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.
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
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
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.
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.
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.
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)
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
