mirror of
https://github.com/alsa-project/alsa-utils
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dea85a314b
Halve the double negative ‘if no loopback mode is not available’. Signed-off-by: Tobias Geerinckx-Rice <me@tobias.gr> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
185 lines
5.9 KiB
Groff
185 lines
5.9 KiB
Groff
.TH ALSABAT 1 "20th October 2015"
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.SH NAME
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alsabat \- command\-line sound tester for ALSA sound card driver
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.SH SYNOPSIS
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\fBalsabat\fP [\fIflags\fP]
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.SH DESCRIPTION
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\fBALSABAT(ALSA Basic Audio Tester)\fP is a simple command\-line utility
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intended to help automate audio driver and sound server testing with little
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human interaction. ALSABAT can be used to test audio quality, stress test
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features and test audio before and after PM state changes.
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ALSABAT's design is relatively simple. ALSABAT plays an audio stream and
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captures the same stream in either a digital or analog loop back. It then
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compares the captured stream using a FFT to the original to determine if
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the test case passes or fails.
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ALSABAT can either run wholly on the target machine being tested (standalone
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mode) or can run as a client/server mode where by alsabat client runs on the
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target and runs as a server on a separate tester machine. The client/server
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mode still requires some manual interaction for synchronization, but this
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is actively being developed for future releases.
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The hardware testing configuration may require the use of an analog cable
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connecting target to tester machines or a cable to create an analog
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loopback if no loopback mode is available on the sound hardware that is
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being tested.
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An analog loopback cable can be used to connect the "line in" to "line out"
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jacks to create a loopback. If only headphone and mic jacks (or combo jack)
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are available then the following simple circuit can be used to create an
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analog loopback :-
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https://source.android.com/devices/audio/loopback.html
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If tinyalsa is installed in system, user can choose tinyalsa as backend lib
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of alsabat, with configure option "--enable-alsabat-backend-tiny".
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.SH OPTIONS
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.TP
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\fI\-h, \-\-help\fP
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Help: show syntax.
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.TP
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\fI\-D\fP
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Select sound card to be tested by name.
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.TP
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\fI\-P\fP
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Select the playback PCM device.
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.TP
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\fI\-C\fP
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Select the capture PCM device.
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.TP
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\fI\-f\fP
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Sample format
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.br
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Recognized sample formats are: U8 S16_LE S24_3LE S32_LE
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.br
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Some of these may not be available on selected hardware
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.br
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The available format shortcuts are:
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.nf
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\-f cd (16 bit little endian, 44100, stereo) [\-f S16_LE \-c2 \-r44100]
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\-f dat (16 bit little endian, 48000, stereo) [\-f S16_LE \-c2 \-r48000]
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.fi
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If no format is given S16_LE is used.
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.TP
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\fI\-c\fP
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The number of channels. The default is one channel.
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Valid values at the moment are 1 or 2.
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.TP
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\fI\-r\fP
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Sampling rate in Hertz. The default rate is 44100 Hertz.
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Valid values depends on hardware support.
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.TP
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\fI\-n\fP
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Duration of generated signal.
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The value could be either of the two forms:
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.br
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1. Decimal integer, means number of frames;
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.br
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2. Floating point with suffix 's', means number of seconds.
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.br
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The default is 2 seconds.
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.TP
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\fI\-k\fP
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Sigma k value for analysis.
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.br
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The analysis function reads data from WAV file, run FFT against the data
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to get magnitude of frequency vectors, and then calculates the average
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value and standard deviation of frequency vectors. After that, we define
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a threshold:
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.br
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threshold = k * standard_deviation + mean_value
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.br
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Frequencies with amplitude larger than threshold will be recognized as a
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peak, and the frequency with largest peak value will be recognized as a
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detected frequency.
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.br
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ALSABAT then compares the detected frequency to target frequency, to
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decide if the detecting passes or fails.
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.br
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The default value is 3.0.
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.TP
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\fI\-F\fP
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Target frequency for signal generation and analysis, in Hertz.
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The default is 997.0 Hertz.
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Valid range is (DC_THRESHOLD, 40% * Sampling rate).
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.TP
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\fI\-p\fP
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Total number of periods to play or capture.
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.TP
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\fI\-\-log=#\fP
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Write stderr and stdout output to this log file.
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.TP
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\fI\-\-file=#\fP
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Input WAV file for playback.
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.TP
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\fI\-\-saveplay=#\fP
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Target WAV file to save capture test content.
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.TP
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\fI\-\-local\fP
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Internal loopback mode.
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Playback, capture and analysis internal to ALSABAT only. This is intended
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for developers to test new ALSABAT features as no audio is routed outside
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of ALSABAT.
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.TP
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\fI\-\-standalone\fP
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Add support for standalone mode where ALSABAT will run on a different machine
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to the one being tested.
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In standalone mode, the sound data can be generated, playback and captured
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just like in normal mode, but will not be analyzed.
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The ALSABAT being built without libfftw3 support is always in standalone mode.
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The ALSABAT in normal mode can also bypass data analysis using option
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"--standalone".
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.TP
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\fI\-\-roundtriplatency\fP
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Round trip latency test.
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Audio latency is the time delay as an audio signal passes through a system.
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There are many kinds of audio latency metrics. One useful metric is the
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round trip latency, which is the sum of output latency and input latency.
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.TP
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\fI\-\-snr\-db=#\fP
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Noise detection threshold in SNR (dB). 26dB indicates 5% noise in amplitude.
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ALSABAT will return error if signal SNR is smaller than the threshold.
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.TP
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\fI\-\-snr\-pc=#\fP
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Noise detection threshold in percentage of noise amplitude (%).
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ALSABAT will return error if the noise amplitude is larger than the threshold.
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.SH EXAMPLES
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.TP
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\fBalsabat \-P plughw:0,0 \-C plughw:0,0 \-c 2 \-f S32_LE \-F 250\fR
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Generate and play a sine wave of 250 Hertz with 2 channel and S32_LE format,
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and then capture and analyze.
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.TP
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\fBalsabat \-P plughw:0,0 \-C plughw:0,0 \-\-file 500Hz.wav\fR
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Play the RIFF WAV file "500Hz.wav" which contains 500 Hertz waveform LPCM
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data, and then capture and analyze.
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.SH RETURN VALUE
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.br
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On success, returns 0.
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.br
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If no peak be detected, returns -1001;
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.br
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If only DC be detected, returns -1002;
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.br
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If peak frequency does not match with the target frequency, returns -1003.
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.SH SEE ALSO
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\fB
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aplay(1)
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\fP
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.SH BUGS
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Currently only support RIFF WAV format with PCM data. Please report any bugs to
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the alsa-devel mailing list.
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.SH AUTHOR
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\fBalsabat\fP is by Liam Girdwood <liam.r.girdwood@linux.intel.com>, Bernard
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Gautier <bernard.gautier@intel.com> and Han Lu <han.lu@intel.com>.
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This document is by Liam Girdwood <liam.r.girdwood@linux.intel.com> and Han Lu
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<han.lu@intel.com>.
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