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"AudioSTFT" (Net Encoder)

[Experimental]

NetEncoder["AudioSTFT"]

represents an encoder that converts an audio file or object into its short-time Fourier transform.

NetEncoder[{"AudioSTFT","param"->val,…}]

represents an encoder with specific parameters for preprocessing.

Details

  • The "AudioSTFT" encoder partitions the signal, multiplies each partition with a window function and computes the Fourier transform on each of them. The result of a Fourier transform is a complex number, and for each of them the encoder returns a list of the real and imaginary parts. The original signal can be reconstructed from the STFT as there is no loss of information.
  • NetEncoder[…][input] applies the encoder to an input to produce a "Real32" output.
  • NetEncoder[…][{input1,input2,…}] applies the encoder to a list of inputs to produce a list of outputs.
  • When given a NumericArray as input, the output will be a NumericArray.
  • The input to the encoder can be an Audio object or a File[…] expression.
  • The output of the encoder is a rank-3 tensor of dimensions {n,ws,2}, where n is the number of partitions after the preprocessing is applied and ws is the length of the partitions used for the computation. The last dimension represents the real and imaginary parts of the result.
  • An encoder can be attached to an input port of a net by specifying "port"->NetEncoder[…] when constructing the net.
    • Parameters
  • The following general parameters are supported:
  • "Augmentation"Noneaugmentation to be applied
    "Normalization"Nonewhether to apply normalization
    "SampleRate"16000target sample rate
    "TargetLength"Alltarget output length
  • Additional partitioning parameters:
  • "WindowSize"Automaticlength of the partitions
    "Offset"Automaticoffset of the partitions
    "WindowFunction"Automaticwindow to be applied to the partitions
  • The following settings and suboptions can be specified for each encoder parameter.
  • "Normalization" can take the following settings:
  • Noneno normalization
    "Max"absolute maximum value normalized to 1
    {"Max",val}absolute maximum value normalized to val
    {"RMS",val}RMS of input audio signal normalized to val
  • "TargetLength" can take the following settings:
  • Allsame as input signal
    durthe duration dur specified as a time quantity
    nthe first n partitions
  • If the specified "TargetLength" does not match the length of the input signal, padding or trimming are applied accordingly.
  • "Augmentation" can be specified as a list of rules with the following keys:
  • "Convolution"Noneconvolves an impulse response to the input
    "Noise"Noneadds noise to the input
    "TimeShift"Noneshifts the input by a specified amount
    "Volume"Nonemultiplies the input with a constant
  • Any augmentation parameter that accepts a numeric value can also be specified as a list of two numbers or a univariate distribution. In the first case, the value will be randomized according to a uniform distribution between the given bounds. In the second, the user-provided distribution will be used.
  • Possible values for "Convolution" include:
  • Noneno augmentation
    signalFile or Audio object to be convolved with input
    {mix,signal}signal to be convolved with input and mix parameter
  • Possible values for "Noise" include:
  • Noneno augmentation
    ampwhite noise with amplitude amp
    noiseFile or Audio object containing the noise signal to be added
    {amp,noise}
  • noise signal and its with the specified amplitude
  • Use "TimeShift"->t to shift the input by t seconds, padding or trimming if necessary. Use Scaled[s] to shift the input by s×dur seconds, where dur is the duration of the input signal. Use {t1,t2} or Scaled[{ts1,t2}] to randomize the shift between the specified times.
  • Use "Volume"->val to specify a constant multiplier.
  • With the parameter "WindowSize"->Automatic, a partition length of 25 milliseconds is used. Use "WindowSize"->dur to select a partition length of duration dur. Use "WindowSize"->n to select a partition length of n samples.
  • With the parameter "Offset"->Automatic, a partition offset of 8.33 milliseconds is used. Use "Offset"->dur to select a partition offset of duration dur. Use "Offset"->n to select a partition offset of n samples.
  • Parameter "WindowFunction" applies a window to each partition. Possible settings are:
  • Noneno windowing applied to the input audio
    Automatic
    functhe window is computed using the function func
    listthe sampled window list is explicitly specified

Examples

open allclose all

Basic Examples  (2)Summary of the most common use cases

Create an audio STFT NetEncoder:

In[1]:=1
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-n8wn6b
Out[1]=1

Create an Audio object:

In[2]:=2
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-53m125
Out[2]=2

Apply the encoder to the Audio object:

In[3]:=3
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-6hg6sa
Out[3]=3

Plot the real part of the result of the encoder:

In[1]:=1
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-1x59x7
Out[1]=1

Scope  (3)Survey of the scope of standard use cases

NetEncoder["AudioSTFT"] can encode either File or Audio objects. Create an audio STFT encoder:

In[1]:=1
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-mmydiq
Out[1]=1

Apply the encoder to a File object:

In[2]:=2
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-59rlwe
Out[2]=2

Apply the encoder to an in-core Audio object:

In[1]:=1
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-nvusko
Out[1]=1
In[4]:=4
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-sryw8o
Out[4]=4

Apply the encoder to an out-of-core Audio object:

In[2]:=2
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-v257i1
Out[2]=2
In[6]:=6
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-fye1ir
Out[6]=6

Create a list of Audio objects:

In[1]:=1
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-y528t8

NetEncoder["AudioSTFT"] maps across a batch of inputs:

In[2]:=2
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-9cdqxr
Out[2]=2

Create an audio STFT NetEncoder:

In[1]:=1
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-y9m591
Out[1]=1

Attach the encoder to the input of a net:

In[2]:=2
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-504ex7
Out[2]=2

Apply the net to an Audio object:

In[3]:=3
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-486h0v
Out[3]=3

Parameters  (6)

"Normalization"  (1)

Create an Audio object:

In[1]:=1
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-pekrv3
Out[1]=1

Use an encoder with "Normalization"->None to avoid any normalization:

In[2]:=2
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-hve4de

Since the normalization is applied to the signal before the short-time Fourier transform is computed, there are no guarantees on the bounds of the result:

In[3]:=3
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-q9lym8
Out[3]=3

Use an encoder with "Normalization"->Automatic to normalize the maximum absolute value of the waveform samples to 1.:

In[4]:=4
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-8p5lvc

Find the minimum and maximum value of the result:

In[5]:=5
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-h8oyh3
Out[5]=5

"SampleRate"  (2)

Create an Audio object:

In[1]:=1
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-0jwi3b
Out[1]=1

Using an encoder with "SampleRate"8000 resamples the signal to 8000Hz before performing the short-time Fourier transform:

In[5]:=5
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-dms1vc

The "SampleRate" parameter affects the computation of the default window size:

In[1]:=1
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-gdgeu9
Out[1]=1

An encoder with a lower sample rate than the original audio will result in a shorter window length:

In[2]:=2
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-ndg4im
Out[2]=2

An encoder with a higher sample rate than the original audio will result in a longer window length:

In[3]:=3
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-ynw3p7
Out[3]=3

"TargetLength"  (1)

Create an Audio object:

In[1]:=1
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-tfs6vj
Out[1]=1

Using an encoder with "TargetLength"All returns the short-time Fourier transform for all the data:

In[5]:=5
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-fxb5xc
Out[5]=5

Using an encoder with "TargetLength"10 zero-pads the output to be of length 10:

In[6]:=6
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-ouchw6
Out[6]=6

Using an encoder with "TargetLength"2 takes only the first two partitions:

In[7]:=7
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-gob6v3
Out[7]=7

"WindowSize"  (1)

Create an Audio object:

In[1]:=1
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-5m9wdu
Out[1]=1

The partition length is automatically computed to be 25ms:

In[6]:=6
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-2qs700
Out[6]=6

Using an encoder with "WindowSize"600 returns the short-time Fourier transform using partitions of 600 samples:

In[7]:=7
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-v935th
Out[7]=7

"Offset"  (1)

Create an Audio object:

In[1]:=1
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-ij9wtt
Out[1]=1

The partition offset is automatically computed to be 1/3 of the partition length:

In[3]:=3
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-t966dh
Out[3]=3

Using an encoder with "Offset"10 returns the short-time Fourier transform computed using partitions with an offset of 10 samples:

In[4]:=4
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-iq8p96
Out[4]=4

Properties & Relations  (2)Properties of the function, and connections to other functions

Create an Audio object:

In[1]:=1
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-6l3wo2
Out[1]=1

Create an audio STFT NetEncoder:

In[2]:=2
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-gasgyo
Out[3]=3

The length of the result can be computed as Ceiling[length/offset], where length is the length of the signal after resampling and offset is the "Offset" parameter of the encoder:

In[4]:=4
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-v7sqlx
Out[4]=4

The equivalent computation for the "AudioSTFT" encoder is based on ShortTimeFourier:

In[1]:=1
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-fbb9px

Compare the results:

In[2]:=2
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-z3t6fe
Out[2]=2

Possible Issues  (1)Common pitfalls and unexpected behavior

If the input is a multichannel signal, the mean of the channels is used for the computation:

In[1]:=1
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-1w4gel

Check that the results are the same:

In[3]:=3
✖
https://wolfram.com/xid/0kxwo3lcghg8zdtwsj8vrzad3-1k2g0f
Out[3]=3