alsabat

ALSABAT(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 automate 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 interaction
       for synchronization, but this is actively being developed for future
       releases.

       The hardware testing configuration may require the use of an analog
       cable connecting target 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, playback 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 return 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 Girdwood <liam.r.girdwood@linux.intel.com> and
       Han Lu <han.lu@intel.com>.



                               20th October 2015                    ALSABAT(1)