Wim Taymans [Fri, 6 Jun 2008 14:19:54 +0000 (14:19 +0000)]
examples/app/appsrc-stream.c: Use deep-notify until we can depend on a playbin2 with support for the source property.
Original commit message from CVS:
* examples/app/appsrc-stream.c: (found_source), (main):
Use deep-notify until we can depend on a playbin2 with support for the
source property.
Wim Taymans [Fri, 6 Jun 2008 13:01:05 +0000 (13:01 +0000)]
gst/rtpmanager/gstrtpbin.c: Fix deadlock when shutting down, use a new lock instead to properly shutdown.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_init),
(gst_rtp_bin_finalize), (gst_rtp_bin_change_state):
Fix deadlock when shutting down, use a new lock instead to properly
shutdown.
Wim Taymans [Thu, 5 Jun 2008 16:38:50 +0000 (16:38 +0000)]
examples/app/: Added an example on how to use appsrc in playbin in streaming mode from an mmapped file.
Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsrc-stream.c: (read_data), (start_feed),
(stop_feed), (found_source), (bus_message), (main):
Added an example on how to use appsrc in playbin in streaming mode from
an mmapped file.
* examples/app/appsrc_ex.c: (main):
Set pipeline to NULL to free queued buffers.
* gst-libs/gst/app/gstapp-marshal.list:
* gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_flush_queued), (gst_app_src_dispose),
(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_is_seekable),
(gst_app_src_check_get_range), (gst_app_src_do_seek),
(gst_app_src_create), (gst_app_src_set_stream_type),
(gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
(gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
(gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
(gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
* gst-libs/gst/app/gstappsrc.h:
Measure max queue size in bytes instead.
Add support for 3 modes of operation, streaming, seekable and
random-access, making basesrc handle the scheduling modes for each.
Add appsrc:// uri handler so that automatic plugging can be done from
playbin2 or uridecodebin, for example.
Added support for custom segment formats.
Add support for push and pull based operations from the application.
Expand the methods so that errors can be detected.
Flush the queued buffers on seeks and when shutting down.
Add signals to inform the app that a seek must happen.
Sebastian Dröge [Thu, 5 Jun 2008 11:07:17 +0000 (11:07 +0000)]
gst/interleave/: Properly implement duration and position queries in bytes format. We have to take the upstream reply...
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_add_new_pads),
(gst_deinterleave_src_query):
* gst/interleave/interleave.c: (gst_interleave_src_query_duration),
(gst_interleave_src_query):
Properly implement duration and position queries in bytes format. We
have to take the upstream reply and divide/multiply it by the number
of channels to get the correct result.
Michael Smith [Wed, 4 Jun 2008 21:18:53 +0000 (21:18 +0000)]
sys/dshowvideosink/: Fix up copyright notice on new plugin.
Original commit message from CVS:
* sys/dshowvideosink/dshowvideofakesrc.cpp:
* sys/dshowvideosink/dshowvideofakesrc.h:
* sys/dshowvideosink/dshowvideosink.cpp:
* sys/dshowvideosink/dshowvideosink.h:
Fix up copyright notice on new plugin.
Jon Trowbridge [Wed, 4 Jun 2008 17:02:38 +0000 (17:02 +0000)]
ext/dirac/gstdiracenc.cc: Update properties for recent dirac changes. Patch from Jonathan Rosser.
Original commit message from CVS:
* ext/dirac/gstdiracenc.cc: Update properties for recent
dirac changes. Patch from Jonathan Rosser.
Tim-Philipp Müller [Wed, 4 Jun 2008 11:33:21 +0000 (11:33 +0000)]
ext/x264/gstx264enc.c: Try harder not to crash when we get an EOS event but haven't set up the encoder yet (as may ha...
Original commit message from CVS:
* ext/x264/gstx264enc.c: (gst_x264_enc_header_buf),
(gst_x264_enc_sink_event), (gst_x264_enc_chain),
(gst_x264_enc_encode_frame):
Try harder not to crash when we get an EOS event but haven't set
up the encoder yet (as may happen when upstream errors out with
not-negotiated, for example). Also, always push the EOS event
downstream.
Sebastian Dröge [Wed, 4 Jun 2008 06:48:46 +0000 (06:48 +0000)]
gst/interleave/interleave.*: Use an always increasing integer for the number in the name of the requested sink pads t...
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_pad_get_type),
(gst_interleave_pad_get_property), (gst_interleave_pad_class_init),
(gst_interleave_request_new_pad), (gst_interleave_release_pad):
* gst/interleave/interleave.h:
Use an always increasing integer for the number in the name of the
requested sink pads to guarantuee a unique name. Add a "channel"
property to GstInterleavePad to make it possible for applications
to retrieve the channel number in the output for every pad.
Use g_type_register_static_simple() instead of
g_type_register_static() to save some relocations.
Christian Schaller [Tue, 3 Jun 2008 15:41:05 +0000 (15:41 +0000)]
fix package name
Original commit message from CVS:
fix package name
Sebastian Dröge [Tue, 3 Jun 2008 14:35:59 +0000 (14:35 +0000)]
gst/interleave/interleave.c: Stop GstCollectPads before calling the parent's state change function when going from PA...
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_pad_get_type),
(gst_interleave_change_state):
Stop GstCollectPads before calling the parent's state change function
when going from PAUSED to READY as we otherwise deadlock.
Fixes bug #536258.
Wim Taymans [Tue, 3 Jun 2008 11:10:32 +0000 (11:10 +0000)]
gst/h264parse/gsth264parse.*: Parse codec_data and use the nalu_size_length field to get the NALU length in packetize...
Original commit message from CVS:
* gst/h264parse/gsth264parse.c: (gst_nal_bs_init),
(gst_h264_parse_sink_setcaps), (gst_h264_parse_chain_forward),
(gst_h264_parse_queue_buffer), (gst_h264_parse_chain_reverse),
(gst_h264_parse_chain):
* gst/h264parse/gsth264parse.h:
Parse codec_data and use the nalu_size_length field to get the NALU
length in packetized h264.
When queueing a packetized buffer in reverse mode, don't unref the
buffer twice.
Avoid accessing the buffer TIMESTAMP field after we pushed it on
the adaptor.
Sebastian Dröge [Tue, 3 Jun 2008 09:03:19 +0000 (09:03 +0000)]
gst/interleave/interleave.c: Use new gst_audio_check_channel_positions() function and register the GstInterleavePad t...
Original commit message from CVS:
* gst/interleave/interleave.c:
(gst_interleave_check_channel_positions),
(gst_interleave_set_channel_positions),
(gst_interleave_class_init):
Use new gst_audio_check_channel_positions() function and register
the GstInterleavePad type from a threadsafe context.
Michael Smith [Mon, 2 Jun 2008 21:51:52 +0000 (21:51 +0000)]
configure.ac: Revert accidental addition in configure.ac. Sorry.
Original commit message from CVS:
* configure.ac:
Revert accidental addition in configure.ac. Sorry.
Michael Smith [Mon, 2 Jun 2008 18:23:54 +0000 (18:23 +0000)]
Add a new win32 videosink. Uses the DirectShow renderers for high-performance video rendering on win32.
Original commit message from CVS:
* configure.ac:
* sys/Makefile.am:
* sys/dshowvideosink/Makefile.am:
* sys/dshowvideosink/README:
* sys/dshowvideosink/dshowvideofakesrc.cpp:
* sys/dshowvideosink/dshowvideofakesrc.h:
* sys/dshowvideosink/dshowvideosink.cpp:
* sys/dshowvideosink/dshowvideosink.h:
Add a new win32 videosink. Uses the DirectShow renderers for
high-performance video rendering on win32.
Currently only supports some YUV formats.
Rank PRIMARY, since it's much more useful for the common cases that the
directdraw sink (which only does RGB).
Tim-Philipp Müller [Mon, 2 Jun 2008 18:06:37 +0000 (18:06 +0000)]
ext/spc/Makefile.am: Dist tag.h
Original commit message from CVS:
* ext/spc/Makefile.am:
Dist tag.h
Wim Taymans [Mon, 2 Jun 2008 17:06:34 +0000 (17:06 +0000)]
ext/faad/gstfaad.c: Always drain before activating the new segment.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_sink_event):
Always drain before activating the new segment.
Sebastian Dröge [Mon, 2 Jun 2008 12:42:14 +0000 (12:42 +0000)]
gst/interleave/interleave.*: Allow setting channel positions via a property and allow using the channel positions on ...
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_pad_get_type),
(gst_interleave_finalize), (gst_audio_check_channel_positions),
(gst_interleave_set_channel_positions),
(gst_interleave_class_init), (gst_interleave_init),
(gst_interleave_set_property), (gst_interleave_get_property),
(gst_interleave_request_new_pad), (gst_interleave_release_pad),
(gst_interleave_sink_setcaps), (gst_interleave_src_query_duration),
(gst_interleave_src_query_latency), (gst_interleave_collected):
* gst/interleave/interleave.h:
Allow setting channel positions via a property and allow using the
channel positions on the input as the channel positions of the output.
Fix some broken logic and memory leaks.
* tests/check/Makefile.am:
* tests/check/elements/interleave.c: (src_handoff_float32),
(sink_handoff_float32), (GST_START_TEST), (interleave_suite):
Add unit tests for checking correct handling of channel positions.
Wim Taymans [Mon, 2 Jun 2008 10:18:25 +0000 (10:18 +0000)]
ext/faad/gstfaad.*: Add basic reverse playback support.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_dispose), (clear_queued),
(flush_queued), (gst_faad_drain), (gst_faad_do_raw_seek),
(gst_faad_src_event), (gst_faad_sink_event), (gst_faad_chain),
(gst_faad_change_state):
* ext/faad/gstfaad.h:
Add basic reverse playback support.
Clear decoder state after disconts.
Remove some unused code.
Mark output buffers with a discont after a decoding error.
Sjoerd Simons [Mon, 2 Jun 2008 07:37:31 +0000 (07:37 +0000)]
gst/mpeg4videoparse/mpeg4videoparse.c: Fix mpeg4videoparse on big endian architectures. Fixes bug #536042.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/mpeg4videoparse/mpeg4videoparse.c:
(gst_mpeg4vparse_handle_vos):
Fix mpeg4videoparse on big endian architectures. Fixes bug #536042.
Sebastian Dröge [Thu, 29 May 2008 19:56:53 +0000 (19:56 +0000)]
tests/check/elements/mplex.c: Don't use the deprecated gst_element_get_pad().
Original commit message from CVS:
* tests/check/elements/mplex.c: (setup_src_pad),
(teardown_src_pad):
Don't use the deprecated gst_element_get_pad().
Sebastian Dröge [Thu, 29 May 2008 19:11:47 +0000 (19:11 +0000)]
examples/directfb/gstdfb.c: Don't use the deprecated gst_element_get_pad().
Original commit message from CVS:
* examples/directfb/gstdfb.c: (main):
Don't use the deprecated gst_element_get_pad().
Onkar Shinde [Wed, 28 May 2008 08:53:00 +0000 (08:53 +0000)]
sys/vcd/vcdsrc.c: Allow the track to be set by using the uri. Fixes #535043.
Original commit message from CVS:
Based on patch by: <onkarshinde at gmail dot com>
* sys/vcd/vcdsrc.c: (gst_vcdsrc_uri_get_uri),
(gst_vcdsrc_uri_set_uri):
Allow the track to be set by using the uri. Fixes #535043.
Sebastian Dröge [Wed, 28 May 2008 08:14:16 +0000 (08:14 +0000)]
gst/interleave/interleave.c: Implement latency query.
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_src_query_duration),
(gst_interleave_src_query_latency), (gst_interleave_src_query):
Implement latency query.
Thijs Vermeir [Tue, 27 May 2008 17:53:58 +0000 (17:53 +0000)]
gst/mpegvideoparse/mpegvideoparse.c: Add GST_BUFFER_FLAG_DELTA_UNIT to not I frame buffers
Original commit message from CVS:
* gst/mpegvideoparse/mpegvideoparse.c:
Add GST_BUFFER_FLAG_DELTA_UNIT to not I frame buffers
Wim Taymans [Tue, 27 May 2008 16:48:10 +0000 (16:48 +0000)]
gst/rtpmanager/gstrtpbin.c: Break out of callbacks when we are shutting down.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
(gst_rtp_bin_propagate_property_to_jitterbuffer),
(gst_rtp_bin_change_state), (new_payload_found),
(new_ssrc_pad_found):
Break out of callbacks when we are shutting down.
Make sure no state changes can happen when we reconfigure.
Wim Taymans [Tue, 27 May 2008 16:32:18 +0000 (16:32 +0000)]
configure.ac: Require CVS core and base for new audio clock reset method.
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.
Wim Taymans [Mon, 26 May 2008 17:52:21 +0000 (17:52 +0000)]
ext/jack/gstjackaudiosink.c: Include the element name in the port name to 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.
Sebastian Dröge [Mon, 26 May 2008 10:28:47 +0000 (10:28 +0000)]
gst/interleave/deinterleave.c: Add another example launch line.
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.
Wim Taymans [Mon, 26 May 2008 10:09:29 +0000 (10:09 +0000)]
gst/rtpmanager/gstrtpjitterbuffer.c: When checking the seqnum, reset the 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..
Håvard Graff [Mon, 26 May 2008 10:00:24 +0000 (10:00 +0000)]
gst/rtpmanager/gstrtpbin.c: Propagate the do-lost and latency properties to 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.
Wim Taymans [Mon, 26 May 2008 09:57:40 +0000 (09:57 +0000)]
Don't use _gst_pad().
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().
Sebastian Dröge [Thu, 22 May 2008 19:47:53 +0000 (19:47 +0000)]
docs/plugins/: Add interleave/deinterleave to the docs and while at that run 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.
Sebastian Dröge [Thu, 22 May 2008 18:55:09 +0000 (18:55 +0000)]
gst/interleave/deinterleave.*: Don't set a getcaps() function on the src 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.
Tim-Philipp Müller [Thu, 22 May 2008 16:33:25 +0000 (16:33 +0000)]
tests/icles/: Small oss4 test that probes for available devices and 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.
Tim-Philipp Müller [Thu, 22 May 2008 15:14:26 +0000 (15:14 +0000)]
sys/oss4/: Make device-name probing in NULL state work better (e.g. for the 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).
Sjoerd Simons [Thu, 22 May 2008 14:03:05 +0000 (14:03 +0000)]
gst/mpeg4videoparse/mpeg4videoparse.c: Move some code around to integrate 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.
Bastien Nocera [Tue, 20 May 2008 09:36:56 +0000 (09:36 +0000)]
ext/mythtv/gstmythtvsrc.c: Correctly set duration to get a more correct seek 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.
Brian Koropoff [Tue, 20 May 2008 09:04:48 +0000 (09:04 +0000)]
ext/spc/: Add support for some essential features like seeking, reading song 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.
Sebastian Dröge [Mon, 19 May 2008 12:32:06 +0000 (12:32 +0000)]
tests/check/elements/deinterleave.c: Set keep-positions property to TRUE for 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.
Sebastian Dröge [Mon, 19 May 2008 07:46:05 +0000 (07:46 +0000)]
gst/interleave/deinterleave.*: Add a property to select whether channel 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.
Jan Schmidt [Sun, 18 May 2008 10:27:25 +0000 (10:27 +0000)]
docs/Makefile.am: Oops - fix the spelling of the variable I added.
Original commit message from CVS:
* docs/Makefile.am:
Oops - fix the spelling of the variable I added.
Sebastian Dröge [Sat, 17 May 2008 19:39:53 +0000 (19:39 +0000)]
gst/interleave/deinterleave.*: Queue events until src pads were added and 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.
Sebastian Dröge [Sat, 17 May 2008 14:05:03 +0000 (14:05 +0000)]
gst/interleave/deinterleave.c: Always set the channel positions when 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.
Sebastian Dröge [Fri, 16 May 2008 22:00:49 +0000 (22:00 +0000)]
ext/timidity/gstwildmidi.c: Check some more common locations for a valid 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.
Sebastian Dröge [Fri, 16 May 2008 21:56:24 +0000 (21:56 +0000)]
gst/interleave/: Add support for all raw audio formats and provide better 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.
Jan Schmidt [Fri, 16 May 2008 19:56:30 +0000 (19:56 +0000)]
docs/Makefile.am: Don't attempt to build plugin docs when they're disabled.
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
Jan Schmidt [Fri, 16 May 2008 14:49:07 +0000 (14:49 +0000)]
docs/plugins/gst-plugins-bad-plugins.types: Remove bogus attempt to pull '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.
Wim Taymans [Wed, 14 May 2008 21:02:19 +0000 (21:02 +0000)]
gst/rtpmanager/gstrtpjitterbuffer.c: Simply drop bad RTP packets with a 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.
Wim Taymans [Wed, 14 May 2008 20:57:31 +0000 (20:57 +0000)]
gst/mpeg4videoparse/mpeg4videoparse.c: Set fixed caps on the srcpad after we 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...
Tim-Philipp Müller [Wed, 14 May 2008 16:21:05 +0000 (16:21 +0000)]
tests/check/Makefile.am: Remove deinterleave test from VALGRIND_TO_FIX again 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.
Tim-Philipp Müller [Wed, 14 May 2008 14:19:47 +0000 (14:19 +0000)]
tests/check/Makefile.am: Add deinterleave unit test to VALGRIND_TO_FIX, 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.
Tim-Philipp Müller [Wed, 14 May 2008 13:57:41 +0000 (13:57 +0000)]
gst/audioresample/gstaudioresample.c: Revert previous change which made 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.
Sebastian Dröge [Wed, 14 May 2008 07:32:44 +0000 (07:32 +0000)]
gst/interleave/: Split definitions into separate header files for better 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.
Sjoerd Simons [Tue, 13 May 2008 17:21:07 +0000 (17:21 +0000)]
gst/mpeg4videoparse/mpeg4videoparse.*: Parse the config data (either 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.
Wim Taymans [Tue, 13 May 2008 16:16:35 +0000 (16:16 +0000)]
gst/real/gstrealvideodec.c: Add some debug for where we are searching for libraries.
Original commit message from CVS:
* gst/real/gstrealvideodec.c: (open_library):
Add some debug for where we are searching for libraries.
Sjoerd Simons [Tue, 13 May 2008 10:59:49 +0000 (10:59 +0000)]
tests/check/elements/audioresample.c: Add unit test for the latest 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.
Wim Taymans [Tue, 13 May 2008 09:06:51 +0000 (09:06 +0000)]
gst/rtpmanager/gstrtpbin.c: Actually add the do-lost property to the object.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
Actually add the do-lost property to the object.
Wim Taymans [Mon, 12 May 2008 18:43:41 +0000 (18:43 +0000)]
gst/rtpmanager/gstrtpjitterbuffer.c: Avoid waiting for a negative (huge) 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.
Peter Kjellerstedt [Mon, 12 May 2008 14:28:09 +0000 (14:28 +0000)]
gst/rtpmanager/gstrtpsession.c: Make sure to unref the rtpsession returned 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.
Jan Schmidt [Mon, 12 May 2008 14:17:06 +0000 (14:17 +0000)]
docs/plugins/gst-plugins-bad-plugins-sections.txt: Quieten some docs output
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
Quieten some docs output
Jan Schmidt [Mon, 12 May 2008 14:12:08 +0000 (14:12 +0000)]
gst/rtpmanager/gstrtpjitterbuffer.c: Initialise with GST_CLOCK_TIME_NONE to 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.
Jan Schmidt [Sun, 11 May 2008 17:23:20 +0000 (17:23 +0000)]
Random doc of the day: the deinterlace element.
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.
Zaheer Abbas Merali [Fri, 9 May 2008 10:21:07 +0000 (10:21 +0000)]
gst/mpegtsparse/: Make sure all schedule EIT and non-actual transport stream
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.
Peter Kjellerstedt [Fri, 9 May 2008 07:41:58 +0000 (07:41 +0000)]
gst/rtpmanager/rtpsource.c: Make sure to unref the caps used by RTPSource to 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.
Clive Wright [Thu, 8 May 2008 19:16:17 +0000 (19:16 +0000)]
sys/oss4/oss4-mixer-slider.c: Apparently mono sliders have the mono value 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.
Olivier Crete [Thu, 8 May 2008 09:43:33 +0000 (09:43 +0000)]
gst/rtpmanager/rtpsession.c: Unlock the session lock when calling one of our 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.
Sjoerd Simons [Thu, 8 May 2008 06:23:39 +0000 (06:23 +0000)]
gst/rtpmanager/gstrtpsession.c: Send RTP BYE command on EOS. Fixes bug #531955.
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.
Sjoerd Simons [Thu, 8 May 2008 06:20:42 +0000 (06:20 +0000)]
gst/audioresample/gstaudioresample.c: Let audioresample use the buffer 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.
Ole André Vadla Ravnås [Wed, 7 May 2008 20:25:09 +0000 (20:25 +0000)]
win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC 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.
Wim Taymans [Wed, 7 May 2008 10:38:23 +0000 (10:38 +0000)]
gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.
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.
Tim-Philipp Müller [Wed, 7 May 2008 07:51:36 +0000 (07:51 +0000)]
configure.ac: Error out if we don't have the required versions of core/base.
Original commit message from CVS:
* configure.ac:
Error out if we don't have the required versions of core/base.
Wim Taymans [Mon, 5 May 2008 10:27:45 +0000 (10:27 +0000)]
gst-libs/gst/app/gstappsink.*: Start some docs.
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.
Christian Schaller [Fri, 2 May 2008 14:40:08 +0000 (14:40 +0000)]
add wildmidi plugin
Original commit message from CVS:
add wildmidi plugin
Jens Granseuer [Tue, 29 Apr 2008 19:11:56 +0000 (19:11 +0000)]
gst/subenc/gstsrtenc.c: Declare variables at the beginning of blocks. Fixes 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.
Zaheer Abbas Merali [Tue, 29 Apr 2008 09:02:35 +0000 (09:02 +0000)]
gst/mpegtsparse/: Detect SI pids (NIT, SDT, EIT etc.) based on table id and 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.
Sebastian Dröge [Fri, 25 Apr 2008 23:22:56 +0000 (23:22 +0000)]
ext/: Cast NULL sentinels to void * as NULL is defined as an integer constant in most environments when using C++ and...
Original commit message from CVS:
* ext/mpeg2enc/gstmpeg2enc.cc:
* ext/soundtouch/gstbpmdetect.cc:
Cast NULL sentinels to void * as NULL is defined as an integer
constant in most environments when using C++ and it's size might
be different from a pointer.
Wim Taymans [Fri, 25 Apr 2008 18:18:47 +0000 (18:18 +0000)]
gst-libs/gst/app/gstappsink.*: Add more docs.
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_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals),
(gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers),
(gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Add more docs.
Add signals for when preroll and render buffers are available.
Add property to control signal emission.
Add property to control the max queue size.
Michael Smith [Fri, 25 Apr 2008 18:13:07 +0000 (18:13 +0000)]
gst-libs/gst/dshow/Makefile.am: Use CXXFLAGS rather than CFLAGS; these are C++ files.
Original commit message from CVS:
* gst-libs/gst/dshow/Makefile.am:
Use CXXFLAGS rather than CFLAGS; these are C++ files.
Define required constants appropriately.
* sys/dshowdecwrapper/Makefile.am:
Add required include dir, libraries.
Define required constants appropriately.
Wim Taymans [Fri, 25 Apr 2008 11:32:09 +0000 (11:32 +0000)]
gst/rtpmanager/gstrtpbin.*: Expose new jitterbuffer property in rtpbin too.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_stream), (gst_rtp_bin_init),
(gst_rtp_bin_set_property), (gst_rtp_bin_get_property):
* gst/rtpmanager/gstrtpbin.h:
Expose new jitterbuffer property in rtpbin too.
Wim Taymans [Fri, 25 Apr 2008 11:22:13 +0000 (11:22 +0000)]
gst/rtpmanager/gstrtpjitterbuffer.c: Disable sending out rtp packet lost events by default and make a property to ena...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init),
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Disable sending out rtp packet lost events by default and make a
property to enabe it. We will likely enable it by default when the base
depayloaders have a default handler for them so that we don't send these
events all through the pipeline for now.
Wim Taymans [Fri, 25 Apr 2008 09:35:43 +0000 (09:35 +0000)]
gst/rtpmanager/gstrtpjitterbuffer.c: Remove private version of a function that is in -base now.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_src_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Remove private version of a function that is in -base now.
Add src event handler.
Rework the jitterbuffer pushing loop so that it can quickly react to
lost packets and instruct the depayloader of them. This can then be used
to implement error concealment data.
Wim Taymans [Fri, 25 Apr 2008 08:21:06 +0000 (08:21 +0000)]
gst/rtpmanager/gstrtpsession.c: Set up some internal links functions for the RTCP and sync pads because the defaults ...
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_query_send_rtcp_src), (create_recv_rtcp_sink),
(create_send_rtcp_src):
Set up some internal links functions for the RTCP and sync pads because
the defaults are really not correct.
Implement a query handler for the RTCP src pad, mostly to correctly
report about the latency.
Wim Taymans [Fri, 25 Apr 2008 08:15:58 +0000 (08:15 +0000)]
gst/rtpmanager/: Also keep track of the first buffer timestamp together with the first
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain):
* gst/rtpmanager/rtpsession.c: (update_arrival_stats),
(rtp_session_process_sr), (rtp_session_on_timeout):
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(calculate_jitter):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Also keep track of the first buffer timestamp together with the first
RTP timestamp as they both are needed to construct the timing of
outgoing packets in the jitterbuffer and are therefore also needed to
manage lip-sync. This fixes lip-sync if the first RTP packets arrive
with a wildly different gap.
Wim Taymans [Fri, 25 Apr 2008 08:07:36 +0000 (08:07 +0000)]
gst/flv/gstflvdemux.c: Forward unknown queries upstream instead of returning FALSE on them.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_query):
Forward unknown queries upstream instead of returning FALSE on them.
Sebastian Dröge [Thu, 24 Apr 2008 22:19:48 +0000 (22:19 +0000)]
Add support for the new libmpcdec API which magically gets us support for SV8 files. Also do some random cleanup. Fix...
Original commit message from CVS:
* configure.ac:
* ext/musepack/gstmusepackdec.c: (gst_musepackdec_base_init),
(gst_musepackdec_init), (gst_musepackdec_dispose),
(gst_musepackdec_handle_seek_event), (gst_musepack_stream_init),
(gst_musepackdec_loop), (plugin_init):
* ext/musepack/gstmusepackdec.h:
* ext/musepack/gstmusepackreader.c:
* ext/musepack/gstmusepackreader.h:
Add support for the new libmpcdec API which magically gets us support
for SV8 files. Also do some random cleanup. Fixes bug #526905.
Jan Schmidt [Thu, 24 Apr 2008 21:24:18 +0000 (21:24 +0000)]
configure.ac: Back to development -> 0.10.7.1
Original commit message from CVS:
* configure.ac:
Back to development -> 0.10.7.1
=== release 0.10.7 ===
Jan Schmidt [Thu, 24 Apr 2008 00:18:30 +0000 (00:18 +0000)]
Release 0.10.7
Original commit message from CVS:
Release 0.10.7
Jan Schmidt [Thu, 24 Apr 2008 00:15:27 +0000 (00:15 +0000)]
Update .po files
Original commit message from CVS:
Update .po files
Stefan Kost [Tue, 22 Apr 2008 15:07:35 +0000 (15:07 +0000)]
ext/faad/gstfaad.c: Don't leak GstAudioChannelPosition. Fixes #529378.
Original commit message from CVS:
* ext/faad/gstfaad.c:
Don't leak GstAudioChannelPosition. Fixes #529378.
Wim Taymans [Tue, 22 Apr 2008 08:18:05 +0000 (08:18 +0000)]
gst/sdp/gstsdpdemux.c: Ref caps, see #528245.
Original commit message from CVS:
* gst/sdp/gstsdpdemux.c: (request_pt_map):
Ref caps, see #528245.
Jan Schmidt [Tue, 22 Apr 2008 00:21:56 +0000 (00:21 +0000)]
configure.ac: 0.10.6.4 pre-release
Original commit message from CVS:
* configure.ac:
0.10.6.4 pre-release
Sebastian Dröge [Mon, 21 Apr 2008 21:54:11 +0000 (21:54 +0000)]
tests/check/elements/rganalysis.c: Don't leak a tag list. Fixes bug #529285.
Original commit message from CVS:
* tests/check/elements/rganalysis.c: (GST_START_TEST):
Don't leak a tag list. Fixes bug #529285.
Sebastian Dröge [Mon, 21 Apr 2008 21:52:30 +0000 (21:52 +0000)]
tests/check/elements/ofa.c: Don't leak the tags string and tag list. Fixes bug #529283.
Original commit message from CVS:
* tests/check/elements/ofa.c: (bus_handler):
Don't leak the tags string and tag list. Fixes bug #529283.
Olivier Crete [Mon, 21 Apr 2008 08:26:37 +0000 (08:26 +0000)]
gst/rtpmanager/gstrtpbin.c: Ref caps when inserting into the cache.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(new_ssrc_pad_found):
Ref caps when inserting into the cache.
Don't leak pads.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_get_clock_rate),
(gst_rtp_jitter_buffer_query):
Avoid a caps leak.
Don't leak refcount in query.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps),
(gst_rtp_pt_demux_chain):
Avoid caps leaks.
* gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure),
(gst_rtp_session_init), (return_true),
(gst_rtp_session_clear_pt_map), (gst_rtp_session_cache_caps),
(gst_rtp_session_clock_rate):
Ref caps when inserting into the cache.
Fix some more caps leaks. Fixes #528245.
Tim-Philipp Müller [Fri, 18 Apr 2008 18:51:08 +0000 (18:51 +0000)]
tests/icles/metadata_editor.c: Add cast to placate gcc 4.1.2.
Original commit message from CVS:
* tests/icles/metadata_editor.c:
Add cast to placate gcc 4.1.2.
Jan Schmidt [Thu, 17 Apr 2008 23:01:11 +0000 (23:01 +0000)]
configure.ac: 0.10.6.3 pre-release
Original commit message from CVS:
* configure.ac:
0.10.6.3 pre-release
Zaheer Abbas Merali [Thu, 17 Apr 2008 18:28:05 +0000 (18:28 +0000)]
sys/dvb/gstdvbsrc.c: Revert patch that added a loop timeout.
Original commit message from CVS:
* sys/dvb/gstdvbsrc.c:
Revert patch that added a loop timeout.
Fixes #528614.
Wim Taymans [Thu, 17 Apr 2008 07:31:44 +0000 (07:31 +0000)]
gst/rtpmanager/: Unset GValues after g_signal_emitv so that we avoid a refcount leak.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (get_pt_map), (free_client),
(gst_rtp_bin_associate), (gst_rtp_bin_get_free_pad_name):
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_get_clock_rate):
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
Unset GValues after g_signal_emitv so that we avoid a refcount leak.
Don't leak a padname.
Don't leak client streams list.
Lock rtpbin when associating streams. Fixes #528245.
Sebastian Dröge [Wed, 16 Apr 2008 09:50:17 +0000 (09:50 +0000)]
tests/check/Makefile.am: Don't inlcude dc1394src in the generic/states test as it requires special hardware. Fixes bu...
Original commit message from CVS:
* tests/check/Makefile.am:
Don't inlcude dc1394src in the generic/states test as it requires
special hardware. Fixes bug #528011.
Sebastian Dröge [Wed, 16 Apr 2008 09:48:06 +0000 (09:48 +0000)]
tests/check/elements/ofa.c: Only check if the generated fingerprints are valid Base64. The fingerprints are different...
Original commit message from CVS:
* tests/check/elements/ofa.c: (bus_handler), (GST_START_TEST):
Only check if the generated fingerprints are valid Base64. The
fingerprints are different when running on different architectures
which is a) no problem because the fingerprints are tolerant enough
and b) is caused by libofa. Fixes bug #528266.
projects / platform / upstream / gst-plugins-bad.git / log