X-Git-Url: https://git.creatis.insa-lyon.fr/pubgit/?p=CreaPhase.git;a=blobdiff_plain;f=octave_packages%2Faudio-1.1.4%2Fdoc-cache;fp=octave_packages%2Faudio-1.1.4%2Fdoc-cache;h=98374702204506d4e6ddec0514824342fed16d43;hp=0000000000000000000000000000000000000000;hb=f5f7a74bd8a4900f0b797da6783be80e11a68d86;hpb=1705066eceaaea976f010f669ce8e972f3734b05 diff --git a/octave_packages/audio-1.1.4/doc-cache b/octave_packages/audio-1.1.4/doc-cache new file mode 100644 index 0000000..9837470 --- /dev/null +++ b/octave_packages/audio-1.1.4/doc-cache @@ -0,0 +1,331 @@ +# Created by Octave 3.6.1, Tue Mar 20 07:03:19 2012 UTC +# name: cache +# type: cell +# rows: 3 +# columns: 7 +# name: +# type: sq_string +# elements: 1 +# length: 2 +au + + +# name: +# 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: +# type: sq_string +# elements: 1 +# length: 26 + y = au(x, fs, lo [, hi]) + + + + +# name: +# type: sq_string +# elements: 1 +# length: 6 +auload + + +# name: +# 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: +# type: sq_string +# elements: 1 +# length: 65 +Reads an audio waveform from a file given by the string FILENAME. + + + +# name: +# type: sq_string +# elements: 1 +# length: 6 +auplot + + +# name: +# 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: +# type: sq_string +# elements: 1 +# length: 54 +Plot the waveform data, displaying time on the X axis. + + + +# name: +# type: sq_string +# elements: 1 +# length: 6 +ausave + + +# name: +# 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: +# type: sq_string +# elements: 1 +# length: 25 + usage: ausave('filename. + + + +# name: +# type: sq_string +# elements: 1 +# length: 4 +clip + + +# name: +# 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: +# type: sq_string +# elements: 1 +# length: 74 + Clip values outside the range to the value at the boundary of the + range. + + + +# name: +# type: sq_string +# elements: 1 +# length: 5 +sound + + +# name: +# 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: +# type: sq_string +# elements: 1 +# length: 28 + usage: sound(x [, fs, bs]) + + + + +# name: +# type: sq_string +# elements: 1 +# length: 7 +soundsc + + +# name: +# 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: +# type: sq_string +# elements: 1 +# length: 60 + usage: soundsc(x, fs, limit) or soundsc(x, fs, [ lo, hi ]) + + + + + +