xenial (1) alsabat.1.gz

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

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.

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)