1AXFER-TRANSFER(1) General Commands Manual AXFER-TRANSFER(1)
2
6 axfer-transfer - transferrer of audio data frame for sound devices and
7 nodes.
8
11 axfer transfer direction [ common-options ] [ backend-options ] [
12 filepath ]
13 axfer transfer direction [ common-options ] [ backend-options ] -I |
15 --separate-channels filepath ...
16 direction = capture | playback
18 common-options = ( read OPTIONS section )
20 backend-options = ( read OPTIONS section )
22 filepaths = ( read OPTIONS section )
24
27 The transfer subcommand of axfer performs transmission of audio data
28 frames for devices available in supported backends. This program is
29 essentially designed to use alsa-lib APIs (libasound backend) to handle
30 sound devices supported by Linux sound subsystem (ALSA).
31
34 Direction
35 capture
36 Operates for capture transmission.
37 playback
40 Operates for playback transmission.
41 Filepath
44 Filepath is handled as a path relative to current working directory of
45 run time if it's not full path from root directory.
46 The standard input or output is used if filepath is not specified or
48 given as '-' .
49 For playback transmission, container format of given I filepath is
51 detected automatically and metadata is used for parameters of sample
52 format, channels, rate, duration. If nothing detected, content of given
53 file path is handled as raw data. In this case, the parameters should
54 be indicated as options.
55 Multiple filepaths are allowed with -I | --separate-channels option. In
57 this case, standard input and output is not available. The same
58 filepath is not allowed except for paths listed below:
59 - /dev/null
60 - /dev/zero
61 - /dev/full
62 - /dev/random
63 - /dev/urandom
64 Common options
67 -h, --help
68 Print help messages and finish run time. Not yet implemented.
69 -q, --quiet
72 Quiet mode. Suppress messages (not sound :))
73 -v, --verbose
76 Verbose mode. Runtime dumps supplemental information according
77 to the number of this option given in command line.
78 -d, --duration=#
81 Interrupt after # seconds. A value of zero means infinity. The
82 default is zero, so if this option is omitted then the transmis‐
83 sion process will run until it is killed. Either -d or -s option
84 is available exclusively.
85 -s, --samples=#
88 Interrupt after transmission of # number of data frames. A value
89 of zero means infinity. The default is zero, so if this options
90 is omitted then the transmission process will run until it is
91 killed. Either -d or -s option is available exclusively.
92 -f, --format=FORMAT
95 Indicate format of audio sample. This is required for capture
96 transmission, or playback transmission with files including raw
97 audio data.
98 Available sample format is listed below:
100 - [S8|U8|S16|U16|S32|U32][_LE|_BE]
101 - [S24|U24][_LE|_BE]
102 - FLOAT[_LE|_BE]
103 - FLOAT64[_LE|_BE]
104 - IEC958_SUBFRAME[_LE|_BE]
105 - MU_LAW
106 - A_LAW
107 - [S20|U20][_LE|_BE]
108 - [S24|U24][_3LE|_3BE]
109 - [S20|U20][_3LE|_3BE]
110 - [S18|U18][_3LE|_3BE]
111 - DSD_U8
112 - DSD_[U16|U32][_LE|_BE]
113 If endian-ness is omitted, host endian-ness is used.
115 Some special formats are available:
117 - cd (16 bit little endian, 44100, stereo) [= -f S16_LE -c 2 -r
118 44100]
119 - cdr (16 bit big endian, 44100, stereo) [= -f S16_BE -c 2 -f
120 44100]
121 - dat (16 bit little endian, 48000, stereo) [= -f S16_LE -c 2
122 -r 48000]
123 If omitted, U8 is used as a default. Actual available formats
125 are restricted by each transmission backend.
126 Unavailable sample format is listed below. These format has size
128 of data frame unaligned to byte unit.
129 - IMA_ADPCM
131 - MPEG
132 - GSM
133 - SPECIAL
134 - G723_24
135 - G723_24_1B
136 - G723_40
137 - G723_40_1B
138 -c, --channels=#
141 Indicate the number of audio data samples per frame. This is
142 required for capture transmission, or playback transmission with
143 files including raw audio data. The value should be between 1 to
144 256 . If omitted, 1 is used as a default.
145 -r, --rate=#
148 Indicate the number of audio data frame per second. This is
149 required for capture transmission, or playback transmission with
150 files including raw audio data. If the value is less than 1000 ,
151 it's interpreted by kHz unit. The value should be between 2000
152 and 192000 . If omitted, 8000 is used as a default.
153 -t, --file-type=TYPES
156 Indicate the type of file. This is required for capture trans‐
157 mission. Available types are listed below:
158 - wav: Microsoft/IBM RIFF/Wave format
159 - au, sparc: Sparc AU format
160 - voc: Creative Tech. voice format
161 - raw: raw data
162 When nothing is indicated, for capture transmission, the type is
164 decided according to suffix of filepath , and raw type is used
165 for fallback.
166 -I, --separate-channels
169 Indicate this option when several files are going to be handled.
170 For capture transmission, if one filepath is given as filepath ,
171 a list of filepaths is generated in a formula
172 '<filepath>-<sequential number>[.suffix]'. The suffix is omit‐
173 ted when raw format of container is used.
174 --dump-hw-params
177 Dump hardware parameters and finish run time if backend supports
178 it.
179 --xfer-backend=BACKEND
182 Select backend of transmission from a list below. The default is
183 libasound.
184 - libasound
185 - libffado (optional if compiled)
186 Backend options for libasound
189 -D, --device
190 This option is used to select PCM node in libasound configura‐
192 tion space. Available nodes are listed by pcm operation of list
193 subcommand.
194 -N, --nonblock
197 With this option, PCM substream is opened in non-blocking mode.
199 When audio data frame is not available in buffer of the PCM sub‐
200 stream, I/O operation immediately returns without blocking
201 process. This option implicitly uses --waiter-type option as
202 well to prevent heavy consumption of CPU time.
203 -M, --mmap
206 With this option, audio data frame is processed directly in buf‐
208 fer of PCM substream if selected node supports this operation.
209 Without the option, temporary buffers are used to copy audio
210 data frame for buffer of PCM substream. This option implicitly
211 uses --waiter-type option as well to prevent heavy consumption
212 of CPU time.
213 -F, --period-size
216 This option configures given value to period_size hardware
218 parameter of PCM substream. The parameter indicates the number
219 of audio data frame per period in buffer of the PCM substream.
220 Actual number is decided as a result of interaction between each
221 implementation of PCM plugin chained from the selected PCM node,
222 and in-kernel driver or PCM I/O plugins.
223 Ideally, the same amount of audio data frame as the value should
225 be handled in one I/O operation. Actually, it is not, depending
226 on implementation of the PCM plugins, in-kernel driver, PCM I/O
227 plugins and scheduling model. For 'hw' PCM plugin in 'irq'
228 scheduling model, the value is used to decide intervals of hard‐
229 ware interrupt, thus the same amount of audio data frame as the
230 value is expected to be available for one I/O operation.
231 --period-time
234 This option configures given value to period_time hardware
236 parameter of PCM substream. This option is similar to
237 --period-size option, however its unit is micro-second.
238 -B, --buffer-size
241 This option configures given value to buffer_size hardware
243 parameter of PCM substream. The parameter indicates the number
244 of audio data frame in buffer of PCM substream. Actual number is
245 decided as a result of interaction between each implementation
246 of PCM plugin chained from the selected PCM node, and in-kernel
247 driver or PCM I/O plugins.
248 Ideally, this is multiples of the number of audio data frame per
250 period, thus the size of period. Actually, it is not, depending
251 on implementation of the PCM plugins, in-kernel driver and PCM
252 I/O plugins.
253 --buffer-time
256 This option configures given value to buffer_time hardware
258 parameter of PCM substream. This option is similar to --buf‐
259 fer-size option, however its unit is micro-second.
260 --waiter-type
263 This option indicates the type of waiter for event notification.
265 At present, four types are available; default , select , poll
266 and epoll . With default type, 'snd_pcm_wait()' is used. With
267 select type, 'select(2)' system call is used. With poll type,
268 'poll(2)' system call is used. With epoll type, Linux-specific
269 'epoll(7)' system call is used.
270 This option should correspond to one of --nonblock or --mmap
272 options, or timer value of --sched-model option. Neither this
273 option nor --test-nowait is available at the same time.
274 --sched-model
277 This option selects scheduling model for process of this pro‐
279 gram. One of irq or timer is available. In detail, please read
280 'SCHEDULING MODEL' section.
281 When nothing specified, irq model is used.
283 -A, --avail-min
286 This option configures given value to avail-min software parame‐
288 ter of PCM substream. In blocking mode, the value is used as
289 threshold of the number of available audio data frames in buffer
290 of PCM substream to wake up process blocked by I/O operation. In
291 non-blocking mode, any I/O operation returns -EAGAIN untill the
292 available number of audio data frame reaches the threshold.
293 This option has an effect in cases neither --mmap nor timer
295 value of --sched-model option is used.
296 -R, --start-delay
299 This option configures given value to start_threshold software
301 parameter of PCM substream. The value is used as threshold to
302 start PCM substream automatically. At present, this option has
303 an effect in cases neither --mmap nor timer value of
304 --sched-model option is used.
305 For playback transmission, when the number of accumulated audio
307 data frame in buffer of PCM substream to which this program
308 writes out reaches the threshold, the PCM substream starts auto‐
309 matically without an explicit call of snd_pcm_start() to the PCM
310 substream.
311 For capture transmission, this option is useless. The number of
313 accumulated audio data frame is not increased without an
314 explicit call of snd_pcm_start() to the PCM substream.
315 This option has an effect in cases neither --mmap nor timer
317 value of --sched-model option is used.
318 -T, --stop-delay
321 This option configures given value to stop_threshold software
323 parameter of PCM substream. The value is used as threshold to
324 stop PCM substream automatically. At present, this option has an
325 effect in cases neither --mmap nor timer value of --sched-model
326 option is used.
327 For capture transmission, when the number of accumulated audio
329 data frame in buffer of PCM substream to which a driver or
330 alsa-lib PCM plugins write reaches the threshold, the PCM sub‐
331 stream stops automatically without an explicit call of
332 snd_pcm_stop() to the PCM substream. This is a case that this
333 program leaves the audio data frames without reading for a
334 while.
335 For playback transmission, when the number available audio data
337 frame in buffer of PCM substream from which a driver or alsa-lib
338 PCM plugins read reaches the threshold, the PCM substream stops
339 automatically without an explicit call of snd_pcm_stop() to the
340 PCM substream. This is a case that this program leaves the audio
341 data frames without writing for a while.
342 This option has an effect in cases neither --mmap nor timer
344 value of --sched-model option is used.
345 --disable-resample
348 This option has an effect for 'plug' plugin in alsa-lib to sup‐
350 press conversion of sampling rate for audio data frame.
351 --disable-channels
354 This option has an effect for 'plug' plugin in alsa-lib to sup‐
356 press conversion of channels for audio data frame.
357 --disable-format
360 This option has an effect for 'plug' plugin in alsa-lib to sup‐
362 press conversion of sample format for audio data frame.
363 --disable-softvol
366 This option has an effect for 'softvol' plugin in alsa-lib to
368 suppress conversion of samples for audio data frame via addi‐
369 tional control element.
370 --fatal-errors
373 This option suppresses recovery operation from XRUN state of
375 running PCM substream, then process of this program is going to
376 finish as usual.
377 --test-nowait
380 This option disables any waiter for I/O event notification. I/O
382 operations are iterated till any of audio data frame is avail‐
383 able. The option brings heavy load in consumption of CPU time.
384 Backend options for libffado
387 This backend is automatically available when configure script detects
388 ffado_streaming_init() symbol in libffado shared object.
389 -p, --port
392 This option uses given value to decide which 1394 OHCI con‐
394 troller is used to communicate. When Linux system has two 1394
395 OHCI controllers, 0 or 1 are available. Neither this option nor
396 -g is available at the same time. If nothing specified, libffado
397 performs to communicate to units on IEEE 1394 bus managed by all
398 of 1394 OHCI controller available in Linux system.
399 -n, --node
402 This option uses given value to decide which unit is used to
404 communicate. This option requires -p option to indicate which
405 1394 OHCI controller is used to communicate to the specified
406 unit.
407 -g, --guid
410 This option uses given value to decide a target unit to communi‐
412 cate. The value should be prefixed with '0x' and consists of
413 hexadecimal literal letters (0-9, a-f, A-F). Neither this option
414 nor -p is available at the same time. If nothing specified,
415 libffado performs to communicate to units on IEEE 1394 bus man‐
416 aged by all of 1394 OHCI controller available in Linux system.
417 --frames-per-period
420 This option uses given value to decide the number of audio data
422 frame in one read/write operation. The operation is blocked till
423 the number of available audio data frame exceeds the given
424 value. As a default, 512 audio data frames is used.
425 --periods-per-buffer
428 This option uses given value to decide the size of intermediate
430 buffer between this program and libffado. As a default, 2 peri‐
431 ods per buffer is used.
432 --slave
435 This option allows this program to run slave mode. In this mode,
437 libffado adds unit directory into configuration ROM of 1394 OHCI
438 controller where Linux system runs. The unit directory can be
439 found by the other node on the same bus. Linux system running
440 on the node can transfer isochronous packet with audio data
441 frame to the unit. This program can receive the packet and
442 demultiplex the audio data frame.
443 --snoop
446 This option allows this program to run snoop mode. In this mode,
448 libffado listens all isochronous channels. When isochronous com‐
449 munication starts by any unit on the same bus, the packets can
450 be handled by this program.
451 --sched-priority
454 This option executes pthread_setschedparam() in a call of
456 ffado_streaming_init() to configure scheduling policy and given
457 value as its priority for threads related to isochronous commu‐
458 nication. The given value should be within RLIMIT_RTPRIO param‐
459 eter of process. Please read getrlimit(2) for details.
460
463 During transmission, SIGINT and SIGTERM will close handled files and
464 PCM substream to be going to finish run time.
465 SIGTSTP will suspend PCM substream and SIGCONT will resume it. No XRUNs
467 are expected. With libffado backend, the suspend/resume is not sup‐
468 ported and runtime is aboeted immediately.
469 The other signals perform default behaviours.
471
474 $ axfer transfer playback -d 1 something
475 The above will transfer audio data frame in 'something' file for play‐
477 back during 1 second. The sample format is detected automatically as a
478 result to parse 'something' as long as it's compliant to one of Micro‐
479 soft/IBM RIFF/Wave, Sparc AU, Creative Tech. voice formats. If nothing
480 detected, -r , -c and -f should be given, or -f should be given with
481 special format.
482 $ axfer transfer playback -r 22050 -c 1 -f S16_LE -t raw something
485 The above will transfer audio data frame in 'something' file including
487 no information of sample format, as sample format of 22050 Hz, monau‐
488 ral, signed 16 bit little endian PCM for playback. The transmission
489 continues till catching SIGINT from keyboard or SIGTERM by kill(1) .
490 $ axfer transfer capture -d 10 -f cd something.wav
493 The above will transfer audio data frame to 'something.wav' file as
495 sample format of 44.1 kHz, 2 channels, signed 16 bit little endian PCM,
496 during 10 seconds. The file format is Microsoft/IBM RIFF/Wave according
497 to suffix of the given filepath .
498 $ axfer transfer capture -s 1024 -r 48000 -c 2 -f S32_BE -I -t au channels
501 The above will transfer audio data frame as sample format of 48.0 kHz,
503 2 channels, signed 32 bit big endian PCM for 1,024 number of data
504 frames to files named 'channels-1.au' and 'channels-2.au'.
505
508 In a design of ALSA PCM core, runtime of PCM substream supports two
509 modes; period-wakeup and no-period-wakeup. These two modes are for
510 different scheduling models.
511 IRQ-based scheduling model
514 As a default, period-wakeup mode is used. In this mode, in-kernel driv‐
515 ers should operate hardware to generate periodical notification for
516 transmission of audio data frame. The interval of notification is
517 equivalent to the same amount of audio data frame as one period of buf‐
518 fer, against actual time.
519 In a handler assigned to the notification, a helper function of ALSA
521 PCM core is called to update a position to head of hardware transmis‐
522 sion, then compare it with a position to head of application operation
523 to judge overrun/underrun (XRUN) and to wake up blocked processes.
524 For this purpose, hardware IRQ of controller for serial audio bus such
526 as Inter-IC sound is typically used. In this case, the controller gen‐
527 erates the IRQ according to transmission on the serial audio bus. In
528 the handler assigned to the IRQ, direct media access (DMA) transmission
529 is requested between dedicated host memory and device memory.
530 If target hardware doesn't support this kind of mechanism, the periodi‐
532 cal notification should be emulated by any timer; e.g. hrtimer, kernel
533 timer. External PCM plugins generated by PCM plugin SDK in alsa-lib
534 should also emulate the above behaviour.
535 In this mode, PCM applications are programmed according to typical way
537 of I/O operations. They execute blocking system calls to read/write
538 audio data frame in buffer of PCM substream, or blocking system calls
539 to wait until any audio data frame is available. In axfer , this is
540 called IRQ-based scheduling model and a default behaviour. Users can
541 explicitly configure this mode by usage of --sched-model option with
542 irq value.
543 Timer-based scheduling model
546 The no-period-wakeup mode is an optional mode of runtime of PCM sub‐
547 stream. The mode assumes a specific feature of hardware and assist of
548 in-kernel driver and PCM applications. In this mode, in-kernel drivers
549 don't operate hardware to generate periodical notification for trans‐
550 mission of audio data frame. The hardware should automatically con‐
551 tinue transmission of audio data frame without periodical operation of
552 the drivers; e.g. according to auto-triggered DMA transmission, a chain
553 of registered descriptors.
554 In this mode, nothing wakes up blocked processes, therefore PCM appli‐
556 cations should be programmed without any blocking operation. For this
557 reason, this mode is enabled when the PCM applications explicitly con‐
558 figure hardware parameter to runtime of PCM substream, to prevent dis‐
559 order of existing applications. Additionally, nothing maintains timing
560 for transmission of audio data frame, therefore the PCM applications
561 should voluntarily handle any timer to queue audio data frame in buffer
562 of the PCM substream for lapse of time. Furthermore, instead of driver,
563 the PCM application should call a helper function of ALSA PCM core to
564 update a position to head of hardware transmission and to check XRUN.
565 In axfer , this is called timer-based scheduling model and available as
567 long as hardware/driver assists no-period-wakeup runtime. Users should
568 explicitly set this mode by usage of --sched-model option with timer
569 value.
570 In the scheduling model, PCM applications need to care of available
572 space on PCM buffer by lapse of time, typically by yielding CPU and
573 wait for rescheduling. For the yielding, timeout is calculated for
574 preferable amount of PCM frames to process. This is convenient to a
575 kind of applications, like sound servers. when an I/O thread of the
576 server wait for the timeout, the other threads can process audio data
577 frames for server clients. Furthermore, with usage of rewinding/for‐
578 warding, applications can achieve low latency between transmission
579 position and handling position even if they uses large size of PCM buf‐
580 fers.
581 Advantages and issues
584 Ideally, timer-based scheduling model has some advantages than
585 IRQ-based scheduling model. At first, no interrupt context runs for PCM
586 substream. The PCM substream is handled in any process context only. No
587 need to care of race conditions between IRQ and process contexts. This
588 reduces some concerns for some developers of drivers and applications.
589 Secondary, CPU time is not used for handlers on the interrupt context.
590 The CPU time can be dedicated for the other tasks. This is good in a
591 point of Time Sharing System. Thirdly, hardware is not configured to
592 generate interrupts. This is good in a point of reduction of overall
593 power consumption possibly.
594 In either scheduling model, the hardware should allow drivers to read
596 the number of audio data frame transferred between the dedicated memory
597 and the device memory for audio serial bus. However, in timer-based
598 scheduling model, fine granularity and accuracy of the value is impor‐
599 tant. Actually hardware performs transmission between dedicated memory
600 and device memory for a small batch of audio data frames or bytes. In a
601 view of PCM applications, the granularity in current transmission is
602 required to decide correct timeout for each I/O operation. As of Linux
603 kernel v4.21, ALSA PCM interface between kernel/userspace has no fea‐
604 ture to report it.
605
608 The transfer subcommand of axfer is designed to keep compatibility to
609 aplay(1). However some options below are not compatible due to several
610 technical reasons.
611 -I, --separate-channels
614 This option is supported just for files to store audio data
615 frames corresponding to each channel. In aplay(1) implementa‐
616 tion, this option has an additional effect to use PCM buffer
617 aligned to non-interleaved order if a target device supports. As
618 of 2018, PCM buffer of non-interleaved order is hardly used by
619 sound devices.
620 -A, --avail-min
623 This option indicates threshold to wake up blocked process in a
624 unit of audio data frame. Against aplay(1) implementation, this
625 option has no effect with --mmap option as well as timer of
626 --sched-model option.
627 -R, --start-delay
630 This option indicates threshold to start prepared PCM substream
631 in a unit of audio data frame. Against aplay(1) implementation,
632 this option has no effect with --mmap option as well as timer of
633 --sched-model option.
634 -T, --stop-delay
637 This option indicates threshold to stop running PCM substream in
638 a unit of audio data frame. Against aplay(1) implementation,
639 this option has no effect with --mmap option as well as timer of
640 --sched-model option.
641 --max-file-time=#
644 This option is unsupported. In aplay(1) implementation, the
645 option has an effect for capture transmission to save files up
646 to the same number of data frames as the given value by second
647 unit, or the maximum number of data frames supported by used
648 file format. When reaching to the limitation, used file is
649 closed, then new file is opened and audio data frames are writ‐
650 ten. However, this option requires extra handling of files and
651 shall increase complexity of main loop of axfer.
652 --use-strftime=FORMAT
655 This option is unsupported. In aplay(1) implementation, the
656 option has an effect for capture transmission to generate file
657 paths according to given format in which some extra formats are
658 available as well as formats supported by strftime(3). However,
659 this option requires extra string processing for file paths and
660 it's bothersome if written in C language.
661 --process-id-file=FILEPATH
664 This option is unsupported. In aplay(1) implementation, the
665 option has an effect to create a file for given value and write
666 out process ID to it. This file allows users to get process ID
667 and send any POSIX signal to aplay process. However, this idea
668 has some troubles for file locking when multiple aplay processes
669 run with the same file.
670 -V, --vumeter=TYPE
673 This option is not supported at present. In aplay(1) implementa‐
674 tion, this option has an effect to occupy stdout with some ter‐
675 minal control characters and display vumeter for monaural and
676 stereo channels. However, some problems lay; this feature is
677 just for audio data frames with PCM format, this feature brings
678 disorder of terminal after aborting, stdout is not available for
679 pipeline.
680 -i, --interactive
683 This option is not supported at present. In aplay(1) implementa‐
684 tion, this option has an effect to occupy stdin for key input
685 and suspend/resume PCM substream according to pushed enter key.
686 However, this feature requires an additional input handling in
687 main loop and leave bothersome operation to maintain PCM sub‐
688 stream.
689 SIGTSTP, SIGCONT
692 This performs suspend/resume of PCM substream. In aplay(1)
693 implementation, these operations bring XRUN state to the sub‐
694 stream, and suspend/resume is done in interactive mode in the
695 above. Some developers use the signal for recovery test from
696 XRUN. At present, no alternative is supported for the test.
697 SIGUSR1
700 This is not supported. In aplay(1) implementation, this signal
701 is assigned to a handler to close a current file to store audio
702 data frame and open a new file to continue processing. However,
703 as well as --max-file-time option, this option should increase
704 complexity of main loop of axfer.
705
708 Modular structure
709 This program consists of three modules; xfer , mapper and container .
710 Each module has an abstraction layer to enable actual implementation.
711 -------- ---------- -------------
713 device <-> | xfer | <-> | mapper | <-> | container | <-> file
714 -------- ---------- -------------
715 libasound single wav
716 libffado multiple au
717 voc
718 raw
719 The xfer module performs actual transmission to devices and nodes. The
721 module can have several transmission backends. As a default backend,
722 libasound backend is used to perform transmission via alsa-lib APIs.
723 The module allows each backend to parse own command line options.
724 The container module performs to read/write audio data frame via
726 descriptor for file/stream of multimedia container or raw data. The
727 module automatically detect type of multimedia container and parse
728 parameters in its metadata of data header. At present, three types of
729 multimedia containers are supported; Microsoft/IBM RIFF/Wave ( wav ),
730 Sparc AU ( au ) and Creative Technology voice ( voc ). Additionally, a
731 special container is prepared for raw audio data ( raw ).
732 The mapper module handles buffer layout and alignment for transmission
734 of audio data frame. The module has two implementations; single and
735 multiple . The single backend uses one container to construct the buf‐
736 fer. The multiple backend uses several containers to construct it.
737 Care of copying audio data frame
740 Between the xfer module and mapper module, a pointer to buffer includ‐
741 ing audio data frames is passed. This buffer has two shapes for inter‐
742 leaved and non-interleaved order. For the former, the pointer points to
743 one buffer. For the latter, the pointer points to an array in which
744 each element points to one buffer. Between the mapper module and con‐
745 tainer module, a pointer to one buffer is passed because supported
746 media containers including raw type store audio data frames in inter‐
747 leaved order.
748 In passing audio data frame between the modules, axfer is programmed to
750 avoid copying between a buffer to another buffer as much as possible.
751 For example, in some scenarios below, no copying occurs between mod‐
752 ules.
753 - xfer(mmap/interleaved), mapper(single), container(any)
755 - xfer(mmap/non-interleaved), mapper(multiple), containers(any)
756
759 axfer(1), axfer-list(1), alsamixer(1), amixer(1)
760
763 Takashi Sakamoto <o-takashi@sakamocchi.jp>
764alsa-utils 28 November 2018 AXFER-TRANSFER(1)