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 in-
       tended to help automate audio driver and sound server testing with lit-
       tle  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 de-
       termine 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 ma-
       chine. 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 ca-
       ble 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  head-
       phone  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 val-
              ues 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 calcu-
              lates the average value and standard deviation of frequency vec-
              tors. After that, we define a threshold:
              threshold = k * standard_deviation + mean_value
              Frequencies  with amplitude larger than threshold will be recog-
              nized 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 fre-
              quency, 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 dif-
              ferent 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 op-
              tion "--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  la-
              tency, 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)