Original commit message from CVS:
* configure.ac:
* ext/flac/gstflacenc.c: (gst_flacenc_init),
(gst_flacenc_seek_callback), (gst_flacenc_write_callback),
(gst_flacenc_tell_callback), (gst_flacenc_chain),
(gst_flacenc_change_state):
* ext/flac/gstflacenc.h:
Update for API change in flac-1.1.1. Update requirement in
configure.ac. Fixes#162974.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flacdec_src_query):
Only return true if we actually filled something in. Prevents
player applications from showing a random length for flac files.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
(gst_riff_read_use_event), (gst_riff_read_handle_event),
(gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh),
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs):
OK, ok, so I implemented event handling. Apparently it's normal
that we receive random events at random points without asking
for it.
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index), (gst_avi_demux_stream_header),
(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Implement non-lineair chunk handling and subchunk processing.
The first solves playback of AVI files where the audio and video
data of individual buffers that we read are not synchronized.
This should not happen according to the wonderful AVI specs, but
of course it does happen in reality. It is also a prerequisite for
the second. Subchunk processing allows us to cut chunks in small
pieces and process each of these pieces separately. This is
required because I've seen several AVI files with incredibly large
audio chunks, even some files with only one audio chunk for the
whole file. This allows for proper playback including seeking.
This patch is supposed to fix all AVI A/V sync issues.
* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
(flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop):
Work.
* gst/modplug/gstmodplug.cc:
Proper return value setting for the query() function.
* gst/playback/gstplaybasebin.c: (setup_source):
Being in non-playing state (after, e.g., EOS) is not necessarily
a bad thing. Allow for that. This fixes playback of short files.
They don't actually playback fully now, because the clock already
runs. This means that small files (<500kB) with a small length
(<2sec) will still not or barely play. Other files, such as mod
or flx, will work correctly, however.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flacdec_class_init),
(gst_flacdec_write): Actually, GST_PAD_CAPS() has nothing to
do with the logic.
Original commit message from CVS:
2004-06-01 Christophe Fergeau <teuf@gnome.org>
* ext/flac/gstflactag.c: strip ending framing bit from vorbiscomment
buffer since libflac doesn't expect it (reports a sync error when
it encounters that)
Original commit message from CVS:
second batch :
remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
(in gst-plugins/ext/ this time)
Original commit message from CVS:
2004-03-06 Christophe Fergeau <teuf@gnome.org>
* ext/faac/gstfaac.c: (gst_faac_chain):
* ext/flac/gstflactag.c: (gst_flac_tag_chain):
* ext/libpng/gstpngenc.c: (user_write_data):
* ext/mikmod/gstmikmod.c: (gst_mikmod_loop):
* gst/ac3parse/gstac3parse.c: (gst_ac3parse_chain):
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_chain_subtitle):
* gst/mpegstream/gstrfc2250enc.c: (gst_rfc2250_enc_add_slice):
Fix several misuse of gst_buffer_merge (it doesn't take ownership
of any buffer), should fix some leaks. I hope I didn't unref buffers
that shouldn't be...
Original commit message from CVS:
2004-02-22 Benjamin Otte <otte@gnome.org>
reported by: Stefan Kost <kost@imn.htwk-leipzig.de>
* gst/audioconvert/gstaudioconvert.c: (plugin_init):
set rank to PRIMARY
* gst/volume/gstvolume.c: (plugin_init):
set rank to NONE
fixes#134960
2004-02-22 Julio M. Merino Vidal <jmmv@menta.net>
reviewed by Benjamin Otte <otte@gnome.org>
* ext/flac/gstflacenc.c: (gst_flacenc_chain):
escape NULL strings in GST_ELEMENT_ERROR properly (fixes#135116)
Original commit message from CVS:
Replace use of GST_PAD_FORMATS_FUNCTION() and similar macros
with the code that they would expand to.
* ext/flac/gstflacdec.c: (gst_flacdec_get_src_formats),
(gst_flacdec_get_src_query_types),
(gst_flacdec_get_src_event_masks):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get_formats),
(gst_gnomevfssrc_get_query_types),
(gst_gnomevfssrc_get_event_mask):
Original commit message from CVS:
2004-01-31 Jeremy Simon <jesimon@libertysurf.fr>
* ext/flac/gstflacdec.c: (gst_flacdec_class_init):
* ext/flac/gstflacdec.h:
* ext/flac/gstflacenc.h:
Fix typos
Original commit message from CVS:
2004-01-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/flac/gstflacdec.c: (gst_flacdec_update_metadata):
* ext/flac/gstflacenc.c: (add_one_tag):
length is already host endian, no need to convert. Fixes playback of
tagged files on PPC. (bug #128384)
Original commit message from CVS:
2004-01-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/flac/gstflacdec.c: (gst_flacdec_update_metadata):
length is already host endian, no need to convert. Fixes playback of
tagged files on PPC.
Original commit message from CVS:
2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
* ext/flac/gstflacdec.c: (gst_flacdec_loop):
Finish flac decoder on EOS. See #116178.
Original commit message from CVS:
2004-01-03 Ronald Bultje <rbultje@ronald.bitfreak.net>
* ext/flac/gstflacdec.c:
* ext/libpng/gstpngenc.h:
* ext/mikmod/gstmikmod.h:
OS X fixes (see #126628).
Original commit message from CVS:
Fixed bug when filling tags in gstflacenc
Added merging support, and a mode which stops after outputting tags to flactag
Original commit message from CVS:
Added element to rewrite vorbiscomments embedded in flac files. It currently
doesn't honour the tag merge mode setting.
Changed application/x-flac to audio/x-flac in gstflacenc.c
Original commit message from CVS:
Apply patch from jesimon@libertysurf.fr (Jeremy SIMON) in #122368.
FLAC io functions should be called every time the encoder is initialized.
Original commit message from CVS:
-fix segfault on PAUSED => READY when metadata wasn't set
- fix caps mimetype that always caused a failure on caps nego
Original commit message from CVS:
Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files
Original commit message from CVS:
merge TYPEFIND branch. Major changes:
- totally reworked type(find) system
- all typefind functions are in gst/typefind now
- more typefind functions then before
- some plugins might fail to compile now because I don't have them installed and they
a) require bytestream or
b) haven't had their typefind fixed.
Please fix those plugins and put the typefind functions into gst/typefind if they don't have dependencies
Original commit message from CVS:
New typefind system:
* bytestream is now part of the core
* all plugins have been modified to use this new typefind system
* asf typefinding added
* mpeg video stream typefiding removed because it's broken
* duplicate typefind entries removed
* extra id3 typefinding added, because we've seen 4 types of files
(riff/wav, flac, vorbis, mp3) with id3 headers and each of these needs
to work. Instead, I've added an id3 element and let it redo typefiding
after the id3 header. this needs a hack because spider only typefinds
once. We can remove this hack once spider supports multiple typefinds.
* with all this, mp3 typefinding is semi-rewritten
* id3 typefinding in flac/vorbis is removed, it's no longer needed
* fixed spider and gst-typefind to use this, too.
* Other general cleanups
Original commit message from CVS:
New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs
Original commit message from CVS:
compatibility fix for new GST_DEBUG stuff.
Includes fixes for missing includes for config.h and unistd.h
I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately.
Original commit message from CVS:
_ Change from StreamEncoder to SeekableStreamEncoder
_ Add a seek callback to fill STREAMINFO blocks correctly
_ Add metadata (vorbis comments)
Original commit message from CVS:
fixes:
- wrong type of 0
- only write data into buffer if buffer is big enough
- write data into a buffer which we actually may write
Original commit message from CVS:
another batch of connect->link fixes
please let me know about issues
and please refrain of making them yourself, so that I don't spend double
the time resolving conflicts
Original commit message from CVS:
add ranks only for plugins who participate in autoplugging. If you have a file that used to autoplug but doesn't anymore, then let me know or add a rank to the missing element.
Original commit message from CVS:
* a hack to work around intltool's brokenness
* a current check for mpeg2dec
* details->klass reorganizations
* an element browser that uses details->klass
* separated cdxa parse out from the avi directory
Original commit message from CVS:
* removal of //-style comments
* don't link plugins to core libs -- the versioning is done internally to the plugins with the plugin_info struct,
and symbol resolution is lazy, so we can always know if a plugin can be loaded by the plugin_info data. in theory.
Original commit message from CVS:
s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/
@-substitued variables variables are defined as make variables automagically,
and this gives the user the freedom to say make GST_PLUGIN_LDFLAGS=-myflag
