Original commit message from CVS:
add Sun Audio plugin. Verified that nothing breaks and that make check works.
Don't think the docs gets properly built yet, but I don't understand exactly how to enable that.
Original commit message from CVS:
2005-01-07 Philippe Khalaf <philippe.kalaf@collabora.co.uk>
* gst-plugins-good/gst/udp/gstdynudpsink.c:
* gst-plugins-good/gst/udp/gstudpsrc.c:
Allow udpsrc and dynudpsink to take a sockfd as a parameter. For udpsrc,
overrides the port or multicast parameters. Fixes bugs #323021.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad):
Add gst_element_no_more_pads() for proper decodebin behaviour.
* gst/id3demux/id3v2frames.c: (parse_comment_frame),
(parse_text_identification_frame), (parse_split_strings):
Failure to decode some tags is not a GST_ERROR() but a
GST_WARNING()
When iterating over a chunk of text, check that we haven't gone too
far.
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* ext/flac/gstflacdec.c: (gst_flac_dec_write),
(gst_flac_dec_convert_src), (gst_flac_dec_src_query),
(gst_flac_dec_change_state):
Don't g_assert() where we should just return FALSE; remove
unnecessary g_assert(); initialize some fields properly in
state change function (fixes#325504). Also, use
GST_DEBUG_OBJECT in two more places.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render),
(gst_multiudpsink_remove), (gst_multiudpsink_get_stats):
* gst/udp/gstmultiudpsink.h:
Track packets sent per client in addition to bytes sent; provide
this info through get-stats signal
Original commit message from CVS:
* gst/id3demux/id3tags.c: (id3demux_read_id3v2_tag):
If a broken tag has 0 bytes payload, at least still skip
the 10 byte header
Original commit message from CVS:
2005-12-22 Philippe Khalaf <burger@speedy.org>
* gst-plugins-good/gst/rtp/gstrtph263pdepay.h:
* gst-plugins-good/gst/rtp/gstrtph263pdepay.c:
* gst-plugins-good/gst/rtp/gstrtpmp4vdepay.h:
* gst-plugins-good/gst/rtp/gstrtpmp4vdepay.c:
Making these depayloaders (H263+ and mpeg4 video) inherit from
RtpBaseDepayloaderClass. Fixes bugs #323922 and #323908.
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
Regenerate the plugin hiearchy.
Original commit message from CVS:
2005-12-21 Jan Schmidt <thaytan@mad.scientist.com>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* gst/id3demux/gstid3demux.c: (gst_id3demux_get_type),
(gst_id3demux_base_init), (gst_id3demux_class_init),
(gst_id3demux_chain):
* gst/id3demux/gstid3demux.h:
Add documentation for id3demux.
Don't fail if the first buffer is not at offset 0, just
attempt to typefind and do pass through
Rename the gst_type function from gst_gst_id3demux..
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render),
(gst_multiudpsink_add), (gst_multiudpsink_remove),
(gst_multiudpsink_get_stats):
* gst/udp/gstmultiudpsink.h:
Collect statistics; return them from get_stats.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream),
(gst_avi_demux_parse_odml), (gst_avi_demux_peek_tag),
(gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan),
(gst_avi_demux_stream_header), (gst_avi_demux_loop):
Construct index for indexless files.
Make sure pad/buffers are correctly reset to NULL once we don't need
them anymore, else we get lovely segfaults/assertions.
* gst/wavparse/gstwavparse.c:
Yes, you can have 96KHz audio and wma in wav :(
Original commit message from CVS:
* ext/esd/esdmon.c: (gst_esdmon_open_audio):
* ext/esd/esdsink.c: (gst_esdsink_prepare):
* gst/multipart/multipartdemux.c:
change some char* into char[]
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
(gst_wavparse_other), (gst_wavparse_perform_seek),
(gst_wavparse_stream_headers), (gst_wavparse_stream_data),
(gst_wavparse_loop), (gst_wavparse_pad_convert),
(gst_wavparse_srcpad_event), (gst_wavparse_sink_activate_pull):
* gst/wavparse/gstwavparse.h:
Use GstSegment to implement more seeking features.
Original commit message from CVS:
* gst/rtsp/rtspconnection.c:
Add <netinet/in.h> include and move <arpa/inet.h> include
to make things work on OpenBSD as well (fixes#323717;
patch by: Benjamin Pineau)
Original commit message from CVS:
2005-12-14 Philippe Khalaf <burger@speedy.org>
* gst-plugins-good/gst/rtp/gstasteriskh263.c:
* gst-plugins-good/gst/rtp/gstrtpamrdepay.c:
* gst-plugins-good/gst/rtp/gstrtpamrpay.c:
* gst-plugins-good/gst/rtp/gstrtpg711depay.c:
* gst-plugins-good/gst/rtp/gstrtpg711depay.c:
* gst-plugins-good/gst/rtp/gstrtpgsmdepay.c:
* gst-plugins-good/gst/rtp/gstrtph263pay.c:
* gst-plugins-good/gst/rtp/gstrtph263pdepay.c:
* gst-plugins-good/gst/rtp/gstrtph263ppay.c:
* gst-plugins-good/gst/rtp/gstrtpmp4vdepay.c:
* gst-plugins-good/gst/rtp/gstrtpmp4vpay.c:
* gst-plugins-good/gst/rtp/gstrtpmpadepay.c:
* gst-plugins-good/gst/rtp/gstrtpmpapay.c:
* gst-plugins-good/gst/rtp/README:
Fixed payload range in payloder caps. Removed payload range completly from
depayloaders as they don't require payload type in their caps. In effect,
there isn't any specific payload type for any given codec, only suggestions.
Fixes bug #324011.
Original commit message from CVS:
2005-12-13 Julien MOUTTE <julien@moutte.net>
* gst/videomixer/videomixer.c: (gst_videomixer_init),
(gst_videomixer_fill_queues), (gst_videomixer_blend_buffers),
(gst_videomixer_collected): Code cleanup and re-enabling
queued time validity check for correct EOS handling.
Original commit message from CVS:
* sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init),
(gst_oss_mixer_element_get_property),
(gst_oss_mixer_element_change_state):
Add 'device-name' property and fix state change function.
Original commit message from CVS:
* gst/flx/gstflxdec.c: (gst_flxdec_chain):
If the speed of the file is null in the header, set the frame_time to the default
setting of GST_SECOND / 70. Which is the default frame_delay for .fli files as
stated in this document : http://www.compuphase.com/flic.htm
Would be nice to have the time conversion done properly too
(duration = flxh->frames * flxdec->frame_time)
Original commit message from CVS:
2005-12-12 Julien MOUTTE <julien@moutte.net>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* gst/videomixer/videomixer.c:
(gst_videomixer_pad_sink_setcaps),
(gst_videomixer_getcaps), (gst_videomixer_fill_queues),
(gst_videomixer_update_queues), (gst_videomixer_collected):
Adding
documentation for videomixer on my way with a funny sample
pipeline.
Original commit message from CVS:
* gst/auparse/gstauparse.c: (gst_au_parse_base_init),
(gst_au_parse_class_init), (gst_au_parse_init),
(gst_au_parse_dispose), (gst_au_parse_chain),
(gst_au_parse_change_state), (plugin_init):
* gst/auparse/gstauparse.h:
Use gst_object_unref() for GstObjects instead of
g_object_unref() and fix a mem leak in a debug
statement; while we're at it, also borgify, use
boilerplate macros and clean up a little bit.
Original commit message from CVS:
* gst/goom/gstgoom.c:
* gst/level/level-example.c: (main):
* gst/smoothwave/demo-osssrc.c: (main):
Use audiotestsrc instead of sinesrc (#323798).
Original commit message from CVS:
* gst/debug/efence.c: (gst_efence_init), (gst_efence_chain),
(gst_fenced_buffer_copy):
Make sure GST_BUFFER_DATA is set on fenced copied buffers; fix
GST_DEBUG crasher where GST_TIME_FORMAT was not used in
conjunction with GST_TIME_ARGS. Also, don't leak pad templates
and use GST_DEBUG_FUNCPTR for pad functions.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_dec_base_init),
(gst_flac_dec_class_init), (gst_flac_dec_init),
(gst_flac_dec_metadata_callback), (gst_flac_dec_error_callback),
(gst_flac_dec_eof), (gst_flac_dec_write), (gst_flac_dec_loop),
(gst_flac_dec_convert_src), (gst_flac_dec_get_src_query_types),
(gst_flac_dec_src_query), (gst_flac_dec_send_newsegment),
(gst_flac_dec_handle_seek_event), (gst_flac_dec_src_event),
(gst_flac_dec_change_state):
* ext/flac/gstflacdec.h:
Rewrite flacdec a bit, so that even seeking might work now. Most
importantly, don't act upon any flow return values we get, just tell
the decoder everything's dandy and act on the flow return values
later on in the loop function. We don't want to mess up the internal
decoder state for non-fatal things like flushing pads etc. Other
than that, use GstSegment (segment seeks don't work yet though, but
should be easy to add), use boilerplate macros, drop the superfluous
'flacdec:' from debug messages, use gst_util_uint64_scale_int, and
lots of other things.
Original commit message from CVS:
* configure.ac:
Update comment in OSS includes check.
* sys/oss/gstossdmabuffer.c:
* sys/oss/gstosshelper.c:
* sys/oss/gstossmixer.c:
* sys/oss/gstossmixertrack.c:
* sys/oss/gstosssink.c:
* sys/oss/gstosssrc.c:
* sys/oss/oss_probe.c:
Don't assume the OSS soundcard.h include is always in
the sys/ directory. Instead, use the existing defines
from config.h to include the right file. Fixes
compilation on OpenBSD 3.8 (#323718).
Original commit message from CVS:
* ext/flac/gstflacdec.c: (raw_caps_factory), (gst_flacdec_write):
Accept a wider range of flac files, more closely matching flac sp
