Takashi Iwai [Tue, 30 Oct 2007 11:31:55 +0000 (12:31 +0100)]
Remove sequencer instrument layer
Remove obsoleted sequencer instrument layer from alsa-lib.
The old symbols are compiled in as default as dummy functions
(unless --disable-old-symbols is given to configure) so that
the old binaries can still work more or less.
Takashi Iwai [Wed, 24 Oct 2007 10:53:08 +0000 (12:53 +0200)]
Change assert condition in error message handler
Activating assert() in the default error message handler isn't always
good for producitve systems. Make this optional and enable only when
a special configure option is given (i.e. for explicit debugging).
Takashi Iwai [Thu, 18 Oct 2007 09:10:35 +0000 (11:10 +0200)]
snd_pcm_dmix_close: raise semaphore if unable to discard
This patch causes snd_pcm_dmix_close() to up a semaphore after downing it
if it is unable to discard it. It prevents some deadlock that I am
getting when a couple of applications interact and one of them closes the
device and later re-opens it.
Takashi Iwai [Wed, 29 Aug 2007 12:48:31 +0000 (14:48 +0200)]
Handle "Input Source" as a capture element
Some drivers use "Input Source" as the capture source mixer element because
mixer abstraction layer can't handle multiple "Capture Source" elements.
This patch adds a hack to handle Input Source as a capture route, and let
mixer apps know that it's a capture stuff, at least.
Takashi Iwai [Wed, 22 Aug 2007 09:41:58 +0000 (11:41 +0200)]
Continue build even if python-config doesn't exist
It's annoying that the configure script stops with an error if
python-config doesn't exist. It's no fatal error, so the configure
should disable the python components and continue.
Clemens Ladisch [Fri, 10 Aug 2007 07:39:24 +0000 (09:39 +0200)]
seq_midi_event: prevent running status after system messages
Reset the event type after encoding a system message to prevent any
following data bytes from being interpreted as data for a running status
system message, which is not allowed in MIDI.
Clemens Ladisch [Fri, 10 Aug 2007 07:38:47 +0000 (09:38 +0200)]
seq_midi_event: fix encoding of data bytes after end of sysex
Create a new state ST_INVALID for the encoder to prevent data bytes at
the beginning of a stream or after a sysex message being interpreted as
note-off parameters.
Jaroslav Kysela [Wed, 11 Jul 2007 08:10:12 +0000 (10:10 +0200)]
mixer simple basic abstraction - added python binding
reasons:
- rapid development
- class-like code structure
- more readable code
features:
- hcontrol binding is managed from python (opportunity to create
virtual mixer without driver or join multiple cards to behave as one)
i386/x86_64 alsa-lib may need to handle big-endian formats, e.g.
when running via qemu on PPC. The generic dmix code already has
both endian support, so let's use it as fallback.
Clemens Ladisch [Mon, 21 May 2007 07:13:19 +0000 (09:13 +0200)]
dmix/dshare/dsnoop plugin: enable slowptr by default
Enabling the slowptr options does not make snd_pcm_delay() and related
functions much slower than they would have been with a hw device, while
disabling this option greatly reduces the accuracy of those functions,
thus creating more jitter in any media player application that
synchronizes its output to the sound device.
Therefore, it is preferrable to have this option enabled by default.
Steve Longerbeam [Fri, 18 May 2007 13:04:12 +0000 (15:04 +0200)]
Add support for gain in softvol plugin
This patch allows for gain in the softvol plugin, in addition to attenuation.
The plugin now has a "max_dB" parameter (up to 50 dB) as well as the
original "min_dB" parameter (down to -51 dB). max_dB defaults to 0 dB, so
unless max_dB is specified in a device conf, the behavior of the plugin will
be the same as before (attenuation only).
HDA-Intel.conf is also modified to use softvol for its default capture.
So now, capture is filtered through softvol (range -30 to +30 dB) before
being passed on to dsnoop as before.
The softvol plugin allows a range of -51 to +50 dB, so max_dB could be
increased to 50. But eventually samples are going to get clipped. At 40
dB I was beginning to get clipping when recording a sample sound at a
"reasonably soft" volume using a digital mic on the stac9205 HDA codec.
The motivation for this work is that some HDA codecs have no hardware gain
control for some paths. For instance, the stac9205 has support for digital
mics, but there is no gain control widget for this signal before it is placed
on the Azalia link (only a mute). Therefore gain can only be accomplished
via software.
Signed-off-by: Steve Longerbeam <stevel@embeddedalley.com>
Takashi Iwai [Thu, 3 May 2007 18:55:54 +0000 (20:55 +0200)]
Add config and plugin directory options to configure
Added --with-configdir and --with-plugindir options to configure
which specify the directories for config files and plugin objects
respectively. The default paths when these options are not
specified are unchanged.
Added --enable-symbolic-functions configure option. This will detect
and pass -Bsymbolic-functions linker option, which gives you better
performance and smaller binary size. Only recent binutils supports
this option.
Takashi Iwai [Wed, 21 Mar 2007 23:58:42 +0000 (00:58 +0100)]
String array for default rate plugin list
Change the rate converter type to allow string arrays in addition to
a string. When a string array is given, the rate plugin probes each
string and try to load the converter plugin in the order of the list.
For example, you can set
defaults.pcm.rate_converter [ "samplerate" "linear" ]
so that samplerate plugin is preferred to linear plugin if it's
installed.
Rene Herman [Wed, 21 Mar 2007 11:21:38 +0000 (12:21 +0100)]
Fix alsa-devel ML address
This replaces all occurences of alsa-devel@lists.s[ource]f[orge].net
that a simple recursive grep found in the current HG ALSA repos by
alsa-devel@alsa-project.org.
Signed-off-by: Rene Herman <rene.herman@gmail.com>
Takashi Iwai [Tue, 13 Mar 2007 01:52:33 +0000 (02:52 +0100)]
More better fix for linked start/stop
Instead of link_fd, more generic callback link_slaves is introduced.
This is called for linking the slave streams as the source to the
given master stream.
Takashi Iwai [Fri, 9 Mar 2007 14:30:38 +0000 (15:30 +0100)]
Check _POSIX_SOURCE for timeval definition
Check _POSIX_SOURCE for struct timeval definition.
In rare cases, _POSIX_C_SOURCE isn't defined but only _POSIX_SOURCE
is set, and it results in the confliction of timeval definition.
Clemens Ladisch [Mon, 12 Feb 2007 12:45:03 +0000 (13:45 +0100)]
remove superfluous strdup() calls
Remove same calls of strdup() that were unnecessary because the
temporary string would not be modified or held longer than the lifetime
of the original string.
Takashi Iwai [Fri, 19 Jan 2007 17:36:48 +0000 (18:36 +0100)]
Update emu10k1.h for new version
Updated emu10k1.h for newer version.
The size of emu10k1_fx8010_control_gpr_t is changed. The apps need to
issue SNDRV_EMU10K1_IOCTL_PVERSION first to switch to the new ABI.
Jaroslav Kysela [Wed, 20 Dec 2006 15:01:27 +0000 (16:01 +0100)]
pcm_ladspa - fix none policy
In some configurations, channel samples were not copied correctly.
Also, fix memory leaks and remove extra end-of-line chars from SNDERR
strings.