Commit graph

60 commits

Author SHA1 Message Date
Tim-Philipp Müller
4b0dce5148 Merge remote-tracking branch 'origin/master' into 0.11
Conflicts:
	gst-libs/gst/audio/Makefile.am
	gst-libs/gst/audio/audio.h
	tests/examples/seek/jsseek.c
	tests/examples/seek/seek.c
	tests/icles/test-colorkey.c
2011-11-13 13:36:29 +00:00
Tim-Philipp Müller
cd21e69913 audio: add GST_AUDIO_INFO_IS_VALID macro and use in audio decoder base class
API: GST_AUDIO_INFO_IS_VALID
2011-11-13 13:18:16 +00:00
Wim Taymans
c42e257751 audio: fix docs 2011-11-11 19:13:52 +01:00
Wim Taymans
b645287775 audio: fix headers
Add const to some methods.
Add padding.
Add GType for GstAudioInfo and GstAudioFormatInfo.
Add new/copy/free for GstAudioInfo.
2011-11-11 17:53:03 +01:00
Wim Taymans
4bf9022e0c docs: improve docs 2011-09-27 11:19:24 +02:00
Edward Hervey
17bfba09f1 Merge branch 'master' into 0.11
Conflicts:
	ext/ogg/gstoggdemux.c
	ext/pango/gsttextoverlay.c
	gst-libs/gst/audio/gstaudioencoder.c
	gst-libs/gst/audio/gstbaseaudiosrc.c
	gst/playback/gstsubtitleoverlay.c
	gst/videorate/gstvideorate.c
2011-09-23 18:27:11 +02:00
Mark Nauwelaerts
7fa7de9221 audio: some more accessor macros for GstAudioInfo 2011-09-22 15:45:05 +02:00
Tim-Philipp Müller
4529c6dc32 Merge remote-tracking branch 'origin/master' into 0.11
Merge in doc updates for audio enums from 0.10, and get rid
of the #if #else in the enum list, since that confuses gtk-doc.

Conflicts:
	gst-libs/gst/audio/audio.c
	gst-libs/gst/audio/audio.h
2011-09-06 16:42:42 +01:00
Wim Taymans
dc28bd1b63 audio: rename IS_LE/BE to _IS_LITTLE_ENDIAN/BIG_ENDIAN 2011-09-06 16:27:27 +01:00
Wim Taymans
f04b8fd8af audio/video add descriptions
Add a description to the audio and video format info in case we want to use this
later.
2011-09-06 16:46:48 +02:00
Wim Taymans
c0d31dd555 rename IS_LE/BE to _IS_LITTLE_ENDIAN/BIG_ENDIAN 2011-09-06 16:46:02 +02:00
Tim-Philipp Müller
91d1112360 audio: update audio format enums to match changes in 0.11
And add new audio format info stuff to docs.
2011-09-06 15:36:51 +01:00
Wim Taymans
7012e88090 Merge branch 'master' into 0.11
Conflicts:
	gst-libs/gst/audio/audio.h
	gst-libs/gst/audio/gstaudiodecoder.c
	gst-libs/gst/audio/gstaudiodecoder.h
	gst-libs/gst/audio/gstaudioencoder.c
	gst-libs/gst/audio/gstbaseaudioencoder.h
	gst/playback/Makefile.am
	gst/playback/gstplaybin.c
	gst/playback/gstplaysink.c
	gst/playback/gstplaysinkvideoconvert.c
	gst/playback/gstsubtitleoverlay.c
	gst/videorate/gstvideorate.c
	gst/videoscale/gstvideoscale.c
	win32/common/libgstaudio.def
2011-09-06 15:24:32 +02:00
Wim Taymans
33196cdd2c audio: change audio format syntax a little
Remove the _ in front of the endianness prefix.
Remove the _3 postfix for the 24 bits formats.
Add a _32 postfix after the formats that occupy extra space beyond their
natural size.
The result is that the GST_AUDIO_NE() macro can simply append the endianness
after all formats and that we only specify a different sample width when it is
different from the natural size of the sample. This makes things more consistent
and follows the pulseaudio conventions instead of the alsa ones.
2011-09-06 12:06:39 +02:00
Tim-Philipp Müller
5e61db25b5 audio: fix GST_AUDIO_FORMAT_INFO_IS_*() macros to return a boolean 2011-09-05 23:28:20 +01:00
Tim-Philipp Müller
946ddb6462 audio: add gst_audio_info_{init,clear} and gst_audio_info_{copy,free} 2011-08-27 14:47:49 +01:00
Tim-Philipp Müller
63a3d360dc audio: add GstAudioFormat, GstAudioFormatInfo and GstAudioInfo
Same as in 0.11, but with caps parsing/serialising for 0.10 style
caps. Add setting default channel positions.
2011-08-27 14:47:01 +01:00
Wim Taymans
24ea19935f audio/video: add format of the pack functions
Replace the unpack_size with an unpack_format, which is more descriptive of the
kind of data the unpack function will create.
2011-08-24 16:40:43 +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
c6758ecfa9 audio: move function to convert 2011-08-22 16:11:27 +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
7db6fa37b4 audio: add function to build audio format 2011-08-19 16:00:33 +02:00
Wim Taymans
17dd31b0f4 audio: add more macros 2011-08-19 14:03:23 +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
Wim Taymans
d6740006d4 audio: remove deprecated methods 2011-08-16 16:59:15 +02:00
Tim-Philipp Müller
e836151009 docs: more helper libraries docs fixes
Quieten gtk-doc a bit more.
2010-03-16 00:44:50 +00:00
Sebastian Dröge
6dfc0270ec audio: Use rounding scaling functions for GST_CLOCK_TIME_TO_FRAMES and _FRAMES_TO_CLOCK_TIME
Fixes bug #607381.
2010-01-19 09:26:37 +01:00
Tim-Philipp Müller
b579580991 gst-libs/gst/audio/audio.h: Remove trailing comma from enum list, which causes problems with -pendantic (#550729).
Original commit message from CVS:
* gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
Remove trailing comma from enum list, which causes problems
with -pendantic (#550729).
2008-09-13 11:04:02 +00:00
Stefan Kost
f73aa5b817 gst-libs/gst/: Reducing number of dundocumented symbols.
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/interfaces/propertyprobe.h:
* gst-libs/gst/tag/gsttagdemux.h:
Reducing number of dundocumented symbols.
2008-08-11 08:34:56 +00:00
Stefan Kost
28b46c1e5d gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
Readd the deprecation guards, but preserve compilability.
2007-11-01 08:06:13 +00:00
Tim-Philipp Müller
55a3eaafea gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
(ie. normal cvs builds) will fail.
2007-10-31 15:30:15 +00:00
Stefan Kost
e37568c196 tell gtk-doc about the deprecation guard. Apply more doc fixes.
Original commit message from CVS:
* docs/libs/Makefile.am:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/interfaces/mixer.c:
tell gtk-doc about the deprecation guard. Apply more doc fixes.
2007-10-31 12:47:41 +00:00
Sebastian Dröge
846ddaa550 gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
Use gst_util_uint64_scale() instead of doing the math
with double for GST_FRAMES_TO_CLOCK_TIME() and
GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
prevents rounding errors. Fixes #467667.
2007-08-17 15:24:43 +00:00
Sebastian Dröge
6be2524031 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
* gst-libs/gst/audio/audio.h:
* tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
API: Add buffer clipping function for raw audio buffers. Fixes #456656.
Also add deprecation guards for gst_audio_structure_set_int() to the
header.
2007-07-23 18:26:09 +00:00
Stefan Kost
b2f9c0f289 More docs coverage and some ChangeLog surgery (add missing names)
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.h:
* ext/ogg/gstoggdemux.h:
* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
(gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
(gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/interfaces/videoorientation.h:
* gst/adder/gstadder.h:
More docs coverage and some ChangeLog surgery (add missing names)
2007-02-15 15:17:23 +00:00
Stefan Kost
7ee1b714f0 Add crossreferences to glib/gobject/gstream docs.
Original commit message from CVS:
* configure.ac:
* docs/libs/Makefile.am:
* docs/plugins/Makefile.am:
Add crossreferences to glib/gobject/gstream docs.
* gst-libs/gst/audio/audio.h:
Source formatting.
* gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
Add own debug category.
2007-02-12 20:42:23 +00:00
Sébastien Moutte
dc46970cdf gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
Original commit message from CVS:
* gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
* gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
use of gst_guint64_to_gdouble to be compliant with vs6
* gst/playback/gstdecodebin.c: (try_to_link_1)
* gst/videorate/videorate.c: (gst_video_rate_blank_data)
use of G_GINT64_CONSTANT for int64 constants
* win32/common/libgstinterfaces.def:
export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
* win32/vs6:
update and add new project files
2006-01-29 19:13:39 +00:00
Thomas Vander Stichele
08cd3b973f remove some deprecated functions
Original commit message from CVS:
remove some deprecated functions
2005-11-22 13:14:07 +00:00
Thomas Vander Stichele
1c3b6d42a9 gst-libs/gst/audio/audio.*: fix prototype - wondering why the test worked regardless
Original commit message from CVS:

* gst-libs/gst/audio/audio.c: (gst_audio_duration_from_pad_buffer):
* gst-libs/gst/audio/audio.h:
fix prototype - wondering why the test worked regardless
2005-11-21 23:51:45 +00:00
Thomas Vander Stichele
be5a7cd625 add a method that returns a proper GstClockTime
Original commit message from CVS:
add a method that returns a proper GstClockTime
2005-11-21 22:56:33 +00:00
Wim Taymans
fc8ce00673 Bye bye buffer-frames.
Original commit message from CVS:
* check/elements/audioconvert.c:
* docs/libs/tmpl/gstaudio.sgml:
* docs/libs/tmpl/gstcolorbalance.sgml:
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_handle_identification_packet), (vorbis_handle_data_packet):
* ext/vorbis/vorbisenc.c: (raw_caps_factory):
* gst-libs/gst/audio/audio.c: (gst_audio_structure_set_int):
* gst-libs/gst/audio/audio.h:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps):
* gst/volume/gstvolume.c:
Bye bye buffer-frames.
2005-10-19 17:02:46 +00:00
Thomas Vander Stichele
272aad79bb add/fix 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:
* gst/volume/gstvolume.c:
add/fix docs
* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size):
* gst-libs/gst/audio/audio.h:
add conversion macros for frames <-> clocktime
2005-09-23 18:14:54 +00:00
Thomas Vander Stichele
9e8a11d3ce use overridable ERROR_CFLAGS; more macro splitting
Original commit message from CVS:
use overridable ERROR_CFLAGS; more macro splitting
2005-07-10 12:03:58 +00:00
Wim Taymans
235ea5989c Make ringbuffer faster and more simple by removing the locks in the playback thread.
Original commit message from CVS:
Make ringbuffer faster and more simple by removing the locks
in the playback thread.
Add sample accurate playback based on buffer sample offsets.
Make the baseaudiosink provide a clock.
Parse caps in the base class.
Correctly handle seeking, flushing and state changes.
2005-04-28 16:15:42 +00:00
Benjamin Otte
37af33bdda ext/alsa/gstalsa.c: buffer-frames property was missing
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps_internal):
buffer-frames property was missing
* ext/arts/gst_arts.c:
rate missing from sinkcaps
* ext/audiofile/gstafparse.c:
* ext/audiofile/gstafsink.c:
* ext/audiofile/gstafsrc.c:
* ext/swfdec/gstswfdec.c:
int audio doesn't know buffer-frames
* ext/cdparanoia/gstcdparanoia.c:
int audio doesn't know chunksize either
* ext/nas/nassink.c:
it's endianness, not endianess
* gst-libs/gst/audio/audio.h:
make float standard pad template caps really describe float
* gst/law/mulaw.c: (linear_factory):
signed only, please
* gst/mpegstream/gstdvddemux.c:
widths of 20 are not valid
2004-11-09 06:08:22 +00:00
Stéphane Loeuillet
2800919549 gst/auparse/gstauparse.c: eee (32, 64) only unsupported formats are ADPCM/CCITT G.72x
Original commit message from CVS:
* gst/auparse/gstauparse.c:
fixes a-law, adds mu-law, linear pcm (8,16,24,32), ieee (32, 64)
only unsupported formats are ADPCM/CCITT G.72x
reviewed by Ronald
* gst-libs/gst/audio/audio.h:
adds 24bit depth to PCM (x-raw-int)
2004-05-10 23:55:17 +00:00
Johan Dahlin
bf07c7cc9b *.h: Revert indenting
Original commit message from CVS:
* *.h: Revert indenting
2004-03-15 16:32:55 +00:00
Thomas Vander Stichele
4df3f18839 gst-indent
Original commit message from CVS:
gst-indent
2004-03-14 22:34:34 +00:00
Benjamin Otte
e4a3aa6fba gst-libs/gst/audio/audio.h: add macro to make sure header isn't included twice
Original commit message from CVS:
2004-02-27  Benjamin Otte  <otte@gnome.org>

* gst-libs/gst/audio/audio.h:
add macro to make sure header isn't included twice
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_chunk):
don't use gst_buffer_free
* gst/playondemand/filter.func:
don't usae gst_data_free. Free data only once.
2004-02-27 21:47:49 +00:00
Benjamin Otte
b552a2d9f0 gst-libs/gst/audio/audio.h: remove buffer-frames from audio caps
Original commit message from CVS:
2004-01-26  Benjamin Otte  <in7y118@public.uni-hamburg.de>

* gst-libs/gst/audio/audio.h:
remove buffer-frames from audio caps
* gst/audioconvert/gstaudioconvert.c:
fix plugin to really work.
2004-01-26 03:54:21 +00:00