With the patch, dmix allows apps to use more flexible buffer sizes.
The max buffer size is unlimited, and the minimal buffer size is
(period size * 2). The buffer size is aligned to period size.
The period size is still bound to the period size of slave PCM.
To back to the old behavior (the fixed buffer size), you can set
defaults.pcm.dmix_variable_buffer false
in your configuration.
Here's a patch for generic dmix which fixes S16 byte swapping.
Tested on powerpc with snd-usb-audio. (Without the patch I get crackling.)
Signed-off-by: Juergen Kreileder <jk@blackdown.de>
Add restriction parameters to pcm hw layer.
The PCM hw has optional parameters, format, rate and channels, to restrict
the configuration. This is useful for definition of surround slave PCMs,
for example.
When calculating the size of the second fragment, do not assume that the
entire size is one period size (which is not true in the draining state)
but use the actual size passed by the caller.
When calculating the continuous part till the end of the buffer, we can
use the slave_frames value that has already been calculated by
snd_pcm_mmap_begin().
Add configuration options to change the default device path from the
default /dev/snd. This is useful for embedded systems that do not want
subdirectories in /dev.
Remove several memory leaks by not aborting prematurely from a
snd_xxx_close() function when some operation fails.
This can happen when a USB device was unplugged.
Fix the check of nonblock option for all hw layer.
Instead of passing in asound.conf, check the option in snd_pcm_hw_open()
so that the nonblock option is referred in the case of "type hw ..." style
definition, too.
It turned out that plugins that had control outputs were not being set
up properly if there was no corresponding "output" section.
Signed-off-by: Nathan Kurz <nate@verse.com>
- Support multi-card/device for dmix/dsnoop/dshare plugins
The unique ipc key is calculated based on card/device/sub index
- Clean up and share the code among all d* plugins
- Refer the defaults.pcm.* configuration
The base ipc_key number, ipc_gid and ipc_perm are referred.
Added a new "nonblock" option for hw layer. This controls the non-blocking
"open" mode as default.
This option is set to TRUE as the default configuration. If the old behavior
is preferred, set "defaults.pcm.nonblock" to 0 in /etc/asound.conf.
The previous code did not allocated a separate buffer for all channels
when a NONINTERLEAVED access was used. The result was that only one
"shared" buffer was incorrectly allocated.
Also, the code was a bit cleaned (cosmetic change only).
Add to the dmix plugin support for the S24_3LE sample format which is
used by 24-bit USB devices.
The optimized assembler version uses only 23 bits for sample data so
that the lowest bit can be used for synchronization because there is no
24-bit cmpxchg instruction.
Add --enable-* and --with-pcm-plugins configure options for partial builds.
User can choose the core components (pcm, mixer, rawmidi, hwdep, seq, instr)
via --enable-xxx or --disable-xxx option. As default, all components are
enabled.
The PCM plugins to build can be selected via --with-pcm-plugins option.
For example, to build only rate and linear plugin, pass
--with-pcm-plugins=rate,linear
Passing "all" will select all plugins (it's the default value).
The plug plugin will select linear and copy plugins automatically.
The other auto conversions of plug plugin are enabled only when the
corresponding plugin is selected.
From: Sascha Sommer <saschasommer@freenet.de>
this is a fix for bug 0001559.
Unlike my first guesses the real problem is not in the kernel driver but in
alsa-lib. Whenever the current dmix status is xrun and snd_pcm_dmix_drain
gets called the process will hang forever in the poll function. The reason is
that poll gets called even though the timer already stopped.
As described in the bugtracking system this bug was not noticable with alsa
versions that used the old IOCTLS because the SNDRV_TIMER_IOCTL_STOP ioctl
never reached the timer kernel module.
Attached patch fixes this problem for alsa-lib 1.0.10 by simply calling
snd_pcm_dmix_drop when snd_pcm_dmix_drain gets called in the state
SND_PCM_STATE_XRUN.
