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136 commits

Author SHA1 Message Date
Tim-Philipp Müller
5ee51e47a1 ext, gst, gst-libs, tests: update for tag list API changes 2011-10-31 14:22:39 +00:00
Tim-Philipp Müller
a586547b0c audiotestsrc: fix crash when setting the wave property before having negotiated a format 
https://bugzilla.gnome.org/show_bug.cgi?id=661911
 2011-10-17 15:47:31 +01:00
Thiago Santos
6eb5f5b13e audiotestsrc: update blocksize when caps or samples-per-buffer change 
Blocksize needs to be updated so we get a correct size buffer on
_fill function.
 2011-10-10 12:31:46 -03:00
Wim Taymans
f1088ed647 update for UNEXPECTED -> EOS flowreturn 2011-10-10 11:39:52 +02:00
Wim Taymans
73b894107a Merge branch 'master' into 0.11 
Conflicts:
	ext/vorbis/gstvorbisdec.c
	ext/vorbis/gstvorbisenc.c
	ext/vorbis/gstvorbisenc.h
	gst/audiotestsrc/gstaudiotestsrc.c
 2011-10-08 10:19:06 +02:00
Vincent Penquerc'h
70239887e8 audiotestsrc: add missing break 
And make violet noise usable

https://bugzilla.gnome.org/show_bug.cgi?id=661105
 2011-10-06 20:45:09 +02:00
Stefan Sauer
7ce811f1ed auditestsrc: indent fix 2011-10-04 23:10:05 +02:00
Sebastian Dröge
0f654f3feb Merge branch 'master' into 0.11 
Conflicts:
	docs/libs/Makefile.am
	tests/check/elements/decodebin2.c
 2011-09-08 14:42:00 +02:00
Stefan Sauer
abc96efb2a docs: add two mising enum docs 2011-09-07 14:14:02 +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
Wim Taymans
81457756f0 audiotestsrc: use base class fill method 2011-08-25 13:21:14 +02:00
Wim Taymans
b0b6d9124d audiotestsrc: fix build 2011-08-24 11:05:05 +02:00
Wim Taymans
2ce5c8b8be audio: use convert audio helper 2011-08-22 16:21:02 +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
0290df6fc5 audiotestsrc: properly override fixate 2011-08-17 17:22:03 +02:00
Tim-Philipp Müller
dd56714b14 ffmpegcolorspace -> videoconvert 2011-07-07 23:59:59 +01:00
Wim Taymans
40d567153a Merge branch 'master' into 0.11 2011-06-13 19:09:05 +02:00
David Schleef
4db89c82bb convert M_PI to G_PI, for msvc 2011-06-10 23:56:34 -07:00
Sebastian Dröge
bf08ca7020 Merge branch 'master' into 0.11 2011-05-26 13:54:09 +02:00
Stefan Kost
5cd0e0f666 audiotestsrc: add blue and violet noise by using spectral inversion 
Add blue and violet noise by spectral inversion of pink and red noise.
Fixes #649969
 2011-05-26 00:18:55 +03:00
Stefan Kost
1cf831e74e audiotestsrc: add red (brownian) noise generator 
Add another noise generator which produces a quite dark noise color.

Fixes parts of #649969.
 2011-05-25 23:43:56 +03:00
Wim Taymans
010add200a scheduling: port to new scheduling query 2011-05-24 17:37:45 +02:00
Sebastian Dröge
318ed07598 Revert "-base_port to new query API" 
This reverts commit c9f4e0676b.
 2011-05-17 11:25:31 +02:00
Wim Taymans
94dfe80f71 -base: port to new SEGMENT API 2011-05-16 13:48:11 +02:00
Wim Taymans
c9f4e0676b -base_port to new query API 2011-05-10 18:39:07 +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
Wim Taymans
86a4771f8e remove buffer_alloc 2011-04-29 13:28:17 +02:00
Sebastian Dröge
f10a8f0986 gst: Use G_DEFINE_TYPE instead of GST_BOILERPLATE 2011-04-19 11:35:53 +02:00
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
Leo Singer
82199c5815 audiotestsrc: each element gets its own instance of GRand, if needed 
As a result, pipelines that contain multiple instances of audiotestsrc
with the 'wave' property set to 'white-noise', 'pink-noise', or
'gaussian-noise' will run much faster, since they won't be competing
for access to the global, lock-protected instance of GRand.

Fixes bug #642720.
 2011-02-19 08:37:46 +01:00
Stefan Kost
45b39fcfc1 audiotestsrc: swap timestamps in forward and reverse mode. 
In reverse mode we want use the next next timestamp (and not the other way
around). Fixes the tests again. Also readd a log line that was dropped with
previous commit.
 2010-04-03 22:52:01 +03:00
Stefan Kost
718edb5c14 audiotestsrc: implement reverse playback 
Support playback at negative rates. When having a GstController assigned, the
element will produce time dependend output.
 2010-04-02 21:04:37 +03:00
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
Wim Taymans
0bb9b75a75 audiotestsrc: call send_event directly 
We can't call gst_element_send_event() from a streaming thread as it gets the
state lock. Instead call the send_event method directly until we have a nice API
for this in basesrc.

Fixes #588746
 2009-07-20 13:15:32 +02:00
Edward Hervey
196b38d4ef audiotestsrc: Make sure tags are properly serialized. Fixes #588746 
We do this by letting the basesrc base class handle the tags.
 2009-07-20 08:47:50 +02:00
Sebastian Dröge
dc706f7f2f audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian" 
Also make all the function arrays constant.
 2009-06-21 19:43:18 +02:00
Kipp Cannon
620391b300 audiotestsrc: Add support for generating gaussian white noise 
This patch adds support for stationary white Gaussian noise.
The Box-Muller algorithm is used to generate pairs of independent
normally-distributed random numbers.

Fixes bug #586519.
 2009-06-21 12:29:03 +02:00
Tim-Philipp Müller
8d326479a5 audiotestsrc: fix broken enum nick - it should have a hyphen 
The enum nick should be 'sine-table', not 'sine table'. Technically this is
an API/ABI change I guess, but anyone who was using this and didn't report
it deserves this.
 2009-05-12 17:18:37 +01:00
Tim-Philipp Müller
21228a6934 audiotestsrc: seek to the requested byte offset, not the expected byte offset 2009-05-12 15:32:02 +01:00
Tim-Philipp Müller
72c5884f4a audiotestsrc: support more than just one channel 2009-05-12 15:32:02 +01:00
Wim Taymans
c3ec18af97 gst/audiotestsrc/gstaudiotestsrc.*: Add property to control pull/push based scheduling. 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_check_get_range),
(gst_audio_test_src_set_property),
(gst_audio_test_src_get_property):
* gst/audiotestsrc/gstaudiotestsrc.h:
Add property to control pull/push based scheduling.
 2009-01-05 10:59:35 +00:00
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
Wim Taymans
5ad1ebcf4c gst/audiotestsrc/gstaudiotestsrc.c: Set the default blocksize to -1 because we will then use the configured samplespe... 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init):
Set the default blocksize to -1 because we will then use the configured
samplesperbuffer to create our output buffer.
 2008-10-16 13:50:00 +00:00
Wim Taymans
81f5117fa9 gst/audiotestsrc/gstaudiotestsrc.*: Define the default property values in the usual place. 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
(gst_audio_test_src_start), (gst_audio_test_src_stop),
(gst_audio_test_src_do_seek), (gst_audio_test_src_check_get_range),
(gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Define the default property values in the usual place.
Implement start/stop to reset values correctly.
Calculate the sample size only once when we negotiate.
Rename some values to make more sense.
Keep track of our byte range.
Add support for pull based scheduling. Disabled for now until we have
the whole stack working.
Set the BUFFER_OFFSET correctly.
 2008-10-10 15:45:15 +00:00
Andy Wingo
79930b61bf gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important documentation fix. 
Original commit message from CVS:
2008-08-04  Andy Wingo  <wingo@pobox.com>

* gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
documentation fix.
 2008-08-04 09:11:08 +00:00
David Schleef
cc74285d12 gst/audiotestsrc/gstaudiotestsrc.*: Add 'ticks', a 1/30 second sine wave pulse every second. 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/audiotestsrc/gstaudiotestsrc.h:
Add 'ticks', a 1/30 second sine wave pulse every second.
 2008-07-17 02:30:24 +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
Stefan Kost
e6528c39fe gst/audiotestsrc/gstaudiotestsrc.c: Remove cpp style commented old code. 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
Remove cpp style commented old code.
 2008-04-15 19:06:00 +00:00
Sebastian Dröge
49deb0c05d Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... 
Original commit message from CVS:
* configure.ac:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_class_init):
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_class_init):
* ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
* ext/pango/gsttextrender.c: (gst_text_render_class_init):
* ext/theora/theoradec.c: (gst_theora_dec_class_init):
* ext/theora/theoraenc.c: (gst_theora_enc_class_init):
* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_class_init):
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init):
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(preroll_unlinked):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
* gst/playback/gstplaysink.c: (gst_play_sink_class_init):
* gst/playback/gstqueue2.c: (gst_queue_class_init):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
(gst_stream_selector_class_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init):
* gst/volume/gstvolume.c: (gst_volume_class_init):
* sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
static strings (i.e. all). This gives us less memory usage,
fewer allocations and thus less memory defragmentation. Depend
on core CVS for this. Fixes bug #523806.
 2008-03-22 15:00:53 +00:00
Stefan Kost
7278c5871c gst/audiotestsrc/gstaudiotestsrc.c: Mark buffers as GAP,if volume is 0.0 and fix the previous logic. 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
 2008-02-21 08:05:10 +00:00
Stefan Kost
1cfef609d0 gst/: Add GAP-flag support. 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/volume/gstvolume.c:
* gst/volume/gstvolume.h:
Add GAP-flag support.
 2007-11-26 12:25:55 +00:00
Sebastian Dröge
5310373def gst/audiotestsrc/gstaudiotestsrc.c: Properly respond to conversion queries. Fixes #464079. 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
Properly respond to conversion queries. Fixes #464079.
 2007-08-06 16:42:22 +00:00
Sebastian Dröge
6f397125d1 gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422. 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
(gst_audio_test_src_init_sine_table),
(gst_audio_test_src_change_wave), (gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Add float/double and int32 support to audiotestsrc. Fixes #460422.
Also set the default volume to the default value specified in the
GParamSpec.
 2007-08-03 19:53:11 +00:00
Michael Smith
6499fcdc2e gst/audiotestsrc/gstaudiotestsrc.c: Don't overflow intermediate values when seeking to large time values in audiotest... 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
Don't overflow intermediate values when seeking to large time values
in audiotestsrc.
 2007-06-05 17:08:04 +00:00
Michael Smith
ab76fa091a gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta... 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
Use the segment->last_stop value to calculate the next timestamp to
generate after a seek; not the segment->start value.
 2007-05-17 11:16:14 +00:00
Stefan Kost
64a9674bd2 gst/: gst/audiotestsrc/gstaudiotestsrc.c 
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst/adder/gstadder.c:
* gst/audiotestsrc/gstaudiotestsrc.c
(gst_audio_test_src_create_white_noise):
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
volume_sink_template, volume_src_template, gst_volume_init,
volume_process_double, volume_process_int16,
volume_process_int16_clamp):
Doc fixes and formatting.
 2007-05-04 13:10:07 +00:00
Loïc Minier
2770d0fbea Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963). 
Original commit message from CVS:
Patch by: Loïc Minier <lool+gnome at via ecp fr>
* configure.ac:
* ext/alsa/Makefile.am:
* gst/audiotestsrc/Makefile.am:
Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
 2007-02-24 20:12:49 +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
Stefan Kost
fae581d596 gst/: remove obsolete _factory_init protos 
Original commit message from CVS:
* gst/adder/gstadder.h:
* gst/audiotestsrc/gstaudiotestsrc.h:
remove obsolete _factory_init protos
 2006-11-24 08:56:10 +00:00
Stefan Kost
267a068e70 ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection with gnomevfssrc 
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c:
Add docs about icydemux usage in connection with gnomevfssrc
* ext/libvisual/visual.c:
* ext/ogg/gstoggaviparse.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst/audiorate/gstaudiorate.c:
More G_OBJECT macro fixing.
* gst/audiotestsrc/gstaudiotestsrc.h:
Fix wrong info in header due to copy & paste
 2006-09-16 21:54:48 +00:00
Stefan Kost
131fb86b4b Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass 
Original commit message from CVS:
* ext/alsa/gstalsamixerelement.h:
* ext/alsa/gstalsamixeroptions.h:
* ext/alsa/gstalsamixertrack.h:
* ext/gnomevfs/gstgnomevfssink.h:
* ext/gnomevfs/gstgnomevfssrc.h:
* ext/theora/gsttheoradec.h:
* ext/theora/gsttheoraenc.h:
* ext/theora/gsttheoraparse.h:
* ext/vorbis/vorbisparse.h:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/audioresample/gstaudioresample.h:
* gst/audiotestsrc/gstaudiotestsrc.h:
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
* gst/playback/gststreamselector.h:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcpclientsink.h:
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpserversink.h:
* gst/tcp/gsttcpserversrc.h:
* gst/videorate/gstvideorate.h:
* gst/videoscale/gstvideoscale.h:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/volume/gstvolume.h:
* sys/v4l/gstv4ljpegsrc.h:
* sys/v4l/gstv4lmjpegsink.h:
* sys/v4l/gstv4lmjpegsrc.h:
* sys/v4l/gstv4lsrc.h:
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.h:
* tests/old/testsuite/alsa/sinesrc.h:
Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
 2006-06-01 19:19:51 +00:00
Tim-Philipp Müller
d8965c30fb Const-ify GEnumValue and GFlagsValue arrays. Use 
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
(gst_text_overlay_halign_get_type),
(gst_text_overlay_wrap_mode_get_type):
* ext/theora/theoradec.c: (theora_handle_type_packet),
(theora_handle_data_packet):
* ext/theora/theoraenc.c: (gst_border_mode_get_type),
(theora_enc_sink_setcaps), (theora_enc_chain):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_mode_get_type):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type):
* gst/playback/gststreaminfo.c: (gst_stream_type_get_type):
* gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
* gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
(gst_sync_method_get_type), (gst_unit_type_get_type),
(gst_client_status_get_type):
* gst/videoscale/gstvideoscale.c:
(gst_video_scale_method_get_type):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_pattern_get_type):
* gst/videotestsrc/videotestsrc.c: (paint_setup_I420),
(paint_setup_YV12), (paint_setup_YUY2), (paint_setup_UYVY),
(paint_setup_YVYU), (paint_setup_IYU2), (paint_setup_Y41B),
(paint_setup_Y42B), (paint_setup_Y800), (paint_setup_YVU9),
(paint_setup_YUV9), (paint_setup_RGB888), (paint_setup_BGR888),
(paint_setup_RGB565), (paint_setup_xRGB1555):
Const-ify GEnumValue and GFlagsValue arrays. Use
GST_ROUND_UP_* macros instead of home-made ones.
 2006-05-09 19:24:46 +00:00
Stefan Kost
e972defd3e make GstElementDetails const 
Original commit message from CVS:
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsasrc.c:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* ext/ogg/gstogmparse.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/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiorate/gstaudiorate.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
* gst/subparse/gstsubparse.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/v4l/gstv4ljpegsrc.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* sys/v4l/gstv4lsrc.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
* tests/check/libs/cddabasesrc.c:
make GstElementDetails const
 2006-04-28 19:46:37 +00:00
Andy Wingo
a8e9a6d7a1 gst/videorate/gstvideorate.c (gst_video_rate_reset) 
Original commit message from CVS:
2006-04-06  Andy Wingo  <wingo@pobox.com>

* gst/videorate/gstvideorate.c (gst_video_rate_reset)
(gst_video_rate_init): Caps-related parameters should not be reset
by a flush -- move their inits to the instance init function.
(gst_video_rate_flush_prev): Don't complain if gst_pad_push
is not OK, just return the result.

* gst/audiotestsrc/gstaudiotestsrc.c
(gst_audio_test_src_class_init)
(gst_audio_test_src_get_times): Re-enable is-live=true, as was
broken by Stefan's commit on 24 March.
 2006-04-06 11:40:45 +00:00
Thomas Vander Stichele
d02bbc0fb5 gst/audiotestsrc/gstaudiotestsrc.c: Fixed the sample pipeline (see #323798) 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
Fixed the sample pipeline (see #323798)
 2006-04-01 11:21:30 +00:00
Stefan Kost
2d826700fa Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top 
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/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
(gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_base_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_base_init):
* gst/adder/gstadder.c: (gst_adder_get_type):
* gst/adder/gstadder.h:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
(gst_audio_test_src_create):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
(gst_stream_selector_base_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
* gst/volume/gstvolume.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* tests/check/libs/cddabasesrc.c:
* tests/old/examples/gob/gst-identity2.gob:
Add docs for adder, use GST_ELEMENT_DETAILS macro,
define GstElementDetails at the top
 2006-03-24 10:42:11 +00:00
Wim Taymans
81e72d14b4 Updated/added documentation. 
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/pango/gstclockoverlay.h:
* ext/pango/gsttextoverlay.h:
* ext/pango/gsttextrender.h:
* ext/pango/gsttimeoverlay.h:
* ext/theora/gsttheoradec.h:
* ext/theora/gsttheoraenc.h:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* gst/audioconvert/gstaudioconvert.h:
* gst/audiotestsrc/gstaudiotestsrc.h:
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
* gst/tcp/gstmultifdsink.h:
Updated/added documentation.
* ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
(gst_text_overlay_halign_get_type),
(gst_text_overlay_wrap_mode_get_type),
(gst_text_overlay_base_init), (gst_text_overlay_class_init),
(gst_text_overlay_init), (gst_text_overlay_set_property),
(gst_text_overlay_get_property):
Fix up properties to be enums instead of string to make bindings,
introspection and automatic GUI creation possible.
Add getters for the properties.
 2006-03-01 16:24:37 +00:00
Sébastien Moutte
81311ccfc1 gst/audiotestsrc/gstaudiotestsrc.c: added defines of M_PI and M_PI_2 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
added defines of M_PI and M_PI_2
* gst/ffmpegcolorspace/avcodec.h:
removed #include "stdint.h" for win32 as _stdint.h is
autogenerated to win32/common
* win32/common/libgstaudio.def:
* win32/common/libgsttag.def:
added some exports
* win32/vs6:
some project files bugs corrected
* win32/vs7:
project files are reset to the default vs7 configuration
(they link to msvcr71.dll using default optimizations)
 2006-02-28 21:21:07 +00:00
Tim-Philipp Müller
51ce1f6179 gst/: Pass unhandled queries to the parent class's query function. 
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_query),
(gst_cdda_base_src_handle_event):
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
Pass unhandled queries to the parent class's query function.
 2006-02-01 11:59:47 +00:00
Stefan Kost
b5398e7a9d gst/: initialize gst_controller before using 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_create_sine_table), (plugin_init):
* gst/volume/gstvolume.c: (plugin_init):
initialize gst_controller before using
 2006-01-31 17:19:09 +00:00
Stefan Kost
0d85e2cc15 gst/audiotestsrc/gstaudiotestsrc.*: update to basesrc changes, implement segmented seeking and eos handling, add a 's... 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
(gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
(gst_audio_test_src_query), (gst_audio_test_src_create_sine),
(gst_audio_test_src_create_square),
(gst_audio_test_src_create_saw),
(gst_audio_test_src_create_triangle),
(gst_audio_test_src_create_silence),
(gst_audio_test_src_create_white_noise),
(gst_audio_test_src_create_pink_noise),
(gst_audio_test_src_init_sine_table),
(gst_audio_test_src_create_sine_table),
(gst_audio_test_src_change_wave),
(gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
(gst_audio_test_src_create), (gst_audio_test_src_set_property):
* gst/audiotestsrc/gstaudiotestsrc.h:
update to basesrc changes, implement segmented seeking and eos handling,
add a 'sine-tab' waveform for performance critical playback
 2005-12-29 20:37:23 +00:00
Stefan Kost
90736bc6e7 ext/libvisual/visual.c: change some char* into char[] 
Original commit message from CVS:
* ext/libvisual/visual.c: (make_valid_name):
change some char* into char[]
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_do_seek),
(gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
prepare to handle EOS and SEGMENT_DONE
 2005-12-16 22:00:07 +00:00
Stefan Kost
7906b3c945 gst/audiotestsrc/gstaudiotestsrc.*: Adjust to some recent api changes and add wtays new cool seeking capabillities 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
(gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
(gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Adjust to some recent api changes and add wtays new cool seeking
capabillities
 2005-12-14 20:42:11 +00:00
Thomas Vander Stichele
f39b477379 borgify further clean up docs a little 
Original commit message from CVS:
borgify further
clean up docs a little
 2005-12-01 01:12:55 +00:00
Thomas Vander Stichele
c0759093eb remove deprecated properties fix up enums to correctly have short lowercase dashed nicks 
Original commit message from CVS:
remove deprecated properties
fix up enums to correctly have short lowercase dashed nicks
 2005-11-22 17:39:29 +00:00
Jan Schmidt
1cc82e9138 Rename gst_caps_structure_fixate_* to gst_structure_fixate_* (#322027) 
Original commit message from CVS:
* ext/libvisual/visual.c: (get_buffer):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_fixate):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_caps):
* gst/audioscale/gstaudioscale.c: (gst_audioscale_fixate):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiotestsrc_src_fixate):
* gst/sine/gstsinesrc.c: (gst_sinesrc_src_fixate):
* gst/videorate/gstvideorate.c: (gst_videorate_setcaps):
* gst/videoscale/gstvideoscale.c: (gst_videoscale_fixate_caps):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_src_fixate):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_fixate):
Rename gst_caps_structure_fixate_* to gst_structure_fixate_*
(#322027)
 2005-11-21 14:29:53 +00:00
Wim Taymans
31461ebd0b gst/: The base class can now sync for us. 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiotestsrc_class_init), (gst_audiotestsrc_get_times),
(gst_audiotestsrc_create):
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_get_times), (gst_sinesrc_create):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_class_init), (gst_videotestsrc_get_times),
(gst_videotestsrc_create):
The base class can now sync for us.
 2005-11-10 14:58:41 +00:00
Tim-Philipp Müller
2548818edc gst/: Don't use functions for position queries when handling duration queries. 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiotestsrc_src_query):
* gst/sine/gstsinesrc.c: (gst_sinesrc_src_query):
Don't use functions for position queries when handling
duration queries.
 2005-10-20 16:44:17 +00:00
Wim Taymans
f88e6c08fd Query API update. 
Original commit message from CVS:
* examples/seeking/seek.c: (make_avi_msmpeg4v3_mp3_pipeline),
(query_positions_elems), (query_positions_pads), (update_scale),
(do_seek), (set_update_scale), (message_received), (main):
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_find_chains),
(gst_ogg_demux_loop):
* ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
* ext/theora/theoradec.c: (theora_dec_src_query),
(theora_dec_sink_event):
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_sink_event), (vorbis_handle_data_packet):
* gst/adder/gstadder.c: (gst_adder_query):
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiotestsrc_src_query):
* gst/playback/test3.c: (update_scale):
* gst/playback/test5.c: (new_pad), (no_more_pads), (start_finding),
(dump_element_stats), (main):
* gst/playback/test6.c: (main):
* gst/sine/gstsinesrc.c: (gst_sinesrc_src_query):
Query API update.
 2005-10-19 15:55:33 +00:00
Stefan Kost
456bd88950 gst/: fix broken build of controllerized plugins 
Original commit message from CVS:
* gst/audiotestsrc/Makefile.am:
* gst/sine/Makefile.am:
* gst/volume/Makefile.am:
fix broken build of controllerized plugins
 2005-10-17 19:57:05 +00:00
Thomas Vander Stichele
fc0e077a93 gst/: doc updates 
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_get_size):
* gst/audiotestsrc/gstaudiotestsrc.c:
doc updates
 2005-10-17 15:37:45 +00:00
Thomas Vander Stichele
4f8f42b0b6 restructure configure.ac, use correct libtool LDFLAGS, fix up defines 
Original commit message from CVS:
restructure configure.ac, use correct libtool LDFLAGS, fix up defines
 2005-10-16 13:54:44 +00:00
Stefan Kost
7f3f034d44 gst/audiotestsrc/gstaudiotestsrc.*: fixed typo, added pink noise 
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audiotestsrc_class_init),
(gst_audiotestsrc_init), (gst_audiotestsrc_create_sine),
(gst_audiotestsrc_create_square), (gst_audiotestsrc_create_saw),
(gst_audiotestsrc_create_triangle),
(gst_audiotestsrc_create_silence),
(gst_audiotestsrc_create_white_noise),
(gst_audiotestsrc_init_pink_noise),
(gst_audiotestsrc_generate_pink_noise_value),
(gst_audiotestsrc_create_pink_noise),
(gst_audiotestsrc_change_wave):
* gst/audiotestsrc/gstaudiotestsrc.h:
fixed typo, added pink noise
 2005-10-09 20:47:31 +00:00
Stefan Kost
9be025e197 add new plugin and element 
Original commit message from CVS:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* gst/audioscale/gstaudioscale.c: (gst_audioscale_method_get_type):
* gst/audiotestsrc/Makefile.am:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audiotestsrc_base_init),
(gst_audiotestsrc_class_init), (gst_audiotestsrc_init),
(gst_audiotestsrc_src_fixate), (gst_audiotestsrc_setcaps),
(gst_audiotestsrc_get_query_types), (gst_audiotestsrc_src_query),
(gst_audiotestsrc_wait), (gst_audiotestsrc_unlock),
(gst_audiotestsrc_create_sine), (gst_audiotestsrc_create_square),
(gst_audiotestsrc_create_saw), (gst_audiotestsrc_create_triangle),
(gst_audiotestsrc_create_silence),
(gst_audiotestsrc_create_white_noise),
(gst_audiotestsrc_change_wave), (gst_audiotestsrc_create),
(gst_audiotestsrc_set_property), (gst_audiotestsrc_get_property),
(gst_audiotestsrc_start), (plugin_init):
* gst/audiotestsrc/gstaudiotestsrc.h:
add new plugin and element
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init):
use gobject_class
 2005-10-09 18:34:44 +00:00