| Commit message (Collapse) | Author | Age | Files | Lines |
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Original commit message from CVS:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofx.c: (plugin_init):
* gst/audiofx/audiovoice.c: (gst_audio_voice_base_init),
(gst_audio_voice_class_init), (gst_audio_voice_init),
(update_filter), (gst_audio_voice_set_property),
(gst_audio_voice_get_property), (gst_audio_voice_setup),
(gst_audio_voice_transform_int), (gst_audio_voice_transform_float),
(gst_audio_voice_transform_ip):
* gst/audiofx/audiovoice.h:
Add simple voice removal element. Yay karaoke.
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Original commit message from CVS:
Patch by: William M. Brack <wbrack at mmm dot com dot hk>
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format):
Fix potential caps leak.
If we can't get the framerate with an ioctl, try to get it with the
current norm. Fixes #520092.
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don't touch the framerates on the template ca...
Original commit message from CVS:
Patch by: William M. Brack <wbrack at mmm dot com dot hk>
* sys/v4l2/v4l2src_calls.c:
(gst_v4l2src_probe_caps_for_format_and_size):
If we fail to get the frame intervals, simply don't touch the framerates
on the template caps instead of discarding the format. See #520092.
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Original commit message from CVS:
Patch by: William M. Brack <wbrack at mmm dot com dot hk>
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_structure),
(gst_v4l2_get_caps_info):
Add NV12, NV21 and bayer support. See #520092.
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Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_find_segment),
(gst_qtdemux_activate_segment):
Unbreak segment activation again. Fixes #531672.
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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.
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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().
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channels as defined by the FLAC spec. Fix...
Original commit message from CVS:
* ext/flac/Makefile.am:
* ext/flac/gstflacdec.c: (gst_flac_dec_write):
Set the channel layout when decoding FLAC files with more than 2
channels as defined by the FLAC spec. Fixes bug #534570.
Also don't try to decode left/side, right/side and mid/side files
as we don't support this at all.
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Original commit message from CVS:
* configure.ac:
We need -base CVS (rtsp).
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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.
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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.
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EAI_ADDRFAMILY when including netdb.h when bui...
Original commit message from CVS:
* gst/udp/Makefile.am:
Add -D_GNU_SOURCE to CFLAGS so we get things like EAI_ADDRFAMILY
when including netdb.h when building against glibc >= 2.8.
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Original commit message from CVS:
2008-05-22 Julien Moutte <julien@fluendo.com>
* gst/smpte/gstsmptealpha.c: (gst_smpte_alpha_setcaps): Fix
debug statement arguments.
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_setup_qos_dscp):
* gst/udp/gstudpnetutils.c: (gst_udp_join_group),
(gst_udp_leave_group): Fix IP and IPV6 options to make it work
on more platforms.
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Original commit message from CVS:
* tests/check/elements/avimux.c: (setup_src_pad),
(teardown_src_pad):
* tests/check/elements/icydemux.c: (icydemux_found_pad),
(GST_START_TEST):
* tests/check/elements/matroskamux.c: (setup_src_pad),
(teardown_src_pad), (setup_sink_pad), (teardown_sink_pad):
* tests/check/elements/videocrop.c: (video_crop_get_test_caps),
(GST_START_TEST):
* tests/check/elements/wavpackparse.c: (wavpackparse_found_pad),
(setup_wavpackparse), (cleanup_wavpackparse):
Don't use gst_element_get_pad(), it's a bad, bad method.
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Original commit message from CVS:
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_reset),
(do_toggle_element):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset),
(do_toggle_element):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_reset),
(do_toggle_element):
* ext/gconf/gstswitchsink.c: (gst_switch_commit_new_kid):
* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_reset),
(do_toggle_element):
* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_reset),
(do_toggle_element):
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_reset),
(gst_auto_audio_sink_detect):
* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_reset),
(gst_auto_video_sink_detect):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_stream_free), (gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink), (gst_rtspsrc_skip_lws),
(gst_rtspsrc_unskip_lws), (gst_rtspsrc_skip_commas),
(gst_rtspsrc_skip_item), (gst_rtsp_decode_quoted_string),
(gst_rtspsrc_parse_digest_challenge), (gst_rtspsrc_parse_auth_hdr):
* tests/icles/videocrop-test.c: (test_with_caps),
(video_crop_get_test_caps):
Don't use gst_element_get_pad(), it's a bad method.
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totally unrelated tasks are must be separ...
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send),
(gst_multiudpsink_add_internal):
* gst/udp/gstudpnetutils.c: (gst_udp_set_loop_ttl),
(gst_udp_join_group):
* gst/udp/gstudpnetutils.h:
* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
Joining a multicast group and setting the loop/ttl properties are
totally unrelated tasks are must be separated.
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Original commit message from CVS:
* gst/avi/gstavimux.c:
Also support alaw/mulaw.
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Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_setup_qos_dscp), (gst_multiudpsink_add_internal):
* gst/udp/gstmultiudpsink.h:
Add a fixme for the auto-multicast property.
Fix some confusing debug messages.
Disable setting a qos value by default.
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Original commit message from CVS:
Patch by: Gustaf Räntilä <g dot rantila at gmail dot com>
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render):
Ignore EPERM errors from sendto. Fixes #533619.
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service.
Original commit message from CVS:
Patch by: Henrik Eriksson <henriken at axis dot com>
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_init), (gst_multiudpsink_setup_qos_dscp),
(gst_multiudpsink_set_property), (gst_multiudpsink_get_property),
(gst_multiudpsink_init_send), (gst_multiudpsink_add_internal):
* gst/udp/gstmultiudpsink.h:
Add qos-dscp property to manage the Quality of service.
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Original commit message from CVS:
* gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_process):
Improve debugging of the ident.
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IPv4 and IPv6 transparently.
Original commit message from CVS:
Patch by: Bruno Santos <brunof at ua dot pt>
* gst/udp/gstudpnetutils.c: (gst_udp_get_addr),
(gst_udp_join_group), (gst_udp_leave_group),
(gst_udp_is_multicast):
* gst/udp/gstudpnetutils.h:
Provide a bunch of helper methods to deal with IPv4 and IPv6
transparently.
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_init), (gst_multiudpsink_set_property),
(gst_multiudpsink_get_property), (join_multicast),
(gst_multiudpsink_init_send), (gst_multiudpsink_add_internal),
(gst_multiudpsink_remove):
* gst/udp/gstmultiudpsink.h:
Add multicast TTL and loopback properties.
Use the helper methods to implement ip4 and ip6.
* gst/udp/gstudpsrc.c: (gst_udpsrc_create), (gst_udpsrc_start):
* gst/udp/gstudpsrc.h:
Use the helper methods to implement ip4 and ip6.
Fixes #515962.
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name because they not always map directl...
Original commit message from CVS:
Patch by: Patrick Radizi <patrick dot radizi at axis dot com>
* gst/multipart/multipartdemux.c: (gst_multipart_demux_class_init),
(gst_multipart_demux_get_gstname),
(gst_multipart_find_pad_by_mime), (gst_multipart_demux_chain):
* gst/multipart/multipartdemux.h:
Don't blindly copy the mime-type as the caps name because they not
always map directly. Instead use a hashtable with common mappings.
Fixes #533287.
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parent know that we cannot write the segment e...
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_write):
When we post an error, we must return -1 to let the parent know that we
cannot write the segment else it will loop and continue to call us again
forever. Patch by Michael Meeks.
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Original commit message from CVS:
* gst/videomixer/videomixer.c:
Add missing incudes.
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Original commit message from CVS:
* gst/rtp/gstrtph264pay.c: (gst_h264_scan_mode_get_type),
(gst_rtp_h264_pay_handle_buffer):
* gst/rtp/gstrtph264pay.h:
Correct a typo (sinle -> single).
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quicktime-like AVC output in addition to the exist...
Original commit message from CVS:
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
(gst_rtp_h264_depay_init), (gst_rtp_h264_depay_set_property),
(gst_rtp_h264_depay_get_property), (gst_rtp_h264_depay_setcaps),
(gst_rtp_h264_depay_process):
* gst/rtp/gstrtph264depay.h:
Add experimental support for outputting quicktime-like AVC output in
addition to the existing bytestream output.
* gst/rtp/gstrtph264pay.c: (gst_h264_scan_mode_get_type),
(gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init),
(gst_rtp_h264_pay_setcaps), (gst_rtp_h264_pay_payload_nal),
(gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property),
(gst_rtp_h264_pay_get_property):
* gst/rtp/gstrtph264pay.h:
Make the parsing mode configurable, for some inputs we don't need to
scan every byte for start codes.
Only set the marker bit on ACCESS units.
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intermediate results to prevent overflows....
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
Use a bigger type in integer mode for the intermediate results to
prevent overflows. This fixes the crippled sound when using the
equalizer in integer mode. Fixes bug #510865.
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id's, use a counter.
Original commit message from CVS:
* gst/videomixer/videomixer.c:
* gst/videomixer/videomixer.h:
Instead of a random number for the request pad id's,
use a counter.
Register the videomixerpad class from the element's class_init
where it's safer, and allows the docs generator to scan it.
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channel of
Original commit message from CVS:
* gst/smpte/Makefile.am:
* gst/smpte/gstsmpte.c: (gst_smpte_plugin_init):
* gst/smpte/gstsmpte.h:
* gst/smpte/gstsmptealpha.c:
(gst_smpte_alpha_transition_type_get_type),
(gst_smpte_alpha_get_type), (gst_smpte_alpha_base_init),
(gst_smpte_alpha_class_init), (gst_smpte_alpha_update_mask),
(gst_smpte_alpha_setcaps), (gst_smpte_alpha_get_unit_size),
(gst_smpte_alpha_init), (gst_smpte_alpha_finalize),
(gst_smpte_alpha_do_ayuv), (gst_smpte_alpha_do_i420),
(gst_smpte_alpha_transform), (gst_smpte_alpha_set_property),
(gst_smpte_alpha_get_property), (gst_smpte_alpha_plugin_init):
* gst/smpte/gstsmptealpha.h:
* gst/smpte/plugin.c: (plugin_init):
Add new plugin that adds the SMPTE transition in the alpha channel of
I420 and AYUV frames so that they can be blended with videomixer later
on. Uses all niceties such as using base transform for efficient alloc
and negotiation. It currently requires GstController to control the
position in the transition effect.
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Aparently works for the Eq. For VideoMixer th...
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* docs/plugins/gst-plugins-good-plugins.interfaces:
* docs/plugins/gst-plugins-good-plugins.types:
* gst/videomixer/videomixer.c:
Try using thaytans new mechanism to get extra classes into plugin
docs. Aparently works for the Eq. For VideoMixer the GObject stuff is
missing still.
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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.
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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.
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also do some more sanity checks on caps ...
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_audsink_set_caps):
Set proper rate in avi stream header for PCM audio, and also do some
more sanity checks on caps in this case. Fixes #511489.
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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.
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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.
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XOverlay support isn't implemented at all ye...
Original commit message from CVS:
* sys/v4l2/gstv4l2src.c:
Don't include the gstv4l2xoverlay.h header as the XOverlay support
isn't implemented at all yet and this requires X headers to be
installed. Fixes bug #533264.
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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.
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Original commit message from CVS:
* ext/wavpack/gstwavpackstreamreader.c:
* tests/examples/spectrum/demo-audiotest.c:
* tests/examples/spectrum/demo-osssrc.c:
Fix some compiler warnings.
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Original commit message from CVS:
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
Small comment added.
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_class_init),
(gst_rtp_h264_pay_decode_nal), (gst_rtp_h264_pay_parse_sps_pps),
(gst_rtp_h264_pay_payload_nal), (gst_rtp_h264_pay_handle_buffer):
Debug string cleanups (remove trailing \n)
Refactor and clean up the payloader a bit and make sure that we only
put one NAL unit in an RTP packet even if the input buffer contains
multiple NAL units.
Add suport for AVC format input.
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sprop-parameter-sets using properties in ca...
Original commit message from CVS:
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_class_init),
(gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_handle_buffer),
(gst_rtp_h264_pay_set_property), (gst_rtp_h264_pay_get_property):
* gst/rtp/gstrtph264pay.h:
Make it possible to specify profile-level-id and sprop-parameter-sets
using properties in case they are not available in-stream.
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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.
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Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-ladspa.xml:
Remove ladspa fro plugin-docs, its in gst-plugins-bad.
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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.
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later seeking, and to forego sync attempts.
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_start_file):
Send an initial BYTE segment to inform downstream of later seeking,
and to forego sync attempts.
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Original commit message from CVS:
* gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_setcaps):
Fix wrong caps string.
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Original commit message from CVS:
Based on patch by: Olivier Crete <tester at tester dot ca>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_base_init),
(gst_rtp_g729_depay_class_init), (gst_rtp_g729_depay_init),
(gst_rtp_g729_depay_setcaps), (gst_rtp_g729_depay_process),
(gst_rtp_g729_depay_plugin_init):
* gst/rtp/gstrtpg729depay.h:
* gst/rtp/gstrtpg729pay.c: (gst_rtpg729pay_base_init),
(gst_rtpg729pay_class_init), (gst_rtpg729pay_init),
(gst_rtpg729pay_setcaps), (gst_rtp_g729_pay_plugin_init):
* gst/rtp/gstrtpg729pay.h:
Added G729 pay and depayloaders. Fixes #532409.
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concealment packets.
Original commit message from CVS:
* ext/speex/gstspeexdec.c: (speex_dec_sink_event):
Fix the calculation of the duration of the concealment packets.
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Original commit message from CVS:
Based on patch by: Olivier Crete <tester at tester dot ca>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpdvdepay.c: (gst_rtp_dv_depay_base_init),
(gst_rtp_dv_depay_class_init), (gst_rtp_dv_depay_init),
(parse_encode), (gst_rtp_dv_depay_setcaps),
(calculate_difblock_location), (gst_rtp_dv_depay_process),
(gst_rtp_dv_depay_reset), (gst_rtp_dv_depay_change_state),
(gst_rtp_dv_depay_plugin_init):
* gst/rtp/gstrtpdvdepay.h:
* gst/rtp/gstrtpdvpay.c: (gst_dv_pay_mode_get_type),
(gst_rtp_dv_pay_base_init), (gst_rtp_dv_pay_class_init),
(gst_rtp_dv_pay_init), (gst_dv_pay_set_property),
(gst_dv_pay_get_property), (gst_rtp_dv_pay_setcaps),
(gst_dv_pay_negotiate), (include_dif),
(gst_rtp_dv_pay_handle_buffer), (gst_rtp_dv_pay_plugin_init):
* gst/rtp/gstrtpdvpay.h:
Add DV pay and depayloaders. Fixes #532423.
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private data from VobSub's (buggy) RGB to YUV.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_push_dvd_clut_change_event):
Convert subtitle palette info in VobSub private data from VobSub's
(buggy) RGB to YUV.
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