- Jun 18, 2008
-
-
Jan Schmidt authored
Original commit message from CVS: Release 0.10.20
-
Jan Schmidt authored
Original commit message from CVS: Update .po files
-
Stefan Kost authored
Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * examples/app/appsrc-ra.c: * examples/app/appsrc-seekable.c: * examples/app/appsrc-stream.c: * examples/app/appsrc-stream2.c: * ext/directfb/dfbvideosink.h: * ext/metadata/gstbasemetadata.c: * ext/metadata/gstbasemetadata.h: * ext/metadata/metadata.c: * ext/metadata/metadataexif.c: * ext/theora/theoradec.h: * gst/deinterlace2/gstdeinterlace2.h: * gst/deinterlace2/tvtime/speedy.c: * gst/deinterlace2/tvtime/speedy.h: * gst/deinterlace2/tvtime/vfir.c: Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments.
-
- Jun 16, 2008
-
-
Andy Wingo Wingo authored
Original commit message from CVS: 2008-06-16 Andy Wingo <wingo@pobox.com> * gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes) (gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().
-
Stefan Kost authored
Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/speed/gstspeed.c: * gst/speexresample/gstspeexresample.c: * gst/videosignal/gstvideoanalyse.c: * gst/videosignal/gstvideodetect.c: * gst/videosignal/gstvideomark.c: * sys/dvb/gstdvbsrc.c: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: * sys/wininet/gstwininetsrc.c: Final round of doc updates.
-
- Jun 13, 2008
-
-
Stefan Kost authored
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-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-gstinterlace.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-mythtv.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-oss4.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-subenc.xml * docs/plugins/inspect/plugin-timidity.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: * ext/amrwb/gstamrwbdec.c: * ext/amrwb/gstamrwbenc.c: * ext/amrwb/gstamrwbparse.c: * ext/dc1394/gstdc1394.c: * ext/directfb/dfbvideosink.c: * ext/ivorbis/vorbisdec.c: * ext/jack/gstjackaudiosink.c: * ext/mpeg2enc/gstmpeg2enc.cc: * ext/mplex/gstmplex.cc: * ext/musicbrainz/gsttrm.c: * ext/mythtv/gstmythtvsrc.c: * ext/theora/theoradec.c: * ext/timidity/gsttimidity.c: * ext/timidity/gstwildmidi.c: * gst-libs/gst/app/gstappsink.c: * gst/deinterlace/gstdeinterlace.c: * gst/dvdspu/gstdvdspu.c: * gst/festival/gstfestival.c: * gst/freeze/gstfreeze.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/modplug/gstmodplug.cc: * gst/nuvdemux/gstnuvdemux.c: Add missing elements to docs. Fix doc-markup: use convinience syntax for examples (produces valid docbook), add several refsec2 when we have several titles. Fix some types.
-
- Jun 12, 2008
-
-
Wim Taymans authored
examples/app/: Add beefed up example app from bug #413418. It now also uses appsink instead of fakesink for more ulti... Original commit message from CVS: * examples/app/.cvsignore: * examples/app/Makefile.am: * examples/app/appsink-src.c: (on_new_buffer_from_source), (on_source_message), (on_sink_message), (main): Add beefed up example app from bug #413418. It now also uses appsink instead of fakesink for more ultimate coolness. * 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_create), (gst_app_src_set_max_bytes), (gst_app_src_push_buffer), (gst_app_src_end_of_stream): * gst-libs/gst/app/gstappsrc.h: Add block property to allow push based implementation to block when we fill up the appsrc queues. Emit the enough-data signal while releasing our lock.
-
Stefan Kost authored
Original commit message from CVS: * examples/app/.cvsignore: Ignore more.
-
Stefan Kost authored
Do not use short_description in section docs for elements. We extract them from element details and there will be war... Original commit message from CVS: * ext/dc1394/gstdc1394.c: * ext/ivorbis/vorbisdec.c: * ext/jack/gstjackaudiosink.c: * ext/metadata/gstmetadatademux.c: * ext/mythtv/gstmythtvsrc.c: * ext/theora/theoradec.c: * gst-libs/gst/app/gstappsink.c: * gst/bayer/gstbayer2rgb.c: * gst/deinterlace/gstdeinterlace.c: * gst/rawparse/gstaudioparse.c: * gst/rawparse/gstvideoparse.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/selector/gstinputselector.c: * gst/selector/gstoutputselector.c: * gst/videosignal/gstvideoanalyse.c: * gst/videosignal/gstvideodetect.c: * gst/videosignal/gstvideomark.c: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: Do not use short_description in section docs for elements. We extract them from element details and there will be warnings if they differ. Also fixing up the ChangeLog order.
-
- Jun 11, 2008
-
-
Jan Schmidt authored
Original commit message from CVS: * configure.ac: 0.10.19.3 pre-release
-
David Schleef authored
Original commit message from CVS: * gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32. Patch By: David Schleef <ds@schleef.org> Fixes: #536874
-
Sebastian Dröge authored
ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first read returns less than requeste... Original commit message from CVS: * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize), (gst_gio_base_src_create): * ext/gio/gstgiobasesrc.h: Try to read the requested number of bytes, even if the first read returns less than requested, until nothing is read anymore or we have the requested amount of bytes. This fixes playback of files via Samba as Samba only allows to read 64k at once. Implement a caching algorithm that makes sure that we read at least 4k of data every time. Some elements will try to read a few bytes, then seek, read again a few bytes and so on and this is painfully slow as every operation has to go over DBus if GVfs is used as backend. Fixes bug #536849 and #536848. * ext/gio/gstgiosrc.c: (gst_gio_src_class_init), (gst_gio_src_check_get_range): Override check_get_range() to blacklist http/https URIs and whitelist file URIs. More to be added on demand.
-
- Jun 06, 2008
-
-
Wim Taymans authored
examples/app/: Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull ... Original commit message from CVS: * examples/app/Makefile.am: * examples/app/appsrc-ra.c: (feed_data), (seek_data), (found_source), (bus_message), (main): * examples/app/appsrc-seekable.c: (feed_data), (seek_data), (found_source), (bus_message), (main): * examples/app/appsrc-stream2.c: (feed_data), (found_source), (bus_message), (main): Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull mode seekable. * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init), (gst_app_src_start), (gst_app_src_do_get_size), (gst_app_src_create): * gst-libs/gst/app/gstappsrc.h: Make stream-type property writable. Unset flushing when starting so that we reuse appsrc. Inform basesrc about the configured size. Emit seek-data signal when we are going to a different offset in random-access mode.
-
Wim Taymans authored
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.
-
- Jun 05, 2008
-
-
Wim Taymans authored
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.
-
Jan Schmidt authored
Original commit message from CVS: * configure.ac: 0.10.19.2 pre-release
-
- Jun 04, 2008
-
-
Jan Schmidt authored
Original commit message from CVS: * win32/common/libgstrtsp.def: * win32/common/libgsttag.def: Add new API functions to the dll exports
-
Michael Smith authored
gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo... Original commit message from CVS: * gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avoid crashes if the decodebin is eventually disposed after the playbin itself (possible if the app takes a reference on the decodebin). Fixes #536521.
-
Tim-Philipp Müller authored
gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo... Original commit message from CVS: * gst/typefind/gsttypefindfunctions.c: (aac_type_find), (mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE), (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find), (h264_video_type_find), (mpeg_video_stream_type_find), (dv_type_find), (mmsh_type_find): Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps for no good reason (this may be desirable to make it easier to detect leaks, but then it should probably be done for all caps in the typefinder somewhere).
-
Peter Kjellerstedt authored
Original commit message from CVS: * tests/check/Makefile.am: Do not try to run the check tests for subparse unless it has been built.
-
Peter Kjellerstedt authored
tests/check/pipelines/streamheader.c: Do not try to run a test which requires vorbisenc unless we have actually built... Original commit message from CVS: * tests/check/pipelines/streamheader.c: (buffer_probe_cb), (test_multifdsink_gdp_vorbisenc), (streamheader_suite): Do not try to run a test which requires vorbisenc unless we have actually built it.
-
Peter Kjellerstedt authored
Original commit message from CVS: * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param), (gst_rtsp_connection_clear_auth_params), (gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip): * gst-libs/gst/rtsp/gstrtspconnection.h: Add a couple of missing argument guards. Add a way of setting the DSCP for an RTSP connection. Add an accessor method for the ip member of GstRTSPConnection as all members are supposed to be private.
-
Peter Kjellerstedt authored
Original commit message from CVS: * gst/tcp/gstmultifdsink.c: (setup_dscp_client): Fixed accidental use of IPv4 options for all IPv6 addresses.
-
Tim-Philipp Müller authored
Original commit message from CVS: * gst-libs/gst/interfaces/mixertrack.h: Document mixer track flags.
-
gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul... Original commit message from CVS: Patch by: Antoine Tremblay <hexa00 at gmail dot com> * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader): Don't set caps on the buffers that contain a copy of the buffer including the caps of them resulting in an always increasing refcount of the caps and insanely large caps. Instead include a buffer without caps in the new caps. Fixes bug #536475.
-
Sebastian Dröge authored
gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ... Original commit message from CVS: * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps): Transform a given PAR to a range on the struct with the generic height/width instead of the struct with the possibly restricted height/width.
-
Sebastian Dröge authored
gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ... Original commit message from CVS: * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps): Prefer the given format if it contains something stricter than [1,MAX] for height or width and only put a structure that requires rescaling as second. This makes it possible to use videoscale in pipelines where the source can actually produce the wanted height/width but usually selects a different one from the requested.
-
- Jun 03, 2008
-
-
Original commit message from CVS: Based on patch by: John Millikin <jmillikin gmail com> * gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add), (gst_vorbis_tag_add_coverart): Retrieve COVERART tags from vorbis comments (#512333)
-
Tim-Philipp Müller authored
gst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use glib-mkenums here...). Original commit message from CVS: * gst-libs/gst/tag/tag.h: * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum): Don't forget to add new enum value here too (should probably use glib-mkenums here...).
-
Tim-Philipp Müller authored
Original commit message from CVS: * gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image): * gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE), * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum), (gst_tag_image_type_get_type), (gst_tag_image_type_is_valid), (gst_tag_image_data_to_image_buffer): Add two utility functions to avoid code duplication (#512333): API: add gst_tag_image_data_to_image_buffer() API: add gst_tag_list_add_id3_image()
-
Sebastian Dröge authored
Original commit message from CVS: * win32/common/libgstaudio.def: Add gst_audio_check_channel_positions() to the exported symbols.
-
Sebastian Dröge authored
Original commit message from CVS: * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/audio/multichannel.c: (gst_audio_check_channel_positions): * gst-libs/gst/audio/multichannel.h: API: Make gst_audio_check_channel_positions() public. * tests/check/libs/audio.c: (GST_START_TEST): Add some simple checks for gst_audio_check_channel_positions().
-
- Jun 02, 2008
-
-
Tim-Philipp Müller authored
Original commit message from CVS: * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names): minrange and maxrange are scaled according to the frequency multiplier.
-
Tim-Philipp Müller authored
ext/pango/: Use gstvideo functions to calculate strides and plane offsets. Fixes rendering issue ('ghost' images of t... Original commit message from CVS: * ext/pango/Makefile.am: * ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y), (gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame): Use gstvideo functions to calculate strides and plane offsets. Fixes rendering issue ('ghost' images of the text on the chroma planes) with widths or heights that are not multiples of 8 (#506659 and probably also #485729). * tests/icles/test-textoverlay.c: (show_text), (test_textoverlay), (main): Test with odd height/width too.
-
Sebastian Dröge authored
Original commit message from CVS: * gst/adder/gstadder.c: (gst_adder_query_duration), (gst_adder_query_latency): When using gst_element_iterate_pads() one has to unref every pad after usage.
-
- May 31, 2008
-
-
Mark Nauwelaerts authored
gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication. Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): Add a gtk-doc chunk for the new properties to have a Since: indication.
-
Mark Nauwelaerts authored
Original commit message from CVS: ChangeLog surgery, mark API change
-
Mark Nauwelaerts authored
gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref... Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init), (gst_base_audio_src_dispose), (gst_base_audio_src_get_property), (gst_base_audio_src_setcaps), (gst_base_audio_src_change_state): Provide readable actual-buffer-time and actual-latency-time properties that reflect the configured ringbuffer values. Fixes #524724.
-
- May 30, 2008
-
-
Wim Taymans authored
gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on... Original commit message from CVS: * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push), (gst_basertppayload_change_state): Simply converting the running time into an RTP timestamp by scaling it based on the clock-rate is good enough for making an RTP timestamp. This has the added benefit that we can later on expose a property with the RTP timestamp of running time 0, as is needed for RTSP servers to generate the response of the PLAY request.
-
Sebastian Dröge authored
gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ... Original commit message from CVS: * gst/audioconvert/gstaudioconvert.c: (structure_has_fixed_channel_positions), (gst_audio_convert_transform_caps): Allow up to 11 positioned channels now that audioconvert can handle this but add no default positions for > 8 channels. * tests/check/elements/audioconvert.c: (GST_START_TEST): Add some unit tests for the above change: Test conversion of 11 positioned channels to stereo and the other way around, test conversion of 15 unpositioned channels in different ways.
-