alsabat

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Az alsabat linux parancs manual oldala és súgója. Az ALSABAT (ALSA Basic Audio Tester) egy egyszerű parancssori segédprogram, amelynek célja az audio meghajtók és hangszerverek tesztelésének automatizálása, kevés emberi interakcióval. Az ALSABAT használható a hangminőség, a stresszteszt jellemzőinek tesztelésére és a hang tesztelésére a PM állapot változása előtt és után.

 

 

Man oldal kimenet

man 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 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)

 

 

Súgó kimenet

alsabat --help
alsa-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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