alsaloop - Online in the Cloud

PROGRAM:

NAME

alsaloop - command-line PCM loopback

SYNOPSIS

alsaloop [-option] [cmd]

DESCRIPTION

alsaloop allows create a PCM loopback between a PCM capture device and a PCM playback
device.

alsaloop supports multiple soundcards, adaptive clock synchronization, adaptive rate
resampling using the samplerate library (if available in the system). Also, mixer controls
can be redirected from one card to another (for example Master and PCM).

OPTIONS

-h | --help

Prints the help information.

-g <file> | --config=<file>

Use given configuration file. The syntax of this file is simple: one line contains
the command line options for one job. The '#' means comment and rest of line is
ignored. Example:

# First line - comment, second line - first job
-C hw:1,0 -P hw:0,0 -t 50000 -T 1
# Third line - comment, fourth line - second job
-C hw:1,1 -P hw:0,1 -t 40000 -T 2

-d | --daemonize

Daemonize the main process and use syslog for messages.

-P <device> | --pdevice=<device>

Use given playback device.

-C <device> | --cdevice=<device>

Use given capture device.

-X <device> | --pctl=<device>

Use given CTL device for playback.

-Y <device> | --cctl=<device>

Use given CTL device for capture.

-l <latency> | --latency=<frames>

Requested latency in frames.

-t <usec> | --tlatency=<usec>

Requested latency in usec (1/1000000sec).

-f <format> | --format=<format>

Format specification (usually S16_LE S32_LE). Use -h to list all formats. Default
format is S16_LE.

-c <channels> | --channels=<channels>

Channel count specification. Default value is 2.

-c <rate> | --rate=<rate>

Rate specification. Default value is 48000 (Hz).

-n | --resample

Allow rate resampling using alsa-lib.

-A <converter> | --samplerate=<converter>

Use libsamplerate and choose a converter:

0 or sincbest - best quality
1 or sincmedium - medium quality
2 or sincfastest - lowest quality
3 or zerohold - hold zero samples
4 or linear - worst quality - linear resampling
5 or auto - choose best method

-B <size> | --buffer=<size>

Buffer size in frames.

-E <size> | --period=<size>

Period size in frames.

-s <secs> | --seconds=<secs>

Duration of loop in seconds.

-b | --nblock

Non-block mode (very early process wakeup). Eats more CPU.

-S <mode> | --sync=<mode>

Sync mode specification for capture to playback stream:
0 or none - do not touch the stream
1 or simple - add or remove samples to keep
both streams synchronized
2 or captshift - use driver for the capture device
(if supported) to compensate
the rate shift
3 or playshift - use driver for the playback device
(if supported) to compensate
the rate shift
4 or samplerate - use samplerate library to do rate resampling
5 or auto - automatically selects the best method
in this order: captshift, playshift,
samplerate, simple

-T <num> | --thread=<num>

Thread number (-1 means create a unique thread). All jobs with same thread numbers
are run within one thread.

-m <mixid> | --mixer=<midid>

Redirect mixer control from the playback card to the capture card. Format of mixid
is SRCID(PLAYBACK)[@DSTID(PLAYBACK)]:

"name='Master Playback Switch'@name='Another Switch'"
"name='PCM Playback Volume'"

Known attributes:

name - control ID name
index - control ID index
device - control ID device
subdevice - control ID subdevice
iface - control ID interface
numid - control ID numid

-O <ossmixid> | --ossmixer=<midid>

Redirect mixer control from the OSS Mixer emulation layer (capture card) to the
ALSA layer (capture card). Format of ossmixid is ALSAID[,INDEX]@OSSID:

"Master@VOLUME"
"PCM,1@ALTPCM"

Known OSS attributes:

VOLUME, BASS, TREBLE, SYNTH, PCM, SPEAKER, LINE, MIC, CD, IMIX, ALTPCM,
RECLEV, IGAIN, OGAIN, LINE1, LINE2, LINE3, DIGITAL1, DIGITAL2, DIGITAL3,
PHONEIN, PHONEOUT, VIDEO, RADIO, MONITOR

-v | --verbose

Verbose mode. Use multiple times to increase verbosity.

-U | --xrun

Verbose xrun profiling.

-W <timeout> | --wake=<timeout>

Set process wake timeout.

EXAMPLES

alsaloop -C hw:0,0 -P hw:1,0 -t 50000

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