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5408 commits

Author SHA1 Message Date
Stefan Kost
3ad40bebe5 docs/libs/Makefile.am: Remove cruft and do some cleanups.
Original commit message from CVS:
* docs/libs/Makefile.am:
Remove cruft and do some cleanups.
* docs/libs/gst-plugins-base-libs-docs.sgml:
Prepare for comming gtkdoc features (rebase against online docs).
2007-08-10 17:35:52 +00:00
Michael Smith
1b7a0df57e gst/audiorate/gstaudiorate.c: Debug output fixes.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
Debug output fixes.
* tests/check/elements/audiorate.c: (do_perfect_stream_test),
(GST_START_TEST):
Change the number of buffers used; 500 is too many and leads to
timeouts.
2007-08-10 13:55:44 +00:00
Tim-Philipp Müller
2c9bef0180 gst/: Printf format fixes (#465028).
Original commit message from CVS:
* gst/playback/gstqueue2.c:
* gst/videorate/gstvideorate.c:
Printf format fixes (#465028).
2007-08-10 10:08:05 +00:00
Michael Smith
9f9e76bc99 gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
If we have a large (> 1 second) discontinuity, push a series of
smaller buffers rather than a single very large buffer. Avoids
unreasonably large single buffer allocations when encountering a
large gap.
* tests/check/elements/audiorate.c: (GST_START_TEST),
(audiorate_suite):
Add a test for this.
2007-08-09 15:44:02 +00:00
Josep Torra Valles
9730f452ee gst/playback/gstplaybasebin.c: Fixes: #465015
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_commit),
(queue_remove_probe), (queue_threshold_reached):
Patch by: Josep Torra Valles <josep@fluendo.com>
Fixes: #465015
Make sure we remove the check_queues buffer probe from the
correct queue to avoid racily going back to "buffering 99%" when
buffering is actually complete.
Also, fix the spelling of Josep's surname in the ChangeLog.
2007-08-09 12:06:43 +00:00
Stefan Kost
87d96c656a ext/ogg/gstoggmux.c: Do not leak oggmux instance.
Original commit message from CVS:
* ext/ogg/gstoggmux.c:
Do not leak oggmux instance.
* ext/vorbis/vorbisenc.c:
Also log values.
2007-08-09 11:37:28 +00:00
Thomas Vander Stichele
f6be63b93b po/: Updated translations.
Original commit message from CVS:
* po/hu.po:
* po/it.po:
* po/nl.po:
* po/uk.po:
* po/vi.po:
Updated translations.
2007-08-09 10:51:55 +00:00
Yang Hong
afd8b931a9 ext/pango/gsttextoverlay.*: Add 'silent' property to GstTimeOverlay. Fixes #462979
Original commit message from CVS:
patch by: Yang Hong <hongyang@redflag-linux.com>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
Add 'silent' property to GstTimeOverlay. Fixes #462979
2007-08-08 16:07:21 +00:00
Josep Torre Valles
382b710277 Add connection-speed property. Fixes #464690.
Original commit message from CVS:
Patch by: Josep Torre Valles <josep@fluendo.com>
* docs/plugins/gst-plugins-base-plugins.args:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
(gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (gen_source_element):
Add connection-speed property. Fixes #464690.
2007-08-08 15:05:22 +00:00
Damien Lespiau
9b8c837165 Fix compilation on windows. Fixes #464320.
Original commit message from CVS:
Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
* configure.ac:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect):
Fix compilation on windows. Fixes #464320.
2007-08-07 15:13:46 +00:00
Josep Torre Valles
5e5aa7b402 gst/playback/: Move connection-speed property from playbin to playbasebin so that we can also configure it in source ...
Original commit message from CVS:
Patch by: Josep Torre Valles <josep@fluendo.com>
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(gst_play_base_bin_init), (queue_threshold_reached),
(gen_source_element), (setup_substreams),
(gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
(gst_play_base_bin_get_streaminfo_value_array):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_set_property), (gst_play_bin_get_property),
(gst_play_bin_handle_redirect_message):
Move connection-speed property from playbin to playbasebin so that we
can also configure it in source elements that have the connection-speed
property. Fixes #464028.
Add some debug info here and there.
2007-08-07 14:14:54 +00:00
Sebastian Dröge
5310373def gst/audiotestsrc/gstaudiotestsrc.c: Properly respond to conversion queries. Fixes #464079.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
Properly respond to conversion queries. Fixes #464079.
2007-08-06 16:42:22 +00:00
Sebastian Dröge
6f397125d1 gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
(gst_audio_test_src_init_sine_table),
(gst_audio_test_src_change_wave), (gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Add float/double and int32 support to audiotestsrc. Fixes #460422.
Also set the default volume to the default value specified in the
GParamSpec.
2007-08-03 19:53:11 +00:00
Jens Granseuer
ef33f2fdc4 gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx dot net>
* gst/audioconvert/gstaudioquantize.c:
Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
2007-08-03 19:40:14 +00:00
Wim Taymans
607fa48ad8 gst-libs/gst/rtsp/gstrtsptransport.c: Add rdt manager for rdt transport.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
Add rdt manager for rdt transport.
Fix parsing of RDT transport.
2007-08-03 15:44:01 +00:00
Jan Schmidt
2f6e0e7b57 configure.ac: Back to CVS
Original commit message from CVS:
* configure.ac:
Back to CVS
2007-08-03 14:43:15 +00:00
Jan Schmidt
221ae4ebd7 Release 0.10.14
Original commit message from CVS:
Release 0.10.14
2007-08-03 14:41:46 +00:00
Jan Schmidt
42771c4f3d Update .po files
Original commit message from CVS:
Update .po files
2007-08-03 14:24:08 +00:00
Jan Schmidt
a9f63daff7 tests/check/libs/audio.c: Fix the test to reflect the behaviour of gst_audio_clip_buffer.
Original commit message from CVS:
* tests/check/libs/audio.c: (GST_START_TEST):
Fix the test to reflect the behaviour of gst_audio_clip_buffer.
2007-07-27 17:37:19 +00:00
Jan Schmidt
d5dc054ea3 gst-libs/gst/audio/audio.c: When clipping a buffer with no timestamp, assume it is within the segment without warnings.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
When clipping a buffer with no timestamp, assume it is
within the segment without warnings.
Fixes: #460978
2007-07-27 17:10:47 +00:00
Wim Taymans
be5ef4b0ad gst-libs/gst/rtsp/gstrtspextension.c: Fire the signal on the object, not the interface.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
Fire the signal on the object, not the interface.
2007-07-27 11:16:23 +00:00
Jan Schmidt
1846b1a84d gst-libs/gst/rtsp/.cvsignore: Ber. Don't include the full path, idiot.
Original commit message from CVS:
* gst-libs/gst/rtsp/.cvsignore:
Ber. Don't include the full path, idiot.
2007-07-27 09:17:19 +00:00
Jan Schmidt
c339ca80c3 gst-libs/gst/rtsp/.cvsignore: Ignore generated files.
Original commit message from CVS:
* gst-libs/gst/rtsp/.cvsignore:
Ignore generated files.
2007-07-27 08:29:29 +00:00
Jan Schmidt
aa14635c47 gst-libs/gst/: Move the rtspextension.h interface into gstrtspextension.h as part of libgstrtsp instead of libgstinte...
Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/interfaces-marshal.list:
* gst-libs/gst/interfaces/rtspextension.c:
* gst-libs/gst/interfaces/rtspextension.h:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtsp.h:
* gst-libs/gst/rtsp/gstrtspextension.c:
(gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
(gst_rtsp_extension_detect_server),
(gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
(gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
(gst_rtsp_extension_configure_stream),
(gst_rtsp_extension_get_transports),
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
* gst-libs/gst/rtsp/gstrtspextension.h:
* gst-libs/gst/rtsp/rtsp-marshal.list:
Move the rtspextension.h interface into gstrtspextension.h
as part of libgstrtsp instead of libgstinterfaces, because it's
only for use within plugins, not applications.
Add stuff to do the enum & marshal generation needed in libgstrtsp now.
Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
is abstract.
2007-07-26 19:57:15 +00:00
Wim Taymans
6d1a34eff2 gst-libs/gst/interfaces/: Fix marshaller for the send signal.
Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/interfaces-marshal.list:
* gst-libs/gst/interfaces/rtspextension.c:
(gst_rtsp_extension_iface_init),
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
* gst-libs/gst/interfaces/rtspextension.h:
Fix marshaller for the send signal.
Add URL to stream selection interface method.
2007-07-26 15:48:01 +00:00
Jan Schmidt
50a3a239a0 gst-libs/gst/riff/Makefile.am: Pull in our dependencies from -base before those from outside.
Original commit message from CVS:
* gst-libs/gst/riff/Makefile.am:
Pull in our dependencies from -base before those from outside.
2007-07-26 15:35:43 +00:00
Wim Taymans
2c35823bdf API: gst_rtsp_base64_decode_ip()
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
* gst-libs/gst/rtsp/gstrtspbase64.h:
API: gst_rtsp_base64_decode_ip()
Added function to decode Base64 in-place.
2007-07-26 14:33:01 +00:00
Jan Schmidt
58afe32d55 tests/check/libs/.cvsignore: Ignore the mixer test binary.
Original commit message from CVS:
* tests/check/libs/.cvsignore:
Ignore the mixer test binary.
2007-07-26 14:08:01 +00:00
Jan Schmidt
b947924e28 ext/vorbis/vorbisdec.c: Gratuitous comment change to trigger a rebuild on the buildbots.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
Gratuitous comment change to trigger a rebuild on the buildbots.
2007-07-26 10:00:37 +00:00
Wim Taymans
8db50d49f7 gst-libs/gst/sdp/gstsdpmessage.*: Constify args where we can.
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
(gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
(gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
(gst_sdp_media_get_format), (gst_sdp_media_get_information),
(gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
(gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
(gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
(gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
(gst_sdp_media_get_attribute_val):
* gst-libs/gst/sdp/gstsdpmessage.h:
Constify args where we can.
2007-07-25 18:20:36 +00:00
Wim Taymans
256d005e49 gst-libs/gst/interfaces/: Move interface for RTSP extensions from -good to here.
Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/rtspextension.c:
(gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
(gst_rtsp_extension_detect_server),
(gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
(gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
(gst_rtsp_extension_configure_stream),
(gst_rtsp_extension_get_transports),
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
* gst-libs/gst/interfaces/rtspextension.h:
Move interface for RTSP extensions from -good to here.
Added helper methods to invoke interface methods.
2007-07-25 18:18:49 +00:00
Wim Taymans
77c284a31f Fix some more RTSP docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
(gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
(gst_rtsp_message_init_response),
(gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
(gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
(gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
(gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
(gst_rtsp_message_get_body), (dump_key_value):
* gst-libs/gst/rtsp/gstrtspmessage.h:
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
(gst_rtsp_range_parse):
* gst-libs/gst/rtsp/gstrtsprange.h:
* gst-libs/gst/rtsp/gstrtsptransport.c:
* gst-libs/gst/rtsp/gstrtspurl.c:
Fix some more RTSP docs.
Add some missing methods for dealing with messages.
2007-07-25 11:22:30 +00:00
Wim Taymans
3dff14d6b1 Added beginnings of RTSP documentation.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
* gst-libs/gst/rtsp/gstrtspbase64.h:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (add_auth_header),
(gst_rtsp_connection_write), (gst_rtsp_connection_send),
(read_body), (gst_rtsp_connection_receive),
(gst_rtsp_connection_next_timeout),
(gst_rtsp_connection_reset_timeout),
(gst_rtsp_connection_set_auth):
* gst-libs/gst/rtsp/gstrtspconnection.h:
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.h:
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
(gst_rtsp_range_parse):
* gst-libs/gst/rtsp/gstrtspurl.h:
Added beginnings of RTSP documentation.
2007-07-24 19:19:33 +00:00
Wim Taymans
ee42361c89 Document the SDP library.
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:
* gst-libs/gst/sdp/gstsdp.h:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
(gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
(gst_sdp_message_add_time), (gst_sdp_message_add_zone),
(gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
(gst_sdp_message_get_attribute_val),
(gst_sdp_message_add_attribute), (gst_sdp_media_new),
(gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
(gst_sdp_media_get_media), (gst_sdp_media_set_media),
(gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
(gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
(gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
(gst_sdp_media_get_format), (gst_sdp_media_add_format),
(gst_sdp_media_get_information), (gst_sdp_media_set_information),
(gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
(gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
(gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
(gst_sdp_media_set_key), (gst_sdp_media_get_key),
(gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
(gst_sdp_media_get_attribute_val_n),
(gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
(print_media), (gst_sdp_message_dump):
* gst-libs/gst/sdp/gstsdpmessage.h:
Document the SDP library.
Add some of the missing SDPMedia methods.
2007-07-24 17:37:03 +00:00
Wim Taymans
19e0dd3140 Move SDP and RTSP from helper objects in -good to a reusable library.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
* gst-libs/gst/rtsp/gstrtspbase64.h:
* gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
(gst_rtsp_connection_create), (gst_rtsp_connection_connect),
(add_auth_header), (add_date_header), (gst_rtsp_connection_write),
(gst_rtsp_connection_send), (read_line), (read_string), (read_key),
(parse_response_status), (parse_request_line), (parse_line),
(gst_rtsp_connection_read), (read_body),
(gst_rtsp_connection_receive), (gst_rtsp_connection_close),
(gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
(gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
(gst_rtsp_connection_set_auth):
* gst-libs/gst/rtsp/gstrtspconnection.h:
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
(gst_rtsp_strresult), (gst_rtsp_method_as_text),
(gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
(gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
(gst_rtsp_find_method):
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
(gst_rtsp_message_new), (gst_rtsp_message_init),
(gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
(gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
(gst_rtsp_message_init_data), (gst_rtsp_message_unset),
(gst_rtsp_message_free), (gst_rtsp_message_add_header),
(gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
(gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
(gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
(gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
(gst_rtsp_message_dump):
* gst-libs/gst/rtsp/gstrtspmessage.h:
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
(gst_rtsp_range_parse), (gst_rtsp_range_free):
* gst-libs/gst/rtsp/gstrtsprange.h:
* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
(gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
(gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
(range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
(gst_rtsp_transport_free):
* gst-libs/gst/rtsp/gstrtsptransport.h:
* gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
(gst_rtsp_url_free), (gst_rtsp_url_set_port),
(gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
* gst-libs/gst/rtsp/gstrtspurl.h:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/sdp/gstsdp.h:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
(gst_sdp_connection_init), (gst_sdp_bandwidth_init),
(gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
(gst_sdp_attribute_init), (gst_sdp_message_new),
(gst_sdp_message_init), (gst_sdp_message_uninit),
(gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
(gst_sdp_media_uninit), (gst_sdp_media_free),
(gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
(gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
(gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
(gst_sdp_message_add_zone), (gst_sdp_message_set_key),
(gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
(gst_sdp_message_get_attribute_val),
(gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
(gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
(gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
(gst_sdp_media_get_attribute_val_n),
(gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
(read_string), (read_string_del), (gst_sdp_parse_line),
(gst_sdp_message_parse_buffer), (print_media),
(gst_sdp_message_dump):
* gst-libs/gst/sdp/gstsdpmessage.h:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
Move SDP and RTSP from helper objects in -good to a reusable library.
Use a proper gst_ namespace.
2007-07-24 11:52:56 +00:00
Sebastian Dröge
9137e98926 ext/vorbis/vorbisdec.c: Use the new buffer clipping function from gstaudio here.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
(vorbis_dec_flush_decode):
Use the new buffer clipping function from gstaudio here.
2007-07-23 18:42:22 +00:00
Sebastian Dröge
6be2524031 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
* gst-libs/gst/audio/audio.h:
* tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
API: Add buffer clipping function for raw audio buffers. Fixes #456656.
Also add deprecation guards for gst_audio_structure_set_int() to the
header.
2007-07-23 18:26:09 +00:00
Stefan Kost
14e301026d docs/libs/gst-plugins-base-libs-sections.txt: Cleanup the docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
Cleanup the docs.
2007-07-23 14:45:16 +00:00
Dan Williams
ace9335ae3 gst/playback/gstplaybasebin.c: Don't return NULL when querying the stream info value array but instead return an empt...
Original commit message from CVS:
Patch by: Dan Williams <dcbw at redhat dot com>
* gst/playback/gstplaybasebin.c:
(gst_play_base_bin_get_streaminfo_value_array):
Don't return NULL when querying the stream info value array but instead
return an empty array. Fixes #459204.
2007-07-23 11:18:35 +00:00
Tim-Philipp Müller
2271ec928f gst/playback/gsturidecodebin.c: Init debug category before using it.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c:
Init debug category before using it.
2007-07-23 10:41:18 +00:00
Jan Schmidt
0776d87e32 gst-libs/gst/interfaces/mixer.h: Add padding vars in place of the signal pointers when building with DISABLE_DEPRECAT...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.h:
Add padding vars in place of the signal pointers
when building with DISABLE_DEPRECATED so that the
interface structure doesn't change size.
2007-07-21 09:56:09 +00:00
Marc-Andre Lureau
c161e29307 Fixes: #152864
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsamixertrack.c:
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixeroptions.h:
* gst-libs/gst/interfaces/mixertrack.c:
* gst-libs/gst/interfaces/mixertrack.h:
* tests/check/Makefile.am:
* tests/check/libs/mixer.c:
Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
Fixes: #152864
Add support for notifying mixer changes on the message bus, and
implement it in alsamixer.
API: gst_mixer_get_mixer_flags
API: gst_mixer_message_parse_mute_toggled
API: gst_mixer_message_parse_record_toggled
API: gst_mixer_message_parse_volume_changed
API: gst_mixer_message_parse_option_changed
API: GstMixerMessageType
API: GstMixerFlags
2007-07-21 09:21:12 +00:00
Michael Smith
11cf0dcd6b sys/xvimage/xvimagesink.c: xcontext->im_format is only for testing XShm support (as the header file comments document...
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps):
xcontext->im_format is only for testing XShm support (as the header
file comments document). Use xvimage->im_format for everything else.
Avoids spurious warnings on buffer allocation before setcaps.
2007-07-20 16:09:03 +00:00
Stefan Kost
9b2fb4d824 tests/: We should use $(LIBM).
Original commit message from CVS:
* tests/examples/volume/Makefile.am:
* tests/icles/Makefile.am:
We should use $(LIBM).
2007-07-20 07:22:15 +00:00
Stefan Kost
f50d9ab580 tests/icles/Makefile.am: This needs -lm.
Original commit message from CVS:
* tests/icles/Makefile.am:
This needs -lm.
2007-07-20 06:13:21 +00:00
Stefan Kost
dcd847c611 Add stdlib include (free, atoi, exit).
Original commit message from CVS:
* examples/app/appsrc_ex.c:
* examples/switch/switcher.c:
* ext/neon/gstneonhttpsrc.c:
* ext/timidity/gstwildmidi.c:
* ext/x264/gstx264enc.c:
* gst/mve/mveaudioenc.c: (mve_compress_audio):
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/spectrum/demo-audiotest.c:
* gst/spectrum/demo-osssrc.c:
* sys/dvb/gstdvbsrc.c:
Add stdlib include (free, atoi, exit).
2007-07-18 07:35:32 +00:00
Wim Taymans
d0e9a76a95 gst-libs/gst/rtp/gstbasertppayload.c: Don't break ABI, restore previous ranges. Keep the default random selection of ...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_set_property),
(gst_basertppayload_get_property):
Don't break ABI, restore previous ranges. Keep the default random
selection of timestamp and seqnum offset but as soon as the app sets a
specific value, use that one.
2007-07-16 10:10:28 +00:00
Bastien Nocera
312c0bd5ba sys/xvimage/xvimagesink.*: Add option to turn off double-buffering for debugging purposes.
Original commit message from CVS:
Patch by: Bastien Nocera <hadess at hadess dot net>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
(gst_xvimagesink_init), (gst_xvimagesink_class_init):
* sys/xvimage/xvimagesink.h:
Add option to turn off double-buffering for debugging purposes.
Fixes #437169.
2007-07-14 18:33:15 +00:00
Jorn Baayen
877fa6035a sys/: add 'handle-expose' property. Useful for video widgets which may want to be in control of Expose behaviour. Fix...
Original commit message from CVS:
Patch by: Jorn Baayen <jorn at openedhand dot com>
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
(gst_ximagesink_set_property), (gst_ximagesink_get_property),
(gst_ximagesink_init), (gst_ximagesink_class_init):
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
(gst_xvimagesink_init), (gst_xvimagesink_class_init):
* sys/xvimage/xvimagesink.h:
add 'handle-expose' property. Useful for video widgets which may want to
be in control of Expose behaviour. Fixes #380625
2007-07-14 18:20:41 +00:00
Wim Taymans
c82275a51d gst-libs/gst/rtp/gstbasertppayload.*: Fix ranges of rtp payloader properties so that the full range can be used in ad...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_event), (gst_basertppayload_push),
(gst_basertppayload_set_property),
(gst_basertppayload_get_property),
(gst_basertppayload_change_state):
* gst-libs/gst/rtp/gstbasertppayload.h:
Fix ranges of rtp payloader properties so that the full range can be
used in addition to -1 (random).
Fix wrong seqnum reporting in caps.
Fixes #420326.
2007-07-14 17:23:42 +00:00