alsa-utils/axfer/axfer-transfer.1

.TH AXFER\-TRANSFER 1 "28 November 2018" "alsa\-utils"

.SH NAME
axfer\-transfer \- transferrer of audio data frame for sound devices and nodes.

.SH SYNOPSIS

.B axfer transfer
.I direction
[
.I common\-options
] [
.I backend\-options
] [
.I filepath
]

.B axfer transfer
.I direction
[
.I common\-options
] [
.I backend\-options
]
.I \-I
|
.I \-\-separate\-channels filepath ...

direction =
.B capture
|
.B playback

common\-options = ( read
.I OPTIONS
section )

backend\-options = ( read
.I OPTIONS
section )

filepaths = ( read
.I OPTIONS
section )

.SH DESCRIPTION
The
.B transfer
subcommand of
.B axfer
performs transmission of audio data frames for devices available in supported
backends. This program is essentially designed to use alsa\-lib APIs
(libasound backend) to handle sound devices supported by Linux sound subsystem
(ALSA).

.SH OPTIONS

.SS Direction

.TP
.B capture
Operates for capture transmission.

.TP
.B playback
Operates for playback transmission.

.SS Filepath

Filepath is handled as a path relative to current working directory of run time
if it\(aqs not full path from root directory.

The standard input or output is used if filepath is not specified or given as
.I \(aq\-\(aq
\&.

For playback transmission, container format of given
.I filepath
is detected automatically and metadata is used for parameters of sample format,
channels, rate, duration. If nothing detected, content of given file path is
handled as raw data. In this case, the parameters should be indicated as
options.

Multiple
.I filepaths
are allowed with
.I \-I
|
.I \-\-separate\-channels
option. In this case, standard input and output is not available. The same
.I filepath
is not allowed except for paths listed below:
 \- /dev/null
 \- /dev/zero
 \- /dev/full
 \- /dev/random
 \- /dev/urandom

.SS Common options

.TP
.B \-h, \-\-help
Print help messages and finish run time.

.TP
.B \-q, \-\-quiet
Quiet mode. Suppress messages (not sound :))

.TP
.B \-v, \-\-verbose
Verbose mode. Runtime dumps supplemental information according to the number of
this option given in command line.

.TP
.B \-d, \-\-duration=#
Interrupt after # seconds. A value of zero means infinity. The default is zero,
so if this option is omitted then the transmission process will run until it is
killed. Either
.I \-d
or
.I \-s
option is available exclusively.

.TP
.B \-s, \-\-samples=#
Interrupt after transmission of # number of data frames. A value of zero means
infinity. The default is zero, so if this options is omitted then the
transmission process will run until it is killed. Either
.I \-d
or
.I \-s
option is available exclusively.

.TP
.B \-f, \-\-format=FORMAT
Indicate format of audio sample. This is required for capture transmission, or
playback transmission with files including raw audio data.

Available sample format is listed below:
 - [S8|U8|S16|U16|S32|U32][_LE|_BE]
 - [S24|U24][_LE|_BE]
 - FLOAT[_LE|_BE]
 - FLOAT64[_LE|_BE]
 - IEC958_SUBFRAME[_LE|_BE]
 - MU_LAW
 - A_LAW
 - [S20|U20][_LE|_BE]
 - [S24|U24][_3LE|_3BE]
 - [S20|U20][_3LE|_3BE]
 - [S18|U18][_3LE|_3BE]
 - DSD_U8
 - DSD_[U16|U32][_LE|_BE]

If endian\-ness is omitted, host endian\-ness is used.

Some special formats are available:
 - cd (16 bit little endian, 44100, stereo) [= \-f S16_LE \-c 2 \-r 44100]
 - cdr (16 bit big endian, 44100, stereo) [= \-f S16_BE \-c 2 \-f 44100]
 - dat (16 bit little endian, 48000, stereo) [= \-f S16_LE \-c 2 \-r 48000]

If omitted,
.I U8
is used as a default. Actual available formats are restricted by each
transmission backend.

Unavailable sample format is listed below. These format has size of data frame
unaligned to byte unit.

 - IMA_ADPCM
 - MPEG
 - GSM
 - SPECIAL
 - G723_24
 - G723_24_1B
 - G723_40
 - G723_40_1B

.TP
.B \-c, \-\-channels=#
Indicate the number of audio data samples per frame. This is required for
capture transmission, or playback transmission with files including raw audio
data. The value should be between
.I 1 to
.I 256
\&. If omitted,
.I 1
is used as a default.

.TP
.B \-r, \-\-rate=#
Indicate the number of audio data frame per second. This is required for
capture transmission, or playback transmission with files including raw audio
data. If the value is less than
.I 1000
, it\(aqs interpreted by
.I kHz
unit. The value should be between
.I 2000
and
.I 192000
\&. If omitted,
.I 8000
is used as a default.

.TP
.B \-t, \-\-file\-type=TYPE
Indicate the type of file. This is required for capture transmission. Available
types are listed below:
 - wav: Microsoft/IBM RIFF/Wave format
 - au, sparc: Sparc AU format
 - voc: Creative Tech. voice format
 - raw: raw data

When nothing is indicated, for capture transmission, the type is decided
according to suffix of
.I filepath
, and
.I raw
type is used for fallback.

.TP
.B \-I, \-\-separate\-channels
Indicate this option when several files are going to be handled. For capture
transmission, if one filepath is given as
.I filepath
, a list of
.I filepaths
is generated in a formula \(aq<filepath>\-<sequential number>[.suffix]\(aq.
The suffix is omitted when raw format of container is used.

.TP
.B \-\-dump\-hw\-params
Dump hardware parameters and finish run time if backend supports it.

.TP
.B \-\-xfer\-backend=BACKEND
Select backend of transmission from a list below. The default is libasound.
.br
 - libasound
 - libffado (optional if compiled)

.SS Backend options for libasound

.TP
.B \-D, \-\-device=NODE

This option is used to select PCM node in libasound configuration space.
Available nodes are listed by
.I pcm
operation of
.I list
subcommand.

.TP
.B \-N, \-\-nonblock

With this option, PCM substream is opened in non\-blocking mode. When audio
data frame is not available in buffer of the PCM substream, I/O operation
immediately returns without blocking process. This option implicitly uses
.I \-\-waiter\-type
option as well to prevent heavy consumption of CPU time.

.TP
.B \-M, \-\-mmap

With this option, audio data frame is processed directly in buffer of PCM
substream if selected node supports this operation. Without the option,
temporary buffers are used to copy audio data frame for buffer of PCM substream.
This option implicitly uses
.I \-\-waiter\-type
option as well to prevent heavy consumption of CPU time.

.TP
.B \-F, \-\-period\-size=#

This option configures given value to
.I period_size
hardware parameter of PCM substream. The parameter indicates the number of audio
data frame per period in buffer of the PCM substream. Actual number is decided
as a result of interaction between each implementation of PCM plugin chained
from the selected PCM node, and in\-kernel driver or PCM I/O plugins.

Ideally, the same amount of audio data frame as the value should be handled in
one I/O operation. Actually, it is not, depending on implementation of the PCM
plugins, in\-kernel driver, PCM I/O plugins and scheduling model. For \(aqhw\(aq
PCM plugin in \(aqirq\(aq scheduling model, the value is used to decide
intervals of hardware interrupt, thus the same amount of audio data frame as
the value is expected to be available for one I/O operation.

.TP
.B \-\-period\-time=#

This option configures given value to
.I period_time
hardware parameter of PCM substream. This option is similar to
.I \-\-period\-size
option, however its unit is micro\-second.

.TP
.B \-B, \-\-buffer\-size=#

This option configures given value to
.I buffer_size
hardware parameter of PCM substream. The parameter indicates the number of audio
data frame in buffer of PCM substream. Actual number is decided as a result of
interaction between each implementation of PCM plugin chained from the selected
PCM node, and in\-kernel driver or PCM I/O plugins.

Ideally, this is multiples of the number of audio data frame per period, thus
the size of period. Actually, it is not, depending on implementation of the PCM
plugins, in\-kernel driver and PCM I/O plugins.

.TP
.B \-\-buffer\-time=#

This option configures given value to
.I buffer_time
hardware parameter of PCM substream. This option is similar to
.I \-\-buffer\-size
option, however its unit is micro\-second.

.TP
.B \-\-waiter\-type=TYPE

This option indicates the type of waiter for event notification. At present,
four types are available;
.I default
,
.I select
,
.I poll
and
.I epoll
\&. With
.I default
type, \(aqsnd_pcm_wait()\(aq is used. With
.I select
type, \(aqselect(2)\(aq system call is used. With
.I poll
type, \(aqpoll(2)\(aq system call is used. With
.I epoll
type, Linux\-specific \(aqepoll(7)\(aq system call is used.

This option should correspond to one of
.I \-\-nonblock
or
.I \-\-mmap
options, or
.I timer
value of
.I \-\-sched\-model
option.
Neither this option nor
.I \-\-test\-nowait
is available at the same time.

.TP
.B \-\-sched\-model=MODEL

This option selects scheduling model for process of this program. One of
.I irq
or
.I timer
is available. In detail, please read \(aqSCHEDULING MODEL\(aq section.

When nothing specified,
.I irq
model is used.

.TP
.B \-A, \-\-avail\-min=#

This option configures given value to
.I avail\-min
software parameter of PCM substream. In blocking mode, the value is used as
threshold of the number of available audio data frames in buffer of PCM
substream to wake up process blocked by I/O operation. In non\-blocking mode,
any I/O operation returns \-EAGAIN until the available number of audio data frame reaches the threshold.

This option has an effect in cases neither
.I \-\-mmap
nor
.I timer
value of
.I \-\-sched\-model
option is used.

.TP
.B \-R, \-\-start\-delay=#

This option configures given value to
.I start_threshold
software parameter of PCM substream. The value is used as threshold to start
PCM substream automatically. At present, this option has an effect in cases
neither
.I \-\-mmap
nor
.I timer
value of
.I \-\-sched\-model
option is used.

For playback transmission, when the number of accumulated audio data frame
in buffer of PCM substream to which this program writes out reaches the
threshold, the PCM substream starts automatically without an explicit call of
.I snd_pcm_start()
to the PCM substream.

For capture transmission, this option is useless. The number of
accumulated audio data frame is not increased without an explicit call of
.I snd_pcm_start()
to the PCM substream.

This option has an effect in cases neither
.I \-\-mmap
nor
.I timer
value of
.I \-\-sched\-model
option is used.

.TP
.B \-T, \-\-stop\-delay=#

This option configures given value to
.I stop_threshold
software parameter of PCM substream. The value is used as threshold to stop PCM
substream automatically. At present, this option has an effect in cases neither
.I \-\-mmap
nor
.I timer
value of
.I \-\-sched\-model
option is used.

For capture transmission, when the number of accumulated audio data frame
in buffer of PCM substream to which a driver or alsa\-lib PCM plugins write
reaches the threshold, the PCM substream stops automatically without an explicit
call of
.I snd_pcm_stop()
to the PCM substream. This is a case that this program leaves the audio data
frames without reading for a while.

For playback transmission, when the number available audio data frame in buffer
of PCM substream from which a driver or alsa\-lib PCM plugins read reaches the
threshold, the PCM substream stops automatically without an explicit call of
.I snd_pcm_stop()
to the PCM substream. This is a case that this program leaves the audio data
frames without writing for a while.

This option has an effect in cases neither
.I \-\-mmap
nor
.I timer
value of
.I \-\-sched\-model
option is used.

.TP
.B \-\-disable\-resample

This option has an effect for \(aqplug\(aq plugin in alsa\-lib to suppress
conversion of sampling rate for audio data frame.

.TP
.B \-\-disable\-channels

This option has an effect for \(aqplug\(aq plugin in alsa\-lib to suppress
conversion of channels for audio data frame.

.TP
.B \-\-disable\-format

This option has an effect for \(aqplug\(aq plugin in alsa\-lib to suppress
conversion of sample format for audio data frame.

.TP
.B \-\-disable\-softvol

This option has an effect for \(aqsoftvol\(aq plugin in alsa\-lib to suppress
conversion of samples for audio data frame via additional control element.

.TP
.B \-\-fatal\-errors

This option suppresses recovery operation from XRUN state of running PCM
substream, then process of this program is going to finish as usual.

.TP
.B \-\-test\-nowait

This option disables any waiter for I/O event notification. I/O operations are
iterated till any of audio data frame is available. The option brings heavy
load in consumption of CPU time.

.SS Backend options for libffado

This backend is automatically available when configure script detects
.I ffado_streaming_init()
symbol in libffado shared object.

.TP
.B \-p, \-\-port=#

This option uses given value to decide which 1394 OHCI controller is used to
communicate. When Linux system has two 1394 OHCI controllers,
.I 0
or
.I 1
are available. Neither this option nor
.I \-g
is available at the same time. If nothing specified, libffado performs to
communicate to units on IEEE 1394 bus managed by all of 1394 OHCI controller available in Linux system.

.TP
.B \-n, \-\-node=#

This option uses given value to decide which unit is used to communicate. This
option requires
.I \-p
option to indicate which 1394 OHCI controller is used to communicate to the
specified unit.

.TP
.B \-g, \-\-guid=HEXADECIMAL

This option uses given value to decide a target unit to communicate. The value
should be prefixed with '0x' and consists of hexadecimal literal letters
(0\-9, a\-f, A\-F). Neither this option nor
.I \-p
is available at the same time. If nothing specified, libffado performs to
communicate to units on IEEE 1394 bus managed by all of 1394 OHCI controller
available in Linux system.

.TP
.B \-\-frames\-per\-period=#

This option uses given value to decide the number of audio data frame in one
read/write operation. The operation is blocked till the number of available
audio data frame exceeds the given value. As a default, 512 audio data frames
is used.

.TP
.B \-\-periods\-per\-buffer=#

This option uses given value to decide the size of intermediate buffer between
this program and libffado. As a default, 2 periods per buffer is used.

.TP
.B \-\-slave

This option allows this program to run slave mode. In this mode, libffado
adds unit directory into configuration ROM of 1394 OHCI controller where Linux
system runs. The unit directory can be found by the other node on the same bus.
Linux system running on the node can transfer isochronous packet with audio
data frame to the unit. This program can receive the packet and demultiplex the
audio data frame.

.TP
.B \-\-snoop

This option allows this program to run snoop mode. In this mode, libffado
listens isochronous channels to which device transfers isochronous packet. When
isochronous communication starts by any unit on the same bus, the packets can
be handled by this program.

.TP
.B \-\-sched\-priority=#

This option executes
.I pthread_setschedparam()
in a call of
.I ffado_streaming_init()
to configure
scheduling policy and given value as its priority for threads related to
isochronous communication.
The given value should be within
.I RLIMIT_RTPRIO
parameter of process. Please read
.I getrlimit(2)
for details.

.SH POSIX SIGNALS
During transmission,
.I SIGINT
and
.I SIGTERM
will close handled files and PCM substream to be going to finish run time.

.I SIGTSTP
will suspend PCM substream and
.I SIGCONT
will resume it. No XRUNs are expected. With libffado backend, the suspend/resume
is not supported and runtime is aboeted immediately.

The other signals perform default behaviours.

.SH EXAMPLES

.PP
.in +4n
.EX
.B $ axfer transfer playback \-d 1 something
.EE
.in
.PP

The above will transfer audio data frame in \(aqsomething\(aq file for playback
during 1 second.  The sample format is detected automatically as a result to
parse \(aqsomething\(aq as long as it\(aqs compliant to one of Microsoft/IBM
RIFF/Wave, Sparc AU, Creative Tech. voice formats. If nothing detected,
.I \-r
,
.I \-c
and
.I \-f
should be given,
or
.I \-f
should be given with special format.

.PP
.in +4n
.EX
.B $ axfer transfer playback \-r 22050 \-c 1 \-f S16_LE \-t raw something
.EE
.in
.PP

The above will transfer audio data frame in \(aqsomething\(aq file including no
information of sample format, as sample format of 22050 Hz, monaural, signed 16
bit little endian PCM for playback. The transmission continues till catching
.I SIGINT
from keyboard or
.I SIGTERM
by
.I kill(1)
\&.

.PP
.in +4n
.EX
.B $ axfer transfer capture \-d 10 \-f cd something.wav
.EE
.in
.PP

The above will transfer audio data frame to \(aqsomething.wav\(aq file as
sample format of 44.1 kHz, 2 channels, signed 16 bit little endian PCM, during
10 seconds. The file format is Microsoft/IBM RIFF/Wave according to suffix of
the given
.I filepath
\&.

.PP
.in +4n
.EX
.B $ axfer transfer capture \-s 1024 \-r 48000 \-c 2 \-f S32_BE \-I \-t au channels
.EE
.in
.PP

The above will transfer audio data frame as sample format of 48.0 kHz, 2
channels, signed 32 bit big endian PCM for 1,024 number of data frames to files
named \(aqchannels\-1.au\(aq and \(aqchannels\-2.au\(aq.

.SH SCHEDULING MODEL

In a design of ALSA PCM core, runtime of PCM substream supports two modes;
.I period\-wakeup
and
.I no\-period\-wakeup.
These two modes are for different scheduling models.

.SS IRQ\-based scheduling model

As a default,
.I period\-wakeup
mode is used. In this mode, in\-kernel drivers should operate hardware to
generate periodical notification for transmission of audio data frame. The
interval of notification is equivalent to the same amount of audio data frame
as one period of buffer, against actual time.

In a handler assigned to the notification, a helper function of ALSA PCM core
is called to update a position to head of hardware transmission, then compare
it with a position to head of application operation to judge overrun/underrun
(XRUN) and to wake up blocked processes.

For this purpose, hardware IRQ of controller for serial audio bus such as
Inter\-IC sound is typically used. In this case, the controller generates the
IRQ according to transmission on the serial audio bus. In the handler assigned
to the IRQ, direct media access (DMA) transmission is requested between
dedicated host memory and device memory.

If target hardware doesn't support this kind of mechanism, the periodical
notification should be emulated by any timer; e.g. hrtimer, kernel timer.
External PCM plugins generated by PCM plugin SDK in alsa\-lib should also
emulate the above behaviour.

In this mode, PCM applications are programmed according to typical way of I/O
operations. They execute blocking system calls to read/write audio data frame
in buffer of PCM substream, or blocking system calls to wait until any audio
data frame is available. In
.I axfer
, this is called
.I IRQ\-based
scheduling model and a default behaviour. Users can explicitly configure this
mode by usage of
.I \-\-sched\-model
option with
.I irq
value.

.SS Timer\-based scheduling model

The
.I no\-period\-wakeup
mode is an optional mode of runtime of PCM substream. The mode assumes a
specific feature of hardware and assist of in\-kernel driver and PCM
applications. In this mode, in\-kernel drivers don't operate hardware to
generate periodical notification for transmission of audio data frame.
The hardware should automatically continue transmission of audio data frame
without periodical operation of the drivers; e.g. according to auto\-triggered
DMA transmission, a chain of registered descriptors.

In this mode, nothing wakes up blocked processes, therefore PCM applications
should be programmed without any blocking operation. For this reason, this mode
is enabled when the PCM applications explicitly configure hardware parameter to
runtime of PCM substream, to prevent disorder of existing applications.
Additionally, nothing maintains timing for transmission of audio data frame,
therefore the PCM applications should voluntarily handle any timer to queue
audio data frame in buffer of the PCM substream for lapse of time. Furthermore,
instead of driver, the PCM application should call a helper function of ALSA
PCM core to update a position to head of hardware transmission and to check
XRUN.

In
.I axfer
, this is called
.I timer\-based
scheduling model and available as long as hardware/driver assists
.I no\-period\-wakeup
runtime. Users should explicitly set this mode by usage of
.I \-\-sched\-model
option with
.I timer
value.

In the scheduling model, PCM applications need to care of available space on
PCM buffer by lapse of time, typically by yielding CPU and wait for
rescheduling. For the yielding, timeout is calculated for preferable amount of
PCM frames to process. This is convenient to a kind of applications, like sound
servers. when an I/O thread of the server wait for the timeout, the other
threads can process audio data frames for server clients. Furthermore, with
usage of rewinding/forwarding, applications can achieve low latency between
transmission position and handling position even if they uses large size of
PCM buffers.

.SS Advantages and issues

Ideally, timer\-based scheduling model has some advantages than IRQ\-based
scheduling model. At first, no interrupt context runs for PCM substream. The
PCM substream is handled in any process context only. No need to care of race
conditions between IRQ and process contexts. This reduces some concerns for
some developers of drivers and applications. Secondary, CPU time is not used
for handlers on the interrupt context. The CPU time can be dedicated for the
other tasks. This is good in a point of Time Sharing System. Thirdly, hardware
is not configured to generate interrupts. This is good in a point of reduction
of overall power consumption possibly.

In either scheduling model, the hardware should allow drivers to read the
number of audio data frame transferred between the dedicated memory and the
device memory for audio serial bus. However, in timer\-based scheduling model,
fine granularity and accuracy of the value is important. Actually hardware
performs transmission between dedicated memory and device memory for a small
batch of audio data frames or bytes. In a view of PCM applications, the
granularity in current transmission is required to decide correct timeout for
each I/O operation. As of Linux kernel v4.21, ALSA PCM interface between
kernel/userspace has no feature to report it.

.SH COMPATIBILITY TO APLAY

The
.B transfer
subcommand of
.B axfer
is designed to keep compatibility to aplay(1). However some options below are
not compatible due to several technical reasons.

.TP
.I \-I, \-\-separate\-channels
This option is supported just for files to store audio data frames corresponding
to each channel. In aplay(1) implementation, this option has an additional
effect to use PCM buffer aligned to non\-interleaved order if a target device
supports. As of 2018, PCM buffer of non\-interleaved order is hardly used by
sound devices.

.TP
.I \-A, \-\-avail\-min=#
This option indicates threshold to wake up blocked process in a unit of
audio data frame. Against aplay(1) implementation, this option has no effect
with
.I \-\-mmap
option as well as
.I timer
of
.I \-\-sched\-model
option.

.TP
.I \-R, \-\-start\-delay=#
This option indicates threshold to start prepared PCM substream in a unit of
audio data frame. Against aplay(1) implementation, this option has no effect
with
.I \-\-mmap
option as well as
.I timer
of
.I \-\-sched\-model
option.

.TP
.I \-T, \-\-stop\-delay=#
This option indicates threshold to stop running PCM substream in a unit of
audio data frame. Against aplay(1) implementation, this option has no effect
with
.I \-\-mmap
option as well as
.I timer
of
.I \-\-sched\-model
option.

.TP
.I \-\-max\-file\-time=#
This option is unsupported. In aplay(1) implementation, the option has an
effect for capture transmission to save files up to the same number of data
frames as the given value by second unit, or the maximum number of data frames
supported by used file format. When reaching to the limitation, used file is
closed, then new file is opened and audio data frames are written. However, this
option requires extra handling of files and shall increase complexity of main
loop of axfer.

.TP
.I \-\-use\-strftime=FORMAT
This option is unsupported. In aplay(1) implementation, the option has an effect
for capture transmission to generate file paths according to given format in
which some extra formats are available as well as formats supported by
strftime(3). However, this option requires extra string processing for file
paths and it\(aqs bothersome if written in C language.

.TP
.I \-\-process\-id\-file=FILEPATH
This option is unsupported. In aplay(1) implementation, the option has an effect
to create a file for given value and write out process ID to it. This file
allows users to get process ID and send any POSIX signal to aplay process.
However, this idea has some troubles for file locking when multiple aplay
processes run with the same file.

.TP
.I \-V, \-\-vumeter=TYPE
This option is not supported at present. In aplay(1) implementation, this option
has an effect to occupy stdout with some terminal control characters and display
vumeter for monaural and stereo channels. However, some problems lay; this
feature is just for audio data frames with PCM format, this feature brings
disorder of terminal after aborting, stdout is not available for pipeline.

.TP
.I \-i, \-\-interactive
This option is not supported at present. In aplay(1) implementation, this option
has an effect to occupy stdin for key input and suspend/resume PCM substream
according to pushed enter key. However, this feature requires an additional
input handling in main loop and leave bothersome operation to maintain PCM
substream.

.TP
.I \-m, \-\-chmap=CH1,CH2,...
ALSA PCM core and control core doesn't support this feature, therefore
remapping should be done in userspace. This brings overhead to align audio
data frames, especially for mmap operation. Furthermore, as of alsa-lib v1.1.8,
some plugins don't support this feature expectedly, thus this option is a lack
of transparent operation. At present, this option is not supported yet not to
confuse users.

.TP
.I SIGTSTP, SIGCONT
This performs suspend/resume of PCM substream. In aplay(1) implementation,
these operations bring XRUN state to the substream, and suspend/resume is done
in interactive mode in the above. Some developers use the signal for recovery
test from XRUN. At present, no alternative is supported for the test.

.TP
.I SIGUSR1
This is not supported. In aplay(1) implementation, this signal is assigned to a
handler to close a current file to store audio data frame and open a new file
to continue processing. However, as well as
.I \-\-max\-file\-time
option, this option should increase complexity of main loop of axfer.

.SH DESIGN

.SS Modular structure

This program consists of three modules;
.I xfer
,
.I mapper
and
.I container
\&.
Each module has an abstraction layer to enable actual implementation.

.nf
           --------     ----------     -------------
device <-> | xfer | <-> | mapper | <-> | container | <-> file
           --------     ----------     -------------
            libasound    single         wav
            libffado     multiple       au
                                        voc
                                        raw
.fi

The
.I xfer
module performs actual transmission to devices and nodes. The module can have
several transmission backends. As a default backend,
.I libasound
backend is used to perform transmission via alsa\-lib APIs. The module allows
each backend to parse own command line options.

The
.I container
module performs to read/write audio data frame via descriptor for file/stream
of multimedia container or raw data. The module automatically detect type of
multimedia container and parse parameters in its metadata of data header. At
present, three types of multimedia containers are supported; Microsoft/IBM
RIFF/Wave (
.I wav
), Sparc AU (
.I au
) and Creative Technology voice (
.I voc
). Additionally, a special container is prepared for raw audio data (
.I raw
).

The
.I mapper
module handles buffer layout and alignment for transmission of audio data frame.
The module has two implementations;
.I single
and
.I multiple
\&.
The
.I single
backend uses one container to construct the buffer. The
.I multiple
backend uses several containers to construct it.

.SS Care of copying audio data frame

Between the
.I xfer
module and
.I mapper
module, a pointer to buffer including audio data frames is passed. This buffer
has two shapes for interleaved and non\-interleaved order. For the former, the
pointer points to one buffer. For the latter, the pointer points to an array in
which each element points to one buffer. Between the
.I mapper
module and
.I container
module, a pointer to one buffer is passed because supported media containers
including raw type store audio data frames in interleaved order.

In passing audio data frame between the modules, axfer is programmed to avoid
copying between a buffer to another buffer as much as possible. For example, in
some scenarios below, no copying occurs between modules.

 - xfer(mmap/interleaved), mapper(single), container(any)
 - xfer(mmap/non\-interleaved), mapper(multiple), containers(any)

.SS Unit test

For each of the
.I mapper
and
.I container
module, unit test is available. To run the tests, execute below command:

.nf
$ make test
.fi

Each test iterates writing to file and reading to the file for many times and it
takes long time to finish. Please take care of the execution time if running on
any CI environment.

.SH SEE ALSO
\fB
axfer(1),
axfer\-list(1),
alsamixer(1),
amixer(1)
\fP

.SH AUTHOR
Takashi Sakamoto <o\-takashi@sakamocchi.jp>