Wim Taymans [Fri, 13 Jul 2007 18:12:19 +0000 (18:12 +0000)]
gst/videorate/gstvideorate.c: Use boilerplate.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_init),
(gst_video_rate_query):
Use boilerplate.
Add latency query, might not be perfect yet but already works a lot
better. Fixes #442557.
Jan Schmidt [Fri, 13 Jul 2007 16:05:17 +0000 (16:05 +0000)]
sys/xvimage/xvimagesink.*: After a caps change, redraw our borders to avoid garbage left there when the image format ...
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_setcaps):
* sys/xvimage/xvimagesink.h:
After a caps change, redraw our borders to avoid garbage left there
when the image format changes to a smaller size, like 16:9 -> 4:3
Also, hold the flow_lock a bit longer in the set_caps while we're
fiddling with the xcontext.
Jan Schmidt [Fri, 13 Jul 2007 16:02:23 +0000 (16:02 +0000)]
Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and...
Original commit message from CVS:
* Makefile.am:
* configure.ac:
* tests/Makefile.am:
Remove bogus check for libcheck, since we check for
gstreamer-check and it pulls in the required info from there, and we
weren't actually _using_ the information for libcheck ourselves
anyway.
Jan Schmidt [Fri, 13 Jul 2007 15:52:02 +0000 (15:52 +0000)]
gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix the r_mask test for RGBA32 on little-endian.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_caps_to_pixfmt):
Fix the r_mask test for RGBA32 on little-endian.
Fix a stupid typo that would have obviously broken
compilation on big-endian, if anyone was testing.
Wim Taymans [Thu, 12 Jul 2007 15:02:43 +0000 (15:02 +0000)]
gst/videotestsrc/videotestsrc.*: Add alpha to the color struct.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
(paint_hline_str4):
* gst/videotestsrc/videotestsrc.h:
Add alpha to the color struct.
Use a default alpha value of 255 instead of 128.
Wim Taymans [Thu, 12 Jul 2007 12:01:20 +0000 (12:01 +0000)]
gst/playback/gstplaybasebin.c: Clear the dynamic pads counter when starting a new uri. This makes reusing playbin wor...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (no_more_pads_full),
(setup_source):
Clear the dynamic pads counter when starting a new uri. This makes
reusing playbin work again.
Fixes #454264.
Stefan Kost [Thu, 12 Jul 2007 11:13:32 +0000 (11:13 +0000)]
configure.ac: Use pkg-config to locate check.
Original commit message from CVS:
* configure.ac:
Use pkg-config to locate check.
Tim-Philipp Müller [Wed, 11 Jul 2007 23:12:12 +0000 (23:12 +0000)]
Fix 'make check' build against core CVS.
Original commit message from CVS:
* configure.ac:
* tests/check/elements/volume.c: (GST_START_TEST):
Fix 'make check' build against core CVS.
Stefan Kost [Tue, 10 Jul 2007 20:46:41 +0000 (20:46 +0000)]
gst-libs/gst/: Make gtk-doc happy.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/tag/gstvorbistag.c:
Make gtk-doc happy.
Tim-Philipp Müller [Sun, 8 Jul 2007 13:07:38 +0000 (13:07 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c: Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs t...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_callback):
Quick hack to make audiosinks stop at EOS when operating in
pull-mode; needs to be fixed properly some day.
Stefan Kost [Fri, 6 Jul 2007 18:19:39 +0000 (18:19 +0000)]
docs/libs/gst-plugins-base-libs-sections.txt: Fix location of includes in the docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
Fix location of includes in the docs.
Jan Schmidt [Fri, 6 Jul 2007 11:40:45 +0000 (11:40 +0000)]
gst/ffmpegcolorspace/: Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and ...
Original commit message from CVS:
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/imgconvert.c: (img_convert),
(img_get_alpha_info):
Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
of the existing BGRA32 and RGBA32 formats with the alpha at the other
end of the word. Partially fixes #451908
Stefan Kost [Thu, 5 Jul 2007 08:43:30 +0000 (08:43 +0000)]
docs/: Simplify --extra-dir as gtkdoc scans recursively.
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/plugins/Makefile.am:
Simplify --extra-dir as gtkdoc scans recursively.
Wim Taymans [Tue, 3 Jul 2007 11:52:47 +0000 (11:52 +0000)]
gst/adder/gstadder.c: Make getcaps more robust by not using the proxycaps function. This makes sure that we don't end...
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_sink_getcaps),
(gst_adder_request_new_pad):
Make getcaps more robust by not using the proxycaps function. This makes
sure that we don't end up recursively calling getcaps upstream.
See #316248.
Wim Taymans [Fri, 29 Jun 2007 17:21:18 +0000 (17:21 +0000)]
gst/audioconvert/audioconvert.c: Include math.h to fix compilation.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c:
Include math.h to fix compilation.
Jan Schmidt [Fri, 29 Jun 2007 14:47:42 +0000 (14:47 +0000)]
gst/ffmpegcolorspace/gstffmpegcodecmap.c: Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel format, ...
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
format, as produced by some dc1394 cameras like the iSight.
See http://www.fourcc.org/yuv.php#IYU1
Sebastian Dröge [Thu, 28 Jun 2007 20:37:58 +0000 (20:37 +0000)]
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_dithering_get_type),
(gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
(gst_audio_convert_init), (gst_audio_convert_set_caps),
(gst_audio_convert_set_property), (gst_audio_convert_get_property):
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_noise_shaping),
(gst_audio_quantize_free_noise_shaping),
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither),
(gst_audio_quantize_setup_quantize_func),
(gst_audio_quantize_setup), (gst_audio_quantize_free):
* gst/audioconvert/gstaudioquantize.h:
Implement dithering and noise shaping in audioconvert. By default now
TPDF dithering (and no noise shaping) will be used when converting
from a higher bit depth to 20 bit depth or smaller, otherwise
everything will be as it is now.
For the last audioconvert in a pipeline it would make sense to
use some kind of noise shaping, enabling it by default for all
conversions would give undesired results though. Fixes #360246.
* tests/check/elements/audioconvert.c: (setup_audioconvert),
(GST_START_TEST):
Adjust unit test for the new audioconvert.
Wim Taymans [Thu, 28 Jun 2007 11:06:56 +0000 (11:06 +0000)]
gst/playback/gstqueue2.c: Use other metrics as well when estimating the buffer level.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
Use other metrics as well when estimating the buffer level.
Wim Taymans [Thu, 28 Jun 2007 10:21:19 +0000 (10:21 +0000)]
gst/playback/gstplaybasebin.c: Small debug improvement.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
Small debug improvement.
* gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
(plugin_init):
Tweak the rate estimation period.
When calculating the buffer filledness in rate estimation mode, don't
mix it with other metrics.
Wim Taymans [Thu, 28 Jun 2007 09:46:11 +0000 (09:46 +0000)]
gst/playback/gstdecodebin2.c: When creating the groups, allow for a 5 second, unlimited buffers preroll phase after w...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_group_new),
(gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
When creating the groups, allow for a 5 second, unlimited buffers
preroll phase after which we expose the group.
When the group is exposed, use a small number of buffers up to a 2
second limit. Also disconnect the overrun signal from multiqueue when we
exposed the group because it is not needed anymore.
Tim-Philipp Müller [Wed, 27 Jun 2007 22:30:19 +0000 (22:30 +0000)]
gst-libs/gst/tag/tags.c: Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags to utf8-validate; fixes...
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
(#451707); also, output some debugging info when dealing with
freeform strings.
* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
Add unit test for the above.
Tim-Philipp Müller [Wed, 27 Jun 2007 12:55:20 +0000 (12:55 +0000)]
gst-libs/gst/pbutils/descriptions.c: Add description for Windows Media RTP caps.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
Add description for Windows Media RTP caps.
* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
Remove RTP fields that don't define the format from caps.
Tim-Philipp Müller [Wed, 27 Jun 2007 10:14:03 +0000 (10:14 +0000)]
ext/vorbis/vorbisdec.c: Skip empty buffers, but not empty header buffers. That way the original vorbisdec unit test s...
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
Skip empty buffers, but not empty header buffers. That way the original
vorbisdec unit test still passes (#451145); also, take into account
that those empty packets might carry a granulepos.
* tests/check/Makefile.am:
* tests/check/elements/vorbisdec.c:
(_create_codebook_header_buffer), (_create_audio_buffer),
(GST_START_TEST), (vorbisdec_suite):
Add unit test that sends an empty packet.
Wim Taymans [Wed, 27 Jun 2007 09:49:51 +0000 (09:49 +0000)]
ext/vorbis/vorbisdec.c: Don't error out on 0-sized packets, just emit a warning because this is not a fatal error. Fi...
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
Don't error out on 0-sized packets, just emit a warning because this is
not a fatal error. Fixes #451145.
Stefan Kost [Mon, 25 Jun 2007 12:43:01 +0000 (12:43 +0000)]
docs/plugins/: Update docs with caps info.
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-decodebin2.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playbin.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
Update docs with caps info.
Tim-Philipp Müller [Mon, 25 Jun 2007 12:04:15 +0000 (12:04 +0000)]
po/POTFILES.in: Add more files with translatable strings (#450875).
Original commit message from CVS:
* po/POTFILES.in:
Add more files with translatable strings (#450875).
Edward Hervey [Sat, 23 Jun 2007 14:44:07 +0000 (14:44 +0000)]
ext/ogg/gstoggdemux.c: The chain should be freed if we error out here, else it will leak.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
The chain should be freed if we error out here, else it will leak.
* gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
(cleanup_decodebin):
Don't forget to *properly* remove the signals, else it will leak.
Jan Schmidt [Fri, 22 Jun 2007 14:25:27 +0000 (14:25 +0000)]
MAINTAINERS: Updating all the maintainers files
Original commit message from CVS:
* MAINTAINERS:
Updating all the maintainers files
Stefan Kost [Thu, 21 Jun 2007 08:34:46 +0000 (08:34 +0000)]
tests/examples/seek/seek.c: Destroy and recreate parse-launch based pipeline after stop to be able to play again. Reo...
Original commit message from CVS:
* tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
(main):
Destroy and recreate parse-launch based pipeline after stop to be able
to play again. Reorder some code and add more comments.
Wim Taymans [Wed, 20 Jun 2007 11:09:03 +0000 (11:09 +0000)]
gst/playback/gstdecodebin2.c: When handling a delayed-caps notification case, mark the group as dynamic so that the n...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
When handling a delayed-caps notification case, mark
the group as dynamic so that the nbdynamic count is
incremented and decremented correctly. Fixes: #449156
Patch by: Wim Taymans <wim@fluendo.com>
Andy Wingo [Tue, 19 Jun 2007 19:13:04 +0000 (19:13 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-06-19 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_init): Enable pull-mode operation.
Michael Smith [Tue, 19 Jun 2007 09:34:35 +0000 (09:34 +0000)]
gst-libs/gst/riff/riff-media.c: Change minimum rate back to 1000 to allow low-sample-rate wav files to play back.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Change minimum rate back to 1000 to allow low-sample-rate wav files
to play back.
Thomas Vander Stichele [Sun, 17 Jun 2007 17:27:09 +0000 (17:27 +0000)]
po/vi.po: Update translations.
Original commit message from CVS:
* po/vi.po:
Update translations.
David Schleef [Sat, 16 Jun 2007 03:42:14 +0000 (03:42 +0000)]
gst/playback/gstqueue2.c: Fix compile error from ignored return value.
Original commit message from CVS:
* gst/playback/gstqueue2.c:
Fix compile error from ignored return value.
Michael Smith [Fri, 15 Jun 2007 15:23:36 +0000 (15:23 +0000)]
gst/videoscale/vs_4tap.c: Update tmpbuf for all neccesary rows, not just one, as is required when downscaling.
Original commit message from CVS:
* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
Update tmpbuf for all neccesary rows, not just one, as is required
when downscaling.
Fixes #402076.
Michael Smith [Fri, 15 Jun 2007 11:15:28 +0000 (11:15 +0000)]
tests/check/pipelines/oggmux.c: Add a test that ensures we set DELTA_UNIT on all non-header, non-video buffers, if we...
Original commit message from CVS:
* tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
(eos_buffer_probe):
Add a test that ensures we set DELTA_UNIT on all non-header,
non-video buffers, if we have a video stream.
* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
(gst_ogg_mux_process_best_pad):
Move setting delta_pad to earlier, where we inspect all pads, so
that leading audio pages don't get DELTA_UNIT unset if they come
before the first DELTA_UNIT from video pages. Fixes the newly-added
test. Fixes #385527.
Tim-Philipp Müller [Thu, 14 Jun 2007 19:53:27 +0000 (19:53 +0000)]
tests/check/pipelines/streamheader.c: Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it fails on the p5-ppc6...
Original commit message from CVS:
* tests/check/pipelines/streamheader.c: (streamheader_suite):
Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
fails on the p5-ppc64 build bot and the failure looks like it is due
to the same issue as #348114, ie. a compiler bug.
Edward Hervey [Wed, 13 Jun 2007 18:20:57 +0000 (18:20 +0000)]
gst/playback/gstqueue2.c: Fix build on MacOSX.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_create_read):
Fix build on MacOSX.
Wim Taymans [Wed, 13 Jun 2007 09:01:32 +0000 (09:01 +0000)]
ext/ogg/gstoggdemux.c: Fix compilation on mingw. Fixes #446972.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
Fix compilation on mingw. Fixes #446972.
Wim Taymans [Tue, 12 Jun 2007 08:38:06 +0000 (08:38 +0000)]
gst/playback/gstqueue2.c: Fix a division by zero when the max percent is <= 0. Fixes #446572. also update the bufferi...
Original commit message from CVS:
Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_locked_enqueue):
Fix a division by zero when the max percent is <= 0. Fixes #446572.
also update the buffering status when receiving events. Fixes #446551.
Thiago Sousa Santos [Mon, 11 Jun 2007 11:32:26 +0000 (11:32 +0000)]
gst/playback/gstqueue2.c: Wait for preroll before attempting to forward a duration query upstream.
Original commit message from CVS:
Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (gst_queue_peer_query),
(gst_queue_handle_src_query):
Wait for preroll before attempting to forward a duration query upstream.
Fixes #445505.
Sébastien Moutte [Thu, 7 Jun 2007 21:08:38 +0000 (21:08 +0000)]
gst-libs/gst/rtp/gstbasertpdepayload.c: Use G_GINT64_CONSTANT macro for int64 constant.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
Use G_GINT64_CONSTANT macro for int64 constant.
* win32/common/libgstinterfaces.def:
* win32/common/libgsttag.def:
Add new exported functions.
Tim-Philipp Müller [Thu, 7 Jun 2007 14:25:32 +0000 (14:25 +0000)]
ext/ogg/gstoggmux.c: The BOS page of the first Dirac video stream needs to come before the BOS page of any Vorbis str...
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
The BOS page of the first Dirac video stream needs to come before
the BOS page of any Vorbis streams or other audio streams, just like
it is with Theora.
Wim Taymans [Thu, 7 Jun 2007 09:11:27 +0000 (09:11 +0000)]
gst/playback/gstqueue2.c: Fix compilation.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_get_range):
Fix compilation.
Thiago Sousa Santos [Wed, 6 Jun 2007 13:36:26 +0000 (13:36 +0000)]
gst/playback/gstqueue2.c: Add pull based scheduling and fix some deadlocks. Fixes #444523.
Original commit message from CVS:
Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (gst_queue_init),
(gst_queue_handle_sink_event), (gst_queue_chain),
(gst_queue_get_range), (gst_queue_src_checkgetrange_function),
(gst_queue_sink_activate_push), (gst_queue_src_activate_push),
(gst_queue_src_activate_pull):
Add pull based scheduling and fix some deadlocks. Fixes #444523.
Does not yet completely work because duration queries upstream won't
block yet.
Wim Taymans [Wed, 6 Jun 2007 09:08:50 +0000 (09:08 +0000)]
Some more fseeko checks.
Original commit message from CVS:
* configure.ac:
* gst/playback/gstqueue2.c: (gst_queue_create_read):
Some more fseeko checks.
Wim Taymans [Wed, 6 Jun 2007 08:01:42 +0000 (08:01 +0000)]
configure.ac: check for large file support.
Original commit message from CVS:
* configure.ac:
check for large file support.
Sven Arvidsson [Tue, 5 Jun 2007 21:36:11 +0000 (21:36 +0000)]
gst/subparse/gstsubparse.*: Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
Original commit message from CVS:
Based on a patch by Sven Arvidsson <sa at whiz dot se>:
* gst/subparse/gstsubparse.c: (parse_subrip),
(subviewer_unescape_newlines), (parse_subviewer),
(gst_sub_parse_data_format_autodetect),
(gst_sub_parse_format_autodetect), (gst_subparse_type_find):
* gst/subparse/gstsubparse.h:
Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a unit test for both SubViewer formats.
Michael Smith [Tue, 5 Jun 2007 17:08:04 +0000 (17:08 +0000)]
gst/audiotestsrc/gstaudiotestsrc.c: Don't overflow intermediate values when seeking to large time values in audiotest...
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
Don't overflow intermediate values when seeking to large time values
in audiotestsrc.
Wim Taymans [Tue, 5 Jun 2007 17:02:13 +0000 (17:02 +0000)]
gst/playback/gstqueue2.c: Include stdio to define fseeko.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_have_data),
(gst_queue_create_read), (gst_queue_read_item_from_file),
(gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
Include stdio to define fseeko.
Edward Hervey [Tue, 5 Jun 2007 16:37:09 +0000 (16:37 +0000)]
sys/v4l/gstv4lsrc.c: Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
Original commit message from CVS:
Patch by: Edward Hervey <edward@fluendo.com>
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
(gst_v4lsrc_query):
Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
Tim-Philipp Müller [Tue, 5 Jun 2007 16:20:44 +0000 (16:20 +0000)]
gst-libs/gst/riff/: Use gst_tag_utf8_from_freeform_string() from libgsttag instead of our own implementation.
Original commit message from CVS:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
our own implementation.
Wim Taymans [Tue, 5 Jun 2007 16:19:30 +0000 (16:19 +0000)]
gst-libs/gst/rtp/gstbasertpdepayload.c: Handle timestamp wraparound.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
Handle timestamp wraparound.
Wim Taymans [Tue, 5 Jun 2007 16:17:30 +0000 (16:17 +0000)]
gst/playback/gsturidecodebin.c: Make sure we name srcpads uniquely even when using different internal decodebins.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c: (no_more_pads_full),
(new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
(gst_uri_decode_bin_change_state):
Make sure we name srcpads uniquely even when using different internal
decodebins.
Signal no-more-pads when no more dynamic elements exist.
Remove pads on cleanup.
Thiago Sousa Santos [Tue, 5 Jun 2007 16:14:23 +0000 (16:14 +0000)]
gst/playback/gstqueue2.c: Add support for filebased buffering. Fixes #441264.
Original commit message from CVS:
Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (gst_queue_class_init),
(gst_queue_init), (gst_queue_finalize),
(gst_queue_write_buffer_to_file), (gst_queue_have_data),
(gst_queue_create_read), (gst_queue_read_item_from_file),
(gst_queue_open_temp_location_file),
(gst_queue_close_temp_location_file), (gst_queue_locked_flush),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_is_empty), (gst_queue_is_filled),
(gst_queue_change_state), (gst_queue_set_temp_location),
(gst_queue_set_property):
Add support for filebased buffering. Fixes #441264.
Wim Taymans [Tue, 5 Jun 2007 16:05:19 +0000 (16:05 +0000)]
gst/playback/gstdecodebin2.c: Add support for delayed caps fixation when autoplugging.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
(analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
(caps_notify_group_cb), (gst_decode_group_new),
(gst_decode_group_free):
Add support for delayed caps fixation when autoplugging.
Optimize cases where a multiqueue is not needed/wanted, like right after
anything that is not a demuxer.
Wim Taymans [Tue, 5 Jun 2007 16:02:57 +0000 (16:02 +0000)]
ext/ogg/gstoggdemux.c: consideratly speedup ogg chain detection by not trying to find a base timestamp for skeleton s...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
(gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
consideratly speedup ogg chain detection by not trying to find a base
timestamp for skeleton streams.
Wim Taymans [Tue, 5 Jun 2007 16:00:33 +0000 (16:00 +0000)]
gst/tcp/gstmultifdsink.*: Add support for remuve_flush.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
(gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
(gst_multi_fd_sink_remove_flush),
(gst_multi_fd_sink_remove_client_link),
(gst_multi_fd_sink_handle_client_write),
(gst_multi_fd_sink_handle_clients):
* gst/tcp/gstmultifdsink.h:
Add support for remuve_flush.
Wim Taymans [Tue, 5 Jun 2007 15:59:00 +0000 (15:59 +0000)]
Add draft design for forcing keyframes in encoders and implement in theoraenc.
Original commit message from CVS:
* docs/design/draft-keyframe-force.txt:
* ext/theora/theoraenc.c: (theora_enc_sink_event),
(theora_enc_chain):
Add draft design for forcing keyframes in encoders and implement in
theoraenc.
Jan Schmidt [Tue, 5 Jun 2007 13:22:18 +0000 (13:22 +0000)]
configure.ac: Back to CVS
Original commit message from CVS:
* configure.ac:
Back to CVS
Jan Schmidt [Tue, 5 Jun 2007 12:50:24 +0000 (12:50 +0000)]
Release 0.10.13 "What's going on?"
Original commit message from CVS:
Release 0.10.13 "What's going on?"
Jan Schmidt [Tue, 5 Jun 2007 12:32:03 +0000 (12:32 +0000)]
Update .po files
Original commit message from CVS:
Update .po files
Wim Taymans [Thu, 31 May 2007 17:08:58 +0000 (17:08 +0000)]
gst-libs/gst/riff/riff-media.c: In riff, the depth is stored in the size field but it just means that the least signi...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
In riff, the depth is stored in the size field but it just means that
the least significant bits are cleared. We can therefore just play
the sample as if it had a depth == width. Fixes: #440997
Patch by: Wim Taymans <wim@fluendo.com>
Patch by: Sebastian Dröge <slomo@circular-chaos.org>
Jan Schmidt [Thu, 31 May 2007 16:36:22 +0000 (16:36 +0000)]
gst-libs/gst/floatcast/floatcast.h: Define inline when needed on win32 builds. Fixes: #441295
Original commit message from CVS:
* gst-libs/gst/floatcast/floatcast.h:
Define inline when needed on win32 builds. Fixes: #441295
Wim Taymans [Tue, 29 May 2007 13:38:35 +0000 (13:38 +0000)]
gst/playback/gstplaybasebin.c: Stop buffering when the group is commited because the queues filled up.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_overrun),
(no_more_pads_full):
Stop buffering when the group is commited because the queues filled up.
Fixes #442024.
Jan Schmidt [Fri, 25 May 2007 10:07:26 +0000 (10:07 +0000)]
Revert commits towards #152864 made so far. We'll pick it up again after the 0.10.13 release.
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
(gst_alsa_mixer_free), (gst_alsa_mixer_update),
(gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
(gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
(gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_interface_supported),
(gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
(gst_alsa_mixer_element_set_property),
(gst_alsa_mixer_element_get_property),
(gst_alsa_mixer_element_change_state):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
* gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
(gst_mixer_option_changed):
* gst-libs/gst/interfaces/mixer.h:
Revert commits towards #152864 made so far. We'll pick it up again
after the 0.10.13 release.
Wim Taymans [Thu, 24 May 2007 16:22:23 +0000 (16:22 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
After an interrupt (PAUSED/flush) assume that the next sample should not
be aligned to the previous sample. Fixes #417992.
Tim-Philipp Müller [Thu, 24 May 2007 15:16:59 +0000 (15:16 +0000)]
gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the template caps for audio/x-dts, as wavparse ...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Don't add channels and rate fields to the template caps for
audio/x-dts, as wavparse might not always be able to set them,
which would then lead to 'caps are not a real subset of the
template caps' warnings.
Jan Schmidt [Thu, 24 May 2007 11:15:32 +0000 (11:15 +0000)]
gst/playback/gstplaybasebin.c: Handle unknown or invalid pads without crashing, as might occur if a media file like a...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
Handle unknown or invalid pads without crashing, as might occur if
a media file like an mp3 is specified as a subtitle file.
Fixes: #410039
Jan Schmidt [Thu, 24 May 2007 10:19:54 +0000 (10:19 +0000)]
gst/playback/gstplaybin.c: Block the subtitle bin output queue before ghosting it and linking, then unblock after. Th...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
(setup_sinks):
Block the subtitle bin output queue before ghosting it and linking,
then unblock after. This avoids spurious not-linked errors caused
by the queue starting up (because it gets linked when it is ghosted).
Fixes: #350299
Jan Schmidt [Wed, 23 May 2007 15:54:28 +0000 (15:54 +0000)]
tests/check/elements/playbin.c: Use /dev/zero instead of /dev/urandom to produce an invalid subtitle file. Avoids flu...
Original commit message from CVS:
* tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
file. Avoids flukes where the input gets typefound to some valid but
useless type.
Tim-Philipp Müller [Tue, 22 May 2007 15:45:19 +0000 (15:45 +0000)]
tests/check/: Add unit test for gnomevfssink seeking and position reporting for file:// URIs.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
(cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
Add unit test for gnomevfssink seeking and position reporting for
file:// URIs.
Mark Nauwelaerts [Tue, 22 May 2007 15:30:26 +0000 (15:30 +0000)]
ext/gnomevfs/gstgnomevfssink.*: see #412648.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
(gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
(gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
* ext/gnomevfs/gstgnomevfssink.h:
Fix position reporting, especially after a seek (from upstream),
see #412648.
Tim-Philipp Müller [Tue, 22 May 2007 15:04:41 +0000 (15:04 +0000)]
ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut.
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.c:
Repair umlaut.
Jan Schmidt [Tue, 22 May 2007 11:40:31 +0000 (11:40 +0000)]
gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3 samplerates. Fixes a regression caused by extra ...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Specify the full valid range for MP3 samplerates. Fixes a regression
caused by extra header checks since the last release.
Mike Smith [Mon, 21 May 2007 15:32:42 +0000 (15:32 +0000)]
sys/: Fix a locking-order bug I introduced with my changes the other day.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
Fix a locking-order bug I introduced with my changes the other day.
Patch by Mike Smith.
Michael Smith [Mon, 21 May 2007 15:24:21 +0000 (15:24 +0000)]
ext/theora/theoradec.c: Don't look inside 0-length packets (which indicate duplicated frames)
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_handle_data_packet):
Don't look inside 0-length packets (which indicate duplicated
frames)
Wim Taymans [Mon, 21 May 2007 10:25:44 +0000 (10:25 +0000)]
Small cleanups.
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_read_sector):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Small cleanups.
* ext/theora/theoradec.c: (theora_dec_sink_event):
Fix typo.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
Add some FIXME
* gst/playback/gstdecodebin.c: (queue_underrun_cb):
And some debug info when a FIXME path is hit.
Wim Taymans [Mon, 21 May 2007 09:45:28 +0000 (09:45 +0000)]
gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init),
(gst_base_rtp_audio_payload_init),
(gst_base_rtp_audio_payload_finalize),
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer),
(gst_base_rtp_payload_audio_handle_event):
Some cleanups, remove minptime property as it is now in the parent
class.
Override parent class event function.
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_event), (gst_basertppayload_set_property),
(gst_basertppayload_get_property):
* gst-libs/gst/rtp/gstbasertppayload.h:
Add min-ptime property.
Add handle-event vmethod. Fixes #415001.
Christian Schaller [Fri, 18 May 2007 17:10:03 +0000 (17:10 +0000)]
update spec
Original commit message from CVS:
update spec
Stefan Kost [Fri, 18 May 2007 15:23:43 +0000 (15:23 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_change_state):
Fix typo in comment.
* gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
free_dynamics, pad_probe, close_pad_link, try_to_link_1,
get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
close_link):
* gst/playback/gstplaybin.c (gst_play_bin_set_property,
gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
Remove trailing whitespaces in comments.
* gst/volume/Makefile.am:
Fix tabs.
Marc-Andre Lureau [Fri, 18 May 2007 15:10:08 +0000 (15:10 +0000)]
gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed, set_option, get_option, _gst_reserved):
Original commit message from CVS:
patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
* gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
set_option, get_option, _gst_reserved):
Revert reordering functions (keep ABI).
Jan Schmidt [Thu, 17 May 2007 17:35:46 +0000 (17:35 +0000)]
sys/: When we create our own window, indicate that we handle the
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
(gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
(gst_ximagesink_show_frame):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
(gst_xvimagesink_show_frame):
When we create our own window, indicate that we handle the
WM_DELETE client message from the window manager, so that it won't
kill our window (and our app) along with it. Handle ClientMessage,
post an error on the bus, and close the window. Further buffers
arriving will result in a FlowError because the window has been
destroyed.
Fixes: #393975
Clean up the X event handling loop and make them the same for
both xvimagesink and ximagesink while I'm at it.
Wim Taymans [Thu, 17 May 2007 16:27:32 +0000 (16:27 +0000)]
gst/playback/gstdecodebin2.c: Make decodebin2 autoplug depayloaders too.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
Make decodebin2 autoplug depayloaders too.
* gst/playback/gsturidecodebin.c: (source_new_pad):
Set the newly created decoder in a usable state when autoplugging a
dynamic source such as RTSP.
Tim-Philipp Müller [Thu, 17 May 2007 16:11:03 +0000 (16:11 +0000)]
gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and ignore audio-codec tags for video streams....
Original commit message from CVS:
* gst/playback/gststreaminfo.c: (cb_probe):
Ignore video-codec tag for audio streams and ignore audio-codec tags
for video streams. Should make codec name collection a bit more
robust against sloppy demuxers that send tag events containing both
tags down each pad.
Wim Taymans [Thu, 17 May 2007 15:22:44 +0000 (15:22 +0000)]
gst/playback/gstqueue2.c: Tweak the buffering thresholds a little.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_rates):
Tweak the buffering thresholds a little.
Update the buffer size with the previously calculate rate instead of
only when we calculate a new rate so that we get smoother buffering
updates.
* gst/playback/Makefile.am:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (unknown_type),
(add_element_stream), (no_more_pads_full), (no_more_pads),
(source_no_more_pads), (new_decoded_pad), (array_has_value),
(gen_source_element), (has_all_raw_caps), (analyse_source),
(remove_decoders), (make_decoder), (remove_source),
(source_new_pad), (setup_source), (decoder_query_init),
(decoder_query_duration_fold), (decoder_query_duration_done),
(decoder_query_position_fold), (decoder_query_position_done),
(decoder_query_latency_fold), (decoder_query_latency_done),
(decoder_query_seeking_fold), (decoder_query_seeking_done),
(decoder_query_generic_fold), (gst_uri_decode_bin_query),
(gst_uri_decode_bin_change_state), (plugin_init):
New element that intergrates a source, optional buffering element and
decodebin.
Tim-Philipp Müller [Thu, 17 May 2007 14:17:17 +0000 (14:17 +0000)]
configure.ac: Bump libtheora requirement to 1.0alpha5 for the pixformat check (also has a .pc file, so we don't need ...
Original commit message from CVS:
* configure.ac:
Bump libtheora requirement to 1.0alpha5 for the pixformat check
(also has a .pc file, so we don't need the fallback check any
longer). Fixes #438840.
Wim Taymans [Thu, 17 May 2007 13:36:11 +0000 (13:36 +0000)]
gst/playback/gstqueue2.c: fix build.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_get_type),
(gst_queue_class_init), (gst_queue_finalize), (update_time_level),
(apply_segment), (apply_buffer), (update_buffering),
(reset_rate_timer), (update_rates), (gst_queue_locked_flush),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_handle_sink_event), (gst_queue_is_filled),
(gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
(plugin_init):
fix build.
Wim Taymans [Thu, 17 May 2007 11:57:44 +0000 (11:57 +0000)]
gst/playback/: On our way to playbin2 this is the new network queue that does buffering all by itself using high and ...
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstqueue2.c: (gst_queue_get_type),
(gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
(gst_queue_getcaps), (gst_queue_bufferalloc),
(gst_queue_acceptcaps), (update_time_level), (apply_segment),
(apply_buffer), (update_buffering), (reset_rate_timer),
(update_rates), (gst_queue_locked_flush),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_handle_sink_event), (gst_queue_is_empty),
(gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
(gst_queue_loop), (gst_queue_handle_src_event),
(gst_queue_handle_src_query), (gst_queue_sink_activate_push),
(gst_queue_src_activate_push), (gst_queue_change_state),
(gst_queue_set_property), (gst_queue_get_property), (plugin_init):
On our way to playbin2 this is the new network queue that does buffering
all by itself using high and low watermarks. It can also measure up and
downstream bandwidth to optimally size the queue.
Michael Smith [Thu, 17 May 2007 11:16:14 +0000 (11:16 +0000)]
gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta...
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
Use the segment->last_stop value to calculate the next timestamp to
generate after a seek; not the segment->start value.
David Schleef [Tue, 15 May 2007 20:14:06 +0000 (20:14 +0000)]
docs/Makefile.am: Install docs even when --disable-gtk-doc is disabled. This matches the behavior of gtk+. Fixes #3...
Original commit message from CVS:
* docs/Makefile.am: Install docs even when --disable-gtk-doc
is disabled. This matches the behavior of gtk+. Fixes #349099.
Wim Taymans [Tue, 15 May 2007 17:11:09 +0000 (17:11 +0000)]
ext/ogg/gstoggdemux.c: Some more chained streaming ogg timestamp fixes.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
Some more chained streaming ogg timestamp fixes.
Wim Taymans [Tue, 15 May 2007 16:46:10 +0000 (16:46 +0000)]
ext/ogg/gstoggdemux.c: Add some FIXMEs.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_handle_page):
Add some FIXMEs.
Fix chain start/stop segment handling based on patch by
<ahalda at cs dot mcgill dot ca> see #320984.
Michael Smith [Tue, 15 May 2007 15:33:54 +0000 (15:33 +0000)]
configure.ac: We don't require a C++ compiler. So don't require one.
Original commit message from CVS:
* configure.ac:
We don't require a C++ compiler. So don't require one.
Stefan Kost [Tue, 15 May 2007 15:29:17 +0000 (15:29 +0000)]
ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds, gst_alsa_mixer_check, gst_alsa_mixer_dispatch, gst_alsa_mixer_...
Original commit message from CVS:
* ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
gst_alsa_mixer_update_track):
Apply some of the cleanup Tim suggested in #152864 afterwards.
Marc-Andre Lureau [Tue, 15 May 2007 14:01:26 +0000 (14:01 +0000)]
ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
Original commit message from CVS:
patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
* ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
_GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
gst_alsa_mixer_handle_source_callback,
gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
* ext/alsa/gstalsamixer.h (handle_source, interface, dir):
* ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
gst_alsa_mixer_element_interface_supported,
gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
gst_alsa_mixer_element_set_property,
gst_alsa_mixer_element_get_property,
gst_alsa_mixer_element_change_state):
* ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
* gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
gst_mixer_option_changed):
* gst-libs/gst/interfaces/mixer.h (set_option, get_option,
volume_changed, option_changed, _gst_reserved):
Implement notification for alsamixer. Fixes #152864
David Schleef [Tue, 15 May 2007 03:53:11 +0000 (03:53 +0000)]
gst/videotestsrc/videotestsrc.*: Add support for video/x-raw-bayer.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add support for video/x-raw-bayer.
David Schleef [Sun, 13 May 2007 01:06:19 +0000 (01:06 +0000)]
sys/xvimage/xvimagesink.c: Add some sanity checking for the XVImage size returned by X.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Add some sanity checking for the XVImage size returned by X.
Related to #377400.
Wim Taymans [Sat, 12 May 2007 16:18:39 +0000 (16:18 +0000)]
gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields as described in updated application/x-rt...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_set_gst_timestamp):
Parse and use additional caps fields as described in updated
application/x-rtp caps spec.
Wim Taymans [Sat, 12 May 2007 16:16:22 +0000 (16:16 +0000)]
ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data packets, ignore the stream in the total lengt...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_collect_chain_info):
If there is a stream in a chain without any data packets, ignore the
stream in the total length calculations. Might be related to #436820.