octave-audio 1.1.4-4build1 / usr / share / octave / packages / audio-1.1.4 / doc-cache

# Created by Octave 3.8.0, Mon Feb 24 19:30:04 2014 UTC <root@toyol>
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au


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 y = au(x, fs, lo [, hi])

 Extract data from x for time range lo to hi in milliseconds.  If lo
 is [], start at the beginning.  If hi is [], go to the end.  If hi is
 not specified, return the single element at lo.  If lo<0, prepad the
 signal to time lo.  If hi is beyond the end, postpad the signal to
 time hi.



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 y = au(x, fs, lo [, hi])



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auload


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 -- Function File: [X,FS,SAMPLEFORMAT] = auload (FILENAME)

     Reads an audio waveform from a file given by the string FILENAME.
     Returns the audio samples in data, one column per channel, one row
     per time slice.  Also returns the sample rate and stored format
     (one of ulaw, alaw, char, int16, int24, int32, float, double).  The
     sample value will be normalized to the range [-1,1] regardless of
     the stored format.

             [x, fs] = auload(file_in_loadpath("sample.wav"));
             auplot(x,fs);

     Note that translating the asymmetric range [-2^n,2^n-1] into the
     symmetric range [-1,1] requires a DC offset of 2/2^n.  The inverse
     process used by ausave requires a DC offset of -2/2^n, so loading
     and saving a file will not change the contents.  Other applications
     may compensate for the asymmetry in a different way (including
     previous versions of auload/ausave) so you may find small
     differences in calculated DC offsets for the same file.




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Reads an audio waveform from a file given by the string FILENAME.



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auplot


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 -- Function File: [Y,T,SCALE] = auplot (X)
 -- Function File: [Y,T,SCALE] = auplot (X,FS)
 -- Function File: [Y,T,SCALE] = auplot (X,FS,OFFSET)
 -- Function File: [Y,T,SCALE] = auplot (...,PLOTSTR)

     Plot the waveform data, displaying time on the X axis.  If you are
     plotting a slice from the middle of an array, you may want to
     specify the OFFSET into the array to retain the appropriate time
     index.  If the waveform contains multiple channels, then the data
     are scaled to the range [-1,1] and shifted so that they do not
     overlap.  If a PLOTSTR is given, it is passed as the third argument
     to the plot command.  This allows you to set the linestyle easily.
     FS defaults to 8000 Hz, and OFFSET defaults to 0 samples.

     Instead of plotting directly, you can ask for the returned
     processed vectors.  If Y has multiple channels, the plot should
     have the y-range [-1 2*size(y,2)-1].  scale specifies how much the
     matrix was scaled so that each signal would fit in the specified
     range.

     Since speech samples can be very long, we need a way to plot them
     rapidly.  For long signals, auplot windows the data and keeps the
     minimum and maximum values in the window.  Together, these values
     define the minimal polygon which contains the signal.  The number
     of points in the polygon is set with the global variable
     auplot_points.  The polygon may be either 'filled' or 'outline', as
     set by the global variable auplot_format.  For moderately long
     data, the window does not contain enough points to draw an
     interesting polygon.  In this case, simply choosing an arbitrary
     point from the window looks best.  The global variable
     auplot_window sets the size of the window required for creating
     polygons.  You can turn off the polygons entirely by setting
     auplot_format to 'sampled'.  To turn off fast plotting entirely,
     set auplot_format to 'direct', or set auplot_points=1.  There is no
     reason to do this since your screen resolution is limited and
     increasing the number of points plotted will not add any
     information.  auplot_format, auplot_points and auplot_window may be
     set in .octaverc.  By default auplot_format is 'outline',
     auplot_points=1000 and auplot_window=7.




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Plot the waveform data, displaying time on the X axis.



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ausave


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 usage: ausave('filename.ext', x, fs, format)

 Writes an audio file with the appropriate header. The extension on
 the filename determines the layout of the header. Currently supports
 .wav and .au layouts.  Data is a matrix of audio samples in the
 range [-1,1] (inclusive), one row per time step, one column per 
 channel. Fs defaults to 8000 Hz.  Format is one of ulaw, alaw, char, 
 short, long, float, double

 Note that translating the symmetric range [-1,1] into the asymmetric
 range [-2^n,2^n-1] requires a DC offset of -2/2^n.  The inverse 
 process used by auload requires a DC offset of 2/2^n, so loading and 
 saving a file will not change the contents.  Other applications may 
 compensate for the asymmetry in a different way (including previous 
 versions of auload/ausave) so you may find small differences in 
 calculated DC offsets for the same file.



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 usage: ausave('filename.ext', x, fs, format)



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clip


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 Clip values outside the range to the value at the boundary of the
 range.

 X = clip(X)
   Clip to range [0, 1]

 X = clip(X, hi)
   Clip to range [0, hi]

 X = clip(X, [lo, hi])
   Clip to range [lo, hi]



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 Clip values outside the range to the value at the boundary of the
 range.



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sound


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 usage: sound(x [, fs, bs])

 Play the signal through the speakers.  Data is a matrix with
 one column per channel.  Rate fs defaults to 8000 Hz.  The signal
 is clipped to [-1, 1].  Buffer size bs controls how many audio samples 
 are clipped and buffered before sending them to the audio player.  bs 
 defaults to fs, which is equivalent to 1 second of audio.  

 Note that if $DISPLAY != $HOSTNAME:n then a remote shell is opened
 to the host specified in $HOSTNAME to play the audio.  See manual
 pages for ssh, ssh-keygen, ssh-agent and ssh-add to learn how to 
 set it up.

 This function writes the audio data through a pipe to the program
 "play" from the sox distribution.  sox runs pretty much anywhere,
 but it only has audio drivers for OSS (primarily linux and freebsd)
 and SunOS.  In case your local machine is not one of these, write
 a shell script such as ~/bin/octaveplay, substituting AUDIO_UTILITY
 with whatever audio utility you happen to have on your system:
   #!/bin/sh
   cat > ~/.octave_play.au
   SYSTEM_AUDIO_UTILITY ~/.octave_play.au
   rm -f ~/.octave_play.au
 and set the global variable (e.g., in .octaverc)
   global sound_play_utility="~/bin/octaveplay";

 If your audio utility can accept an AU file via a pipe, then you
 can use it directly:
   global sound_play_utility="SYSTEM_AUDIO_UTILITY flags"
 where flags are whatever you need to tell it that it is receiving
 an AU file.

 With clever use of the command dd, you can chop out the header and
 dump the data directly to the audio device in big-endian format:
   global sound_play_utility="dd of=/dev/audio ibs=2 skip=12"
 or little-endian format:
   global sound_play_utility="dd of=/dev/dsp ibs=2 skip=12 conv=swab"
 but you lose the sampling rate in the process.  

 Finally, you could modify sound.m to produce data in a format that 
 you can dump directly to your audio device and use "cat >/dev/audio" 
 as your sound_play_utility.  Things you may want to do are resample
 so that the rate is appropriate for your machine and convert the data
 to mulaw and output as bytes.
 
 If you experience buffer underruns while playing audio data, the bs
 buffer size parameter can be increased to tradeoff interactivity
 for smoother playback.  If bs=Inf, then all the data is clipped and 
 buffered before sending it to the audio player pipe.  By default, 1 
 sec of audio is buffered.



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 usage: sound(x [, fs, bs])



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soundsc


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 usage: soundsc(x, fs, limit) or soundsc(x, fs, [ lo, hi ])

 soundsc(x)
    Scale the signal so that [min(x), max(x)] -> [-1, 1], then 
    play it through the speakers at 8000 Hz sampling rate.  The
    signal has one column per channel.  

 soundsc(x,fs)
    Scale the signal and play it at sampling rate fs.

 soundsc(x, fs, limit)
    Scale the signal so that [-|limit|, |limit|] -> [-1, 1], then
    play it at sampling rate fs.  If fs is empty, then the default
    8000 Hz sampling rate is used.

 soundsc(x, fs, [ lo, hi ])
    Scale the signal so that [lo, hi] -> [-1, 1], then play it
    at sampling rate fs.  If fs is empty, then the default 8000 Hz
    sampling rate is used.

 y=soundsc(...)
    return the scaled waveform rather than play it.

 See sound for more information.



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 usage: soundsc(x, fs, limit) or soundsc(x, fs, [ lo, hi ])