alsa-utils.git
3 days agoRelease v1.1.7 master v1.1.7
Jaroslav Kysela [Tue, 16 Oct 2018 08:08:36 +0000 (10:08 +0200)]
Release v1.1.7

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

5 days agoalsamixer: fix gcc warnings
Jaroslav Kysela [Sun, 14 Oct 2018 15:03:08 +0000 (17:03 +0200)]
alsamixer: fix gcc warnings

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

5 days agoalsactl: lock - fix the array size (gcc warning)
Jaroslav Kysela [Sun, 14 Oct 2018 15:01:38 +0000 (17:01 +0200)]
alsactl: lock - fix the array size (gcc warning)

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

5 days agoaplay: add missing block brackets
Jaroslav Kysela [Sun, 14 Oct 2018 15:00:26 +0000 (17:00 +0200)]
aplay: add missing block brackets

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

5 days agoalsaloop: pcmjob - fix few warnings
Jaroslav Kysela [Sun, 14 Oct 2018 14:59:28 +0000 (16:59 +0200)]
alsaloop: pcmjob - fix few warnings

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

5 days agoalsactl: use signalfd to catch UNIX signal
Takashi Sakamoto [Sun, 14 Oct 2018 14:36:34 +0000 (23:36 +0900)]
alsactl: use signalfd to catch UNIX signal

In a mode of 'monitor, event loop runs to dispatch asynchronous event
emitted by control node. In this case, UNIX signal is used to terminate
the event loop.

This commit uses signalfd to catch the UNIX signal.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

5 days agoalsactl: handle detection of new sound card
Takashi Sakamoto [Sun, 14 Oct 2018 14:36:33 +0000 (23:36 +0900)]
alsactl: handle detection of new sound card

At present, plug-and-play is not supported in a mode of 'monitor',
thus new sound card is not handled during runtime. This is not happy.

This commit uses Linux-specific inotify(7) to monitor '/dev/snd'
directory. When some files are newly added to the directory,
event dispatcher is suspended. Event sources are scanned again and the
dispatcher continue to run.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

5 days agoalsactl: handle disconnection of sound card
Takashi Sakamoto [Sun, 14 Oct 2018 14:36:32 +0000 (23:36 +0900)]
alsactl: handle disconnection of sound card

Once sound card becomes disconnection state, corresponding control node
becomes to emit error event for listeners. When catching this type of
event, event dispatcher should stop observation of the node. However,
at present, a mode of monitor can't handle this correctly. As a result,
poll(2) is executed quite frequently in loop with no wait. This results
100% consumption of CPU time.

This commit takes the dispatcher to remove the node from observation
list when detecting the disconnection state.

Reported-by: Thomas Gläßle <thomas@coldfix.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

5 days agoalsactl: obsolete array for maintenance of handlers
Takashi Sakamoto [Sun, 14 Oct 2018 14:36:31 +0000 (23:36 +0900)]
alsactl: obsolete array for maintenance of handlers

In former commits, handlers of control node are maintained by link list,
instead of one-dimensional array.

This commit obsoletes the array and split source preparation to a
function.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

5 days agoalsactl: use a list of source for event dispatcher instead of an array of source
Takashi Sakamoto [Sun, 14 Oct 2018 14:36:30 +0000 (23:36 +0900)]
alsactl: use a list of source for event dispatcher instead of an array of source

In a previous commit, handlers of control nodes are maintained by link
list.

This commit uses the list to register/unregister event sources to
dispatcher.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

5 days agoalsactl: use link list to maintain source of events
Takashi Sakamoto [Sun, 14 Oct 2018 14:36:29 +0000 (23:36 +0900)]
alsactl: use link list to maintain source of events

At present, handlers for control nodes are maintained by one-dimensional
array. This is not necessarily useful to maintain handlers with
associated information.

This commit adds link-list for the maintenance.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

5 days agoalsactl: use epoll(7) instead of poll(2)
Takashi Sakamoto [Sun, 14 Oct 2018 14:36:28 +0000 (23:36 +0900)]
alsactl: use epoll(7) instead of poll(2)

Linux kernel supports unique system call; epoll(7). This allows
applications to make associations for descriptor-unique data in a
easy way.

This commit uses epoll(7) instead of poll(2) for this point.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

5 days agoalsactl: add an iterator of registered instances of sound card
Takashi Sakamoto [Sun, 14 Oct 2018 14:36:27 +0000 (23:36 +0900)]
alsactl: add an iterator of registered instances of sound card

In a mode of 'monitor', when given no argument, all of available control
node is observed for their events. At present, discovering the nodes is
done according to sound card number, instead of listing nodes in
configuration space of alsa-lib.

This commit adds a structure to discover sound cards with a simple
interface.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

5 days agoalsactl: split event loop code to a function
Takashi Sakamoto [Sun, 14 Oct 2018 14:36:26 +0000 (23:36 +0900)]
alsactl: split event loop code to a function

In a mode of 'monitor', an event loop runs.

This commit applies a small refactoring to splits the loop into a
function for readability.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

10 days agoalsatplg: add man file
Jaroslav Kysela [Tue, 9 Oct 2018 08:53:08 +0000 (10:53 +0200)]
alsatplg: add man file

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

6 weeks agoalsaucm: add alsa-ucm udev rules for PAZ00 (Toshiba AC100/Dynabook AZ).
Jaroslav Kysela [Fri, 7 Sep 2018 08:53:19 +0000 (10:53 +0200)]
alsaucm: add alsa-ucm udev rules for PAZ00 (Toshiba AC100/Dynabook AZ).

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

2 months agoalsabat: Allow custom sample format for round trip latency test
Jonathan Liu [Sun, 5 Aug 2018 03:59:35 +0000 (13:59 +1000)]
alsabat: Allow custom sample format for round trip latency test

Setting the format to BAT_PCM_FORMAT_S16_LE in the round trip latency
test initialization is redundant as it is already set by default to
BAT_PCM_FORMAT_S16_LE unless a sample format is specified on the command
line.

Signed-off-by: Jonathan Liu <net147@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

4 months agospeaker-test: Allow sampling rates up to 768000
Julian Scheel [Thu, 7 Jun 2018 09:10:55 +0000 (11:10 +0200)]
speaker-test: Allow sampling rates up to 768000

There are audio devices around that support up to 768kHz playback, allow
testing them by increasing the maximum supported sampling rate.

Signed-off-by: Julian Scheel <julian@jusst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

4 months agospeaker-test: Remove unused variable
Julian Scheel [Wed, 23 May 2018 13:42:21 +0000 (15:42 +0200)]
speaker-test: Remove unused variable

Signed-off-by: Julian Scheel <julian@jusst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

4 months agospeaker-test: Support S24_3LE sample format
Julian Scheel [Wed, 23 May 2018 13:42:20 +0000 (15:42 +0200)]
speaker-test: Support S24_3LE sample format

Implement support signed 24 bit samples, packed in 3 bytes.

Signed-off-by: Julian Scheel <julian@jusst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

5 months agoaplay: Fix invalid file size check for non-regular files
Takashi Iwai [Tue, 15 May 2018 20:17:01 +0000 (22:17 +0200)]
aplay: Fix invalid file size check for non-regular files

aplay tries to check the file size via fstat() at parsing the format
headers and avoids parsing when the size is shorter than the given
size.  This works fine for regular files, but when a special file like
pipe is passed, it fails, eventually leading to the fallback mode
wrongly.

A proper fix is to do this sanity check only for a regular file.

Reported-by: Jay Foster <jay@systech.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

6 months agoRelease v1.1.6
Jaroslav Kysela [Tue, 3 Apr 2018 08:58:10 +0000 (10:58 +0200)]
Release v1.1.6

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

6 months agobat: alsa.c - move the thread cleanup pop before goto exit3
Jaroslav Kysela [Tue, 3 Apr 2018 08:48:52 +0000 (10:48 +0200)]
bat: alsa.c - move the thread cleanup pop before goto exit3

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

7 months agoaplay: Fix wav file not being split on 32 bit platforms
erwin [Tue, 13 Mar 2018 18:51:24 +0000 (19:51 +0100)]
aplay: Fix wav file not being split on 32 bit platforms

On my 32 bit armhf board arecord exits because of write() returning EFBIG
when the output file size reaches 2147483647 bytes.

To fix this, include generated header file before system header files
so that _FILE_OFFSET_BITS=64 is used properly, as required in documentation
"man feature_test_macros".

Signed-off-by: Takashi Iwai <tiwai@suse.de>

7 months agospeaker-test: Refactor the tone-generator codes
Takashi Iwai [Mon, 5 Mar 2018 15:10:42 +0000 (16:10 +0100)]
speaker-test: Refactor the tone-generator codes

There are many redundant open codes in speaker-test for performing the
similar things, and especially the tone generator codes are ugly.
Let's clean up a bit.  This patch combines all open-codes into a
single common helper with the callback for generating the tone.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

8 months agoalsaloop: fix a typo in the comparison
Kirill Marinushkin [Mon, 5 Feb 2018 06:47:11 +0000 (07:47 +0100)]
alsaloop: fix a typo in the comparison

Hello maintainers,

I would like to suggest you a patch which fixes a typo in the alsa-utils
alsaloop.

Best Regards,
Kirill Marinushkin

Signed-off-by: Kirill Marinushkin <k.marinushkin@gmail.com>
Cc: patch@alsa-project.org
Cc: alsa-devel@alsa-project.org
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

10 months agoalsactl: Only start restore service when asoundrc file exists
Ikey Doherty [Tue, 12 Dec 2017 13:32:34 +0000 (13:32 +0000)]
alsactl: Only start restore service when asoundrc file exists

This solves the chicken and egg problem on fresh installations whereby
the alsa state file does not yet exist, and alsa-restore unit attempted
to launch without first having a state file.

Signed-off-by: Ikey Doherty <ikey@solus-project.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

10 months agoaplay: Adjust sample rate limits to support newer hardware
Jussi Laako [Thu, 7 Dec 2017 11:57:14 +0000 (13:57 +0200)]
aplay: Adjust sample rate limits to support newer hardware

There are number of devices that support up to 384 kHz sampling rate and
some devices up to 768 kHz sampling rate. This patch increases sanity
check limit to 768k in order to support testing of such hardware.

Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

11 months agoChange FSF address (Franklin Street)
Jaroslav Kysela [Tue, 14 Nov 2017 13:28:51 +0000 (14:28 +0100)]
Change FSF address (Franklin Street)

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

11 months agoRelease v1.1.5 v1.1.5
Jaroslav Kysela [Tue, 14 Nov 2017 07:52:09 +0000 (08:52 +0100)]
Release v1.1.5

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

12 months agoalsactl: Move systemd unit start-up from basic.target to sound.target
Chris Mayo [Tue, 26 Sep 2017 18:36:12 +0000 (19:36 +0100)]
alsactl: Move systemd unit start-up from basic.target to sound.target

Ensures soundcard is ready before restoring state.

sound.target added to systemd in v18:
https://cgit.freedesktop.org/systemd/systemd/commit/?id=88dfa2938af

Simplify dependencies:
 - After=alsa-state.service is not needed because both units test for
   @daemonswitch@ with opposite outcomes.

 - After=sysinit.target is automatically added by systemd.

First proposed by Tom Yan.

Signed-off-by: Chris Mayo <aklhfex@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

13 months agoalsabat: fix one uninitialized warning issue
Zhang Keqiao [Wed, 30 Aug 2017 01:26:16 +0000 (09:26 +0800)]
alsabat: fix one uninitialized warning issue

Fix a variable uninitialized issue, adding the initialized assignment to fix it.

Signed-off-by: Zhang Keqiao <keqiaox.k.zhang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

14 months agoaplay: Fix playback for small raw files
Daniel Baluta [Tue, 8 Aug 2017 22:06:00 +0000 (01:06 +0300)]
aplay: Fix playback for small raw files

This fixes a bug when trying to play files with size
smaller than maximum supported header size.

Lets have a look at the following example:

$ aplay -s 2 sample.raw

-> playback_go(fd = 10, loaded = 26, count = 2, name="sample.raw")
--> l = loaded = 26
--> c = count - written = 2
--> c -= l = 2 - 26 = -24
---> r = safe_read(fd, audiobuf + 26, -24)
---> r = -1, EXIT_FAILURE

In this case we have already 'loaded' from the input file more
bytes that we need to send to pcm device. So, we need to adjust
the number of bytes loaded and avoid reading a negative number
of bytes.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

14 months agoaplay: Refactor playback code
Daniel Baluta [Tue, 8 Aug 2017 22:05:59 +0000 (01:05 +0300)]
aplay: Refactor playback code

This introduces read_header function which tries
to read the header of an audio file in order to determine
its type.

This has the following effects:
(1) makes code easier to read
(2) don't abort if file size is less than expected header

(2), allows us to play small files with size smaller than any
supported audio file headers.

Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

14 months agoaplay: Add samples argument for playing/recording a given number of samples
Ion-Horia Petrisor [Tue, 8 Aug 2017 22:05:58 +0000 (01:05 +0300)]
aplay: Add samples argument for playing/recording a given number of samples

-s --samples allows aplay to be used for playback/capture a given
number of samples per channel

Signed-off-by: Ion-Horia Petrisor <ion-horia.petrisor@nxp.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

15 months agoaplay: interrupt streaming via signal in voc_pcm_write
Srikanth Krishnakar [Mon, 19 Dec 2016 08:34:39 +0000 (14:04 +0530)]
aplay: interrupt streaming via signal in voc_pcm_write

aplay/arecord (alsa-utils v1.1.2) cannot interrupt streaming
via CTRL-C. Fixed the issue by properly handling 'in_aborting'
flag in appropriate functions.

Signed-off-by: Anant Agrawal <Anant_Agrawal@mentor.com>
Signed-off-by: Mikhail Durnev <mikhail_durnev@mentor.com>
Signed-off-by: Srikanth Krishnakar <Srikanth_Krishnakar@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

15 months agoaplay: Fix --max-file-time option 32 bits overflow
Scott Gilliland [Fri, 23 Jun 2017 18:35:03 +0000 (18:35 +0000)]
aplay: Fix --max-file-time option 32 bits overflow

Fix bug in arecord --max-file-time where the file size could overflow
32 bits.

Signed-off-by: Scott Gilliland <scott.gilliland@gatech.edu>
Acked-by: John Sauter <John_Sauter@systemeyescomputerstore.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

16 months agoalsatplg: fix topology compiler long option parsing
Liam Girdwood [Wed, 14 Jun 2017 11:25:33 +0000 (12:25 +0100)]
alsatplg: fix topology compiler long option parsing

verbose, compile and output options all have a parameter.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

16 months agotopology: delete output file if parsing fails.
Liam Girdwood [Fri, 9 Jun 2017 15:33:42 +0000 (16:33 +0100)]
topology: delete output file if parsing fails.

Currently the binary output file is left when parsing fails. This confuses
GNU Make if the parsing fails and causes the compilation to partially
complete.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

17 months agoRelease v1.1.4 v1.1.4
Jaroslav Kysela [Fri, 12 May 2017 08:01:46 +0000 (10:01 +0200)]
Release v1.1.4

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

18 months agoaplay: Introduce and use xwrite helper
Daniel Baluta [Mon, 10 Apr 2017 07:04:33 +0000 (10:04 +0300)]
aplay: Introduce and use xwrite helper

Write can return less then requested bytes, but we treat this as
an error thus ending up with confusing error messages.

Fix this by introducing xwrite helper, which makes sure all bytes
are written or an error is returned.

With this patch an usecase where disk is filled by recording will
print:
$ /mnt/msc/audio.wav: No space left on device

instead of random messages like:

$/mnt/msc/audio.wav: No such file or directory

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

19 months agoamidi: optarg might be NULL, fix 't' argument parsing
Jaroslav Kysela [Mon, 27 Feb 2017 08:04:08 +0000 (09:04 +0100)]
amidi: optarg might be NULL, fix 't' argument parsing

21 months agoalsa-info: add ACPI device status
Pierre-Louis Bossart [Tue, 10 Jan 2017 00:32:24 +0000 (18:32 -0600)]
alsa-info: add ACPI device status

BIOS vendors typically reuse the same definitions between different
platforms and expose the relevant hardware by changing the value of
the _STA method.

For example on the Asus T100HA, there are 3 HID values for audio
codecs in the DSDT table but two have a zero status and will be
ignored by the ACPI subsystem.

$ more /sys/bus/acpi/devices/10EC*/status
::::::::::::::
/sys/bus/acpi/devices/10EC3270:00/status
::::::::::::::
15
::::::::::::::
/sys/bus/acpi/devices/10EC5640:00/status
::::::::::::::
0
::::::::::::::
/sys/bus/acpi/devices/10EC5648:00/status
::::::::::::::
0

This information is very useful to figure out which HIDs/quirks need
to be supported. Add log to alsa-info.sh to only expose non-zero
results of the ACPI _STA method, e.g.

!!ACPI Device Status Information
!!---------------

/sys/bus/acpi/devices/10EC3270:00/status   15

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

21 months agoalsa-info: provide more DMI information
Pierre-Louis Bossart [Tue, 10 Jan 2017 00:32:23 +0000 (18:32 -0600)]
alsa-info: provide more DMI information

Some manufacturers don't provide useful information for Manufacturer
and Product Name but instead use Board Vendor and Board Name fields,
add them to alsa-info log

Example on Intel NUC:

!!DMI Information
!!---------------

Manufacturer:
Product Name:
Product Version:
Firmware Version:  KYSKLi70.86A.0042.2016.0929.1933
Board Vendor:      Intel Corporation
Board Name:        NUC6i7KYB

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

21 months agoalsactl: Remove standard output definition in systemd unit
Paul Menzel [Tue, 8 Jul 2014 07:23:06 +0000 (07:23 +0000)]
alsactl: Remove standard output definition in systemd unit

`/lib/systemd/system/alsa-restore.service` specifies
`StandardOutput=syslog`. This overrides the `DefaultStandardOutput`
setting from `/etc/systemd/system.conf`, which the system administrator
can use to specify how output gets logged. In particular, the sysadmin
may want output to go to the journal, or to syslog, or nowhere at all [1].

This patch removes the definition entirely, so the units can use the
system default.

Upstream the patch from the Debian package [2].

[1] https://bugs.debian.org/741123
    "systemd services should not use StandardOutput=syslog; should rely
     on DefaultStandardOutput"
[2] https://sources.debian.net/src/alsa-utils/1.1.2-1/debian/patches/systemd_standardoutput.patch/

Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
CC: Jordi Mallach <jordi@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

21 months agoalsaucm: Add alsaucm.rst to EXTRA_DIST
Takashi Iwai [Wed, 28 Dec 2016 14:58:51 +0000 (15:58 +0100)]
alsaucm: Add alsaucm.rst to EXTRA_DIST

Otherwise it's missing in the tarball.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

21 months agoRelease v1.1.3 v1.1.3
Jaroslav Kysela [Tue, 20 Dec 2016 09:12:29 +0000 (10:12 +0100)]
Release v1.1.3

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

22 months agoalsactl: Fix potential NULL dereferences in daemon mode
Takashi Iwai [Fri, 9 Dec 2016 16:28:47 +0000 (17:28 +0100)]
alsactl: Fix potential NULL dereferences in daemon mode

The code releasing the each card object may access to NULL when a
bogus count is given.  Add a NULL check just to make sure.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

22 months agoINSTALL: document how to configure a build for installation in a local dir
Antonio Ospite [Fri, 9 Dec 2016 13:02:32 +0000 (14:02 +0100)]
INSTALL: document how to configure a build for installation in a local dir

Signed-off-by: Antonio Ospite <ao2@ao2.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

22 months agoAdd alsaucm.1 to .gitignore
Takashi Iwai [Fri, 9 Dec 2016 16:24:14 +0000 (17:24 +0100)]
Add alsaucm.1 to .gitignore

Signed-off-by: Takashi Iwai <tiwai@suse.de>

22 months agoalsaucm: add a man page, generated from reStructuredText
Antonio Ospite [Fri, 9 Dec 2016 13:02:31 +0000 (14:02 +0100)]
alsaucm: add a man page, generated from reStructuredText

Signed-off-by: Antonio Ospite <ao2@ao2.it>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

22 months agoconfigure.ac: add a check for rst2man, a reStructuredText man page generator
Antonio Ospite [Fri, 9 Dec 2016 13:02:30 +0000 (14:02 +0100)]
configure.ac: add a check for rst2man, a reStructuredText man page generator

Define a USE_RST2MAN conditional so that, when available, rst2man can be
used to generate man pages from reStructuredText source files.

The code follows what is done to check for xmlto.

On Debian system, the rst2man executable is provided by python-docutils
or python3-docutils.

Signed-off-by: Antonio Ospite <ao2@ao2.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

22 months agoconfigure.ac: fix the check for xmlto availability
Antonio Ospite [Fri, 9 Dec 2016 13:02:29 +0000 (14:02 +0100)]
configure.ac: fix the check for xmlto availability

The same $xmlto variable is used both in AC_ARG_ENABLE and
AC_CHECK_PROG, but the latter is not setting a value to it when the
program is not found.

These two facts result in the "yes" value from the AC_ARG_ENABLE macro
to be still kept in the variable when the program is not found by
AC_CHECK_PROG, causing USE_XMLTO to be always set, finally resulting in
a build failure in case the xmlto program is not actually in the PATH.

As possible fix could have been to set "no" as a value in AC_CHECK_PROG
when program is not found.

However using two separate variables is more explicit, so fix the issue
this way.

Signed-off-by: Antonio Ospite <ao2@ao2.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

22 months agoalsaucm: mention the "list1" command in the usage output
Antonio Ospite [Fri, 9 Dec 2016 13:02:28 +0000 (14:02 +0100)]
alsaucm: mention the "list1" command in the usage output

Signed-off-by: Antonio Ospite <ao2@ao2.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

22 months agospeaker-test: Fix chmap wav file selection.
Arnaud Pouliquen [Wed, 7 Dec 2016 16:44:26 +0000 (17:44 +0100)]
speaker-test: Fix chmap wav file selection.

The channel selection currently does not work properly when a channel
map control is provided but no manual channel map was explicitly
requested with "-m".

For example, the CEA/HDMI 6ch (surround 5.1) map is:
 FL, FR, LFE, FC, RL, RR.

Tested command: speaker-test -D hdmi -c 6 -t wav

Speaker-test tries to play channels in this following order:
 0 - Front Left
 3 - Front Center
 1 - Front Right
 5 - Rear Right
 4 - Rear Left
 2 - LFE

But wav file played on associated speakers are not aligned. Here are
the real files played:
 0- /usr/share/sounds/alsa/Front_Left.wav => OK
 3- /usr/share/sounds/alsa/Rear_Right.wav  => OK
 1- /usr/share/sounds/alsa/Front_Right.wav  => OK
 5- /usr/share/sounds/alsa/Rear_Center.wav => KO
 4- found file /usr/share/sounds/alsa/Front_Center.wav => KO
 2- /usr/share/sounds/alsa/Rear_Left.wav  => KO

Issue is that associated wav files ordering is reworked only if
channel_map_set variable is set.

Fix consists in allowing wavs re-ordering if a channel mapping as been
get or set, i.e. channel_map is not null.

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

22 months agoaplay: Fix to handle pause when system is suspended/Resumed
Jeeja KP [Mon, 28 Nov 2016 16:32:00 +0000 (22:02 +0530)]
aplay: Fix to handle pause when system is suspended/Resumed

If PCM is paused and then we do system supend-resume, the stream throws
error(EBADF) when stream is paused released.

Check the pcm state before pause/release and if stream is suspended,
call snd_pcm_resume to resume the stream.

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsaloop: try adapt avail_min for playback to avoid 100% CPU usage
Jaroslav Kysela [Tue, 18 Oct 2016 11:57:19 +0000 (13:57 +0200)]
alsaloop: try adapt avail_min for playback to avoid 100% CPU usage

2 years agoamidi: add sysex-interval option
Felipe F. Tonello [Tue, 30 Aug 2016 16:02:48 +0000 (17:02 +0100)]
amidi: add sysex-interval option

This patch adds a new option to amidi tool: sysex-interval.

It adds a delay (in milliseconds) in between each SysEx message - it searches
for a 0xF7 byte.

This is very useful when sending firmware updates to a remote device via SysEx
or any other use that requires this delay in between SysEx messages.

`amidi' manual was updated with an example usage as well.

Signed-off-by: Felipe F. Tonello <eu@felipetonello.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: add system power management S3 test
Keqiao, Zhang [Fri, 26 Aug 2016 15:37:55 +0000 (23:37 +0800)]
alsabat: add system power management S3 test

Support audio pause/resume for playback and capture. The user can
pause alsabat playback/capture threads by sending a signal. The patch
provides a method for QA to quick test audio during system s3.

Signed-off-by: Keqiao, Zhang <keqiao.zhang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: fix alsabat -86 error
Keqiao, Zhang [Fri, 26 Aug 2016 15:37:54 +0000 (23:37 +0800)]
alsabat: fix alsabat -86 error

alsabat reports -86 error when system suspend and resume. Check the
return value of read_to_pcm() and write_to_pcm(), when -x8 err is
detected, do resume and wait for read/write to pcm to complete.

Write PCM device error: Streams pipe error(-86)
Read PCM device error: Streams pipe error(-86)
*** Error in alsabat: double free or corruption (out): 0x00007fb438001810 ***

Signed-off-by: Keqiao, Zhang <keqiao.zhang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: add amixer config files
Focus Luo [Mon, 22 Aug 2016 16:16:50 +0000 (00:16 +0800)]
alsabat: add amixer config files

This patch includes the reference asound.state config files
on Intel Skylake, Broadwell and Hsawell platforms

Signed-off-by: Focus Luo <focus.luo@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: automation test scripts
Focus Luo [Mon, 22 Aug 2016 16:16:49 +0000 (00:16 +0800)]
alsabat: automation test scripts

This patch includes automated test scripts for linux audio driver
based on alsa-lib interface by using alsabat as test tool.
It supports analog and display(HDMI/DP) audio test.
The package needs the alsa-utils, alsa-lib installed environment.

Signed-off-by: Focus Luo <focus.luo@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoamidi: fix timeout handling
Clemens Ladisch [Sat, 13 Aug 2016 14:41:58 +0000 (16:41 +0200)]
amidi: fix timeout handling

The timeout is not supposed to expire when ignored messages are
received.  This cannot be handled with the poll() timeout, so add
a separate timer.

Reported-by: Martin Tarenskeen <m.tarenskeen@gmail.com>
Reviewd-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoamidi: ignore not only Active Sensing but also Clock bytes
Clemens Ladisch [Sat, 13 Aug 2016 14:41:23 +0000 (16:41 +0200)]
amidi: ignore not only Active Sensing but also Clock bytes

Active Sensing messages are sent by many devices in the background and
would only interfere with the actual messages that amidi is supposed to
capture.  Therefore, amidi ignores them by default.  However, there are
also devices that send Clock messages with the same problem, so it is
a better idea to filter them out, too.

Reported-by: Martin Tarenskeen <m.tarenskeen@gmail.com>
Reviewd-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoRelease v1.1.2 v1.1.2
Jaroslav Kysela [Tue, 2 Aug 2016 17:09:45 +0000 (19:09 +0200)]
Release v1.1.2

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

2 years agoalsabat: make snr_is_valid static
Vinod Koul [Tue, 19 Jul 2016 09:05:09 +0000 (14:35 +0530)]
alsabat: make snr_is_valid static

The compilation fails due to multiple defination of snr_is_valid

common.o: In function `snr_is_valid':
bat/common.h:99: multiple definition of `snr_is_valid'
bat.o:bat/common.h:99: first defined here

signal.o: In function `snr_is_valid':
bat/common.h:99: multiple definition of `snr_is_valid'
bat.o:bat/common.h:99: first defined here

latencytest.o: In function `snr_is_valid':
bat/common.h:99: multiple definition of `snr_is_valid'
bat.o:bat/common.h:99: first defined here

analyze.o: In function `snr_is_valid':
bat/common.h:99: multiple definition of `snr_is_valid'
bat.o:bat/common.h:99: first defined here

alsa.o: In function `snr_is_valid':
bat/common.h:99: multiple definition of `snr_is_valid'
bat.o:bat/common.h:99: first defined here

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: fix a missing break in switch
Lu, Han [Sun, 12 Jun 2016 10:17:00 +0000 (18:17 +0800)]
alsabat: fix a missing break in switch

Add the break line for OPT_ROUNDUPLATENCY case.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: add noise detection
Lu, Han [Wed, 8 Jun 2016 19:42:49 +0000 (03:42 +0800)]
alsabat: add noise detection

Alsabat reports error when noise above threshold be detected.
Use either of the options below to designate the threshold. (e.g.
if the ratio of noise to signal is 5%, the snr is about 26dB.)
    --snr-db <value in dB>
    --snr-pc <value in %>

The noise detection is performed in time domain. On each period
of the sine wave being analyzed, alsabat substracts a clean sine
wave from the source, calculates the RMS value of the residual,
and compares the result with the threshold. At last, alsabat
returns the number of periods with noise above threshold. 0 is
returned when the source is clean.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: add a single channel sine wave generator
Lu, Han [Wed, 8 Jun 2016 19:42:48 +0000 (03:42 +0800)]
alsabat: add a single channel sine wave generator

Add function generate_sine_wave_raw_mono(). It serves as a single
channel sine wave generator, to provide data for calculation (e.g.
for noise analysis).
The function is similar to generate_sine_wave(), but a lite revision.
It has no dependency on bat channels and target frequency, no malloc
inside, no data conversion from float to integer samples, and supports
one channel only.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: add channels parameter for adjust_waveform()
Lu, Han [Wed, 8 Jun 2016 19:42:47 +0000 (03:42 +0800)]
alsabat: add channels parameter for adjust_waveform()

The function adjust_waveform() is a component of generate_sine_wave(),
and depended on bat->channels parameter. Add parameter "channels" to
remove the dependency, and then adjust_waveform() can be applied on
other use cases, e.g. a single channel sine wave generator.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: align the data type on float
Lu, Han [Wed, 8 Jun 2016 19:42:46 +0000 (03:42 +0800)]
alsabat: align the data type on float

Aligning the data type of fftw analyzer, sample converter and other
components on float, because:
  1. avoid unnecessary data type conversion;
  2. using float is more efficient than using double;
  3. the extra double accuracy is not required.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: add round trip audio latency test
vivian,zhang [Fri, 3 Jun 2016 02:05:08 +0000 (10:05 +0800)]
alsabat: add round trip audio latency test

Audio latency is the time delay as an audio signal passes through
a system. There are many kinds of audio latency metrics. One useful
metric is the round trip latency, which is the sum of output latency
and input latency.

The measurement step works like below:
1. Listen and measure the average loudness of the environment for
one second;
2. Create a threshold value 16 decibels higher than the average
loudness;
3. Begin playing a ~1000 Hz sine wave and start counting the samples
elapsed;
4. Stop counting and playing if the input's loudness is higher than
the threshold, as the output wave is probably coming back;
5. Calculate the audio latency value in milliseconds.

Signed-off-by: Zhang Vivian <vivian.zhang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: fix a possible memory leak
Lu, Han [Wed, 1 Jun 2016 08:54:28 +0000 (16:54 +0800)]
alsabat: fix a possible memory leak

Fix a possible memory leak in generate_sine_wave(). Memory free was
ignored when the function return an error.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: add buffer size and period size settings
vivian,zhang [Tue, 31 May 2016 07:31:32 +0000 (15:31 +0800)]
alsabat: add buffer size and period size settings

Add buffer size and period size settings in alsabat.
With -E and -B options, alsabat performs the test with
specified buffer size and period size

Signed-off-by: Zhang Vivian <vivian.zhang@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: add terminate status check for capture thread
Lu, Han [Sun, 17 Apr 2016 01:26:45 +0000 (09:26 +0800)]
alsabat: add terminate status check for capture thread

In loopback test, alsabat use pthread_join(pthread_t thread, **retval)
to wait for the capture thread to terminate. If the capture thread was
canceled, PTHREAD_CANCELED is placed in *retval, and the access to the
**retval will fail. Add status check to prevent illegal access to the
**retval.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoaplay: fix lurking capture file overwrite bug
David Fries [Thu, 14 Apr 2016 04:32:46 +0000 (23:32 -0500)]
aplay: fix lurking capture file overwrite bug

If -d was given to arecord while commit
8aa13eec80eac312e4b99423909387660fb99b8f (now reverted) was in effect,
the last read would be shorter than the chunk size, but pcm_read would
read and return the chunk size, the samples were discarded, and
capture() continued in a loop because count never reached 0.  arecord
opens a new file each loop iteration, if arecord is dynamically naming
files, --use-strftime option or beyond the wave 2GB limit, this will
generate a series of header only wave files.  If the file is unique
the originally recorded data is lost and it will continue overwriting
the same file with a header only wave file.

While the current pcm_read can't fail (it can exit), it is better to
just fix this lurking bug in case it is "fixed" again.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoRelease v1.1.1 v1.1.1
Jaroslav Kysela [Thu, 31 Mar 2016 14:37:02 +0000 (16:37 +0200)]
Release v1.1.1

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

2 years agoalsa-info: add alsa-info.sh.1 to EXTRA_DIST
Jaroslav Kysela [Thu, 31 Mar 2016 14:35:26 +0000 (16:35 +0200)]
alsa-info: add alsa-info.sh.1 to EXTRA_DIST

2 years agoalsabat: add tinyalsa support
Lu, Han [Wed, 23 Mar 2016 07:52:47 +0000 (15:52 +0800)]
alsabat: add tinyalsa support

Use "configure --enable-alsabat-backend-tiny" for alsabat to use
tinyalsa as backend lib. On a system that has both ALSA and tinyalsa
installed, alsabat will use ALSA library by default.
The intention is for alsabat to run on tinyalsa platforms such as
Android or some Internet of Things(IoT) devices.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: move alsa process to a single block
Lu, Han [Wed, 23 Mar 2016 07:52:46 +0000 (15:52 +0800)]
alsabat: move alsa process to a single block

Move all alsa callings to a single block (alsa.c), so other blocks
such as the main structure, the signal process and the data analysis
modules will be independent to alsa, and new modules such as a
tinyalsa interface can be easily embedded into alsabat.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: use general data generator function
Lu, Han [Wed, 23 Mar 2016 07:52:45 +0000 (15:52 +0800)]
alsabat: use general data generator function

Use general data generator to replace local function, so other
modules can reuse the data generator rather than re-implement it.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: clean return value for playback and capture threads
Lu, Han [Wed, 23 Mar 2016 07:52:44 +0000 (15:52 +0800)]
alsabat: clean return value for playback and capture threads

Remove unnecessary prints in playback and capture threads, and replace
the return value "0" with error code for convenience of maintaining.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: use general function for wav header update
Lu, Han [Wed, 23 Mar 2016 07:52:43 +0000 (15:52 +0800)]
alsabat: use general function for wav header update

In playback thread, use general function update_wav_header()
to replace a bunch of code, so the structure is cleaner and
no need to define variable "wav".

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: refactoring alsa capture thread
Lu, Han [Wed, 23 Mar 2016 07:52:42 +0000 (15:52 +0800)]
alsabat: refactoring alsa capture thread

Refactoring ALSA capture thread:
  1. Move file open/seek operations to sub function, so all file
  processes are now on a single function (read_from_pcm_loop()), so
  the structure is more reasonable, the function API is simplified
  and no need file cleanup in thread loop.
  2. Replace the wav header processing lines with a general function
  (update_wav_header()), which can be reused in other sections.
  3. Add pthread_exit() for thread to exit safely in single line mode,
  and correct comment.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsactl: init/ca0106, init/hda - use CTL{values} instead CTL{value}
Jaroslav Kysela [Tue, 22 Mar 2016 15:53:30 +0000 (16:53 +0100)]
alsactl: init/ca0106, init/hda - use CTL{values} instead CTL{value}

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

2 years agoalsactl: remove debug line in set_ctl_values()
Jaroslav Kysela [Tue, 22 Mar 2016 15:50:31 +0000 (16:50 +0100)]
alsactl: remove debug line in set_ctl_values()

Signed-off-by: Jaroslav Kysela <perex@perex.cz>

2 years agoalsabat: use variable for thread return value
Lu, Han [Mon, 21 Mar 2016 11:05:49 +0000 (19:05 +0800)]
alsabat: use variable for thread return value

Use variable instead of 0/1 to indicate the return value of
playback and capture threads.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: fix fopen and messages
Lu, Han [Mon, 21 Mar 2016 11:05:48 +0000 (19:05 +0800)]
alsabat: fix fopen and messages

All files should be opened in either "rb" or "wb" in current
usage.
Remove incorrect and unneccesary prints.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: fix misusing of errno
Lu, Han [Mon, 21 Mar 2016 11:05:47 +0000 (19:05 +0800)]
alsabat: fix misusing of errno

Preserve errno value before use, since the value might be
changed by another library call.
Add "#include <errno.h>" and remove redundant include.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: add bash test script
Lu, Han [Tue, 15 Mar 2016 03:18:54 +0000 (11:18 +0800)]
alsabat: add bash test script

Add bash script for alsabat feature test. It covers basic alsabat
features, including waveform and wav file generate, playback,
capture and analysis with configurable parameters under different
work modes. (loopback, single line, standalone and local mode)
Usage examples:
        alsabat-test.sh
        alsabat-test.sh plughw:1,0
        alsabat-test.sh plughw:1,0 plughw:1,0

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: add standalone mode
Lu, Han [Tue, 15 Mar 2016 03:18:53 +0000 (11:18 +0800)]
alsabat: add standalone mode

Add support for standalone mode where alsabat will run on a
different machine to the one being tested.
In standalone mode, the alsabat just generates, playback and
capture sound data like in normal mode, but does not analyze.
The alsabat being built without libfftw3 support is always work
in standalone mode.
The alsabat in normal mode can also bypass data analysis using
option "--standalone".

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: add default device name for playback and capture
Lu, Han [Tue, 15 Mar 2016 03:18:52 +0000 (11:18 +0800)]
alsabat: add default device name for playback and capture

Add default name for the playback and capture devices, in case
they were not set by user through '-D', '-P' or '-C' options.
Previously, if no device be specified, the alsabat will start
a playback thread and a capture thread, and then exit the
threads with error log.
If only one of playback and capture is specified, the alsabat
will work on single line mode as before, where only one thread
(playback or capture) will be started.
The patch was tested on Ubuntu and Chrome OS.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoaplay/arecord: handle parsing errors of parameter values
erwin [Fri, 11 Mar 2016 11:40:30 +0000 (11:40 +0000)]
aplay/arecord: handle parsing errors of parameter values

when a user enters a command, he expects his command to be executed
as specified or aborted if it is impossible to fulfill his request

right now a command like "arecord -d hw:1,0 --max-file-time 1h recording.wav"
will happily record something and exit without error status while the resulting
recording contains definitely not what the user requested

to fix this, the patch handles the number parsing function's error channel
and checks whether the parsed number has any trailing characters

Signed-off-by: erwin <nebelbank@posteo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoaconnect: Show sequencer sound card numer/PID via aconnect
Martin Koegler [Tue, 8 Mar 2016 22:06:47 +0000 (23:06 +0100)]
aconnect: Show sequencer sound card numer/PID via aconnect

rawmidi devices expose the card number via IOCTLs, which allows to
find the corresponding device in sysfs.

The sequencer provides no identifing data. Chromium works around this
issue by scanning rawmidi as well as sequencer devices and matching
them by using assumtions, how the kernel register sequencer devices.

This patch adds support for displaying the sound card number/PID to
aconnect.

Signed-off-by: Martin Koegler <martin.koegler@chello.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: truncate sample frames for faster FFT analysis
Lu, Han [Mon, 29 Feb 2016 02:33:45 +0000 (10:33 +0800)]
alsabat: truncate sample frames for faster FFT analysis

Truncate the sample frames to powers of 2, since the FFTW algorithm
runs especially fast in this case, and other sizes may be computed
by means of a slow, general-purpose algorithm.
In my test environment applying the patch, a sound clip of 33072
frames is cut off to 32768 frames before analysis, and the time
cost is reduced from 6.128s to 0.224s.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoUpdate .gitignore
Takashi Iwai [Wed, 3 Feb 2016 07:50:28 +0000 (08:50 +0100)]
Update .gitignore

bat was renamed to alsabat.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsabat: rename to avoid naming conflict
Lu, Han [Wed, 3 Feb 2016 06:20:33 +0000 (14:20 +0800)]
alsabat: rename to avoid naming conflict

alsa-utils as well as bareos-bat (as well a some Bacula packages)
all contain a program called /usr/bin/bat, which causes conflicts on
various distributions ("basic audio tester" vs "bareos administration
tool"("bacula administration tool")).
Rename to avoid conflict.

Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoamixer: skip showing asoc tlv byte controls
Vinod Koul [Thu, 28 Jan 2016 08:32:18 +0000 (14:02 +0530)]
amixer: skip showing asoc tlv byte controls

ASoC TLV Byte controls are very large size controls so we should add new
options for these. So skip dumping contents for these.

$amixer -c0 cget numid=16
numid=16,iface=MIXER,name='mdl params'
  ; type=BYTES,access=-----RW-,values=30336
  ; ASoC TLV Byte control, skipping bytes dump

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 years agoalsa-info.sh: add man page
Jaroslav Kysela [Wed, 13 Jan 2016 08:46:31 +0000 (09:46 +0100)]
alsa-info.sh: add man page

2 years agobat: Don't pass incompatible function pointers to pthread_cleanup_push()
Takashi Iwai [Mon, 9 Nov 2015 13:09:50 +0000 (14:09 +0100)]
bat: Don't pass incompatible function pointers to pthread_cleanup_push()

pthread_cleanup_push() takes a function pointer for void (void *).
Although it may work in most cases, we shouldn't pass an incompatible
function pointer there, as some old gcc complains:
  alsa.c:560: warning: initialization from incompatible pointer type
  alsa.c:562: warning: initialization from incompatible pointer type

Signed-off-by: Takashi Iwai <tiwai@suse.de>