CompAudio [options] AFileA [AFileB]
Compare audio files, printing statistics
This program gathers and prints statistics for one or two input audio files.
With one input file, this program prints the statistics for that file. With
two input files, the signal-to-noise ratio (SNR) of the second file relative
to the first file is also printed. For this calculation, the first audio
file is used as the reference signal. The "noise" is the difference between
sample values in the files.
For SNR comparisons between two-channel audio files, the data is treated as
complex values. For more than two channels, the audio files are treated as
if they were single channel files.
For each file, the following statistical quantities are calculated and
printed for each channel.
Xm = SUM x(i) / N
sd = sqrt [ (SUM x(i)^2 - Xm^2) / (N-1) ]
Xmax = max (x(i))
Xmin = min (x(i))
These values are reported as a percent of full scale. For instance for
16-bit data, full scale is 32768.
Number of Overloads:
Count of values at or exceeding the full scale value. For integer data
from a saturating A/D converter, the presence of such values is an
indication of a clipped signal.
Number of Anomalous Transitions:
An anomalous transition is defined as a transition from a sample value
greater than +0.5 full scale directly to a sample value less than -0.5
full scale or vice versa. A large number of such transitions is an
indication of byte-swapped data.
This measurement calculates the active speech level using Method B of
ITU-T Recommendation P.56. The smoothed envelope of the signal is used
to divide the signal into active and non-active regions. The active
level is the average envelope value for the active region. The activity
factor in percent is also reported.
An optional delay range can be specified when comparing files. The samples
in file B are delayed relative to those in file A by each of the delay values
in the delay range. For each delay, the SNR with optimized gain factor (see
below) is calculated. For the delay corresponding to the largest SNR, the
full regalia of file comparison values is reported.
SNR = --------------------- .
SUM |xa(i) - xb(i)|^2
The corresponding value in dB is printed.
SNR with optimized gain factor:
SNR = 1 / (1 - r^2) ,
where r is the (normalized) correlation coefficient,
r = -------------------------------------- .
sqrt [ SUM |xa(i)|^2 * SUM |xb(i)|^2 ]
The SNR value in dB is printed. This SNR calculation corresponds to
using an optimized gain factor Sf for file B,
Sf = ---------------- .
This is the average of SNR values calculated for segments of data. The
default segment length corresponds to 16 ms (128 samples at a sampling
rate of 8000 Hz). However if the sampling rate is such that the segment
length is less than 64 samples or more than 1024 samples, the segment
length is set to 256 samples. For each segment, the SNR is calculated
SS(k) = 1 + ------------------------- .
eps + SUM [xa(i)-xb(i)]^2
The term eps in the denominator prevents divides by zero. The additive
unity term discounts segments with SNR's less than unity. The final
average segmental SNR in dB is calculated as
SSNR = 10 * log10 ( 10^[SUM log10 (SS(k)) / N] - 1 ) dB.
The first term in the bracket is the geometric mean of SS(k). The
subtraction of the unity term tends to compensate for the unity term
If any of these SNR values is infinite, only the optimal gain factor is
printed as part of the message (Sf is the optimized gain factor),
"File A = Sf * File B".
Input file(s): AFileA [AFileB]:
The environment variable AUDIOPATH specifies a list of directories to be
searched for the input audio file(s). Specifying "-" as the input file
indicates that input is from standard input.
-d DL:DU, --delay=DL:DU
Specify a delay range (in samples). Each delay in the delay range
represents a delay of file B relative to file A. The default range is
-s SAMP, --segment=SAMP
Segment length (in samples) to be used for calculating the segmental
signal-to-noise ratio. The default is a length corresponding to 16 ms.
-g GAIN, --gain=GAIN
A gain factor applied to the data from the input files. This gain
applies to all channels in a file. The gain value can be given as a
real number (e.g., "0.003") or as a ratio (e.g., "1/256"). This option
may be given more than once. Each invocation applies to the input files
that follow the option.
-l L:U, --limits=L:U
Sample limits for the input files (numbered from zero). Each invocation
applies to the input files that follow the option. The specification
":" means the entire file; "L:" means from sample L to the end; ":U"
means from sample 0 to sample U; "N" means from sample 0 to sample N-1.
-t FTYPE, --type=FTYPE
Input audio file type. In the default automatic mode, the input file
type is determined from the file header. For input from a non-random
access file (e.g. a pipe), the input file type can be explicitly
specified to avoid the lookahead used to read the file header. This
option can be specified more than once. Each invocation applies to the
input files that follow the option. See the description of the
environment variable AF_FILETYPE below for a list of file types.
-P PARMS, --parameters=PARMS
Parameters to be used for headerless input files. This option may be
given more than once. Each invocation applies to the input files that
follow the option. See the description of the environment variable
AF_NOHEADER below for the format of the parameter specification.
Print a list of options and exit.
Print the version number and exit.
This environment variable defines the input audio file type. In the default
mode, the input file type is determined from the file header.
"auto" - determine the input file type from the file header
"AU" or "au" - AU audio file
"WAVE" or "wave" - WAVE file
"AIFF" or "aiff" - AIFF or AIFF-C sound file
"noheader" - headerless (non-standard or no header) audio file
"SPHERE" - NIST SPHERE audio file
"ESPS" - ESPS sampled data feature file
"IRCAM" - IRCAM soundfile
"SPPACK" - SPPACK file
"INRS" - INRS-Telecom audio file
"SPW" - Comdisco SPW Signal file
"CSL" or "NSP" - CSL NSP file
"text" - Text audio file
This environment variable defines the data format for headerless or
non-standard input audio files. The string consists of a list of parameters
separated by commas. The form of the list is
"Format, Start, Sfreq, Swapb, Nchan, ScaleF"
The default values for the audio file parameters correspond to the following
Format: File data format
"undefined" - Headerless files will be rejected
"mu-law8" - 8-bit mu-law data
"A-law8" - 8-bit A-law data
"unsigned8" - offset-binary 8-bit integer data
"integer8" - two's-complement 8-bit integer data
"integer16" - two's-complement 16-bit integer data
"integer24" - two's-complement 24-bit integer data
"integer32" - two's-complement 32-bit integer data
"float32" - 32-bit floating-point data
"float64" - 64-bit floating-point data
"text" - text data
Start: byte offset to the start of data (integer value)
Sfreq: sampling frequency in Hz (floating point number)
Swapb: Data byte swap parameter
"native" - no byte swapping
"little-endian" - file data is in little-endian byte order
"big-endian" - file data is in big-endian byte order
"swap" - swap the data bytes as the data is read
Nchan: number of channels
The data consists of interleaved samples from Nchan channels
ScaleF: Scale factor
"default" - Scale factor chosen appropriate to the type of data. The
scaling factors shown below are applied to the data in the file.
8-bit mu-law: 1/32768
8-bit A-law: 1/32768
8-bit integer: 128/32768
16-bit integer: 1/32768
24-bit integer: 1/(256*32768)
32-bit integer: 1/(65536*32768)
float data: 1
"<number or ratio>" - Specify the scale factor to be applied to
the data from the file.
"undefined, 0, 8000., native, 1, default"
This environment variable specifies a list of directories to be searched when
opening the input audio files. Directories in the list are separated by
colons (semicolons for Windows).
Author / version
P. Kabal / v5r1 2003-07-11
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