One of the most exciting developments were sequence-to-sequence
models has been the rise of very accurate speech recognition.
We're nearing the end of the course,
we want to take just a couple of videos to give you a sense of how
these sequence-to-sequence models are applied to audio data, such as the speech.
So, what is the speech recognition problem?
You're given an audio clip, x,
and your job is to automatically find a text transcript, y.
So, an audio clip,
if you plot it looks like this,
the horizontal axis here is time,
and what a microphone does is it really measures minuscule changes in air pressure,
and the way you're hearing my voice right now is that
your ear is detecting little changes in air pressure,
probably generated either by your speakers or by a headset.
And some audio clips like this plots with the air pressure against time.
And, if this audio clip is of me saying,
"the quick brown fox", then hopefully,
a speech recognition algorithm can input that audio clip and output that transcript.
And because even the human ear doesn't process raw wave forms,
but the human ear has physical structures that
measures the amounts of intensity of different frequencies,
there is, a common pre-processing step for audio data
is to run your raw audio clip and generate a spectrogram.
So, this is the plots where the horizontal axis is time,
and the vertical axis is frequencies,
and intensity of different colors shows the amount of energy.
So, how loud is the sound at different frequencies? At different times?
And so, these types of spectrograms,
or you might also hear people talk about false back outputs,
is often commonly applied
pre-processing step before audio is pass into in the running algorithm.
And the human ear does a computation pretty similar to this pre-processing step.
So, one of the most exciting trends in speech recognition is that,
once upon a time,
speech recognition systems used to be built using phonemes and this where,
I want to say hand-engineered basic units of cells.
So, the quick brown fox represented as phonemes.
I'm going to simplify a bit, let say,
"The" has a "de" and "e" sound and Quick,
has a "ku" and "wu", "ik", "k" sound,
and linguist used to write off these basic units of sound,
and try the Greek language down to these basic units of sound.
So, brown, this aren't
the official phonemes which are written with more complicated notation,
but linguists use to hypothesize that writing down audio in terms of
these basic units of sound called
phonemes would be the best way to do speech recognition.
But with end-to-end deep learning,
we're finding that phonemes representations are no longer necessary.
But instead, you can built systems that input an audio clip and directly
output a transcript without needing to use hand-engineered representations like these.
One of the things that made this possible was going to much larger data sets.
So, academic data sets on speech recognition might be as a 300 hours, and in academia,
3000 hour data sets of transcribed audio would be considered reasonable size,
so lot of research has been done,
a lot of research papers that are written on data sets there are several thousand voice.
But, the best commercial systems are now trains
on over 10,000 hours and sometimes over a 100,000 hours of audio.
And, it's really moving to a much larger audio data sets,
transcribe audio data sets were both x and y,
together with deep learning algorithm,
that has driven a lot of progress is speech recognition.
So, how do you build a speech recognition system?
In the last video,
we're talking about the attention model.
So, one thing you could do is actually do that,
where on the horizontal axis,
you take in different time frames of the audio input,
and then you have an attention model try to output the transcript like,
"the quick brown fox",
or what it was said.
One other method that seems to work well is to use the CTC cost for speech recognition.
CTC stands for Connection is Temporal Classification and is due to Alex Graves,
Santiago Fernandes, Faustino Gomez, and Jürgen Schmidhuber.
So, here's the idea. Let's say the audio clip was someone saying,
"the quick brown fox".
We're going to use a new network structured
like this with an equal number of input x's and output y's,
and I have drawn a simple of what uni-directional for the RNN for this, but in practice,
this will usually be a bidirectional LSP and
bidirectional GIU and usually, a deeper model.
But notice that the number of time steps here is very large and in speech recognition,
usually the number of input time steps is much
bigger than the number of output time steps.
So, for example, if you have 10 seconds of audio and your features
come at a 100 hertz so 100 samples per second,
then a 10 second audio clip would end up with a thousand inputs.
Right, so it's 100 hertz times 10 seconds,
and so with a thousand inputs.
But your output might not have a thousand alphabets,
might not have a thousand characters.
So, what do you do?
The CTC cost function allows the RNN to generate an output like this ttt,
there's a special character called
the blank character, which we're going to write as an underscore here,
h_eee___, and then maybe a space,
we're going to write like this, so that a space and then ___ qqq__.
And, this is considered a correct output for the first parts of the space,
quick with the Q,
and the basic rule for the CTC cost function is to collapse
repeated characters not separated by "blank".
So, to be clear,
I'm using this underscore to denote
a special blank character and that's different than the space character.
So, there is a space here between the and quick,
so I should output a space.
But, by collapsing repeated characters,
not separated by blank,
it actually collapse the sequence into t, h,
e, and then space, and q,
and this allows your network
to have a thousand outputs by repeating characters allow the times.
So, inserting a bunch of blank characters and still ends
up with a much shorter output text transcript.
So, this phrase here "the quick brown fox" including spaces
actually has 19 characters, and if somehow,
the newer network is forced upwards of
a thousand characters by allowing the network to insert blanks and
repeated characters and can still represent
this 19 character upwards with this 1000 outputs of values of Y.
So, this paper by Alex Grace,
as well as by those deep speech recognition system,
which I was involved in,
used this idea to build effective Speech recognition systems.
So, I hope that gives you a rough sense of how speech recognition models work.
Attention like models work and CTC models work and
present two different options of how to go about building these systems.
Now, today, building effective where
production skills speech recognition system is
a pretty significant effort and requires a very large data set.
But, what I like to do in the next video is share you,
how you can build a trigger word detection system,
where keyword detection system which is actually much easier and can
be done with even a smaller or more reasonable amount of data.
So, let's talk about that in the next video.