Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximage_buffer_free), (gst_ximagesink_ximage_put),
(gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
Avoid type checking in buffer casts.
Avoid caps copy in buffer_alloc when we can.
Use pad_peer_accept.
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:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
(gst_text_overlay_video_chain):
g_markup_escape_text() REALLY doesn't like non-UTF8 input
and doesn't validate its input either (and neither did
textoverlay it seems). Let's do that then and fix#345206.
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:
* tests/check/elements/audioresample.c: (test_reuse),
(audioresample_suite):
Add test case for bug #342789 fixed below.
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:
Patch by: Young-Ho Cha <ganadist at chollian dot net>
* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
Parse extra data better, apparently it's right behind
the normal strf header size. Fixes#343500.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams):
If we fail to set the buffer_time and period_time alsa
parameters, post a warning and leave alsa select a
default instead of failing. Fixes#342085
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/cdda/gstcddabasesrc.h:
Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
out in the header file and shouldn't be listed in the docs.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
Fix it so that it doesn't crash in the debug statement.
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
add remaining symbols into correct setions
* gst-libs/gst/audio/gstringbuffer.c:
fix incomplete docs
* gst-libs/gst/audio/gstringbuffer.h:
comment out not yet implemented function
* gst-libs/gst/floatcast/floatcast.h:
* gst-libs/gst/netbuffer/gstnetbuffer.c:
add short descriptions
* gst-libs/gst/interfaces/propertyprobe.c:
fix return value docs
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
simplify debug logging
* gst-libs/gst/riff/riff-read.h:
sync function prototype and docs
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
remove left over symbol
Original commit message from CVS:
* autogen.sh:
* configure.ac:
* docs/Makefile.am:
Use GST_PLUGIN_DOCS macro in configure.ac, add
--enable-plugin-docs default to autogen.sh and use
ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
(gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
(gst_ogg_demux_loop):
Combine GstFlowReturn from the source pads to give a
meaningfull result to the upstream peer or to stop the
processing task in case of errors.
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:
* ext/ogg/Makefile.am:
* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
Extract language tags from ogm subtitle streams, so that
the subtitle menu choices are labelled correctly in
Totem (fixes#344708).
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:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
first batch of reordering things, add index & hierarchy
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
Add support for burn:// URIs (#343385); const-ify things a bit,
use G_N_ELEMENTS instead of hard-coded array size.
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:
* ext/theora/theoraparse.c: (theora_parse_drain_queue_prematurely),
(theora_parse_drain_queue):
Mark DELTA_UNIT on non-keyframes.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
(gst_ring_buffer_samples_done):
* gst-libs/gst/audio/gstringbuffer.h:
Document better the fact that latency_time and buffer_time are values
stored in microseconds, and not the usual GStreamer nanoseconds.
Change the variables (compatibly) that store them from GstClockTime
to guint64 to make it more clear that they're not storing clock times.
Also, remove the bogus property description that says the user can
specify -1 to get the default value, since that's never been the case.
When computing the default segment size for the ring buffer, make it
an integer number of samples.
When the sub-class indicates a delay greater than the number of
samples we've written return 0 from the audio sink get_time method.
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (set_channel_positions),
(get_float_mc_caps), (get_int_mc_caps):
* tests/check/elements/audioresample.c:
* tests/check/elements/audiotestsrc.c: (GST_START_TEST):
* tests/check/elements/videorate.c:
* tests/check/elements/videotestsrc.c: (GST_START_TEST):
* tests/check/elements/volume.c:
* tests/check/elements/vorbisdec.c:
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
Don't busy-wait in tests; this was causing test timeouts very
frequently when running under valgrind.
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:
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
Reinstate limit on channel count. Vorbis does not define the meaning
of > 6 channels, so they're just independent channels. Gstreamer
currently has no mechanism to represent N independent channels.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
Don't arbitrarily restrict channel counts and rate in vorbis.
In terms of effects likely on real-world files, this fixes 96kHz
playback of vorbis.
