# doc-cache created by Octave 9.2.0
# name: cache
# type: cell
# rows: 3
# columns: 5
# name: <cell-element>
# type: sq_string
# elements: 1
# length: 16
voicebox_ditherq


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 448
 -- Function: voicebox_ditherq
     DITHERQ  add dither and quantize [Y,ZF]=(X,M,ZI)
       Inputs:
           x   is the input signal
     	   m   specifies the mode:
               'w'  white dither (default)
               'h'  high-pass dither (filtered by 1 - z^-1)
               'l'  low pass filter  (filtered by 1 + z^-1)
               'n'  no dither
     *Url*:
     <http://ltfat.github.io/doc/thirdparty/voicebox/voicebox_ditherq.html>


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
DITHERQ  add dither and quantize [Y,ZF]=(X,M,ZI)
  Inputs:
      x   is the i...



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 17
voicebox_pcma2lin


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 936
 -- Function: voicebox_pcma2lin
     PCMU2LIN Convert A-law PCM to linear X=(P,M,S)
     	lin = pcma2lin(pcma,m,s) where pcma contains a vector or matrix
     	of A-law values in the range 0 to 255.
     	No checking is performed to see that numbers are in this range.
     
     	Input values are exclusive ored with m (default=85)
     
     	Output values are divided by the scale factor s:
     
     		   s		Output Range
     
     		   1		+-4032	(integer values)
     		2017.396342	+-1.998616 (default)
     		4032		+-1
     		4096		+-0.984375 (+-1 nominal full scale)
     
     	The default value of s is 2017.396342 which equals
     	sqrt((1120^2 + 2624^2)/2). This factor follows ITU standard G.711 and
     	the sine wave with PCM-A values [225 244 244 225 97 116 116 97]
     	has a mean square value of unity corresponding to 0 dBm0.
     *Url*:
     <http://ltfat.github.io/doc/thirdparty/voicebox/voicebox_pcma2lin.html>


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
PCMU2LIN Convert A-law PCM to linear X=(P,M,S)
	lin = pcma2lin(pcma,m,s) wher...



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 17
voicebox_pcmu2lin


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 842
 -- Function: voicebox_pcmu2lin
     PCMU2LIN Convert Mu-law PCM to linear X=(P,S)
     	lin = pcmu2lin(pcmu) where pcmu contains a vector
     	of mu-law values in the range 0 to 255.
     	No checking is performed to see that numbers are in this range.
     
     	Output values are divided by the scale factor s:
     
     		   s		Output Range
     
     		   1		+-8031	(integer values)
     		4004.2	+-2.005649 (default)
     		8031		+-1
     		8159		+-0.9843118 (+-1 nominal full scale)
     
     	The default scaling factor 4004.189931 is equal to
     	sqrt((2207^2 + 5215^2)/2) this follows ITU standard G.711.
     	The sine wave with PCM-Mu values [158 139 139 158 30 11 11 30]
     	has a mean square value of unity corresponding to 0 dBm0.
     *Url*:
     <http://ltfat.github.io/doc/thirdparty/voicebox/voicebox_pcmu2lin.html>


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
PCMU2LIN Convert Mu-law PCM to linear X=(P,S)
	lin = pcmu2lin(pcmu) where pcm...



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7
wavload


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 5147
 -- Function: wavload
     WAVLOAD  Read a .WAV format sound file [Y,FS,WMODE,FIDX]=(FILENAME,MODE,NMAX,NSKIP)
     
      Input Parameters:
     
     	FILENAME gives the name of the file (with optional .WAV extension) or alternatively
                      can be the FIDX output from a previous call to WAVLOAD
     	MODE		specifies the following (*=default):
     
         Scaling: 's'    Auto scale to make data peak = +-1
                  'r'    Raw unscaled data (integer values)
                  'q'    Scaled to make 0dBm0 be unity mean square
                  'p' 	 Scaled to make +-1 equal full scale
                  'o'    Scale to bin centre rather than bin edge (e.g. 127 rather than 127.5 for 8 bit values)
                          (can be combined with n+p,r,s modes)
                  'n'    Scale to negative peak rather than positive peak (e.g. 128.5 rather than 127.5 for 8 bit values)
                          (can be combined with o+p,r,s modes)
                  'g'    Scale by the gain written by the "g" option in "writewav" to restore original level
          Offset: 'y' 	 Correct for offset in <=8 bit PCM data
                  'z'    No offset correction
        File I/O: 'f'    Do not close file on exit
        Display;  'h'    Print header information
                  'w'    Plot waveform
                  'W'    Plot spectrogram (max 10 seconds)
                  'a'    play audio (max 10 seconds)
                  'A'    play all audio even if very long
                  'i'    Read header only.
     
     	NMAX     maximum number of samples to read (or -1 for unlimited [default])
     	NSKIP    number of samples to skip from start of file
                    (or -1 to continue from previous read when FIDX is given instead of FILENAME [default])
     
      Output Parameters:
     
     	Y        data matrix of dimension (samples,channels)
     	FS       sample frequency in Hz
     	WMODE    mode string needed for WRITEWAV to recreate the data file
     	FIDX     Information row vector containing the element listed below.
     
                (1)  file id
                (2)  current position in file
                (3)  dataoff	byte offset in file to start of data
                (4)  nsamp	number of samples
                (5)  nchan	number of channels
                (6)  nbyte	bytes per data value
                (7)  bits	number of bits of precision
                (8)  code	Data format: 1=PCM, 2=ADPCM, 3=floating point, 6=A-law, 7=Mu-law
                (9)  fs	    sample frequency
                (10) mask   channel mask
                (11) gain   gain factor in dB
     
        If no output parameters are specified, header information will be printed.
     
        !!WARNING!!
        Please note that this function cannot handle compressed wav files.
     
        For stereo data, y(:,1) is the left channel and y(:,2) the right
        The mask, if specified, is a bit field giving the channels present in the following order:
        0=FL, 1=FR, 2=FC, 3=W, 4=BL, 5=BR, 6=FLC, 7=FRC, 8=BC, 9=SL, 10=SR, 11=TC, 12=TFL, 13=TFC, 14=TFR, 15=TBL, 16=TBC, 17=TBR
        where F=front, L=left, C=centre, W=woofer (low frequency), B=back, LC=left of centre, RC=right of centre, S=side, T=top
        
        * Note on scaling **
        If we want to scale signal values in the range +-1 to an integer in the
        range [-128,127] then we have four plausible choices corresponding to
        scale factors of (a) 127, (b) 127.5, (c) 128 or (d) 128.5 but each choice
        has disadvantages.
        For forward scaling: (c) and (d) cause clipping on inputs of +1.
        For reverse scaling: (a) and (b) can generate output values < -1.
        Any of these scalings can be selected via the mode input: (a) 'o', (b) default, (c) 'on', (d) 'n'
     
     	   Copyright (C) Mike Brookes 1998-2011
           Version: Id: readwav.m 713 2011-10-16 14:45:43Z dmb 
     
        VOICEBOX is a MATLAB toolbox for speech processing.
        Home page: http://www.ee.ic.ac.uk/hp/staff/dmb/voicebox/voicebox.html
     
        * NOTE ON CHANGES **
        The original name of this function in VOICEBOX was readwav. It was
        renamed to avoid possible namespace clash.
        Modified by: Zdenek Prusa 2015
     
     
        This program is free software; you can redistribute it and/or modify
        it under the terms of the GNU General Public License as published by
        the Free Software Foundation; either version 2 of the License, or
        (at your option) any later version.
     
        This program is distributed in the hope that it will be useful,
        but WITHOUT ANY WARRANTY; without even the implied warranty of
        MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
        GNU General Public License for more details.
     
        You can obtain a copy of the GNU General Public License from
        http://www.gnu.org/copyleft/gpl.html or by writing to
        Free Software Foundation, Inc.,675 Mass Ave, Cambridge, MA 02139, USA.
     
     *Url*:
     <http://ltfat.github.io/doc/thirdparty/voicebox/wavload.html>


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 80
WAVLOAD  Read a .WAV format sound file [Y,FS,WMODE,FIDX]=(FILENAME,MODE,NMAX,...



# name: <cell-element>
# type: sq_string
# elements: 1
# length: 7
wavsave


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 4238
 -- Function: wavsave
     WAVSAVE Creates .WAV format sound files FIDX=(D,FS,FILENAME,MODE,NSKIP,MASK)
     
        The input arguments for WAVSAVE are as follows:
     
            D           The sampled data to save
            FS          The rate at which the data was sampled
            FILENAME    A string containing the name of the .WAV file to create or
                             alternatively the FIDX output from a previous wavsave call
            MODE        String containing any reasonable mixture of flags below (*=default):
            NSKIP       Number of samples to skip before writing or -1[default] to continue from previous write
                        Only valid if FIDX is specified for FILENAME
            MASK        specifies the speaker positions included as a bit mask (see readwav)
     
      MODE flags (*=default):
       Precision: 'a'    for 8-bit A-law PCM
                  'u'    for 8-bit mu-law PCM
                 '16' 	 for 16 bit PCM data
                  '8'    for 8 bit PCM data
                  ...    any number in the range 2 to 32 for PCM
                  'v'    32-bit floating point
                  'V'    64-bit floating point
                  'c'    embed in 16 bits
                  'C'    embed in 24 bits
                  'L'    embed in 32 bits
     	  Dither: 'w'    White triangular dither of amplitude +-1 LSB (PCM modes only)
                  'h'    High pass dither (filtered by 1-1/z) (PCM modes only)
                  'l'    Low pass dither (filtered by 1+1/z) (PCM modes only)
         Scaling: 's'    Auto scale to make data peak = +-1
                  'r'    Raw unscaled data (integer values)
                  'q'    Scaled to make unity mean square correspond to 0dBm according to G.711
                  'p'   Scaled to make +-1 equal full scale
                  'o'    Scale to bin centre rather than bin edge (e.g. 127 rather than 127.5 for 8 bit values)
                          (can be combined with n+p,r,s modes)
                  'n'    Scale to negative peak rather than positive peak (e.g. 128.5 rather than 127.5 for 8 bit values)
                          (can be combined with o+p,r,s modes)
                  'g'    Include a gain factor so that "readwav" will restore the correct level
          Offset: 'y' 	 Correct for offset in <=8 bit PCM data
                  'z'    Do not apply offset correction
          Format: 'x'    use WAVEFORMATEX format (default for non PCM)
                  'X'    use WAVEFORMATEXTENSIBLE (default if MASK input is given)
                  'e'    use original WAVEFORMAT (default for PCM)
                  'E'    include a 'fact' chunk (default for non-PCM)
        File I/O: 'f'    Do not close file on exit
     
     
      Output Parameter:
     
     	FIDX     Information row vector containing the element listed below.
     
                (1)  file id
                (2)  current position in file (in samples, 0=start of file)
                (3)  dataoff	length of file header in bytes
                (4)  nsamp	number of samples
                (5)  nchan	number of channels
                (6)  nbyte	bytes per data value
                (7)  bits	number of bits of precision
                (8)  code	Data format: 1=PCM, 2=ADPCM, 6=A-law, 7=Mu-law
                (9)  fs	sample frequency
                (10) dither state variable
                (11) gain in dB (in INST chunk)
     
        Note: WAVSAVE will create an 16-bit PCM, auto-scaled wave file by default.
        For stereo data, d(:,1) is the left channel and d(:,2) the right
     
        * Note on scaling **
        If we want to scale signal values in the range +-1 to an integer in the
        range [-128,127] then we have four plausible choices corresponding to
        scale factors of (a) 127, (b) 127.5, (c) 128 or (d) 128.5 but each choice
        has disadvantages.
        For forward scaling: (c) and (d) cause clipping on inputs of +1.
        For reverse scaling: (a) and (b) can generate output values < -1.
        Any of these scalings can be selected via the mode input: (a) 'o', (b) default, (c) 'on', (d) 'n'
     *Url*:
     <http://ltfat.github.io/doc/thirdparty/voicebox/wavsave.html>


# name: <cell-element>
# type: sq_string
# elements: 1
# length: 76
WAVSAVE Creates .WAV format sound files FIDX=(D,FS,FILENAME,MODE,NSKIP,MASK)





