Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
Don't crash on truncated files - check that we got an 8 byte buffer
before trying to memcmp it.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (get_active_source):
Make stream-switching appear instant to the application
(ie. make sure that a g_object_get on 'current-foo' returns
the stream previously set with g_object_set(). Totem needs
this to update stream-related meta-info (like audio-codec)
correctly when switching streams.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
(gst_audio_convert_transform_caps):
Get structure-name just once.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c:
Add docs about icydemux usage in connection with gnomevfssrc
* ext/libvisual/visual.c:
* ext/ogg/gstoggaviparse.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst/audiorate/gstaudiorate.c:
More G_OBJECT macro fixing.
* gst/audiotestsrc/gstaudiotestsrc.h:
Fix wrong info in header due to copy & paste
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(gst_play_base_bin_init), (fill_buffer), (check_queue),
(queue_threshold_reached), (gst_play_base_bin_set_property),
(gst_play_base_bin_get_property):
* gst/playback/gstplaybasebin.h:
Don't use a 0 low watermark when buffering, it is catching starvation
way too late. Instead, use a 3 second queue with 30 and 95
percent low/high watermarks.
Added queue-min-threshold property to configure low watermark.
Use new _buffering message API.
Make queue_threshold variable big enough to store a uint64 time value.
API: playbin::queue-min-threshold property.
Original commit message from CVS:
patch by: Michael Smith <msmith at fluendo dot com>
* gst/tcp/gstmultifdsink.c: (is_sync_frame),
(gst_multi_fd_sink_client_queue_buffer),
(gst_multi_fd_sink_new_client):
* tests/check/elements/multifdsink.c: (GST_START_TEST),
(multifdsink_suite):
Fix implementation of sync-method 'next-keyframe'
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure),
(paint_get_structure), (gst_video_test_src_get_size),
(gst_video_test_src_smpte), (gst_video_test_src_snow),
(gst_video_test_src_unicolor), (paint_setup_AYUV),
(paint_hline_AYUV), (paint_setup_ARGB8888), (paint_setup_ABGR8888),
(paint_setup_RGBA8888), (paint_setup_BGRA8888), (paint_hline_str4):
* gst/videotestsrc/videotestsrc.h:
Add support for AYUV and the various RGBA formats. Initialise
fields of paintinfo structs allocated on the stack.
* tests/check/elements/videotestsrc.c: (right_shift_colour),
(fix_expected_colour), (check_rgb_buf), (got_buf_cb),
(GST_START_TEST), (videotestsrc_suite):
Add unit tests for videotestsrc's RGB output.
Original commit message from CVS:
* gst/adder/gstadder.c: (forward_event_func),
(gst_adder_src_event), (gst_adder_collected),
(gst_adder_change_state):
* gst/adder/gstadder.h:
Remember the start position asked in the incoming seeks, so we can
output GST_EVENT_NEW_SEGMENT with a correct position value (instead
of assuming it will always be 0).
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size):
Return FALSE instead of returning a random false unit
size when the format isn't known/supported (even if
this shouldn't happen under normal circumstances).
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
Don't rely on incoming buffers offset anymore, since it is completely
broken when using multiple segments.
Instead convert the incoming buffers timestamp to running time, and
then convert that value to the offsets.
Also inform GstSegment of the last outputted stop position, which is
needed if we received several segments with an unknown stop value.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
(gst_audio_rate_chain):
Make the metadata of the buffer writable before changing its
flags.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
(gst_audio_rate_setcaps), (gst_audio_rate_init),
(gst_audio_rate_sink_event), (gst_audio_rate_src_event),
(gst_audio_rate_chain), (gst_audio_rate_change_state):
Fix audiorate some more.
Reset and resync counters on flush and READY.
Handle the DISCONT flag correctly.
Use GstSegment to track position.
Fail when not negotiated.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
Small cleanups.
If a buffer is received with no caps, make the buffer metadata
writable and set the caps, making sure that we don't screw up the
refcounts.
Original commit message from CVS:
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
(gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
Fix memory leaks and misleading debug messages, add a couple of
comments.
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
(gst_multi_fd_sink_render):
Do not use gst_buffer_make_writable() in a basesink render method,
as it may incorrectly unref the buffer. Instead, use convoluted
dance to avoid copying the buffer except when we need to.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_get_unit_size), (set_structure_widths):
Lower debug, use g_assert in _get_unit_size
* gst/audioresample/gstaudioresample.c:
(audioresample_get_unit_size):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
use g_assert in _get_unit_size
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_set_property), (gst_play_bin_get_property),
(value_list_append_structure_list),
(gst_play_bin_handle_redirect_message),
(gst_play_bin_handle_message):
Add "connection-speed" property; re-order redirect messages with
multiple redirect locations depending on the minimum bitrate if
that information is available and a connection speed is set
(#350399).
Original commit message from CVS:
2006-08-11 Andy Wingo <wingo@pobox.com>
* gst/tcp/gsttcp.h: For now, always disable deprecation here --
fixes the build.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
The checks here are not even close to anything that would
justify MAXIMUM probability, lowering to POSSIBLE until someone
fixes the checks (case at hand: quicktime redirection files
might start with 00 00 01 XX and pass the checks here just
fine, see #350399).
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon net>
* gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
Better detection for multipart/x-mixed-replace: accept leading
whitespaces before the boundary marker as well (as our very own
multipartmux used to produce) (#349068).
Original commit message from CVS:
2006-08-04 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
Fix event parsing by gdpdepay. Fixes#349916.
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx net>
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinder for Interplay's MVE format (#348973).
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
(gst_gdp_depay_sink_event), (gst_gdp_depay_chain):
Consume all events except EOS because we generate events from
the gdp payload instead. Fixes#349204
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (audioresample_stop),
(audioresample_set_caps):
Don't leak references to the incoming caps. Clean them up when
stopping.
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
(gst_video_scale_finalize):
Don't leak our temporary pixel buffer.
* tests/check/Makefile.am:
* tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
(GST_START_TEST), (simple_launch_lines_suite):
Fix leaks and re-enable the test for valgrind checking.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon net>
* gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
(plugin_init):
Add typefind function for multipart/x-mixed-replace (#348916).
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps),
(gst_adder_query_duration):
Fix leak in duration query.
Reflow some docs and notes.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
(gst_multi_fd_sink_new_client):
debug a little more understandably
do not use goto as a substitute for break, especially if
break is also being used
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init):
proxying get/set caps is the wrong thing to do, since we really
do change caps quite fundamentally
* tests/check/elements/gdpdepay.c:
* tests/check/elements/gdppay.c:
remove declaration of buffers, it's already done in gstcheck.h
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (xml_check_first_element):
Limit search for the first markup tag to the first few kB of
the file. If we don't find one there, it's highly unlikely that
this is an XML(-ish) file.
Original commit message from CVS:
* gst/playback/test.c: (gen_video_element), (gen_audio_element),
(cb_newpad), (main):
Example of a small audio/video player using decodebin.
Original commit message from CVS:
2006-07-17 Thomas Vander Stichele <thomas at apestaart dot org>
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_class_init):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
remove parent_class setting, BOILERPLATE does this
(gst_gdp_pay_reset_streamheader):
fix typo in comment
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
add more plugins and elements to docs
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
fix segfaults due to wrong g_free
add example
* gst/gdp/gstgdppay.c:
add example
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (find_compatibles):
Fix a caps leak when linking (#347304)
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
(gst_ximagesink_change_state):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
(gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
Don't leak shared memory resources. Use the object lock to protect
against the xcontext disappearing while returning a buffer from the
pipeline. (#347304)
Original commit message from CVS:
* gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
(gst_ssa_parse_parse_line):
Don't include the terminating NUL in the buffer size,
it's only there for extra paranoia (would add random
'*' characters at the end of each subtitle since the
terminator itself is not valid UTF-8 technically).
Also fix indenting after boilerplate macro.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (close_pad_link):
Also emit 'unknown-type' signal (which should really be
called unhandled-type) if we found potential decoders/demuxers
in the registry but none of them worked in the end (as in the
case where the plugins don't exist any longer but are still
listed in the registry). Fixes#329798.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
(xml_check_first_element), (xml_type_find), (smil_type_find):
Fix SMIL typefinding, make xml_check_first_element() more
useful.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
(gst_play_base_bin_finalize), (decodebin_element_added_cb),
(decodebin_element_removed_cb), (gst_play_base_bin_set_property):
* gst/playback/gstplaybasebin.h:
Protect list of elements with a subtitle-encoding property and
the subtitle encoding member itself with a lock of their own
instead of using the object lock. This prevents a dead-lock in
the element-remove callback in some circumstances when shutting
down playbin.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
Improve checking if we are dealing with a stream. Added some
more uris that need buffering.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
(gst_decode_bin_init), (gst_decode_bin_finalize), (add_fakesink),
(remove_fakesink), (pad_probe), (gst_decode_bin_change_state):
Protect remove_fakesink using a mutex, so that we don't try and
remove the fakesink simultaneously from multiple threads.
When going from READY to PAUSED, restore the fakesink, so that
it is there when decodebin gets reused.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
Fix warnings with gst-inspect: "buffers-min" property
should be of G_TYPE_INT and not G_TYPE_INT64. Also fix
typo in property description.
Original commit message from CVS:
Patch by: Philip Jaegenstedt <philip at lysator dot liu dot se>
* gst/videoscale/gstvideoscale.c: (gst_video_scale_prepare_size),
(gst_video_scale_transform):
Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes#345131
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init), (gst_audioresample_init),
(audioresample_start), (audioresample_stop),
(gst_audioresample_set_property), (gst_audioresample_get_property):
Implement GstBaseTransform::start and ::stop so that audioresample
can clear its internal state properly and be reused insted of
causing non-negotiated errors with playbin under some circumstances
(#342789).
* tests/check/elements/audioresample.c: (setup_audioresample),
(cleanup_audioresample):
Need to set element state here so that ::start and ::stop are
called.
Original commit message from CVS:
* gst/playback/gststreaminfo.c: (cb_probe):
Try GST_TAG_CODEC as fallback when extracting the
codec name; more debug info.
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* gst/subparse/samiparse.c: (sami_context_pop_state),
(handle_start_font), (end_sami_element):
Honour font face tags in SAMI subtitles (#344503).
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* gst/subparse/samiparse.c: (fix_invalid_entities), (parse_sami):
Fix up broken entities before passing them to libxml *sigh*.
(#343303).
Original commit message from CVS:
* configure.ac:
enable building of GDP elements
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
(gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
(gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
(gst_gdp_pay_reset_streamheader), (gst_gdp_pay_sink_event),
(gst_gdp_pay_set_property), (gst_gdp_pay_get_property),
(gst_gdp_pay_change_state):
* gst/gdp/gstgdppay.h:
add version 1.0
Original commit message from CVS:
* gst/tcp/README:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_init),
(gst_multi_fd_sink_remove_client_link),
(gst_multi_fd_sink_client_queue_caps),
(gst_multi_fd_sink_client_queue_buffer),
(gst_multi_fd_sink_handle_client_write),
(gst_multi_fd_sink_render):
* gst/tcp/gstmultifdsink.h:
make multifdsink properly deal with streamheader:
- streamheader is taken from caps
- buffers marked with IN_CAPS are not sent
- streamheaders are sent, on connection, from the caps of the
buffer where the client gets positioned to
- further streamheader changes are done every time the client
will receive a buffer with different caps
* tests/check/elements/multifdsink.c: (GST_START_TEST),
(gst_multifdsink_create_streamheader):
add tests for this
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_create), (group_commit),
(setup_source):
* gst/playback/gstplaybasebin.h:
Make the subtitle detection work from any thread so we don't
deadlock. Fixes#343397.
Original commit message from CVS:
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
(gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
(gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
(gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain),
(gst_gdp_pay_sink_event), (gst_gdp_pay_set_property),
(gst_gdp_pay_get_property):
add crc-header and crc-payload properties
don't error out on some things that are recoverable
* tests/check/elements/gdppay.c: (GST_START_TEST), (gdppay_suite):
add test for crc
Original commit message from CVS:
* gst/volume/Makefile.am:
Seriously, it's not *that* hard to get compilation right. Even
a drunk can do it ! Add LIBOIL CFLAGS and LIBS
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_update_real_volume), (gst_volume_class_init),
(gst_volume_init), (volume_process_float), (volume_process_int16),
(volume_process_int16_clamp), (volume_set_caps),
(volume_transform_ip), (plugin_init):
* gst/volume/gstvolume.h:
rewrite the passthrough check, split _int16 and _int16_clamp, fix
another property desc., remove unused param from process function
* tests/check/elements/volume.c: (volume_suite):
reactivate the passthrough test
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.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
(gst_adder_init):
send events from src-pad to all sink-pads fixes#338657
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_remove_client_link):
* gst/tcp/gstmultifdsink.h:
Fix race condition in multifdsink that can lead to spurious
duplicate clients. this patch adds a new signal that is fired when
multifdsink has removed all references to the fd.
Fixes#339574.
Updated documentation.
API: client-fd-removed signal added
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats):
When asking g_value_array_new to prealloc elements, we may as well
ask for the right number of elements.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
(gst_video_rate_swap_prev), (gst_video_rate_chain):
fix up docs
fix a leak when no caps negotiated
fix counting of input frames
* tests/check/elements/.cvsignore:
* tests/check/elements/videorate.c: (assert_videorate_stats),
(GST_START_TEST), (videorate_suite):
add tests for these
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(resample_set_state_from_caps):
Add support for other formats audioresample can handle such as
32 bits in and float and 64 bits float. Fixes#301759
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (wavpack_type_find):
Fix wavpack typefinding to work in more cases (don't peek
for chunks of multiple hundred kBs at once, but process
things step-by-step in smaller units). Fixes#339786.
Original commit message from CVS:
Patch by: Edward Hervey <edward@fluendo.com>
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
* tests/check/Makefile.am:
* tests/check/elements/videorate.c: (assert_videorate_stats),
(setup_videorate), (cleanup_videorate), (GST_START_TEST),
(videorate_suite), (main):
Fix an infinite loop if frames are passed in with wrongly ordered
timestamps. Fixes#339013.
Original commit message from CVS:
Patch by: Tim-Philipp Müller <tim at centricular dot net>
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
fix typefinding on some ISO files. Fixes#339212.
Original commit message from CVS:
Patch by: Jan Schmidt
* gst/playback/gststreamselector.c:
(gst_stream_selector_bufferalloc):
Restore old StreamSelector behaviour.
Fixes#338419.
Original commit message from CVS:
* gst/audioresample/debug.h:
replace debug macros with variable number of parameters
by a simple alias to gstreamer standard debug macros
(#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
supported by MSVC 6.0 and 7.1)
* gst/audioresample/resample.h:
define M_PI and rint for WIN32
* win32/common/libgstaudio.def:
* win32/common/libgstriff.def:
* win32/common/libgsttag.def:
* win32/common/libgstvideo.def:
add new exported functions
* win32/vs6:
update project files
