About half a yr in the past, this weblog featured a publish, written by Daniel Falbel, on find out how to use Keras to categorise items of spoken language. The article received a whole lot of consideration and never surprisingly, questions arose find out how to apply that code to completely different datasets. We’ll take this as a motivation to discover in additional depth the preprocessing finished in that publish: If we all know why the enter to the community appears to be like the way in which it appears to be like, we can modify the mannequin specification appropriately if want be.
In case you have got a background in speech recognition, and even basic sign processing, for you the introductory a part of this publish will most likely not include a lot information. Nevertheless, you would possibly nonetheless have an interest within the code half, which exhibits find out how 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 referring to one of many higher mysteries of this universe.
We’ll use the identical dataset as Daniel did in his publish, that’s, version 1 of the Google speech commands dataset(Warden 2018)
The dataset consists of ~ 65,000 WAV recordsdata, of size one second or much less. Every file is a recording of one among thirty phrases, uttered by completely different audio system.
The objective then is to coach a community to discriminate between spoken phrases. How ought to the enter to the community look? The WAV recordsdata include amplitudes of sound waves over time. Listed here are just a few examples, akin to the phrases fowl, 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 apparent structure to make use of in modeling it thus appears to be a recurrent neural community.
Nevertheless, the data contained within the sound wave could be represented in another approach: 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 could seem as one of many biggest mysteries on this world that you would be able to convert between these two with out lack of info, that’s: Each representations are primarily equal!
Conversion from the time area to the frequency area is finished utilizing the Fourier remodel; to transform again, the Inverse Fourier Remodel is used. There exist various kinds of Fourier transforms relying on whether or not time is considered as steady or discrete, and whether or not the sign itself is steady or discrete. Within the “actual world,” the place often for us, actual means digital as we’re working with digitized indicators, the time area in addition to the sign are represented as discrete and so, the Discrete Fourier Remodel (DFT) is used. The DFT itself is computed utilizing the FFT (Quick Fourier Remodel) algorithm, leading to vital speedup over a naive implementation.
Wanting 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, appears to be like primarily onedimensional.
Nevertheless, once we ask Praat to create a spectrogram of one among our instance sounds (a seven), it might seem 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 an alternative of an RNN.
Spectrograms will look completely 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 at all times 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 known as sampling.
On this strategy of discretization, a vital resolution to be made is the sampling charge to make use of. The sampling charge must be not less than double the best frequency within the sign. If it’s not, lack of info will happen. The best way that is most frequently put is the opposite approach spherical: To protect all info, the analog sign might not include frequencies above onehalf the sampling charge. This frequency – half the sampling charge – is known as the Nyquist charge.
If the sampling charge is simply too low, aliasing takes place: Greater frequencies alias themselves as decrease frequencies. Which means 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 once we ask Praat for a spectogram, we must always not ask for frequencies larger than 8kHz. Here’s what occurs if we ask for frequencies as much as 16kHz as an alternative – 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. Nevertheless 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 Remodel (STFT).
After we compute the spectrogram by way of the STFT, we have to inform it what measurement 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. Nevertheless, 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 basic 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 completely 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 an alternative?
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’d wish 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 completely 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 prefer to experiment with this.
Previously, coefficients reworked to Mel scale have typically 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 typically used these days) see this post 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 find out how to carry out these transformations in TensorFlow.
Code will probably be represented in snippets in keeping with the performance it offers, so we might instantly map it to what was defined conceptually above.
A whole instance is offered here. The entire instance builds on Daniel’s original code as a lot as attainable, 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 instantly go on to TensorFlow features 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 attainable to run on the following main model of TF. One large distinction is that there’ll not be a
contrib
module. Within the unique publish,contrib
was used to learn within the.wav
recordsdata in addition to compute the spectrograms. Right here, we’ll use performance fromtf.audio
andtf.sign
as an alternative.
All operations proven beneath will run inside tf.dataset
code, which on the R facet is completed utilizing the tfdatasets
bundle.
To clarify the person operations, we take a look at a single file, however later we’ll additionally show the info generator as an entire.
For stepping by means of particular person strains, it’s at all times useful to have keen mode enabled, independently of whether or not in the end we’ll execute in keen or graph mode:
We decide a random .wav
file and decode it utilizing tf$audio$decode_wav
.This may give us entry to 2 tensors: the samples themselves, and the sampling charge.
fname < "knowledge/speech_commands_v0.01/fowl/00b01445_nohash_0.wav"
wav < tf$audio$decode_wav(tf$read_file(fname))
wav$sample_rate
incorporates the sampling charge. 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 now we have to transpose the tensor to get the standard (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 Remodel). stft
expects three nondefault arguments: Moreover the enter sign itself, there are the window measurement, frame_length
, and the stride to make use of when figuring out the overlapping home windows, frame_step
. Each are expressed in models 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 prematurely, 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 Remodel of measurement smallest energy of two higher 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 prematurely:
Again to the output of the STFT. Taking the elementwise magnitude of the complicated values, we acquire an power spectrogram:
magnitude_spectrograms < tf$abs(stft_out)
If we cease preprocessing right here, we’ll often wish to log remodel the values to raised match the sensitivity of the human auditory system:
log_magnitude_spectrograms = tf$log(magnitude_spectrograms + 1e6)
Mel spectrograms and MelFrequency Cepstral Coefficients (MFCCs)
If as an alternative we select to make use of Mel spectrograms, we are able to acquire a change 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 acquire a tensor of measurement (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 embrace this within the full instance as with MFCCs, we would wish 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 charge from the primary file learn, thus assuming all recordings have been sampled on the identical charge. We do permit 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/fowl/1746d7b6_nohash_0.wav"
we’d have ended up with simply 63 intervals within the spectrogram. As now we have to outline a set input_size
for the primary conv layer, we have to pad the corresponding dimension to the utmost attainable 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 an entire, leaving out the attainable era of Mel spectrograms.
data_generator < perform(df,
window_size_ms,
window_stride_ms) {
# assume sampling charge is similar in all samples
sampling_rate <
tf$audio$decode_wav(tf$read_file(tf$reshape(df$fname[[1]], list()))) %>% .$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$solid(samples_per_window, tf$float32)
) / tf$log(2)) /
2 + 1L) %>% tf$solid(tf$int32)
ds < tensor_slices_dataset(df) %>%
dataset_shuffle(buffer_size = buffer_size)
ds < ds %>%
dataset_map(perform(obs) {
wav <
tf$audio$decode_wav(tf$read_file(tf$reshape(obs$fname, list())))
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$transpose(log_magnitude_spectrograms, perm = c(1L, 2L, 0L))
list(enter, response)
})
ds < ds %>%
dataset_repeat()
ds %>%
dataset_padded_batch(
batch_size = batch_size,
padded_shapes = list(tf$stack(list(
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 complete example, now’s the time for experimentation: How do completely different window sizes have an effect on classification accuracy? Does transformation to the mel scale yield improved outcomes? You may additionally wish to attempt 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 a whole lot of room for enchancment.
Talking of the community: Now that we’ve gained extra perception into what’s contained in a spectrogram, we’d begin asking, is a convnet actually an enough answer right here? Usually we use convnets on pictures: twodimensional knowledge the place each dimensions characterize the identical form of info. Thus with pictures, it’s pure to have sq. filter kernels.
In a spectrogram although, the time axis and the frequency axis characterize basically various kinds of info, and it isn’t clear in any respect that we must always deal with them equally. Additionally, whereas in pictures, 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 result of deeper information in regards to the topic area, we’re in a greater place to motive about (hopefully) profitable community architectures. We go away it to the creativity of our readers to proceed the search…