Record
[
NextMusic Home Page
|
Sound and Music Programs Page
|
Program List ]
Author: Jean Laroche
Platforms: NeXT, ?
Prerequisites: NS2.2 or higher?
Price: free
Demo: N/A
To acquire: ftp://ftp.eecs.umich.edu/pub/nextmusic/Record.tar.Z (25k)
Information: ?
Entry updated: 12 November 1996
Description: A command-line utility that records soundfiles at any sampling rate from an A/D64X.
RECORD(1) UNIX Programmer's Manual RECORD(1)
NAME
record - record sounds at any sampling rate from A/D64X.
SYNOPSIS
record [ -UDSKtovsosibcr ] soundFile
DESCRIPTION
record records from the A/D64X interface, simultaneously
converting to any specified sampling rate. record can
either record from the digital input (for direct digital
recording from a DAT machine or a CD player) or from the
analog input (using the A/D64X analog-digital converters).
When analog input is selected, record chooses the A/D64X
quartz frequency (32, 44.1 or 48kHz) that makes the conver-
sion to the desired sampling rate as easy as possible,
unless the A/D64X sampling rate is also specified. When
recording from digital input, you need to indicate the
SPDIFF sampling rate (either 32, 44.1 or 48kHz). record can
record in stereo, or in mono (selecting one of the two chan-
nels, or mixing them), and write the output sound with or
without a header. record can also swap the output sound's
bytes, for compatibility with DEC machines, PCs etc...
record can be forced to do or not do direct-to-disk record-
ing.
OPTIONS
-v Tells record to shut up. By default, record gives all
recording informations.
-S sampling rate
This specifies the sound's output sampling rate. If
this sampling rate is not 32kHz, 44.1kHz nor 48kHz,
record will perform a real-time sampling rate conver-
sion to the specified sampling rate.
-L length in seconds
With this option, you can specify the duration in
seconds of the signal that will be recorded. record
then asks you to start the recording by hiting the
"return" key. If this option is not used, record will
ask you to start and to stop the recording.
-a (default) used to select the analog input (no sampling
rate specified, record will choose the most appropriate
one according to the requested output sampling rate).
-dh used to select digital input, indicating that the sig-
nal coming from the digital input is in 48kHz.
-dm used to select digital input, indicating that the sig-
nal coming from the digital input is in 44.1kHz.
-dl used to select digital input, indicating that the sig-
nal coming from the digital input is in 32kHz.
-b Soundfile output. With this option, the converted sound
is written in a soundfile, with a 28 byte header. If -b
is not specified, the output is written in a raw,
headerless file (short ints).
-cs records the signal in stereo mode. Both channels are
interleaved, starting with the left channel.
-cl record only the signal's left channel.
-cr record only the signal's right channel.
-c+ mixes the signal's two channels, and records in mono.
-w This option makes it possible to rescale the recorded
sound. Use this if the recorded sound clicks. The
default rescaling value is 0.9. This option is disabled
when bit faithful recording is performed.
-s This can be used to swap the output sound bytes. Use
this if you want the recorded sound to be in DEC or in
PC format. This option is valid only with the -b
option.
-A Disables direct-to-disk recording. In default mode, the
sound is saved on the disk at the same time it's
recorded (direct-to-disk recording), which limits the
size of the virtual memory allocated by the record pro-
cess. However, in some cases, direct-to-disk recording
fails because the disk or the network cannot keep up
with the rate at which samples are coming in. This typ-
ically happens when recording at a high sampling rate
in stereo mode and storing in a file located on another
machine. In that case, the -A flag disables direct-to-
disk recording: the sound is stored in memory, and save
at the end of the recording. The danger with this is
that the record process can become extremely fat
memorywise.
-al This is used to specify analog input, with low A/D64X
sampling rate (32kHz). You don't normally use this
option.
-am This is used to specify analog input, with medium
A/D64X sampling rate (44.1kHz). You don't normally use
this option.
-ah This is used to specify analog input, with high A/D64X
sampling rate (48kHz). You don't normally use this
option.
-r forces the output header to contain the exact frequency
you specified with the -S option. Due to the way record
works, it sometimes can't convert exactly to the
requested sampling rate, but only to an approximation.
Normally, the approximated sampling rate is written in
the output sound's header. If you use the -r option,
the exact requested sampling rate is written instead.
-K filter length
This can be use to specify the filter length you want
to use. This length is normally calculated
automatically, so you shouldn't need to use this
option.
-B This can be use to specify the type of filter windowing
you want to use. By default, a Hanning window is used,
giving a rejection of about -40dB to -50dB. It may be
necessary in some cases to use a Blackman window to get
more rejection (about -60dB), at the expense of a wider
transition band. This can be done by use of the -B
flag.
-U upward conversion factor
upward conversion factor. Normally, you don't need to
use this option.
-D downward conversion factor
downward conversion factor. Normally, you don't need to
use this option.
EXAMPLE
"record -a -b -S16000 file" records a stereo soundfile at
16kHz from analog input. The output soundfile has a 28 bit
header.
"record -dh -cl file" records a headerless mono file (left
channel) from digital input (at 48kHz). The transfer is
bit-faithful (no conversion, digital transfer).
"record -dh -c+ -S16000 file" records a headerless mono file
(both channels mixed) from digital input (48kHz) and down
samples it to 16kHz.
SEE ALSO
sndinfo(1), srconv, play, fromsnd, tosnd
AUTHOR
Jean Laroche, June 1992, TELECOM PARIS.
Eric M. Mrozek (mrozek@umich.edu), EECS-Systems, University of Michigan