--- /dev/null
+# Created by Octave 3.6.1, Tue Mar 20 07:03:19 2012 UTC <root@t61>
+# name: cache
+# type: cell
+# rows: 3
+# columns: 7
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 2
+au
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 315
+ 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.
+
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 26
+ y = au(x, fs, lo [, hi])
+
+
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 6
+auload
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 1008
+ -- 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.
+
+
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 65
+Reads an audio waveform from a file given by the string FILENAME.
+
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 6
+auplot
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 2310
+ -- 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.
+
+
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 54
+Plot the waveform data, displaying time on the X axis.
+
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 6
+ausave
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 871
+ 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.
+
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 25
+ usage: ausave('filename.
+
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 4
+clip
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 206
+ 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]
+
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 74
+ Clip values outside the range to the value at the boundary of the
+ range.
+
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 5
+sound
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 2377
+ 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.
+
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 28
+ usage: sound(x [, fs, bs])
+
+
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 7
+soundsc
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 793
+ 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.
+
+
+
+# name: <cell-element>
+# type: sq_string
+# elements: 1
+# length: 60
+ usage: soundsc(x, fs, limit) or soundsc(x, fs, [ lo, hi ])
+
+
+
+
+
+
git://git.creatis.insa-lyon.fr / CreaPhase.git / blobdiff