Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init), (gst_auto_audio_sink_class_init),
(gst_auto_audio_sink_find_best):
* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_detect):
Small cleanups.
don't try to set "sync" property when it is not available.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/alpha/gstalpha.c:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtsp/gstrtspsrc.c:
* gst/udp/gstudpsrc.c:
* gst/videomixer/videomixer.c:
Include stdlib.h in some more places, makes things compile
with uClibc and -Werror (#357592).
Original commit message from CVS:
* ext/jpeg/gstjpegdec.c:
Set minimum height to 8 (from 16), our code should handle
that fine. Some of the buttons on the apple trailer site
are apparently only 15 pixels high (see #357470).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index):
Don't check for a tag that is never there and check if we read the
correct tag. Fixes seeking again.
We must post an error when all pads are unlinked.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
(gst_rtp_vorbis_pay_reset_packet),
(gst_rtp_vorbis_pay_init_packet),
(gst_rtp_vorbis_pay_flush_packet), (gst_rtp_vorbis_pay_parse_id),
(gst_rtp_vorbis_pay_handle_buffer):
More fixage, set endoder-params correctly in the payloader.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_base_init):
Make static pad templates static to appease valgrind's leak
detector.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/autodetect.c: (GST_START_TEST),
(autodetect_suite):
Add simple test for the ghostpad lockup on shutdown fixed in core
CVS (audio bit disabled because it would need dozens of alsa
suppressions and I'm too lazy to add those now).
Original commit message from CVS:
* gst/rtp/README:
Update README with some examples.
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_init),
(gst_rtp_mp4g_pay_finalize), (gst_rtp_mp4g_pay_parse_audio_config),
(gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps),
(gst_rtp_mp4g_pay_setcaps):
* gst/rtp/gstrtpmp4gpay.h:
Make optional RTP parameters of type STRING, as required by the
application/x-rtp caps specification.
Original commit message from CVS:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
Correctly calculate size of each H263+ RTP buffer taking into account MTU and
RTP header.
Original commit message from CVS:
* gst/rtsp/URLS:
Added some test URLS.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
When creating streams, give access to the complete SDP.
Fix some leaks.
Collect and merge global stream properties in stream caps.
Preliminary support for WMServer.
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (rtsp_connection_read), (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspconnection.h:
Make connection interruptable.
Refactor to make it reconnectable.
Don't fail on short reads when reading data packets.
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port),
(rtsp_url_get_port):
* gst/rtsp/rtspurl.h:
Add methods for getting/setting the port.
* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
(sdp_message_get_attribute_val), (sdp_media_get_attribute),
(sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val),
(sdp_media_get_format), (sdp_parse_line),
(sdp_message_parse_buffer):
Fix headers.
Add methods for getting multiple attributes with the same name.
Increase buffer size when parsing.
Fix parsing of a=foo fields.
* gst/rtsp/test.c: (main):
Update to new connection API.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_free):
* gst/rtsp/rtsptransport.h:
* gst/rtsp/sdp.h:
* gst/rtsp/sdpmessage.h:
* gst/rtsp/gstrtsp.c:
* gst/rtsp/gstrtsp.h:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/rtsp.h:
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Dual licensed under MIT and LGPL now.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (find_stream_by_pt),
(gst_rtspsrc_create_stream), (gst_rtspsrc_free_stream),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
(gst_rtspsrc_push_event), (gst_rtspsrc_loop), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
Reorganize stream parsing and creation.
Detect container formats in interleaved mode.
Keep more state about the streams.
Assume a server also supports PLAY if it does not say.
Add unicast and interleaved properties to TCP transport requests to make
some servers happy (WMServer).
* gst/rtsp/sdpmessage.h:
Add some defines for the standard Bandwidth types.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_base_init),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
Export sometimes source pad with correct caps on the template, create
the ghostpad from the template.
Remove RTCP template as we never expose RTCP.
Protect against invalid body size.
Avoid memcpy when creating the output buffer.
Properly post an error and send EOS when the loop function is shut down.
Original commit message from CVS:
Based on patch by: Lutz Mueller <lutz at topfrose dot de>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_set_property), (gst_rtspsrc_open),
(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
Make sure we can never set an invalid location.
* gst/rtsp/rtspmessage.c: (rtsp_message_steal_body):
* gst/rtsp/rtspmessage.h:
Added _steal_body method for future use.
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free):
Make freeing of NULL url return immediatly.
Original commit message from CVS:
Based on patch by: Lutz Mueller <lutz at topfrose dot de>
* gst/rtsp/gstrtspsrc.c: (_do_init), (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Use boilerplate.
Make rtspsrc subclass GstBin to make state changes easier.
Add Range header field on the PLAY request.
Original commit message from CVS:
Based on patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause):
* gst/rtsp/rtspconnection.c: (inet_aton):
Small cleanups.
when multicast is selected as the transport, create UDP sources and
connect to the multicast group.
Move parsing and setting of caps to a common place.
Fixes#349894.
Original commit message from CVS:
Patch by: Yves Lefebvre <ivanohe at abacom dot com>
* gst/avi/gstavimux.c: (gst_avi_mux_stop_file):
Correctly set the dwLength in strh.
With this patch, the file duration is now displayed correctly in window
media player and the AVI plays completely. Fixes#356147
Original commit message from CVS:
Patch by: Darren Kenny <darren dot kenny at sun dot com>
* sys/sunaudio/gstsunaudiomixerctrl.c:
(gst_sunaudiomixer_ctrl_build_list):
Set the output track as the MASTER so that the gnome-settings-daemon
keybindings for changing the volume using the keyboard works.
Fixes#356142.
Original commit message from CVS:
* gst/multipart/multipartdemux.c: (gst_multipart_demux_chain):
Fix documentation, it is not possible to control the framerate of jpegdec
using filtered caps yet. Fixes#355210.
Return the downstream GstFlowReturn instead of GST_FLOW_OK so that we
stop when there is an error.
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag):
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
Don't interpret a first buffer with an offset of NONE as
'from the middle of the stream', but only a first buffer
that has a valid buffer offset that's non-zero (see #345449).
Original commit message from CVS:
* gst/icydemux/gsticydemux.c: (gst_icydemux_reset),
(gst_icydemux_typefind_or_forward):
* gst/icydemux/gsticydemux.h:
When we merge/collect multiple incoming buffers for typefinding
purposes, keep an initial 0 offset on the first outgoing buffer
as well (otherwise id3demux won't work right). Fixes#345449.
Also Make buffer metadata writable before setting buffer caps.
* tests/check/elements/icydemux.c: (typefind_succeed),
(cleanup_icydemux), (push_data), (GST_START_TEST),
(icydemux_suite):
Small test case for the above.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_peek_chunk),
(gst_avi_demux_stream_index), (gst_avi_demux_sync),
(gst_avi_demux_stream_header_push),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_loop):
More code reuse and better logging in _peek_chunk(). Reintroduce check
for chunk sizes before reading them (avoid oom). Better handling for
invalid chunksizes when streaming.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_set_property):
* gst/level/gstlevel.h:
Fix type mixup in level->interval (gdouble<->guint64). Spotted by
René Stadler
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
(gst_avi_demux_sync), (gst_avi_demux_stream_header_push),
(gst_avi_demux_stream_data):
Revert one change to fix streaming avi (adapter size != data size).
Original commit message from CVS:
Patch by: Frédéric Riss <frederic.riss at gmail dot com>
* gst/matroska/matroska-demux.c: (gst_matroska_track_free),
(gst_matroska_demux_reset),
(gst_matroska_demux_read_track_encodings),
(gst_matroska_demux_add_stream), (gst_matroska_decode_buffer),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_subtitle_caps):
* gst/matroska/matroska-ids.h:
Add support for VOBSUB subtitle tracks and zlib-compressed
tracks. Make sure we start on a keyframe after a seek. (#343348)
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_push_hdr_buf),
(gst_matroska_demux_push_flac_codec_priv_data),
(gst_matroska_demux_push_xiph_codec_priv_data),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
* gst/matroska/matroska-ids.h:
Add basic FLAC support (#311586), not perfect yet though, needs some
tweaking in flacdec; also, seeking could be better.
Do better bounds checking when deserialising vorbis stream headers
to make sure we don't read beyond the end of the buffer on bad input.
Original commit message from CVS:
Patch by: Alessandro Decina <alessandro at nnva dot org>
* ext/annodex/gstcmmldec.c: (gst_cmml_dec_chain):
Seeking back in a file containing a CMML stream errors out if the seek
goes back up to the CMML headers. This is because after the seek the xml
processing instruction <?xml ...?> is submitted to the xml parser again,
which results in an error. The attached patch fixes the problem.
Fixes#353908.
* ext/annodex/gstcmmlenc.h:
Fix authors name.
Original commit message from CVS:
2006-08-28 Andy Wingo <wingo@pobox.com>
* ext/raw1394/gstdv1394src.c (gst_dv1394src_from_raw1394handle):
New helper function to lessen the ifdefs.
(GST_INFO_OBJECT):
(gst_dv1394src_iso_receive): Use it.
(gst_dv1394src_create): Also use the control sockets in iec61883
mode.
(gst_dv1394src_start, gst_dv1394src_stop): Always use a separate
handle for AVC operations; fixes#348233.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_init), (gst_avi_demux_finalize),
(gst_avi_demux_reset), (gst_avi_demux_index_last),
(gst_avi_demux_index_next), (gst_avi_demux_index_entry_for_time),
(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_index),
(gst_avi_demux_stream_index), (gst_avi_demux_peek_tag),
(gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_stream_header_pull), (gst_avi_demux_do_seek),
(gst_avi_demux_process_next_entry), (gst_avi_demux_loop),
(gst_avi_demux_chain), (gst_avi_demux_sink_activate),
(gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
More attempts to turn this into readable code.
Don't leak adapters.
Calculate duration according to index more efficiently.
Don't try to act like we drive the pipeline in chain mode.
Original commit message from CVS:
Patch by: Alessandro Decina <alessandro at nnva dot org>
* ext/annodex/gstannodex.c: (gst_annodex_granule_to_time):
Do some extra sanity checks.
Fixes#350340.
* ext/annodex/gstcmmlenc.c: (gst_cmml_enc_change_state),
(gst_cmml_enc_parse_tag_head), (gst_cmml_enc_parse_tag_clip),
(gst_cmml_enc_push_clip), (gst_cmml_enc_push):
Check if clip->start_time is valid before adding the clip to the
track list.
Reset enc->preamble going from PAUSED to READY.
Don't use GST_FLOW_UNEXPECTED for wrong usage of the element, it is
only used for EOS.
Only post an error message if we were the one that created the fatal
GstFlowReturn value.
* ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt),
(gst_cmml_clock_time_to_granule), (gst_cmml_track_list_has_clip):
Parse the seconds field of the npt-sec time format using %llu rather than
%d and check that the value scaled by GST_SECOND doesn't overflow.
Use guint64(s) to represent the keyindex and keyoffset fields of a granulepos.
Lookup a clip's track with clip->track rather than clip->id which
makes no sense.
Identify a clip by its track and start time and not its xml id.
do some more input checking and make sure we don't do undefined shifts.
* tests/check/elements/cmmldec.c: (setup_cmmldec),
(teardown_cmmldec), (check_output_buffer_is_equal), (push_data),
(cmml_tag_message_pop), (check_headers), (push_clip_full),
(push_clip), (push_empty_clip), (check_output_clip),
(GST_START_TEST), (cmmldec_suite):
* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
(teardown_cmmlenc), (check_output_buffer_is_equal), (push_data),
(check_headers), (push_clip), (check_clip_times), (check_clip),
(check_empty_clip), (GST_START_TEST), (cmmlenc_suite):
Added some more checks.
Original commit message from CVS:
* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_class_init),
(gst_audio_panorama_set_property),
(gst_audio_panorama_get_property),
(gst_audio_panorama_transform_m2s_int),
(gst_audio_panorama_transform_s2s_int),
(gst_audio_panorama_transform_m2s_float),
(gst_audio_panorama_transform_s2s_float):
* gst/audiofxgood/audiopanorama.h:
* tests/check/elements/audiopanorama.c: (GST_START_TEST):
Make also the pan-property float (saves scaling and yields better
resolution)
Original commit message from CVS:
* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_set_caps),
(gst_audio_panorama_transform_m2s_float),
(gst_audio_panorama_transform_s2s_float):
ChangeLog surgery to add cymax's real name
Original commit message from CVS:
* gst/audiofxgood/audiopanorama.c:
(gst_audio_panorama_transform_m2s):
Fix docs & debug category. Add Fixme for volume pan levels.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
(gst_avi_demux_sync), (gst_avi_demux_stream_header_push),
(gst_avi_demux_stream_header_pull),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_chain):
unbreak AVI index handling, some more debug, remove an obsolete
adapter_flush that caused streaming to wander off in the wild
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_parse_superindex), (gst_avi_demux_parse_subindex),
(gst_avi_demux_parse_stream), (gst_avi_demux_parse_odml),
(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_stream_header_push),
(gst_avi_demux_stream_header_pull):
* gst/avi/gstavidemux.h:
Some more cleanups.
Fix totalFrames parsing in ODML.
Disable use of index for length calculation in case of ODML as this is
broken now.
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
There is no taglibmux element ...
* gst/rtsp/gstrtspsrc.c:
Use '%' rather than '&perc;' in gtk-doc blurb, docs build
was complaining about unknown entity here.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_do_seek), (gst_avi_demux_handle_seek),
(gst_avi_demux_process_next_entry):
* gst/avi/gstavidemux.h:
Mark DISCONT.
Remove old unused fields and reorder the struct a bit.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_index_entry_for_time),
(gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
(gst_avi_demux_stream_init), (gst_avi_demux_parse_stream),
(gst_avi_demux_stream_index), (gst_avi_demux_peek_tag),
(gst_avi_demux_next_data_buffer),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_stream_header), (gst_avi_demux_do_seek),
(gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow),
(gst_avi_demux_process_next_entry), (gst_avi_demux_loop),
(gst_avi_demux_sink_activate_pull), (gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
Precalc most of the duration query for each stream.
Make seeking more correct.
Use GstSegment to track position and duration.
Code cleanups and leak fixes.
Calculate correct total duration based on index length.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_text_identification_frame),
(parse_insert_string_field):
If strings in text fields are marked ISO8859-1, but contain
valid UTF-8 already, then handle them as UTF-8 and ignore
the encoding. (#351794)
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_dec_scan_got_frame),
(gst_flac_dec_write), (gst_flac_dec_loop),
(gst_flac_dec_sink_event), (gst_flac_dec_chain),
(gst_flac_dec_src_query):
* ext/flac/gstflacdec.h:
Make flac-in-ogg work (#352100).
Original commit message from CVS:
* gst/monoscope/gstmonoscope.c: (gst_monoscope_chain):
Don't unref buffers of which we've already given away
ownership to the adapter.
Original commit message from CVS:
* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_comments):
Make metadata extraction actually work.
* ext/speex/gstspeexenc.c: (gst_speexenc_base_init),
(gst_speexenc_init), (gst_speexenc_create_metadata_buffer),
(gst_speexenc_chain):
Fix metadata writing: replace old code which wrote completely
broken tags with libgsttag-based code. Plus miscellaneous
code cleanups (use static pad templates etc.) and a bunch
of leak fixes.
Original commit message from CVS:
* gst/audiopanorama/.cvsignore:
* gst/audiopanorama/Makefile.am:
* gst/audiopanorama/audiofx.c:
* gst/audiopanorama/audiopanorama.c:
* gst/audiopanorama/audiopanorama.h:
die! die! die! you should never have been there
Original commit message from CVS:
* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_get_unit_size):
* gst/videobox/gstvideobox.c: (gst_video_box_get_unit_size):
use g_assert in _get_unit_size
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/inspect/plugin-audiofxgood.xml:
cleanup -unused.txt to make it useful, add previously missing docs
* ext/Makefile.am:
* ext/esd/esdmon.c:
* ext/esd/esdsink.c:
* ext/esd/gstesd.c: (plugin_init):
reflow to get rid of two external symbols
* gst/audiofxgood/audiofx.c: (plugin_init):
re-add
Original commit message from CVS:
* ext/dv/gstdvdemux.c: (gst_dvdemux_handle_pull_seek),
(gst_dvdemux_loop), (gst_dvdemux_change_state):
* ext/dv/gstdvdemux.h:
When handling seek requests, don't send the newsegment event from the
calling thread. Instead save it so it can be sent from the streaming
thread.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/multipart/multipartdemux.c: (multipart_parse_header):
Accept leading whitespace before the boundary
This patch makes the demuxer allow some whitespace before the actual
boundary. This makes the demuxer work with the ``old'' gstreamer
multipartmuxer again (which placed an extra \n before the start
of the stream) Fixes#349068.
Original commit message from CVS:
* ext/ladspa/gstladspa.c: (gst_ladspa_base_init):
Convert ' ' into '_'. Try to keep as many characters in the padtemplate
names as possible.
Original commit message from CVS:
* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_flush),
(gst_signal_processor_do_pushes):
A push() gives away our refcount so we should not use the buffer on the
pen anymore.
Original commit message from CVS:
* configure.ac:
Require CVS of GStreamer core and -base (for
GST_TAG_EXTENDED_COMMENT and gst_tag_parse_extended_comment()).
* ext/taglib/gstid3v2mux.cc:
Write extended comment tags properly (#348762).
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_comment_frame):
Extract COMM frames into extended comments, which makes it
easier to properly retain the description bit of the tag
and maintain this information when re-tagging (#348762).
Original commit message from CVS:
* tests/check/Makefile.am:
Don't try to run annodex unit tests if the annodex
plugin has not been built (Fixes#351116).
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_find_best):
When we can't find a usable audiosink, don't error out,
but use a fake sink instead and post a warning message
on the bus (#341278).
Original commit message from CVS:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpmp4gdepay.c:
Caps extra properties must be defined as strings for
depayloaders because they are generated from an SDP.
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_base_init),
(gst_rtp_h264_depay_class_init), (gst_rtp_h264_depay_init),
(gst_rtp_h264_depay_finalize), (decode_base64),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
(gst_rtp_h264_depay_set_property),
(gst_rtp_h264_depay_get_property),
(gst_rtp_h264_depay_change_state),
(gst_rtp_h264_depay_plugin_init):
* gst/rtp/gstrtph264depay.h:
Added basic, not completely functional RFC 3984 H264 depayloader.
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx net>
* gconf/Makefile.am:
Make --disable-schemas work right (they still need
to be copied to the installation directory, just not
applied). Fixes#351347 (also #344100).
Original commit message from CVS:
Patch by: Edward Hervey <edward@fluendo.com>
* configure.ac:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
(gst_wavparse_stream_data):
Send the newsegment event in the streaming thread.
Fixes#347529
Original commit message from CVS:
* ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_setcaps),
(gst_smokeenc_resync), (gst_smokeenc_chain):
Refuse sink caps in the encoder if width or height is not a
multiple of 16, the encoder does not support that yet; along the
same lines, check the return value of the encoder setup function;
also remove some debug log clutter.
Original commit message from CVS:
2006-08-04 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstsignalprocessor.h: Add infrastructure for storing
whether a processor can work in place or not, and for keeping
track of its state. Change the FlowReturn instance variable from
"state" to "flow_state", all callers changed.
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setup)
(gst_signal_processor_start, gst_signal_processor_stop)
(gst_signal_processor_cleanup): New functions to manage the
processor's state.
(gst_signal_processor_setcaps): start() as well as setup() here.
(gst_signal_processor_prepare): Respect CAN_PROCESS_IN_PLACE.
(gst_signal_processor_change_state): Stop and cleanup the
processor as we go to NULL.
* ext/ladspa/gstladspa.c (gst_ladspa_base_init): Reuse buffers if
INPLACE_BROKEN is not set.
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_prepare):
Do the alloc_buffer in bytes, not frames.
Original commit message from CVS:
2006-08-04 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* sys/ximage/ximageutil.c: (ximageutil_xcontext_get):
Fix rgb masks when recording in < 24bpp.
Original commit message from CVS:
2006-08-04 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setcaps)
(gst_signal_processor_prepare)
(gst_signal_processor_update_inputs)
(gst_signal_processor_process, gst_signal_processor_pen_buffer)
(gst_signal_processor_flush)
(gst_signal_processor_sink_activate_push)
(gst_signal_processor_src_activate_pull)
(gst_signal_processor_change_state): Remove the last of the code
that assumes that we process whole buffers at a time. Fix some
debugging. Seems to work now in some cases.
Original commit message from CVS:
2006-08-01 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_process):
Fix nframes-choosing.
(gst_signal_processor_init): Init pending_in and pending_out.
Original commit message from CVS:
2006-08-01 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_init): No
more default sample rate, although we never check that the sample
rate actually gets set. Something for the future.
(gst_signal_processor_setcaps): Some refcount fixes, flow fixes.
(gst_signal_processor_event): Refcount fixen.
(gst_signal_processor_process): Pull the number of frames to
process from the sizes of the buffers in the input pens.
(gst_signal_processor_pen_buffer): Remove an incorrect FIXME :)
(gst_signal_processor_do_pulls): Add an nframes argument, and use
it instead of buffer_frames.
(gst_signal_processor_getrange): Refcount fixen, pass nframes on
to do_pulls.
(gst_signal_processor_chain)
(gst_signal_processor_sink_activate_push)
(gst_signal_processor_src_activate_pull): Refcount fixen.
* ext/ladspa/gstsignalprocessor.h: No more buffer_frames, yay.
Original commit message from CVS:
* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setcaps),
(gst_signal_processor_process):
don't query buffer-frames from caps, add lots of debug-log,
try fix for assert (#349189)
Original commit message from CVS:
2006-07-29 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps),
(gst_smokeenc_setcaps), (gst_smokeenc_chain):
Set caps on buffer correctly. Fixes bug #349155.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init),
(gst_multipart_demux_class_init), (gst_multipart_demux_init),
(gst_multipart_demux_finalize), (get_line_end),
(multipart_parse_header), (multipart_find_boundary),
(gst_multipart_demux_chain), (gst_multipart_demux_change_state),
(gst_multipart_set_property), (gst_multipart_get_property):
Uses GstAdapter instead of own buffering.
Actually parses the mime-type correctly (In tests the mime-type was
always "" with the old version).
Uses the Content-length header if available to speed up things.
Reliably autoscans the boundary name by default.
Fixes#349068.
* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
Don't start the stream with a \n.
Original commit message from CVS:
Patch by: Brian Cameron <brian dot cameron at sun com>
* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
Open source with O_NONBLOCK (#349015).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_massage_index):
* gst/avi/gstavidemux.h:
Whitespace fixes and more debug
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_create_element_with_pretty_name),
(gst_auto_audio_sink_find_best),
(gst_auto_audio_sink_change_state):
Get rid of old and unused magic sound-server properties stuff.
Add suffix to child sink's name that makes it easy to see from
the name alone which type it actually is (alsa, oss, esd, etc.).
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
(gst_udpsrc_set_property), (gst_udpsrc_get_property),
(gst_udpsrc_start):
* gst/udp/gstudpsrc.h:
Rename "buffer" to "buffer-size" to make clear it is a size we set and
not some sort of feature we enable.
Original commit message from CVS:
* ext/taglib/gstid3v2mux.cc:
Fix writing of comment frames (should be COMM not TCOM),
is still sub-optimal though, since we don't retain or
extract the comment descriptions properly (#334375,
also see #334375).
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
#define 'fact' RIFF chunk if we are not compiling against
-base CVS (we don't want to depend on -base CVS for this
one define only, and also not for release order reasons).
Original commit message from CVS:
* ext/taglib/gstid3v2mux.cc:
Handle multiple tags of the same type properly. Re-inject
unparsed ID3v2 frames that we get as binary blobs from
id3demux into the tag again so we don't lose information
when retagging (#334375).
Original commit message from CVS:
* sys/ximage/gstximagesrc.c: (gst_ximage_src_class_init):
Document newly-added properties properly, so that there is a
'Since: 0.10.4' in the plugin docs. Convert some property
names into canonical GObject style (GObject will do that
internally anyway).
Original commit message from CVS:
* gst/id3demux/id3tags.c:
(id3demux_add_id3v2_frame_blob_to_taglist):
Extract frames for ID3v2 versions prior to ID3v2.3.0 properly as
well, and add the version to the blob's buffer caps, since that
information will be needed for deserialisation later on (#348644).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes),
(gst_avi_demux_parse_stream):
Moved win32 variant of GST_DEBUG_CATEGORY_EXTERN to gstinfo.h. Fixed
indentation and spacing.
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_open),
(gst_esdsink_factory_init):
Prevent libesd from auto-spawning a sound daemon if it
is not already running. Now that we don't do evil stuff
like that any longer we can give esdsink a rank so that
autoaudiosink will try it as well if all other audio
sinks fail (#343051).
Original commit message from CVS:
* ext/esd/README:
Remove, it contains nothing useful anyway.
* ext/esd/esdsink.c: (gst_esdsink_init), (gst_esdsink_prepare),
(gst_esdsink_delay):
Some small clean-ups; use GST_BOILERPLATE etc.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
(gst_wavparse_other), (gst_wavparse_perform_seek),
(gst_wavparse_get_upstream_size), (gst_wavparse_stream_headers),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_pad_query):
* gst/wavparse/gstwavparse.h:
Use information from 'fact' chunk for length calculation of compressed
samples. Calculate bps if bogus value is found in wav header (embeded
mp2/mp3).
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (plugin_init):
* gst/id3demux/id3tags.c:
(id3demux_add_id3v2_frame_blob_to_taglist):
* gst/id3demux/id3tags.h:
On second thought, it might be wiser and more efficient
not to do tag registration from a streaming thread.
Original commit message from CVS:
* gst/id3demux/id3tags.c:
(id3demux_add_id3v2_frame_blob_to_taglist),
(id3demux_id3v2_frames_to_tag_list):
Put ID3v2 frames we can't parse as binary blobs into private
tags, so that they are not lost when retagging, at least once
id3v2mux has been taught to re-inject those frames again.
See bug #334375.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_process_next_entry):
Fix some leaks.
* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
Don't use \n in debug lines.
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
Add annodex and icydemux, cleanup the sections a bit
Original commit message from CVS:
Patch by: Alex Lancaster <alexl at users sourceforge net>
* ext/taglib/gstid3v2mux.cc:
Write GST_TAG_ENCODER and GST_TAG_ENCODER_VERSION as
ID3v2 TSSE frames (#347898).
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps):
Respect mpegversion for "video/mpeg" and give message in case of
unhandled versions.
Original commit message from CVS:
* ext/libpng/gstpngdec.c: (gst_pngdec_init), (buffer_clip),
(gst_pngdec_caps_create_and_set), (gst_pngdec_task),
(gst_pngdec_chain), (gst_pngdec_sink_event),
(gst_pngdec_libpng_init), (gst_pngdec_change_state),
(gst_pngdec_sink_activate_push):
* ext/libpng/gstpngdec.h:
Use statically allocated segment instead of leaking.
Various cleanups.
Fix flush and seek handling.
Original commit message from CVS:
2006-07-14 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get),
(gst_ximage_src_get_caps), (gst_ximage_src_class_init):
Fix segfault when moving mouse pointer to the bottom right corner.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_stream_header), (push_tag_lists):
* gst/avi/gstavidemux.h:
Don't push tag events found by gst_riff_parse_info() before outputting
GST_EVENT_NEWSEGMENT.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/rtspconnection.c: (rtsp_connection_send),
(rtsp_connection_close):
* gst/rtsp/rtspdefs.h:
replaced closesocket and close in code with one CLOSE_SOCKET.
Some more cleanups. Fixes#345301.
Original commit message from CVS:
Patch by: Rob Taylor <robtaylor at floopily dot org>
* gst/udp/gstmultiudpsink.c: (join_multicast),
(gst_multiudpsink_init_send), (gst_multiudpsink_add):
If a destination is added before the stream is set to PAUSED, the
multicast group is not joined as the socket is not created yet.
Also TTL and LOOP should also be set. Fixes#346921.
Original commit message from CVS:
2006-07-09 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get),
(gst_ximage_src_set_property), (gst_ximage_src_get_property),
(gst_ximage_src_get_caps), (gst_ximage_src_class_init),
(gst_ximage_src_init):
* sys/ximage/gstximagesrc.h:
Fix use-damage property to actually work :)
Add startx, starty, endx, endy properties so screencasts other than full
screen ones can work.
Original commit message from CVS:
2006-07-08 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get),
(gst_ximage_src_set_property), (gst_ximage_src_get_property),
(gst_ximage_src_class_init), (gst_ximage_src_init):
* sys/ximage/gstximagesrc.h:
Add use_damage property to offer ability to choose whether to use
XDamage or not.
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_trim_buffer),
(gst_tag_demux_read_range):
* gst/id3demux/gstid3demux.c: (gst_id3demux_trim_buffer),
(gst_id3demux_read_range):
Don't return FLOW_UNEXPECTED when a buffer is before
the start of the stream (which might happen with
large ID3v2 tags if the tag reading was done pullrange
based and we then switched to push mode later on).
Fixes regression introduced by commit from June 29th.
Original commit message from CVS:
* ext/taglib/gstid3v2mux.cc:
Make UTF-8 the default encoding when writing string
tags (before, our UTF-8 strings would automatically
be converted to ISO-8859-1 by taglib and written as
ISO-8859-1 fields if that was possible).
* tests/check/elements/id3v2mux.c: (utf8_string_in_buf),
(test_taglib_id3mux_check_tag_buffer), (identity_cb),
(test_taglib_id3mux_with_tags):
Add test case that makes sure our UTF-8 strings have
actually been written into the tag as UTF-8.
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain):
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
Return FLOW_UNEXPECTED when at the end of the file, not
FLOW_ERROR. Fixes 'internal stream error' errors that
would sometimes occur in totem when scrubbing to the
end of an ID3v1 tagged mp3 file.
Original commit message from CVS:
* ext/libpng/gstpngdec.c: (gst_pngdec_init), (user_info_callback),
(buffer_clip), (user_end_callback), (gst_pngdec_chain),
(gst_pngdec_sink_event), (gst_pngdec_change_state):
* ext/libpng/gstpngdec.h:
Implement buffer clipping/dropping using GstSegment.
This provides accurate seeking.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_read_subindexes), (gst_avi_demux_parse_stream),
(gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow),
(gst_avi_demux_process_next_entry), (push_tag_lists),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Proper aggregation of each stream's GstFlowReturn in order to figure out
whether the task should stop or not.
Don't send inline events before pushing out a NEW_SEGMENT, more
specifically for GST_TAG_EVENT.
Change a GST_ERROR to a GST_WARNING for a non-fatal situation in reading
sub-indexes.
Original commit message from CVS:
Patch by: Brian Cameron <brian dot cameron at sun dot com>
* sys/sunaudio/gstsunaudiomixerctrl.c:
(gst_sunaudiomixer_ctrl_build_list):
Move "Monitor" slider to input tab so it works more like
sdtaudiocontrol, which is what people on Solaris are used
to using for their mixer program (#346259).
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
(gst_matroska_demux_send_event),
(gst_matroska_demux_loop_stream_parse_id):
* gst/matroska/matroska-ids.h:
Send tag event after newsegment event.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_trim_buffer),
(gst_id3demux_read_range):
Make sure we don't return GST_FLOW_OK with a NULL buffer in
certain cases where a read beyond the end of the file is
requested. Fixes#345930.
* gst/apetag/gsttagdemux.c: (gst_tag_demux_trim_buffer),
(gst_tag_demux_read_range):
Fix same issue here as well.
Original commit message from CVS:
2006-06-29 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get):
Fix hypothetical crash.
Original commit message from CVS:
Patch by: Brian Cameron <brian dot cameron at sun dot com>
* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_prepare):
Do not modify the ports value. If the user has turned off the
built-in speakers, then we should not reset it in the prepare
function, since this causes the built-in speakers to turn
back on anytime the user changes a track in totem, rhythmbox,
etc. (#346066).
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_check_subtitle_buffer),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_subtitle_caps):
* gst/matroska/matroska-ids.c:
(gst_matroska_track_init_subtitle_context):
* gst/matroska/matroska-ids.h:
Try to fix up broken matroska files containing subtitle
streams with non-UTF8 character encodings (courtesy of
mkvmerge) using either the encoding specified in the
GST_SUBTITLE_ENCODING environment variable or the
current locale's character set if it is non-UTF8.
Fixes#337076.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_picture_frame):
Set image type from APIC frame as "image-type" field
of GST_TAG_IMAGE buffer caps (#344605).
Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send),
(rtsp_connection_close), (rtsp_connection_free):
Use better G_OS_* macros. Fixes#345301 some more.
Original commit message from CVS:
Patch by: Brian Cameron <brian dot cameron at sun dot com>
* sys/sunaudio/Makefile.am:
* sys/sunaudio/gstsunaudio.c: (plugin_init):
* sys/sunaudio/gstsunaudiomixerctrl.c:
(gst_sunaudiomixer_ctrl_build_list), (gst_sunaudiomixer_ctrl_new),
(gst_sunaudiomixer_ctrl_list_tracks),
(gst_sunaudiomixer_ctrl_get_volume),
(gst_sunaudiomixer_ctrl_set_volume),
(gst_sunaudiomixer_ctrl_set_mute),
(gst_sunaudiomixer_ctrl_set_record):
* sys/sunaudio/gstsunaudiomixerctrl.h:
* sys/sunaudio/gstsunaudiomixertrack.c:
(gst_sunaudiomixer_track_init), (gst_sunaudiomixer_track_new):
* sys/sunaudio/gstsunaudiomixertrack.h:
* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose),
(gst_sunaudiosrc_base_init), (gst_sunaudiosrc_class_init),
(gst_sunaudiosrc_init), (gst_sunaudiosrc_set_property),
(gst_sunaudiosrc_get_property), (gst_sunaudiosrc_getcaps),
(gst_sunaudiosrc_open), (gst_sunaudiosrc_close),
(gst_sunaudiosrc_prepare), (gst_sunaudiosrc_unprepare),
(gst_sunaudiosrc_read), (gst_sunaudiosrc_delay),
(gst_sunaudiosrc_reset):
* sys/sunaudio/gstsunaudiosrc.h:
Add a SunAudio source plugin.
Support stereo and right/left channel gain in the mixer plugin.
Support the RECORD flag so that you can switch between line-input and
microphone in gnome-volume-control.
Code cleanups like using an enumerator for track number instead of an
integer. Fixes#344923.
Original commit message from CVS:
Patch by: Joni Valtanen <joni dot valtanen at movial dot fi>
* gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send),
(rtsp_connection_close):
Make RTSP plugin compile on windows. Fixes#345301.
Some changes to original patch to catch errors better.
use ifdef WIN32 instead of ifndef.
Original commit message from CVS:
2006-06-19 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* configure.ac:
If we have libraw1394 >= 1.2.1, then we need libiec61883.
Original commit message from CVS:
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
After a failed buffer alloc, we need to abort the jpeg decoding (it
started when parsing headers to figure out how many bytes we need
to request downstream).
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek):
Make sure we don't read beyond the end of the file (#345232).
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
No language specified means the implied language is English
according to the matroska spec (partially fixes#344708);
add some more debug output.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_peek_chunk_info),
(gst_wavparse_peek_chunk), (gst_wavparse_stream_headers),
(gst_wavparse_chain):
When operating chain-based, don't make any assumptions about the
chunking of the incoming data and make streaming work on days other
than the second Thursday after a full moon. Also fix up debug
messages here and there and make use of the most excellent new
gst_pad_query_peer_duration() utility function.
Skip any 'bext' chunks in front of the 'fmt ' chunk. Fixes#343837.
* gst/wavparse/gstwavparse.h:
Remove trailing comma after last enum value, some compilers don't
like that.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_seek):
Prevent out of bounds array access when scrubbing towards
the end of the file between the last index entry and the
end. Fixes occasional 'start <= stop' newsegment event
assertions when scrubbing in MJPEG files.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
(scan_encoded_string), (parse_picture_frame):
Extract images from ID3v2 tags (APIC frames). Fixes#339704.
* configure.ac:
Require core >= 0.10.8 (for GST_TAG_IMAGE and
GST_TAG_PPEVIEW_IMAGE used in the patch above).
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_get_upstream_size):
* gst/id3demux/gstid3demux.c: (id3demux_get_upstream_size):
Use gst_pad_query_peer_duration() utility function here.
Original commit message from CVS:
* tests/examples/level/Makefile.am:
Add -lm to LIBS for pow() function, don't assume one of our
dependencies (such as libxml-2.0) drags it in automatically
(#343603).
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis dot com>
* configure.ac:
We should use $SED and not $(SED) in configure.ac (#343678).
Original commit message from CVS:
Patch by: Brian Cameron <brian dot cameron at sun dot com>
* sys/sunaudio/gstsunaudiomixerctrl.c:
(gst_sunaudiomixer_ctrl_open), (gst_sunaudiomixer_ctrl_build_list),
(gst_sunaudiomixer_ctrl_new), (gst_sunaudiomixer_ctrl_set_volume),
(gst_sunaudiomixer_ctrl_set_mute):
* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_class_init),
(gst_sunaudiosink_init), (gst_sunaudiosink_prepare),
(gst_sunaudiosink_write):
Attached find a patch that fixes a number of bugs with the SunAudio mixer
plugin and fixes#344101:
1. The gst_sunaudiomixer_ctrl_build_list kept appending the same 3 tracks onto
the tracklist causing gnome-volume-control's preferences dialog to be messed
up and would core dump if you checked/unchecked any item.
2. We weren't previously setting the MUTE flag properly. Fixing this makes
gnome-volume-control work better.
3. Now we properly define the input track to be GST_MIXER_TRACK_INPUT and
the monitor to be GST_MIXER_TRACK_OUTPUT, so that makes gnome-volume-control
look better.
Also some minor cleanup in gstsunaudiosink.c.
Original commit message from CVS:
2006-06-07 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* configure.ac:
We now require libraw1394 >= 1.1.0 and that version onwards all
have .pc files.
Original commit message from CVS:
2006-05-31 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* ext/raw1394/gstdv1394src.c: (gst_dv1394src_bus_reset):
Fix bus reset when using libiec61883
Original commit message from CVS:
* gst/avi/gstavidemux.c:
add an explicit dll imported declaration for GST_CAT_EVENT+WIN32
* win32/MANIFEST:
sort file listing
* win32/vs6/libgstavi.dsp:
add gstavimux.c to the project
* win32/vs6/libgstid3demux.dsp:
add link to zlib library
* win32/vs6/libgstmatroska.dsp:
add matroska-ids.c to the project
Original commit message from CVS:
* gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps):
* gst/debug/negotiation.c: (gst_negotiation_update_caps):
* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps):
GST_PTR_FORMAT should be used to print caps in debug statements.
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo at ubuntu dot com>
* gst/apetag/gstapedemux.c: (ape_demux_get_gst_tag_from_tag),
(ape_demux_parse_tags):
Some clean-ups and additions: map APE 'file' tag to
GST_TAG_LOCATION (#343123); add support for extracting
the track count and clean up parsing a bit (#343127).
