Categories
Dragon goldman sach

Goldman Sachs not Liable

Here is the Bloomberg’s piece.   Sounds like it’s a real case closed.

Of course, we are all still feeling the consequences.

Arthur

Categories
Amazon Dragon Goldman Ivona Trial

Speech-related Readings at Jan 30, 2013

Amazon acquired Ivona:

I am aware of Amazon’s involvement in ASR.   Though it’s a question on the domain.

Goldman-Dragon Trial:

I simply hope Dr. Baker has a closure on the whole thing.   In fact, when you think about it,  the whole L&H fallout is the reason why the ASR industry has a virtual monopoly now.  So if you are interested in ASR, you should be concerned.

Arthur

Categories
biphone diphone hetaphone phone phonemes quadphone quinphone subword units triphone

Subword Units and their Occasionally Non-Trivial Meanings

While I was writing the next article on bw,  I found myself forget the meaning of different type of subword units (i.e phones, biphones, diphones, triphones and such).  So I decide to write a little note.

On this kind of topics, someone would likely to come up and say “X always mean Y bla bla bla etc and so on.”  My view (and hope) is that the wording of a certain should reflect what it means.  So when I hear a term and can come up with multiple definition in your head, I would say the naming convention is a little bit broken.

Phones vs Phonemes

Linguist distinguish between phoneme and phone The former usually means a kind of abstract categorization of a sound, whereas the latter usually mean the actual realization of a sound.

In a decoder though, what you see most is the term phone.

Biphones vs Diphones

(Ref here) ” ….. one important difference between the two units. Biphones are understood tobe just left or right context dependent (mono)phones. On the other hand, diphones represent the transition regions that strech between the two ”centres” of the subsequent phones. “
So that’s why there can be left-biphone and right-biphone.  Diphones is intuitively better in synthesis.
Possible combination of left-biphones/right-biphones/diphones are all N^2.  With N equals to the number of phones. 
Btw, the link I gave also has a term called “bi-diphone”, which I don’t think it’s a standard term. 

Triphones

For most of the time, it means considering both left and right context.  Possible combinations N^3. 

Quinphones

For most of the time, it means considering both the two left and two right contexts. Possible combinations N^5. 

Heptaphones


For most of the time, it means considering both three left and three right  contexts. Possible combinations N^7. 

“Quadphones” and Other possible confusions in terminology. 

I guess what I don’t feel comfortable are terms such as “Quadphones”.   Even quinphones and heptaphones can potentially means different things from time-to-time.  
For example, if you look at LID literature, occasionally, you will see the term quadphone.  But it seems the term “phone 4-gram” (or more correctly quadgram…… if you think too much,) might be a nicer choice.  
Then there is how the context looks like:  2 left 1 right? 1 right 2 left?   Come to think of it, this terminology is confusing for even triphones because we can also mean a phone depend on 2 left or 2 right phones.  ASR people don’t feel that ways probably because of a well-established convention.  Of course, the same can be said for quinphone and hetaphones. 
Arthur
Categories
Bayesian method C++ C11 iOS tools

Readings at Jan 28, 2013

Tools of the Trade : Mainly an iOS article but it has many tools on maintaining contacts, task lists and requests.
C11 : I have no idea C99 tried to implement variable length array.  It’s certainly not very successful in the past 10 years…..   Another great book to check out is Ning’s C book.
How to make iPhone App that actually sells : Again, probably not just for iOS but generally for writing free/shareware.
Bayesian vs Non-Bayesian:  Nice post.  I don’t fully grok Bayesian/Non-Bayesian but if you know better, they are essentially two schools of thoughts. (ASR? The whole training process starts from a flat-start, you figure.)

Categories
ASR Kurzweil Speech Recognition

On Kurzweil : a perspective of an ASR practitioner

Many people who don’t work on the fields of AI, NLP and ASR have heard of Kurzweil.   To my surprise, many seem to give little thought on what he said and just follow his theory wholeheartedly.

In this post, I just want to comment on one single little thing, which is whether real-time speech-to-speech translation can be achieved in 2010s.  This is a very specific prediction from Kurzweil’s book “The Singularity is Near“.

My discussion would mainly focus on ASR first.  So even though my examples below are not exactly pure ASR systems, I will skip the long winding wording of saying “ASR of System X”.  And to be frank, MT and Response system probably goes through similar torturous development process anyway.   So, please, don’t tell me that “System X is actually ASR + Y”, that sort of besides the point.

Oh well, you probably ask why bother, don’t we have a demo of real-time speech-to-speech translation from Microsoft already?

True, but if you observe the demo carefully, it is based on read speech.  I don’t want to speculate much but I doubt it is a generic language model which wasn’t tuned to the lecture.   In a nutshell, I disbelieve it is something you can use it in real-life.

Let’s think of a more real-life example: Siri, are we really getting 100% correct response now?  (Not to boil down to ASR WER …… yet)  I don’t think so. Even with adaptation, I don’t think Siri understand what I said every single time.    For most of the time, I follow the unofficial command list of Siri, let it improve with adaptation….. but still, it is not perfect.

Why? It is the hard cold reality: ASR is still not perfect, with all the advancement in HMM-based speech recognition.  All the key technologies we know in the last 20 years: CMLLR, MMIE, MPE, fMPE, DMT, consensus network, multipass decodings, SAT, SAT with MMIE or all the nicest front-ends, all the engineerings.   Nope, we are not yet having a feel-good accuracy.  Indeed, human speech recognition is not 0% WER neither but for some reasons, the current state-of-the-art ASR performance is not reaching there.

And Siri, we all know is the state-of-the-art.

Just digress a little bit: Now most of the critics when they write to this point, will then lament that “oh, there is just some invisible barrier out there and human just couldn’t make a Tower Babel, blabla….”.  I believe most of these “critics” have absolutely no ideas what they are talking about.   To identify these air-head critics, just try to see if they put “cognitive science” into the articles, then you don’t know they never work on real-life ASR system.

I, on the hand, do believe one day we can get there.  Why?  Because when people work on one of these speech recognition evaluation tasks, many would tell you : given a certain test set and with enough time and gumption, you would be able to create a system without any errors.  So to me, it is more of an issue of whether some guys grinding on the problem, but not feasibility issue.

So where are we now in ASR?  Recently, In ICASSP 2012,  a Google paper, trained 87 thousand hour of data.  That is probably the largest scale of training I know.  Oh well, where are we now? 10%.  Go down from 12%.  So the last big experiment I know, it’s probably the 3000 hours experiment back in 2006-7.  The Google authors are probably using a tougher test set.  So the initial recognition rate was yet again lower.

Okay, speculation time.  So let’s assume, that human can always collect 10 times more labelled data for every 6-7 years AND we can do an AM training on them. When will we go to have say 2% WER on the current Google test set?   If we just think of very simple linear interpolation.  It will take 4 * 6 years = 24 years to collect 10000 times more data (or 8 billion hour of data).    So we are way-way past the 2010s deadline from Kurzweil.

And that’s a wild speculation.   Computation resources probably will work out itself by that time.  What I doubt most is whether the progress would be linear.  

Of course, it might be non-linearly better too.  But here is another point: it’s not just about the training set, it’s about the test set.  If we truly want a recognizer to work for *everyone* in the planet, then the very right thing to do is test your recognizer on our whole population.  If we can’t then you want to sample enough human speech to represent the Earth’s population, the current test set might not be representative enough.   So it is possible that when we increase our test set, we found that the initial recognition rate has go down again.   And it seems to me our test set is still in the state of mimicking human population.

My discussion so far are mostly on acoustic model.  On the language model side,  the problem will mainly on domain specificity.   Also bear in mind, human language can evolve.  So, say we want to build a system which build a customized language model for each human being in the planet.  At a particular moment of time, you might not be able to get enough data to build such a language model.

For me, the point of the whole discussion is that ASR is an engineering system, not some idealistic discussion topic.  There will always be tradeoff.   You may say: “What if a certain technology Y emerge in the next 50 years?” I heard that a lot Y could be quantum computing or brain simulation or brain-human interface or machine implementation of brain.    Guys….. those, I got to admit are very smart idea in our time, and give it another 30-40 years, we might see something useful.   For now, ASR really has nothing to do with them.  I never heard of machine implementation of the audio cortex, or even an accurate construction of audio pathway.  Nor, there is an easy progress of dissecting mammal inner ear and bring understanding on what’s going on in human ear.   From what I know, we seem to know some, but there are lots of other things we don’t know.

That’s why I think it’s better to buckle down and just to try to work out our stuffs.  Meaning, try to come up with more interesting mathematical model, try to come up with more computational efficient method.   Those …. I think are meaningful discussion.   As for Kurzweil, no doubt he is a very smart guy, but at least on ASR, I don’t think he knows what he talks about.

Of course, I am certainly not the only person who complains Kurzweil.  Look at how Douglas Hofstadter’s criticism:

“It’s as if you took a lot of very good food and some dog excrement and blended it all up so that you can’t possibly figure out what’s good or bad. It’s an intimate mixture of rubbish and good ideas, and it’s very hard to disentangle the two, because these are smart people; they’re not stupid.”

Sounds like very reasonable to me.

Arthur

Categories
acoustic score cmu sphinx linguistic score. Speech Recognition

Acoustic Score and Its Signness

Over the years, I got asked about why acoustic score could be a positive number all the time. That occasionally lead to a kind of big confusion from beginner users. So I write this article as a kind of road sign for people.

Acoustic score per frame is essentially the log value of continuous distribution function (cdf). In Sphinx’s case, the cdf is a multi-dimensional Gaussian distribution. So Acoustic score per phone will be the log likelihood of the phone HMM. You can extend this definition to word HMM.

For the sign. If you think of a discrete probability distribution, then this acoustic score thingy should always be negative. (Because log of a decimal number is negative.) In the case of a Gaussian distribution though, when the standard deviation is small, it is possible that the value is larger than 1. (Also see this link). So those are the time you will see a positive value.

One thing you might feel disharmonious is the magnitude of the likelihood you see. Bear in mind, Sphinx2 or Sphinx3 are using a very small logbase. We are also talking about a multi-dimensional Gaussian distribution. It makes numerical values become bigger.

Arthur

Also see:
My answer on the Sphinx Forum

Categories
bw state sequence

Commentary on SphinxTrain1.07’s bw (Part II : next_state_utt.c’s First Half)

I will go on with the analysis of bw.  In my last post, we understand the high-level structure of the bw program.  So we now turns to the details of how the state lattice was built.  Or how next_state_utt.c works.

As always, here is the standard disclaimer: this is another long and super technical post I only expect a small group of programmer with exoteric interest of Baum-Welch algorithm can read.   It’s not for faint of heart but if you understand the whole thing, you would have some basic but genuine understanding of Baum-Welch algorithm in real life.

The name of the module, next_state_utt.c, is quite unassuming but it is an important key of understanding the Baum-Welch algorithm.  The way how the state lattice is structured affects how parameter estimation works.   The same statement says for not only Baum-Welch estimation but also other estimation algorithm in speech.

But what so difficult about the coding the lattice?  Here are two points I think it is worthwhile to point out:

  1. I guess an average programmer can probably work out a correct concatenation of all phone HMMs if all phones are context-independent in 3 days to 1 week.  But in many advanced systems, most of them are using context-dependent phones.  So you go to make sure at the right time, the right triphone state was used.
  2. In Sphinx, it got a bit more complicated because there is a distinction between positions of triphones.  This is quite specific to Sphinx and you won’t find it in systems such as HTK.  So it further complicates coding.  You will see in the following discussion, Sphinx has back off cases of position-dependent triphone estimation from time-to-time.   In my view, it might not be too different from the position-independent triphone system.  (It’s certainly fun to read. 🙂 ) 
My goal here is to analyze the next_state_utt.c, how it works and to state some part one can improve.

High-Level Layout of next_utt_state.c:

 next_utt_state.c  
-> mk_wordlist (mk_wordlist.c)
-> mk_phone_list (mk_phonelist.c)
-> cvt2triphone (cvt2triphone.c)
-> state_seq_make (state_seq_make.c)

Nothing fancy here. We first make a wordlist (mk_wordlist) , then make a phone list (mk_phone_list), then convert the phone list to triphone (cvt2triphones), then create the state sequence (state_seq_make).

Now before we go on, just at this level, you may already discover one issue of Sphinx’s bw.  It is using a list to represent phone models.   So, let’s assume if you want to model a certain word with multiple pronunciations, you probably can’t do it without changing the code.

Another important thing to note: just like many non-WFST systems, it is not that easy to make a simple phoneme system with Sphinx.  (HTK is an exception but you can always turn on a flag to expand context. Just look up the manual.)  Say if you want to express your phoneme system to be one phoneme word, then you would want your dictionary look like:

AA AA
AE AE
B B
.
.
.
.
.

But then, if a word is a phone, should you actually want to build a network of cross-word triphones?  You probably want to if you want to shoot for performance – all of the most accurate phoneme-based system has some sort of context-dependency there.  (The Brno’s recognizer probably has some, but I don’t really grok why it is so good.)

But if you want to do your own interesting experiments, this fixed behavior may not suit your appetite.   Maybe you just want to use a context-independent phone system for some toy experiments.  But then, you are probably always building a triphone system.  So, it might or might not be what you like.

So if you really want to trigger the CI-model behavior, what can you do?  Take a look of my next post, in cvt2triphone.c, if the model definition file only specify CI states, then no triphone conversion will occur.   In a way, that is to say the system assume if you just train the CI model, you will get the CI model but there is no explicit way to turn it off.

mk_wordlist 

mk_wordlist is rather trivial:

 char **mk_wordlist(char *str,  
uint32 *n_word)
{
uint32 n_w;
uint32 i;
char **wl;
n_w = n_words(str);
wl = ckd_calloc(n_w, sizeof(char *));
wl[0] = strtok(str, " t");
for (i = 1; i < n_w; i++) {
wl[i] = strtok(NULL, " t");
}
assert(strtok(NULL, " t") == NULL);
*n_word = n_w;
return wl;
}

With one line of transcripts, mk_wordlist transform it to an array of C-string.  Memory of the string are allocated.

mk_phone_list

mk_phone_list is still trivial but there is a bit more detail


1:  acmod_id_t *mk_phone_list(char **btw_mark,  
2: uint32 *n_phone,
3: char **word,
4: uint32 n_word,
5: lexicon_t *lex)
6: {
7: uint32 n_p;
8: lex_entry_t *e;
9: char *btw;
10: unsigned int i, j, k;
11: acmod_id_t *p;
12: /*
13: * Determine the # of phones in the sequence.
14: */
15: for (i = 0, n_p = 0; i < n_word; i++) {
16: e = lexicon_lookup(lex, word[i]);
17: if (e == NULL) {
18: E_WARN("Unable to lookup word '%s' in the lexiconn", word[i]);
19: return NULL;
20: }
21: n_p += e->phone_cnt;
22: }
23: /*
24: * Allocate the phone sequence
25: */
26: p = ckd_calloc(n_p, sizeof(acmod_id_t));
27: /*
28: * Allocate the between word markers
29: */
30: btw = ckd_calloc(n_p, sizeof(char));
31: for (i = 0, k = 0; i < n_word; i++) { /* for each word */
32: e = lexicon_lookup(lex, word[i]);
33: if (e->phone_cnt == 0) /* Ignore words with no pronunciation */
34: continue;
35: for (j = 0; j < e->phone_cnt-1; j++, k++) { /* for all but the last phone in the word */
36: p[k] = e->ci_acmod_id[j];
37: }
38: p[k] = e->ci_acmod_id[j]; /* move over the last phone */
39: btw[k] = TRUE; /* mark word boundary following
40: kth phone */
41: ++k;
42: }
43: *btw_mark = btw;
44: *n_phone = n_p;
45: assert(k == n_p);
46: return p;
47: }

In line 15-22:, we first look up the pronunciations of the words. (Remember, right now we can only look up one.) It then allocate the an array of phones with ID (in the type of acmod_id_t).

Now here is special part of the code, other the array of phones, it also allocate an array call “between word markers”.  So what’s the mechanism?  Let me give an example.

Suppose you have a transcript with word sequence “MY NAME IS CLINTON”

       mk_word_list would create

     word[0] -> MY  
word[1] -> NAME
word[2] -> IS
word[3] -> CLINTON

       mk_print_list (with my best guess of pronunciations) would create

     ph[0] -> M      btw[0] -> 0
ph[1] -> AY btw[1] -> 1
ph[2] -> N btw[2] -> 0
ph[3] -> EY btw[3] -> 0
ph[4] -> M btw[4] -> 1
ph[5] -> IY btw[5] -> 0
ph[6] -> S btw[6] -> 1
ph[7] -> K btw[7] -> 0
ph[8] -> L btw[8] -> 0
ph[9] -> IY btw[9] -> 0
ph[10] -> N btw[10] -> 0
ph[11] -> T btw[11] -> 0
ph[12] -> AX btw[12] -> 0
pH[13] -> N btw[13] -> 1

So essentially it would indicate there is a word end at a certain phone.

I believe such as representation are for convenience purpose: it facilitate determination of whether a word is at the beginning, the middle or the end.

An alternative here is to do an optional silence.  This, according to HTK handbook, usually reduce the WER slightly.  It seems to be reasonable to figure out where the location of a phone is using silences as a marker.

A digression: acmod_id_t

1:  typedef unsigned char ci_acmod_id_t;  
2: #define NO_CI_ACMOD (0xff)
3: #define MAX_CI_ACMOD (0xfe)
4: typedef uint32 acmod_id_t;
5: #define NO_ACMOD (0xffffffff) /* The ID representing no acoustic
6: * model. */
7: #define MAX_ACMOD (0xfffffffe) /* The max ID possible */

Now, we used acmod_id_t in mk_phone_list, but what is it really?  So let’s take a detour of acmod_id_ds.t (“ds” stands for data structure.)

acmod_id_t is essentially a uint32, which is just a the size of unsigned integer or 2^32 -1.  Why -1? Notice that MAX_CI_ACMOD was defined as 0xfe?

The more interesting part here: we saw ci_acmod_id_t is only a character type.  This is obviously another problem here, in some languages, one may be interested to express it with more than 255 phones. (Why 255?)

We’ll meet acmod_set a little bit more.  But let us move on first – sometimes code tracing will be more motivated when you see the code before the data structure.   Many suggest otherwise: Indeed, once you know the data, code will make more sense.   But in practice, you will most likely read the code first and needs to connect things together.   Thus IMO: both approach has their merit in code tracing.

So far ….. and next post

To avoid clutter a single post, I will stop and put the rest of next_utt_states.c (cvt2phones and state_seq_make) on another post.   But I want to summarize several things I have observed so far:
  1. Current bw doesn’t create a word network so it has issues to handle multiple pronunciations. 
  2. Current bw automatically expand triphone contexts. There is no explicit way to turn it off.
  3. bw is not doing optional silence in the network. 
  4. bw doesn’t work for more than 255 CI phones. 
Btw, SphinxTrain1.08 has several changed which replace mk_word_list with data structure from sphinxbase.  Those are encouraging changes.  If I have time, I will cover them. 
Arthur

Categories
C++ DBN HMM java learning NLTK Python Ruby scipy wfst writing

Learning vs Writing

I haven’t done any serious writings for a week.  Mostly post interesting readings just to keep up the momentum.   Work is busy so I slowed down.  Another concern is what to write.   Some of the topics I have been writing such as Sphinx4 and SphinxTrain take a little bit of research to get them right.

So far I think I am on the right track.  There are not many bloggers on  speech recognition.  (Nick is an exception.)   To really increase awareness of how ASR is done in practice, blogging is a good way to go.

I also describe myself as “recovering” because there are couple of years I hadn’t seriously thought about open source Sphinx.  In fact though I was working on speech related stuffs, I didn’t spend too much time on mainstream ASR neither because my topic is too esoteric.

Not to say, there are many new technologies emerged in the last few years.   The major one I would say is the use of neural network in speech recognition.  It probably won’t replace HMM soon but it is a mainstay for many sites already.   WFST, with more tutorial type of literature available, has become more and more popular.    In programming, Python now is a mainstay plus job-proof type of language.  The many useful toolkit such as scipy, nltk by themselves deserves book-length treatment.  Java starts to be like C++, a kind of necessary evil you need to learn.  C++ has a new standard.   Ruby is huge in the Web world and by itself is fun to learn.

All of these new technologies took me back to a kind of learning mode.   So some of my articles become longer and in more detail.   For now, they probably cater to only a small group of people in the world.   But it’s okay, when you blog, you just want to build landmarks on the blogosphere.   Let people come to search for them and get benefit from it.   That’s my philosophy of going on with this site.

Arthur

Categories
Frederick Jelinek Linguistic Noam Chomsky Peter Norvig

How Norvig refutes Chomsky

In a well-written and elaborated post, Peter Norvig reputed Noam Chomsky’s arguments against statistical models. (Links: I, II)

“Then, to get language from this abstract, eternal, mathematical realm into the heads of people, he [Chomsky] must fabricate a mystical facility that is exactly tuned to the eternal realm. This may be very interesting from a mathematical point of view, but it misses the point about what language is, and how it works.”

On speech recognition, I cannot help but think Norvig is right.  There are many times introduction of linguistic knowledge proven to be not as helpful as we hope.   If you are interested in the topic, you should also read Jelinek’s piece on Some of my Best Friends are Linguists.

Arthur

Categories
C++ Management Python Scheme time

Readings from Jan 19, 2012

C spiral law of parsing

Python vs Scheme

How do you manage geeks?

Things you know about time.

Arthur