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:
* Trying to correct the GST_CHECK_CONFIGPROG macro. It works perfectly for
I hope i dont break anything.
* Shifting the MAS plugin back from gst-rotten.
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
