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.
From: Pierre Ossman <drzeus-list@drzeus.cx>
Freeing the global config update structure when it had been created
without any configs present caused an assertion to trigger. Since this
is a valid scenario and the assertion didn't really protect against
anything, it should simply be removed. Attached patch does exactly that.
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.
Example failure error message before fix:
speaker-test 0.0.8
Playback device is spdif
Stream parameters are 48000Hz, S16_LE, 2 channels
Using 16 octaves of pink noise
ALSA lib setup.c:549:(add_elem) Cannot obtain info for CTL elem (MIXER,'SPDIF Front',0,0,0): No such file or directory
Playback open error: -2,No such file or directory
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.
From: Nicolas Boullis <nboullis@debian.org>
While working on an ALSA driver for the audio part of em8300-based MPEG
decoder boards (Sigma Designs' Hollywood+ boards and Creative Labs' DXR3
boards), I think I discovered a bug in the code that generates IEC958
subframes from linear samples (iec958 plugin): M and W preambles are
swapped, and B preambles should only be used on even subframes. The
attached patch fixes this issue.
