Original commit message from CVS:
* tests/examples/seek/seek.c: (stop_seek):
When we stop scrubbing, don't leave the pipeline PLAYING when we
requested a PAUSED state.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Parse date strings in vorbis comments that have an invalid (zero)
month or day (#410396).
* tests/check/libs/tag.c: (GST_START_TEST):
Test case for the above.
Original commit message from CVS:
Patch by: Loïc Minier <lool+gnome at via ecp fr>
* configure.ac:
* ext/alsa/Makefile.am:
* gst/audiotestsrc/Makefile.am:
Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
Original commit message from CVS:
* gst-libs/gst/utils/install-plugins.c:
* gst-libs/gst/utils/missing-plugins.c:
* tests/check/libs/utils.c: (missing_msg_check_getters):
Change GStreamer marker prefix in detail string from 'gstreamer.net'
to just 'gstreamer'. Document the caps string component of the
decoder/encoder detail a bit better, since not everyone will be
familiar with the GStreamer media type/caps system (but they better
enjoy nested itemized lists).
Original commit message from CVS:
* gst-libs/gst/netbuffer/gstnetbuffer.c:
(notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
Fix copying of GstNetBuffer (would crash before, or at least lead to
invalid memory access, #410772), for now by copying the GstBuffer copy
code from the core over here so we can copy the GstBuffer fields on a
provided buffer instance (of type GstNetBuffer in this case). Would be
better to fix this with some support by the core though (and in the long
run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
* tests/check/Makefile.am:
Enable unit test for GstNetBuffer.
Original commit message from CVS:
2007-02-22 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_init): Disable pull-mode activation until we
figure out how to make audio sinks go to PLAYING.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
(gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
* gst/audioconvert/gstchannelmix.h:
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add float as an intermediate format, as well as float mixing. Enable
test that was failing before. Fixes#339837
Original commit message from CVS:
* tests/examples/seek/seek.c: (do_seek):
Undo the previous commit: -1 as a stop time implies that the stop
time is the end of file, clearing any previously configured segment.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_process_int16),
(volume_process_int16_clamp), (volume_set_caps):
Unbreak volume, value remains gint.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
(multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
(sort_end_pads), (gst_decode_group_expose),
(gst_decode_group_hide):
Don't free groups from the streaming threads. Just put them aside and
free them in dispose.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (connect_element),
(pad_added_group_cb), (gst_decode_group_check_if_blocked),
(sort_end_pads), (gst_decode_group_expose):
Handle dynamic pads within groups.
Sort pads before exposing them in order to make playbin happy.
There still is a race with the multiqueue filling up. This should be
solved separately.
Fixes#398721
Original commit message from CVS:
* gst-libs/gst/utils/base-utils.c:
* gst-libs/gst/utils/descriptions.c:
* gst-libs/gst/utils/install-plugins.c:
* gst-libs/gst/utils/missing-plugins.c:
Some more docs (and descriptions for two subtitle formats).
Original commit message from CVS:
* sys/ximage/ximagesink.c:
(gst_ximagesink_calculate_pixel_aspect_ratio):
* sys/xvimage/xvimagesink.c:
(gst_xvimagesink_calculate_pixel_aspect_ratio):
Small constifications.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_reset):
Ignore errors in reset, these are not fatal. They also grab the element
lock which is already taking when this function is called. Fixes
#405451.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c:
Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
(#403597).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source):
When we have external subtitles and wait for the subtitle decodebin
to get up and running, we set up a (sync) bus handler for the
subtitle decodebin, so we can stop waiting when it posts an error
message. However, we should do that before we set the subtitle
decodebin's state to playing, otherwise things are racy and we might
miss error messages posted before we had a chance to set up the bus.
This should finally fix totem hanging on .txt pseudo-subtitle files.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
(subrip_remove_unhandled_tags), (parse_subrip):
For SubRip (.srt) subtitles, ignore all markup tags we don't
handle (like font tags, for example).
* tests/check/elements/subparse.c:
Add test for this.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_fakesink),
(gst_decode_bin_change_state):
* gst/playback/gstdecodebin2.c: (add_fakesink),
(gst_decode_bin_change_state):
Don't error out if there is no fakesink in the READY to NULL state
change, since when decodebin is re-used, we're only adding the
fakesink element in READY to PAUSED.
* tests/check/elements/decodebin.c:
(new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
(decodebin_suite):
Minimal unit test to make sure we can use the same decodebin
instance twice (at least with audiotestsrc input).
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
Try to get devic-name from device string first, and from handle only
as fallback (seems to yield better results and is more robust
against buggy probing code on the application side).
Original commit message from CVS:
Based on patch by: Julien Puydt <julien.puydt at laposte net>
* ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
(gst_alsa_find_device_name):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
Improve device-name detection a bit, especially in the case where
the device is not actually open (#405020, #405024). Move common code
into gstalsa.c instead of duplicating it.
Original commit message from CVS:
2007-02-06 Julien MOUTTE <julien@moutte.net>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
(gst_xvimagesink_get_xv_support),
(gst_xvimagesink_xcontext_clear),
(gst_xvimagesink_interface_supported),
(gst_xvimagesink_probe_get_properties),
(gst_xvimagesink_probe_probe_property),
(gst_xvimagesink_probe_needs_probe),
(gst_xvimagesink_probe_get_values),
(gst_xvimagesink_property_probe_interface_init),
(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
(gst_xvimagesink_init), (gst_xvimagesink_class_init),
(gst_xvimagesink_get_type):
* sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
for XVAdaptors so that one can choose the adaptor to use with
gstreamer-properties.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init), (gst_audio_filter_change_state):
Clear our formats structure and free the caps contained in it when
shutting down.
Original commit message from CVS:
2007-02-05 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_callback): Update basesink->offset so that we
pull monotonically increasing offsets instead of, um, seeking back
to 0 each time. Fixes alsasrc ! alsasink!
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c:
A width and height of 1 makes us crash, so increase minimum size to
2x2 pixels until someone feels like fixing this (#404512).
Original commit message from CVS:
* tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
Add small test to make sure request pads are cleaned up properly
even if oggmux never changes state out of NULL.
Original commit message from CVS:
* tests/check/libs/utils.c: (GST_START_TEST):
Fix unit test. Turns out things work much better when you
NULL-terminate string arrays. Should make p5 build bot happy again.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
(gst_audio_filter_class_init), (gst_audio_filter_init),
(gst_audio_filter_set_caps),
(gst_audio_filter_class_add_pad_templates):
* gst-libs/gst/audio/gstaudiofilter.h:
Port GstAudioFilter to 0.10. This change technically breaks
API and ABI (and thus also every library developer's heart),
but seems justifiable on the grounds that the base class was
completely unusable before (ie. would crash immediately when
actually used). Fixes#403963 (and eventually also #403572).
Also document all of this a bit.
Original commit message from CVS:
* gst-libs/gst/utils/install-plugins.c:
(gst_install_plugins_spawn_child):
* tests/check/libs/utils.c:
(test_base_utils_install_plugins_do_callout):
Lowering log level to see why things fail on the p5 build bot;
fix some typos in unit test messages.
Original commit message from CVS:
* tests/check/libs/utils.c:
(test_base_utils_install_plugins_do_callout):
Don't hard-code temp directory for test helper; use GLib functions
to write out file and do error checking etc.
Original commit message from CVS:
* gst-libs/gst/utils/Makefile.am:
* gst-libs/gst/utils/base-utils.h:
* gst-libs/gst/utils/install-plugins.c:
(gst_install_plugins_context_set_xid),
(gst_install_plugins_context_new),
(gst_install_plugins_context_free),
(gst_install_plugins_get_helper),
(gst_install_plugins_spawn_child),
(gst_install_plugins_return_from_status),
(gst_install_plugins_installer_exited),
(gst_install_plugins_async), (gst_install_plugins_sync),
(gst_install_plugins_return_get_name),
(gst_install_plugins_installation_in_progress):
* gst-libs/gst/utils/install-plugins.h:
API: add API for applications to initiate installation of missing
plugins, ie. gst_install_plugins_async() primarily.
Based on libgimme-codec by Ryan Lortie.
* configure.ac:
Add --with-install-plugins-helper configure option so distros can specify
the path of the helper script or program to call when plugin installation
is requested (distros: please do any argument munging in this helper
script instead of patching GStreamer to pass arguments differently
to another program directly).
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Build and document new API.
* tests/check/libs/utils.c: (result_cb),
(test_base_utils_install_plugins_do_callout), (GST_START_TEST),
(libgstbaseutils_suite):
Some simple checks for the new API.
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (test_float_conversion):
Add small test for 32bit float <=> 64bit float conversion (works
only one way so far, 32=>64 produces structured noise).
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(set_structure_widths_32_and_64), (make_lossless_changes):
We don't support floats with a width of 40, 48 or 56 bits.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double),
(audio_convert_get_func_index):
* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
(make_lossless_changes):
Support for 64-bit float audio in audioconvert (#339837)
