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.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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)
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Either '--samples' ('-s') and '--duration' ('-d') option is available
exclusively, according to its semantics and actual implementation.
This commit improves description of manual at this point.
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>
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: 3d44e2bc15 ('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>
A '--sleep-min' option was already obsoleted for aplay. On the other hand,
a paragraph for the option was left as is.
This commit deletes the paragraph.
Fixes: 4cb74aed89 ('Remove sleep_min from aplay')
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>
This include is not used/needed and prevents building on systems that
don't provide <dlfcn.h>.
Signed-off-by: Thomas Petazzoni <thomas.petazzoni@bootlin.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
1) do not increase avail_min forever
It seems that there are broken plugins like pulse which returns from poll()
immediately regardless avail_min settings.
2) remove ommited debug printf()
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
