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2008-05-27configure.ac: Require CVS core and base for new audio clock reset method.Wim Taymans3-2/+11
Original commit message from CVS: * configure.ac: Require CVS core and base for new audio clock reset method. * ext/alsaspdif/alsaspdifsink.c: (alsaspdifsink_change_state): Reset the audio clock. See #521761.
2008-05-26ext/jack/gstjackaudiosink.c: Include the element name in the port name to ↵Wim Taymans2-5/+14
avoid duplicate port names. Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (gst_jack_audio_sink_allocate_channels): Include the element name in the port name to avoid duplicate port names.
2008-05-26gst/interleave/deinterleave.c: Add another example launch line.Sebastian Dröge5-455/+1157
Original commit message from CVS: * gst/interleave/deinterleave.c: Add another example launch line. * gst/interleave/interleave.c: (interleave_24), (gst_interleave_finalize), (gst_interleave_base_init), (gst_interleave_class_init), (gst_interleave_init), (gst_interleave_request_new_pad), (gst_interleave_release_pad), (gst_interleave_change_state), (__remove_channels), (__set_channels), (gst_interleave_sink_getcaps), (gst_interleave_set_process_function), (gst_interleave_sink_setcaps), (gst_interleave_sink_event), (gst_interleave_src_query_duration), (gst_interleave_src_query), (forward_event_func), (forward_event), (gst_interleave_src_event), (gst_interleave_collected): * gst/interleave/interleave.h: Major rewrite of interleave using GstCollectpads. This new version also supports almost all raw audio formats and has better caps negotiation. Fixes bug #506594. Also update docs and add some more examples. * tests/check/elements/interleave.c: (interleave_chain_func), (GST_START_TEST), (src_handoff_float32), (sink_handoff_float32), (interleave_suite): Add some more extensive unit tests for interleave.
2008-05-26gst/rtpmanager/gstrtpjitterbuffer.c: When checking the seqnum, reset the ↵Wim Taymans4-38/+81
jitterbuffer if the gap is too big, we need ... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): When checking the seqnum, reset the jitterbuffer if the gap is too big, we need to do this so that we can better handle a restarted source. Fix some comments. * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew), (rtp_jitter_buffer_insert): Tweak the skew resync diff. Use our working seqnum compare function in -base. Rework the jitterbuffer insert code to make it clearer and more performant by only retrieving the seqnum of the input buffer once and by adding some G_LIKELY compiler hints. Improve debugging for duplicate packets. * gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp): Fix a comment, we don't do skew correction here..
2008-05-26gst/rtpmanager/gstrtpbin.c: Propagate the do-lost and latency properties to ↵Håvard Graff2-0/+34
the jitterbuffers when they are changed o... Original commit message from CVS: Patch by: Håvard Graff <havard dot graff at tandberg dot com> * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_propagate_property_to_jitterbuffer), (gst_rtp_bin_set_property): Propagate the do-lost and latency properties to the jitterbuffers when they are changed on rtpbin.
2008-05-26Don't use _gst_pad().Wim Taymans8-21/+37
Original commit message from CVS: * examples/switch/switcher.c: (switch_timer): * gst/replaygain/gstrgvolume.c: (gst_rg_volume_init): * gst/rtpmanager/gstrtpclient.c: (create_stream): * gst/sdp/gstsdpdemux.c: (gst_sdp_demux_stream_configure_udp), (gst_sdp_demux_stream_configure_udp_sink): * tests/check/elements/deinterleave.c: (GST_START_TEST), (pad_added_setup_data_check_float32_8ch_cb): * tests/check/elements/rganalysis.c: (send_eos_event), (send_tag_event): Don't use _gst_pad().
2008-05-22docs/plugins/: Add interleave/deinterleave to the docs and while at that run ↵Sebastian Dröge69-642/+2700
make update in docs/plugins. 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.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.prerequisites: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-alsaspdif.xml: * docs/plugins/inspect/plugin-amrwb.xml: * docs/plugins/inspect/plugin-app.xml: * docs/plugins/inspect/plugin-bayer.xml: * docs/plugins/inspect/plugin-bz2.xml: * docs/plugins/inspect/plugin-cdaudio.xml: * docs/plugins/inspect/plugin-cdxaparse.xml: * docs/plugins/inspect/plugin-dfbvideosink.xml: * docs/plugins/inspect/plugin-dtsdec.xml: * docs/plugins/inspect/plugin-dvb.xml: * docs/plugins/inspect/plugin-dvdspu.xml: * docs/plugins/inspect/plugin-faac.xml: * docs/plugins/inspect/plugin-faad.xml: * docs/plugins/inspect/plugin-fbdevsink.xml: * docs/plugins/inspect/plugin-festival.xml: * docs/plugins/inspect/plugin-filter.xml: * docs/plugins/inspect/plugin-flvdemux.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-interleave.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-ladspa.xml: * docs/plugins/inspect/plugin-metadata.xml: * docs/plugins/inspect/plugin-mms.xml: * docs/plugins/inspect/plugin-modplug.xml: * docs/plugins/inspect/plugin-mpeg2enc.xml: * docs/plugins/inspect/plugin-mpeg4videoparse.xml: * docs/plugins/inspect/plugin-mpegtsparse.xml: * docs/plugins/inspect/plugin-mpegvideoparse.xml: * docs/plugins/inspect/plugin-musepack.xml: * docs/plugins/inspect/plugin-musicbrainz.xml: * docs/plugins/inspect/plugin-mve.xml: * docs/plugins/inspect/plugin-nas.xml: * docs/plugins/inspect/plugin-neon.xml: * docs/plugins/inspect/plugin-nsfdec.xml: * docs/plugins/inspect/plugin-nuvdemux.xml: * docs/plugins/inspect/plugin-rawparse.xml: * docs/plugins/inspect/plugin-real.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rfbsrc.xml: * docs/plugins/inspect/plugin-sdl.xml: * docs/plugins/inspect/plugin-sdp.xml: * docs/plugins/inspect/plugin-selector.xml: * docs/plugins/inspect/plugin-sndfile.xml: * docs/plugins/inspect/plugin-soundtouch.xml: * docs/plugins/inspect/plugin-spcdec.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-speexresample.xml: * docs/plugins/inspect/plugin-stereo.xml: * docs/plugins/inspect/plugin-tta.xml: * docs/plugins/inspect/plugin-vcdsrc.xml: * docs/plugins/inspect/plugin-videosignal.xml: * docs/plugins/inspect/plugin-vmnc.xml: * docs/plugins/inspect/plugin-wildmidi.xml: * docs/plugins/inspect/plugin-x264.xml: * docs/plugins/inspect/plugin-xvid.xml: * docs/plugins/inspect/plugin-y4menc.xml: Add interleave/deinterleave to the docs and while at that run make update in docs/plugins. * gst/interleave/deinterleave.c: Add a parapraph about using a queue and audioconvert after the source pads to the docs.
2008-05-22gst/interleave/deinterleave.*: Don't set a getcaps() function on the src ↵Sebastian Dröge3-8/+37
pads as it's not required and the default ge... Original commit message from CVS: * gst/interleave/deinterleave.c: (gst_deinterleave_base_init), (gst_deinterleave_class_init), (gst_deinterleave_init), (gst_deinterleave_add_new_pads), (gst_deinterleave_sink_getcaps): * gst/interleave/deinterleave.h: Don't set a getcaps() function on the src pads as it's not required and the default getcaps() function returns the correct results for our src pads. Complete documentation and add myself to the authors of the element.
2008-05-22tests/icles/: Small oss4 test that probes for available devices and ↵Tim-Philipp Müller4-2/+274
retrieves their caps and mixer tracks and all tha... Original commit message from CVS: * tests/icles/.cvsignore: * tests/icles/Makefile.am: * tests/icles/test-oss4.c: (opt_show_mixer_messages), (WAIT_TIME), (show_mixer_messages), (probe_mixer_tracks), (probe_pad), (probe_details), (probe_element), (main): Small oss4 test that probes for available devices and retrieves their caps and mixer tracks and all that. Also allows testing of mixer change messages on the bus.
2008-05-22sys/oss4/: Make device-name probing in NULL state work better (e.g. for the ↵Tim-Philipp Müller6-2/+52
gnome-control-center sound capplet). Original commit message from CVS: * sys/oss4/oss4-mixer.c: (gst_oss4_mixer_open): * sys/oss4/oss4-property-probe.c: (gst_oss4_property_probe_find_device_name), (gst_oss4_property_probe_find_device_name_nofd): * sys/oss4/oss4-property-probe.h: * sys/oss4/oss4-sink.c: (gst_oss4_sink_get_property): * sys/oss4/oss4-source.c: (gst_oss4_source_get_property): Make device-name probing in NULL state work better (e.g. for the gnome-control-center sound capplet).
2008-05-22gst/mpeg4videoparse/mpeg4videoparse.c: Move some code around to integrate ↵Sjoerd Simons3-56/+38
the startcode searching with the other bits... Original commit message from CVS: Patch by: Sjoerd Simons <sjoerd at luon dot net> * gst/mpeg4videoparse/mpeg4videoparse.c: (gst_mpeg4vparse_push), (gst_mpeg4vparse_drain), (gst_mpeg4vparse_chain), (gst_mpeg4vparse_change_state): Move some code around to integrate the startcode searching with the other bits of parsing, avoid a whole bunch of peeks. Get rid of invalid data that should not happen according to the specs. Fixes #533559.
2008-05-20ext/mythtv/gstmythtvsrc.c: Correctly set duration to get a more correct seek ↵Bastien Nocera2-11/+46
bar in totem. Original commit message from CVS: Patch by: Bastien Nocera <hadess at hadess dot net> * ext/mythtv/gstmythtvsrc.c: (gst_mythtv_src_class_init), (gst_mythtv_src_init), (gst_mythtv_src_clear), (do_read_request_response), (gst_mythtv_src_create), (gst_mythtv_src_start): Correctly set duration to get a more correct seek bar in totem. Disable query and event functions as they don't work and do some smaller cleanup. Fixes bug #533736.
2008-05-20ext/spc/: Add support for some essential features like seeking, reading song ↵Brian Koropoff6-35/+697
duration and extended tags. Fixes bug #4... Original commit message from CVS: Patch by: Brian Koropoff <brianhk at cs dot washington dot edu> * ext/spc/Makefile.am: * ext/spc/gstspc.c: (gst_spc_dec_class_init), (gst_spc_dec_src_query_type), (gst_spc_dec_init), (gst_spc_dec_dispose), (gst_spc_dec_sink_event), (gst_spc_duration), (gst_spc_fadeout), (gst_spc_dec_src_event), (gst_spc_dec_src_query), (spc_play), (spc_setup): * ext/spc/gstspc.h: * ext/spc/tag.c: (spc_tag_is_extended), (spc_tag_is_text_format), (spc_tag_is_present), (spc_tag_unpack_date), (spc_tag_clear), (spc_tag_get_info), (spc_tag_free): * ext/spc/tag.h: Add support for some essential features like seeking, reading song duration and extended tags. Fixes bug #454151.
2008-05-19tests/check/elements/deinterleave.c: Set keep-positions property to TRUE for ↵Sebastian Dröge2-0/+7
the 8 channel test to ensure that the or... Original commit message from CVS: * tests/check/elements/deinterleave.c: (GST_START_TEST): Set keep-positions property to TRUE for the 8 channel test to ensure that the original channel position is set on the output.
2008-05-19gst/interleave/deinterleave.*: Add a property to select whether channel ↵Sebastian Dröge3-6/+78
positions should be kept on the mono output b... Original commit message from CVS: * gst/interleave/deinterleave.c: (gst_deinterleave_class_init), (gst_deinterleave_init), (gst_deinterleave_add_new_pads), (gst_deinterleave_set_pads_caps), (gst_deinterleave_set_property), (gst_deinterleave_get_property): * gst/interleave/deinterleave.h: Add a property to select whether channel positions should be kept on the mono output buffers or should be dropped.
2008-05-18docs/Makefile.am: Oops - fix the spelling of the variable I added.Jan Schmidt2-2/+7
Original commit message from CVS: * docs/Makefile.am: Oops - fix the spelling of the variable I added.
2008-05-17gst/interleave/deinterleave.*: Queue events until src pads were added and ↵Sebastian Dröge4-0/+103
they can be sent. Otherwise downstream will... Original commit message from CVS: * gst/interleave/deinterleave.c: (gst_deinterleave_finalize), (gst_deinterleave_init), (gst_deinterleave_sink_event), (gst_deinterleave_process), (gst_deinterleave_sink_activate_push): * gst/interleave/deinterleave.h: Queue events until src pads were added and they can be sent. Otherwise downstream will never get the first newsegment event.
2008-05-17gst/interleave/deinterleave.c: Always set the channel positions when ↵Sebastian Dröge2-20/+35
gst_audio_get_channel_positions() returns someth... Original commit message from CVS: * gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps), (gst_deinterleave_getcaps): Always set the channel positions when gst_audio_get_channel_positions() returns something, even if they're not set in the caps. This makes sure that the output channels can be interleaved again correctly in the mono/stereo cases too. Don't ask for the peercaps of the current pad in getcaps() as this might call getcaps() again and deadlock.
2008-05-16ext/timidity/gstwildmidi.c: Check some more common locations for a valid ↵Sebastian Dröge2-0/+29
configuration file. Original commit message from CVS: * ext/timidity/gstwildmidi.c: (wildmidi_open_config): Check some more common locations for a valid configuration file. Fixes bug #533435. Packagers should still #define WILDMIDI_CFG to the distributions default location.
2008-05-16gst/interleave/: Add support for all raw audio formats and provide better ↵Sebastian Dröge5-64/+403
negotiation if the caps are changing. Original commit message from CVS: * gst/interleave/Makefile.am: * gst/interleave/deinterleave.c: (deinterleave_24), (gst_deinterleave_finalize), (gst_deinterleave_base_init), (gst_deinterleave_class_init), (gst_deinterleave_init), (gst_deinterleave_add_new_pads), (gst_deinterleave_set_pads_caps), (gst_deinterleave_set_process_function), (gst_deinterleave_sink_setcaps), (__remove_channels), (__set_channels), (gst_deinterleave_getcaps), (gst_deinterleave_process), (gst_deinterleave_chain), (gst_deinterleave_sink_activate_push): * gst/interleave/deinterleave.h: Add support for all raw audio formats and provide better negotiation if the caps are changing. Don't allow changes of the channel positions and set the position of the corresponding channel on the src pad caps. General cleanup and smaller bugfixes. * tests/check/elements/deinterleave.c: (float_buffer_check_probe): Check the channel positions on the output buffer caps.
2008-05-16docs/Makefile.am: Don't attempt to build plugin docs when they're disabled.Jan Schmidt5-7/+28
Original commit message from CVS: * docs/Makefile.am: Don't attempt to build plugin docs when they're disabled. * gst/bayer/Makefile.am: Add libgstvideo to the link. * gst/rtpmanager/Makefile.am: Fix link order, and move LIBS things to _LIBS
2008-05-16docs/plugins/gst-plugins-bad-plugins.types: Remove bogus attempt to pull ↵Jan Schmidt2-4/+6
'metadata' plugin's base class into the docs. Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins.types: Remove bogus attempt to pull 'metadata' plugin's base class into the docs.
2008-05-14gst/rtpmanager/gstrtpjitterbuffer.c: Simply drop bad RTP packets with a ↵Wim Taymans2-4/+12
warning instead of just posting an error and ... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain): Simply drop bad RTP packets with a warning instead of just posting an error and stopping. This is a perfectly recoverable event and we don't force people to use an rtpbin to filter out bad packets first.
2008-05-14gst/mpeg4videoparse/mpeg4videoparse.c: Set fixed caps on the srcpad after we ↵Wim Taymans3-1/+6
created the pad... Original commit message from CVS: * gst/mpeg4videoparse/mpeg4videoparse.c: (gst_mpeg4vparse_init): Set fixed caps on the srcpad after we created the pad...
2008-05-14tests/check/Makefile.am: Remove deinterleave test from VALGRIND_TO_FIX again ↵Tim-Philipp Müller3-1/+6
now that there are suppressions in gst.s... Original commit message from CVS: * tests/check/Makefile.am: Remove deinterleave test from VALGRIND_TO_FIX again now that there are suppressions in gst.supp which make this work for me.
2008-05-14tests/check/Makefile.am: Add deinterleave unit test to VALGRIND_TO_FIX, ↵Tim-Philipp Müller3-0/+202
since it causes weird invalid free errors in ... Original commit message from CVS: * tests/check/Makefile.am: Add deinterleave unit test to VALGRIND_TO_FIX, since it causes weird invalid free errors in valgrind/libc after _exit for some reason. * tests/check/elements/deinterleave.c: (pads_created), (set_channel_positions), (src_handoff_float32_8ch), (float_buffer_check_probe), (pad_added_setup_data_check_float32_8ch_cb), (make_fake_src_8chans_float32), (GST_START_TEST), (deinterleave_suite): Add some more deinterleave unit test bits I had locally.
2008-05-14gst/audioresample/gstaudioresample.c: Revert previous change which made ↵Tim-Philipp Müller1-0/+4
basetransform handle buffer_alloc and which b... Original commit message from CVS: * gst/audioresample/gstaudioresample.c: (gst_audioresample_init): Revert previous change which made basetransform handle buffer_alloc and which breaks things badly in the non-passthrough case since it returned buffers with a different (ie. sometimes smaller) size than the size requested.
2008-05-14gst/interleave/: Split definitions into separate header files for better ↵Sebastian Dröge9-120/+594
documentation generation. Original commit message from CVS: * gst/interleave/Makefile.am: * gst/interleave/deinterleave.h: * gst/interleave/interleave.h: * gst/interleave/plugin.h: Split definitions into separate header files for better documentation generation. * gst/interleave/deinterleave.c: (gst_deinterleave_base_init), (gst_deinterleave_class_init), (gst_deinterleave_sink_setcaps), (gst_deinterleave_process): Don't use alloca, allow caps changes as long as the number of channels does not change, don't use g_warning, return NOT_NEGOTIATED as early as possible and some other cleanup. * gst/interleave/interleave.c: (gst_interleave_base_init), (gst_interleave_class_init): Do some random cleanup. * tests/check/Makefile.am: * tests/check/elements/deinterleave.c: (GST_START_TEST), (deinterleave_chain_func), (deinterleave_pad_added), (deinterleave_suite): Add unit tests for the deinterleave element.
2008-05-13gst/mpeg4videoparse/mpeg4videoparse.*: Parse the config data (either ↵Sjoerd Simons3-30/+563
outbound or in the stream) to set width/height, ... Original commit message from CVS: Patch by: Sjoerd Simons <sjoerd at luon dot net> * gst/mpeg4videoparse/mpeg4videoparse.c: (gst_mpeg4vparse_set_new_caps), (gst_mpeg4vparse_align), (get_bits), (next_start_code), (gst_mpeg4vparse_handle_vos), (gst_mpeg4vparse_push), (gst_mpeg4vparse_drain), (gst_mpeg4vparse_chain), (gst_mpeg4vparse_sink_setcaps), (gst_mpeg4vparse_sink_event), (gst_mpeg4vparse_src_query), (gst_mpeg4vparse_set_property), (gst_mpeg4vparse_get_property), (gst_mpeg4vparse_class_init), (gst_mpeg4vparse_init): * gst/mpeg4videoparse/mpeg4videoparse.h: Parse the config data (either outbound or in the stream) to set width/height, apect ration, framerate in the caps if applicable. Mark frames as GST_BUFFER_FLAG_DELTA_UNIT when they are not intra frames Set the timestamps of outgoing buffers to the buffer in which the VOP header was found. Drop incoming data untill configuration is found (by default, configurable using a property). Report a 1 frame latency. Fixes #532723.
2008-05-13gst/real/gstrealvideodec.c: Add some debug for where we are searching for ↵Wim Taymans2-0/+8
libraries. Original commit message from CVS: * gst/real/gstrealvideodec.c: (open_library): Add some debug for where we are searching for libraries.
2008-05-13tests/check/elements/audioresample.c: Add unit test for the latest ↵Sjoerd Simons1-43/+77
basetransform negotiation changes. Original commit message from CVS: Patch by: Sjoerd Simons <sjoerd at luon dot net> * tests/check/elements/audioresample.c: (live_switch_alloc_only_48000), (live_switch_get_sink_caps), (live_switch_push), (GST_START_TEST): Add unit test for the latest basetransform negotiation changes. See bug #526768.
2008-05-13gst/rtpmanager/gstrtpbin.c: Actually add the do-lost property to the object.Wim Taymans2-0/+10
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init): Actually add the do-lost property to the object.
2008-05-12gst/rtpmanager/gstrtpjitterbuffer.c: Avoid waiting for a negative (huge) ↵Wim Taymans2-2/+15
duration when the last packet has a lower ti... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Avoid waiting for a negative (huge) duration when the last packet has a lower timestamp than the current packet.
2008-05-12gst/rtpmanager/gstrtpsession.c: Make sure to unref the rtpsession returned ↵Peter Kjellerstedt2-0/+10
by gst_pad_get_parent() to prevent a memor... Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_query_send_rtcp_src): Make sure to unref the rtpsession returned by gst_pad_get_parent() to prevent a memory leak.
2008-05-12docs/plugins/gst-plugins-bad-plugins-sections.txt: Quieten some docs outputJan Schmidt2-2/+11
Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins-sections.txt: Quieten some docs output
2008-05-12gst/rtpmanager/gstrtpjitterbuffer.c: Initialise with GST_CLOCK_TIME_NONE to ↵Jan Schmidt3-1/+7
avoid compiler warning. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Initialise with GST_CLOCK_TIME_NONE to avoid compiler warning.
2008-05-11Random doc of the day: the deinterlace element.Jan Schmidt7-11/+102
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-gstinterlace.xml: * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: Random doc of the day: the deinterlace element.
2008-05-09gst/mpegtsparse/: Make sure all schedule EIT and non-actual transport streamZaheer Abbas Merali3-0/+31
Original commit message from CVS: * gst/mpegtsparse/mpegtspacketizer.c: * gst/mpegtsparse/mpegtsparse.c: Make sure all schedule EIT and non-actual transport stream EITs are parsed. Also add present-following flag and actual-transport-stream flag to eit bus message.
2008-05-09gst/rtpmanager/rtpsource.c: Make sure to unref the caps used by RTPSource to ↵Peter Kjellerstedt3-0/+7
prevent a memory leak. Original commit message from CVS: * gst/rtpmanager/rtpsource.c: (rtp_source_finalize): Make sure to unref the caps used by RTPSource to prevent a memory leak.
2008-05-08sys/oss4/oss4-mixer-slider.c: Apparently mono sliders have the mono value ↵Clive Wright2-2/+17
repeated in the upper bits, so mask those o... Original commit message from CVS: Based on patch by: Clive Wright <clive_wright ntlworld com> * sys/oss4/oss4-mixer-slider.c: (gst_oss4_mixer_slider_unpack_volume): Apparently mono sliders have the mono value repeated in the upper bits, so mask those out when reading them. Probably makes the mixer applet work properly in some more cases.
2008-05-08gst/rtpmanager/rtpsession.c: Unlock the session lock when calling one of our ↵Olivier Crete2-0/+17
callbacks. Original commit message from CVS: Patch by: Olivier Crete <tester at tester dot ca> * gst/rtpmanager/rtpsession.c: (source_clock_rate), (rtp_session_process_bye), (rtp_session_send_bye_locked): Unlock the session lock when calling one of our callbacks. Fixes #532011.
2008-05-08gst/rtpmanager/gstrtpsession.c: Send RTP BYE command on EOS. Fixes bug #531955.Sjoerd Simons2-0/+9
Original commit message from CVS: Patch by: Sjoerd Simons <sjoerd at luon dot net> * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_send_rtp_sink): Send RTP BYE command on EOS. Fixes bug #531955.
2008-05-08gst/audioresample/gstaudioresample.c: Let audioresample use the buffer ↵Sjoerd Simons2-4/+89
allocation of basetransform instead of it's ow... Original commit message from CVS: Patch by: Sjoerd Simons <sjoerd at luon dot net> * gst/audioresample/gstaudioresample.c: (gst_audioresample_init): Let audioresample use the buffer allocation of basetransform instead of it's own stuff. * tests/check/elements/audioresample.c: (alloc_only_48000), (GST_START_TEST), (audioresample_suite): Add unit test for the recent basetransform bugfix, where upstream changes caps to something that can't be passed through anymore.
2008-05-07win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC ↵Ole André Vadla Ravnås2-3/+6
we'd much rather use the real thing than h... Original commit message from CVS: * win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather use the real thing than having "???" unconditionally.
2008-05-07gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.Wim Taymans8-80/+465
Original commit message from CVS: * gst-libs/gst/app/.cvsignore: * gst-libs/gst/app/Makefile.am: * gst-libs/gst/app/gstapp-marshal.list: Add marshal.list, make it compile and add to cvsignore. * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose), (gst_app_sink_stop): Small cleanups. * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init), (gst_app_src_init), (gst_app_src_set_property), (gst_app_src_get_property), (gst_app_src_unlock), (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop), (gst_app_src_create), (gst_app_src_set_caps), (gst_app_src_get_caps), (gst_app_src_set_size), (gst_app_src_get_size), (gst_app_src_set_seekable), (gst_app_src_get_seekable), (gst_app_src_set_max_buffers), (gst_app_src_get_max_buffers), (gst_app_src_push_buffer), (gst_app_src_end_of_stream): * gst-libs/gst/app/gstappsrc.h: Beat appsrc in shape, add signals and actions. Add some docs. Add properties for caps, size, seekability and max-buffers. Fix unlock/stop code.
2008-05-07configure.ac: Error out if we don't have the required versions of core/base.Tim-Philipp Müller3-6/+10
Original commit message from CVS: * configure.ac: Error out if we don't have the required versions of core/base.
2008-05-05gst-libs/gst/app/gstappsink.*: Start some docs.Wim Taymans3-25/+185
Original commit message from CVS: * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init), (gst_app_sink_init), (gst_app_sink_set_property), (gst_app_sink_get_property), (gst_app_sink_unlock_start), (gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked), (gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll), (gst_app_sink_render), (gst_app_sink_set_caps), (gst_app_sink_set_drop), (gst_app_sink_get_drop): * gst-libs/gst/app/gstappsink.h: Start some docs. Add property to drop buffers when the queue is filled Fix unlocking and flushing when the queues are filled.
2008-05-02add wildmidi pluginChristian Schaller1-1/+4
Original commit message from CVS: add wildmidi plugin
2008-04-29gst/subenc/gstsrtenc.c: Declare variables at the beginning of blocks. Fixes ↵Jens Granseuer2-4/+14
compilation with gcc 2.x and other compil... Original commit message from CVS: Patch by: Jens Granseuer <jensgr at gmx dot net> * gst/subenc/gstsrtenc.c: (gst_srt_enc_timestamp_to_string): Declare variables at the beginning of blocks. Fixes compilation with gcc 2.x and other compilers. Fixes bug #530611.
2008-04-29gst/mpegtsparse/: Detect SI pids (NIT, SDT, EIT etc.) based on table id and ↵Zaheer Abbas Merali4-36/+85
not by pid number. This allows for exampl... Original commit message from CVS: * gst/mpegtsparse/mpegtspacketizer.c: * gst/mpegtsparse/mpegtspacketizer.h: * gst/mpegtsparse/mpegtsparse.c: Detect SI pids (NIT, SDT, EIT etc.) based on table id and not by pid number. This allows for example the EPG data from UK's freesat to be picked up.