Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gen_source_element):
Throw a more comprehensible error for rtsp:// URIs (rather
than erroring out with a negotiation error later on) until
we fix playbin to handle rtspsrc etc.
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(gst_play_base_bin_init), (gst_play_base_bin_dispose),
(set_encoding_element), (decodebin_element_added_cb),
(decodebin_element_removed_cb), (setup_subtitle), (setup_source),
(gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
* gst/playback/gstplaybasebin.h:
Add 'subtitle-encoding' property to playbin, so applications can
force a subtitle encoding for non-UTF8 subtitles (#342268).
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init),
(gst_sub_parse_set_property):
Rename recently-added 'encoding' property to 'subtitle-encoding'
(so it can be proxied by playbin/decodebin in a generic way
with less danger of false positives).
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(append_with_other_format), (set_structure_widths),
(gst_audio_convert_transform_caps):
Patch from #341562: give more specific audio caps in get_caps, so
that basetransform can make better decisions on what caps to
negotiate.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_get_type):
Make it easier to copy&paste
* gst/volume/Makefile.am:
* gst/volume/gstvolume.c: (volume_update_real_volume),
(gst_volume_set_volume), (gst_volume_set_mute),
(gst_volume_class_init), (volume_process_int16), (volume_set_caps),
(volume_transform_ip), (volume_update_mute),
(volume_update_volume):
* gst/volume/gstvolume.h:
Add own debug category, move duplicate code to helper function, fix
property texts, add more comments and prepare ffor liboil-goodness
* tests/check/Makefile.am:
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
add test for mute and passtrough case, be a bit more verbose to track
failure
* tests/check/generic/states.c: (GST_START_TEST):
catch elements that fail to instantiate
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_smpfmt_to_caps):
Marking caps conversion issues as GST_WARNING is way too verbose,
Moving them to GST_LOG.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (remove_fakesink), (pad_probe):
When there is only one unfinished pad and it receives an event that
doesn't match our requirements, we need to set alldone=FALSE so that
the fakesink is not removed yet.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
Else they play perfectly fine with qtdemux.
Original commit message from CVS:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
* gst/audiorate/gstaudiorate.c:
make more debug catagories static
* tests/check/Makefile.am:
* tests/check/elements/adder.c: (message_received),
(test_event_message_received), (GST_START_TEST),
(test_play_twice_message_received), (adder_suite):
added test case for using element twice, extra bonus points for anyone
who can make these test run reliably
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/tcp/Makefile.am:
fdstresstest doesn't need Gtk+, fix compilation if
gtk is not available (#342566).
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
Use the gstutil scaling function to preserve 64 bits while calculating
output width and height from the display-aspect-ratio. (A continuation
of #341542)
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist chollian net>
* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
(gst_sub_parse_class_init), (gst_sub_parse_init),
(gst_sub_parse_set_property), (gst_sub_parse_get_property),
(convert_encoding):
* gst/subparse/gstsubparse.h:
Add 'encoding' property (#341681).
* gst/subparse/samiparse.c: (characters_sami):
Output is pango markup, so we need to escape text
between tags (#342143).
Original commit message from CVS:
2006-05-16 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* gst/tcp/gsttcp.c: (gst_tcp_socket_read):
Return GST_FLOW_UNEXPECTED when we have an eos on the socket so
basesrc can do its job correctly.
Original commit message from CVS:
2006-05-15 Julien MOUTTE <julien@moutte.net>
* gst/playback/gstdecodebin.c: (cleanup_decodebin),
(gst_decode_bin_change_state): Make decodebin reusable
when going from PAUSE_TO_READY and then back to PAUSED.
Fixes#331678.
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect):
Don't use libxml functions in the typefinding code.
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect):
Improve SAMI typefinding: handle case where there are
whitespaces or newlines in front of the first <SAMI>
tag (#169936).
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* gst/subparse/samiparse.c: (handle_start_font):
Need to map "silver" colour explicitly (#169936).
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Fix#341696: crash when mixing L+R+C to mono or stereo.
* tests/check/Makefile.am:
* tests/check/elements/audioconvert.c: (set_channel_positions),
(get_float_mc_caps), (get_int_mc_caps), (GST_START_TEST),
(audioconvert_suite):
Add test for the above, including some generic framework bits for
testing multichannel things.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
Fix the build.
Original commit message from CVS:
2006-05-11 Thomas Vander Stichele <thomas at apestaart dot org>
patch by: Sjoerd Simons (sjoerd@luon.net)
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(group_create), (group_destroy), (add_stream),
(gst_play_base_bin_get_property),
(gst_play_base_bin_get_streaminfo_value_array):
* gst/playback/gstplaybasebin.h:
API: GstPlayBaseBin::stream-info-value-array property
use a more bindings-friendly way of exposing streaminfo
using a GValueArray. Tested in ipython.
Closes#341114
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (try_to_link_1), (queue_enlarge),
(queue_underrun_cb), (queue_filled_cb):
Also catch queue underruns but don't do anything yet.
Refactor and comment queue enlarging code a bit.
* gst/playback/gstplaybasebin.c: (queue_overrun),
(queue_threshold_reached), (queue_out_of_data),
(gen_preroll_element):
If a queue over/underruns check that we don't create nasty
deadlocks when the min-threshold is not reached but the
max-bytes is. In those cases disable max-bytes when we
know that the queue is fed timed data.
Add more comments.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_audio_element):
Make playbin automatically plug an 'audioresample'
element before the audio sink as well. This solves
problems with sinks that only accept a very specific
sample rate, like esdsink (e.g. #340379).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gen_source_element):
Make http sources send special headers so that we receive
icecast metadata if the http stream is an icecast stream
(otherwise the server will just ignore them). This also
means that from now on users will need the 'icydemux'
element from gst-plugins-good installed if they want to
listen to icecast radio streams. (#341432, #333657).
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_remove_client_link),
(gst_multi_fd_sink_new_client), (gst_multi_fd_sink_stop):
remove stupid example from docs - it should come with a simple
C program instead.
Clean up/fix docs
* tests/check/elements/multifdsink.c: (wait_bytes_served),
(fail_if_can_read), (GST_START_TEST),
(gst_multifdsink_create_streamheader), (multifdsink_suite):
add a test for changing streamheader which exposes a bug in
multifdsink
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_stop):
clean up the bufqueue when shutting down
* tests/check/Makefile.am:
* tests/check/elements/multifdsink.c: (setup_multifdsink),
(cleanup_multifdsink), (GST_START_TEST), (multifdsink_suite),
(main):
add a test for the leak that was just fixed
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps),
(gst_adder_query_duration), (gst_adder_query), (forward_event),
(gst_adder_src_event), (gst_adder_sink_event),
(gst_adder_class_init), (gst_adder_finalize),
(gst_adder_request_new_pad), (gst_adder_collected):
* gst/adder/gstadder.h:
Updated some docs. Added comments and FIXMEs all over the place.
Improve debugging info.
Fix leak on finalize by not calling the parent.
Implement duration query.
Make event forwarding threadsafe.
Correctly send NEWSEGMENT at start and after flush.
Handle EOS correctly.
Post error when not negotiated.
* tests/check/elements/adder.c: (GST_START_TEST):
Added FIXME in the test.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon net>
* gst/tcp/gstmultifdsink.c: (gst_client_status_get_type):
Register nick for enum value (#341160).
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_request_new_pad),
(gst_adder_collected):
* gst/adder/gstadder.h:
Remove bogus segment merging and forwarding, we don't
care about timestamps anyway and we just produce a
continuous stream.
Also create a nice NEWSEGMENT event when we start.
Use _scale_int some more.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp4_find_box),
(mp4_type_find), (plugin_init):
Add typefind to distinguish between "audio/x-m4a" and new type
"video/mp4". Fixes#340375
* tests/check/elements/adder.c: (adder_suite):
Raise timeout to make buildbot happy
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_sink_event),
(gst_adder_request_new_pad), (gst_adder_change_state):
* gst/adder/gstadder.h:
* tests/check/Makefile.am:
* tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
(adder_suite), (main):
Add sink-event handling to adder. It tries to merge incomming
newsegment-events. Added test to check if segment_done is comming
through.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find),
(mpeg1_sys_type_find), (ogganx_type_find), (sw_data_destroy):
Rearrange MPEG system stream detection, fixing some memleaks in the
process.
Constify the data for STARTS_WITH and RIFF helper handlers. Make sure
they clean up their data correctly.
Remove unused ogganx caps and move the 'is_annodex' check to inside
the 'is_ogg' if statement.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
(mpeg_ts_probe_headers), (mpeg_ts_type_find):
When typefinding an MP3 in push-based mode, don't penalise the
probability down to 74% when we found 5 valid frames just because we
can't peek the end of the file.
Make the probability for detecting MPEG Transport Streams based on the
number of sequential headers we successfully detected.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_funcfind), (volume_set_caps),
(volume_transform_ip):
Increase "volume" property to 10.0. Fixes#340369.
Set the process function to NULL when capsnego fails so that
we properly error out.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
(plugin_init):
Refine musepack typefinding a bit. Return MAXIMUM
probability when we detect stream version 7 to make
sure the mpeg audio typefinder doesn't trump us.
