Original commit message from CVS:
2006-01-31 Andy Wingo <wingo@pobox.com>
* sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the
caps being set doesn't have a framerate value. Basically a stopgap
measure.
* ext/ogg/gstoggmux.c (GST_BUFFER_END_TIME): New macro. Not
technically correct enough to put into core though.
(gst_ogg_mux_dequeue_page): Use END_TIME instead of TIMESTAMP +
DURATION. Fixes theoraenc ! oggmux.
* sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Fixate to the nearest
fraction, not double.
Original commit message from CVS:
2006-01-30 Andy Wingo <wingo@pobox.com>
* ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare
timestamp + duration, not just timestamp -- ogg pages should be
ordered by stop time. Necessary fix given the change in vorbis
timestamps.
Original commit message from CVS:
2006-01-30 Andy Wingo <wingo@pobox.com>
* ext/theora/theoraenc.c (theora_enc_sink_setcaps)
(gst_theora_enc_init): Pull the granule shift out of the encoder.
(granulepos_add): New function, handles the messiness of adjusting
granulepos values.
(theora_buffer_from_packet):
(theora_enc_chain):
(theora_enc_sink_event): Use granulepos_add, not +.
* tests/check/pipelines/theoraenc.c
(check_buffer_granulepos_from_starttime): Just check the frame
count, not the actual granulepos -- we can't dictate to the
encoder when it should be placing keyframes.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
SERVICE_NOT_AVAILABLE happens for example when you're trying to
play an http:// stream from a server that's not serving
Original commit message from CVS:
2006-01-30 Andy Wingo <wingo@pobox.com>
* tests/check/pipelines/vorbisenc.c (TIMESTAMP_OFFSET):
* tests/check/pipelines/theoraenc.c (TIMESTAMP_OFFSET): Totally
remove the UINT64_CONSTANT macro, doesn't appear to be needed or
available.
Original commit message from CVS:
2006-01-30 Andy Wingo <wingo@pobox.com>
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c: Same changes as were done to vorbisenc,
although theoraenc was timestamping correctly. Added handling of
streams that start with nonzero timestamps.
* tests/check/Makefile.am:
* tests/check/pipelines/theoraenc.c: New file, basically does same
tests as vorbisenc.
* tests/check/pipelines/vorbisenc.c: I claim these bugs.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
(gst_audioringbuffer_pause):
Implement pause that does not wait for completion.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Don't drop buffers when going to PAUSED but perform preroll on
remaining samples now that core base class supports this.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
(gst_ring_buffer_commit):
Pause should not signal waiters.
Implement return value of _commit correctly.
Original commit message from CVS:
2006-01-30 Andy Wingo <wingo@pobox.com>
* tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
* ext/vorbis/vorbisenc.c (gst_vorbisenc_buffer_from_packet): Logic
updated to timestamp from the first sample, not the last.
(gst_vorbisenc_buffer_from_header_packet): New function, takes
special care of granulepos and timestamp for header packets.
(gst_vorbisenc_chain): Reflow, fix some leaks, and handle the case
when the first buffer has a nonzero timestamp.
* ext/vorbis/vorbisenc.h (GstVorbisEnc.granulepos_offset)
(GstVorbisEnc.subgranule_offset): New members. Take care of the
case when the first audio buffer we get has a nonzero timestamp.
(GstVorbisEnc.next_ts): Renamed from prev_ts, because now we
properly timestamp vorbis buffers with the time of the first
sample, not the last.
* ext/vorbis/vorbisenc.c (granulepos_to_clocktime): Renamed from
vorbis_granule_time_copy -- now it takes the granule/subgranule
offset into account.
* tests/check/pipelines/vorbisenc.c: New test for correctness of
timestamps, durations, and granulepos on buffers produced by
vorbisenc.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
Patch from Eric Jonas to support conversions to/from UYVY
(Fixes: #324626)
Original commit message from CVS:
* gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
* gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
use of gst_guint64_to_gdouble to be compliant with vs6
* gst/playback/gstdecodebin.c: (try_to_link_1)
* gst/videorate/videorate.c: (gst_video_rate_blank_data)
use of G_GINT64_CONSTANT for int64 constants
* win32/common/libgstinterfaces.def:
export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
* win32/vs6:
update and add new project files
Original commit message from CVS:
* Makefile.am:
* win32/MANIFEST:
* win32/common/interfaces-enumtypes.c:
(gst_color_balance_type_get_type), (gst_mixer_type_get_type),
(gst_mixer_track_flags_get_type),
(gst_tuner_channel_flags_get_type):
* win32/common/interfaces-enumtypes.h:
* win32/common/multichannel-enumtypes.c:
(gst_audio_channel_position_get_type):
* win32/common/multichannel-enumtypes.h:
add a win32-update rule like in core, and copy over enumtypes files
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_init), (set_hwparams),
(set_swparams), (gst_alsasink_prepare), (gst_alsasink_unprepare),
(gst_alsasink_close), (gst_alsasink_write), (gst_alsasink_reset):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (set_hwparams),
(set_swparams), (gst_alsasrc_open), (gst_alsasrc_prepare),
(gst_alsasrc_unprepare), (gst_alsasrc_read):
Update all error messages. All of them should either use
the default translated message, or actually provide a
translatable string.
Make the string for channel count problems meaningful.
Original commit message from CVS:
* gst/videoscale/vs_scanline.c: Oops, *that's* why I never
checked in this change -- it requires liboil features not
in 0.3.6. Revert parts.
Original commit message from CVS:
* ext/libvisual/visual.c: (get_buffer):
When pad_alloc returns a GstFlowReturn other
than GST_FLOW_OK, make sure it is passed upstream.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
(gst_alsasink_class_init):
Free the device name string.
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
(gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad),
(gst_ogg_mux_handle_src_event), (gst_ogg_mux_clear_collectpads):
Don't remove a pad from the collectpads structure until it
is released - it's a request pad, and may receive data again
if the element gets moved back to PLAYING state.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
Ensure we turn on double buffering on the Xv port, and
set the colour key to something dark and mysterious that
isn't black.
Original commit message from CVS:
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_base_init), (plugin_init):
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
- a library should not call setlocale. see Libraries node in
gettext manual
- make sure all plugins that use translation do bindtextdomain
to point to the localedir
* gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
(setup_sinks), (plugin_init):
all this, and check for NULL when creating sinks
Original commit message from CVS:
2006-01-27 Julien MOUTTE <julien@moutte.net>
* gst/subparse/gstsubparse.c: (gst_subparse_type_find),
(plugin_init): Make typefinding of subtitles work again.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
(mp3_type_frame_length_from_header), (mp3_type_find),
(wavpack_type_find), (m4a_type_find), (ircam_type_find),
(plugin_init):
Backport a bunch of typefinding fixes from the 0.8 branch.
Also, improve wavpack typefinding: if we can't peek the
entire wavpack block, try to parse the bits we can get and
see if we find what we're looking for in those.
Original commit message from CVS:
2006-01-26 Julien MOUTTE <julien@moutte.net>
* sys/ximage/ximagesink.c:
(gst_ximagesink_calculate_pixel_aspect_ratio):
* sys/xvimage/xvimagesink.c:
(gst_xvimagesink_calculate_pixel_aspect_ratio): Handle some
more cases of pixel aspect ratio.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (pad_probe):
Also consider the flush-start and tag events as unblockers
for the pad probes.
Original commit message from CVS:
2006-01-26 Julien MOUTTE <julien@moutte.net>
* gst/playback/gstplaybin.c: (gst_play_bin_init),
(gst_play_bin_dispose), (gst_play_bin_vis_unblocked),
(gst_play_bin_vis_blocked), (gst_play_bin_set_property):
On the fly visualisation switch, works disabling, enabling as
well but it won't be able to enable vis in a playbin that was
created with no visualisation.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Undo previous commit, it breaks resume after pause.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
(gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
Improve debugging.
Post error when caps cannot be parsed.
Resync on discontinuity in the stream.
Clip samples to segment boundaries.
return WRONG_STATE sooner when we are flushing.
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_get_time), (gst_base_audio_src_create):
Make audiosrc operate in TIME.
Set TIMESTAMP and DURATION on buffers.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(free_pad_probes), (remove_fakesink), (pad_probe),
(close_pad_link), (gst_decode_bin_change_state):
Replace GstPadBlockCallback with pad probes that detect
first buffer AND eos before removing fakesink.
Fixes hang with demuxers doing EOS while pre-rolling.
Solves #328279
Original commit message from CVS:
2006-01-22 Julien MOUTTE <julien@moutte.net>
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimagesink_buffer_alloc): Playbin keeps some ref to some
frames. We might get a frame destroyed after changing state to
NULL, adding a safety check on xcontext.
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c:
Fix prepare-xwindow-id code example in the docs - we need to
ignore all messages that aren't element messages as well.
Original commit message from CVS:
2006-01-21 Julien MOUTTE <julien@moutte.net>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
I think one day i'll completely undestand how caps negotiation
is supposed to work. This refactoring handles buffer_alloc
called with caps we can't handle. We definitely don't want a
set_caps with those caps, so we define and allocate a buffer
we would like to receive.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Fix playback of non-synchronised streams by assuming a rate
of 1.0 instead of a random one.
Makes this work again:
gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
endianness=(int)4321, signed=(boolean)true, width=(int)16,
depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
audioresample ! alsasink
Original commit message from CVS:
* gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
* gst/playback/gststreamselector.c:
(gst_stream_selector_set_property):
Comment out broken code that connects to the state-changed signal.
At this point, changing current stream selection is broken, but
stuff like gst-launch playbin current-audio=1 works and filters
to the chosen stream.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_handle_comment_packet):
Post taglist actually on bus instead of just freeing it
(fixes#327114 and totem bug #327080).
* ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
Use gst_element_found_tags_for_pad(), so that the tags
are sent downstream as an event as well.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximagesink_ximage_new), (gst_ximagesink_ximage_put),
(gst_ximagesink_buffer_alloc):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
(gst_xvimagesink_xvimage_put), (gst_xvimagesink_show_frame),
(gst_xvimagesink_buffer_alloc):
move all regularly occurring messages to GST_LOG level
add some more object logs
Original commit message from CVS:
2006-01-14 Thomas Vander Stichele <thomas at apestaart dot org>
* ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
fix a silly segfault
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source):
Fix playback for sources that emit raw audio or
raw video streams (e.g.: cd audio sources) (#325984).
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_track_seek):
No need to post a tag message on the bus when seeking
within the same track, only post it when the current
track changes.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_destroy),
(probe_triggered), (new_decoded_pad), (mute_group_type),
(set_active_source):
* gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
* gst/playback/gststreamselector.c:
(gst_stream_selector_base_init),
(gst_stream_selector_set_property),
(gst_stream_selector_request_new_pad):
Reenable stream selection. These mechanisms need a complete overhaul
in the face of 0.8->0.10 changes though.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c:
Change the pad template to src_%d to match the pads that
are created from it. decodebin needs this information in order
to decide that oggdemux is capable of producing multiple pads
(and hence needs queues inserted).
* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
(gst_ogg_mux_collected):
Make debug output more useful by using GST_PTR_FORMAT.
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
Set depth and width for alaw/mulaw (fixes#326601).
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
(gst_cd_paranoia_paranoia_callback):
Don't try to free NULL pointers.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
(gst_audio_rate_change_state), (plugin_init):
Add debugging category.
Fix type issues.
Add case for incoming buffers without valid offset/offset_end.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_audio_element):
Don't leak an autoaudiosink/alsasink when we generate
a new audio element. (old code, I guess)
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
Support float audio in audiorate.
Use width rather than depth for selecting sample width.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.h:
Use GLib types here (that way we don't have to include the
generated _stdint.h header, which makes life easier for win32
folks that don't use autotools for the build) (#325990, patch
by: Sergey Scobich).
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (wait_segment):
* gst-libs/gst/audio/gstringbuffer.h:
Name (private) union, makes Forte compiler happy (this time
for real) (#324900).
Original commit message from CVS:
* gst/audioresample/resample.h:
Declare struct _ResampleState.buffer as unsigned char *, not void *,
since we do arithmetic on it.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (wait_segment):
* gst-libs/gst/audio/gstringbuffer.h:
Sun's Forte compiler doesn't seem to like anonymous structs,
so use same setup as in GstBaseSrc (fixes#324900).
Original commit message from CVS:
* configure.ac:
* gst/volume/Makefile.am:
* gst/volume/demo.c:
move old example to tests/examples/volume/volune.c
* tests/examples/Makefile.am:
* tests/examples/seek/seek.c: (main):
change window-close event from "delete-event" to "destroy"
* tests/examples/volume/Makefile.am:
* tests/examples/volume/volume.c: (value_changed_callback),
(setup_gui), (message_received), (eos_message_received), (main):
fix event handling and bus usage
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (new_pad):
Fix non-C89 variable declaration not at the start of a block. Should
help some compilers.
Original commit message from CVS:
* configure.ac:
* ext/cdparanoia/Makefile.am:
* ext/cdparanoia/gstcdparanoia.c:
* ext/cdparanoia/gstcdparanoia.h:
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_mode_get_type), (gst_cd_paranoia_src_base_init),
(gst_cd_paranoia_src_init), (gst_cd_paranoia_src_class_init),
(gst_cd_paranoia_src_open), (gst_cd_paranoia_src_close),
(gst_cd_paranoia_paranoia_callback),
(gst_cd_paranoia_src_read_sector), (gst_cd_paranoia_src_finalize),
(gst_cd_paranoia_src_set_property),
(gst_cd_paranoia_src_get_property), (plugin_init):
* ext/cdparanoia/gstcdparanoiasrc.h:
New cdparanoiasrc element based on cddabasesrc; enable cdparanoia
plugin again (there are still fixes required to playbin to make
cdda:// uris work there).
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_update_duration),
(gst_cdda_base_src_calculate_cddb_id):
An integer is not a string. Fix access to uninitialised variable.
* tests/check/Makefile.am:
Add cddabasesrc unit test; also actually enable the vorbis test.
* tests/check/generic/states.c:
Blacklist new cd audio elements as well.
* tests/check/libs/cddabasesrc.c:
Unit test for GstCddaBaseSrc (discid calculation mostly).
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
Add docs for libgstcdda/GstCddaBaseSrc.
* gst-libs/gst/interfaces/mixertrack.h:
Do one struct member per line with a semicolon at the end, that way
even gtk-doc might parse it without complaining.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init), (gst_video_test_src_start):
Add start method to reset running time and number of frames sent
when starting up (fixes#324696; patch by: Michal Benes).
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
update strings, values are in microseconds
change the default sink buffer time to something that is smaller
(to help software volume mixing have a slightly lower delay) but
still be acceptable on Wim's laptop
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):
Made a quack, forgot to add DUCK to the riff video template.
Original commit message from CVS:
* ext/ogg/gstogmparse.c: (gst_ogm_text_parse_base_init),
(gst_ogm_parse_init), (gst_ogm_audio_parse_init),
(gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
(gst_ogm_parse_chain):
Make sure pads are initialized correctly.
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add a whole bunch of FOURCC <=> MimeType.
Extend the riff video pad template to support the newly added fourcc.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
(gst_ogg_demux_activate_chain):
Extra debug output when activating/deactivating chains.
* gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
(is_demuxer_element), (try_to_link_1), (remove_element_chain),
(unlinked):
Remove a queue from our list when it becomes unlinked.
Don't add queues to elements in class 'Demux' if they
can only produce one pad
Original commit message from CVS:
2005-12-17 Philippe Khalaf <burger@speedy.org>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_handle_sink_event):
Handle downstream newsegment by sending our own newsegment before the
next buffer to be released. (#323900)
Original commit message from CVS:
2005-12-17 Philippe Khalaf <burger@speedy.org>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
add queue delay to new segment as well (as opposed to just the first
buffer). (bug #322347)
Original commit message from CVS:
* ext/libvisual/visual.c: (make_valid_name):
change some char* into char[]
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_do_seek),
(gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
prepare to handle EOS and SEGMENT_DONE
Original commit message from CVS:
* gst/tcp/gsttcp.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
Add <string.h> includes for memset and FD_ZERO (fixes#323878;
patch by: Benjamin Pineau).
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_blank_data),
(gst_video_rate_chain):
Fix timestamping for videorate when the first buffer it sees has a
non-zero timestamp. Fix some misleading debug output.
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_collected):
Don't operate on empty text buffers. Strip newlines and
tabs only from the end of the text, but leave them intact
in the middle. Fix typo in gtk-doc description.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybin.c: (handoff):
Make sure the video frame buffer we return to apps via the
"frame" property always has caps set on it. Modify
_gst_gvalue_set_object() macro to handle NULL objects
gracefully too.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
(gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
(gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Adjust to some recent api changes and add wtays new cool seeking
capabillities
Original commit message from CVS:
* ext/alsa/Makefile.am:
* ext/alsa/gstalsadeviceprobe.c:
* ext/alsa/gstalsadeviceprobe.h:
Helper functions to add device probing via the GstPropertyProbe
interface to a class.
* ext/alsa/gstalsamixer.h:
Comment out GST_ALSA_MIXER, it returns a struct that's not
used.
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
Add some debug info.
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_interface_supported),
(gst_implements_interface_init),
(gst_alsa_mixer_element_init_interfaces),
(gst_alsa_mixer_element_class_init),
(gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
(gst_alsa_mixer_element_set_property),
(gst_alsa_mixer_element_get_property),
(gst_alsa_mixer_element_change_state):
* ext/alsa/gstalsamixerelement.h:
Add 'device' and 'device-name' properties. Add GstPropertyProbe
for device handling (gnome-volume-control will need that).
Original commit message from CVS:
* ext/cdparanoia/Makefile.am:
* ext/cdparanoia/gstcdparanoia.c: (gst_paranoia_mode_get_type),
(gst_paranoia_endian_get_type), (_do_init),
(cdparanoia_class_init), (cdparanoia_init),
(cdparanoia_set_property), (cdparanoia_get_property),
(cdparanoia_do_seek), (cdparanoia_is_seekable),
(cdparanoia_create), (cdparanoia_start), (cdparanoia_stop),
(cdparanoia_convert), (cdparanoia_get_query_types),
(cdparanoia_query), (cdparanoia_set_index),
(cdparanoia_uri_set_uri):
* ext/cdparanoia/gstcdparanoia.h:
Partially ported cdparanoia now that basesrc can support a
plugin like this..
Original commit message from CVS:
* tests/examples/seek/scrubby.c: (main):
Set higher priority for bus events so they don't get reordered with
gtk gui events.
* tests/examples/seek/seek.c: (do_seek), (start_seek), (stop_seek),
(flush_toggle_cb), (main):
Added checkbox do disable flushing seeks.
Disable scrubbing when doing non flushing seeks.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (gst_sub_parse_init),
(gst_sub_parse_do_seek), (gst_sub_parse_src_event), (parse_subrip),
(parser_state_init), (handle_buffer), (gst_sub_parse_chain),
(gst_sub_parse_sink_event), (gst_sub_parse_change_state):
Implement some sort of event handling that doesn't rely on
g_return_if_fail; make sure we always push the last chunk of an
.srt out when we receive an EOS; use gst_pad_alloc_buffer; fix
state change function; remove some old cruft. Seeking is still
rather unlikely to work though.
* tools/.cvsignore:
Ignore more.
Original commit message from CVS:
2005-12-11 Julien MOUTTE <julien@moutte.net>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
Fixed a leak of the current image reference when cleaning up.
Thanks to Arwed von Merkatz (alley_cat) for pointing it out.
Original commit message from CVS:
* tools/Makefile.am:
* tools/gst-launch-ext-m.m:
Remove gst-launch-ext. It doesn't work, and is no longer
particularly useful.
Original commit message from CVS:
* ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
don't pass random values to ogmparse convert function.
Make seeking possible in the exile1.ogm file.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
* gst/playback/gstplaybin.c: (gst_play_bin_get_property):
Work around refcount problem with g_value_set_object() that occur
if the core has been compiled against GLib-2.6 (g_value_set_object()
will only g_object_ref() the element, but the caller will
gst_object_unref() it and bad things will happen due to the way
GstObjects are refcounted in the GLib-2.6 case). Fixes problems with
totem for people on FC4 using Thomas's 0.10 RPMs.
Original commit message from CVS:
Time to welcome ogm to 0.10 :)
* ext/ogg/gstoggdemux.c: (internal_element_pad_added_cb),
(gst_ogg_pad_typefind):
Oggdemux can now properly typefind elements with dynamic pads.
* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
Properly set caps on src pad, and set caps on outgoing buffers.
Original commit message from CVS:
* ext/Makefile.am:
Add $(PANGO_DIR) to SUBDIRS
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttimeoverlay.c:
Fix and improve element descriptions.
Original commit message from CVS:
* docs/libs/tmpl/gstcolorbalance.sgml:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtp/gstrtpbuffer.h:
Do burger's rename for rtp payloaders and depayloaders
Original commit message from CVS:
* gst-libs/gst/net/Makefile.am:
* gst-libs/gst/net/README:
* gst-libs/gst/net/gstnetbuffer.c:
* gst-libs/gst/net/gstnetbuffer.h:
this was moved to netbuffer
Original commit message from CVS:
* gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_get_type),
(gst_video_filter_class_init), (gst_video_filter_init):
* gst-libs/gst/video/gstvideofilter.h:
borgify name to bring in line with other classes
Original commit message from CVS:
* gst-libs/gst/netbuffer/Makefile.am: (libgstnetbufferincludedir):
Let's not override libgstnet from core for no reason...
(libgstnetbuffer_@GST_MAJORMINOR@_la_SOURCES):
Ok, maybe not so quick next time.
Original commit message from CVS:
* gst-libs/gst/netbuffer/Makefile.am: (libgstnetbufferincludedir):
Let's not override libgstnet from core for no reason...
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
moved gst-libs/gst/net to netbuffer through CVS surgery
remove old directory
updating build to accomodate
(#322257)
Original commit message from CVS:
2005-11-29 Andy Wingo <wingo@pobox.com>
* pkgconfig/gstreamer-plugins-base.pc.in:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* gst-libs/gst/net/Makefile.am: Rename gstnet to gstnetbuffer
(#322257).
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_stream_selector_chain):
3rd time's the charm. Correct ref-counting for discarded buffers.
Original commit message from CVS:
* gst/playback/gststreamselector.c:
(gst_stream_selector_class_init), (gst_stream_selector_init),
(gst_stream_selector_dispose), (gst_stream_selector_set_property),
(gst_stream_selector_get_property),
(gst_stream_selector_get_linked_pad),
(gst_stream_selector_request_new_pad), (gst_stream_selector_chain):
* gst/playback/gststreamselector.h:
Add the active-pad property for playbin to use shortly. Ignore buffers
from any other pad, returning GST_FLOW_NOT_LINKED
Original commit message from CVS:
2005-11-28 Julien MOUTTE <julien@moutte.net>
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
(gst_ximagesink_setcaps):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_setcaps): Fixed a tricky bug. When caps
renegotiation
happens (only visible on ximagesink but bug is in xv too)
set_caps was
destroying the internal x[v]image used to memcpy non locally
alloced
buffers so that it got renewed on next _chain. The issue is that
_expose will try to put that image as it reffed it in _put.
Using gst_buffer_unref instead of destroy fixes it !
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(try_to_link_1), (queue_filled_cb):
Better use of the queues. Start with a small size queue and only increase
the size of the queues when the other queues are empty.
Original commit message from CVS:
* gst-libs/gst/video/Makefile.am:
compile in copied-over videofilter into the video library
* gst-libs/gst/video/videosink.h:
rename the header to gstvideosink.h since it's a base GstObject class
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.h:
use the new header
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_commit), (probe_triggered):
* gst/playback/gstplaybasebin.h:
Prepare to handle errors betters.
* gst/playback/gstplaybin.c: (add_sink), (setup_sinks):
Set sinks to PAUSED first before adding and linking them so that
we don't interrupt dataflow.
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Updated TODO
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_open_device),
(gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_release):
Small cleanups.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Slave to the master clock when going to PLAYING and unslave when
going to PAUSED.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_acquire), (gst_ring_buffer_release),
(gst_ring_buffer_samples_done), (gst_ring_buffer_set_sample),
(gst_ring_buffer_clear_all), (wait_segment),
(gst_ring_buffer_commit), (gst_ring_buffer_read),
(gst_ring_buffer_advance):
* gst-libs/gst/audio/gstringbuffer.h:
Add some docs and cleanups.
Original commit message from CVS:
2005-11-28 Julien MOUTTE <julien@moutte.net>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
Use calculated video geometry from _setcaps instead of buffer
caps to respect pixel aspect ratio. (fixes#322388)
Original commit message from CVS:
2005-11-28 Julien MOUTTE <julien@moutte.net>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
Refuse to create an XvImage if we can't find the format.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_template_caps):
Add ATRAC3 to the list of riff-possible audio caps.
I know we still don't have a plugin for atrac3, but it's saner to output
that than a cryptic mimetype.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
(gst_decode_bin_init), (close_pad_link), (try_to_link_1):
Remove unused properties, and add queues between demuxers and decoders
so that a lot more files can preroll properly.
Original commit message from CVS:
* gst-libs/gst/net/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
remove silly include
* gst/tags/Makefile.am:
* gst/tags/gsttagediting.c:
* gst/tags/gsttageditingprivate.h:
* gst/tags/tagedit.vcproj:
remove directory, is as good as empty
Original commit message from CVS:
* configure.ac:
added GST_LIB_LDFLAGS and GST_ALL_LDFLAGS
* gst-libs/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/net/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
and use them
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audiorate_chain):
Properly return GstFlowReturn from gst_pad_push in chain functions.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_convert),
(gst_ogg_demux_chain_peer), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
(gst_ogg_demux_send_event), (gst_ogg_demux_loop):
Handle various conditions better when we don't understand a stream.
Removes a heap of CRITICALs on ogg streams containing unknown data.
Original commit message from CVS:
2005-11-24 Andy Wingo <wingo@pobox.com>
* gst/tcp/gstmultifdsink.c (gst_multifdsink_handle_client_write):
Be threadsafe.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_read):
If we are reading too slowly, jump forward in the ringbuffer
instead of blocking.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_set_clock),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Fix for calibration API change.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_get_channel_positions),
(gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(gst_audio_fixate_channel_positions):
Use gst_value_array_*() functions on value arrays, not
gst_value_list_*().
Original commit message from CVS:
* check/Makefile.am:
* check/elements/audiotestsrc.c: (setup_audiotestsrc),
(cleanup_audiotestsrc), (GST_START_TEST), (audiotestsrc_suite),
(main):
add a test for audiotestsrc, testing all waves. Even seems
leak-free at first glance, nice job Stefan
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_set_clock),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
And we provide a clock by default, of course...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init):
This clock can be slaved to a master clock now.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_dispose), (gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_set_clock),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_preroll),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Handle slaving the internal clock to the clock selected in the
pipeline.
Add property to make the basesink not provide a clock.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
(gst_base_rtp_depayload_wait):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
We can use the clock in GstElement, no need to store it ourselves.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_videorate_chain):
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_create):
Use utility method for scaling clocktime for fractional framerates.
Original commit message from CVS:
* examples/seeking/seek.c: (main):
Give higher priority to bus signals than the gtk events
to fix a race condition in the segment looping.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
Call gst_x_overlay_prepare_xwindow_id() to give applications
a final chance to set their own xwindow id before the video
sink creates its own window.
Original commit message from CVS:
2005-11-22 Julien MOUTTE <julien@moutte.net>
* sys/xvimage/xvimagesink.c:
(gst_xvimagesink_navigation_send_event): Handle navigation
events correcly with borders if applicable.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_caps_remove_format_info):
* gst/ffmpegcolorspace/imgconvert.c:
* gst/ffmpegcolorspace/imgconvert_template.h:
Forward-port fixes from the 0.8 branch (patch by Luca Ognibene,
#318353); use gst_structure_has_name().
Original commit message from CVS:
* gst-libs/gst/audio/audio.c: (gst_audio_duration_from_pad_buffer):
* gst-libs/gst/audio/audio.h:
fix prototype - wondering why the test worked regardless
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_base_init),
(gst_x_overlay_got_xwindow_id), (gst_x_overlay_prepare_xwindow_id):
* gst-libs/gst/interfaces/xoverlay.h:
Remove everything having to do with the desired size; add
gst_x_overlay_prepare_xwindow_id() function; remove the
'have-xwindow-id' signal and make gst_x_overlay_got_xwindow_id()
post a message on the bus instead (#321816).
* sys/ximage/ximagesink.c: (gst_ximagesink_xoverlay_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps),
(gst_xvimagesink_xoverlay_init):
Remove desired size stuff (#321816).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find),
(mpeg2_sys_type_find), (mpeg1_sys_type_find),
(mpeg_video_type_find), (mpeg_video_stream_type_find):
Terminate vararg functions with NULL instead of 0 to
make gcc4 happy.
Original commit message from CVS:
2005-11-21 Andy Wingo <wingo@pobox.com>
* gst/playback/gstplaybin.c (gen_audio_element)
(gen_video_element): Use the new MISSING_PLUGIN core error
category.
Original commit message from CVS:
* sys/ximage/Makefile.am:
* sys/ximage/ximage.c:
Rename ximage plugin to ximagesink (#321426) (Don't forget to
remove your old libgstximage.* manually if necessary).
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_setup):
Minimal fix for bug #320200: set the min/max bitrate in the correct
units. A better fix would be to upgrade to the RATEMANAGE2
interface, rather than using the deprecated interface used here, but
that would require an update in our libvorbis dependency (to 1.1),
which is probably undesirable.
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_init),
(gst_ogg_mux_request_new_pad), (gst_ogg_mux_queue_pads),
(gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
* gst/adder/gstadder.c: (gst_adder_init),
(gst_adder_request_new_pad), (gst_adder_collected),
(gst_adder_change_state):
Update for gst_collectpads_foo() to gst_collect_pads_foo()
API change.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_push_buffer),
(gst_vorbisenc_push_packet):
GST_PAD_IS_USABLE is gone, use the return value of
the push or pad_alloc_buffer instead.
Original commit message from CVS:
2005-11-18 Julien MOUTTE <julien@moutte.net>
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximagesink_check_xshm_calls), (gst_ximagesink_ximage_new),
(gst_ximagesink_ximage_destroy),
(gst_ximagesink_xwindow_draw_borders),
(gst_ximagesink_ximage_put),
(gst_ximagesink_xwindow_new),
(gst_ximagesink_xwindow_update_geometry),
(gst_ximagesink_handle_xevents), (gst_ximagesink_event_thread),
(gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
(gst_ximagesink_bufferpool_clear), (gst_ximagesink_getcaps),
(gst_ximagesink_setcaps), (gst_ximagesink_change_state),
(gst_ximagesink_show_frame), (gst_ximagesink_buffer_alloc),
(gst_ximagesink_navigation_send_event),
(gst_ximagesink_set_xwindow_id), (gst_ximagesink_expose),
(gst_ximagesink_set_property), (gst_ximagesink_get_property),
(gst_ximagesink_finalize), (gst_ximagesink_init),
(gst_ximagesink_class_init):
* sys/ximage/ximagesink.h: Refactoring of ximagesink.
This new version brings correct software scaling, non flickering
window while resizing, pixel aspect ratio handling, usage of
hardware buffer pools, out of data flow event thread for
navigation and handling of expose events even when being PAUSED,
a new property to keep video aspect ratio when resizing, etc...
Original commit message from CVS:
2005-11-18 Julien MOUTTE <julien@moutte.net>
* gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
(gst_videoscale_fixate_caps): Introduce back caps fixate with
handling of PAR.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init):
Unsetting IS_SINK flag from the fakesink, so decodebin
never behaves as a sink.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_change_state):
Fix the audiosrc base class again, we did not unflush.
Original commit message from CVS:
2005-11-17 Julien MOUTTE <julien@moutte.net>
* examples/seeking/seek.c: (make_dv_pipeline),
(make_vorbis_theora_pipeline), (make_avi_pipeline),
(make_mpegnt_pipeline): Use VSINK everywhere, add a scaler
to ogg/vorbis/theora pipeline.
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
(gst_ogg_mux_get_headers), (gst_ogg_mux_send_headers),
(gst_ogg_mux_collected), (gst_ogg_mux_change_state):
Fix EOS on multiple streams.
More debugging.
Original commit message from CVS:
* check/generic/states.c: (GST_START_TEST):
fix the test so that it only checks for elements that are part of
this source module
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_change_state):
Set ringbuffer to non-flushing when going to PAUSED, set to
flushing again when going to READY.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_stop):
Start in flushing mode by default.
Don't set flushing in the _stop method, let the app call
this explicitly.
Original commit message from CVS:
2005-11-16 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/video/gstvideosink.c:
(gst_video_sink_center_rect):
* gst-libs/gst/video/videosink.h: Add helper function needed
for video sinks.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_stop):
Set ringbuffer to flushing when stopping so that we don't
block on wait_segment anymore and livelock.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_receive_event),
(gst_ogg_demux_class_init):
Implement GstElement::send_event, so we can send seek events
in GST_STATE_READY
Original commit message from CVS:
2005-11-14 Julien MOUTTE <julien@moutte.net>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new):
Discovered how to take away flickering while resizing the
window. Please don't put that in ximagesink, refactoring in
progress.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_client_queue_data),
(gst_multifdsink_render):
Don't leak GDP headers when using GDP mode (i.e. tcpserversink).
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_audio_element):
Use autoaudiosink, it tends to be more widely available than
autoaudiiosink.
Original commit message from CVS:
2005-11-14 Andy Wingo <wingo@pobox.com>
* gst/playback/gstplaybin.c (gen_audio_element): Use autoaudiosink
as well if it is available. Fixes#316442.
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_clear), (gst_ogg_mux_init),
(gst_ogg_mux_sinkconnect), (gst_ogg_mux_request_new_pad),
(gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
(gst_ogg_mux_pad_queue_page), (gst_ogg_mux_queue_pads),
(gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_collected),
(gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_buffer_from_packet),
(gst_vorbisenc_change_state):
Fix a small memory leak in vorbisenc.
Fix large memory leaks in oggmux, also fix lots of state change
bugs in oggmux.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_class_init), (gst_videotestsrc_init),
(gst_videotestsrc_src_fixate):
move fixation to a fixate function
remove negotiate function, basesrc's is good enough
fixes a bug for check when using the element alone
Original commit message from CVS:
* examples/seeking/seek.c: (do_seek), (accurate_toggle_cb),
(key_toggle_cb), (main):
Added checkboxes for adding/removing the accurate and key_unit seek
flags.
Original commit message from CVS:
* examples/seeking/seek.c: (make_parselaunch_pipeline):
Added parse-launch syntax seeking mode for the seeking example.
This should help stress-test even more cases.
Ex usage : ./seek 15 "filesrc location=uranus.avi ! decodebin ! xvimagesink"
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
(gst_xvimagesink_navigation_send_event):
Check whether peer pad exists before sending navigation events
to it.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_setup),
(gst_vorbisenc_buffer_from_packet):
* ext/vorbis/vorbisenc.h:
Set duration on encoded buffers. This allows oggmux's
max_page_delay parameter to actually work.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_get_palette), (gst_ffmpeg_set_palette),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size), (gst_ffmpegcsp_transform):
Make palettes work again (see #132341). Use our own macros
for rounding up.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_open):
check for ALSA errors properly, instead of relying on ALSA's
error strings to serve to the user.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_commit), (new_decoded_pad),
(setup_substreams), (set_active_source):
Unlock GROUP_LOCK in failure cases, so that we don't deadlock when
trying to go to NULL if we failed to read a file.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiotestsrc_class_init), (gst_audiotestsrc_get_times),
(gst_audiotestsrc_create):
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_get_times), (gst_sinesrc_create):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_class_init), (gst_videotestsrc_get_times),
(gst_videotestsrc_create):
The base class can now sync for us.
Original commit message from CVS:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
Check if the caps have a fourcc field. Fixes crash for
gst-launch-0.9 v4lsrc name=source autoprobe=false autoprobe-fps=false copy-mode=1 device=/dev/video0 ! ffmpegcolorspace ! "video/x-raw-yuv, format=(fourcc)I420" ! xvimagesink
Original commit message from CVS:
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_get_query_types), (gst_sinesrc_src_query),
(gst_sinesrc_newsegment):
Send newsegment event in TIME format, set duration if
num-buffers is set, fix duration querying.
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
(gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
(gst_ogg_mux_pad_queue_page), (gst_ogg_mux_queue_pads),
(gst_ogg_mux_collected):
Fix EOS handling, partially. Now forwarding an EOS event once we have
EOS on all pads works correctly. However, we still don't properly set
EOS on the actual ogg stream pages.
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_init),
(gst_base_rtp_depayload_set_gst_timestamp):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
We need to send a newsegment event for each instance, not
just for the first instance of this class (get rid of
static variable in function). (#321011).
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
(gst_ogg_mux_buffer_from_page), (gst_ogg_mux_push_buffer),
(gst_ogg_mux_dequeue_page), (gst_ogg_mux_pad_queue_page),
(gst_ogg_mux_send_headers), (gst_ogg_mux_collected):
Forward port rewrite of muxing strategy to 0.9 version of oggmux.
This makes us mux things correctly according to the ogg muxing
rules. Still not handling EOS correctly right now, though.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
(gst_ogg_pad_submit_packet), (gst_ogg_chain_new):
Initialise segment_stop to GST_CLOCK_TIME_NONE when
creating a new chain; should fix live streaming. Also
add more debug output and fix a typo.
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst/volume/gstvolume.c: (volume_set_caps):
Fix compilation on Solaris with Forte. (#320923)
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(gst_decode_bin_dispose), (free_dynamics), (remove_fakesink),
(pad_blocked), (close_pad_link), (new_pad), (no_more_pads):
Handle the case where a pad_block failed.
Original commit message from CVS:
2005-10-31 Michael Smith <msmith@fluendo.com>
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_init),
(gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
(gst_ogg_demux_collect_chain_info), (gst_ogg_print):
Patch from Alessandro Decina <alessandro@nnva.org>.
Make oggdemux only find the final time in a chain, not per-pad,
since the per-pad information can be very expensive to locate, and
it isn't used anywhere. This makes reading a file containing
OggSkeleton reasonably fast.
Also, make chain finding work when there are logical bitstreams that
can't be decoded. Fixes#319110.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.h:
Don't break ABI.
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_caps_to_pixfmt):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_set_caps):
Some more comments.
Handle missing required caps fields better.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_get_offset),
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_pause),
(gst_ring_buffer_stop), (wait_segment), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
Add flushing mode to the ringbuffer so that it in all cases does
not try to handle more audio. This makes sure it does not try to
block anymore when flushing and fixes a livelock.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_convert),
(gst_ogg_demux_chain_peer), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain):
Explicitly check for -1 values before doing a conversion
and always map them to -1. (#315545)
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query), (gst_adder_collected),
(gst_adder_change_state):
Fix timestamps and fix deadlock when stopping the collectpads.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (audio_convert_clean_context):
When clearing an audioconvert context, set tmpbufsize to zero, so
we'll allocate it again later if required.
This fixes audioconvert re-negotiating formats, which previously
segfaulted with a NULL destination buffer.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
Correctly flush decoder samples even if we could not
copy them to an output buffer. Fixes#319618.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_offset), (gst_base_audio_sink_render):
Remove g_print
Use sync property from baseclass to disable sync.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_offset), (gst_base_audio_sink_render):
Buffers with no timestamps get aligned with previous buffers or
on underrun, played ASAP.
Original commit message from CVS:
2005-10-24 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/video/video.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
And
here comes my change on caps for framerate and geometry range.
We are now accepting 1 to MAXINT for width and height, and from
0.0 to MAXDOUBLE for framerate. That allows duration less png
frames
to be blended correctly in videomixer.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(gst_decode_bin_dispose), (free_dynamics), (pad_unblocked),
(pad_blocked), (close_pad_link), (new_pad):
Don't try to remove elements twice.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_src_query),
(theora_dec_sink_event):
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_sink_event), (vorbis_handle_identification_packet),
(vorbis_handle_data_packet):
* ext/vorbis/vorbisdec.h:
Fix old naming.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Don't try to sync on buffers without a timestamp.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_query_types),
(gst_vorbisenc_src_query):
Implement position and duration queries.
* gst/playback/test3.c: (update_scale), (main):
Fix for async state changes and print nicer output.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiotestsrc_src_query):
* gst/sine/gstsinesrc.c: (gst_sinesrc_src_query):
Don't use functions for position queries when handling
duration queries.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_handle_data_packet), (vorbis_dec_chain),
(vorbis_dec_change_state):
* ext/vorbis/vorbisdec.h:
Vorbis streams can be embedded in other container formats
than ogg, container formats where the demuxer might set
timestamps on encoded vorbis buffers instead of those silly
granulepos thingies. In short: make vorbisdec handle
timestamps on incoming buffers as well.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_destroy),
(gst_play_base_bin_change_state):
Fix leak.
Handle case where playbasebin is now ASYNC because
decodebin is.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
(xml_check_first_element), (xml_type_find), (smil_type_find),
(plugin_init):
Add typefinding for SMIL and for generic XML. Based on patch by
Akos Maroy (#308663).
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_loop):
Fix for segment-start/stop API change.
Original commit message from CVS:
* check/Makefile.am:
* check/clocks/selection.c: (GST_START_TEST), (volume_suite),
(main):
Add future test for clock selection.
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add support for Indeo-3 (IV32).
Original commit message from CVS:
2005-10-17 Andy Wingo <wingo@pobox.com>
* ext/ogg/gstoggmux.c (gst_ogg_mux_queue_pads): Fix bug introduced
with the collectpads change.
(gst_ogg_mux_send_headers): Elevate warning to a g_critical.
Original commit message from CVS:
2005-10-17 Andy Wingo <wingo@pobox.com>
* gst/tcp/gstmultifdsink.c: Convert to use the boilerplate macro.
* gst/tcp/gsttcp.c (gst_tcp_socket_read): Comment update.
Original commit message from CVS:
2005-10-17 Andy Wingo <wingo@pobox.com>
* ext/theora/theoraenc.c (theora_buffer_from_packet): Pass the
alloc_buffer flow return to callers.
(theora_enc_chain, theora_enc_chain): Adapt to buffer_from_packet
change. Fix some memleaks in theoraenc.
Original commit message from CVS:
2005-10-17 Andy Wingo <wingo@pobox.com>
* ext/ogg/gstoggmux.c (gst_ogg_mux_send_headers): Fix a segfault
in strange circumstance.
Original commit message from CVS:
2005-10-17 Julien MOUTTE <julien@moutte.net>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size): We are asked to compute a buffer
size
from caps, let's use the caps...
Original commit message from CVS:
2005-10-17 Thomas Vander Stichele <thomas at apestaart dot org>
* configure.ac:
put back AX_CREATE_STDINT_H, ffmpegcolorspace includes _stdint.h
Original commit message from CVS:
2005-10-16 Andy Wingo <wingo@pobox.com>
* gst/playback/gstdecodebin.c
(gst_element_set_state_like_a_crazy_man): New kraaaaaaazy
function!
(try_to_link_1): Increase kraziness level.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_new_from_id3v1):
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
(gst_tag_to_vorbis_comments):
Fix handling of GST_TAG_DATE, which is now of GST_TYPE_DATE.
Original commit message from CVS:
- Don't use non-portable LL suffix on constants, since MSVC doesn't allow
them. These constants all fit into ints anyway.
- Continue to hate nano.
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-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Cleanups.
Commit and read from ringbuffer in samples rather than bytes.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Respect segment rate and accum when scheduling samples.
Original commit message from CVS:
2005-10-11 Julien MOUTTE <julien@moutte.net>
* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
(gst_ogg_mux_collected): Quick hack to fix build. We need to
handle
EOS correctly, that needs more work.
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:
* check/elements/audioconvert.c: (setup_audioconvert),
(cleanup_audioconvert), (get_int_caps), (verify_convert),
(GST_START_TEST), (audioconvert_suite):
clean up tests a little, fix some leaks.
Original commit message from CVS:
* ext/alsa/gstalsasink.c:
Also allow unsigned int.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Small cleanup
Original commit message from CVS:
* check/pipelines/simple_launch_lines.c: (run_pipeline):
Small update, use API as stated in design docs.
* examples/seeking/seek.c: (make_avi_msmpeg4v3_mp3_pipeline),
(update_scale), (do_seek), (seek_cb), (set_update_scale),
(start_seek), (stop_seek), (play_cb), (pause_cb), (stop_cb),
(message_received), (main):
Updated seek example for GOption. Some usability improvements.
Original commit message from CVS:
* gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_create),
(gst_tcpserversrc_start):
Don't block in accept while doing the state change, move
to poll and make cancellable.
Original commit message from CVS:
2005-10-09 Philippe Khalaf <burger@speedy.org>
* gst-libs/gst/rtp/rtpbasedepayload.c:
Set timestamp and add queue delay to timestamp
* gst-libs/gst/rtp/rtpbuffer.h:
Set correct payload type for h263
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (wavpack_type_find),
(plugin_init):
Add wavpack and spc typefind functions from 0.8 branch.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (tar_type_find),
(ar_type_find), (msdos_type_find), (plugin_init):
Add typefind functions for tar archives, ar archives,
RAR archives, and msdos-executables (dlls, exe, etc.).
Some of those would be wrongly identified as mpeg
streams of some sort before (#315550).
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query), (gst_adder_class_init),
(gst_adder_init), (gst_adder_request_new_pad),
(gst_adder_change_state):
Add query function to source pad, so adder reports the correct
time/sample position when queried (#315457); fix state change
function; use GST_DEBUG_FUNCPTR() for pad functions.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (utf8_type_find):
Fix leaks in typefind registration
Clean up the gratuitous commenting and whitespacing a little
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
Only actually wait for the thread to be stopped if it's
running.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
If we receive EOS we can start playback of what we had.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
(gst_multifdsink_finalize), (multifdsink_hash_remove),
(gst_multifdsink_stop):
Fix crasher when going to NULL multiple times.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_event),
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_read):
patch from Edgard Lima <edgard.lima@indt.org.br>
Fixed gstbaseaudiosrc adding ring buffer sync to it.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_loop):
Report the FLOW_RETURN as string in the error message.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_clear_all):
Don't assert when clearing an unnegotiated buffer.
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:
* ext/gnomevfs/gstgnomevfssrc.c (gst_gnomevfssrc_uri_get_protocols):
protect gst_gnomevfs_get_supported_uris by a mutex, to make it
MT safe.
Original commit message from CVS:
2005-10-02 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_clear)
(gst_ring_buffer_prepare_read):
* gst-libs/gst/audio/gstaudiosink.c (audioringbuffer_thread_func):
Demote to LOG.
Original commit message from CVS:
2005-09-28 Andy Wingo <wingo@pobox.com>
* gst/videotestsrc/gstvideotestsrc.c: Implement live source mode
and unlocking.
Original commit message from CVS:
2005-09-28 Andy Wingo <wingo@pobox.com>
* gst/tcp/gsttcpclientsink.c (gst_tcpclientsink_base_init):
Actually add the pad template.
(gst_tcpclientsink_get_type): We're a base sink. Woot, works.
* gst/tcp/gsttcpserversrc.c: Go ahead and fix up serversrc while
I'm at it...
Original commit message from CVS:
2005-09-28 Andy Wingo <wingo@pobox.com>
* gst/tcp/gsttcpclientsrc.c: Make interruptable -- code stolen
from fdsrc. Get caps in create() instead of start() so it can be
interrupted. Interruption somewhat untested.
* gst/tcp/gsttcp.c (gst_tcp_read_buffer, gst_tcp_socket_read):
Proper EOS handling.
Original commit message from CVS:
2005-09-27 Andy Wingo <wingo@pobox.com>
* gst/tcp/gsttcpserversrc.c:
* gst/tcp/gsttcpclientsrc.c: Updated for new gsttcp API.
* gst/tcp/gsttcp.h:
* gst/tcp/gsttcp.c (gst_tcp_read_buffer): New function, factored
out of tcpclientsrc.c. Cancellable.
(gst_tcp_socket_read): Made private, cancellable, with better
diagnostics. Also the FIONREAD ioctl takes a int*, not a size_t*.
(gst_tcp_gdp_read_buffer): Made cancellable, actually returns the
whole buffer, and better diagnostics.
(gst_tcp_gdp_read_caps): Same.
* gst/sine/gstsinesrc.c (gst_sinesrc_wait): Add the base time.
Original commit message from CVS:
2005-09-26 Andy Wingo <wingo@pobox.com>
* gst/sine/gstsinesrc.h:
* gst/sine/gstsinesrc.c: Refactor, remove the table lookup code,
change the 'sync' property to 'is-live' and implement it halfway,
update for controller api change.
* gst/volume/gstvolume.c (volume_transform_ip): Update for
controller api change.
Original commit message from CVS:
* gst/audioresample/Makefile.am:
* gst/audioresample/debug.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/resample.c: Convert to using gst debugging
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:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_clear),
(gst_vorbisenc_sink_event), (gst_vorbisenc_change_state):
Don't flush encoder state unless we have an initialised encoder.
Clear out encoder state on PAUSED_TO_READY.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_setcaps), (gst_basertppayload_chain),
(gst_basertppayload_set_options), (gst_basertppayload_set_outcaps),
(gst_basertppayload_is_filled), (gst_basertppayload_push),
(gst_basertppayload_set_property),
(gst_basertppayload_get_property),
(gst_basertppayload_change_state):
* gst-libs/gst/rtp/gstbasertppayload.h:
Added max-ptime to control amount of data in the rtp packets.
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:
* check/Makefile.am:
have some tests be disabled for valgrinding
* check/elements/vorbisdec.c: (cleanup_vorbisdec),
(GST_START_TEST):
* ext/vorbis/vorbisdec.c: (vorbisdec_finalize):
Fix A Leak. Chain To Parent Finalize.
