Commit graph

29 commits

Author SHA1 Message Date
Olivier Crête
2f83ca5c7f alsadeviceprovider: Rank down to secondary so PulseAudio can hide it
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/727>
2020-07-06 16:32:21 +00:00
Thibault Saunier
ac5d0f7da6 alsa: Implement a DeviceProvider
Removing gstalsadeviceprobe.[ch] as it was a relique from the 0.10
century.

This doesn't implement device monitoring but only probing, monitoring
should be implemented in its own commit.
2019-06-04 14:07:37 +00:00
Nicolas Dufresne
d64a4b7a69 Revert "alsa: Implement a DeviceProvider"
This reverts commit 69c3c31608.

All devices have the same name, they are duplicated with pulseaudio one
and the provided does not respond to HW being plugged/unplugged. I think
it's not ready for 1.16.
2019-01-18 11:39:02 -05:00
Thibault Saunier
69c3c31608 alsa: Implement a DeviceProvider
Removing gstalsadeviceprobe.[ch] as it was a relique from the 0.10
century.
2019-01-18 10:18:54 -03:00
Antonio Ospite
2c7ed42292 midi: add an ALSA MIDI sequencer source
The alsamidisrc element allows to get input event from ALSA MIDI
sequencer devices, and possibly convert them to sound using some
downstream element like fluiddec.

Fixes #738687
2015-10-01 21:43:21 +02:00
Sebastian Dröge
d3e0381d3d alsa: Make clang happy with our g_strdup_vprintf() wrapper 2014-02-08 17:01:38 +01:00
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
69b18ab09d gst-libs: Remove interfaces libs and mixer/tuner interfaces
The navigation interface is now in the video library.
2012-04-13 13:14:13 +02:00
Sebastian Dröge
ad42b16375 gst: Update for GST_PLUGIN_DEFINE() API change 2012-04-05 15:11:05 +02:00
Benjamin Otte
420d7b111d More ENABLE_NLS fixes 2010-03-16 18:31:15 +01:00
Frederic Crozat
89be246154 Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
Original commit message from CVS:
Patch by: Frederic Crozat <fcrozat@mandriva.org>
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
* gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
* gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
* gst/playback/gstdecodebin.c: (plugin_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
* gst/playback/gstplayback.c: (plugin_init):
* gst/playback/gstqueue2.c: (plugin_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
* sys/v4l/gstv4l.c: (plugin_init):
Make sure gettext returns translations in UTF-8 encoding rather
than in the current locale encoding (#546822).
2008-08-07 15:58:58 +00:00
Thomas Vander Stichele
4f10b0983d ext/: - a library should not call setlocale. see Libraries node in gettext manual
Original commit message from CVS:

* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_base_init), (plugin_init):
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
- a library should not call setlocale. see Libraries node in
gettext manual
- make sure all plugins that use translation do bindtextdomain
to point to the localedir
* gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
(setup_sinks), (plugin_init):
all this, and check for NULL when creating sinks
2006-01-27 01:06:29 +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
Andy Wingo
13b122a106 gst-libs/gst/audio/gstaudiosrc.*: Implement open_device and close_device in the ring buffer, like gstaudiosink.
Original commit message from CVS:
2005-08-22  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstaudiosrc.c: Implement open_device and
close_device in the ring buffer, like gstaudiosink.

* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixer.c: Not a GObject any more. Include a nifty
macro to implement the interface without much code. Cleanups.

* ext/alsa/gstalsasrc.h:
* ext/alsa/gstalsasrc.c: Be a mixer. Open device and mixer in
READY.

* ext/alsa/Makefile.am: Add new files.
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsamixerelement.c: Split element code out from
mixer code so that alsasrc can be a mixer too.
2005-08-22 15:11:31 +00:00
Andy Wingo
b05796c9d9 ext/alsa/: Port to 0.9.
Original commit message from CVS:
2005-08-19  Andy Wingo  <wingo@pobox.com>

* ext/alsa/gstalsamixertrack.h:
* ext/alsa/gstalsamixertrack.c:
* ext/alsa/gstalsamixeroptions.h:
* ext/alsa/gstalsamixeroptions.c:
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixer.c: Port to 0.9.

* ext/alsa/Makefile.am: Build mixer, mixeroptions, mixertracks.
Remove gstalsa.c and alsaclock. No more cruft here.
2005-08-19 16:13:54 +00:00
Andy Wingo
708deec535 ext/alsa/gstalsaplugin.c (plugin_init): We are primary audio sinks.
Original commit message from CVS:
2005-07-29  Andy Wingo  <wingo@pobox.com>

* ext/alsa/gstalsaplugin.c (plugin_init): We are primary audio
sinks.

* ext/alsa/gstalsasink.c (alsasink_sink_factory): Advertise our
support of both endiannesses.
2005-07-29 15:42:17 +00:00
Wim Taymans
ceb88a7777 Added audiosource base classes.
Original commit message from CVS:
Added audiosource base classes.
Ported alsasrc, still very basic.
2005-07-06 15:27:17 +00:00
Wim Taymans
851547e321 ext/alsa/: Implement alsasink with simple open/write/close API.
Original commit message from CVS:
* ext/alsa/Makefile.am:
* ext/alsa/gstalsa.c: (gst_alsa_init), (gst_alsa_get_caps),
(gst_alsa_fixate_to_mimetype), (gst_alsa_fixate_field_nearest_int),
(gst_alsa_link), (gst_alsa_close_audio):
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/alsa/gstalsasink.c: (gst_alsasink_get_type),
(gst_alsasink_dispose), (gst_alsasink_base_init),
(gst_alsasink_class_init), (gst_alsasink_init),
(gst_alsasink_getcaps), (set_hwparams), (set_swparams),
(alsasink_parse_spec), (gst_alsasink_open), (gst_alsasink_close),
(xrun_recovery), (gst_alsasink_write), (gst_alsasink_delay),
(gst_alsasink_reset):
* ext/alsa/gstalsasink.h:
Implement alsasink with simple open/write/close API.
Make alsa dir build by disabling compilation of code.
2005-04-28 16:19:06 +00:00
David Schleef
129c7e8af1 configure.ac: Remove idct and resample libs
Original commit message from CVS:
* configure.ac: Remove idct and resample libs
* gst-libs/gst/Makefile.am: same
Remove usage of gst_library_load():
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/libvisual/visual.c: (plugin_init):
* ext/ogg/gstogg.c: (plugin_init):
* ext/theora/theora.c: (plugin_init):
* ext/vorbis/vorbis.c: (plugin_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init):
* gst/audioscale/gstaudioscale.c:
* gst/adder/gstadder.c: (plugin_init):
* gst/audioconvert/plugin.c: (plugin_init):
* sys/ximage/ximagesink.c: (plugin_init):
* sys/xvimage/xvimagesink.c: (plugin_init):
* gst/tcp/gsttcpplugin.c: (plugin_init):
Link plugins against libraries:
* ext/ogg/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst/audioconvert/Makefile.am:
Create proper libraries:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/video/Makefile.am:
Move resample library to audioscale plugin directory:
* gst-libs/gst/resample/Makefile.am:
* gst-libs/gst/resample/README:
* gst-libs/gst/resample/dtof.c:
* gst-libs/gst/resample/dtos.c:
* gst-libs/gst/resample/functable.c:
* gst-libs/gst/resample/private.h:
* gst-libs/gst/resample/resample.c:
* gst-libs/gst/resample/resample.h:
* gst-libs/gst/resample/resample.vcproj:
* gst-libs/gst/resample/test.c:
* gst/audioscale/Makefile.am:
* gst/audioscale/README:
* gst/audioscale/dtof.c:
* gst/audioscale/dtos.c:
* gst/audioscale/functable.c:
* gst/audioscale/private.h:
* gst/audioscale/resample.c:
* gst/audioscale/resample.h:
* gst/audioscale/test.c:
Move tagedit library to gst-libs:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gsttagediting.c:
* gst-libs/gst/tag/gsttageditingprivate.h:
* gst-libs/gst/tag/gstvorbistag.c:
* gst/tags/Makefile.am:
* gst/tags/gstid3tag.c:
* gst/tags/gstvorbistag.c:
Fix for core changes:
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_init), (gst_sinesrc_src_fixate), (gst_sinesrc_link),
(gst_sinesrc_getrange):
2005-04-25 00:23:06 +00:00
Ronald S. Bultje
3a0a2898af Surround sound support.
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_channels), (gst_a52dec_push),
(gst_a52dec_reneg), (gst_a52dec_loop), (plugin_init):
* ext/alsa/gstalsa.c: (gst_alsa_get_caps):
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/dts/gstdtsdec.c: (gst_dtsdec_channels),
(gst_dtsdec_renegotiate), (gst_dtsdec_loop), (plugin_init):
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_chanpos_from_gst),
(gst_faad_chanpos_to_gst), (gst_faad_sinkconnect),
(gst_faad_srcgetcaps), (gst_faad_srcconnect), (gst_faad_chain),
(gst_faad_change_state), (plugin_init):
* ext/faad/gstfaad.h:
* ext/vorbis/vorbis.c: (plugin_init):
* ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/audio.c: (plugin_init):
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_get_channel_positions),
(gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
(gst_audio_fixate_channel_positions):
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/audio/testchannels.c: (main):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_init),
(gst_audio_convert_dispose), (gst_audio_convert_getcaps),
(gst_audio_convert_parse_caps), (gst_audio_convert_link),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
* gst/audioconvert/plugin.c: (plugin_init):
Surround sound support.
2004-11-25 20:36:29 +00:00
David Schleef
a0dd75c692 ext/alsa/gstalsaplugin.c: Disable call to gst_debug_log() if debugging is disabled (bug #145118)
Original commit message from CVS:
* ext/alsa/gstalsaplugin.c: (gst_alsa_error_wrapper): Disable
call to gst_debug_log() if debugging is disabled (bug #145118)
2004-07-03 23:35:36 +00:00
Benjamin Otte
f157024ae8 ext/alsa/gstalsa.c: - don't call set_periods_integer anymore, it breaks the configuration randomly
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_probe_hw_params),
(gst_alsa_set_hw_params), (gst_alsa_set_sw_params),
(gst_alsa_start_audio):
- don't call set_periods_integer anymore, it breaks the
configuration randomly
- call snd_pcm_hw_params_set_access directly instead of using masks
- don't fail if the sw_params can't be set, just use the default
params and hope it works. Alsalib has weird issues when you touch
sw_params and does no proper error reporting about what failed.
* ext/alsa/gstalsa.c: (gst_alsa_open_audio),
(gst_alsa_close_audio):
make our alsa debugging go via gst debugging and not conditionally
defined
* ext/alsa/gstalsa.h:
add ALSA_DEBUG_FLUSH macro
* ext/alsa/gstalsaplugin.c: (gst_alsa_error_wrapper),
(plugin_init):
wrap alsa errors to be printed via the gst debugging system and not
spammed to stderr
2004-06-06 17:26:54 +00:00
Benjamin Otte
4ae33d8a98 add experimental kiosrc plugin
Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/kio/Makefile.am:
* ext/kio/kioreceiver.cpp:
* ext/kio/kioreceiver.h:
* ext/kio/kiosrc.cpp:
* ext/kio/kiosrc.h:
add experimental kiosrc plugin
* ext/alsa/gstalsaplugin.c: (plugin_init):
initialize debugging category only when we're sure registering the
plugins worked.
2004-03-30 02:53:00 +00:00
Thomas Vander Stichele
f83cb187de don't mix tabs and spaces
Original commit message from CVS:
don't mix tabs and spaces
2004-03-15 19:32:28 +00:00
Thomas Vander Stichele
4df3f18839 gst-indent
Original commit message from CVS:
gst-indent
2004-03-14 22:34:34 +00:00
Leif Johnson
ba2b9bd9d1 + got the alsa mixer implementation to work !
Original commit message from CVS:
+ got the alsa mixer implementation to work !
2003-12-06 00:02:20 +00:00
Andy Wingo
80fece4f4b remove copyright field from plugins
Original commit message from CVS:
remove copyright field from plugins
2003-12-04 10:37:39 +00:00
Benjamin Otte
280c25766a fixes for recent changes:
Original commit message from CVS:
fixes for recent changes:
- GstAlsaClock is not a GstSystemClock
- initialize debugging system correctly
2003-11-18 15:32:52 +00:00
Leif Johnson
f3b328da39 splitting ALSA code into separate source files
Original commit message from CVS:
splitting ALSA code into separate source files
2003-11-16 00:40:01 +00:00