On Wed, 03 Feb 2016 07:20:33 +0100,
han.lu@xxxxxxxxx wrote:
>
> From: "Lu, Han" <han.lu@xxxxxxxxx>
>
> alsa-utils as well as bareos-bat (as well a some Bacula packages)
> all contain a program called /usr/bin/bat, which causes conflicts on
> various distributions ("basic audio tester" vs "bareos administration
> tool"("bacula administration tool")).
> Rename to avoid conflict.
>
> Signed-off-by: Lu, Han <han.lu@xxxxxxxxx>
Applied, thanks.
Takashi
>
> diff --git a/bat/Makefile.am b/bat/Makefile.am
> index f0dc5ab..8dfafa9 100644
> --- a/bat/Makefile.am
> +++ b/bat/Makefile.am
> @@ -1,9 +1,9 @@
> -bin_PROGRAMS = bat
> -man_MANS = bat.1
> +bin_PROGRAMS = alsabat
> +man_MANS = alsabat.1
>
> -EXTRA_DIST = bat.1
> +EXTRA_DIST = alsabat.1
>
> -bat_SOURCES = \
> +alsabat_SOURCES = \
> bat.c \
> common.c \
> analyze.c \
> @@ -21,4 +21,4 @@ noinst_HEADERS = \
> AM_CPPFLAGS = \
> -Wall -I$(top_srcdir)/include
>
> -bat_LDADD = @FFTW_LIB@
> +alsabat_LDADD = @FFTW_LIB@
> diff --git a/bat/alsabat.1 b/bat/alsabat.1
> new file mode 100644
> index 0000000..8d0b9c0
> --- /dev/null
> +++ b/bat/alsabat.1
> @@ -0,0 +1,159 @@
> +.TH ALSABAT 1 "20th October 2015"
> +.SH NAME
> +alsabat \- command\-line sound tester for ALSA sound card driver
> +
> +.SH SYNOPSIS
> +\fBalsabat\fP [\fIflags\fP]
> +
> +.SH DESCRIPTION
> +\fBALSABAT(ALSA Basic Audio Tester)\fP 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
> +
> +.SH OPTIONS
> +.TP
> +\fI\-h, \-\-help\fP
> +Help: show syntax.
> +.TP
> +\fI\-D\fP
> +Select sound card to be tested by name.
> +.TP
> +\fI\-P\fP
> +Select the playback PCM device.
> +.TP
> +\fI\-C\fP
> +Select the capture PCM device.
> +.TP
> +\fI\-f\fP
> +Sample format
> +.br
> +Recognized sample formats are: U8 S16_LE S24_3LE S32_LE
> +.br
> +Some of these may not be available on selected hardware
> +.br
> +The available format shortcuts are:
> +.nf
> +\-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]
> +.fi
> +If no format is given S16_LE is used.
> +.TP
> +\fI\-c\fP
> +The number of channels. The default is one channel.
> +Valid values at the moment are 1 or 2.
> +.TP
> +\fI\-r\fP
> +Sampling rate in Hertz. The default rate is 44100 Hertz.
> +Valid values depends on hardware support.
> +.TP
> +\fI\-n\fP
> +Duration of generated signal.
> +The value could be either of the two forms:
> +.br
> +1. Decimal integer, means number of frames;
> +.br
> +2. Floating point with suffix 's', means number of seconds.
> +.br
> +The default is 2 seconds.
> +.TP
> +\fI\-k\fP
> +Sigma k value for analysis.
> +.br
> +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:
> +.br
> +threshold = k * standard_deviation + mean_value
> +.br
> +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.
> +.br
> +ALSABAT then compares the detected frequency to target frequency, to
> +decide if the detecting passes or fails.
> +.br
> +The default value is 3.0.
> +.TP
> +\fI\-F\fP
> +Target frequency for signal generation and analysis, in Hertz.
> +The default is 997.0 Hertz.
> +Valid range is (DC_THRESHOLD, 40% * Sampling rate).
> +.TP
> +\fI\-p\fP
> +Total number of periods to play or capture.
> +.TP
> +\fI\-\-log=#\fP
> +Write stderr and stdout output to this log file.
> +.TP
> +\fI\-\-file=#\fP
> +Input WAV file for playback.
> +.TP
> +\fI\-\-saveplay=#\fP
> +Target WAV file to save capture test content.
> +.TP
> +\fI\-\-local\fP
> +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.
> +
> +.SH EXAMPLES
> +
> +.TP
> +\fBalsabat \-P plughw:0,0 \-C plughw:0,0 \-c 2 \-f S32_LE \-F 250\fR
> +Generate and play a sine wave of 250 Hertz with 2 channel and S32_LE format,
> +and then capture and analyze.
> +
> +.TP
> +\fBalsabat \-P plughw:0,0 \-C plughw:0,0 \-\-file 500Hz.wav\fR
> +Play the RIFF WAV file "500Hz.wav" which contains 500 Hertz waveform LPCM
> +data, and then capture and analyze.
> +
> +.SH RETURN VALUE
> +.br
> +On success, returns 0.
> +.br
> +If no peak be detected, returns -1001;
> +.br
> +If only DC be detected, returns -1002;
> +.br
> +If peak frequency does not match with the target frequency, returns -1003.
> +
> +.SH SEE ALSO
> +\fB
> +aplay(1)
> +\fP
> +
> +.SH BUGS
> +Currently only support RIFF WAV format with PCM data. Please report any bugs to
> +the alsa-devel mailing list.
> +
> +.SH AUTHOR
> +\fBalsabat\fP is by Liam Girdwood <liam.r.girdwood@xxxxxxxxxxxxxxx>, Bernard
> +Gautier <bernard.gautier@xxxxxxxxx> and Han Lu <han.lu@xxxxxxxxx>.
> +This document is by Liam Girdwood <liam.r.girdwood@xxxxxxxxxxxxxxx> and Han Lu
> +<han.lu@xxxxxxxxx>.
> diff --git a/bat/bat.1 b/bat/bat.1
> deleted file mode 100644
> index e00fc27..0000000
> --- a/bat/bat.1
> +++ /dev/null
> @@ -1,158 +0,0 @@
> -.TH BAT 1 "20th October 2015"
> -.SH NAME
> -bat \- command\-line sound tester for ALSA sound card driver
> -
> -.SH SYNOPSIS
> -\fBbat\fP [\fIflags\fP]
> -
> -.SH DESCRIPTION
> -\fBBAT(Basic Audio Tester)\fP is a simple command\-line utility intended
> -to help automate audio driver and sound server testing with little human
> -interaction. BAT can be used to test audio quality, stress test features
> -and test audio before and after PM state changes.
> -
> -BAT's design is relatively simple. BAT 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.
> -
> -BAT can either run wholly on the target machine being tested (standalone
> -mode) or can run as a client/server mode where by bat 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
> -
> -.SH OPTIONS
> -.TP
> -\fI\-h, \-\-help\fP
> -Help: show syntax.
> -.TP
> -\fI\-D\fP
> -Select sound card to be tested by name.
> -.TP
> -\fI\-P\fP
> -Select the playback PCM device.
> -.TP
> -\fI\-C\fP
> -Select the capture PCM device.
> -.TP
> -\fI\-f\fP
> -Sample format
> -.br
> -Recognized sample formats are: U8 S16_LE S24_3LE S32_LE
> -.br
> -Some of these may not be available on selected hardware
> -.br
> -The available format shortcuts are:
> -.nf
> -\-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]
> -.fi
> -If no format is given S16_LE is used.
> -.TP
> -\fI\-c\fP
> -The number of channels. The default is one channel.
> -Valid values at the moment are 1 or 2.
> -.TP
> -\fI\-r\fP
> -Sampling rate in Hertz. The default rate is 44100 Hertz.
> -Valid values depends on hardware support.
> -.TP
> -\fI\-n\fP
> -Duration of generated signal.
> -The value could be either of the two forms:
> -.br
> -1. Decimal integer, means number of frames;
> -.br
> -2. Floating point with suffix 's', means number of seconds.
> -.br
> -The default is 2 seconds.
> -.TP
> -\fI\-k\fP
> -Sigma k value for analysis.
> -.br
> -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:
> -.br
> -threshold = k * standard_deviation + mean_value
> -.br
> -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.
> -.br
> -BAT then compares the detected frequency to target frequency, to decide
> -if the detecting passes or fails.
> -.br
> -The default value is 3.0.
> -.TP
> -\fI\-F\fP
> -Target frequency for signal generation and analysis, in Hertz.
> -The default is 997.0 Hertz.
> -Valid range is (DC_THRESHOLD, 40% * Sampling rate).
> -.TP
> -\fI\-p\fP
> -Total number of periods to play or capture.
> -.TP
> -\fI\-\-log=#\fP
> -Write stderr and stdout output to this log file.
> -.TP
> -\fI\-\-file=#\fP
> -Input WAV file for playback.
> -.TP
> -\fI\-\-saveplay=#\fP
> -Target WAV file to save capture test content.
> -.TP
> -\fI\-\-local\fP
> -Internal loopback mode.
> -Playback, capture and analysis internal to BAT only. This is intended for
> -developers to test new BAT features as no audio is routed outside of BAT.
> -
> -.SH EXAMPLES
> -
> -.TP
> -\fBbat \-P plughw:0,0 \-C plughw:0,0 \-c 2 \-f S32_LE \-F 250\fR
> -Generate and play a sine wave of 250 Hertz with 2 channel and S32_LE format,
> -and then capture and analyze.
> -
> -.TP
> -\fBbat \-P plughw:0,0 \-C plughw:0,0 \-\-file 500Hz.wav\fR
> -Play the RIFF WAV file "500Hz.wav" which contains 500 Hertz waveform LPCM
> -data, and then capture and analyze.
> -
> -.SH RETURN VALUE
> -.br
> -On success, returns 0.
> -.br
> -If no peak be detected, returns -1001;
> -.br
> -If only DC be detected, returns -1002;
> -.br
> -If peak frequency does not match with the target frequency, returns -1003.
> -
> -.SH SEE ALSO
> -\fB
> -aplay(1)
> -\fP
> -
> -.SH BUGS
> -Currently only support RIFF WAV format with PCM data. Please report any bugs to
> -the alsa-devel mailing list.
> -
> -.SH AUTHOR
> -\fBbat\fP is by Liam Girdwood <liam.r.girdwood@xxxxxxxxxxxxxxx>, Bernard Gautier
> -<bernard.gautier@xxxxxxxxx> and Han Lu <han.lu@xxxxxxxxx>.
> -This document is by Liam Girdwood <liam.r.girdwood@xxxxxxxxxxxxxxx> and Han Lu
> -<han.lu@xxxxxxxxx>.
> diff --git a/bat/bat.c b/bat/bat.c
> index 086b9fa..ddb60b7 100644
> --- a/bat/bat.c
> +++ b/bat/bat.c
> @@ -272,7 +272,7 @@ static void test_capture(struct bat *bat)
> static void usage(struct bat *bat)
> {
> fprintf(bat->log,
> -_("Usage: bat [-options]...\n"
> +_("Usage: alsabat [-options]...\n"
> "\n"
> " -h, --help this help\n"
> " -D pcm device for both playback and capture\n"
> --
> 2.5.0
>
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