Rename the local functions to snd1_* so that they won't be exported
out of alsa-lib.
Some functions are still kept because aserver requires them. Sigh.
softvol can be also a pass-thru when the given control already exists
as a hardware control, and the name isn't set properly because of
slave creation. This patch fixes it.
The PCM name isn't set properly on empty and asym plugins due to its call
of snd_pcm_open_slave(). Now a new function snd_pcm_open_named_slave()
is created and make snd_pcm_open_slave() an inline function calling the
new one with name=NULL.
The mmap of multi plugin seems broken (for a long time!) due to its
creation of local buffer via snd_pcm_mmap(). Since the multi plugin
just needs to shadow the mmap buffer of each slave, it now has
mmap_shadow=1 and its own mmap/unmap method to do shadowing.
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>
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>
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.
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.
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.
Improve the sample conversion in plug layer for the sample bits
reduction. Use round instead of truncating, which introduces
more quantization noise.
Original patch by Jean-Marc Valin <jean-marc.valin@usherbrooke.ca>
The old format tried to do something when the requested format was not
supported by the hardware, but did not actually select any other format.
Now we try to switch to any format supported by dmix, or any other
format when the plugin is not dmix.
The code to set the number of channels did not work when the requested
channel count was not available and when the min/max channel counts were
not identical.
Replacing the entire selection code with
snd_pcm_hw_params_set_channels_near() gives the same result in the cases
where it previously worked, and works in all other cases.
- add long card name to device description
- create empty PCM plugin to allow right hint description parsing
- reorder devices in alsa.conf
- make namehint more configurable (using default.namehint.showall switch)
- add two levels basic and exteded for hints to default configuration files
- do not show direct device aliases
- removed all known memory leaks
- add snd_device_name_hint() and snd_device_name_free_hint() functions
- add snd_ctl_iface_conf_name() functions
- do not accept parameters for the plugin definition without @args section
- add defaults.pcm.dmix.card/device and dsnoop.card/device definitions
- add hints for HDA-Intel.conf, pcm/dmix.conf, pcm/dsnoop.conf and alsa.conf
- add test/namehint test utility
- doxygen related cleanups
