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.
Original commit message from CVS:
* gst/auparse/gstauparse.c: (gst_auparse_chain):
Fix for some uninitialized variables in previous patch, also
makes it work. Fixes#142286 while we're at it.
Original commit message from CVS:
* gst/auparse/gstauparse.c:
fixes a-law, adds mu-law, linear pcm (8,16,24,32), ieee (32, 64)
only unsupported formats are ADPCM/CCITT G.72x
reviewed by Ronald
* gst-libs/gst/audio/audio.h:
adds 24bit depth to PCM (x-raw-int)
Original commit message from CVS:
* ext/vorbis/Makefile.am:
* ext/vorbis/README:
* ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_get_formats),
(oggvorbisenc_get_type), (vorbis_caps_factory), (raw_caps_factory),
(gst_oggvorbisenc_base_init), (gst_oggvorbisenc_class_init),
(gst_oggvorbisenc_sinkconnect), (gst_oggvorbisenc_convert_src),
(gst_oggvorbisenc_convert_sink),
(gst_oggvorbisenc_get_query_types), (gst_oggvorbisenc_src_query),
(gst_oggvorbisenc_init), (gst_oggvorbisenc_get_tag_value),
(gst_oggvorbisenc_metadata_set1), (gst_oggvorbisenc_set_metadata),
(get_constraints_string), (update_start_message),
(gst_oggvorbisenc_setup), (gst_oggvorbisenc_write_page),
(gst_oggvorbisenc_chain), (gst_oggvorbisenc_get_property),
(gst_oggvorbisenc_set_property), (gst_oggvorbisenc_change_state):
* ext/vorbis/oggvorbisenc.h:
* ext/vorbis/vorbis.c: (plugin_init):
* ext/vorbis/vorbisenc.c: (vorbis_caps_factory),
(raw_caps_factory), (gst_vorbisenc_class_init),
(gst_vorbisenc_init), (gst_vorbisenc_setup),
(gst_vorbisenc_push_packet), (gst_vorbisenc_chain),
(gst_vorbisenc_get_property), (gst_vorbisenc_set_property):
* ext/vorbis/vorbisenc.h:
Added a raw vorbis encoder to be used with the oggmuxer.
We still need the old encoder for some gnome applications,
read the README to find out how that works.
The raw encoder is called "rawvorbisenc" until 0.9.
Original commit message from CVS:
* ext/theora/theoraenc.c: (gst_theora_enc_class_init),
(theora_enc_sink_link), (theora_push_packet), (theora_enc_chain),
(theora_enc_change_state), (theora_enc_set_property),
(theora_enc_get_property):
Mark the last packet with an EOS flag which is not really needed
in gstreamer.
Do some better video framerate initialisation.
Update the buffer timestamp.
Original commit message from CVS:
* ext/a52dec/gsta52dec.c : mark audio/a52, audio/ac3 as deprecated in a comment
* gst/ac3parse/gstac3parse.c : audio/ac3 => audio/x-ac3
* gst/realmedia/rmdemux.c : audio/a52 => audio/x-ac3
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
