.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 untill 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,