-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, 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".
--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 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.