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Manual page and help for the alsabat linux command. ALSABAT (ALSA Basic Audio Tester) is a simple command-line utility designed to automate the testing of audio drives and sound servers with little human interaction. ALSABAT can be used to test sound quality, stress test characteristics, and sound testing before and after a change in PM status.
Man page output
man alsabatALSABAT(1) General Commands Manual ALSABAT(1)
NAME
alsabat - command-line sound tester for ALSA sound card driver
SYNOPSIS
alsabat [flags]
DESCRIPTION
ALSABAT(ALSA Basic Audio Tester) is a simple command-line utility intended to help auto‐
mate audio driver and sound server testing with little human interaction. ALSABAT can be
used to test audio quality, stress test features and test audio before and after PM state
changes.
ALSABAT's design is relatively simple. ALSABAT plays an audio stream and captures the same
stream in either a digital or analog loop back. It then compares the captured stream using
a FFT to the original to determine if the test case passes or fails.
ALSABAT can either run wholly on the target machine being tested (standalone mode) or can
run as a client/server mode where by alsabat client runs on the target and runs as a
server on a separate tester machine. The client/server mode still requires some manual in‐
teraction for synchronization, but this is actively being developed for future releases.
The hardware testing configuration may require the use of an analog cable connecting tar‐
get to tester machines or a cable to create an analog loopback if no loopback mode is not
available on the sound hardware that is being tested. An analog loopback cable can be
used to connect the "line in" to "line out" jacks to create a loopback. If only headphone
and mic jacks (or combo jack) are available then the following simple circuit can be used
to create an analog loopback :-
https://source.android.com/devices/audio/loopback.html
If tinyalsa is installed in system, user can choose tinyalsa as backend lib of alsabat,
with configure option "--enable-alsabat-backend-tiny".
OPTIONS
-h, --help
Help: show syntax.
-D Select sound card to be tested by name.
-P Select the playback PCM device.
-C Select the capture PCM device.
-f Sample format
Recognized sample formats are: U8 S16_LE S24_3LE S32_LE
Some of these may not be available on selected hardware
The available format shortcuts are:
-f cd (16 bit little endian, 44100, stereo) [-f S16_LE -c2 -r44100]
-f dat (16 bit little endian, 48000, stereo) [-f S16_LE -c2 -r48000]
If no format is given S16_LE is used.
-c The number of channels. The default is one channel. Valid values at the moment are
1 or 2.
-r Sampling rate in Hertz. The default rate is 44100 Hertz. Valid values depends on
hardware support.
-n Duration of generated signal. The value could be either of the two forms:
1. Decimal integer, means number of frames;
2. Floating point with suffix 's', means number of seconds.
The default is 2 seconds.
-k Sigma k value for analysis.
The analysis function reads data from WAV file, run FFT against the data to get
magnitude of frequency vectors, and then calculates the average value and standard
deviation of frequency vectors. After that, we define a threshold:
threshold = k * standard_deviation + mean_value
Frequencies with amplitude larger than threshold will be recognized as a peak, and
the frequency with largest peak value will be recognized as a detected frequency.
ALSABAT then compares the detected frequency to target frequency, to decide if the
detecting passes or fails.
The default value is 3.0.
-F Target frequency for signal generation and analysis, in Hertz. The default is
997.0 Hertz. Valid range is (DC_THRESHOLD, 40% * Sampling rate).
-p Total number of periods to play or capture.
--log=#
Write stderr and stdout output to this log file.
--file=#
Input WAV file for playback.
--saveplay=#
Target WAV file to save capture test content.
--local
Internal loopback mode. Playback, capture and analysis internal to ALSABAT only.
This is intended for developers to test new ALSABAT features as no audio is routed
outside of ALSABAT.
--standalone
Add support for standalone mode where ALSABAT will run on a different machine to
the one being tested. In standalone mode, the sound data can be generated, play‐
back and captured just like in normal mode, but will not be analyzed. The ALSABAT
being built without libfftw3 support is always in standalone mode. The ALSABAT in
normal mode can also bypass data analysis using option "--standalone".
--roundtriplatency
Round trip latency test. Audio latency is the time delay as an audio signal passes
through a system. There are many kinds of audio latency metrics. One useful metric
is the round trip latency, which is the sum of output latency and input latency.
--snr-db=#
Noise detection threshold in SNR (dB). 26dB indicates 5% noise in amplitude.
ALSABAT will return error if signal SNR is smaller than the threshold.
--snr-pc=#
Noise detection threshold in percentage of noise amplitude (%). ALSABAT will re‐
turn error if the noise amplitude is larger than the threshold.
EXAMPLES
alsabat -P plughw:0,0 -C plughw:0,0 -c 2 -f S32_LE -F 250
Generate and play a sine wave of 250 Hertz with 2 channel and S32_LE format, and
then capture and analyze.
alsabat -P plughw:0,0 -C plughw:0,0 --file 500Hz.wav
Play the RIFF WAV file "500Hz.wav" which contains 500 Hertz waveform LPCM data, and
then capture and analyze.
RETURN VALUE
On success, returns 0.
If no peak be detected, returns -1001;
If only DC be detected, returns -1002;
If peak frequency does not match with the target frequency, returns -1003.
SEE ALSO
aplay(1)
BUGS
Currently only support RIFF WAV format with PCM data. Please report any bugs to the alsa-
devel mailing list.
AUTHOR
alsabat is by Liam Girdwood <liam.r.girdwood@linux.intel.com>, Bernard Gautier
<bernard.gautier@intel.com> and Han Lu <han.lu@intel.com>. This document is by Liam Gird‐
wood <liam.r.girdwood@linux.intel.com> and Han Lu <han.lu@intel.com>.
20th October 2015 ALSABAT(1)
Help output
alsabat --helpalsa-utils version 1.1.8
Usage: alsabat [-options]...
-h, --help this help
-D pcm device for both playback and capture
-P pcm device for playback
-C pcm device for capture
-f sample format
-c number of channels
-r sampling rate
-n frames to playback or capture
-k parameter for frequency detecting threshold
-F target frequency
-p total number of periods to play/capture
-B buffer size in frames
-E period size in frames
--log=# file that both stdout and strerr redirecting to
--file=# file for playback
--saveplay=# file that storing playback content, for debug
--local internal loop, set to bypass pcm hardware devices
--standalone standalone mode, to bypass analysis
--roundtriplatency round trip latency mode
--snr-db=# noise detect threshold, in SNR(dB)
--snr-pc=# noise detect threshold, in noise percentage(%)
Recognized sample formats are: U8 S16_LE S24_3LE S32_LE
The available format shotcuts are:
-f cd (16 bit little endian, 44100, stereo)
-f dat (16 bit little endian, 48000, stereo)
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