FiltAudio(1audio)

Routine
FiltAudio [options] -f FilterFile AFileI AFileO

Purpose
Filter data from an audio file

Description
This program takes an audio file and a filter coefficient file as input and produces a filtered output audio file. Subsampling and interpolation factors can be optionally specified. This program supports three types of filters, FIR, all-pole, and general IIR. Filters are specified in filter files.

Filter Files:
The first record of a filter file indicates the type of filter.
  !FIR  - FIR filter, direct form
  !IIR  - IIR filter, cascade of biquad sections
  !ALL  - All-pole filter, direct form
Subsequent records contain filter coefficients. Comment records ('!' in the first position of the record) can be interspersed amongst the data. Data records are free form, with data values separated by white space (blanks, tabs and newlines). Commas can also be used to separate data values, but only within records, i.e. a comma should not appear at the end of a record.

FIR filters are specified by the direct-form coefficients h[i],
         N-1       -i
  H(z) = SUM h[i] z    .
         i=0

IIR filters are implemented as the cascade of biquadratic filter sections, where each section has a z-transform,
           h(i,0)*z^2 + h(i,1)*z + h(i,2)
  H(i,z) = ------------------------------ .
              z^2 + h(i,3)*z + h(i,4)

All-pole filters are specified by direct-form feedback coefficients,
          1                    N-1       -i
  H(z) = ----  ,  where C(z) = SUM h[i] z    .
         C(z)                  i=0

For FIR filters, a sample rate change can be affected with interpolation and subsampling. For IIR and all-pole filters, the output can be subsampled. Let Ir and Nsub be the interpolation and subsampling factors, respectively. Conceptually, the rate change is accomplished as follows.
 1: Ir-1 zeros are inserted between adjacent samples of the frequency shifted input to increase the sampling rate by a factor of Ir.
 2: The increased rate signal is filtered.
 3: The result of the filtering is subsampled by a factor of Nsub to form the output signal.

The initial filter alignment and the number of output samples can be specified with options. The filter alignment specifies the position of the filter relative to the input data for calculating the first output sample. For FIR filters, this alignment is relative to the increased rate input sequence. Specifically, let the number of samples in the input file be Nin. The input can be considered to be an array x(0),...,x(Nin-1). The increased rate sequence is xi(.), with xi(k*Ir)=x(k). The first output sample is calculated with the beginning of the impulse response of the filter aligned with xi(idoffs). The array xi(.) can be considered to be of length Nin*Ir; the first non-zero sample is xi(0)=x(0), the last non-zero sample is xi((Nin-1)*Ir). Conceptually, the impulse impulse response is moved in steps of Nsub to create the output samples.

The intent is that the output samples are the subset of the values the would be obtained if the infinite length sequence formed by padding out the input data on either end with zeros were to be filtered. To this end, the filter calculations need warm-up points, particularly for the case that the initial filter alignment is not at the beginning of the input data. For FIR filters, this is taken into account by reading previous input values into the filter memory. For IIR and all-pole filters, previous outputs are also needed as warm-up points. If the initial alignment is near the beginning of the data, the IIR filter is run from the beginning of the data to generate the warm-up points. For larger alignment offsets, the IIR filter is backed up for a maximum of 1000 samples to provide the warm-up points.

If the initial filter alignment is not explicitly specified, it is chosen to be zero, except for odd-length symmetric or anti-symmetric FIR filters for which it is is chosen to be (Ncof-1)/2. If the number of output samples is not explicitly set, it is chosen to be Ir*Nin/Nsub. For the case of Ir=1 and Nsub=1, this results in the same number of output samples as input samples. If the initial filter alignment, offs, is explicitly specified, the number of output samples is chosen to be (Ir*Nin-offs)/Nsub. This value can be overridden by explicitly setting the number of output samples.

Options
Input file name: AFileI:
The environment variable AUDIOPATH specifies a list of directories to be searched for the input audio file. Specifying "-" as the input file indicates that input is from standard input.
Output file name: AFileO:
The second file name is the output file. Specifying "-" as the output file name indicates that output is to be written to standard output. If the output file type is not explicitly given (-F option), the extension of the output file name is used to determine the file type.
  ".au"   - AU audio file
  ".wav"  - RIFF WAVE file
  ".aif"  - AIFF-C audio file
  ".raw"  - Headerless file (native byte order)
  ".txt"  - Headerless file (text data)
-f FILTFILE, --filter_file=FILTFILE
Filter file name.
-i IR/NSUB, --interpolate=IR/NSUB
Filter interpolation ratio, default 1. The interpolation and subsampling factors are specified as a fraction Ir/Nsub. Interpolation can only be used with FIR filters.
-a OFFS, --alignment=OFFS
Alignment of data relative to the filter. The first output sample is calculated with the beginning of the filter response aligned with the specified sample of the interpolated data sequence.
-n NFRAME, --number_frames=NFRAME
Number of sample frames for the output file.
-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. 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. See the description of the environment variable AF_NOHEADER below for the format of the parameter specification.
-g GAIN, --gain=GAIN
A gain factor applied to the data from the input file. 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 fraction (e.g., "1/256").
-F FTYPE, --file_type=FTYPE
Output file type.
  "AU" or "au"             - AU audio file
  "WAVE" or "wave"         - RIFF WAVE file
  "AIFF-C" "aiff-c"        - AIFF-C audio file
  "noheader" or "noheader_native" - Headerless file (native byte
                             order)
  "noheader_swap"          - Headerless file (byte swapped)
  "noheader_big-endian"    - Headerless file (big-endian byte
                             order)
  "noheader_little-endian" - Headerless file (little-endian byte
                             order)
-D DFORMAT, --data_format=DFORMAT
Data format for the output file.
  "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
The data formats available depend on the output file type. AU audio files:
  mu-law, A-law, 8-bit integer, 16-bit integer, 32-bit float, 64-bit
  float
RIFF WAVE files:
  mu-law, A-law, offset-binary 8-bit integer, 16-bit integer
AIFF-C audio files:
  mu-law, A-law, 8-bit integer, 16-bit integer
Headerless files:
  all data formats
-I INFO, --info=INFO
Audio file information string for the output file.
-h, --help
Print a list of options and exit.
-v, --version
Print the version number and exit.

By default, the output file contains a standard audio file information string.
  Standard Audio File Information:
     date: 1994-01-25 19:19:39 UTC    date
     user: kabal@aldebaran            user
     program: FiltAudio               program name
This information can be changed with the header information string which is specified as one of the command line options. Structured information records should adhere to the above format with a named field terminated by a colon, followed by numeric data or text. Comments can follow as unstructured information.
  Record delimiter: Newline character or the two character escape
      sequence "\" + "n".
  Line delimiter: Within records, lines are delimiteded by a carriage
      control character, the two character escape sequence "\" + "r",
      or the two character sequence "\" + newline.
If the information string starts with a record delimiter, the header information string is appended to the standard header information. If not, the user supplied header information string appears alone.

Environment variables
AF_FILETYPE:
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" - RIFF WAVE file
  "AIFF" or "aiff" - AIFF or AIFF-C audio 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
  "text"       - Text audio file

AF_NOHEADER:
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"
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
Scale factor applied to the data from the file
The default values for the audio file parameters correspond to the following string.
  "undefined, 0, 8000., native, 1, 1.0"

AUDIOPATH:
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 / v5r0 2000-11-21 Copyright (C) 2000

See Also
ResampAudio, AFsp