Takashi Iwai [Fri, 11 Dec 2020 22:55:34 +0000 (23:55 +0100)]
alsactl: Remove asound.state file check from alsa-restore.service again
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.
Takashi Iwai [Fri, 11 Dec 2020 22:46:23 +0000 (23:46 +0100)]
alsactl: Fix race at creating a lock file
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.
Takashi Iwai [Fri, 11 Dec 2020 22:41:59 +0000 (23:41 +0100)]
alsactl: Fix double decrease of lock timeout
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.
Takashi Iwai [Wed, 9 Dec 2020 17:35:49 +0000 (18:35 +0100)]
alsa-info: Add lsusb and stream outputs
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.
Hui Wang [Fri, 23 Oct 2020 08:47:10 +0000 (16:47 +0800)]
aplay: try to use 16-bit format to increase capture quality
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>
Jaroslav Kysela [Thu, 15 Oct 2020 11:12:37 +0000 (13:12 +0200)]
aplay: fix the wrong count in compute_max_peak() for 16bit+ sample widths
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.
The system sku is used in SOF machine drivers to set specific quirks,
expose it in alsa-info to help support users. Example on a SoundWire
enabled device:
Marcin Rajwa [Fri, 7 Aug 2020 14:17:25 +0000 (16:17 +0200)]
aplay: capture: write only the amount of read frames
This patch changes the way aborted stream is being saved. Currently when
abort signal happens the write back of read samples is skipped but there
is no reason to not save them. Also, we need to know how much frames have
been read and write only those.
Signed-off-by: Marcin Rajwa <marcin.rajwa@linux.intel.com> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Marcin Rajwa [Fri, 7 Aug 2020 15:35:44 +0000 (17:35 +0200)]
aplay: pcm_read(): return read samples instead of requested upon abort
This patch changes the logic of pcm_read() when abort signal has been
detected. During such condition we should return the amount of frames
actually read instead of the size requested by caller.
Currently functions pcm_read() and pcm_readv() when aborted (in_aborting
flag set) return the amount of requested frames instead of those actually
read prior to interrupt. The consequence of this is repetition of recent X
frames where X stands for amount of frames in one period. This problem is
barely visible or rather audible when the period is small like few
milliseconds because repetition of 1 [ms] of data is not-noticeable
however if we use buffer and period sizes in seconds then the problem
becomes apparent.
Signed-off-by: Marcin Rajwa <marcin.rajwa@linux.intel.com> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Marcin Rajwa [Fri, 7 Aug 2020 13:56:05 +0000 (15:56 +0200)]
aplay: pcm_readv(): return read samples instead of requested upon abort
This patch changes the logic of pcm_readv() when abort signal has been
detected. During such condition we should return the amount of frames
actually read instead of the size requested by caller.
Currently functions pcm_read() and pcm_readv() when aborted (in_aborting
flag set) return the amount of requested frames instead of those actually
read prior to interrupt. The consequence of this is repetition of recent X
frames where X stands for amount of frames in one period. This problem is
barely visible or rather audible when the period is small like few
milliseconds because repetition of 1 [ms] of data is not-noticeable
however if we use buffer and period sizes in seconds then the problem
becomes apparent.
Example issue -> thesofproject/sof#3189
Signed-off-by: Marcin Rajwa <marcin.rajwa@linux.intel.com> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The silence frames generation in xrun_sync() seems to be currently
a bit off as snd_pcm_format_set_silence() uses a wrong buffer address
and size for zero-ing out the data. Consequently instead of clearing
out the last frame(s), snd_pcm_format_set_silence() ends up silencing
samples somewhere else in the buffer. This is partilarly more obvious
for higher frame sizes (e.g. more than 2 channels).
This patch fixes this issue by correcting the parameters passed to
snd_pcm_format_set_silence().
If we use 'alsaucm -n -b -' with 'get "CapturePCM/Internal Mic"' then
the alsaucm will report error and stop immediately. The reason is that
the parse_line in usecase.c appends an empty argument if a command ends
with quotes.
This change adds a patch to fix the parse_line function in usecase.c.
Signed-off-by: Cheng Yueh <cyueh@chromium.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
braph [Thu, 3 Oct 2019 18:18:23 +0000 (20:18 +0200)]
alsamixer: small optimizations
textbox.c, mixer_display.c:
- Replaced manual line drawing code `for (...) mvwaddch(...)` by the
functions `mvwhline/mvwvline` provided by standard curses
card_select.c, proc_files.c:
- Moved duplicate code in the create() functions to menu_widget_create()
in menu_widget.c
card_select.c:
- Removed redundant calls to sprintf()
device_name.c:
- Combined multiple calls to form_opts_off()
- Recognize ASCII DEL (127) as backspace
widget.c:
- Uncommented code for setting a dummy key handler (all existing
widgets do provide a key handling function)
Signed-off-by: Benjamin Abendroth <braph93@gmx.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
braph [Thu, 3 Oct 2019 17:06:11 +0000 (19:06 +0200)]
alsamixer: added configuration file parser
Added configparser.c and curskey.c:
- Lines starting with arbitrary whitespace + '#' are comments
- Words in a command name don't have a fixed order (toggle_mute is the
same as mute_toggle)
Moved read_file() from textbox.c to utils.c, so configparser.c can make
use of it.
Added command line options:
-f/-F to specify/disable configuration file
-m/-M to enable/disable mouse
Signed-off-by: Benjamin Abendroth <braph93@gmx.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
braph [Thu, 3 Oct 2019 15:36:52 +0000 (17:36 +0200)]
alsamixer: Use a struct for storing color attributes
This commit is a preparation for the configuration file parser.
The `int attr_*` variables have been moved into a separate struct.
Members of that struct are alphabetically ordered, so an attribute can
later be accessed by its name.
Added `int get_color_pair(short fg, short bg)` for returning or
creating a color pair number.
Added call to `use_default_colors()` for enabling access to the
terminal's default color (-1).
Signed-off-by: Benjamin Abendroth <braph93@gmx.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Mouse support has been added for mixer_widget.c, card_select.c and
proc_files.c.
In the mixer widget the mouse is handled as follows:
- After an element has been printed in mixer_display.c, a call to
clickable_set() will store the coordinates of the drawn area plus the
command enum that should be executed on click. An optional argument
holds an index which points to the selected mixer control.
- on_mouse_click() searches for a matching rectangle, focuses the mixer
control and returns the command enum.
In the menu widgets, the menu_driver() function handles mouse input.
Signed-off-by: Benjamin Abendroth <braph93@gmx.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
alsamixer: Replaced hardcoded keys by binding arrays
This commit is a preparation for making keybindings configurable.
The hardcoded keys in the switch statements of mixer_widget.c,
textbox.c, proc_files.c and card_select.c have been replaced by
command enums. The command that shall be executed is looked up in
the arrays `mixer_bindings` and `textbox_bindings`.
The channel_mask enum has been made public, since LEFT and RIGHT are
used as command arguments in bindings.c.
Signed-off-by: Benjamin Abendroth <braph93@gmx.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This might become a Pandora's box given
the amount of obscure package managers
on the planet but these two account for
most *nix-like systems with ALSA it seems.
Added support for querying ALSA packages
installed through rpm and dpkg; tested
on ALT (rpm) and OS Elbrus (dpkg).
Suggested-by: Anton Farygin <rider@altlinux.org>
See-also: http://bugzilla.altlinux.org/38416 Signed-off-by: Jaroslav Kysela <perex@perex.cz>
alsactl: fix error handling for sched_setscheduler() call
As man page says: "If successful, the sched_setparam() function shall return zero."
Without update I got this output in the syslog (journalctl):
abr 16 09:25:30 mypc alsactl[1652]: alsactl 1.2.2 daemon started
abr 16 09:25:30 mypc alsactl[1652]: /usr/bin/alsactl: do_nice:165sched_setparam failed: No such file or directory
If sched_setscheduler() returns 0, so it means that the call was successful.
Signed-off-by: Oscar Megía <megia.oscar@gmail.com> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Ruslan Bilovol [Mon, 9 Mar 2020 20:29:54 +0000 (22:29 +0200)]
alsaloop: reduce cumulative error caused by non-atomic samples calculation
When doing loopback between two audio card with
same sampling frequency, I noticed slow increase
of pitch_diff.
When I changed order of get_queued_playback_samples()
vs get_queued_capture_samples(), I noticed same drift
of pitch_diff but if was decreasing this time.
This seems to be caused by non-atomic consecutive
snd_pcm_delay() invocation for playback then for
capture. snd_pcm_delay() measures delay between
read/write call and actual ADC/DAC operation.
So while we get this value for playback path in
get_queued_playback_samples(), next call to
get_queued_capture_samples() will happen a little
bit later so snd_pcm_delay() may return incorrect
value.
Be interleaving get_queued_{playback,capture}_samples()
order, we divide this small error between playback
and capture paths. I do not see any issues anymore
with one-way drift of pitch_diff.
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
git://git.alsa-project.org / alsa-utils.git / log