Routine

CopyAudio [options] AFileA AFileB ... AFileO

Purpose

Copy audio files (combine/concatenate, scale and shift data)

Description

This program copies samples from one or more input audio files to an output audio file. The samples in the output file are linear combinations of the samples in the different channels in the input files. In the combine mode, data from multiple input files is combined. In the concatenate mode, data from multiple input files is concatenated.

The output consists of linear combinations of the input data channels. For instance output channel A can be formed as the average of input channel A and input channel B. For multichannel data, the data is organized into sample frames, with samples from individual channels making up a frame.

Sample limits may be specified for the input data. The same sample limits apply to each channel in a file. Samples in the file are numbered from zero. Negative sample limits may be specified; the samples corresponding to negative indices have zero values. Similarly, sample limits beyond the end-of-file may be specified; samples beyond the end-of-file are assumed to have zero values.

The combine and concatenate modes differ in how they treat multiple input files. For the combine mode, if there is more than one input file, channels assignments cross between input files. Consider two input files, with the first having 3 channels, and the second having 2 channels. The channels are labelled A to E, with channels A, B and C coming from the first input file and channels D and E coming from the second input file. For the combine mode, a single sample limits specification applies to all input files. The default sample limits correspond to the length of the longest input file. If necessary, the data from the input files is padded with zeros to a common length before combining.

In the concatenate mode, the data in the input files is concatenated. The number of channels in each input file must be the same. For the concatenate mode, sample limits can be specified for each input file. The default sample limits for an input file correspond to the length of that file.

For fixed point data, the data is normalized to the range -1 to +1. This normalization becomes important when files with different data formats are combined.

The default data format for the output file is chosen according to a data format promotion rule based on the data types of the input files. For single input files, the output data format will be the same as the input data format as long as that data format is compatible with the output file type.

Options

Input file names: 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 name indicates that input is from standard input. Output file name: AFileO:
The last 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"  - WAVE file
  ".aif"  - AIFF sound file
  ".afc"  - AIFF-C sound file
  ".raw"  - Headerless file (native byte order)
  ".txt"  - Headerless file (text data)

-c, --combine
Use the combine mode for multiple input files (default) -C, --concatenate
Use the concatenate mode for multiple input files -g GAIN, --gain=GAIN
A gain factor applied to the data from the input files. This gain applies to all channels in a file and is applied in addition to the channel scaling factors. 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. 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). For the combine mode, the default sample limits correspond to the limits of the longest input audio file. For the concatenate mode, this option may be given more than once. 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. -s SFREQ, --srate=SFREQ
Sampling frequency for the output file, default from the input audio file(s). This option only changes the sampling frequency field in the output header; the audio data itself is unaffected. -n NSAMPLE, --number_samples=NSAMPLE
Number of samples (per channel) for the output file. -cA CGAINS, --chanA=CGAINS
Channel scaling factors for output channel A. The output data for the specified output channel is created by adding scaled samples from the specified input channels. The default is to have each output channel equal to the corresponding input channel. Input channels are labelled A, B, C, ... , R, S, T. The channel scaling factor string takes the form

  [+|-] [gain *] chan +|- [gain *] chan ... +|- offset

where chan is A through L. The gains can be expressed as ratios, i.e. of the form "n/m". The offset is in normalized units, where an offset of one corresponds to full scale. Note that that the character "*" is a special character to Unix shells and should appear only within quotes to prevent the shell from interpreting it. -cB CGAINS, --chanB=CGAINS
Channel scaling factors for output channel B.

...

-cL CGAINS, --chanL=CGAINS
Channel scaling factors for output channel L. -F FTYPE, --file_type=FTYPE
Output file type. If this option is not specified, the file type is determined by the output file name extension.

  "AU" or "au"             - AU audio file
  "WAVE" or "wave"         - WAVE file
  "WAVE-NOEX" or "wave-noex" - WAVE file (no extensible data)
  "AIFF-C" "aiff-c"        - AIFF-C sound file
  "AIFF" or "aiff"         - AIFF sound 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, 8/16/24/32-bit integer, 32/64-bit float

WAVE files:

  mu-law, A-law, offset-binary 8-bit integer, 16/24/32-bit integer,
  32/64-bit float

AIFF-C sound files:

  mu-law, A-law, 8/16/24/32-bit integer, 32/64-bit float

AIFF sound files:

  8/16/24/32-bit integer

Headerless files:

  all data formats

Optionally, the number of "valid" bits may be specified as a qualifier to the data format, e.g. "integer16/12". The number of valid bits is for informational purposes only. -S SPEAKERS, --speakers=SPEAKERS
Mapping of output channels to speaker positions. The spacial positions of the loudspeakers are specified as a list of white-space separated locations from the list below.

  "FL"  - Front Left
  "FR"  - Front Right
  "FC"  - Front Center
  "LF"  - Low Frequency
  "BL"  - Back Left
  "BR"  - Back Right
  "FLC" - Front Left of Center
  "FRC" - Front Right of Center
  "BC"  - Back Center
  "SL"  - Side Left
  "SR"  - Side Right
  "TC"  - Top Center
  "TFL" - Top Front Left
  "TFC" - Top Front Center
  "TFR" - Top Front Right
  "TBL" - Top Back Lefty
  "TBC" - Top Back Center
  "TBR" - Top Back Right
  "-"   - none

A speaker position can be associated with only one channel. In the case of WAVE files, the subset of spacial positions must appear in the order given above. -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.

This program allows direct specification of the gains for 12 output channels and 20 input channels. The program can handle larger numbers of channels for the case that the input channels are in a one-to-one correspondence with the output channels. A gain factor applying to all channels can be specified with the -g or --gain option.

By default, the output file contains a standard audio file information string.

  Standard Audio File Information:
     date: 2001-01-25 19:19:39 UTC    date
     program: CopyAudio               program name

This information can be changed with the 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.

Examples

1: File convert.
Copy audio file abc.au to abc.wav. The output audio file is a WAVE file with the same data type (if possible) as the input file. The number of channels in the output file is the same as the number of channels in the input file.

  CopyAudio abc.au abc.wav

2: Concatenate audio data.
Form a audio file which is the concatenation of the data from two input files.

  CopyAudio -C abc1.au abc2.au abc12.au

3: Difference between values.
Create an output audio file in which each sample is the difference between corresponding samples in two single channel input audio files.

  CopyAudio --chanA=A-B abc1.au abc2.au diff.au

4: Scale sample values.
Scale the samples in the input single-channel file by 0.5.

  CopyAudio --gain=1/2 abc.au scaled.au

The same result can be obtained by specifying the gain for the (single) output channel.

  CopyAudio --chanA="0.5*A" abc.au scaled.au

5: Byte-swap data values.
Let the input audio file be headerless and contain 16-bit data. Create a headerless audio file with byte-swapped data.

  CopyAudio -P integer16 -F noheader_swap abc.au swap.au

6: Extract samples.
Extract samples 1000 to 1999 inclusive from the input audio file. The output audio file will have 1000 samples.

  CopyAudio -l 1000:1999 abc.au out.au

7: Create a stereo file.
Form a stereo (2-channel) audio file from two single channel audio files.

  CopyAudio abc1.wav abc2.wav -S "FL FR" stereo.wav

8: Add a dc value to a file.
If the input file has a mean value of 8.63/32768, the output file will have a zero mean after adding -8.63/32768 to each sample.

  CopyAudio --chanA="A-8.63/32768" abc.au zeromean.au

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" - 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

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"

The list can be shortened and elements skipped (nothing between adjacent commas. The missing parameters take on default values.
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.

The default values for the audio file parameters correspond to the following string.

  "undefined, 0, 8000., native, 1, default"

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 / v6r0 2003-05-08