Provided by: alsa-utils_1.1.9-0ubuntu1_amd64 bug


       alsabat - command-line sound tester for ALSA sound card driver


       alsabat [flags]


       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 :-

       If tinyalsa is installed in system, user can choose tinyalsa as backend  lib  of  alsabat,
       with configure option "--enable-alsabat-backend-tiny".


       -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.

              Write stderr and stdout output to this log file.

              Input WAV file for playback.

              Target WAV file to save capture test content.

              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.

              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".

              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.

              Noise detection threshold in SNR  (dB).  26dB  indicates  5%  noise  in  amplitude.
              ALSABAT will return error if signal SNR is smaller than the threshold.

              Noise  detection  threshold  in  percentage  of  noise amplitude (%).  ALSABAT will
              return error if the noise amplitude is larger than the threshold.


       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.


       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.




       Currently only support RIFF WAV format with PCM data. Please report any bugs to the  alsa-
       devel mailing list.


       alsabat   is   by   Liam   Girdwood   <>,  Bernard  Gautier
       <> and Han Lu  <>.   This  document  is  by  Liam
       Girdwood <> and Han Lu <>.

                                        20th October 2015                              ALSABAT(1)