Original commit message from CVS:
* gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
(gst_multipart_mux_init), (gst_multipart_mux_loop),
(gst_multipart_mux_get_property), (gst_multipart_mux_set_property),
(gst_multipart_mux_change_state):
Added configurable boundary specifier, added the value as a
caps field as well.
Original commit message from CVS:
2004-06-01 Christophe Fergeau <teuf@gnome.org>
* ext/flac/gstflactag.c: strip ending framing bit from vorbiscomment
buffer since libflac doesn't expect it (reports a sync error when
it encounters that)
Original commit message from CVS:
* gst-libs/gst/tuner/tunerchannel.h:
- add a freq_multiplicator field to make the conversion
between internal frequency unit and Hz
* sys/v4l/gstv4lelement.c:
* sys/v4l2/gstv4l2element.c:
- change default video device to /dev/video0
* sys/v4l/v4l_calls.c:
* sys/v4l2/v4l2_calls.c:
- we only expose frequency to the user in Hz instead of
bastard v4lX unit (either 62.5kHz or 62.5Hz)
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
Initialise b_o_s and e_o_s variables
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
Add some unusual fourcc's from mplayer avi's
* gst/multipart/multipartmux.c: (gst_multipart_mux_plugin_init):
Make the muxer have rank GST_RANK_NONE, so it doesn't mess up
autoplugging.
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:
* 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/avi/gstavidemux.c: (gst_avi_demux_handle_src_query):
Fix potential division by zero error and hopefully get
the position query right to get correct timestamps on avi
audio.
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:
* ext/mad/gstid3tag.c: (gst_id3_tag_get_type),
(gst_id3_tag_class_init), (gst_id3_tag_init),
(gst_id3_tag_set_property), (gst_id3_tag_get_tag_to_render),
(gst_id3_tag_handle_event), (gst_id3_tag_do_caps_nego),
(gst_id3_tag_send_tag_event):
lots of fixes to make id3mux work and id3demux work correctly
Original commit message from CVS:
* ext/Makefile.am:
add rules to build shout2send (was removed by accident
when this module was no more marked experimental/broken)
* ext/shout2/gstshout2.c:
* ext/shout2/gstshout2.h:
adding a "connection problem" signal to shout2send
(fixes#142954)
Original commit message from CVS:
* gst/cdxaparse/gstcdxaparse.c:
* gst/cdxaparse/gstcdxaparse.h:
some renaming
add some checks/sanity
prepare for seek addition
* sys/sunaudio/gstsunaudio.c:
remove exported dupe init function
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_get_formats),
(gst_dvdec_src_convert), (gst_dvdec_sink_convert):
Fix format conversion and position querying.
* gst/debug/progressreport.c: (gst_progressreport_report):
Don't output a bogus total value that we didn't query.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
Always set XV_AUTOPAINT_COLORKEY to true. Fixes xvimagesink showing
only a blank window after xine has been used.
Original commit message from CVS:
* configure.ac: Minor cosmetic change to convince the buildbot to
reautogen.
* sys/sunaudio/gstsunaudio.c: (gst_sunaudiosink_class_init),
(gst_sunaudiosink_init), (gst_sunaudiosink_getcaps),
(gst_sunaudiosink_pad_link), (gst_sunaudiosink_chain),
(gst_sunaudiosink_setparams), (gst_sunaudiosink_open),
(gst_sunaudiosink_close), (gst_sunaudiosink_change_state),
(gst_sunaudiosink_set_property), (gst_sunaudiosink_get_property):
More hacking. Plays audio now.
Original commit message from CVS:
* sys/osxaudio/Makefile.am: New OS X audio plugin by Zaheer Merali
* sys/osxaudio/gstosxaudio.c:
* sys/osxaudio/gstosxaudioelement.c:
* sys/osxaudio/gstosxaudioelement.h:
* sys/osxaudio/gstosxaudiosink.c:
* sys/osxaudio/gstosxaudiosink.h:
* sys/osxaudio/gstosxaudiosrc.c:
* sys/osxaudio/gstosxaudiosrc.h:
Original commit message from CVS:
* configure.ac:
remove -DG_DISABLE_DEPRECATED. It's not usable without workarounds
if you want to work against glib 2.2 and 2.4
Original commit message from CVS:
* gst/debug/testplugin.c:
* gst/debug/tests.c:
* gst/debug/tests.h:
add new extensible and configurable testing element. Current tests
include buffer count, stream length, timestamp/duration matching and
md5.
* gst/debug/Makefile.am:
* gst/debug/gstdebug.c: (plugin_init):
add infrastructure for new element
Original commit message from CVS:
* gst/ffmpegcolorspace/imgconvert.c: (img_convert): Fixes for
warnings (bugs, actually) noticed by gcc but not forte.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_header): Patch from dcm@acm.org (David Moore)
to allow qtdemux to use non-seekable streams. (bug #142272)
Original commit message from CVS:
* gst-libs/gst/resample/resample.c: (gst_resample_sinc_ft_s16),
(gst_resample_sinc_ft_float): Remove use of static temporary
buffer. This code was obviously not supposed to last long, but
it's stuck in our ABI, so it required a little hack to make it
ABI-compatible. Fixes#142585.
* gst-libs/gst/resample/resample.h: same.
Original commit message from CVS:
* configure.ac: Add sunaudio
* examples/Makefile.am: make gstplay depend on gconf
* gst/ffmpegcolorspace/gstffmpegcodecmap.c: Remove c99-isms
* gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette),
(convert_table_lookup), (img_convert): remove c99-isms
* gst/ffmpegcolorspace/imgconvert_template.h: make a constant
unsigned, to fix a warning on Solaris
* gst/mpeg1sys/systems.c: bcopy->memcpy
* gst/rtjpeg/RTjpeg.c: (RTjpeg_yuvrgb8): bcopy->memcpy
* sys/Makefile.am: Add sunaudio
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_init),
(gst_ogg_mux_sinkconnect), (gst_ogg_mux_request_new_pad),
(gst_ogg_mux_next_buffer), (gst_ogg_mux_push_page),
(gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
(gst_ogg_mux_loop):
Fix an ugly memleak where the muxer didn't flush enough ogg
pages. This also resulted in badly muxed ogg files.
Original commit message from CVS:
* ext/mad/gstmad.c: (gst_mad_handle_event):
Fix for when the first format in a discont event is not a
byte-based one. Should fix#137710.
Original commit message from CVS:
* ext/shout2/gstshout2.c:
use application/ogg instead of application/x-ogg (patch by Patrick
Guimond, fixes#142432)
* sys/oss/gstosselement.c: (gst_osselement_reset),
(gst_osselement_sync_parms):
don't set fragment size unless specified
Original commit message from CVS:
* autogen.sh:
* configure.ac:
* ext/mad/gstid3tag.c: (gst_id3_tag_chain):
compute offsets correctly for internal buffers so timestamps are set
correctly when we can't seek. Also handle cases where there are no
offsets. (based on a patch by David Moore, fixes#142507)
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:
* gst/asfdemux/gstasfdemux.c:
- fix a mem leak and always propagate tags
- add WMV3 to known video codecs (but no decoder yet)
- replace "surplus data" at end of audio header for what
it is : codec specific data
- fix a typo
Original commit message from CVS:
reviewed by: Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/audio/audioclock.c:
Fix wrong return type (#142205).
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:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_comment):
- process comments even if they don't end with \0\0
g_convert would ignore them if present and works well without them
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_comment):
don't write to memory we might not write to - g_convert does that
for us anyway
(gst_asf_demux_audio_caps):
conmment out gst_util_dump_mem
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/esd/esdsink.c: (gst_esdsink_chain): Fix crash when ESD
is killed while we're playing.
* gst/qtdemux/qtdemux.c: (qtdemux_parse): call
gst_element_no_more_pads().
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c :
- fix INFO tag extraction in RIFF/AVI files
because gst_event_unref (event) also freed taglist
- avoid a mem leak
Original commit message from CVS:
* ext/mad/gstid3tag.c : move from "Codec/(Dem/M)uxer" to "Codec/(Dem/M)uxer/Audio"
* gst/wavenc/gstwavenc.c : move from "Codec/Encoder/Audio" to "Codec/Muxer/Audio"
* gst/auparse/gstauparse.c :
- add code (commented for now) to support audio/x-adpcm on src pad
(we have no decoder for those layout yet)
* gst/cdxaparse/gstcdxaparse.c :
* gst/cdxaparse/gstcdxaparse.h :
- partial rewrite using RiffRead (ripped iain's wavparse code)
* gst/rtp/gstrtpL16enc.c : typo
* gst/rtp/gstrtpgsmenc.c : typo
Original commit message from CVS:
* ext/audiofile/gstafsrc.c: (gst_afsrc_get):
Remove old debug output
* ext/dv/gstdvdec.c: (gst_dvdec_quality_get_type),
(gst_dvdec_class_init), (gst_dvdec_loop), (gst_dvdec_change_state),
(gst_dvdec_set_property), (gst_dvdec_get_property):
Change the quality setting to an enum, so it works from gst-launch
Don't renegotiate a non-linked pad. Allows audio only decoding.
* gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_getcaps),
(gst_deinterlace_link), (gst_deinterlace_init):
* gst/videodrop/gstvideodrop.c: (gst_videodrop_getcaps),
(gst_videodrop_link):
Some caps negotiation fixes
Original commit message from CVS:
* ext/tarkin/gsttarkin.c :
- Change RANK from NONE to PRIMARY (decoder)
* ext/gdk_pixbuf/gstgdkpixbuf.c :
- Change RANK from NONE to MARGINAL (decoder)
* ext/divx/gstdivxenc.c :
- Change RANK from PRIMARY to NONE (encoder/spider issue)
Original commit message from CVS:
* configure.ac:
enable shout2 by default
* ext/shout2/gstshout2.c: (gst_shout2send_protocol_get_type),
(gst_shout2send_base_init), (gst_shout2send_init),
(gst_shout2send_connect), (gst_shout2send_change_state):
* ext/shout2/gstshout2.h:
make this work again. Based on a patch by Zaheer Merali (fixes
#142262)
* ext/theora/theora.c: (plugin_init):
don't set rank on encoders
Original commit message from CVS:
* gst/auparse/gstauparse.c :
- Document all audio encoding we can encounter from Solaris 9
headers and libsndfile information.
- Increase max. rate from 48000 to 192000 (to match other elements)
- Don't try to play junk data between header and samples
Original commit message from CVS:
* gst/cdxaparse/gstcdxaparse.c :
Add mpegversion to CAPS to make it link
Rank is as GST_RANK_SECONDARY instead of NONE
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_getcaps):
use the right caps depending on endianness (I hope)
* ext/ogg/gstoggmux.c: (gst_ogg_mux_plugin_init):
use GST_RANK_NONE for all non-decoding elements or spider gets
mighty confused
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_comment):
Fix some odd cases and fix BE metadata parsing of unicode16 text.
Original commit message from CVS:
* gst/switch/gstswitch.c: (gst_switch_release_pad),
(gst_switch_request_new_pad), (gst_switch_poll_sinkpads),
(gst_switch_loop), (gst_switch_get_type):
whoever that was: DO NOT IMPORT PRIVATE SYMBOLS THAT ARE NOT IN
HEADERS. Had to be said.
Original commit message from CVS:
* gst/auparse/gstauparse.c: (gst_auparse_class_init),
(gst_auparse_init), (gst_auparse_chain),
(gst_auparse_change_state):
Hack around spider. Remove me some day please.
