platform/upstream/gstreamer.git
16 years agogst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties...
Mark Nauwelaerts [Sat, 31 May 2008 19:57:57 +0000 (19:57 +0000)]
gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
Add a gtk-doc chunk for the new properties to have a Since: indication.

16 years agoChangeLog surgery, mark API change
Mark Nauwelaerts [Sat, 31 May 2008 19:50:59 +0000 (19:50 +0000)]
ChangeLog surgery, mark API change

Original commit message from CVS:
ChangeLog surgery, mark API change

16 years agogst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual...
Mark Nauwelaerts [Sat, 31 May 2008 18:10:47 +0000 (18:10 +0000)]
gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
(gst_base_audio_src_change_state):
Provide readable actual-buffer-time and actual-latency-time properties
that reflect the configured ringbuffer values. Fixes #524724.

16 years agogst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an...
Wim Taymans [Fri, 30 May 2008 15:29:20 +0000 (15:29 +0000)]
gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...

Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
(gst_basertppayload_change_state):
Simply converting the running time into an RTP timestamp by scaling it
based on the clock-rate is good enough for making an RTP timestamp. This
has the added benefit that we can later on expose a property with the
RTP timestamp of running time 0, as is needed for RTSP servers to
generate the response of the PLAY request.

16 years agogst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audio...
Sebastian Dröge [Fri, 30 May 2008 08:42:17 +0000 (08:42 +0000)]
gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...

Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
Allow up to 11 positioned channels now that audioconvert can handle
this but add no default positions for > 8 channels.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some unit tests for the above change: Test conversion of
11 positioned channels to stereo and the other way around, test
conversion of 15 unpositioned channels in different ways.

16 years agowin32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols.
Sebastian Dröge [Thu, 29 May 2008 19:45:40 +0000 (19:45 +0000)]
win32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols.

Original commit message from CVS:
* win32/common/libgstaudio.def:
Add gst_audio_clock_reset to the list of exported symbols.

16 years agotests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit...
Sebastian Dröge [Thu, 29 May 2008 19:37:47 +0000 (19:37 +0000)]
tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit test as we now support 7 (and more...

Original commit message from CVS:
* tests/check/elements/vorbisdec.c: (vorbisdec_suite):
Remove wrong_channels_identification_header unit test as we now
support 7 (and more channels).

16 years agogst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other...
Sebastian Dröge [Thu, 29 May 2008 12:17:16 +0000 (12:17 +0000)]
gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...

Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_one_other):
If mixing left or right to center (or the other way around) only take
the complete value if we don't already have the original position in
the source.

16 years agogst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right...
Sebastian Dröge [Thu, 29 May 2008 11:34:09 +0000 (11:34 +0000)]
gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...

Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(gst_audio_fixate_channel_positions):
Allow rear center together with rear left/right and other previously
conflicting channel positions. The reason why they weren't allowed
was the channel mixing implementation in audioconvert.
Also take this into account when fixing channel layouts.
Allow setting channel positions for 1/2 channels when using
gst_audio_set_structure_channel_position().
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
(gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
Major rewrite of the channel mixing.
We now allow previously conflicting channel positions to appear
together (rear center and rear left/right for example).
Fixes bug #533817.
Rework the way channels are mixed together to take more possible
channel positions into account, properly mix from/to side channels
and don't assume that either center, left&right or nothing of a
specific position is available anymore.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Adjust unit tests with non-standard 1/2 channel layouts to the more
correct new behaviour.
Add a unit test for 5.1->Stereo downmixing.

16 years agoext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined...
Sebastian Dröge [Thu, 29 May 2008 07:02:50 +0000 (07:02 +0000)]
ext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined in the Vorbis spec. Use NONE ch...

Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
Add sane defaults for the 7 and 8 channel layouts as those are
undefined in the Vorbis spec. Use NONE channel layouts when decoding
more than 8 channels instead of erroring out. Fixes bug #535356.

16 years agoAdd theoraparse to the docs and fix some docs.
Wim Taymans [Wed, 28 May 2008 16:10:20 +0000 (16:10 +0000)]
Add theoraparse to the docs and fix some docs.

Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/theora/theoraparse.c:
Add theoraparse to the docs and fix some docs.

16 years agogst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the...
Wim Taymans [Wed, 28 May 2008 15:48:33 +0000 (15:48 +0000)]
gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the track.end sector is included in t...

Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
Fix EOS condition and track addition check, the track.end sector is
included in the track. Fixes #533265.

16 years agogst/videorate/gstvideorate.*: React (more) to NEWSEGMENT
Mark Nauwelaerts [Wed, 28 May 2008 14:49:24 +0000 (14:49 +0000)]
gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT

Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
(gst_video_rate_flush_prev), (gst_video_rate_event),
(gst_video_rate_chain):
* gst/videorate/gstvideorate.h:
React (more) to NEWSEGMENT
Small adjustment in timestamp calculation to prevent mismatches
Fixes #435633.

16 years agotests/examples/seek/seek.c: Initialise error to NULL as we should.
Tim-Philipp Müller [Wed, 28 May 2008 11:31:44 +0000 (11:31 +0000)]
tests/examples/seek/seek.c: Initialise error to NULL as we should.

Original commit message from CVS:
* tests/examples/seek/seek.c: (make_parselaunch_pipeline):
Initialise error to NULL as we should.

16 years agogst/adder/gstadder.c: Implement latency query.
Sebastian Dröge [Wed, 28 May 2008 08:14:47 +0000 (08:14 +0000)]
gst/adder/gstadder.c: Implement latency query.

Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration),
(gst_adder_query_latency), (gst_adder_query):
Implement latency query.

16 years agogst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns
Sebastian Dröge [Tue, 27 May 2008 18:10:00 +0000 (18:10 +0000)]
gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns

Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration):
Correctly resync the iterator if gst_iterator_next() returns
GST_ITERATOR_RESYNC.

16 years agowin32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037).
Tim-Philipp Müller [Tue, 27 May 2008 17:14:07 +0000 (17:14 +0000)]
win32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037).

Original commit message from CVS:
* win32/vs6/libgstpbutils.dsp:
Add pbutils-enumtypes.c to sources (#518037).

16 years agogst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time...
Wim Taymans [Tue, 27 May 2008 16:20:17 +0000 (16:20 +0000)]
gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...

Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
* gst-libs/gst/audio/gstaudioclock.h:
Add method to inform the clock that the time starts from 0 again. We use
this info to calculate a clock offset so that the time we report in
internal_time is monotonically increasing, as required by the clock base
class. Fixes #521761.
API: GstAudioClock::gst_audio_clock_reset()
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Reset reported time when we (re)create the ringbuffer.

16 years agoext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwrit...
Tim-Philipp Müller [Tue, 27 May 2008 16:11:32 +0000 (16:11 +0000)]
ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwritten by capture volumes if an als...

Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c:
(gst_alsa_mixer_track_update_alsa_capabilities):
Make sure playback volumes aren't accidentally overwritten by
capture volumes if an alsa mixer track has both playback and
capture capabilities: we create two GstMixerTracks in that
case, so make sure we query only the alsa capabilities that
refer to the type of GstMixerTrack we created from the dual
capability alsa element. Should fix issues with Audigy2 sound
cards (#518082).

16 years agotests/check/pipelines/oggmux.c: Don't use deprecated function.
Tim-Philipp Müller [Tue, 27 May 2008 10:57:56 +0000 (10:57 +0000)]
tests/check/pipelines/oggmux.c: Don't use deprecated function.

Original commit message from CVS:
* tests/check/pipelines/oggmux.c: (test_pipeline):
Don't use deprecated function.

16 years agogst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads...
Wim Taymans [Tue, 27 May 2008 10:35:55 +0000 (10:35 +0000)]
gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...

Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
(gst_decode_group_control_source_pad), (gst_decode_group_expose):
Check for NULL cases and log them, creating ghostpads can, for example,
fail when the pad returns wrong caps.
* gst/playback/gstplaybin2.c: (perform_eos):
When pushing out the EOS event, collect the return value and warn when
something failed.

16 years agogst-libs/gst/riff/riff-media.c: Add support for DVCPRO.
Wim Taymans [Mon, 26 May 2008 17:18:52 +0000 (17:18 +0000)]
gst-libs/gst/riff/riff-media.c: Add support for DVCPRO.

Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add support for DVCPRO.

16 years agogst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour...
Tim-Philipp Müller [Mon, 26 May 2008 10:29:20 +0000 (10:29 +0000)]
gst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour to bilinear.

Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
Change default scaling method from nearest-neighbour to bilinear.

16 years agotests/check/libs/video.c: More checks.
Tim-Philipp Müller [Mon, 26 May 2008 10:26:00 +0000 (10:26 +0000)]
tests/check/libs/video.c: More checks.

Original commit message from CVS:
* tests/check/libs/video.c:
More checks.

16 years agoLimit duration to a maximum of five seconds for tmplayer format where we can guess...
Tim-Philipp Müller [Sun, 25 May 2008 20:51:35 +0000 (20:51 +0000)]
Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...

Original commit message from CVS:
* gst/subparse/gstsubparse.c: (parser_state_init),
(gst_sub_parse_format_autodetect), (handle_buffer):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (test_tmplayer_style3b):
Limit duration to a maximum of five seconds for tmplayer format where
we can guess the duration only from the timestamp of the next line of
text. We don't want to show a text for eternities just because nothing
else is being said for a while.

16 years agogst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers...
Wim Taymans [Fri, 23 May 2008 14:14:28 +0000 (14:14 +0000)]
gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...

Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_change_state):
Check sequence numbers, mark input buffers with a discont flag for the
subclass when we detected a gap, drop duplicate buffers. We do this
because one can use the element without a jitterbuffer in front and we
don't want to feed the subclasses invalid or reordered data.
Do an error when the subclass did not provide a process function instead
of crashing.
Some other small cleanups.

16 years agogst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride...
Tim-Philipp Müller [Thu, 22 May 2008 22:35:40 +0000 (22:35 +0000)]
gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here.

Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
May just as well use the precalculated uvstride here.

16 years agoAdd some documentation comments, and some new headers to be scanned.
Jan Schmidt [Thu, 22 May 2008 22:09:16 +0000 (22:09 +0000)]
Add some documentation comments, and some new headers to be scanned.

Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* 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-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.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-playback.xml:
* docs/plugins/inspect/plugin-queue2.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-uridecodebin.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:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/gdp/gstgdpdepay.h:
* gst/gdp/gstgdppay.h:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcp.h:
Add some documentation comments, and some new headers to be scanned.
Rename some internal enum declarations (audioconvert's DitherType and
NoiseShapingType, GstUnitType from the TCP elements) to match the
documented GObject type names so that the docs pick them up.
Name the playbin2 docs markups properly so they get picked up. They'll
need renaming back when/if playbin2 becomes playbin.
100% symbol coverage for the plugin docs, booya.

16 years agogst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.
Thijs Vermeir [Thu, 22 May 2008 18:30:15 +0000 (18:30 +0000)]
gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.

Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
Fix generation of NV12/NV21 frames. Fixes bug #532454.

16 years agogst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and...
Sjoerd Simons [Thu, 22 May 2008 11:59:33 +0000 (11:59 +0000)]
gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...

Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/playback/gstdecodebin.c: (remove_fakesink):
Lock the fakesink before setting the state to NULL and removing it from
the bin so that a concurrent state change cannot interfere.
Fixes #534331.

16 years agodocs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled.
Felipe Contreras [Wed, 21 May 2008 17:09:42 +0000 (17:09 +0000)]
docs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled.

Original commit message from CVS:
* docs/Makefile.am:
Fix installing plugin documentation when gtk-doc is disabled.

16 years agogst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h
Felipe Contreras [Wed, 21 May 2008 17:01:16 +0000 (17:01 +0000)]
gst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h

Original commit message from CVS:
* gst-libs/gst/rtsp/Makefile.am:
Distribute, don't install md5.h

16 years agogst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
Julien Moutte [Wed, 21 May 2008 16:47:58 +0000 (16:47 +0000)]
gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.

Original commit message from CVS:
2008-05-21  Julien Moutte  <julien@fluendo.com>

* gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
instead of SOL_IP, works on more platforms.
* gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
arguments.

16 years agoSome debug and comment fixes.
Wim Taymans [Wed, 21 May 2008 16:44:15 +0000 (16:44 +0000)]
Some debug and comment fixes.

Original commit message from CVS:
* ext/vorbis/vorbisdec.c:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
Some debug and comment fixes.
* tests/examples/dynamic/addstream.c: (main):
Fix , to ;

16 years agoDon't use bad gst_element_get_pad().
Wim Taymans [Wed, 21 May 2008 16:36:50 +0000 (16:36 +0000)]
Don't use bad gst_element_get_pad().

Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
* gst/playback/decodetest.c: (new_decoded_pad_cb):
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
(cleanup_decodebin):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
(connect_element), (gst_decode_group_control_demuxer_pad):
* gst/playback/gstplaybasebin.c: (queue_remove_probe),
(queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
(mute_group_type):
* gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
(gst_play_bin_set_property), (handoff), (gen_video_element),
(gen_text_element), (gen_audio_element), (gen_vis_element),
(remove_sinks), (add_sink), (setup_sinks):
* gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
* gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
(gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
(gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
(gen_video_chain), (gen_text_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
(gst_play_sink_request_pad):
* gst/playback/gsturidecodebin.c: (type_found), (setup_source):
* gst/playback/test.c: (gen_video_element), (gen_audio_element),
(cb_newpad):
* gst/playback/test6.c: (new_decoded_pad_cb):
* tests/check/elements/audioconvert.c: (GST_START_TEST):
* tests/check/elements/audiorate.c: (test_injector_chain),
(do_perfect_stream_test):
* tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
* tests/check/elements/gdpdepay.c: (GST_START_TEST):
* tests/check/elements/gnomevfssink.c:
* tests/check/elements/textoverlay.c:
(notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
* tests/check/elements/videotestsrc.c: (GST_START_TEST):
* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
* tests/check/pipelines/oggmux.c: (test_pipeline):
* tests/check/pipelines/streamheader.c: (GST_START_TEST):
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
* tests/examples/seek/scrubby.c: (make_wav_pipeline):
* tests/examples/seek/seek.c: (make_mod_pipeline),
(make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
(make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
(make_theora_pipeline), (make_vorbis_theora_pipeline),
(make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
(make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
(update_fill), (msg_buffering):
Don't use bad gst_element_get_pad().

16 years agogst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of...
Stefan Kost [Wed, 21 May 2008 14:35:41 +0000 (14:35 +0000)]
gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of strf fields for broken mulaw/alaw.

Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
Fix wrong method name in docs. Fix calculation of strf fields for
broken mulaw/alaw.
* gst-libs/gst/riff/riff-read.c:
Whitespace fix and removing double ';'.

16 years agodocs/design/part-playbin2.txt: Add some leftover doc.
Wim Taymans [Wed, 21 May 2008 11:52:30 +0000 (11:52 +0000)]
docs/design/part-playbin2.txt: Add some leftover doc.

Original commit message from CVS:
* docs/design/part-playbin2.txt:
Add some leftover doc.

16 years agogst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
Sebastian Dröge [Wed, 21 May 2008 11:36:37 +0000 (11:36 +0000)]
gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.

Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Fix copy & paste error in last commit.

16 years agogst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_S...
Sebastian Dröge [Wed, 21 May 2008 11:30:58 +0000 (11:30 +0000)]
gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...

Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
other channel positions when source has SIDE channels and dest doesn't
or the other way around.

16 years agogst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.
Henrik Eriksson [Wed, 21 May 2008 11:29:25 +0000 (11:29 +0000)]
gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.

Original commit message from CVS:
Patch by: Henrik Eriksson <henriken at axis dot com>
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
(gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
(gst_multi_fd_sink_get_property):
* gst/tcp/gstmultifdsink.h:
Add support for DSCP QOS. Fixes #469933.

16 years agotests/check/elements/audioconvert.c: Add another test that checks if conversion betwe...
Sebastian Dröge [Wed, 21 May 2008 07:46:02 +0000 (07:46 +0000)]
tests/check/elements/audioconvert.c: Add another test that checks if conversion between standard 1 and 2 channel layo...

Original commit message from CVS:
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add another test that checks if conversion between standard 1 and 2
channel layouts with and without positions set is working.

16 years agogst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
Sebastian Dröge [Wed, 21 May 2008 07:39:56 +0000 (07:39 +0000)]
gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.

Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
Allow non-standard 2 channel layouts.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some tests for converting and remapping non-standard 1 and 2
channel layouts.

16 years agogst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix...
Sebastian Dröge [Wed, 21 May 2008 07:28:04 +0000 (07:28 +0000)]
gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.

Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_normalize):
Prevent division by zero if the channel mix matrix contains only
zeroes.

16 years agogst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.
Antoine Tremblay [Wed, 21 May 2008 06:45:22 +0000 (06:45 +0000)]
gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.

Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
Close a buffer memory leak. Fixes bug #534071.

16 years agogst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public...
Sebastian Dröge [Wed, 21 May 2008 06:39:20 +0000 (06:39 +0000)]
gst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public as there are no setters/getters...

Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtsptransport.h:
Make the GstRTSPTransport struct members public as there are no
setters/getters and it's supposed to be changed directly.
Fixes bug #533087.

16 years agogst/adder/gstadder.c: Adder also doesn't support audio/x-raw-int with width!=depth...
Sebastian Dröge [Wed, 21 May 2008 05:48:05 +0000 (05:48 +0000)]
gst/adder/gstadder.c: Adder also doesn't support audio/x-raw-int with width!=depth so don't claim this on the pad tem...

Original commit message from CVS:
* gst/adder/gstadder.c:
Adder also doesn't support audio/x-raw-int with width!=depth so don't
claim this on the pad template caps.

16 years agogst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if...
Wim Taymans [Tue, 20 May 2008 16:26:53 +0000 (16:26 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_sync_latency):
We can only use our optimal calibration if we prerolled before the
latency expired.

16 years agoconfigure.ac: Require core CVS for GstBaseSrc buffer caps setting magic.
Tim-Philipp Müller [Tue, 20 May 2008 14:35:42 +0000 (14:35 +0000)]
configure.ac: Require core CVS for GstBaseSrc buffer caps setting magic.

Original commit message from CVS:
* configure.ac:
Require core CVS for GstBaseSrc buffer caps setting magic.

16 years agogst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
Sebastian Dröge [Tue, 20 May 2008 12:26:32 +0000 (12:26 +0000)]
gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.

Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Fix logic in last commit.

16 years agogst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number...
Sebastian Dröge [Tue, 20 May 2008 12:15:34 +0000 (12:15 +0000)]
gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...

Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Passthrough the channel positions if the number of output channels is
the same as the number of input channels, the input had a channel
layout and downstream requests no special one. We did this already for
> 2 channels but now it's also done for 1 channel. Fixes bug #533617.

16 years agoext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get...
Wim Taymans [Tue, 20 May 2008 11:13:27 +0000 (11:13 +0000)]
ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...

Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
(gst_gnome_vfs_src_finalize),
(gst_gnome_vfs_src_received_headers_callback),
(gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
* ext/gnomevfs/gstgnomevfssrc.h:
Set the ICY caps on the srcpad from where they get picked up by the base
class now and set on the outgoing buffers.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
BaseSrc now sets the caps on outgoing buffers automatically.

16 years agogst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is...
Wim Taymans [Tue, 20 May 2008 11:09:06 +0000 (11:09 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Change the way in which the ringbuffer is started when dealing with a
slaved clock and latency. We now sync to the clock until we reach
upstream latency before starting the ringbuffer. This has the effect
that we can accurately align the master and slave clocks and let the
rate correction code take care of the initial drift or rounding errors
instead of leaving them uncorrected with the old approach.

16 years agogst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when...
Sebastian Dröge [Tue, 20 May 2008 08:12:19 +0000 (08:12 +0000)]
gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.

Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Correctly set the default channel positions when converting to 8
channels.

16 years agoconfigure.ac: Error out if we don't have the required version of core.
Tim-Philipp Müller [Mon, 19 May 2008 16:13:25 +0000 (16:13 +0000)]
configure.ac: Error out if we don't have the required version of core.

Original commit message from CVS:
* configure.ac:
Error out if we don't have the required version of core.

16 years agogst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop...
Tim-Philipp Müller [Mon, 19 May 2008 15:59:40 +0000 (15:59 +0000)]
gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...

Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find):
Use data scan helper in aac typefinder and stop scanning
for headers when we've found a type. Also fix potential invalid
memory access when calculating the frame length.

16 years agogst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE...
Tim-Philipp Müller [Mon, 19 May 2008 14:09:08 +0000 (14:09 +0000)]
gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...

Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
(mpeg_sys_is_valid_pack):
Don't modify scan context when we return FALSE in ensure_data, so
it's possible to continue scanning, and we don't end up with a NULL
data pointer and a positive size, which might bite us the next time
we're called. Small constification.

16 years agogst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports...
Sebastian Dröge [Fri, 16 May 2008 21:12:02 +0000 (21:12 +0000)]
gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.

Original commit message from CVS:
* gst/adder/gstadder.c:
Adder doesn't support 24 bit samples so don't claim it supports them
in the pad template caps.

16 years agogst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further proces...
Wim Taymans [Wed, 14 May 2008 20:28:02 +0000 (20:28 +0000)]
gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...

Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain):
Validate the RTP packet before further processing it. It's just too
dangerous to accept random packets and people are not forced to use a
jitterbuffer or session manager to filter out the bad packets.
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data),
(gst_rtp_buffer_get_payload_subbuffer):
Small cleanups.
When setting extension data in a buffer that is too small, we fail and
we should not set the extension bit.
Change GST_WARNINGS into g_warning because they really are
programming errors.
* tests/check/libs/rtp.c: (GST_START_TEST):
Catch the g_warnings now in the unit tests and that fact that failing to
set extension data left the extension bit untouched.

16 years agogst/audioresample/gstaudioresample.c: Revert previous change which made basetransform...
Tim-Philipp Müller [Wed, 14 May 2008 13:57:41 +0000 (13:57 +0000)]
gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...

Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Revert previous change which made basetransform handle buffer_alloc
and which breaks things badly in the non-passthrough case since it
returned buffers with a different (ie. sometimes smaller) size than
the size requested.

16 years agogst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values...
Bernard B [Wed, 14 May 2008 13:43:12 +0000 (13:43 +0000)]
gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...

Original commit message from CVS:
Patch by: Bernard B <b-gnome at largestprime dot net>
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
Fix seqnum compare function for bordercase values and fix the docs
again. Fixes #533075.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add a testcase for seqnum compare function.

16 years agogst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates...
Sebastian Dröge [Wed, 14 May 2008 10:58:52 +0000 (10:58 +0000)]
gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...

Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps),
(gst_adder_class_init):
Correctly declare the supported endianness on the pad templates
and check for correct endianness in the set caps function. Adder
only supports native endianness.
Also use gst_element_class_set_details_simple().

16 years agosys/xvimage/xvimagesink.c: Better debug logging in port value handling. Merging separ...
Stefan Kost [Wed, 14 May 2008 09:12:10 +0000 (09:12 +0000)]
sys/xvimage/xvimagesink.c: Better debug logging in port value handling. Merging separate port value loops into one.

Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Better debug logging in port value handling. Merging separate port
value loops into one.

16 years agogst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set...
Hannes Bistry [Tue, 13 May 2008 16:02:19 +0000 (16:02 +0000)]
gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.

Original commit message from CVS:
Patch by: Hannes Bistry <hannesb at gmx dot de>
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
* gst/tcp/gsttcpserversink.c:
(gst_tcp_server_sink_handle_server_read),
(gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
Fix regression in clientsrc because we did not add the fd to the poll
set anymore. Fixes #532364.
Do some cleanups here and there.

16 years agogst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.
Sebastian Dröge [Tue, 13 May 2008 13:04:24 +0000 (13:04 +0000)]
gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.

Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplay-marshal.list:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
Use correct marshallers. GstCaps are a boxed type and no GObject
subclass.

16 years agowin32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param() to...
Sebastian Dröge [Tue, 13 May 2008 11:37:15 +0000 (11:37 +0000)]
win32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param() to the exported symbols.

Original commit message from CVS:
* win32/common/libgstrtsp.def:
Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
symbols.

16 years agotests/check/elements/audioresample.c: Add unit test for the latest basetransform...
Sjoerd Simons [Tue, 13 May 2008 10:59:49 +0000 (10:59 +0000)]
tests/check/elements/audioresample.c: Add unit test for the latest basetransform negotiation changes.

Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* tests/check/elements/audioresample.c:
(live_switch_alloc_only_48000), (live_switch_get_sink_caps),
(live_switch_push), (GST_START_TEST):
Add unit test for the latest basetransform negotiation changes.
See bug #526768.

16 years agogst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger...
Sebastian Dröge [Tue, 13 May 2008 09:14:44 +0000 (09:14 +0000)]
gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width.

Original commit message from CVS:
* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
Fix nv12<->nv21 conversion if stride is larger than width.

16 years agoext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use it as...
j^ [Tue, 13 May 2008 07:28:21 +0000 (07:28 +0000)]
ext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use it as offset for the timestamps. Fixes b...

Original commit message from CVS:
Patch by: j^ <j at oil21 dot org>
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
(gst_ogg_pad_parse_skeleton_fisbone):
* ext/ogg/gstoggdemux.h:
Parse presentation time from skeleton streams and use it as offset
for the timestamps. Fixes bug #530068.

16 years agogst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more...
Wim Taymans [Mon, 12 May 2008 08:45:11 +0000 (08:45 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
Revert previous patch that attempted to more accurately calculate the
initial offset between master and slave clock. The best thing we can do
in general is take the time of both clocks as the diff since we don't
know when the actual preroll happened.

16 years agogst-libs/gst/pbutils/install-plugins.c: Fix docs: type and missing word.
Tim-Philipp Müller [Sun, 11 May 2008 19:52:59 +0000 (19:52 +0000)]
gst-libs/gst/pbutils/install-plugins.c: Fix docs: type and missing word.

Original commit message from CVS:
* gst-libs/gst/pbutils/install-plugins.c:
Fix docs: type and missing word.

16 years agogst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the ...
Tim-Philipp Müller [Sat, 10 May 2008 20:16:21 +0000 (20:16 +0000)]
gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the 'new' data scan helper for this inste...

Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Don't do lots of 4-byte peeks, but use the 'new' data scan helper
for this instead; don't check if we've found enough markers after
each and every step, it's enough to do that only if we've actually
found a new marker.
Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.

16 years agogst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of...
Tim-Philipp Müller [Sat, 10 May 2008 18:19:17 +0000 (18:19 +0000)]
gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...

Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
(data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
(mpeg_video_stream_type_find):
Move scan helper thingy to the beginning of the file so we can use
it in other typefind functions. Rename it to something more
generic. Also improve handling of things towards the end of the
typefind data: peek as much as we can if we know the size of the
data, rather than just min_size.

16 years agoDocument the GstTuner and GstColorBalance interfaces, and some other random API funct...
Jan Schmidt [Fri, 9 May 2008 21:42:26 +0000 (21:42 +0000)]
Document the GstTuner and GstColorBalance interfaces, and some other random API functions that needed it. 70% symbol ...

Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/interfaces/colorbalance.c:
* gst-libs/gst/interfaces/colorbalance.h:
* gst-libs/gst/interfaces/colorbalancechannel.c:
* gst-libs/gst/interfaces/colorbalancechannel.h:
* gst-libs/gst/interfaces/tuner.c:
* gst-libs/gst/interfaces/tunerchannel.c:
* gst-libs/gst/interfaces/tunerchannel.h:
* gst-libs/gst/interfaces/tunernorm.c:
* gst-libs/gst/interfaces/tunernorm.h:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
Document the GstTuner and GstColorBalance interfaces, and some
other random API functions that needed it. 70% symbol coverage, woo.

16 years agogst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require...
Wim Taymans [Fri, 9 May 2008 16:38:10 +0000 (16:38 +0000)]
gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.

Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
Choose to allocate one less segment but require one additional segment
as latency.
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
No need to increment the number of segments in the source.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (clock_convert_external),
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
Remove adding latency when returning the internal time while subtracting
it again when we use the value a little later.
When calculating the end timestamp, we are making a rounding error
with the current algorithm. Ensure that we don't accumulate these
rounding errors when aligning samples by not resampling at all if we
don't need to. Fixes #419351.
Make the initial calibration of the clock slaving a little more
predictable and accurate. Also handle the case where we don't do
clock slaving.

16 years agogst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between...
Sebastian Dröge [Fri, 9 May 2008 08:34:52 +0000 (08:34 +0000)]
gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between those two formats. Fixes bug #53...

Original commit message from CVS:
Based on a patch by:
Björn Benderius <bjoern dot benderius at axis dot com>
* 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: (nv12_to_nv21):
* gst/ffmpegcolorspace/imgconvert_template.h:
Add conversions from/to NV12 and NV21 and conversions between those
two formats. Fixes bug #532166.

16 years agogst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek...
Edward Hervey [Thu, 8 May 2008 17:35:44 +0000 (17:35 +0000)]
gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek() doesn't return anything, which hap...

Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Abort the h264 typefinding as soon as _peek() doesn't return anything,
which happens for example with files smaller than 128kb.

16 years agogst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065.
Wouter Cloetens [Thu, 8 May 2008 14:46:27 +0000 (14:46 +0000)]
gst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065.

Original commit message from CVS:
Patch by: Wouter Cloetens <zombie at e2big dot org>
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create), (md5_digest_to_hex_string),
(auth_digest_compute_hex_urp), (auth_digest_compute_response),
(add_auth_header), (gst_rtsp_connection_free),
(gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
(gst_rtsp_connection_set_auth_param),
(gst_rtsp_connection_clear_auth_params):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Add Digest authorization support for RTSP connections. See #532065.
* gst-libs/gst/rtsp/md5.c:
* gst-libs/gst/rtsp/md5.h:
Yeap, another md5 implementation until we can depend on a glib that has
support for it.

16 years agogst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation...
Sjoerd Simons [Thu, 8 May 2008 06:20:42 +0000 (06:20 +0000)]
gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...

Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Let audioresample use the buffer allocation of basetransform instead
of it's own stuff.
* tests/check/elements/audioresample.c: (alloc_only_48000),
(GST_START_TEST), (audioresample_suite):
Add unit test for the recent basetransform bugfix, where upstream
changes caps to something that can't be passed through anymore.

16 years agowin32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much...
Ole André Vadla Ravnås [Wed, 7 May 2008 19:50:27 +0000 (19:50 +0000)]
win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather use the real thing than h...

Original commit message from CVS:
* win32/common/config.h.in:
Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
use the real thing than having "???" unconditionally.

16 years agogst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency...
Wim Taymans [Wed, 7 May 2008 15:47:03 +0000 (15:47 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter.

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query):
Report the latency with the new seglatency parameter.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
(gst_ring_buffer_acquire):
* gst-libs/gst/audio/gstringbuffer.h:
Add new field to the ringbufferspec to specify the expected latency
between the underlying device read/write pointer, this is needed
when writing sinks that sit a little closer to the hardware.
Add some more docs for other fields.

16 years agogst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.
Wim Taymans [Wed, 7 May 2008 10:38:23 +0000 (10:38 +0000)]
gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.

Original commit message from CVS:
* gst-libs/gst/app/.cvsignore:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp-marshal.list:
Add marshal.list, make it compile and add to cvsignore.
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose),
(gst_app_sink_stop):
Small cleanups.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_unlock),
(gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
(gst_app_src_create), (gst_app_src_set_caps),
(gst_app_src_get_caps), (gst_app_src_set_size),
(gst_app_src_get_size), (gst_app_src_set_seekable),
(gst_app_src_get_seekable), (gst_app_src_set_max_buffers),
(gst_app_src_get_max_buffers), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream):
* gst-libs/gst/app/gstappsrc.h:
Beat appsrc in shape, add signals and actions.
Add some docs.
Add properties for caps, size, seekability and max-buffers.
Fix unlock/stop code.

16 years agogst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function...
Sebastian Dröge [Tue, 6 May 2008 12:35:09 +0000 (12:35 +0000)]
gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function yet for some reason instead of cras...

Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_transform_ip):
Return NOT_NEGOTIATED if we didn't set a process function yet for some
reason instead of crashing later. Might fix bug #509125.

16 years agogst/audioconvert/: Add support for more than 8 channels and NONE channel layouts...
Tim-Philipp Müller [Tue, 6 May 2008 12:12:16 +0000 (12:12 +0000)]
gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...

Original commit message from CVS:
Based on a patch by: Tim-Philipp Müller  <tim.muller at collabora co uk>
* gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps),
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
Add support for more than 8 channels and NONE channel layouts. For
more than 8 channels no channel conversion is supported yet, only
format conversions are supported. Fixes bug #398033.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST), (audioconvert_suite):
Add some unit tests by Tim for checking the NONE channel layouts
and more than 8 channels and add some more unit tests for channel
conversions.

16 years agogst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL...
Wim Taymans [Tue, 6 May 2008 10:16:49 +0000 (10:16 +0000)]
gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL before unreffing it.

Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (connect_pad):
When autoplugging fails, set the element back to NULL before
unreffing it.

16 years agowin32/common/libgstaudio.def: Add gst_base_audio_src_[sg]et_slave_method() to the...
Sebastian Dröge [Tue, 6 May 2008 09:59:43 +0000 (09:59 +0000)]
win32/common/libgstaudio.def: Add gst_base_audio_src_[sg]et_slave_method() to the exported symbols.

Original commit message from CVS:
* win32/common/libgstaudio.def:
Add gst_base_audio_src_[sg]et_slave_method() to the exported
symbols.

16 years agogst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.
Sebastian Dröge [Mon, 5 May 2008 12:33:05 +0000 (12:33 +0000)]
gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.

Original commit message from CVS:
* gst/subparse/samiparse.c: (handle_start_sync),
(end_sami_element), (characters_sami):
Remove trailing, leading and double whitespaces.
Correctly timestamp buffers and output the last buffer too.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a simple unit test for SAMI parsing.

16 years agogst/subparse/samiparse.c: Only output characters inside the "sync" elements. There...
Young-Ho Cha [Mon, 5 May 2008 11:14:48 +0000 (11:14 +0000)]
gst/subparse/samiparse.c: Only output characters inside the "sync" elements. There could be other elements like "styl...

Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian dot net>
* gst/subparse/samiparse.c: (handle_start_sync),
(start_sami_element), (end_sami_element), (characters_sami),
(sami_context_reset):
Only output characters inside the "sync" elements. There could be
other elements like "style" that have some content but should
not be printed. Fixes bug #467911.

16 years agogst-libs/gst/app/gstappsink.*: Start some docs.
Wim Taymans [Mon, 5 May 2008 10:27:45 +0000 (10:27 +0000)]
gst-libs/gst/app/gstappsink.*: Start some docs.

Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (gst_app_sink_unlock_start),
(gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
(gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(gst_app_sink_set_caps), (gst_app_sink_set_drop),
(gst_app_sink_get_drop):
* gst-libs/gst/app/gstappsink.h:
Start some docs.
Add property to drop buffers when the queue is filled
Fix unlocking and flushing when the queues are filled.

16 years agogst/playback/: Allow setting -1 as current-audio to mute the current audio stream...
Sebastian Dröge [Mon, 5 May 2008 10:03:51 +0000 (10:03 +0000)]
gst/playback/: Allow setting -1 as current-audio to mute the current audio stream, similar to what is done for subtit...

Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (set_audio_mute),
(set_active_source):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(playbin_set_audio_mute):
Allow setting -1 as current-audio to mute the current audio stream,
similar to what is done for subtitles. Fixes bug #342294.

16 years agogst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn.
Edward Hervey [Mon, 5 May 2008 07:41:03 +0000 (07:41 +0000)]
gst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn.

Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
It's SorensOn and not SorensEn.

16 years agogst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video.
Tim-Philipp Müller [Sun, 4 May 2008 15:23:36 +0000 (15:23 +0000)]
gst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video.

Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
Fix description of video/x-flash-video.

16 years agoRemove some unused code.
Sebastian Dröge [Sun, 4 May 2008 15:02:20 +0000 (15:02 +0000)]
Remove some unused code.

Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
Remove some unused code.
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_free_noise_shaping):
Don't return before freeing the noise shaping history.

16 years agotests/check/elements/subparse.c: Add unit test for the tmplayer variant from bug...
Tim-Philipp Müller [Sat, 3 May 2008 16:00:04 +0000 (16:00 +0000)]
tests/check/elements/subparse.c: Add unit test for the tmplayer variant from bug #530962.

Original commit message from CVS:
* tests/check/elements/subparse.c: (do_test),
(test_tmplayer_style3b), (subparse_suite):
Add unit test for the tmplayer variant from bug #530962.

16 years agogst/subparse/: Fix parsing of tmplayer subtitle variant where every single line conta...
Tim-Philipp Müller [Sat, 3 May 2008 15:45:23 +0000 (15:45 +0000)]
gst/subparse/: Fix parsing of tmplayer subtitle variant where every single line contains text and there isn't an empt...

Original commit message from CVS:
* gst/subparse/gstsubparse.c: (handle_buffer),
(gst_sub_parse_sink_event):
* gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
(tmplayer_parse_line):
Fix parsing of tmplayer subtitle variant where every single line contains
text and there isn't an empty line after each line to determine the
duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
making sure that we push out the last line of text without a duration if
there's still text left in the buffer at the end.

16 years agogst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer...
Tim-Philipp Müller [Sat, 3 May 2008 15:39:04 +0000 (15:39 +0000)]
gst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer offset (doesn't work so well if no b...

Original commit message from CVS:
* gst/subparse/gstsubparse.c: (feed_textbuf):
Fix detection of discontinuities based on the buffer offset (doesn't work
so well if no buffer offset is set) and also check for the DISCONT buffer
flag. This keeps the parser state from being reset after each buffer in
the unit test.

16 years agogst/typefind/gsttypefindfunctions.c: Further fine-tuning: don't absolutely require...
Tim-Philipp Müller [Sat, 3 May 2008 12:09:16 +0000 (12:09 +0000)]
gst/typefind/gsttypefindfunctions.c: Further fine-tuning: don't absolutely require sequence or GOP headers but adjust...

Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
Further fine-tuning: don't absolutely require sequence or GOP headers
(as introduced in the previous commit), but adjust the typefind
probabilities returned accordingly if we don't see them. Also make sure
picture header and first slice are somewhat close to each other (which
is not perfect but still better than requiring a fixed offset or having
no limit at all).

16 years agogst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally...
Wim Taymans [Fri, 2 May 2008 12:13:08 +0000 (12:13 +0000)]
gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c...

Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
(gst_basertppayload_sink_setcaps),
(gst_basertppayload_sink_getcaps):
Rename the setcaps/getcaps function internally to make it clear that
they are called for the sink pad.

16 years agogst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffe...
Wim Taymans [Fri, 2 May 2008 12:11:07 +0000 (12:11 +0000)]
gst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffer and convert them into a vmetho...

Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init),
(gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
(gst_base_rtp_depayload_packet_lost),
(gst_base_rtp_depayload_set_gst_timestamp):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Catch packet-lost events from the jitterbuffer and convert them into a
vmethod call (lost-packet) so that depayloaders can do something smart.
Also add a default packet-lost function that sends out a segment update
to the decoders.

16 years agogst/playback/: Also include config.h when relying on defines from it. Fixes the build...
Stefan Kost [Fri, 2 May 2008 11:13:05 +0000 (11:13 +0000)]
gst/playback/: Also include config.h when relying on defines from it. Fixes the build. Its been a please to serve :)

Original commit message from CVS:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/playback/test7.c:
Also include config.h when relying on defines from it. Fixes the
build. Its been a please to serve :)

16 years agoAdd support for NV12 and NV21 in videotestsrc
Thijs Vermeir [Fri, 2 May 2008 10:54:51 +0000 (10:54 +0000)]
Add support for NV12 and NV21 in videotestsrc

Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
(paint_setup_NV21), (paint_hline_NV12_NV21):
Add support for NV12 and NV21 in videotestsrc