Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/videorate/gstvideorate.c: (gst_videorate_blank_data),
(gst_videorate_init), (gst_videorate_chain),
(gst_videorate_change_state):
Event handling (fixes#159986).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
Add BLZ0 (Blizzard's version of DivX) fourcc.
Original commit message from CVS:
* gst/ffmpegcolorspace/imgconvert.c: (get_pix_fmt_info),
(avcodec_get_chroma_sub_sample), (avcodec_get_pix_fmt_name),
(avcodec_get_pix_fmt), (avpicture_layout),
(avcodec_get_pix_fmt_loss), (avg_bits_per_pixel), (img_copy),
(get_convert_table_entry), (img_convert), (img_get_alpha_info):
Fix code to not use GCC extensions (and c99 extensions that
Forte does not like.)
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_ebmlnum_uint),
(gst_matroska_ebmlnum_sint), (gst_matroska_demux_parse_blockgroup):
Lace sizes can be zero.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index):
Work for truncated (unfinished download etc.) files. Fixes#160514.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
Fix for integer overflow. Makes #156001 not crash. Probably masks
the real bug.
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:
* configure.ac: look for musepack headers as musepack/*.h
(fixes#159847)
* ext/musepack/gstmusepackdec.h: use <musepack/*.h>
* ext/musepack/gstmusepackreader.h: same
Original commit message from CVS:
* gst/audioscale/gstaudioscale.c: allow passthru of >2 channel
audio. does _not_ attempt or allow conversion unless channels
is 1 or 2.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (aac_rate_idx), (aac_profile_idx),
(gst_matroska_demux_audio_caps):
Some MPEG-AAC hacks, because else it doesn't work...
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_class_init),
(gst_dvd_demux_reset), (gst_dvd_demux_change_state):
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_reset),
(gst_mpeg_demux_change_state):
Reset on ready. Fixes 160276.
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_pad_link):
Fix memleak (#154815).
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init),
(gst_musicbrainz_init), (gst_musicbrainz_chain),
(gst_musicbrainz_set_property), (gst_musicbrainz_get_property):
* ext/musicbrainz/gsttrm.h:
Add support for using a proxy server when getting a trm id from
the MusicBrainz database (#149613).
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* sys/oss/gstosselement.c: (gst_osselement_probe_caps):
* sys/oss/oss_probe.c: (main):
Check for mono/stereo support (similar to samplerate probing),
fixes#159433. Also add missing copyright header to oss_probe.c.
Original commit message from CVS:
* configure.ac: add audioresample and cairo plugins. Remove
HAVE_MMX stuff, because it's not used.
* ext/Makefile.am: same
* ext/audioresample/Makefile.am: You are not ready for an
audio resampling element based on audioresample.
* ext/audioresample/gstaudioresample.c:
* ext/audioresample/gstaudioresample.h:
* ext/cairo/Makefile.am: You are not ready for overlay elements
based on cairo. Don't look too closely, these elements kinda
suck right now.
* ext/cairo/gstcairo.c: new
* ext/cairo/gsttextoverlay.c: new
* ext/cairo/gsttextoverlay.h: new
* ext/cairo/gsttimeoverlay.c: new
* ext/cairo/gsttimeoverlay.h: new
* gst-libs/gst/media-info/media-info-priv.h: fix compile
problem with compilers that don't support variadic macros.
Original commit message from CVS:
Reviewed by: David Schleef <ds@schleef.org>
* sys/sunaudio/gstsunaudio.c: (plugin_init): Apply patch from
Bala, registering sunaudiosrc (oops!), and cleaning up code a
bit. Also ran indent-gst.
* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_init),
(gst_sunaudiosrc_change_state), (gst_sunaudiosrc_get),
(gst_sunaudiosrc_setparams):
Original commit message from CVS:
2004-12-14 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
Add typefinding for mpeg2 pes streams
Original commit message from CVS:
* configure.ac: Applied patch from bug #143659, making default
sources and sinks OS-dependent (for Solaris), and added code
for OS/X.
* gconf/gstreamer.schemas.in: use OS-dependent sinks in gconf.
Original commit message from CVS:
* gst/asfdemux/README
* gst/wavenc/riff.h
* gst-libs/gst/riff/riff-ids.h
* gst-libs/gst/riff/riff-media.c
add new 4CC codes for h263 related codecs
fixes partially bug #155163
Original commit message from CVS:
2004-12-11 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* gst/interleave/deinterleave.c:
fix my name's spelling! :)
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_loop):
Align by packetsize, and assert that we a packet available before
playing. The first makes webstreams work (they often include
trailing padding data in a packet), the second allows pausing a
ASF stream in totem without getting demux errors afterwards.
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:
2004-12-09 Thomas Vander Stichele <thomas at apestaart dot org>
* configure.ac: move GCONF macro outside conditional for the am
conditional. Fixes#160439
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_handle_src_query):
Don't set DEFAULT, unsupported - makes length display incorrectly
in some cases.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps), (gst_alsa_close_audio):
* ext/alsa/gstalsa.h:
refactor big chunks of the core caps negotiation code to make it
a lot faster, because people claim it's really slow
(actually, just cache the getcaps when the device is opened)
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_init),
(gst_a52dec_handle_event), (gst_a52dec_update_streaminfo),
(gst_a52dec_handle_frame), (gst_a52dec_chain),
(gst_a52dec_change_state), (plugin_init):
* ext/a52dec/gsta52dec.h:
Do something useful with timestamps. Make chain-based (since
there's really no reason to be loopbased).
* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
Update current_byte/frame correctly.
Original commit message from CVS:
* gst/matroska/ebml-read.c: (gst_ebml_read_class_init),
(gst_ebml_read_init), (gst_ebml_read_use_event),
(gst_ebml_read_element_id), (gst_ebml_peek_id),
(gst_ebml_read_seek), (gst_ebml_read_skip),
(gst_ebml_read_reserve), (gst_ebml_read_buffer),
(gst_ebml_read_master):
* gst/matroska/ebml-read.h:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_contents),
(gst_matroska_demux_loop_stream), (gst_matroska_demux_audio_caps):
Disgustingly evil hack for working around INTERRUPT events and
their extremely annoying habit of being a pain in the ass. We
simply peek a cluster before reading any of it.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_chanpos_from_gst),
(gst_faad_chanpos_to_gst), (gst_faad_chain):
Set DURATION even if source buffer didn't. Also use increasing
timestamps.
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Block_align can have larger values than 8192.
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_chain):
Make error actually say something useful (fixes#156798).
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add Intel Video 5.0 fourcc (IV50).
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:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/law/alaw-decode.c: (alawdec_getcaps):
* gst/law/mulaw-decode.c: (mulawdec_getcaps):
Prevent warnings when negotiating caps (fixes#159338).
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
Don't forward DISCONT events (fixes#159684).
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:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_massage_index):
Fix quite humiliating bug in omitting 0-sized index chunks but
forgetting to count them for timestamps.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_audio_convert_mix):
more overwriting protection due to modifying channels one by one
instead of all at once
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_audio_convert_mix):
walk the samples backwards if out_channels > in_channels so we don't
overwrite data
Original commit message from CVS:
2004-11-28 Martin Soto <martinsoto@users.sourceforge.net>
* ext/alsa/gstalsasink.c (gst_alsa_sink_loop):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsa.c (gst_alsa_set_clock):
Make alsasink actually honor gst_element_set_clock and use that
clock instead of ist internal one.
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:
* ext/a52dec/gsta52dec.c: (gst_a52dec_loop),
(gst_a52dec_change_state):
Don't do sample adjusting anymore, we use float audio now.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
Don't fixate to non-existing properties.
Original commit message from CVS:
2004-11-27 Martin Soto <martinsoto@users.sourceforge.net>
* gst-libs/gst/audio/audioclock.c (gst_audio_clock_set_active)
(gst_audio_clock_get_internal_time):
Fix active <-> inactive transitions: ensure time value always
grows and avoid abrupt value changes.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps):
Don't omit the last (which incase of dmix is the only :) )
channel count. Don't set channels if <= 2.
Original commit message from CVS:
* ext/vorbis/oggvorbisenc.c
* ext/vorbis/vorbisenc.c :
change description fields of those plugins to differentiate them
(pitivi show Encoders by description, they had the same one)
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:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_push):
Fix position for discont if we're close as well. Nitpicking, but
saves a few milliseconds of extra waiting or skipping.
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:
patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
* ext/ogg/gstoggdemux.c:
Fix sync on broken files. Fixes#158976
Original commit message from CVS:
* gst/synaesthesia/gstsynaesthesia.c:
(gst_synaesthesia_class_init), (gst_synaesthesia_init),
(gst_synaesthesia_dispose), (gst_synaesthesia_finalize),
(gst_synaesthesia_sink_link), (gst_synaesthesia_src_getcaps),
(gst_synaesthesia_src_link), (gst_synaesthesia_chain),
(gst_synaesthesia_change_state), (plugin_init):
Fix up synaesthesia to work under different samplerates/ buffer sizes.
Force 320x200 output, as that's the only thing the underlying
synaesthesia implementation supports. Still needs to be made
re-entrant.
Original commit message from CVS:
reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
Fix for gcc-2.95 (fixes#158221).
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:
* ext/alsa/gstalsa.c: (gst_alsa_pcm_wait):
add debugging
* ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
do a wait when we enter the loop func with no data available to
write instead of getting into an 100% CPU loop by just returning and
being called again by the scheduler
Original commit message from CVS:
* configure.ac:
* ext/libvisual/visual.c: (gst_visual_get_type),
(libvisual_log_handler), (gst_visual_getcaps),
(gst_visual_srclink), (gst_visual_change_state), (make_valid_name),
(plugin_init):
Update libvisual to 0.1.7. Link in the debug handling to gstreamer
* ext/smoothwave/Makefile.am:
* ext/smoothwave/demo-osssrc.c: (main):
* ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init),
(gst_smoothwave_init), (gst_smoothwave_dispose), (gst_sw_sinklink),
(gst_sw_srclink), (gst_smoothwave_chain), (gst_sw_change_state),
(plugin_init):
* ext/smoothwave/gstsmoothwave.h:
Make gstsmoothwave a working element in the 20th century.
* gst/chart/gstchart.c: (gst_chart_init), (gst_chart_srcconnect):
Fix incorrect link function
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
Fix for negotiation order problem. This would show when the
ALSA loopfuction was called before any other function. ALSA
wouldn't do anything because we're not negotiated yet, leading
to an infinite loop. Showed in e.g. Rhythmbox. Fixes#158006.
Original commit message from CVS:
reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
No warnings (#157986).
Original commit message from CVS:
* gst/ffmpegcolorspace/imgconvert.c: (yuv420p_to_yuv422):
Actually test for odd width/height rather than testing whether
a temporary variable that was 0 before we subtracted 1 is now
not equal to zero (which it always is).
Original commit message from CVS:
2004-11-11 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* sys/v4l2/gstv4l2element.c: (gst_v4l2_iface_supported):
Fix compilation if HAVE_XVIDEO is not defined
Original commit message from CVS:
2004-11-11 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* sys/v4l/gstv4lelement.c: (gst_v4l_iface_supported):
Fix compilation if HAVE_XVIDEO is not defined
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_change_state):
Only set hardware parameters *after* negotiation. Before
negotiation, it will set ANY and that seems to cause crashes
(see e.g. #151288, #153227).
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsa_sink_get_time):
This seems to be antique leftover. It needs to pass error
checking.
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_init),
(gst_sdlvideosink_deinitsdl), (gst_sdlvideosink_initsdl),
(gst_sdlvideosink_destroy), (gst_sdlvideosink_create),
(gst_sdlvideosink_sinkconnect), (gst_sdlvideosink_chain):
Fix GstXOverlay implementation (#151059).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
Disable halfway-seek for pending release (since it needs a new
core release).
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:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps_internal):
buffer-frames property was missing
* ext/arts/gst_arts.c:
rate missing from sinkcaps
* ext/audiofile/gstafparse.c:
* ext/audiofile/gstafsink.c:
* ext/audiofile/gstafsrc.c:
* ext/swfdec/gstswfdec.c:
int audio doesn't know buffer-frames
* ext/cdparanoia/gstcdparanoia.c:
int audio doesn't know chunksize either
* ext/nas/nassink.c:
it's endianness, not endianess
* gst-libs/gst/audio/audio.h:
make float standard pad template caps really describe float
* gst/law/mulaw.c: (linear_factory):
signed only, please
* gst/mpegstream/gstdvddemux.c:
widths of 20 are not valid
Original commit message from CVS:
* gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_ayuv),
(gst_videomixer_fill_checker), (gst_videomixer_fill_color),
(gst_videomixer_blend_buffers), (gst_videomixer_loop):
Only mix AYUV for maximum quality.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (get_relative), (gst_ogg_demux_src_query),
(gst_ogg_demux_push), (gst_ogg_pad_push):
Let's act as if we're synchronized now! :).
* ext/theora/theoradec.c: (theora_dec_chain):
Add some debug.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
Actually always send a discont (cornercase when resending the
same serial-tagged chain twice).
Original commit message from CVS:
2004-11-08 Julien MOUTTE <julien@moutte.net>
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
(gst_ximagesink_finalize):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
(gst_xvimagesink_finalize): Some more cleanups, leaks fixed and checks.
