Takashi Sakamoto [Tue, 13 Nov 2018 06:41:34 +0000 (15:41 +0900)]
axfer: add support for non-blocking operation
In alsa-lib PCM API, snd_pcm_read[i|n]() and snd_pcm_write[i|n] can be
used with non-blocking mode. This is available when SND_PCM_NONBLOCK is
used as 'mode' argument for a call of snd_pcm_open().
This commit adds support this type of operation. To reduce CPU usage, this
commit uses 'snd_pcm_wait()' to wait for event notification.
Below lines are examples to execute:
$ axfer transfer -N -P -d 2 -D hw:0,3 /dev/urandom -f dat -vvv
$ axfer transfer -N -C -d 2 -D hw:1,0 /dev/null -r 48000 -vvv
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:33 +0000 (15:41 +0900)]
axfer: add an option to finish transmission at XRUN
In aplay, '--fatal-errors' option has an effect to give up recovery of PCM
substream from XRUN state. This commit adds support for this option.
In original implementation, this option brings program abort. This seems
to generate core dump of process VMA. However, typically, XRUN comes from
timing mismatch between hardware and application, therefore core dump has
less helpful. This commit finishes this program in usual way with this
option at XRUN.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:32 +0000 (15:41 +0900)]
axfer: add options related to duration and obsolete '--max-file-size' option
In aplay, some options are available to stop data transmission by frame
unit. This commit adds support for the options below:
* --duration (-d)
* For duration seconds. The number of data frames transferred in this
* runtime is calculated by this value and sampling rate.
* --samples (-s)
* For the number of data frames to handle in this runtime.
An original aplay has a similar option; '--max-file-time'. This option
is used for capture data transmission to switch file to write data frame
up to maximum number of frames which container format supports, instead
of terminating. However, this may brings complicated file handling to
this program. To reduce maintaining cost, this option is obsoleted.
Additionally, a handler for SIGUSR1 Unix signal has similar feature to
switch the file. For the same reason, the handler is also obsoleted.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:31 +0000 (15:41 +0900)]
axfer: add an option to dump available hardware parameters
In ALSA PCM interface, before configuring hardware actually, applications
can request available set of hardware parameters for runtime of PCM
substream. The set of parameters are represented and delivered by a
structure.
In alsa-lib PCM API, the above design is abstracted by a series of
snd_pcm_hw_params_xxx() functions. An actual layout of the structure is
hidden from applications by an opaque pointer.
In aplay, '--dump-hw-params' option is for this purpose. With this option,
the command output available set of the hardware parameters.
This commit adds support for the option. Unlike aplay, this commit takes
this program to finish after dumping the parameters for simplicity of
usage.
I note that all of combinations in the set are not necessarily available
when the PCM substream includes dependencies of parameters described by
constraints and rules.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:30 +0000 (15:41 +0900)]
axfer: add informative output and an option to suppress it
In current aplay, some informative output is available as a default. This
can be suppressed by a quiet option. This commit adds support for it.
An original aplay implementation has no effect of this option in a case
to handle multiple files. However, in a point of usability, this commit
support this case.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:29 +0000 (15:41 +0900)]
axfer: add a sub-command to transfer data frames
In current aplay, default action is to transfer data frames from/to
devices. This commit adds support for this functionality.
Event loop is included in an added file. In the loop, the number of
handled data frames is manipulated by an appropriate way. As a result, users
can stop data transmission frames by frame.
Unlike aplay, when catching SIGSTP, this application performs to suspend
PCM substream. When catching SIGCONT, it performs to resume the PCM
substream. The aim of this design is to avoid XRUN state of the PCM
substream. If users/developers need to any XRUN-recovery test, it's
better to work for the other ways.
Below lines are examples to execute:
$ axfer transfer -P -D hw:0,3 /dev/urandom -f dat -vvv
$ axfer transfer -C -D hw:1,0 /dev/null -r 48000 -vvv
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:28 +0000 (15:41 +0900)]
axfer: add support for blocking data transmission operation of alsa-lib PCM API
In alsa-lib PCM API, snd_pcm_read[i|n]() and snd_pcm_write[i|n]() are used
to transfer data frames from/to hardware. When a handler is not opened with
specific flags, these functions perform blocking operation; i.e. the
function call doesn't return till all of request number of data frames are
actually handled, or call is interrupted by Unix signals, or PCM substeam
corrupts due to hardware reasons.
This commit adds support for this type of data transmission. For cases that
requested data frames are not processed by container interface, this commit
adds internal cache mechanism to handle rest of data frames in next timing.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:27 +0000 (15:41 +0900)]
axfer: add support to transfer data frames by alsa-lib PCM APIs
This commit adds support fo alsa-lib PCM API as a backend of 'xfer'
module. In a set of alsa-lib PCM API, there're two ways to handle data
frames; by calling ioctl(2) with some specific commands with buffer in
user space, or copying data frames on mapped page frames. To support
both ways, this commit adds an operation structure as abstraction.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:26 +0000 (15:41 +0900)]
axfer: add a parser for command-line options
In aplay, many command-line options are supported. Some of them have
dependency or conflicts. Furthemore, some of them are just for
runtime configuration of alsa-lib(libasound), and some options can
be used by several xfer backends commonly; e.g. options for file name,
sample format and sampling rate.
This commit adds a parser for the common options below.
* --help (-h)
* Just output 'help' string (not written yet).
* --verbose (-v)
* For verbose output, including information about xfer, mapper and
container.
* --format (-f): string. format literals or one of ['cd'|'cdr'|'dat']
* For sample format supported by ALSA PCM interface. Special format
can be used. For playback, this is auto-detected according to actual
file format.
* --channels (-c)
* For the number of samples included in one data frame. For playback,
this is auto-detected according to actual file format, except for
'raw' format. This option can conflict to above format option.
* --rate (-r)
* For the number of data frames transferred in one second. For playback,
this is auto-detected according to actual file format, except for
'raw' format. This option can conflict to format option above.
* --file-type (-f): string. one of ['wav'|'au'|'voc'|'raw']
* For format of files of given paths. For playback, this is optional
because the format is auto-detected. For capture, this is optional too
because the format is decided according to suffix of given path.
Anyway, this option is used for cases to fail to detect or decide.
* --separate-channels (-I)
* When using several files as source or destination for transmission
of data frame, this option can be used with several file paths.
When '--separate-channels' option is used, users can give several file
paths to source/destination of data transmission, else they can give single
file path for the purpose. When multiple files are handled by this option,
for playback, data frames in first channel is used to construct buffer for
data transmission with multi channel. For capture, data frames in each
channel of buffer are written to each of given path. Furthermore, when a
single path is given for capture, file paths are auto-generated according
to available number of channels. For example, 'name.wav' is given for
2 channels capture, 'name-0.wav' and 'name-1.wav' are generated. In a
case of no suffix, 'name-0' and 'name-1' are generated.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:25 +0000 (15:41 +0900)]
axfer: add a common interface to transfer data frames
ALSA has PCM interface to transfer data frames. In userspace, there're
some implementation to utilize this interface to produce application
programming interface; alsa-lib (libasound) and tinyalsa. However, it's
possible to use the interface with raw I/O operations.
This commit adds an common interface to transfer data frames for this
program, named as 'xfer'. This internal interface is designed for users
to select several backend for data transmission. This includes some
functions expected to be called by main program just for data
transmission. In an aspect to maintain PCM substream, suspend feature is
required to handle a pair of SIGTSTP/SIGCONT UNIX signals.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:24 +0000 (15:41 +0900)]
axfer: add a unit test for mapper interface
In former commits, mapper module gets supports of muxer/demuxer for
single/multiple targets for playback source or capture destination. This
commit adds a unit test for them. This includes positive test cases only.
The test cases actually generate I/O to file systems for many test cases.
It takes a bit long time to finish.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:23 +0000 (15:41 +0900)]
axfer: add support for a mapper for multiple target
This commit adds support of mapper for 'multiple' target. This handles
several files via 'container' functions, and constructs data frame buffer
for playback, or splits data frames from data frame buffer for capture.
When playback source files includes data frames with several channels, the
first channel is used to construct buffer. For capture direction, each of
channel of data frame is stored in one file, thus the file includes one
channel of data frame. When handling non-interleaved buffer, a caller
should use an array of buffer for each of channels with non-interleaved
data frames.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:22 +0000 (15:41 +0900)]
axfer: add support for a mapper for single target
In usual use case of aplay, single file is used to playback or capture data
frames.
This commit adds support of single type mapper for this use case. All of
supported file format can include data frame with interleaved alignment,
thus this mapper have a functionality to convert from several types of
data frame alignment to interleaved alignment or vise versa. When
handling non-interleaved buffer, a caller should use an array of buffer
for each of channels with non-interleaved data frames.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:21 +0000 (15:41 +0900)]
axfer: add a common interface to align data frames on different layout
In current aplay, several files can be handled as source of data frames for
playback, or destination of captured data frames by an option
'--separate-channels' (-I).
On the other hand, in ALSA PCM kernel/user interface, several types of
buffer are used to communicate between application/hardware;
- mapped page frame for data frames with interleaved alignment
- mapped page frame for data frames with non-interleaved alignment
- buffer in user space for data frames with interleaved alignment
- a list of buffer in user space for data frames with non-interleaved
alignment
This commit adds a common interface, named as 'mapper' to convert frame
alignment between these two sides. This interface includes two types;
'muxer' and 'demuxer'. The 'muxer' is for playback direction, to
construct playback buffer with PCM frames from several files. The 'demuxer'
is for capture direction, to split PCM frames from capture buffer to
each of file. Unlike multimedia containers such as MPEG 2/4 Systems,
the 'muxer' and 'demuxer' are for playback/capture buffer, not for file
contents.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:20 +0000 (15:41 +0900)]
axfer: add unit test for container interface
In former commits, container module gets supports of parser/builder for
several types of file format. This commit adds a unit test for them.
This includes positive test cases only. The test cases actually generate
I/O to file systems for many test cases. It takes a long time to finish.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:19 +0000 (15:41 +0900)]
axfer: add support for a container of raw data
This commit adds support for raw data without any headers/chunks/blocks.
A parser of container cannot recognize format of sample without
supplemental information.
Additionally, it includes no magic bytes. A parser of container should
process first several bytes as a part of PCM frames, instead of magic
bytes.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:18 +0000 (15:41 +0900)]
axfer: add support for a container of Creative Tech. voice format
This commit adds support for data of Creative Tech. voice format. In this
data format, values in each of field are represented in little-endian byte
order and available formats of data sample are restricted in little-endian
byte order.
In version 1.10 of this format, sampling rate is represented with
reciprocal number of the rate, thus we cannot calculate original sampling
rate precisely just from its header. For example at 44.1kHz, file header
includes 233 (=256-1,000,000/44,100), but we cannot recover the value just
from the code (43478.2...). For my convenience, this commit adds a
pre-computed table and lookup major rates from the table.
Additionally, this format can includes several blocks with different
sample format. When handling this type of file, we need to start/stop
substream for each of the block, while this brings complicated code.
This type of format is enough ancient and presently quite minor. This
commit takes a compromise and handles a first sample block only.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:17 +0000 (15:41 +0900)]
axfer: add support for a container of Sparc AU format
This commit adds support for data of Sparc AU format. In this data format,
values in each of field are encoded in big-endian byte order and available
formats of data sample are restricted in big-endian byte order.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:16 +0000 (15:41 +0900)]
axfer: add support for a container of Microsoft/IBM RIFF/Wave format
This commit adds support for data of Microsoft/IBM RIFF/Wave format. In
this data format, values in each of field are encoded in both bit/little
byte order but inner a file the same order is used. Magic bytes in the
beginning of data indicated which byte order is used for the file.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:15 +0000 (15:41 +0900)]
axfer: add a common interface to handle a file with audio-specific data format
Current aplay supports several types of data format for file; Microsoft/IBM
RIFF/Wave (.wav), Sparc AU (.au) and Creative Tech. voice (.voc). These
formats were designed to handle audio-related data with interleaved frame
alignment.
This commit adds a common interface to handle the file format, named as
'container' module. This includes several functions to build/parse
the format data from any file descriptors. Furthermore, this includes
several helper functions for implementations of each builder/parser.
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:14 +0000 (15:41 +0900)]
axfer: add a sub-command to print list of PCMs/devices
Original aplay implementation has a feature to output two types of list;
devices and PCMs. The list of devices is a result to query sound card and
pcm component structured maintained in kernel land. The list of PCMs is a
result to parse runtime configuration files in alsa-lib. Entries in the
former list is corresponding to ALSA PCM character device
('/dev/snd/pcm%uC%uD[p|c]'), while entries in the latter list includes
some 'virtual' instances in application runtime.
This commit adds an implementation for the above functionality. This is
executed by taking 'list' sub-command. A 'device' option has the same
effect as '--list-devices' and '-L' of aplay. A 'pcm' option has the same
effect as '--list-pcms' and '-l' of aplay. In both cases, an additional
option is required for stream direction. Below is examples of new command
system for this sub-command.
$ axfer list device -C (= arecord --list-devices)
$ axfer list pcm -P (= aplay -l)
Takashi Sakamoto [Tue, 13 Nov 2018 06:41:13 +0000 (15:41 +0900)]
axfer: add an entry point for this command
This commit adds a new command, 'axfer' ('ALSA transfer'), to transfer data
frames described in asound.h. This command is intended to replace current
aplay. The most of features and command line parameters come from aplay as
much as possible, while it has more better feature and code to maintain.
This commit adds an entry point for this command. Current option system of
aplay is still available, while this command has a sub-command system like
commands in iproute2.
Currently, two sub-commands are supported; 'list' and 'transfer'. The
'list' sub-command has the same effect as '-l' and '-L' options of aplay.
The 'transfer' sub-command has the same effect as the main feature of
aplay. For the sub-command system, an option for stream direction is
required; '-P' for playback and '-C' for capture. If you create symbolic
links to this binary for aplay/arecord, please execute:
$ ln -s axfer aplay
$ ln -s axfer arecord
Actual code for each sub-command will be implemented in later commits.
aplay: add a paragraph for '--samples' ('-s') option to aplay manual
A '--samples' ('-s') option was added so that record/playback process is
terminated after handling the same number of PCM frames as a value of the
option. However this option is not described in aplay manual.
This commit adds a paragraph for the option.
Fixes: 3d44e2bc159e ('aplay: Add samples argument for playing/recording a given number of samples') Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Reviewed-by: Daniel Baluta daniel.baluta@nxp.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
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>
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>
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>
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>
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>
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".
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.
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.
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>
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>
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.
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.
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.
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
!!---------------
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
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/
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>
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.
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.
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.
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>
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>
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>
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>
git://git.alsa-project.org / alsa-utils.git / log