Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read), (gst_ring_buffer_clear):
Don't assert on normal stuff.
* gst/playback/gstplaybin.c: (do_playbin_seek):
API fix.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
(do_playbin_seek), (gst_play_bin_send_event):
Override send_event differently, so that we can takes bits of
functionality from GstPipeline (special handling for seeks,
including pausing/resuming, and resetting stream time) and
still get
the appropriate behaviour of only forwarding event to a single
sink,
rather than all of them.
Unfortunately requires a lot of code duplication, but the
alternatives are equally ugly in the end.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_destroy),
(gen_preroll_element), (remove_groups), (setup_source):
* gst/playback/gstplaybin.c: (remove_sinks), (add_sink),
(setup_sinks), (gst_play_bin_send_event),
(gst_play_bin_change_state):
Set state to NULL before removing from bin. Fix refcounting.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_send_event):
Correct refcounting in send_event() function. Previously was wrong
if the first sink was unable to handle the event.
Original commit message from CVS:
2005-10-03 Andy Wingo <wingo@pobox.com>
* gst/playback/gstdecodebin.c (try_to_link_1)
(remove_element_chain): set element to NULL before removing it.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_send_event):
Only seek on one sink, the first one that succeeds.
Original commit message from CVS:
2005-09-21 Andy Wingo <wingo@pobox.com>
* gst/playback/gstplaybasebin.c: Attempt to fix up buffer probe
thingies.
* gst/playback/gstdecodebin.c (gst_decode_bin_dispose): Dispose
can be called multiple times, dogs.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: free plugin list correctly
* gst/playback/gstplaybin.c: emit warning if autovideosink
and autoaudiosink can't be found (instead of segfaulting)
Original commit message from CVS:
* check/generic/states.c:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
* gst/playback/gstdecodebin.c: (gst_decode_bin_init):
Fixes for changes in registry API.
* configure.ac: Only export gst_plugins_desc. Add -no-undefined
to GST_PLUGIN_LDFLAGS.
* ext/libvisual/visual.c: Make the library shut up.
* gst-libs/gst/audio/audio.c: Don't define a plugin in a library.
* gst-libs/gst/audio/gstaudiofilter.c: same
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
Audioconvert derives from GstBaseTransform and should
link to the library with our base elements to avoid
unresolved symbols. Makes things work with MinGW (#316160)
* gst/playback/test4.c: (main):
Fix MinGW build problem and use g_usleep() instead of
sleep() (#316162)
Original commit message from CVS:
* check/Makefile.am:
* configure.ac:
add core's plugins to the mix so that playbin works
* check/generic/states.c: (GST_START_TEST):
set a 0 timeout on pipelines, so they don't force the next
state change
* gst/playback/gstplaybasebin.c: (setup_source), (prepare_output),
(gst_play_base_bin_change_state):
remove the crappy error handling and do GST error handling
Original commit message from CVS:
* check/Makefile.am:
Add CHECK_CFLAGS and LDFLAGS
* gst/playback/gstplaybasebin.c: (fill_buffer):
GST_MESSAGE_SRC became a GObject
Original commit message from CVS:
* gst/playback/gstplaybin.c: (remove_sinks):
Remove visualization from parent explicitely; works around some
apparent refcount issue that I haven't tracked down yet.
Original commit message from CVS:
* common/gtk-doc-plugins.mak:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
renamed to actual element names, so much nicer to look at
* docs/plugins/tmpl/gstmultifdsink.sgml:
remove
* docs/plugins/tmpl/multifdsink.sgml:
* docs/plugins/tmpl/tcpserversink.sgml:
add
* ext/alsa/gstalsa.c:
* ext/cdparanoia/gstcdparanoia.c: (cdparanoia_get_property):
* ext/ogg/gstoggmux.c:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
* gst/playback/gstdecodebin.c:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init):
* gst/tcp/gsttcpserversink.c:
various fixes and documentation additions
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element):
Switch to auto*sink elements as default sinks; add volume element
so that volume control in totem works.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gen_preroll_element):
* gst/playback/gstplaybin.c: (setup_sinks),
(gst_play_bin_change_state):
Refcount fix and more comments.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (fill_buffer):
Use _new_custom() so we can set custom message types for buffering
messages.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered),
(mute_stream), (silence_stream):
Small debug additions.
Original commit message from CVS:
make GST_PLUGIN_LDFLAGS only be flags; GST_LIBS should be
added manually to each Makefile.am so we are sure it goes
*last* and doesn't add -L flags before linking in libs of our
own, like, say, internal .la libs, that then accidentally pick
up the installed copy.
Original commit message from CVS:
2005-07-05 Andy Wingo <wingo@pobox.com>
* gst/playback/gstplaybasebin.c (fill_buffer):
message_new_application fixen.
x
Original commit message from CVS:
2005-06-27 Andy Wingo <wingo@pobox.com>
* gst/videotestsrc/gstvideotestsrc.c
(gst_videotestsrc_activate_push): Activation API changes.
* gst/playback/gstdecodebin.c (gst_decode_bin_change_state)
(gst_decode_bin_dispose): Free dynamics in READY->NULL, because
they have refs on the decodebin.
* ext/ogg/gstoggdemux.c (gst_ogg_pad_class_init): Ref the right
parent class.
(gst_ogg_pad_typefind): Don't leak a pad ref.
(gst_ogg_chain_new_stream): gst_object_unref, not g_object_unref.
(gst_ogg_demux_sink_activate, gst_ogg_demux_sink_activate_push)
(gst_ogg_demux_sink_activate_pull): Changes for activation API.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
2nd argument of 'unknow-type' signal is a GstCaps and not a
GstMiniObject
Original commit message from CVS:
2005-06-25 Jan Schmidt <thaytan@mad.scientist.com>
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
Set the worker thread's running flag to TRUE before starting the
thread.
* gst/playback/gstdecodebin.c: (gst_decode_bin_init):
Catch a failure to add typefind to the bin.
Original commit message from CVS:
2005-06-02 Andy Wingo <wingo@pobox.com>
* pkgconfig/gstreamer-libs-uninstalled.pc.in (prefix):
* pkgconfig/gstreamer-libs.pc.in (prefix): Add gst/tag to the -L
list.
* gst/playback/gstdecodebin.c (gst_decode_bin_dispose): Don't
remove the typefind, the bin dispose will do it for us. When it's
removed and unreffed, the signal handler will be disconnected,
too.
(unlinked): It's too difficult to disconnect from unlinked
handlers, as they are on pads not elements. Just punt if the pads
aren't grandkids of the bin.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (probe_triggered):
Fix missing unlock.
* gst/playback/gstplaybin.c: (add_sink):
First add, then link (otherwise pad link fails).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gen_preroll_element):
Increase buffer for video, decrease buffer for other media types.
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element):
Change names for debugging purposes.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(fill_buffer), (check_queue), (queue_threshold_reached),
(queue_out_of_data):
* gst/playback/gstplaybasebin.h:
Post buffer-fullness on the bus.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_type),
(gst_play_base_bin_class_init), (gst_play_base_bin_finalize),
(get_active_group), (get_building_group), (group_destroy),
(group_commit), (check_queue), (queue_overrun),
(queue_threshold_reached), (queue_out_of_data),
(gen_preroll_element), (remove_groups), (unknown_type),
(add_element_stream), (no_more_pads), (probe_triggered),
(preroll_unlinked), (new_decoded_pad), (setup_subtitle),
(setup_substreams), (setup_source), (finish_source),
(prepare_output), (muted_group_change_state),
(gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
(gst_play_base_bin_change_state):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_init), (gst_play_bin_set_property),
(gen_video_element), (gen_text_element), (gen_audio_element),
(gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
(gst_play_bin_change_state):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init),
(cb_probe), (gst_stream_info_new), (gst_stream_info_dispose),
(stream_info_change_state), (gst_stream_info_set_mute),
(gst_stream_info_get_property):
* gst/playback/gststreaminfo.h:
* gst/playback/gststreamselector.c: (gst_stream_selector_init),
(gst_stream_selector_get_linked_pad),
(gst_stream_selector_getcaps),
(gst_stream_selector_get_linked_pads),
(gst_stream_selector_request_new_pad), (gst_stream_selector_chain):
* gst/playback/gststreamselector.h:
Rough port of playbin. Needs some more work, but is mostly done,
and uses a few locks in important places, which should make stuff
like chain-switches clean. Still uses GST_STATE() in a few places,
which isn't all that good an idea, subtitles/elements disabled
because no elements to test with and thus probably broken, query
and event handling moved to GstBin, internal thread removed
alltogether because the pipeline does that for us now. Can play
Ogg/Vorbis files. Haven't tested anything else yet.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c (vorbis_handle_comment_packet): Post a
message to the bus with the tags. Still not sent downstream tho.
* gst/playback/gstdecodebin.c (remove_element_chain): Unref after
get_parent.
(remove_element_chain): Use OBJECT_PARENT instead of get_parent to
avoid refcounting hassles.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_factory_filter):
* gst/playback/gstdecodebin.c: (find_compatibles):
Work with staticpadtemplates in elementfactories.
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_stream_selector_init),
(gst_stream_selector_get_caps), (gst_stream_selector_chain):
* gst/playback/gststreamselector.h:
Be more selective when we're redoing caps negotiation from
within the chain function on a stream change.
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_push),
(gst_a52dec_handle_event), (gst_a52dec_chain):
Add some debug output. Check that a discont has a valid
time associated.
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event),
(gst_alsa_sink_loop):
Ignore TAG events. A little extra debug for broken timestamps.
* ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_init), (dvdnavsrc_loop),
(dvdnavsrc_change_state):
Ensure we send a discont to engage the link before we send any
other events.
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_init),
(dvdreadsrc_finalize), (_close), (_open), (_seek_title),
(_seek_chapter), (seek_sector), (dvdreadsrc_get),
(dvdreadsrc_uri_get_uri), (dvdreadsrc_uri_set_uri):
Handle URI of the form dvd://title[,chapter[,angle]]. Currently only
dvd://title works in totem because typefinding sends a seek that ends
up going back to chapter 1 regardless.
* ext/mpeg2dec/gstmpeg2dec.c:
* ext/mpeg2dec/gstmpeg2dec.h:
Output correct timestamps and handle disconts.
* ext/ogg/gstoggdemux.c: (get_relative):
Small guard against a null dereference.
* ext/pango/gsttextoverlay.c: (gst_textoverlay_finalize),
(gst_textoverlay_set_property):
Free memory when done. Don't call gst_event_filler_get_duration on
EOS events. Use GST_LOG and GST_WARNING instead of g_message and
g_warning.
* ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init),
(draw_line), (gst_smoothwave_dispose), (gst_sw_sinklink),
(gst_sw_srclink), (gst_smoothwave_chain):
Draw solid lines, prettier colours.
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init):
Add a default palette that'll work for some movies.
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_init),
(gst_dvd_demux_handle_dvd_event), (gst_dvd_demux_send_discont),
(gst_dvd_demux_send_subbuffer), (gst_dvd_demux_reset):
* gst/mpegstream/gstdvddemux.h:
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_send_discont),
(gst_mpeg_demux_parse_syshead), (gst_mpeg_demux_parse_pes):
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_init),
(gst_mpeg_parse_handle_discont), (gst_mpeg_parse_parse_packhead):
* gst/mpegstream/gstmpegparse.h:
Use PTM/NAV events when for timestamp adjustment when connected to
dvdnavsrc. Don't use many discont events where one suffices.
* gst/playback/gstplaybasebin.c: (group_destroy),
(gen_preroll_element), (gst_play_base_bin_add_element):
* gst/playback/gstplaybasebin.h:
Make sure we remove subtitles from the same bin we put them in.
* gst/subparse/gstsubparse.c: (convert_encoding), (parse_subrip),
(gst_subparse_buffer_format_autodetect),
(gst_subparse_change_state):
Fix some memleaks and invalid accesses.
* gst/typefind/gsttypefindfunctions.c: (ogganx_type_find),
(oggskel_type_find), (cmml_type_find), (plugin_init):
Some typefind functions for Annodex v3.0 files
* gst/wavparse/gstwavparse.h:
GstRiffReadClass is the correct parent class.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_change_state):
Explicit state change to workaround refcount bugs.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_element_data),
(gst_riff_read_element_data):
* gst-libs/gst/riff/riff-read.h:
Add _peek version (req'ed in CDXA).
* gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init),
(gst_cdxaparse_loop):
Fix parsing in playbin.
* gst/playback/gstdecodebin.c: (close_pad_link):
Ignore current_ pads, they cause major annoyance.
Original commit message from CVS:
* gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
Elements may already be destroyed when this function is called.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(gst_play_base_bin_init), (gst_play_base_bin_dispose),
(probe_triggered), (new_decoded_pad), (gen_source_element),
(gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_init), (group_switch), (remove_sinks), (setup_sinks),
(gst_play_bin_change_state):
Implement group-switch signal for use in apps to clear metadata
cache, clean up subtitle, add suburi property instead of # hack,
some error-out fixes.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
Don't bail on unknown events.
* gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
Don't crash on events before negotiation.
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Send tags on pads, too.
* gst/playback/gststreamselector.c:
(gst_stream_selector_request_new_pad):
Forward events on first pad if no input was selected yet.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_substreams):
Don't disable streamtype if the stream doesn't exist, since
then playing a video after audio will disable both and nothing
will happen. Fixes the testsuite.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_destroy):
Remove hack to get rid of assert and get rid of unlinked
signals properly.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source):
Set source to NULL so that resources are free'ed. Fixes issues
with playback of CDDA and similar device-accessing things.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
(gen_preroll_element), (remove_groups), (setup_subtitle),
(gen_source_element), (setup_source):
* gst/playback/gstplaybasebin.h:
Multiple .sub files is just a stupid idea... Fix some threading
mistakes. Interestingly, external .sub files cause playbin to
hang, I don't know why... Parsing fixes contributed by Felix
Kooman <fkooman@tuxed.net>.
Original commit message from CVS:
patch by: Tim-Philipp Müller <t.i.m@zen.co.uk>
* gst/playback/gstplaybasebin.c:
Fix for #162924 - free caps after use, not before
Original commit message from CVS:
* configure.ac:
Fix indentation, fix v4l2 plugin detection.
* ext/Makefile.am:
Fix libmms location (Maciej, use diff -u!).
* ext/alsa/gstalsa.c: (gst_alsa_init):
Initialize caps cache to NULL.
* gst/playback/gstplaybin.c: (gst_play_bin_change_state):
Only change state on audiosink if it exists.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_add_element):
Revert patch 1.38 as clock distribution over schedulers does
not work correcly in the core yet.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (compare_ranks):
make sure the facotries are ordered the same every time even if they
have the same rank by using the name
* gst/playback/gstdecodebin.c: (find_compatibles):
make sure we don't add factories to the list twice
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoia.c: (cdparanoia_class_init),
(cdparanoia_set_property), (cdparanoia_get_property):
* ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_class_init),
(dvdnavsrc_set_property), (dvdnavsrc_get_property):
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
(dvdreadsrc_init), (dvdreadsrc_set_property),
(dvdreadsrc_get_property):
* sys/vcd/vcdsrc.c: (gst_vcdsrc_class_init),
(gst_vcdsrc_set_property), (gst_vcdsrc_get_property):
Synchronize property names where not yet the case. Devices are
now device=X, other versions are deprecated (but still exist).
Also use g_free() unconditionally.
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(setup_source), (gst_play_base_bin_get_property):
Expose source.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_change_state):
Don't crash on EMPTY caps (e.g. when the demuxer didn't recognize
the contained stream).
Original commit message from CVS:
* gst/playback/gstplaybin.c: (remove_sinks), (setup_sinks):
Unlink manually since sometimes bin disposal (and therefore
pad unlinking) is delayed, which will cause a new media file
to not be able to start playing instantly.
Original commit message from CVS:
* gst/playback/gststreaminfo.c: (stream_info_mute_pad):
On mute of an unlinked stream, check for pad availability so
we don't crash on unlinked pad.
Original commit message from CVS:
2004-11-27 Christophe Fergeau <teuf@gnome.org>
* gst/playback/gstplaybasebin.c: (setup_source): fixed a caps leak
(gst_play_base_bin_change_state): nullify source and decoder when
going from READY to NULL so that we don't try to do weird stuff with
them when going from NULL to READY
* gst/playback/gstplaybin.c: (gst_play_bin_init): use gst_object_unref
instead of g_object_unref
(gen_video_element), (gen_audio_element): more refcounting fixes, now
it should be correct
(gst_play_bin_change_state): don't call remove_sinks if we are
currently disposing the object
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/playback/gstplaybin.c: (gst_play_bin_dispose),
(gst_play_bin_set_property), (gen_video_element),
(gen_audio_element):
Refcounting fixes for provided audio-/videosinks.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element), (setup_sinks), (gst_play_bin_change_state):
Don't reference all sinks, but only the video- and audiosinks.
The vis. element should be disposed when we're done with it.
We don't have any reason to keep it around. This fixes warnings
when reusing playbin for playing multiple audio files with
vis. enabled. Also release audio device on pause - idea stolen
from Rhythmbox.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter):
We sometimes need parsers for playback, so add those too.
Original commit message from CVS:
patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst/playback/gstplaybasebin.c:
Fix unplayable files error handling. Fixes#158365
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_add_element):
Re-add clock distribution hack (until new core is released).
Fixes#158125.
Original commit message from CVS:
* gst/playback/README:
* gst/playback/gstplaybasebin.c: (group_destroy), (group_is_muted),
(add_stream), (unknown_type), (add_element_stream), (no_more_pads),
(probe_triggered), (preroll_unlinked), (new_decoded_pad),
(gst_play_base_bin_change_state), (gst_play_base_bin_found_tag):
* gst/playback/gstplaybin.c: (gen_vis_element), (remove_sinks),
(setup_sinks):
* gst/playback/gststreaminfo.c: (gst_stream_info_set_mute),
(gst_stream_info_is_mute), (gst_stream_info_set_property):
* gst/playback/gststreaminfo.h:
Updated README.
Only switch groups if all streams have muted (EOSed).
Send Tags in sync with the stream playback instead of in
the playback/preroll phase.
Some cleanups, free the fakesrc elements.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
(try_to_link_1), (get_our_ghost_pad), (remove_element_chain),
(unlinked), (no_more_pads), (close_link):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
(unknown_type), (add_element_stream), (new_decoded_pad),
(removed_decoded_pad), (setup_source):
* gst/playback/gststreaminfo.c: (gst_stream_info_get_type),
(gst_stream_info_class_init), (gst_stream_info_init),
(gst_stream_info_new), (gst_stream_info_dispose),
(stream_info_mute_pad), (gst_stream_info_set_property),
(gst_stream_info_get_property):
* gst/playback/gststreaminfo.h:
Fix playback of multiple files.
a slightly different approach to handling dynamic pad removals.
This one only looks at pads that we have linked.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
(get_unconnected_element), (remove_starting_from), (pad_removed),
(close_link):
Implement support for dynamic pad changing. We listen to "live"
pad removals (i.e. while playing) and re-setup autoplugging
after that. Playbasebin/playbin need some more work for this
to finally work, but decodebin supports (and replugs) chained
ogg now.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_vis_element), (remove_sinks), (setup_sinks):
Added vis plugin support, need to configure the vis
element to activate it.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source):
Cleanup the previous pipeline a little earlier for the
case that a source element provides raw data.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source),
(gst_play_base_bin_change_state):
Actually clean up streaminfo if output fails. This would trigger
if, for example, there was no CD in the drive. No preroll, so
a streaminfo structure is created, but the subsequent state change
of the thread fails.
* gst/playback/gstplaybin.c: (gst_play_bin_change_state):
Don't change state if parent failed.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_init), (gst_play_bin_get_property), (handoff),
(gen_video_element), (remove_sinks):
Add small bits of code for screenshot handling.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_set_property),
(gen_video_element), (gen_audio_element), (setup_sinks):
Don't assume the user provided sinks are named "sink"...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gen_preroll_element),
(unknown_type), (setup_source), (gst_play_base_bin_remove_element),
(gst_play_base_bin_link_stream):
Do not try to autoplug sources that generate raw streams like
cdparanoia.
disconnect the preroll overrun signal when we don't need it anymore.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (play_base_bin_mute_pad),
(gst_play_base_bin_mute_stream), (gst_play_base_bin_link_stream):
* gst/playback/gstplaybin.c: (setup_sinks):
Implement muting/unmuting of streams, mute streams that are not
used.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(find_compatibles), (close_pad_link), (try_to_link_1), (new_pad),
(no_more_pads), (close_link), (type_found):
* gst/playback/gstplaybasebin.c: (new_decoded_pad):
* gst/playback/gstplaybin.c: (gen_video_element):
Do not signal the no_more_pads after the first pad when
we are plugging a non dynamic element with multiple
output pads (like swfdec, dvdec, ...).
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(find_compatibles), (close_pad_link), (try_to_link_1),
(no_more_pads), (close_link), (type_found):
* gst/playback/gstplaybasebin.c: (new_decoded_pad):
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element):
Set state on newly added element to READY so that negotiation
can happen ASAP.
Addes some more debug info.
Do not try to plug pads with multiple caps structures or ANY
because it is too dangerous since we do not do dynamic
replugging.
Original commit message from CVS:
* gst/playback/README:
* gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
(gst_decode_bin_init), (find_compatibles), (close_pad_link),
(try_to_link_1), (no_more_pads), (close_link), (type_found):
Add some debug info to decodebin, update README
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flacdec_src_query):
Only return true if we actually filled something in. Prevents
player applications from showing a random length for flac files.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
(gst_riff_read_use_event), (gst_riff_read_handle_event),
(gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh),
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs):
OK, ok, so I implemented event handling. Apparently it's normal
that we receive random events at random points without asking
for it.
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index), (gst_avi_demux_stream_header),
(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Implement non-lineair chunk handling and subchunk processing.
The first solves playback of AVI files where the audio and video
data of individual buffers that we read are not synchronized.
This should not happen according to the wonderful AVI specs, but
of course it does happen in reality. It is also a prerequisite for
the second. Subchunk processing allows us to cut chunks in small
pieces and process each of these pieces separately. This is
required because I've seen several AVI files with incredibly large
audio chunks, even some files with only one audio chunk for the
whole file. This allows for proper playback including seeking.
This patch is supposed to fix all AVI A/V sync issues.
* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
(flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop):
Work.
* gst/modplug/gstmodplug.cc:
Proper return value setting for the query() function.
* gst/playback/gstplaybasebin.c: (setup_source):
Being in non-playing state (after, e.g., EOS) is not necessarily
a bad thing. Allow for that. This fixes playback of short files.
They don't actually playback fully now, because the clock already
runs. This means that small files (<500kB) with a small length
(<2sec) will still not or barely play. Other files, such as mod
or flx, will work correctly, however.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (new_decoded_pad):
Only signal the no_more_pads signal when we have
added the stream to our list.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (remove_prerolls),
(new_decoded_pad):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (setup_sinks):
Don't try to preroll or decode more than one audio/video
track.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_change_state):
Throw error if we failed to find a suitable output. This should
throw an error if we successfully set up a pipeline (e.g. because
we recognized a media file) but found no decodable streams in it
(e.g. because it contains only media stream types for which we
have no decoders, or because it's not a media type).
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (_read_var_length), (_read_guid),
(gst_asf_demux_process_segment), (gst_asf_demux_handle_data),
(gst_asf_demux_process_chunk), (gst_asf_demux_handle_sink_event):
Prevent infinite loops. More correct error reporting.
* gst/auparse/gstauparse.c: (gst_auparse_chain):
Error out if negotiation fails.
* gst/playback/gstplaybasebin.c: (setup_source),
(gst_play_base_bin_change_state), (gst_play_base_bin_error),
(gst_play_base_bin_found_tag):
Error/tag forwarding. Pre-roll fixes for source errors on state
changes (e.g. "file does not exist") to prevent hangs.
Original commit message from CVS:
* ext/mad/gstmad.c: (gst_mad_check_caps_reset),
(gst_mad_change_state):
Allow for mp3 rate/channels changes. However, only very
conservatively. Reason that we *have* to enable this is smiply
because the mad find_sync() function is not good enough, it will
regularly sync on random data as valid frames and therefore make
us provide random caps as *final* caps of the stream. The best fix
I could think of is to simply require several of the same stream
changes in a row before we change caps.
The actual testcase that works now is #
* ext/ogg/Makefile.am:
* ext/ogg/gstogg.c: (plugin_init):
* ext/ogg/gstogmparse.c:
OGM support (video only for now; I need an audio sample file).
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
(gst_asf_demux_process_stream), (gst_asf_demux_video_caps),
(gst_asf_demux_add_video_stream):
WMV extradata.
* gst/playback/gstplaybasebin.c: (unknown_type):
Don't error out on single unknown-types after all. It's wrong.
If we found type of video and audio but not of a subtitle stream,
it will still error out (which is unwanted). Will find a better fix
later on.
* gst/typefind/gsttypefindfunctions.c: (ogmvideo_type_find),
(ogmaudio_type_find), (plugin_init):
OGM support.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (state_change), (setup_source),
(gst_play_base_bin_change_state):
Handle the case where we failed to setup a clear pipeline. This
will throw an error (or EOS, another nice case) and if you don't
catch that, the app will wait for the signal forever (and thus
hang).
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssink.c:
(gst_gnomevfssink_uri_get_protocols):
* ext/gnomevfs/gstgnomevfssrc.c:
(gst_gnomevfssrc_uri_get_protocols):
* ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
* ext/gnomevfs/gstgnomevfsuri.h:
Use _uri_new() instead of _open(), so it doesn't take as long and
Christophe's computer won't hang.
* gst/playback/gstplaybasebin.c: (unknown_type):
Throw error on unknown media type, so apps actually display it.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_overrun), (no_more_pads),
(setup_source), (gst_play_base_bin_set_property),
(gst_play_base_bin_add_element):
* gst/playback/gstplaybin.c: (gst_play_bin_send_event):
Some more work on making sure seeking pauses the pipeline and
that changing the uri actually does something.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
(gst_play_base_bin_dispose), (gst_play_base_bin_set_property):
Handle double disposals, and proper change of URIs.
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_update),
(gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
(gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
(gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
Update mixer (to sync with other sessions) if we try to obtain
a new value. This makes alsamixer work accross applications.
* ext/alsa/gstalsasink.c: (gst_alsa_sink_get_time):
Only call sync functions if we're running, else alsalib asserts.
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_query):
Sometimes fails to compile. Possibly a gcc bug.
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element):
Add a reference to an application-provided object, because we lose
this same reference if we add it to the bin. If we don't do this,
we can only use this object once and thus crash if we go from
ready to playing, back to ready and back to playing again.
Also add an audioscale element because several cheap soundcards -
like mine - don't support all samplerates.
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
(gst_ximagesink_xcontext_clear), (gst_ximagesink_change_state):
Fix wrong order or PAR calls. Makes automatically obtained PAR
from the X server atually being used.
