Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_sinkconnect),
(gst_faad_chain), (gst_faad_change_state):
* ext/faad/gstfaad.h:
Allow playback of raw (unframed) MPEG AAC files (#148993).
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
Throw error if we didn't recognize the stream. Fixes#152289.
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
Fix memleak.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_audio_caps_with_data):
Add codec_data handling (like asfdemux used to do).
* gst/asfdemux/gstasf.c: (plugin_init):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
(gst_asf_demux_add_audio_stream), (gst_asf_demux_add_video_stream):
Use riff-media for caps creation instead of our own (mostly
broken) copy of its functions.
Original commit message from CVS:
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_try_capture):
Don't actually error out if we get another return value than
-EINVAL. Opposite to what I first thought, drivers have random
return values for this, although -EINVAL is the expected return
value. Since this is not fatal, we shouldn't use
GST_ELEMENT_ERROR() but just GST_ERROR_OBJECT().
Original commit message from CVS:
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
(dvdreadsrc_init), (dvdreadsrc_dispose), (dvdreadsrc_set_property),
(dvdreadsrc_get_property), (_open), (_seek), (_read),
(dvdreadsrc_get), (dvdreadsrc_open_file),
(dvdreadsrc_change_state):
Fix. Don't do one big huge loop around the whole DVD, that will
cache all data and thus eat sizeof(dvd) (several GB) before we
see something.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Actually NULL'ify event after using it.
* gst/matroska/ebml-read.c: (gst_ebml_read_use_event),
(gst_ebml_read_handle_event), (gst_ebml_read_element_id),
(gst_ebml_read_element_length), (gst_ebml_read_element_data),
(gst_ebml_read_seek), (gst_ebml_read_skip):
Handle events.
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_base_init),
(gst_dvd_demux_init), (gst_dvd_demux_get_audio_stream),
(gst_dvd_demux_get_subpicture_stream), (gst_dvd_demux_plugin_init):
Fix timing (this will probably break if I seek using menus, but
I didn't get there yet). VOBs and normal DVDs should now work.
Add a mpeg2-only pad with high rank so this get autoplugged for
MPEG-2 movies.
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_base_init),
(gst_mpeg_demux_class_init), (gst_mpeg_demux_init),
(gst_mpeg_demux_new_output_pad), (gst_mpeg_demux_get_video_stream),
(gst_mpeg_demux_get_audio_stream),
(gst_mpeg_demux_get_private_stream), (gst_mpeg_demux_parse_packet),
(gst_mpeg_demux_parse_pes), (gst_mpeg_demux_plugin_init):
Use this as second rank for MPEG-1 and MPEG-2. Still use this for
MPEG-1 but use dvddemux for MPEG-2.
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_class_init),
(gst_mpeg_parse_init), (gst_mpeg_parse_new_pad),
(gst_mpeg_parse_parse_packhead):
Timing. Only add pad template if it exists. Add sink template from
class and not from ourselves. This means we will always use the
correct sink template even if it is not the one defined in this
file.
Original commit message from CVS:
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_packet),
(gst_mpeg_demux_parse_pes):
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_parse_packhead):
Fix playback of mpeg again, timestamps where screwed up by
patch 1.61.
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:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_add),
(gst_multifdsink_remove), (gst_multifdsink_remove_client_link),
(is_sync_frame), (gst_multifdsink_client_queue_buffer),
(gst_multifdsink_new_client),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
(gst_multifdsink_handle_clients):
* gst/tcp/gstmultifdsink.h:
Make syncing to keyframes actually work for new clients and lagging
clients.
Original commit message from CVS:
Company's wisdom:
Events should be passed on using the sinkpad's default handler not the src
Seek events only go upstream, so send a discont downstream instead.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (new_decoded_pad):
Only signal the no_more_pads signal when we have
added the stream to our list.
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:
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
For completeness, XSync in the destroy function as xvimage does.
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:
* ext/jpeg/gstjpegdec.c (gst_jpegdec_chain): Allocate the buffer
after setting caps. Fixes mysterious segfault. Blessed by Wim.
Original commit message from CVS:
2004-09-19 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* gst/wavenc/gstwavenc.c: (gst_wavenc_init), (gst_wavenc_chain):
* gst/wavenc/gstwavenc.h:
Added newmedia support to wavenc
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:
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_get_audio_stream):
Caps are only set if the type of the stream is unknown, but this
is initialized in ->init_stream(), so set to UNKNOWN after calling
->init_stream() so that capsnego starts.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_stream_data):
Just hardcode for raw audio then. AVI audio sucks.
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/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_add_stream), (gst_avi_demux_stream_data):
* gst/avi/gstavidemux.h:
Fix for compressed audio (mp3) timestamp generation. How did this
ever work?
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:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer),
(gst_ogg_mux_send_headers), (gst_ogg_mux_loop):
Fix the case where the muxer would mark pages as delta
frames when they are not (vorbis only ogg).
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:
2004-09-14 Brian Cameron <brian.cameron@sun.com
* tools/gst-launch-ext-m.m: Changed ~ to $ENV{HOME} to allow
this script to work on Solaris since bash shell handles echo
differenly than bash.
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/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
(qtdemux_parse_trak):
Don't crash by dividing by zero (see sample movie in #126922).
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/gnomevfs/Makefile.am:
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_get_type),
(gst_gnomevfssink_dispose), (gst_gnomevfssink_init),
(gst_gnomevfssink_uri_get_type),
(gst_gnomevfssink_uri_get_protocols),
(gst_gnomevfssink_uri_get_uri), (gst_gnomevfssink_uri_set_uri),
(gst_gnomevfssink_uri_handler_init),
(gst_gnomevfssink_set_property), (gst_gnomevfssink_get_property),
(gst_gnomevfssink_open_file), (gst_gnomevfssink_close_file):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get_type),
(gst_gnomevfssrc_init), (gst_gnomevfssrc_dispose),
(gst_gnomevfssrc_uri_get_type),
(gst_gnomevfssrc_uri_get_protocols), (gst_gnomevfssrc_uri_get_uri),
(gst_gnomevfssrc_uri_set_uri), (gst_gnomevfssrc_uri_handler_init),
(gst_gnomevfssrc_set_property), (gst_gnomevfssrc_get_property),
(gst_gnomevfssrc_open_file), (gst_gnomevfssrc_close_file):
* ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
* ext/gnomevfs/gstgnomevfsuri.h:
Add URI support to Gnome-VFS plugins. Tries to load a fixed list
of fake URIs to see which this version of Gnome-VFS likes, and
uses that for the Gst-URI interface. Makes playbin support http://
streams. Also fix up some stupid behaviour in gnomevfssrc.
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/videomixer/videomixer.c: (gst_videomixer_blend_buffers):
Copy timestamps from the master pad to the output buffers.
Original commit message from CVS:
Write track and segment UIDs, write muxing date, write TRACKDEFAULTDURATION for TTA audio, write BLOCKDURATION if known.
Original commit message from CVS:
* sys/glsink/glimagesink.c: (gst_glimagesink_ximage_put),
(gst_glimagesink_xwindow_new), (gst_glimagesink_xcontext_get),
(gst_glimagesink_fixate): Move local variable declarations to
make gcc-2.95 happy.
Original commit message from CVS:
Fix byte order reversion for writing ebml floats.
Write segment duration and muxing application in matroska.
Added TTA codec to the list of supported codecs to mux into matroska.
Original commit message from CVS:
2004-08-26 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* ext/lame/gstlame.c: (gst_lame_init), (gst_lame_chain):
* ext/lame/gstlame.h:
Added new media support to lame
Original commit message from CVS:
Interpret BLOCKDURATION and set buffer duration accordingly, enable demuxing
of TTA audio from matroska, fixes bugs #148950 and #148951.
Original commit message from CVS:
2004-08-24 Sebastien Cote <sc5@hermes.usherb.ca>
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head),
(gst_riff_read_element_data), (gst_riff_read_seek),
(gst_riff_read_skip): fix infinite loop in wavparse, fixes bug
#144616, patch reviewed by Ronald and committed by Christophe Fergeau
<teuf@gnome.org>
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer),
(gst_ogg_mux_send_headers), (gst_ogg_mux_loop):
Make sure we never send -1 granulepos.
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer),
(gst_ogg_mux_loop):
I will accept bitchslappings with non sharp objects.
Original commit message from CVS:
2004-08-19 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* configure.ac:
* ext/lame/Makefile.am:
* ext/lame/gstlame.c: (gst_lame_class_init),
(gst_lame_set_property), (gst_lame_get_property), (gst_lame_setup):
Only enable lame presets if version of lame has presets in API
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_get):
* gst/udp/gstudpsrc.h:
Don't call gst_pad_push in a get function. Fixes#150449
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:
* sys/sunaudio/Makefile.am:
* sys/sunaudio/gstsunaudio.c: Fix caps to handle full range
of rates and channels. Make debugging less obnoxious.
Patch from Balamurali Viswanathan implementing a mixer for
Sun audio. (bug #144091):
* sys/sunaudio/gstsunelement.c:
* sys/sunaudio/gstsunelement.h:
* sys/sunaudio/gstsunmixer.c:
* sys/sunaudio/gstsunmixer.h:
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
