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Author SHA1 Message Date
Tim-Philipp Müller
f317061b9c audioconvert: simplify fixate_format function some more 
If we have no output format yet, any format will do. The
!out_info condition existed in every path, so just split
it our for clarity. KISS.
 2013-08-23 19:43:14 +01:00
Tim-Philipp Müller
7a481c13ae audioconvert: make fixate function more readable 
Use some variables to replace accessor macros to make code
a little bit mor readable.
 2013-08-23 19:11:17 +01:00
Tim-Philipp Müller
f448977dbd audioconvert: remove unnecessary deep nesting in fixate function 
Makes it easier to read and removes two levels of indentation.
 2013-08-23 18:53:48 +01:00
Sebastian Dröge
cebae4514a audioconvert: If we have to lose precision, try to lose as less precision as possible 
https://bugzilla.gnome.org/show_bug.cgi?id=706624
 2013-08-23 18:52:50 +02:00
Sebastian Dröge
ff5d3313d4 Release 1.1.1 2013-06-05 18:31:27 +02:00
Tim-Philipp Müller
f5c0d61be7 Update disted orc backup files 
Generated with 0.4.17 now.
 2013-04-22 13:58:33 +01:00
Sebastian Dröge
948a4a3632 gst: Add better support for static plugins 2013-04-15 15:52:58 +02:00
Tim-Philipp Müller
5f59b4f7ee Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-03 23:05:09 +00:00
Sebastian Dröge
9e6021fe4b audioconvert: Always prefer the input format if possible 
Previously we could've chosen another format with the same
depth even if the input format was possible.

Also make sure to chose according to the order in the
caps.
 2012-11-01 16:44:05 +01:00
Sebastian Dröge
bc4389806d audioconvert: Also ignore the SIGNED flag when matching an output format 2012-11-01 14:31:29 +01:00
Rasmus Rohde
c286f8ffa2 audioconvert: Prefer output formats with the same depth or at least a higher depth 
Enhance current code to prefer an exact match on sample depth if
possible. Also ignore GST_AUDIO_FORMAT_FLAG_UNPACK when checking
equality on the flags.
 2012-11-01 14:29:43 +01:00
Sebastian Dröge
3c1041d5eb Revert "gst: Add better support for static plugins" 
This reverts commit d2d79e3bc2,
which was accidentially pushed.
 2012-10-24 13:26:26 +02:00
Sebastian Dröge
d2d79e3bc2 gst: Add better support for static plugins 2012-10-24 12:10:44 +02:00
Mark Nauwelaerts
a66ff00908 audioconvert: enhance transforming caps 
... so as to preserve input format precision,
and preferably not convert at all.
 2012-10-19 16:02:44 +02:00
Tim-Philipp Müller
6842698f0d Purge all references to liboil 
And remove unused ffmpegcolorspace tests in the process.

https://bugzilla.gnome.org/show_bug.cgi?id=673285
 2012-09-29 11:47:52 +01:00
Tim-Philipp Müller
f7c6aa5abd Release 0.11.94 2012-09-14 02:47:54 +01:00
Mark Nauwelaerts
22d7149ba6 audioconvert: plug leak 2012-09-06 14:02:07 +02:00
Mark Nauwelaerts
88e73f8515 audioconvert: prefer channels of base caps when fixating 
... which in turn prefers to preserve input channels when converting.
 2012-07-25 15:58:19 +02:00
Wim Taymans
5d3b56e9c4 audioconvert: prefix orc functions with audio_convert_orc_ 2012-07-23 17:24:13 +02:00
Tim-Philipp Müller
3c6a3ad629 Use new gst_element_class_set_static_metadata() 2012-04-10 00:45:16 +01:00
Sebastian Dröge
ad42b16375 gst: Update for GST_PLUGIN_DEFINE() API change 2012-04-05 15:11:05 +02:00
Sebastian Dröge
65307dd132 gst: Update versioning 2012-04-04 14:55:15 +02:00
Wim Taymans
b52c035f13 audioconvert: remove useless transform_ip function 2012-04-02 11:21:26 +02:00
Mark Nauwelaerts
aaf84a941e audioconvert: plug caps leak 2012-03-30 16:56:40 +02:00
Wim Taymans
1982d1ce12 Release 0.11.3 2012-03-22 15:51:39 +01:00
Wim Taymans
25137962ad fix for caps API changes 2012-03-11 19:04:41 +01:00
Wim Taymans
ef980bc09b audioconvert: improve fixation 2012-02-27 12:52:07 +01:00
Wim Taymans
9212619549 update for new fixate_caps function 2012-02-22 12:32:44 +01:00
Wim Taymans
fcdc385aa1 port to new map API 2012-01-25 12:30:53 +01:00
Sebastian Dröge
241de164ee audioconvert: Fix channel-mask handling 2012-01-05 10:34:25 +01:00
Sebastian Dröge
5bdf6b3383 gst: Add new layout field to the raw audio caps 2012-01-05 10:34:25 +01:00
Sebastian Dröge
e0f9b4fffc audioconvert: Port to the new multichannel caps 
audioconvert still needs support for mixing all the new
channel positions, see:
https://bugzilla.gnome.org/show_bug.cgi?id=666506
 2012-01-05 10:34:19 +01:00
Wim Taymans
ff4efd075f audioconvert: handle unpositioned channels 
Refuse to convert between unpositioned layouts.
 2012-01-02 15:03:54 +01:00
Tim-Philipp Müller
177525f89f Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	gst-libs/gst/netbuffer/gstnetbuffer.c
	gst/ffmpegcolorspace/avcodec.h
	gst/ffmpegcolorspace/gstffmpegcodecmap.c
	gst/ffmpegcolorspace/imgconvert.c
	gst/ffmpegcolorspace/imgconvert_template.h
	gst/ffmpegcolorspace/mem.c
	gst/playback/README
	gst/playback/gstplaybasebin.c
	gst/playback/gstplaybasebin.h
	gst/playback/gstplaybin.c
	sys/v4l/v4lmjpegsrc_calls.c
	sys/v4l/videodev_mjpeg.h
	tests/check/elements/gnomevfssink.c
 2011-12-02 11:10:17 +00:00
Piotr Fusik
14644457b0 various: typo fixes 
Fix typos in code and docs. Fixes. #658984
 2011-12-02 12:03:27 +01:00
Vincent Penquerc'h
96374054ac various: fix pad template leaks 
https://bugzilla.gnome.org/show_bug.cgi?id=662664
 2011-11-28 13:09:02 +00:00
René Stadler
5f3c8eb680 audioconvert, videoconvert: fix caps leak in transform_caps 2011-11-12 01:38:37 +01:00
René Stadler
7651fa27dc audioconvert: fix leak of channel matrix 
gst_channel_mix_unset_matrix relies on the channel count to free the matrix
array, so run it before resetting it to zero with gst_audio_info_init.
 2011-11-11 20:19:53 +01:00
Vincent Penquerc'h
7ca4b51b01 audioconvert: truncate caps in _fixate 
Otherwise the resulting caps may not be fixed.
 2011-11-10 14:38:09 +00:00
Wim Taymans
016d036137 Merge branch 'master' into 0.11 
Conflicts:
	configure.ac
	gst-libs/gst/audio/gstbaseaudiosink.c
	gst/audioconvert/channelmixtest.c
	gst/playback/gstplaybasebin.c
	gst/playback/gstsubtitleoverlay.c
	tests/examples/Makefile.am
	tests/examples/audio/Makefile.am
 2011-10-27 15:44:58 +02:00
René Stadler
0a5fcbb080 audioconvert: bury dead test program 2011-10-21 22:24:14 +02:00
Edward Hervey
1c10fbcd33 audioconvert: We can handle channels conversion 2011-10-17 12:28:58 +02:00
Tim-Philipp Müller
754b22d7ee libs: remove unused floatcast header-only library 
There's no code whatsoever that uses these macros. If anyone
ever feels the need to resurrect them, we should add them to
gstutils.h in core or libgstaudio or so.
 2011-09-23 21:18:47 +01:00
Wim Taymans
c0d31dd555 rename IS_LE/BE to _IS_LITTLE_ENDIAN/BIG_ENDIAN 2011-09-06 16:46:02 +02:00
Wim Taymans
0a1874461a audio: rename UNPOSITIONED to DEFAULT_POSITIONS 
Rename the UNPOSITIONED flag to the DEFAULT_POSITIONS flag because that is
really what the resulting GstAudioInfo will contain as the chanel mappings.
 2011-08-24 14:13:33 +02:00
Wim Taymans
0213407fbc audio: rename INT -> INTEGER 
Spell INTEGER fully instead of using the int abreviation.
Remove some old functions.
 2011-08-20 10:49:17 +02:00
Wim Taymans
8023f49d19 more audio caps porting 2011-08-19 17:41:22 +02:00
Wim Taymans
dae848818d audio: rework audio caps. 
Rework the audio caps similar to the video caps. Remove
width/depth/endianness/signed fields and replace with a simple string
format and media type audio/x-raw.
Create a GstAudioInfo and some helper methods to parse caps.
Remove duplicate code from the ringbuffer and replace with audio info.
Use AudioInfo in the base audio filter class.
Port elements to new API.
 2011-08-18 19:15:03 +02:00
Josep Torra
5629ed74b3 Fix debug statements 
Fixes build on MacOSX

Signed-off-by: Edward Hervey <edward.hervey@collabora.co.uk>
 2011-08-10 11:15:41 +02:00
Wim Taymans
43bee0022a audioconvert: update orc dist files 2011-07-07 10:28:08 +02:00
Wim Taymans
d051f3cb5b audioconvert: don't use .init function 
Don't use the .init function but compile all functions when needed instead of
when the plugin is registered.
 2011-07-07 10:24:55 +02:00
Sebastian Dröge
4fcd621101 audioconvert: Use new gst_caps_is_subset_structure() API 
This prevents one copy of every structure and creating a new caps
instance.
 2011-05-27 14:10:50 +02:00
Sebastian Dröge
d590bce5f7 audioconvert: Optimize transform_caps() 
If the second and next caps structures are a subset of the already existing
transformed caps we can safely skip them because we would transform them to
the same caps again.

This makes gst_pad_get_caps() on an audiotestsrc ! audioconvert !
audioconvert ! audioconvert ! fakesink pipeline about 1.7 times faster.
 2011-05-27 13:13:42 +02:00
Sebastian Dröge
d8e0af1fc1 gst: Update for the GstBaseTransform::transform_caps() changes 2011-05-27 12:13:14 +02:00
Sebastian Dröge
a9b134d1a9 Merge branch 'master' into 0.11 
Conflicts:
	docs/plugins/gst-plugins-base-plugins.hierarchy
	docs/plugins/gst-plugins-base-plugins.interfaces
	docs/plugins/gst-plugins-base-plugins.prerequisites
 2011-05-20 12:26:57 +02:00
Stefan Kost
f514be993c audioconvert: cleanup helper code 
make_lossless_changes() returns the same structure that we're passing (probably
to enable chaining). Instead of reusing s and making it point to s2 as well,
keep using s2. Drop the assignment which in the 2nd case is a dead one anyway.
 2011-05-19 23:41:08 +03:00
Sebastian Dröge
c020add91e audioconvert: Update for negotiation related API changes 2011-05-16 15:35:40 +02:00
Wim Taymans
ec57868488 -base: don't use buffer caps 
Port to newest 0.11 core API, remove GST_PAD_CAPS and GST_BUFFER_CAPS.
 2011-05-09 13:05:12 +02:00
Sebastian Dröge
f10a8f0986 gst: Use G_DEFINE_TYPE instead of GST_BOILERPLATE 2011-04-19 11:35:53 +02:00
Sebastian Dröge
0759ce8533 Merge branch 'master' into 0.11 2011-04-18 13:23:32 +02:00
Tim-Philipp Müller
82a791519c gst: update disted orc backup code 2011-04-16 15:59:45 +01:00
Wim Taymans
6e160bed3d Merge branch 'master' into 0.11 
Conflicts:
	android/alsa.mk
	android/app.mk
	android/app_plugin.mk
	android/audio.mk
	android/audioconvert.mk
	android/decodebin.mk
	android/decodebin2.mk
	android/gdp.mk
	android/interfaces.mk
	android/netbuffer.mk
	android/pbutils.mk
	android/playbin.mk
	android/queue2.mk
	android/riff.mk
	android/rtp.mk
	android/rtsp.mk
	android/sdp.mk
	android/tag.mk
	android/tcp.mk
	android/typefindfunctions.mk
	android/video.mk
 2011-04-11 11:37:51 +02:00
Alessandro Decina
030f639a8e android: make it ready for androgenizer 
Remove the android/ top dir
Fixe the Makefile.am to be androgenized

To build gstreamer for android we are now using androgenizer which generates the
needed Android.mk files.
Androgenizer can be found here:
http://git.collabora.co.uk/?p=user/derek/androgenizer.git
 2011-04-11 07:23:21 +02:00
Wim Taymans
3b03e23559 plugins: port some plugins to the new memory API 2011-03-27 16:35:28 +02:00
Sebastian Dröge
19b9460e60 audioconvert: Update generated orc files 2010-10-08 00:01:15 +02:00
Sebastian Dröge
f5e9d8bb62 audioconvert: Implement remaining conversion functions from/to doubles to orc 
This requires orc 0.4.10
 2010-10-08 00:01:14 +02:00
David Schleef
bec69e20ae orc: update generated files to fix MSVC compile issues 2010-09-16 18:03:23 -07:00
Sebastian Dröge
18b282e49f orc: Fix generated source files 2010-09-10 08:43:17 +02:00
Sebastian Dröge
3c43dbfc51 orc: Update generated source files everywhere 2010-09-09 10:59:59 +02:00
Sebastian Dröge
8ba4b70118 Revert "Revert "Use init functions for Orc code"" 
This reverts commit 93aa13639d.

Everything should work now after regenerating the disted source files.
 2010-09-09 10:57:41 +02:00
Sebastian Dröge
65e5984634 audioconvert: Simplify float->s32 conversion 
orc 0.4.7 is doing saturated conversion from floats to integers
and it's not necessary to do this manually anymore.
 2010-09-05 12:57:36 +02:00
Sebastian Dröge
dd910ceaf4 audioconvert: Update disted orc files 2010-09-05 12:12:43 +02:00
Sebastian Dröge
24831973c0 audioconvert: Use the ORC double support 2010-09-05 12:09:42 +02:00
Wim Taymans
93aa13639d Revert "Use init functions for Orc code" 
This reverts commit b2051090b4.

Fixes the build again until someone pushes the regenerated .c/.h
files too.
 2010-08-27 11:49:47 +02:00
David Schleef
b2051090b4 Use init functions for Orc code 2010-08-26 17:03:13 -07:00
Sebastian Dröge
2ee9360cf6 audioconvert: Require ORC 0.4.7 for the loadl/storel opcodes 
And update disted files to allow compilation with no or too old ORC.
 2010-08-24 15:07:40 +02:00
Sebastian Dröge
5e0706c74f audioconvert: Use ORC for the float<->int32 conversion 
This should speed up standard Vorbis encoding and decoding pipelines a bit.

Thanks to David Schleef for the assistance to get the ORC code right
and explaining everything.
 2010-08-24 11:11:49 +02:00
Tim-Philipp Müller
b16e7e8fa2 gst: update orc files 2010-06-26 18:19:33 +01:00
David Schleef
d7f7e1cc23 audioconvert, videotestsrc: Update generated Orc code 
Fixes compile errors with initialization of unions.
 2010-06-08 00:33:31 -07:00
David Schleef
c49806ed16 audioconvert: convert from liboil to orc 2010-06-07 23:58:53 -07:00
Stefan Kost
4965782c48 audioconvert: disambigue comment due to popular demand 
Write "target depth" instead of "our depth" or previous ambigous "out depth".
 2010-05-07 00:10:22 +03:00
Stefan Kost
51739d562c audioconvert: fix typo in comment 2010-05-06 08:22:36 +03:00
Tim-Philipp Müller
b5f0b7c221 build: use LDADD instead of LDFLAGS to specify libs to link to when building executables 
Use foo_LDADD instead of foo_LDFLAGS to specify the libraries to link to.
This should make sure arguments are passed to the linker in the right
order, and makes LDFLAGS usable again.

Based on initial patch by Brian Cameron <brian.cameron@oracle.com>

Fixes #615697.
 2010-04-14 14:08:15 +01:00
Benjamin Otte
253d9acbcd Fix for -Wold-style-definition 
I didn't add the flag to configure because libvisual ships headers that
trigger this warning.
 2010-03-17 12:09:25 +01:00
Benjamin Otte
5e21fa5e0e gst_element_class_set_details => gst_element_class_set_details_simple 
Also change my email from the old university one to the current one.
 2010-03-16 17:41:50 +01:00
Benjamin Otte
3a7d632a59 Add -Wredundant-decls to warning flags 
... and fix all the warnings that flag throws.
 2010-03-11 15:38:18 +01:00
Stefan Kost
bbb531619c audioconvert: remove unused array 2009-11-16 22:51:17 +02:00
Stefan Kost
319baefeba audioconvert: track active conversion in perf log 2009-10-12 21:43:42 +03:00
Josep Torra
7bba1217a5 audioconvert: fixes warning: format not a string literal and no format arguments 
redo of valid part of my previous revert.
 2009-10-09 15:29:15 +02:00
Josep Torra
7b77138667 Revert "audioconvert: fixes warning: format not a string literal and no format arguments" 
Revert this commit as unintentionally I've changed common.

This reverts commit 49ea013822.
 2009-10-09 15:19:42 +02:00
Josep Torra
49ea013822 audioconvert: fixes warning: format not a string literal and no format arguments 2009-10-09 14:14:15 +02:00
Edward Hervey
8cd1b5209b gst: Remove dead assignments and resulting unused variables 2009-08-08 15:54:02 +02:00
Philip Jägenstedt
fa0a5a667f audioconvert: Fix compilation when debugging is disabled 
Fixes bug #587980.
 2009-07-08 15:08:32 +02:00
Stefan Kost
2cd4c7e2b9 Don't install static libs for plugins. Fixes #550851 for base. 
Original commit message from CVS:
* ext/alsa/Makefile.am:
* ext/cdparanoia/Makefile.am:
* ext/gio/Makefile.am:
* ext/gnomevfs/Makefile.am:
* ext/libvisual/Makefile.am:
* ext/ogg/Makefile.am:
* ext/pango/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst/adder/Makefile.am:
* gst/audioconvert/Makefile.am:
* gst/audiorate/Makefile.am:
* gst/audioresample/Makefile.am:
* gst/audiotestsrc/Makefile.am:
* gst/ffmpegcolorspace/Makefile.am:
* gst/gdp/Makefile.am:
* gst/playback/Makefile.am:
* gst/subparse/Makefile.am:
* gst/tcp/Makefile.am:
* gst/typefind/Makefile.am:
* gst/videorate/Makefile.am:
* gst/videoscale/Makefile.am:
* gst/videotestsrc/Makefile.am:
* gst/volume/Makefile.am:
* sys/v4l/Makefile.am:
* sys/ximage/Makefile.am:
* sys/xvimage/Makefile.am:
Don't install static libs for plugins. Fixes #550851 for base.
 2008-10-16 15:07:00 +00:00
Sebastian Dröge
c915582c17 gst/audioconvert/audioconvert.c: Prevent overflows with big buffer when calculating the size of the intermediate buff... 
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (audio_convert_convert):
Prevent overflows with big buffer when calculating the size of
the intermediate buffer by using gst_util_uint64_scale() instead of
plain arithmetics. Fixes bug #552801.
 2008-10-08 11:50:50 +00:00
Tim-Philipp Müller
58c48279dc gst/audioconvert/Makefile.am: Dist recently-added gstfastrandom.h. 
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
Dist recently-added gstfastrandom.h.
 2008-07-30 19:51:36 +00:00
Sebastian Dröge
ef5004e56e gst/audioconvert/: Implement a linear congruential generator as pseudo random number generator for the dither noise. ... 
Original commit message from CVS:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither):
* gst/audioconvert/gstfastrandom.h:
Implement a linear congruential generator as pseudo random number
generator for the dither noise. This is about 2 times faster than
using GLib's mersenne twister. Also this uses only integer math for
generating integers while GLib internally uses floating point math.
 2008-07-23 18:34:19 +00:00
Stefan Kost
8b24a3a057 gst/audioconvert/gstaudioconvert.c: Remove now obsolete note in the docs. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Remove now obsolete note in the docs.
 2008-07-11 18:06:33 +00:00
Stefan Kost
2b33c755b6 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe... 
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* 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/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-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-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/alsa/gstalsamixer.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosrc.c:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/pango/gsttimeoverlay.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/theora/theoraparse.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* ext/vorbis/vorbisparse.c:
* ext/vorbis/vorbistag.c:
* gst/adder/gstadder.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gstqueue2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpserversink.c:
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
Cleanup Plugin docs. Link to signals and properties. Fix sub-section
titles. Drop mentining that all our example pipelines are "simple"
pipelines.
 2008-07-10 21:06:06 +00:00
Sebastian Dröge
fdd708c418 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.
 2008-05-30 08:42:17 +00:00