I'm a beginner to Deep Learning, and I'm using a deep CNN structure to generate masks. I hope every channel of the output contains exactly one mask, but I found some channels may contain 0 or 2 masks, so what should I do to solve the problem.
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I'm trying to build a keras model to classify text for 45 different classes. I'm a little confused about preparing my data for the input as required by google's BERT model.
Some blog posts insert data as a tf dataset with input_ids, segment ids, and mask ids, as in this guide, but then some only go with input_ids and masks, as in this guide.
Also in the second guide, it notes that the segment mask and attention mask inputs are optional.
Can anyone explain whether or not those two are required for a multiclass classification task?
If it helps, each row of my data can consist of any number of sentences within a reasonably sized paragraph. I want to be able to classify each paragraph/input to a single label.
I can't seem to find many guides/blogs about using BERT with Keras (Tensorflow 2) for a multiclass problem, indeed many of them are for multi-label problems.
I guess it is too late to answer but I had the same question. I went through huggingface code and found that if attention_mask and segment_type ids are None then by default it pays attention to all tokens and all the segments are given id 0.
If you want to check it out, you can find the code here
Let me know if this clarifies it or you think otherwise.
In this article, I've come across the following network structure:
Figure 1(b). https://wx4.sinaimg.cn/mw690/5396ee05ly1fg9vi5phcbj20vj0kb0ty.jpg
Each layer is a fully connected one.
The weights shared by the two parts are denoted by Wc.
The pairs of the top fully connected layers of dimension 500 are concatenated to create a layer of dimension 1000 which is then used directly to reconstruct the input of size 784.
I want to implement it with keras, however I am not skilled with keras.
any ideas on how to implement this?
thank you very much!
I'd like to create an audio classification system with Keras that simply determines whether a given sample contains human voice or not. Nothing else. This would be my first machine learning attempt.
This audio preprocessor exists. It claims not to be done, but it's been forked a few times:
https://github.com/drscotthawley/audio-classifier-keras-cnn
I don't understand how this one would work, but I'm ready to give it a try:
https://github.com/keunwoochoi/kapre
But let's say I got one of those to work, would the rest of the process be similar to image classification? Basically, I've never fully understood when to use Softmax and when to use ReLu. Would this be similar with sound as it would with images once I've got the data mapped as a tensor?
Sounds can be seen as a 1D image and be worked with with 1D convolutions.
Often, dilated convolutions may do a good work, see Wave Nets
Sounds can also be seen as sequences and be worked with RNN layers (but maybe they're too bulky in amount of data for that)
For your case, you need only one output with a 'sigmoid' activation at the end and a 'binary_crossentropy' loss.
Result = 0 -> no voice
Result = 1 -> there's voice
When to use 'softmax'?
The softmax function is good for multiclass problems (not your case) where you want only one class as a result. All the results of a softmax function will sum 1. It's intended to be like a probability of each class.
It's mainly used at the final layer, because you only get classes as the final result.
It's good for cases when only one class is correct. And in this case, it goes well with the loss categorical_crossentropy.
Relu and other activations in the middle of the model
These are not very ruled. There are lots of possibilities. I often see relu in image convolutional models.
Important things to know are they "ranges". What are the limits of their outputs?
Sigmoid: from 0 to 1 -- at the end of the model this will be the best option for your presence/abscence classification. Also good for models that want many possible classes together.
Tanh: from -1 to 1
Relu: from 0 to limitless (it simply cuts negative values)
Softmax: from 0 to 1, but making sure the sum of all values is 1. Good at the end of models that want only 1 class among many classes.
Oftentimes it is useful to preprocess the audio to a spectrogram:
Using this as input, you can use classical image classification approaches (like convolutional neural networks). In your case you could divide the input audio in frames of around 20ms-100ms (depending on the time resolution you need) and convert those frames to spectograms. Convolutional networks can also be combined with recurrent units to take a larger time context into account.
It is also possible to train neural networks on raw waveforms using 1D Convolutions. However research has shown that preprocessing approaches using a frequency transformation achieve better results in general.
I am a newby to the convolutional neural nets... so this may be an ignorant question.
I have followed many examples and tutorials now on the MNIST example in TensforFlow. In the CNN examples, all authors talk bout using the 'input filters' to run in the CNN. But no one that I can find mentions WHERE they come from. Can anyone answer where these come from? Or are they magically obtained from the input images.
Thanks! Chris
This is an image that one professor uses, be he does not exaplain if he made them or TensorFlow auto-extracts these somehow.
Disclaimer: I am not an expert, more of an enthusiast.
To cut a long story short: filters are the CNN equivalent of weights, and all a neural network essentially does is learning their optimal values.
Which it does by iterating through a training dataset, making predictions, comparing them to the label/value already assigned to each training unit (usually an image in case of a CNN) and adjusting weights to minimize the error function (the difference between the predicted value and the actual value).
Initial values of filters/weights do not matter that much, so although they might affect the speed of convergence to a small degree, I believe they are often assigned random values.
It is the job of the neural network to figure out the optimal weights, not of the person implementing it.
I want to extract Audio Features using RBM (Restricted Boltzmann Machine). For this, I am giving the spectrogram (PCA whitened) as an input to the RBM.
For each audio file, The spectrogram is a matrix with no. of columns fixed but with different number of rows for each audio file. My question how can I train my RBM, or how can I extract the features from audio using RBM, given this spectrogram matrix. I read in a paper by Honglak Lee, paper title Unsupervised Feature Learning for Audio Classification using convolutional deep belief networks. http://machinelearning.wustl.edu/mlpapers/paper_files/NIPS2009_1171.pdf
"We then trained 300 first layer bases with a filter length of 6 and a max-pooling ratio of 3."
First, what is meant by bases here. (They have used Convolutional Deep Belief Networks, so I guess, bases do not mean weights here).
Second, what do they mean by using a filter length of 6? How can I do it? Any hint will be appreciated. (I am new to RBM)
I think what is confusing here is they add a convolutional layer to their deep belief network. The idea of the convolutional layer is they use kernels that are specific to a small region of the image, in their case a 6 element window. I'm not an expert in audio problems, but I believe bases refer to the different bands in the spectrograph.