
Wim Taymans [Wed, 25 Apr 2007 13:19:36 +0000 (13:19 +0000)]
gst/rtpmanager/gstrtpjitterbuffer.c: Report NO_PREROLL when going to PAUSED.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_change_state):
Report NO_PREROLL when going to PAUSED.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
Don't send RTCP right before we are shutting down.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_rtp),
(rtp_session_process_sr), (session_report_blocks),
(rtp_session_perform_reporting):
Improve report blocks.
* gst/rtpmanager/rtpsource.c: (calculate_jitter), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr),
(rtp_source_process_rb), (rtp_source_get_last_sr),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Cleanups, add methods to access stats.
Wim Taymans [Wed, 25 Apr 2007 08:30:48 +0000 (08:30 +0000)]
gst/rtpmanager/gstrtpbin.c: fix for pad name change
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_rtcp):
fix for pad name change
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate):
Fix for renamed methods.
* gst/rtpmanager/rtpsession.c: (rtp_session_init),
(rtp_session_finalize), (rtp_session_set_cname),
(rtp_session_get_cname), (rtp_session_set_name),
(rtp_session_get_name), (rtp_session_set_email),
(rtp_session_get_email), (rtp_session_set_phone),
(rtp_session_get_phone), (rtp_session_set_location),
(rtp_session_get_location), (rtp_session_set_tool),
(rtp_session_get_tool), (rtp_session_set_note),
(rtp_session_get_note), (source_push_rtp), (obtain_source),
(rtp_session_add_source), (rtp_session_get_source_by_ssrc),
(rtp_session_create_source), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_sdes),
(rtp_session_process_rtcp), (rtp_session_send_rtp),
(rtp_session_get_reporting_interval), (session_report_blocks),
(session_sdes), (rtp_session_perform_reporting):
* gst/rtpmanager/rtpsession.h:
Prepare for implementing SSRC sampling.
Create SSRC for the session.
Add methods to set the SDES entries.
fix accounting of senders/receivers.
Implement SR/RR/SDES RTCP reporting.
* gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr):
* gst/rtpmanager/rtpsource.h:
Implement extended sequence number.
* gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
* gst/rtpmanager/rtpstats.h:
Rename some fields.
Tim-Philipp Müller [Tue, 24 Apr 2007 15:49:18 +0000 (15:49 +0000)]
Plug some leaks; try to make build bot happy again.
Original commit message from CVS:
* gst/y4m/gsty4mencode.c: (gst_y4m_encode_init),
(gst_y4m_encode_setcaps):
* tests/check/elements/y4menc.c: (GST_START_TEST):
Plug some leaks; try to make build bot happy again.
Tim-Philipp Müller [Sat, 21 Apr 2007 19:39:03 +0000 (19:39 +0000)]
gst/Makefile.am: Fix distcheck, hopefully (rtpmanager is already in GST_PLUGINS_ALL).
Original commit message from CVS:
* gst/Makefile.am:
Fix distcheck, hopefully (rtpmanager is already in GST_PLUGINS_ALL).
Tim-Philipp Müller [Sat, 21 Apr 2007 19:21:49 +0000 (19:21 +0000)]
gst/rtpmanager/rtpsession.c: Don't use GLib-2.10 API, we only require GLib 2.8 at the moment.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_finalize):
Don't use GLib-2.10 API, we only require GLib 2.8 at the moment.
Tim-Philipp Müller [Sat, 21 Apr 2007 14:14:24 +0000 (14:14 +0000)]
gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Make more functions static, just because we can.
Tim-Philipp Müller [Sat, 21 Apr 2007 13:54:39 +0000 (13:54 +0000)]
tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).
Original commit message from CVS:
* tests/check/elements/audioresample.c:
Add unit test for audioresample shutdown crasher (#420106).
Michael Smith [Fri, 20 Apr 2007 15:31:32 +0000 (15:31 +0000)]
ext/faad/gstfaad.c: FAAD fails to decode low (e.g. 8 kHz) sample rate AAC data in quicktime because of sample rate mi...
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_open_decoder):
FAAD fails to decode low (e.g. 8 kHz) sample rate AAC data in
quicktime because of sample rate mismatches.
Reenable overriding the implicit SBR behaviour (accidently changed?)
to allow playback of these files.
David Schleef [Thu, 19 Apr 2007 15:43:26 +0000 (15:43 +0000)]
configure.ac: Change rtpmanager disabling to keep -bad releasable.
Original commit message from CVS:
* configure.ac:
Change rtpmanager disabling to keep -bad releasable.
David Schleef [Wed, 18 Apr 2007 19:45:32 +0000 (19:45 +0000)]
Fix wtay's hack. rtpmanager is disabled in configure.ac on line 268.
Original commit message from CVS:
* configure.ac:
* gst/Makefile.am:
Fix wtay's hack. rtpmanager is disabled in configure.ac on
line 268.
Wim Taymans [Wed, 18 Apr 2007 19:26:52 +0000 (19:26 +0000)]
gst/Makefile.am: Add rtpmanager dir to dist.
Original commit message from CVS:
* gst/Makefile.am:
Add rtpmanager dir to dist.
Wim Taymans [Wed, 18 Apr 2007 18:58:53 +0000 (18:58 +0000)]
configure.ac: Disable rtpmanager for now because it depends on CVS -base.
Original commit message from CVS:
* configure.ac:
Disable rtpmanager for now because it depends on CVS -base.
* gst/rtpmanager/Makefile.am:
Added new files for session manager.
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (pt_map_requested), (new_ssrc_pad_found):
Some cleanups.
the session manager can now also request a pt-map.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate),
(gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_send_rtcp_src),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpsession.h:
We can ask for pt-map now too when the session manager needs it.
Hook up to the new session manager, implement the needed callbacks for
pushing data, getting clock time and requesting clock-rates.
Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to
be send to clients.
Add code to start and stop the thread that will schedule RTCP through
the session manager.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated),
(on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks),
(rtp_session_set_bandwidth), (rtp_session_get_bandwidth),
(rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth),
(source_push_rtp), (source_clock_rate), (check_collision),
(obtain_source), (rtp_session_add_source),
(rtp_session_get_num_sources),
(rtp_session_get_num_active_sources),
(rtp_session_get_source_by_ssrc),
(rtp_session_get_source_by_cname), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye),
(rtp_session_process_app), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_get_rtcp_interval),
(rtp_session_produce_rtcp):
* gst/rtpmanager/rtpsession.h:
The advanced beginnings of the main session manager that handles the
participant database of RTPSources, SSRC probation, SSRC collisions,
parse RTCP to update source stats. etc..
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_finalize), (rtp_source_new),
(rtp_source_set_callbacks), (rtp_source_set_as_csrc),
(rtp_source_set_rtp_from), (rtp_source_set_rtcp_from),
(push_packet), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_process_bye),
(rtp_source_send_rtp), (rtp_source_process_sr),
(rtp_source_process_rb):
* gst/rtpmanager/rtpsource.h:
Object that encapsulates an SSRC and its state in the database.
Calculates the jitter and transit times of data packets.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter):
* gst/rtpmanager/rtpstats.h:
Various stats regarding the session and sources.
Used to calculate the RTCP interval.
Tim-Philipp Müller [Tue, 17 Apr 2007 10:56:37 +0000 (10:56 +0000)]
gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
Original commit message from CVS:
* gst/app/Makefile.am:
Fix CFLAGS and hopefully #430594.
Tim-Philipp Müller [Tue, 17 Apr 2007 08:48:34 +0000 (08:48 +0000)]
gst/nsf/types.h: Rename #ifndef header guard symbol to something less generic, so types.h doesn't get skipped over wh...
Original commit message from CVS:
* gst/nsf/types.h:
Rename #ifndef header guard symbol to something less generic, so
types.h doesn't get skipped over when compiling on MingW. Include
GLib headers and use those to set the endianness and the basic
types so that this isn't entirely broken for non-x86 architectures.
Tim-Philipp Müller [Tue, 17 Apr 2007 08:04:43 +0000 (08:04 +0000)]
gst/mve/gstmvedemux.c: Use G_LITTLE_ENDIAN instead of LITTLE_ENDIAN, so stuff compiles on
Original commit message from CVS:
* gst/mve/gstmvedemux.c: (gst_mve_audio_init):
Use G_LITTLE_ENDIAN instead of LITTLE_ENDIAN, so stuff compiles on
MingW (no idea though why we add a BYTE_ORDER endianness field if
the audio is compressed).