PCM direct plugins didn't update the timestamp properly.
Now it always starts the slave PCM with MMAP tstamp_mode so that the
timestamp will be being updated. When a client is set up as MMAP
tstamp_mode as well, simply copy this slave timestamp. Otherwise
status callback calculates the current timestamp as usual.
The rate plugin has ugly hacks in poll_descriptors callback to adjust
avail_min when partial read/write occurs. This causes often unexpected
problems like XRUNs, especially with two-period cases.
Let's remove that beast, it's rather harmful than useful.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
When calculating the value range of a control, the variables cannot be
initialized with zero because this would prevent the minimum from having
a value above zero or the maximum from having a value below zero.
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.
Putting assert in the public macros isn't good idea at all.
Let's get rid of them.
Also, clean up snd*_alloca() functions to use a helper macro
instead of copy and paste.
Export helper functions to convert dB level and range.
snd_tlv_*dB*() are to convert dB level or range directly from TLV data.
snd_ctl_*dB*() are to get dB level or range from a control element.
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).
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.
From: Mike Gorse <mgorse@mgorse.dhs.org>
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.
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.
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.
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.
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.
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>
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.
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.
