Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (remove_prerolls),
(new_decoded_pad):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (setup_sinks):
Don't try to preroll or decode more than one audio/video
track.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_change_state):
Throw error if we failed to find a suitable output. This should
throw an error if we successfully set up a pipeline (e.g. because
we recognized a media file) but found no decodable streams in it
(e.g. because it contains only media stream types for which we
have no decoders, or because it's not a media type).
Original commit message from CVS:
* ext/dirac/Makefile.am:
* ext/dirac/gstdirac.cc:
* ext/dirac/gstdiracdec.cc:
* ext/dirac/gstdiracdec.h:
Do something. Don't actually know if this works because I don't
have a demuxer yet.
* ext/gsm/gstgsmdec.c: (gst_gsmdec_getcaps):
Add channels=1 to caps returned from _getcaps().
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_get_type),
(gst_ogm_video_parse_get_type), (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_parse_init),
(gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
(gst_ogm_parse_sink_convert), (gst_ogm_parse_chain),
(gst_ogm_parse_change_state):
Separate between audio/video so ogmaudioparse actually uses the
audio pad templates. Both audio and video work now, including
autoplugging. Also use sometimes-srcpad hack.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Handle events better. Don't hang on infinite loops.
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_init), (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_stream_header), (gst_avi_demux_stream_data),
(gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
Improve A/V sync. Still not perfect.
* gst/matroska/ebml-read.c: (gst_ebml_read_seek),
(gst_ebml_read_skip):
Handle events better.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
(gst_qtdemux_loop_header), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Add IMA4. Improve event handling. Save offset after a seek when
the headers are at the end of the file so that we don't end up in
an infinite loop.
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
Add low-priority typefind support for files with no length.
Original commit message from CVS:
* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_getcaps):
Correct caps negotiation
* gst/volume/gstvolume.c: (volume_chain_float),
(volume_chain_int16):
Modify debug output to be little more informative
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
(gst_xvimagesink_xvimage_destroy):
Add XSync calls after detaching from the shared memory segment to
avoid a crash.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (_read_var_length), (_read_guid),
(gst_asf_demux_process_segment), (gst_asf_demux_handle_data),
(gst_asf_demux_process_chunk), (gst_asf_demux_handle_sink_event):
Prevent infinite loops. More correct error reporting.
* gst/auparse/gstauparse.c: (gst_auparse_chain):
Error out if negotiation fails.
* gst/playback/gstplaybasebin.c: (setup_source),
(gst_play_base_bin_change_state), (gst_play_base_bin_error),
(gst_play_base_bin_found_tag):
Error/tag forwarding. Pre-roll fixes for source errors on state
changes (e.g. "file does not exist") to prevent hangs.
Original commit message from CVS:
* ext/mad/gstmad.c: (gst_mad_check_caps_reset),
(gst_mad_change_state):
Allow for mp3 rate/channels changes. However, only very
conservatively. Reason that we *have* to enable this is smiply
because the mad find_sync() function is not good enough, it will
regularly sync on random data as valid frames and therefore make
us provide random caps as *final* caps of the stream. The best fix
I could think of is to simply require several of the same stream
changes in a row before we change caps.
The actual testcase that works now is #
* ext/ogg/Makefile.am:
* ext/ogg/gstogg.c: (plugin_init):
* ext/ogg/gstogmparse.c:
OGM support (video only for now; I need an audio sample file).
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
(gst_asf_demux_process_stream), (gst_asf_demux_video_caps),
(gst_asf_demux_add_video_stream):
WMV extradata.
* gst/playback/gstplaybasebin.c: (unknown_type):
Don't error out on single unknown-types after all. It's wrong.
If we found type of video and audio but not of a subtitle stream,
it will still error out (which is unwanted). Will find a better fix
later on.
* gst/typefind/gsttypefindfunctions.c: (ogmvideo_type_find),
(ogmaudio_type_find), (plugin_init):
OGM support.
Original commit message from CVS:
* gst/tcp/gstfdset.c: (gst_fdset_fd_has_closed),
(gst_fdset_fd_has_error), (gst_fdset_fd_can_read),
(gst_fdset_fd_can_write), (gst_fdset_wait):
* gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
(gst_multifdsink_init), (gst_multifdsink_add),
(gst_multifdsink_remove), (gst_multifdsink_get_stats),
(gst_multifdsink_remove_client_link),
(gst_multifdsink_client_queue_buffer),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_recover_client), (gst_multifdsink_handle_clients),
(gst_multifdsink_close), (gst_multifdsink_change_state):
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcpserversink.c: (gst_tcpserversink_class_init),
(gst_tcpserversink_removed):
Small cleanups in fdset.c
Use a hastable to map fd to the client structure for faster
lookup in _remove and get_stats.
Added virtual function to close the fds.
Handle clients even when the select/poll call was unblocked because
of a command.
Implement syncing to keyframe in the recovery procedure.
Original commit message from CVS:
* configure.ac: remove NASM check, since we don't use it. Update
dirac check to 0.4
* ext/dirac/gstdiracdec.cc: update to current 0.4 API
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
Initialized variables.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_header), (qtdemux_parse), (qtdemux_parse_trak),
(gst_qtdemux_handle_esds), (qtdemux_audio_caps): Fix seeking, add
SVQ3 format
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_remove_client_link),
(gst_multifdsink_handle_clients), (gst_multifdsink_change_state):
Don't close the fd in multifdsink as we didn't open it in the
first place. Some cleanups.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (state_change), (setup_source),
(gst_play_base_bin_change_state):
Handle the case where we failed to setup a clear pipeline. This
will throw an error (or EOS, another nice case) and if you don't
catch that, the app will wait for the signal forever (and thus
hang).
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssink.c:
(gst_gnomevfssink_uri_get_protocols):
* ext/gnomevfs/gstgnomevfssrc.c:
(gst_gnomevfssrc_uri_get_protocols):
* ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
* ext/gnomevfs/gstgnomevfsuri.h:
Use _uri_new() instead of _open(), so it doesn't take as long and
Christophe's computer won't hang.
* gst/playback/gstplaybasebin.c: (unknown_type):
Throw error on unknown media type, so apps actually display it.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_overrun), (no_more_pads),
(setup_source), (gst_play_base_bin_set_property),
(gst_play_base_bin_add_element):
* gst/playback/gstplaybin.c: (gst_play_bin_send_event):
Some more work on making sure seeking pauses the pipeline and
that changing the uri actually does something.
Original commit message from CVS:
* gst/tcp/gstfdset.c: (gst_fdset_wait):
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_close):
* gst/tcp/gsttcpserversink.c: (gst_tcpserversink_init_send),
(gst_tcpserversink_close):
Be a bit more paranoid when freeing memory.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
(gst_play_base_bin_dispose), (gst_play_base_bin_set_property):
Handle double disposals, and proper change of URIs.
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_update),
(gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
(gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
(gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
Update mixer (to sync with other sessions) if we try to obtain
a new value. This makes alsamixer work accross applications.
* ext/alsa/gstalsasink.c: (gst_alsa_sink_get_time):
Only call sync functions if we're running, else alsalib asserts.
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_query):
Sometimes fails to compile. Possibly a gcc bug.
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element):
Add a reference to an application-provided object, because we lose
this same reference if we add it to the bin. If we don't do this,
we can only use this object once and thus crash if we go from
ready to playing, back to ready and back to playing again.
Also add an audioscale element because several cheap soundcards -
like mine - don't support all samplerates.
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
(gst_ximagesink_xcontext_clear), (gst_ximagesink_change_state):
Fix wrong order or PAR calls. Makes automatically obtained PAR
from the X server atually being used.
Original commit message from CVS:
* gst/tcp/gstfdset.c: (gst_fdset_free), (gst_fdset_set_mode),
(gst_fdset_get_mode), (gst_fdset_add_fd), (gst_fdset_remove_fd),
(gst_fdset_fd_ctl_write), (gst_fdset_fd_ctl_read),
(gst_fdset_fd_has_closed), (gst_fdset_fd_has_error),
(gst_fdset_fd_can_read), (gst_fdset_fd_can_write),
(gst_fdset_wait):
* gst/tcp/gstfdset.h:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_add),
(gst_multifdsink_client_queue_buffer),
(gst_multifdsink_handle_client_write):
* gst/tcp/gstmultifdsink.h:
Some extra checks in gstfdset.
Only use send() when the fd is a socket. Don't try to
read from write only fds.
Original commit message from CVS:
* gst/tcp/gstfdset.c: (ensure_size), (gst_fdset_wait):
Realloc test fdset in the lock and right before starting
the poll call. Bump the limit to 4096.
Original commit message from CVS:
2004-08-17 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* gst/audioscale/gstaudioscale.c:
* gst/audioscale/gstaudioscale.h:
made audioscale resample from any sample rate to any sample rate
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
(gst_multifdsink_init), (gst_multifdsink_add),
(gst_multifdsink_client_queue_buffer),
(gst_multifdsink_set_property), (gst_multifdsink_get_property):
* gst/tcp/gstmultifdsink.h:
Added option to send a keyframe to clients as the first buffer.
Make timeout property writable.
Original commit message from CVS:
* gst/tcp/gstfdset.c: (ensure_size), (gst_fdset_new),
(gst_fdset_add_fd), (gst_fdset_remove_fd),
(gst_fdset_fd_has_closed), (gst_fdset_fd_has_error),
(gst_fdset_fd_can_read), (gst_fdset_fd_can_write),
(gst_fdset_wait):
Make sure the pollfds are not changed when the poll call is
running. Protect against array out of bounds.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_unit_type_get_type),
(gst_client_status_get_type), (gst_multifdsink_class_init),
(gst_multifdsink_init), (gst_multifdsink_remove_client_link),
(gst_multifdsink_handle_client_read),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
(gst_multifdsink_handle_clients), (gst_multifdsink_set_property),
(gst_multifdsink_get_property):
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcp-marshal.list:
Starting to prepare for specifying buffer time in other units
than buffers. Expose remove reason in signal.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_add),
(gst_multifdsink_remove), (gst_multifdsink_clear),
(gst_multifdsink_remove_client_link),
(gst_multifdsink_handle_client_read),
(gst_multifdsink_client_queue_data),
(gst_multifdsink_client_queue_buffer),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients),
(gst_multifdsink_chain), (gst_multifdsink_close):
* gst/tcp/gstmultifdsink.h:
Added more debugging info. Changed the way clients are
removed from the lists. Fixed a bug where a bad file descriptor
could cause many clients to be removed.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
(gst_multifdsink_add), (gst_multifdsink_get_stats),
(gst_multifdsink_client_remove),
(gst_multifdsink_handle_client_read),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients):
Do a bit more logging, make the client_read code more robust.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
(gst_multifdsink_add), (gst_multifdsink_get_stats),
(gst_multifdsink_client_remove),
(gst_multifdsink_handle_client_read),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients):
Make sure we don't try to read more from a client that what
ioctl says us or we deadlock.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_get_capslist), (generate_capslist),
(plugin_init):
generate the list of supported caps at startup and reuse it instead
of always generating it
Original commit message from CVS:
* gst/audioscale/gstaudioscale.c:
- fix templates to only support S16, it's the only format that works
- make caps nego code use try_set_caps_nonfixed and fixation instead
of try_set_caps twice, which is not nice for autopluggers
- change rank to secondary, so autopluggers can pick it up after
audioconvert
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
(gst_multifdsink_add), (gst_multifdsink_remove),
(gst_multifdsink_clear), (gst_multifdsink_get_stats),
(gst_multifdsink_client_remove),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients):
* gst/tcp/gstmultifdsink.h:
Recover from a select with a bad file descriptor by removing
the client.
Original commit message from CVS:
* gst/tcp/gsttcpclientsrc.c (gst_tcpclientsrc_get): Make sure that
the pad is negotiated.
* gst/ffmpegcolorspace/gstffmpegcolorspace.c (gst_ffmpegcolorspace_chain): Ditto
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init): Add typefind
for ELF files, since they can easily be recognized as audio/mpeg.
(bug #147441)
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
(gst_multifdsink_add), (gst_multifdsink_get_stats),
(gst_multifdsink_client_remove),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_queue_buffer):
* gst/tcp/gstmultifdsink.h:
More multifdsink stats. Avoid deadlock by releasing locks
before sending out a signal.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
(gst_multifdsink_init), (gst_multifdsink_add),
(gst_multifdsink_client_remove),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_queue_buffer), (gst_multifdsink_chain),
(gst_multifdsink_set_property), (gst_multifdsink_get_property),
(gst_multifdsink_init_send):
* gst/tcp/gstmultifdsink.h:
Added more stats, added timeout for a client, fixed some typos
and added some comments.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
mp42/mp43 (no caps) exist too.
* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
Set pixel_width/height; we've got them in-caps.
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
* gst/wavparse/gstwavparse.c: (plugin_init):
Both are valid primary.
* sys/oss/gstossmixer.c:
Remove i18n hack and enable translations.
Original commit message from CVS:
2004-07-11 Andy Wingo <wingo@pobox.com>
* gst/audioconvert/gstaudioconvert.c (gst_audio_convert_link): For
float, "any" caps -> buffer_frames=[0,MAX].
* gst/interleave/interleave.c (interleave_getcaps): Seems the core
doesn't intersect our caps with the template any more. Do it
ourselves.
(interleave_buffered_loop): Use g_newa instead of malloc/free.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
(gst_multifdsink_class_init), (gst_multifdsink_add),
(gst_multifdsink_remove), (gst_multifdsink_clear),
(gst_multifdsink_client_remove),
(gst_multifdsink_handle_client_read),
(gst_multifdsink_client_queue_data),
(gst_multifdsink_client_queue_caps),
(gst_multifdsink_client_queue_buffer),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
(gst_multifdsink_handle_clients), (gst_multifdsink_thread),
(gst_multifdsink_init_send), (gst_multifdsink_close):
Fix wrong GList iteration that could crash the server when
more then 2 clients disconnect at the same time. Read all the
pending commands in one batch to recover from command storms under
very heavy load.
Original commit message from CVS:
* gst/tcp/gsttcpserversink.c: (gst_tcpserversink_class_init),
(gst_tcpserversink_init), (gst_tcpserversink_handle_server_read),
(gst_tcpserversink_client_remove),
(gst_tcpserversink_handle_client_read),
(gst_tcpserversink_client_queue_data),
(gst_tcpserversink_client_queue_caps),
(gst_tcpserversink_client_queue_buffer),
(gst_tcpserversink_handle_client_write),
(gst_tcpserversink_queue_buffer),
(gst_tcpserversink_handle_clients), (gst_tcpserversink_thread),
(gst_tcpserversink_chain), (gst_tcpserversink_set_property),
(gst_tcpserversink_get_property), (gst_tcpserversink_init_send),
(gst_tcpserversink_close):
* gst/tcp/gsttcpserversink.h:
Serversink rewrite. Really do non blocking writes to clients and
maintain an internal queue to handle slower clients while not
disturbing fast clients.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audiorate_link),
(gst_audiorate_init), (gst_audiorate_chain),
(gst_audiorate_set_property), (gst_audiorate_get_property):
* gst/videorate/gstvideorate.c: (gst_videorate_class_init),
(gst_videorate_chain):
Added some logging, fixed an overflow bug in videorate.
Original commit message from CVS:
* gst-libs/gst/colorbalance/Makefile.am:
* gst-libs/gst/mixer/Makefile.am:
* gst-libs/gst/play/Makefile.am:
* gst-libs/gst/tuner/Makefile.am:
* gst/tcp/Makefile.am:
* sys/dxr3/Makefile.am:
don't include -enumtypes.[ch] or -marshal.[ch] files in the disted
tarball.
Also add all *.list files that were missing.
* Makefile.am:
add a distcheck hook to ensure the above doesn't happen again.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_videorate_class_init),
(gst_videorate_init), (gst_videorate_chain),
(gst_videorate_set_property), (gst_videorate_get_property):
Add property to make videorate silent.
Add property to prefer new frames over old ones.
Original commit message from CVS:
* ext/dvdnav/gst-dvd: Grab the gconf key from the right spot
* gst/debug/gstnavseek.c: (gst_navseek_init),
(gst_navseek_segseek), (gst_navseek_handle_src_event),
(gst_navseek_chain):
* gst/debug/gstnavseek.h: Add 's', 'e' and 'l' keypresses to navseek
to define the start,end and loop parameters of a segment seek.
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_init),
(gst_videotestsrc_get_event_masks),
(gst_videotestsrc_handle_src_event), (gst_videotestsrc_get):
* gst/videotestsrc/gstvideotestsrc.h:
Add seeking support to videotestsrc
Initialise the timestamp_offset variable.
Original commit message from CVS:
* gst/ffmpegcolorspace/imgconvert.c: (img_convert):
Patch 1.3 broke the ordering of the colorspace info and
made the plugin basically work by coincidence, reodered
the info.
Original commit message from CVS:
2004-06-12 Christophe Fergeau <teuf@gnome.org>
* gst/tags/gstvorbistag.c: replaced a g_warning which I added in my
previous commit with GST_DEBUG
Original commit message from CVS:
2004-06-12 Zaheer Abbas Merali <zaheerabbas@merali.org>
* gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_init_send):
* gst/tcp/gsttcpclientsink.h:
* gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_init_receive):
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpserversink.c: (gst_tcpserversink_init),
(gst_tcpserversink_handle_server_read),
(gst_tcpserversink_init_send):
* gst/tcp/gsttcpserversink.h:
* gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_init_receive):
* gst/tcp/gsttcpserversrc.h:
Modified the tcp plugins so they are portable (IPv4,IPv6, any future
version of IP)
Original commit message from CVS:
2004-06-10 Christophe Fergeau <teuf@gnome.org>
* gst/tags/gstvorbistag.c: (gst_vorbis_tag_add): make sure parsed
vorbis comments are properly encoded in UTF-8 before adding them
to a GstTagList
Original commit message from CVS:
reviewed by Benjamin Otte <otte@gnome.org>
* gst/adder/gstadder.c: (gst_adder_loop):
properly error out when no negotiation has happened yet. (fixes
#143032)
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c: that's
G_HAVE_GNUC_VARARGS, not G_HAVE_GNU_VARARGS. Should fix compile
problems on several systems.
Original commit message from CVS:
* gst/tcp/gsttcp.c: portability (Solaris 10/FreeBSD)
* gst/tcp/gsttcpclientsrc.h: idem
- define MSG_NOSIGNAL if not done
- include unistd.h for off_t
(fixes#143749)
patch by Andrew Turner <zxombie@hotpop.com>
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
fixate nicely even when the peer is not negotiating
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps):
make sure we don't allow depth > width
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
fixate endianness to G_BYTE_ORDER as default
* gst/audioscale/gstaudioscale.c:
we don't handle another endianness as host-endianness
Original commit message from CVS:
* ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_sinkconnect),
(gst_oggvorbisenc_setup):
properly fail when we can't setup the vorbis encoder due to
unsupported settings
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_sinkconnect),
(gst_vorbisenc_setup):
same
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
fix case where warnings occured when one pad was unlinked while the
other's link function was called
Original commit message from CVS:
* gst/videoscale/videoscale.c: (gst_videoscale_scale_nearest),
(gst_videoscale_scale_nearest_str2),
(gst_videoscale_scale_nearest_str4),
(gst_videoscale_scale_nearest_32bit),
(gst_videoscale_scale_nearest_24bit),
(gst_videoscale_scale_nearest_16bit):
Fix the scaling algorithm and avoid a buffer overflow.
removed the while loop in the scaling function as it
was used for point sampling only.
Original commit message from CVS:
- change sunaudio category to Sink/Audio
- change HAVE_FIONREAD macro to GST_CHECK_FIONREAD
- add conditional include for FIONREAD ioctl on more files
