Take the card iterator idea from the monitor code and
use it for all card loops. It reduces the code duplications
and makes things easy to review.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Also remove extra snd_config_update_free_global() call for dump-state.
There's a global call in the main() function.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add missing channel params to the amixer man page. Also call out that
the channel param must come before the value to take effect.
signed-off-by: Matthew Campbell <mcampbell@izotope.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
It is handy to remove all card controls created by applications.
This change allows to remove those controls for all cards, selected
card or selected card with a control id filter list like:
alsactl clean 0 "name='PCM'" "name='Mic Phantom'"
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
We added the check of asound.state file presence some time ago to
assure that alsactl gets called only if the state file is already
present. Since then, the situation has changed significantly:
e.g. now alsactl does initialize if the state file isn't present, and
the same alsa-restore.service is used to save the state. This means
that we should start this service no matter the state file exists at
the boot time or not. So, revert the old change again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A race at creating a lock file in state_lock() was discovered
recently: namely, between the first open(O_RDWR) and the second
open(O_RDWR|O_CREAT|O_EXCL) calls, another alsactl invocation may
already create a lock file, then the second open() will return EEXIST,
which isn't handled properly and treated as a fatal error.
In this patch, we check EEXIST case and try again open() with O_RDWR.
This must succeed usually, and if it fails, handle finally as the
fatal error.
BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1179904
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The state_lock() has a loop to wait for the lock file creation, and
the timeout value gets decremented twice mistakenly, which leads to a
half timeout (5 seconds) than expected 10 seconds. Fix it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We need more detailed information for USB-audio devices, at least the
lsusb -v output and the contents of stream* proc files.
Let's add them to alsa-info.sh output.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the standard poll mechanism to ensure that there's
something in the input to avoid busy loop on the file
descriptor with the non-block mode set.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Recently users reported a bug, I tested it and found it is a common
issue on Laptop or Desktop machines.
The issue is users plug a headset and use "arecord test.wav" to
record a sound with default input volume, the recorded sound has
poor quality and nearly can't distinguish it is the sound we want
to record.
This is because the input volume is low and the default format is U8.
The driver records sound with 16bit, because the input volume is low,
most of samples are within (-256,+256), when converting 16bit to U8,
those samples will be 0x7f. This is called quantization noise and we
could only workaround it by increase the input volume or adding -f to
arecord.
But users want to record a better quality sound with default input
volume (after installing a new OS, the volume is the default volume),
and they don't want to add parameters to the arecord because most of
new linux users just use "arecord test.wav".
So this patch tries to change the default format from U8 to S16_LE/BE.
If the machine doesn't support S16_LE/BE, it still uses U8 as default
format.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The count argument was renamed to samples to correctly represent
the value meaning. Also, remove the wrong count recalculation lines
for 16-bit, 24-bit and 32-bit samples.
BugLink: https://github.com/alsa-project/alsa-utils/issues/57
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
