Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_type_get), (qtdemux_audio_caps):
* gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
(plugin_init):
Add 3GP (variables name Q3GP because they can't start with a
number). Add samr audio fourcc (used in .3gp files), decoder
is work in progress. Also do a GST_WARNING instead of ERROR
in case of unknown nodes, to decrease output.
Original commit message from CVS:
* configure.ac:
* ext/speex/gstspeexdec.h:
* ext/speex/gstspeexenc.h:
Fixes#158382. Make speex plugin compatible with both 1.0 and 1.1.
Fix detection code in configure.ac
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup):
Save position, so that queries give proper return values. Don't
know how this could ever have worked before...
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_stream_scan):
Add some more debug. Fix logic error when setting movi offset
while reading index.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_stream_scan), (gst_avi_demux_handle_seek),
(gst_avi_demux_process_next_entry):
Add some debugging. Better detection of broken indexes and the
accompanying index recovery. No infinite loops on state changes
when we're still in our loopfunction.
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* sys/sunaudio/gstsunmixer.c: (gst_sunaudiomixer_set_volume):
Normalizing the value before setting
(gst_sunaudiomixer_get_volume):
Normalizing the value after getting. Fixes bug# 161980
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:
* sys/v4l/gstv4lelement.c: (gst_v4l_iface_supported):
* sys/v4l2/gstv4l2element.c: (gst_v4l2_iface_supported):
g_assert() can be a macro, don't use #ifdef inside it.
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
