Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_open),
(gst_alsasink_reset):
Also release lock when we get an error in _reset();
fix an error message.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
(gst_alsasink_init), (get_channel_free_structure),
(caps_add_channel_configuration), (gst_alsasink_getcaps),
(gst_alsasink_close):
* ext/alsa/gstalsasink.h:
Add support for more than 2 channels (#326720).
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
(gst_alsasink_class_init), (gst_alsasink_init),
(gst_alsasink_write), (gst_alsasink_reset):
* ext/alsa/gstalsasink.h:
Add lock to protect alsa calls.
Implement reset to flush samples ASAP, does not work
with dmix though.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_init), (set_hwparams),
(set_swparams), (gst_alsasink_prepare), (gst_alsasink_unprepare),
(gst_alsasink_close), (gst_alsasink_write), (gst_alsasink_reset):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (set_hwparams),
(set_swparams), (gst_alsasrc_open), (gst_alsasrc_prepare),
(gst_alsasrc_unprepare), (gst_alsasrc_read):
Update all error messages. All of them should either use
the default translated message, or actually provide a
translatable string.
Make the string for channel count problems meaningful.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
(gst_alsasink_class_init):
Free the device name string.
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
(gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad),
(gst_ogg_mux_handle_src_event), (gst_ogg_mux_clear_collectpads):
Don't remove a pad from the collectpads structure until it
is released - it's a request pad, and may receive data again
if the element gets moved back to PLAYING state.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
Ensure we turn on double buffering on the Xv port, and
set the colour key to something dark and mysterious that
isn't black.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_open):
check for ALSA errors properly, instead of relying on ALSA's
error strings to serve to the user.
Original commit message from CVS:
* ext/alsa/gstalsasink.c:
Also allow unsigned int.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Small cleanup
Original commit message from CVS:
2005-08-22 Andy Wingo <wingo@pobox.com>
* ext/alsa/gstalsasink.c (gst_alsasink_get_property):
* ext/alsa/gstalsasrc.c (gst_alsasrc_get_property): Add a
device-name property.
Original commit message from CVS:
2005-08-08 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_change_state): Open the device in NULL->READY
like good elements should. Close on READY->NULL too.
* gst-libs/gst/audio/gstaudiosink.c
(gst_audioringbuffer_open_device,
(gst_audioringbuffer_close_device, gst_audioringbuffer_acquire)
(gst_audioringbuffer_release): Updates for new ring buffer API,
hook into the new audio sink api.
* gst-libs/gst/audio/gstaudiosink.h (GstAudioSinkClass.open)
(GstAudioSinkClass.close): Just open and close the device -- no
resource allocation or configuration.
(GstAudioSinkClass.prepare, GstAudioSinkClass.unprepare): New
vmethods, handle device setup and resource allocation.
* ext/alsa/gstalsasink.c (gst_alsasink_open, gst_alsasink_close)
(gst_alsasink_prepare, gst_alsasink_unprepare): Update for new
base class API.
* gst-libs/gst/audio/gstringbuffer.h
(GstRingBufferClass.open_device, GstRingBufferClass.close_device):
New vmethods.
* gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_open_device)
(gst_ring_buffer_close_device, gst_ring_buffer_device_is_open):
New API functions. The device should be opened before acquiring
and closed after releasing.
Original commit message from CVS:
2005-07-29 Andy Wingo <wingo@pobox.com>
* ext/alsa/gstalsaplugin.c (plugin_init): We are primary audio
sinks.
* ext/alsa/gstalsasink.c (alsasink_sink_factory): Advertise our
support of both endiannesses.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_open):
Get actual segment size and buffer size after opening
the device.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_open):
Open non-blocking, set to blocking mode afterwards to avoid
lockups when audio device is busy.
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_push),
(gst_a52dec_handle_event), (gst_a52dec_chain):
Add some debug output. Check that a discont has a valid
time associated.
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event),
(gst_alsa_sink_loop):
Ignore TAG events. A little extra debug for broken timestamps.
* ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_init), (dvdnavsrc_loop),
(dvdnavsrc_change_state):
Ensure we send a discont to engage the link before we send any
other events.
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_init),
(dvdreadsrc_finalize), (_close), (_open), (_seek_title),
(_seek_chapter), (seek_sector), (dvdreadsrc_get),
(dvdreadsrc_uri_get_uri), (dvdreadsrc_uri_set_uri):
Handle URI of the form dvd://title[,chapter[,angle]]. Currently only
dvd://title works in totem because typefinding sends a seek that ends
up going back to chapter 1 regardless.
* ext/mpeg2dec/gstmpeg2dec.c:
* ext/mpeg2dec/gstmpeg2dec.h:
Output correct timestamps and handle disconts.
* ext/ogg/gstoggdemux.c: (get_relative):
Small guard against a null dereference.
* ext/pango/gsttextoverlay.c: (gst_textoverlay_finalize),
(gst_textoverlay_set_property):
Free memory when done. Don't call gst_event_filler_get_duration on
EOS events. Use GST_LOG and GST_WARNING instead of g_message and
g_warning.
* ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init),
(draw_line), (gst_smoothwave_dispose), (gst_sw_sinklink),
(gst_sw_srclink), (gst_smoothwave_chain):
Draw solid lines, prettier colours.
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init):
Add a default palette that'll work for some movies.
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_init),
(gst_dvd_demux_handle_dvd_event), (gst_dvd_demux_send_discont),
(gst_dvd_demux_send_subbuffer), (gst_dvd_demux_reset):
* gst/mpegstream/gstdvddemux.h:
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_send_discont),
(gst_mpeg_demux_parse_syshead), (gst_mpeg_demux_parse_pes):
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_init),
(gst_mpeg_parse_handle_discont), (gst_mpeg_parse_parse_packhead):
* gst/mpegstream/gstmpegparse.h:
Use PTM/NAV events when for timestamp adjustment when connected to
dvdnavsrc. Don't use many discont events where one suffices.
* gst/playback/gstplaybasebin.c: (group_destroy),
(gen_preroll_element), (gst_play_base_bin_add_element):
* gst/playback/gstplaybasebin.h:
Make sure we remove subtitles from the same bin we put them in.
* gst/subparse/gstsubparse.c: (convert_encoding), (parse_subrip),
(gst_subparse_buffer_format_autodetect),
(gst_subparse_change_state):
Fix some memleaks and invalid accesses.
* gst/typefind/gsttypefindfunctions.c: (ogganx_type_find),
(oggskel_type_find), (cmml_type_find), (plugin_init):
Some typefind functions for Annodex v3.0 files
* gst/wavparse/gstwavparse.h:
GstRiffReadClass is the correct parent class.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
Fix for integer overflow. Makes #156001 not crash. Probably masks
the real bug.
Original commit message from CVS:
2004-11-28 Martin Soto <martinsoto@users.sourceforge.net>
* ext/alsa/gstalsasink.c (gst_alsa_sink_loop):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsa.c (gst_alsa_set_clock):
Make alsasink actually honor gst_element_set_clock and use that
clock instead of ist internal one.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_pcm_wait):
add debugging
* ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
do a wait when we enter the loop func with no data available to
write instead of getting into an 100% CPU loop by just returning and
being called again by the scheduler
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
Fix for negotiation order problem. This would show when the
ALSA loopfuction was called before any other function. ALSA
wouldn't do anything because we're not negotiated yet, leading
to an infinite loop. Showed in e.g. Rhythmbox. Fixes#158006.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsa_sink_get_time):
This seems to be antique leftover. It needs to pass error
checking.
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_init),
(gst_sdlvideosink_deinitsdl), (gst_sdlvideosink_initsdl),
(gst_sdlvideosink_destroy), (gst_sdlvideosink_create),
(gst_sdlvideosink_sinkconnect), (gst_sdlvideosink_chain):
Fix GstXOverlay implementation (#151059).
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:
* ext/alsa/gstalsa.c: (gst_alsa_xrun_recovery):
use our own functions for restarting the alsa device.
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
I should apply patches myself - use MIN for the third argument, not
the second, this fixes seeking
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_change_state), (gst_alsa_start),
(gst_alsa_xrun_recovery):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event),
(gst_alsa_sink_loop), (gst_alsa_sink_get_time):
* ext/alsa/gstalsasrc.c: (gst_alsa_src_init),
(gst_alsa_src_get_time), (gst_alsa_src_update_avail),
(gst_alsa_src_loop):
Add clock to alsasrc. Take new capture timestamp when
restarting after an overrun. Split up some functions between
alsasrc ans alsasink.
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/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/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/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:
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:
2004-01-15 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)
