Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_samples_to_timestamp):
cast to GstClockTime to get higher granularity
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
use gst_element_set_time_delay to get the exact time
* ext/mad/gstmad.c: (gst_mad_chain):
use the negotiated rate instead of the current frame's rate which
might be wrong because of bit errors. This avoids emitting totally
bogus timestamps and screwing sync.
(fixes#143454)
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_probe_hw_params),
(gst_alsa_set_hw_params), (gst_alsa_set_sw_params),
(gst_alsa_start_audio):
- don't call set_periods_integer anymore, it breaks the
configuration randomly
- call snd_pcm_hw_params_set_access directly instead of using masks
- don't fail if the sw_params can't be set, just use the default
params and hope it works. Alsalib has weird issues when you touch
sw_params and does no proper error reporting about what failed.
* ext/alsa/gstalsa.c: (gst_alsa_open_audio),
(gst_alsa_close_audio):
make our alsa debugging go via gst debugging and not conditionally
defined
* ext/alsa/gstalsa.h:
add ALSA_DEBUG_FLUSH macro
* ext/alsa/gstalsaplugin.c: (gst_alsa_error_wrapper),
(plugin_init):
wrap alsa errors to be printed via the gst debugging system and not
spammed to stderr
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_build_list):
Select first track as master track. Not sure how else to handle
that...
* ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer):
Discard discont events. Should fix#142962.
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:
* ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
use correct variable when determining amount of data to skip so we
don't skip into the void and segfault
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
(gst_alsa_mixer_close), (gst_alsa_mixer_supported),
(gst_alsa_mixer_build_list), (gst_alsa_mixer_free_list),
(gst_alsa_mixer_change_state), (gst_alsa_mixer_list_tracks),
(gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
(gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record):
Fix for cases where we fail to attach to a mixer.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
compute correct expected timestamps after seek (broken since
last commit)
* ext/gdk_pixbuf/pixbufscale.c: (pixbufscale_init):
rename element and debugging category to gdkpixbufscale
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
add error checking to snd_pcm_delay and remove duplicate call to
snd_pcm_delay that caused issues (see inline code comments)
* ext/alsa/gstalsasink.c: (gst_alsa_sink_get_time):
make more readable and fix return value when snd_pcm_delay fails
Original commit message from CVS:
* ext/alsa/gstalsasrc.c: (gst_alsa_src_loop):
don't use a fixed buffer size when writing variable length data to
it. Fixes memory corruption and makes alsasrc work
Original commit message from CVS:
* ext/alsa/gstalsa.c: (device_list),
(gst_alsa_class_probe_devices):
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
Fix alsa oddness in mixer after the combination of using mixer
in source/sink elements and using hw:x,y instead of just hw:x.
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_build_list):
* ext/alsa/gstalsasink.c: (gst_alsa_sink_get_type),
(gst_alsa_sink_class_init):
* ext/alsa/gstalsasink.h:
* ext/alsa/gstalsasrc.c: (gst_alsa_src_get_type),
(gst_alsa_src_class_init):
* ext/alsa/gstalsasrc.h:
Make alsasink/src a subclass of alsamixer so that mixer stuff
shows up in gst-rec. Needs some finetuning.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_class_probe_devices):
Don't probe for playback device if we're a source element. Fixes
#139658.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_change_state),
(gst_alsa_close_audio):
handle case better where a soundcard can't pause
* ext/ogg/gstoggdemux.c:
don't crash when we get events but don't have pads yet
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_fixate): Don't fixate fields that
aren't in the caps.
* gst/sine/gstsinesrc.c: change rate caps to [1,MAX]
* gst/videocrop/gstvideocrop.c: (plugin_init): Change rank to NONE.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_property),
(gst_alsa_open_audio), (gst_alsa_close_audio):
* ext/alsa/gstalsa.c:
Don't open the device if we're a mixer (= padless).
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_class_init),
(gst_alsa_mixer_init), (gst_alsa_mixer_open),
(gst_alsa_mixer_close), (gst_alsa_mixer_change_state):
Open mixer during state change rather than during object
initialization. Also, get a device name. Currently in a somewhat
hackish fashion, but I didn't really find something better.
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_free_list):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init),
(gst_alsa_mixer_track_new):
* ext/alsa/gstalsamixertrack.h:
Fix ancient leftovers... MixerTrack is a GObject.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_class_probe_devices):
* sys/oss/gstosselement.c: (gst_osselement_class_probe_devices):
Don't block during probing...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_type), (gst_alsa_class_init),
(gst_alsa_get_property), (gst_alsa_probe_get_properties),
(gst_alsa_class_probe_devices), (gst_alsa_class_list_devices),
(gst_alsa_probe_probe_property), (gst_alsa_probe_needs_probe),
(gst_alsa_probe_get_values), (gst_alsa_probe_interface_init),
(gst_alsa_open_audio), (gst_alsa_close_audio):
* ext/alsa/gstalsa.h:
Add propertyprobe interface implementation, add some device-name
property, all this so that it looks good in gnome-volume-control.
Original commit message from CVS:
2004-02-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_pcm_wait),
(gst_alsa_xrun_recovery):
* ext/alsa/gstalsa.h:
try xrun recovery when wait failed. Make xrun recovery function
return TRUE/FALSE to indicate success. (might fix#134354)
Original commit message from CVS:
* ext/alsa/Makefile.am: Fix linking against libgstinterfaces.
(bug #133886) Noticed by bugs@leroutier.net (Stephane LOEUILLET)
Original commit message from CVS:
2004-02-05 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_change_state):
be sure to stop the clock when going to paused
* sys/oss/gstosssink.c: (gst_osssink_change_state):
reset number of transmitted when going to ready.
fixes#132935
2004-02-05 Charles Schmidt <cschmidt2@emich.edu>
reviewed by Benjamin Otte
* ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list):
extract track count (fixes#133410)
Original commit message from CVS:
2004-01-31 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_change_state), (gst_alsa_start),
(gst_alsa_drain_audio), (gst_alsa_stop_audio):
really start/stop clock only on PLAYING <=> PAUSED
* ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
remove \n from debugging lines
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain):
make it work when seeking does not
* ext/vorbis/vorbisdec.c: (vorbis_dec_event):
reset on DISCONT
Original commit message from CVS:
2004-01-31 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_change_state), (gst_alsa_start):
start clock on PAUSED=>PLAYING, not later
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
extract correct time for different discont formats
(gst_alsa_sink_get_time):
don't segfault when no format is negotiated yet, just return 0
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_event),
(gst_ogg_demux_handle_event), (gst_ogg_demux_push),
(gst_ogg_pad_push):
handle flush and discont events correctly
* ext/vorbis/vorbisdec.c: (vorbis_dec_event), (vorbis_dec_chain):
handle discont events correctly
Original commit message from CVS:
2004-01-28 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_query_func):
use gst_element_get_time to get correct time
Original commit message from CVS:
2004-01-15 Julien MOUTTE <julien@moutte.net>
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_interface_init): Setting
mixer interface type to HARDWARE.
* gst-libs/gst/mixer/mixer.c: (gst_mixer_class_init): Adding a default
type to SOFTWARE.
* gst-libs/gst/mixer/mixer.h: Adding mixer interface type and macro.
* gst-libs/gst/mixer/mixertrack.h: Adding mixertrack flag SOFTWARE.
* gst/volume/gstvolume.c: (gst_volume_interface_supported),
(gst_volume_interface_init), (gst_volume_list_tracks),
(gst_volume_set_volume), (gst_volume_get_volume),
(gst_volume_set_mute), (gst_volume_mixer_init),
(gst_volume_dispose), (gst_volume_get_type), (volume_class_init),
(volume_init): Implementing mixer interface.
* gst/volume/gstvolume.h: Adding tracklist for mixer interface.
* sys/oss/gstosselement.c: (gst_osselement_get_type),
(gst_osselement_change_state): Removing some trailing commas in
structures.
* sys/oss/gstossmixer.c: (gst_ossmixer_interface_init): Setting mixer
interface type to HARDWARE.
* sys/v4l/gstv4lcolorbalance.c:
(gst_v4l_color_balance_interface_init): Setting colorbalance interface
type to HARDWARE.
* sys/v4l2/gstv4l2colorbalance.c:
(gst_v4l2_color_balance_interface_init): Setting colorbalance
interface type to HARDWARE.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain): use exactly the
same code than ximagesink for event handling.
Original commit message from CVS:
2004-01-15 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
Don't update the time of the clock
(gst_alsa_sink_loop):
sync to the clock given to alsasink, not the own clock
* sys/oss/gstosssink.c: (gst_osssink_chain):
sync to the clock
(gst_osssink_change_state):
activate the clock
* sys/ximage/ximagesink.c: (gst_ximagesink_chain):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain):
remove bogus code that made DISCONT events unhandled
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_video_caps):
explicitly case to double in _set_simple. (fixes 2nd warning in bug
#131502)
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_read_object_header),
(gst_asf_demux_handle_sink_event), (gst_asf_demux_audio_caps),
(gst_asf_demux_add_audio_stream), (gst_asf_demux_video_caps):
convert g_warning because of wrong asf data to GST_WARNINGs (fixes
2nd warning in bug #131502)
Original commit message from CVS:
Remove all usage of gst_pad_get_caps(), and replace it with
gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap().
Original commit message from CVS:
2004-01-03 Thomas Canty <tommydal@optushome.com.au>
reviewed by: Ronald Bultje <rbultje@ronald.bitfreak.net>
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_dispose):
Correct logic of dispose function (see #129306).
Original commit message from CVS:
* ext/alsa/gstalsasrc.c: (gst_alsa_src_pad_factory),
(gst_alsa_src_base_init): Remove bogus "src" request pad.
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_base_init),
(gst_mpeg_parse_class_init): Move pad template registration
to class_init, since the derived class (mpegdemux) doesn't
want them.
Original commit message from CVS:
2003-12-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_open_audio):
Don't send ALSA debugging to stderr.
* ext/alsa/gstalsa.h:
Use GST_WARNING instead of g_warning when ALSA functions fail.
Original commit message from CVS:
2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_get_caps), (gst_alsa_link):
Fix remaining caps handling errors due to CAPS merge.
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:
next big bunch of stuff:
- proper caps setting in alsasrc
- query / conversion functions
WARNING: Alsa 0.9.2 is heavily borked wrt recording - expect segfaults
Original commit message from CVS:
bugfixes:
- better error reporting
- segfault when using alsasrc without alsasink (d'oh)
- don't try to round when doing samples => time conversion
Original commit message from CVS:
total code reorganization as a start to get alsasrc working - sink and src are now really different classes, not just on paper - includes a fix that makes the testsuite work that might be an older bug
Original commit message from CVS:
fix clock - seeking, xruns etc should be handled correctly now
includes bugfix to not play the rest of the audio buffer when going PAUSED => READY
