AXFER-TRANSFER(1) General Commands Manual AXFER-TRANSFER(1)
NAME
axfer-transfer - transferrer of audio data frame for sound devices and
nodes.
SYNOPSIS
axfer transfer direction [ common-options ] [ backend-options ] [
filepath ]
axfer transfer direction [ common-options ] [ backend-options ] -I |
--separate-channels filepath ...
direction = capture | playback
common-options = ( read OPTIONS section )
backend-options = ( read OPTIONS section )
filepaths = ( read OPTIONS section )
DESCRIPTION
The transfer subcommand of axfer performs transmission of audio data
frames for devices available in supported backends. This program is es-
sentially designed to use alsa-lib APIs (libasound backend) to handle
sound devices supported by Linux sound subsystem (ALSA).
OPTIONS
Direction
capture
Operates for capture transmission.
playback
Operates for playback transmission.
Filepath
Filepath is handled as a path relative to current working directory of
run time if it's not full path from root directory.
The standard input or output is used if filepath is not specified or
given as '-' .
For playback transmission, container format of given filepath is de-
tected automatically and metadata is used for parameters of sample for-
mat, 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 filepaths are allowed with -I | --separate-channels option. In
this case, standard input and output is not available. The same filepath
is not allowed except for paths listed below:
- /dev/null
- /dev/zero
- /dev/full
- /dev/random
- /dev/urandom
Common options
-h, --help
Print help messages and finish run time.
-q, --quiet
Quiet mode. Suppress messages (not sound :))
-v, --verbose
Verbose mode. Runtime dumps supplemental information according to
the number of this option given in command line.
-d, --duration=#
Interrupt after # seconds. A value of zero means infinity. The
default is zero, so if this option is omitted then the transmis-
sion process will run until it is killed. Either -d or -s option
is available exclusively.
-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 -d or -s option is available exclusively.
-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, 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
-c, --channels=#
Indicate the number of audio data samples per frame. This is re-
quired for capture transmission, or playback transmission with
files including raw audio data. The value should be between 1 to
256 . If omitted, 1 is used as a default.
-r, --rate=#
Indicate the number of audio data frame per second. This is re-
quired for capture transmission, or playback transmission with
files including raw audio data. If the value is less than 1000 ,
it's interpreted by kHz unit. The value should be between 2000
and 192000 . If omitted, 8000 is used as a default.
-t, --file-type=TYPE
Indicate the type of file. This is required for capture transmis-
sion. 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 filepath , and raw type is used
for fallback.
-I, --separate-channels
Indicate this option when several files are going to be handled.
For capture transmission, if one filepath is given as filepath ,
a list of filepaths is generated in a formula '<filepath>-<se-
quential number>[.suffix]'. The suffix is omitted when raw for-
mat of container is used.
--dump-hw-params
Dump hardware parameters and finish run time if backend supports
it.
--xfer-backend=BACKEND
Select backend of transmission from a list below. The default is
libasound.
- libasound
- libffado (optional if compiled)
Backend options for libasound
-D, --device=NODE
This option is used to select PCM node in libasound configuration
space. Available nodes are listed by pcm operation of list sub-
command.
-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 sub-
stream, I/O operation immediately returns without blocking
process. This option implicitly uses --waiter-type option as well
to prevent heavy consumption of CPU time.
-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
--waiter-type option as well to prevent heavy consumption of CPU
time.
-F, --period-size=#
This option configures given value to period_size hardware para-
meter of PCM substream. The parameter indicates the number of au-
dio data frame per period in buffer of the PCM substream. Actual
number is decided as a result of interaction between each imple-
mentation 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 'hw' PCM plugin in 'irq' sched-
uling 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.
--period-time=#
This option configures given value to period_time hardware para-
meter of PCM substream. This option is similar to --period-size
option, however its unit is micro-second.
-B, --buffer-size=#
This option configures given value to buffer_size hardware para-
meter of PCM substream. The parameter indicates the number of au-
dio data frame in buffer of PCM substream. Actual number is de-
cided as a result of interaction between each implementation of
PCM plugin chained from the selected PCM node, and in-kernel dri-
ver 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.
--buffer-time=#
This option configures given value to buffer_time hardware para-
meter of PCM substream. This option is similar to --buffer-size
option, however its unit is micro-second.
--waiter-type=TYPE
This option indicates the type of waiter for event notification.
At present, four types are available; default , select , poll and
epoll . With default type, 'snd_pcm_wait()' is used. With select
type, 'select(2)' system call is used. With poll type, 'poll(2)'
system call is used. With epoll type, Linux-specific 'epoll(7)'
system call is used.
This option should correspond to one of --nonblock or --mmap op-
tions, or timer value of --sched-model option. Neither this op-
tion nor --test-nowait is available at the same time.
--sched-model=MODEL
This option selects scheduling model for process of this program.
One of irq or timer is available. In detail, please read 'SCHED-
ULING MODEL' section.
When nothing specified, irq model is used.
-A, --avail-min=#
This option configures given value to avail-min software parame-
ter 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 --mmap nor timer value
of --sched-model option is used.
-R, --start-delay=#
This option configures given value to 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 --mmap nor timer value of --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 auto-
matically without an explicit call of 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 snd_pcm_start() to the PCM substream.
This option has an effect in cases neither --mmap nor timer value
of --sched-model option is used.
-T, --stop-delay=#
This option configures given value to stop_threshold software pa-
rameter of PCM substream. The value is used as threshold to stop
PCM substream automatically. At present, this option has an ef-
fect in cases neither --mmap nor timer value of --sched-model op-
tion 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 sub-
stream stops automatically without an explicit call of
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 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 --mmap nor timer value
of --sched-model option is used.
--disable-resample
This option has an effect for 'plug' plugin in alsa-lib to sup-
press conversion of sampling rate for audio data frame.
--disable-channels
This option has an effect for 'plug' plugin in alsa-lib to sup-
press conversion of channels for audio data frame.
--disable-format
This option has an effect for 'plug' plugin in alsa-lib to sup-
press conversion of sample format for audio data frame.
--disable-softvol
This option has an effect for 'softvol' plugin in alsa-lib to
suppress conversion of samples for audio data frame via addi-
tional control element.
--fatal-errors
This option suppresses recovery operation from XRUN state of run-
ning PCM substream, then process of this program is going to fin-
ish as usual.
--test-nowait
This option disables any waiter for I/O event notification. I/O
operations are iterated till any of audio data frame is avail-
able. The option brings heavy load in consumption of CPU time.
Backend options for libffado
This backend is automatically available when configure script detects
ffado_streaming_init() symbol in libffado shared object.
-p, --port=#
This option uses given value to decide which 1394 OHCI controller
is used to communicate. When Linux system has two 1394 OHCI con-
trollers, 0 or 1 are available. Neither this option nor -g is
available at the same time. If nothing specified, libffado per-
forms to communicate to units on IEEE 1394 bus managed by all of
1394 OHCI controller available in Linux system.
-n, --node=#
This option uses given value to decide which unit is used to com-
municate. This option requires -p option to indicate which 1394
OHCI controller is used to communicate to the specified unit.
-g, --guid=HEXADECIMAL
This option uses given value to decide a target unit to communi-
cate. The value should be prefixed with '0x' and consists of
hexadecimal literal letters (0-9, a-f, A-F). Neither this option
nor -p is available at the same time. If nothing specified, libf-
fado performs to communicate to units on IEEE 1394 bus managed by
all of 1394 OHCI controller available in Linux system.
--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.
--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.
--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.
--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.
--sched-priority=#
This option executes pthread_setschedparam() in a call of
ffado_streaming_init() to configure scheduling policy and given
value as its priority for threads related to isochronous communi-
cation. The given value should be within RLIMIT_RTPRIO parameter
of process. Please read getrlimit(2) for details.
POSIX SIGNALS
During transmission, SIGINT and SIGTERM will close handled files and PCM
substream to be going to finish run time.
SIGTSTP will suspend PCM substream and SIGCONT will resume it. No XRUNs
are expected. With libffado backend, the suspend/resume is not supported
and runtime is aborted immediately.
The other signals perform default behaviours.
EXAMPLES
$ axfer transfer playback -d 1 something
The above will transfer audio data frame in 'something' file for play-
back during 1 second. The sample format is detected automatically as a
result to parse 'something' as long as it's compliant to one of Mi-
crosoft/IBM RIFF/Wave, Sparc AU, Creative Tech. voice formats. If noth-
ing detected, -r , -c and -f should be given, or -f should be given with
special format.
$ axfer transfer playback -r 22050 -c 1 -f S16_LE -t raw something
The above will transfer audio data frame in 'something' 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 SIGINT from keyboard or SIGTERM by kill(1) .
$ axfer transfer capture -d 10 -f cd something.wav
The above will transfer audio data frame to 'something.wav' file as sam-
ple 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 filepath .
$ axfer transfer capture -s 1024 -r 48000 -c 2 -f S32_BE -I -t au channels
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 'channels-1.au' and 'channels-2.au'.
SCHEDULING MODEL
In a design of ALSA PCM core, runtime of PCM substream supports two
modes; period-wakeup and no-period-wakeup. These two modes are for dif-
ferent scheduling models.
IRQ-based scheduling model
As a default, period-wakeup mode is used. In this mode, in-kernel dri-
vers should operate hardware to generate periodical notification for
transmission of audio data frame. The interval of notification is equiv-
alent 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 gener-
ates 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 periodi-
cal 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 au-
dio data frame in buffer of PCM substream, or blocking system calls to
wait until any audio data frame is available. In axfer , this is called
IRQ-based scheduling model and a default behaviour. Users can explicitly
configure this mode by usage of --sched-model option with irq value.
Timer-based scheduling model
The no-period-wakeup mode is an optional mode of runtime of PCM sub-
stream. 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 transmis-
sion 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 applica-
tions should be programmed without any blocking operation. For this rea-
son, 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 axfer , this is called timer-based scheduling model and available as
long as hardware/driver assists no-period-wakeup runtime. Users should
explicitly set this mode by usage of --sched-model option with 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 appli-
cations, 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 posi-
tion even if they uses large size of PCM buffers.
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. Sec-
ondary, 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 con-
sumption 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 impor-
tant. 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 re-
quired 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.
COMPATIBILITY TO APLAY
The transfer subcommand of axfer is designed to keep compatibility to
aplay(1). However some options below are not compatible due to several
technical reasons.
-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 de-
vices.
-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 --mmap option as well as timer of
--sched-model option.
-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 --mmap option as well as timer of
--sched-model option.
-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 --mmap option as well as timer of
--sched-model option.
--max-file-time=#
This option is unsupported. In aplay(1) implementation, the op-
tion 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 for-
mat. 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.
--use-strftime=FORMAT
This option is unsupported. In aplay(1) implementation, the op-
tion 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's bothersome if written in C language.
--process-id-file=FILEPATH
This option is unsupported. In aplay(1) implementation, the op-
tion 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.
-V, --vumeter=TYPE
This option is not supported at present. In aplay(1) implementa-
tion, this option has an effect to occupy stdout with some termi-
nal 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 disor-
der of terminal after aborting, stdout is not available for
pipeline.
-i, --interactive
This option is not supported at present. In aplay(1) implementa-
tion, this option has an effect to occupy stdin for key input and
suspend/resume PCM substream according to pushed enter key. How-
ever, this feature requires an additional input handling in main
loop and leave bothersome operation to maintain PCM substream.
-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 opera-
tion. Furthermore, as of alsa-lib v1.1.8, some plugins don't sup-
port this feature expectedly, thus this option is a lack of
transparent operation. At present, this option is not supported
yet not to confuse users.
SIGTSTP, SIGCONT
This performs suspend/resume of PCM substream. In aplay(1) imple-
mentation, 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.
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 --max-file-time option, this option should increase com-
plexity of main loop of axfer.
DESIGN
Modular structure
This program consists of three modules; xfer , mapper and container .
Each module has an abstraction layer to enable actual implementation.
-------- ---------- -------------
device <-> | xfer | <-> | mapper | <-> | container | <-> file
-------- ---------- -------------
libasound single wav
libffado multiple au
voc
raw
The xfer module performs actual transmission to devices and nodes. The
module can have several transmission backends. As a default backend,
libasound backend is used to perform transmission via alsa-lib APIs. The
module allows each backend to parse own command line options.
The container module performs to read/write audio data frame via de-
scriptor 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 ( wav ), Sparc AU ( au
) and Creative Technology voice ( voc ). Additionally, a special con-
tainer is prepared for raw audio data ( raw ).
The mapper module handles buffer layout and alignment for transmission
of audio data frame. The module has two implementations; single and
multiple . The single backend uses one container to construct the
buffer. The multiple backend uses several containers to construct it.
Care of copying audio data frame
Between the xfer module and 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 el-
ement points to one buffer. Between the mapper module and container mod-
ule, a pointer to one buffer is passed because supported media contain-
ers 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)
Unit test
For each of the mapper and container module, unit test is available. To
run the tests, execute below command:
$ make test
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 execu-
tion time if running on any CI environment.
SEE ALSO
axfer(1), axfer-list(1), alsamixer(1), amixer(1)
AUTHOR
Takashi Sakamoto <o-takashi@sakamocchi.jp>
alsa-utils 28 November 2018 AXFER-TRANSFER(1)
Generated by dwww version 1.16 on Sat Oct 3 07:42:54 CEST 2026.