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About half a yr in the past, this weblog featured a submit, written by Daniel Falbel, on the way to use Keras to categorise items of spoken language. The article received quite a lot of consideration and never surprisingly, questions arose the way to apply that code to totally different datasets. We’ll take this as a motivation to discover in additional depth the preprocessing carried out in that submit: If we all know why the enter to the community seems the best way it seems, we will modify the mannequin specification appropriately if want be.
In case you’ve a background in speech recognition, and even common sign processing, for you the introductory a part of this submit will most likely not comprise a lot information. Nonetheless, you would possibly nonetheless have an interest within the code half, which reveals the way to do issues like creating spectrograms with present variations of TensorFlow.
In case you don’t have that background, we’re inviting you on a (hopefully) fascinating journey, barely bearing on one of many better mysteries of this universe.
We’ll use the identical dataset as Daniel did in his submit, that’s, model 1 of the Google speech instructions dataset(Warden 2018)
The dataset consists of ~ 65,000 WAV information, of size one second or much less. Every file is a recording of considered one of thirty phrases, uttered by totally different audio system.
The purpose then is to coach a community to discriminate between spoken phrases. How ought to the enter to the community look? The WAV information comprise amplitudes of sound waves over time. Listed below are just a few examples, equivalent to the phrases hen, down, sheila, and visible:
A sound wave is a sign extending in time, analogously to how what enters our visible system extends in area.
At every time limit, the present sign depends on its previous. The plain structure to make use of in modeling it thus appears to be a recurrent neural community.
Nonetheless, the knowledge contained within the sound wave could be represented in another manner: specifically, utilizing the frequencies that make up the sign.
Right here we see a sound wave (high) and its frequency illustration (backside).
Within the time illustration (known as the time area), the sign consists of consecutive amplitudes over time. Within the frequency area, it’s represented as magnitudes of various frequencies. It might seem as one of many best mysteries on this world that you would be able to convert between these two with out lack of data, that’s: Each representations are primarily equal!
Conversion from the time area to the frequency area is completed utilizing the Fourier remodel; to transform again, the Inverse Fourier Rework is used. There exist several types of Fourier transforms relying on whether or not time is seen as steady or discrete, and whether or not the sign itself is steady or discrete. Within the “actual world,” the place normally for us, actual means digital as we’re working with digitized alerts, the time area in addition to the sign are represented as discrete and so, the Discrete Fourier Rework (DFT) is used. The DFT itself is computed utilizing the FFT (Quick Fourier Rework) algorithm, leading to important speedup over a naive implementation.
Trying again on the above instance sound wave, it’s a compound of 4 sine waves, of frequencies 8Hz, 16Hz, 32Hz, and 64Hz, whose amplitudes are added and displayed over time. The compound wave right here is assumed to increase infinitely in time. Not like speech, which modifications over time, it may be characterised by a single enumeration of the magnitudes of the frequencies it’s composed of. So right here the spectrogram, the characterization of a sign by magnitudes of constituent frequencies various over time, seems primarily onedimensional.
Nonetheless, after we ask Praat to create a spectrogram of considered one of our instance sounds (a seven), it might appear like this:
Right here we see a twodimensional picture of frequency magnitudes over time (larger magnitudes indicated by darker coloring). This twodimensional illustration could also be fed to a community, instead of the onedimensional amplitudes. Accordingly, if we determine to take action we’ll use a convnet as a substitute of an RNN.
Spectrograms will look totally different relying on how we create them. We’ll check out the important choices in a minute. First although, let’s see what we can’t all the time do: ask for all frequencies that have been contained within the analog sign.
Above, we stated that each representations, time area and frequency area, have been primarily equal. In our digital actual world, that is solely true if the sign we’re working with has been digitized appropriately, or as that is generally phrased, if it has been “correctly sampled.”
Take speech for instance: As an analog sign, speech per se is steady in time; for us to have the ability to work with it on a pc, it must be transformed to occur in discrete time. This conversion of the unbiased variable (time in our case, area in e.g. picture processing) from steady to discrete is named sampling.
On this technique of discretization, an important choice to be made is the sampling fee to make use of. The sampling fee must be a minimum of double the best frequency within the sign. If it’s not, lack of data will happen. The best way that is most frequently put is the opposite manner spherical: To protect all data, the analog sign could not comprise frequencies above onehalf the sampling fee. This frequency – half the sampling fee – is named the Nyquist fee.
If the sampling fee is simply too low, aliasing takes place: Increased frequencies alias themselves as decrease frequencies. Which means that not solely can’t we get them, in addition they corrupt the magnitudes of corresponding decrease frequencies they’re being added to.
Right here’s a schematic instance of how a highfrequency sign might alias itself as being lowerfrequency. Think about the highfrequency wave being sampled at integer factors (gray circles) solely:
Within the case of the speech instructions dataset, all sound waves have been sampled at 16 kHz. Which means that after we ask Praat for a spectogram, we should always not ask for frequencies larger than 8kHz. Here’s what occurs if we ask for frequencies as much as 16kHz as a substitute – we simply don’t get them:
Now let’s see what choices we do have when creating spectrograms.
Within the above easy sine wave instance, the sign stayed fixed over time. Nonetheless in speech utterances, the magnitudes of constituent frequencies change over time. Ideally thus, we’d have an actual frequency illustration for each time limit. As an approximation to this superb, the sign is split into overlapping home windows, and the Fourier remodel is computed for every time slice individually. That is referred to as the Brief Time Fourier Rework (STFT).
Once we compute the spectrogram by way of the STFT, we have to inform it what dimension home windows to make use of, and the way large to make the overlap. The longer the home windows we use, the higher the decision we get within the frequency area. Nonetheless, what we achieve in decision there, we lose within the time area, as we’ll have fewer home windows representing the sign. It is a common precept in sign processing: Decision within the time and frequency domains are inversely associated.
To make this extra concrete, let’s once more take a look at a easy instance. Right here is the spectrogram of an artificial sine wave, composed of two parts at 1000 Hz and 1200 Hz. The window size was left at its (Praat) default, 5 milliseconds:
We see that with a brief window like that, the 2 totally different frequencies are mangled into one within the spectrogram.
Now enlarge the window to 30 milliseconds, and they’re clearly differentiated:
The above spectrogram of the phrase “seven” was produced utilizing Praats default of 5 milliseconds. What occurs if we use 30 milliseconds as a substitute?
We get higher frequency decision, however on the value of decrease decision within the time area. The window size used throughout preprocessing is a parameter we would need to experiment with later, when coaching a community.
One other enter to the STFT to play with is the kind of window used to weight the samples in a time slice. Right here once more are three spectrograms of the above recording of seven, utilizing, respectively, a Hamming, a Hann, and a Gaussian window:
Whereas the spectrograms utilizing the Hann and Gaussian home windows don’t look a lot totally different, the Hamming window appears to have launched some artifacts.
Preprocessing choices don’t finish with the spectrogram. A wellliked transformation utilized to the spectrogram is conversion to mel scale, a scale primarily based on how people truly understand variations in pitch. We don’t elaborate additional on this right here, however we do briefly touch upon the respective TensorFlow code beneath, in case you’d wish to experiment with this.
Up to now, coefficients reworked to Mel scale have generally been additional processed to acquire the socalled MelFrequency Cepstral Coefficients (MFCCs). Once more, we simply present the code. For glorious studying on Mel scale conversion and MFCCs (together with the rationale why MFCCs are much less usually used these days) see this submit by Haytham Fayek.
Again to our unique activity of speech classification. Now that we’ve gained a little bit of perception in what’s concerned, let’s see the way to carry out these transformations in TensorFlow.
Code will likely be represented in snippets in line with the performance it supplies, so we could straight map it to what was defined conceptually above.
An entire instance is accessible right here. The whole instance builds on Daniel’s unique code as a lot as doable, with two exceptions:

The code runs in keen in addition to in static graph mode. In case you determine you solely ever want keen mode, there are just a few locations that may be simplified. That is partly associated to the truth that in keen mode, TensorFlow operations instead of tensors return values, which we are able to straight go on to TensorFlow capabilities anticipating values, not tensors. As well as, much less conversion code is required when manipulating intermediate values in R.

With TensorFlow 1.13 being launched any day, and preparations for TF 2.0 operating at full pace, we wish the code to necessitate as few modifications as doable to run on the subsequent main model of TF. One large distinction is that there’ll not be a
contrib
module. Within the unique submit,contrib
was used to learn within the.wav
information in addition to compute the spectrograms. Right here, we’ll use performance fromtf.audio
andtf.sign
as a substitute.
All operations proven beneath will run inside tf.dataset
code, which on the R facet is achieved utilizing the tfdatasets
package deal.
To clarify the person operations, we take a look at a single file, however later we’ll additionally show the information generator as a complete.
For stepping via particular person traces, it’s all the time useful to have keen mode enabled, independently of whether or not finally we’ll execute in keen or graph mode:
We choose a random .wav
file and decode it utilizing tf$audio$decode_wav
.This can give us entry to 2 tensors: the samples themselves, and the sampling fee.
fname < "knowledge/speech_commands_v0.01/hen/00b01445_nohash_0.wav"
wav < tf$audio$decode_wav(tf$read_file(fname))
wav$sample_rate
accommodates the sampling fee. As anticipated, it’s 16000, or 16kHz:
sampling_rate < wav$sample_rate %>% as.numeric()
sampling_rate
16000
The samples themselves are accessible as wav$audio
, however their form is (16000, 1), so we’ve got to transpose the tensor to get the same old (batch_size, variety of samples) format we’d like for additional processing.
samples < wav$audio
samples < samples %>% tf$transpose(perm = c(1L, 0L))
samples
tf.Tensor(
[[0.00750732 0.04653931 0.02041626 ... 0.01004028 0.01300049
0.00250244]], form=(1, 16000), dtype=float32)
Computing the spectogram
To compute the spectrogram, we use tf$sign$stft
(the place stft stands for Brief Time Fourier Rework). stft
expects three nondefault arguments: Moreover the enter sign itself, there are the window dimension, frame_length
, and the stride to make use of when figuring out the overlapping home windows, frame_step
. Each are expressed in items of variety of samples
. So if we determine on a window size of 30 milliseconds and a stride of 10 milliseconds …
window_size_ms < 30
window_stride_ms < 10
… we arrive on the following name:
samples_per_window < sampling_rate * window_size_ms/1000
stride_samples < sampling_rate * window_stride_ms/1000
stft_out < tf$sign$stft(
samples,
frame_length = as.integer(samples_per_window),
frame_step = as.integer(stride_samples)
)
Inspecting the tensor we received again, stft_out
, we see, for our single enter wave, a matrix of 98 x 257 complicated values:
tf.Tensor(
[[[ 1.03279948e04+0.00000000e+00j 1.95371482e046.41121820e04j
1.60833192e03+4.97534114e04j ... 3.61620914e051.07343149e04j
2.82576875e055.88812982e05j 2.66879797e05+0.00000000e+00j]
...
]],
form=(1, 98, 257), dtype=complex64)
Right here 98 is the variety of intervals, which we are able to compute upfront, primarily based on the variety of samples in a window and the dimensions of the stride:
257 is the variety of frequencies we obtained magnitudes for. By default, stft
will apply a Quick Fourier Rework of dimension smallest energy of two better or equal to the variety of samples in a window, after which return the fft_length / 2 + 1 distinctive parts of the FFT: the zerofrequency time period and the positivefrequency phrases.
In our case, the variety of samples in a window is 480. The closest enclosing energy of two being 512, we find yourself with 512/2 + 1 = 257 coefficients.
This too we are able to compute upfront:
Again to the output of the STFT. Taking the elementwise magnitude of the complicated values, we receive an power spectrogram:
magnitude_spectrograms < tf$abs(stft_out)
If we cease preprocessing right here, we’ll normally need to log remodel the values to raised match the sensitivity of the human auditory system:
log_magnitude_spectrograms = tf$log(magnitude_spectrograms + 1e6)
Mel spectrograms and MelFrequency Cepstral Coefficients (MFCCs)
If as a substitute we select to make use of Mel spectrograms, we are able to receive a metamorphosis matrix that may convert the unique spectrograms to Mel scale:
lower_edge_hertz < 0
upper_edge_hertz < 2595 * log10(1 + (sampling_rate/2)/700)
num_mel_bins < 64L
num_spectrogram_bins < magnitude_spectrograms$form[1]$worth
linear_to_mel_weight_matrix < tf$sign$linear_to_mel_weight_matrix(
num_mel_bins,
num_spectrogram_bins,
sampling_rate,
lower_edge_hertz,
upper_edge_hertz
)
Making use of that matrix, we receive a tensor of dimension (batch_size, variety of intervals, variety of Mel coefficients) which once more, we are able to logcompress if we wish:
mel_spectrograms < tf$tensordot(magnitude_spectrograms, linear_to_mel_weight_matrix, 1L)
log_mel_spectrograms < tf$log(mel_spectrograms + 1e6)
Only for completeness’ sake, lastly we present the TensorFlow code used to additional compute MFCCs. We don’t embody this within the full instance as with MFCCs, we would want a special community structure.
num_mfccs < 13
mfccs < tf$sign$mfccs_from_log_mel_spectrograms(log_mel_spectrograms)[, , 1:num_mfccs]
Accommodating differentlength inputs
In our full instance, we decide the sampling fee from the primary file learn, thus assuming all recordings have been sampled on the identical fee. We do enable for various lengths although. For instance in our dataset, had we used this file, simply 0.65 seconds lengthy, for demonstration functions:
fname < "knowledge/speech_commands_v0.01/hen/1746d7b6_nohash_0.wav"
we’d have ended up with simply 63 intervals within the spectrogram. As we’ve got to outline a hard and fast input_size
for the primary conv layer, we have to pad the corresponding dimension to the utmost doable size, which is n_periods
computed above.
The padding truly takes place as a part of dataset definition. Let’s shortly see dataset definition as a complete, leaving out the doable technology of Mel spectrograms.
data_generator < operate(df,
window_size_ms,
window_stride_ms) {
# assume sampling fee is similar in all samples
sampling_rate <
tf$audio$decode_wav(tf$read_file(tf$reshape(df$fname[[1]], checklist()))) %>% .$sample_rate
samples_per_window < (sampling_rate * window_size_ms) %/% 1000L
stride_samples < (sampling_rate * window_stride_ms) %/% 1000L
n_periods <
tf$form(
tf$vary(
samples_per_window %/% 2L,
16000L  samples_per_window %/% 2L,
stride_samples
)
)[1] + 1L
n_fft_coefs <
(2 ^ tf$ceil(tf$log(
tf$forged(samples_per_window, tf$float32)
) / tf$log(2)) /
2 + 1L) %>% tf$forged(tf$int32)
ds < tensor_slices_dataset(df) %>%
dataset_shuffle(buffer_size = buffer_size)
ds < ds %>%
dataset_map(operate(obs) {
wav <
tf$audio$decode_wav(tf$read_file(tf$reshape(obs$fname, checklist())))
samples < wav$audio
samples < samples %>% tf$transpose(perm = c(1L, 0L))
stft_out < tf$sign$stft(samples,
frame_length = samples_per_window,
frame_step = stride_samples)
magnitude_spectrograms < tf$abs(stft_out)
log_magnitude_spectrograms < tf$log(magnitude_spectrograms + 1e6)
response < tf$one_hot(obs$class_id, 30L)
enter < tf$transpose(log_magnitude_spectrograms, perm = c(1L, 2L, 0L))
checklist(enter, response)
})
ds < ds %>%
dataset_repeat()
ds %>%
dataset_padded_batch(
batch_size = batch_size,
padded_shapes = checklist(tf$stack(checklist(
n_periods, n_fft_coefs,1L
)),
tf$fixed(1L, form = form(1L))),
drop_remainder = TRUE
)
}
The logic is similar as described above, solely the code has been generalized to work in keen in addition to graph mode. The padding is taken care of by dataset_padded_batch(), which must be advised the utmost variety of intervals and the utmost variety of coefficients.
Time for experimentation
Constructing on the full instance, now’s the time for experimentation: How do totally different window sizes have an effect on classification accuracy? Does transformation to the mel scale yield improved outcomes? You may also need to strive passing a nondefault window_fn
to stft
(the default being the Hann window) and see how that impacts the outcomes. And naturally, the easy definition of the community leaves quite a lot of room for enchancment.
Talking of the community: Now that we’ve gained extra perception into what’s contained in a spectrogram, we would begin asking, is a convnet actually an sufficient resolution right here? Usually we use convnets on photos: twodimensional knowledge the place each dimensions characterize the identical sort of data. Thus with photos, it’s pure to have sq. filter kernels.
In a spectrogram although, the time axis and the frequency axis characterize essentially several types of data, and it’s not clear in any respect that we should always deal with them equally. Additionally, whereas in photos, the interpretation invariance of convnets is a desired function, this isn’t the case for the frequency axis in a spectrogram.
Closing the circle, we uncover that as a consequence of deeper information concerning the topic area, we’re in a greater place to purpose about (hopefully) profitable community architectures. We go away it to the creativity of our readers to proceed the search…
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