Leveraging Speech Production Knowledge for Improved Speech Recognition Abhijeet Sangwan, John H.L. Hansen Center for Robust Speech Systems (CRSS) University of Texas at Dallas (UTD), Richardson, Texas, U.S.A {abhijeet.sangwan,John.Hansen}@utdallas.edu

Abstract—This study presents a novel phonological methodology for speech recognition based on phonological features (PFs) which leverages the relationship between speech phonology and phonetics. In particular, the proposed scheme estimates the likelihood of observing speech phonology given an associative lexicon. In this manner, the scheme is capable of choosing the most likely hypothesis (word candidate) among a group of competing alternative hypotheses. The framework employs the Maximum Entropy (ME) model to learn the relationship between phonetics and phonology. Subsequently, we extend the ME model to a ME-HMM (maximum entropy-hidden Markov model) which captures the speech production and linguistic relationship between phonology and words. The proposed MEHMM model is applied to the task of re-processing N-best lists where an absolute WRA (word recognition rate) increase of 1.7%, 1.9% and 1% are reported for TIMIT, NTIMIT, and the SPINE (speech in noise) corpora (15.5% and 22.5% relative reduction in word error rate for TIMIT and NTIMIT).

I. I NTRODUCTION In this study, we propose a disambiguation scheme based on phonological features (PFs) that exploits speech production knowledge embedded in phonology. Our motivation stems from research which shows that PFs capture production variability in greater detail than phones [1], [2], [3], [4]. Herein, the inability of phones to effectively model production variability is exposed in the errors made by standard ASR. As shown in Fig. I, production related difficulty is often presented as ambiguous output in standard ASR structures such as Nbest lists, lattices or a word-mesh. In these structures, ASR performance can be improved by selecting the correct alternative among ambiguous words. In view of this observation, the proposed scheme is designed to exploit phonological knowledge in order to consistently select correct words from ambiguous alternatives. This paradigm is shown in Fig. I, where the proposed scheme [Block (3)] first extracts PF sequences from the speech signal [Block (2a)], and then uses the PF information to disambiguate confusable words in the standard ASR output [Block (2b)]. In particular, the proposed scheme computes and assigns a “phonological score” to each and every ambiguous word based on the likelihood of jointly observing that word and the corresponding PFs. It is the leveraging of these two domains that makes the proposed solution unique. Thereafter, the new scores are used in conjunction with language model (LM) scores to re-rank the alternate hypotheses, and choose the best candidate. In summary, our contribution is the proposed PF-based framework that allows an efficient and

TABLE I G OVERNMENT P HONOLOGY Property Resonance Primes Manner Source Headedness

Attributes A, I, E, U S, h, N H a, i, u

Examples A=/ae/, I=/ih/, A+I=/eh/ A+S+N=/n/, E+h=/z/ E+h+H=/s/ A+a=/aa/,A+I+a+i=/ay/

meaningful integration of phonology knowledge into standard ASR systems. The proposed system can always be combined with existing and continuously improving back-end solutions, thereby leveraging the orthogonal knowledge of phonology to solve a common problem of improving ASR performance. The ability to compute phonological scores for words (or corresponding phone-sequences) lies at the heart of the proposed scheme. In this study, we design and develop a MEHMM (maximum entropy-hidden Markov model) to learn the probabilistic relationship between phones and their corresponding spectro-temporal articulation variability (via PFs). Subsequently, we apply this newly acquired knowledge to score the joint word-phonology observations. Specifically, the MEM (maximum entropy model) models the phonological variability of phones, and the HMM structure models the temporal evolution of words from a succession of its component phones. Furthermore, a robust contextually-aware input feature-set is also proposed for the MEM. The proposed feature-set uses the knowledge of phonetic-context and cophonological states to build accurate models of phones-PFs relationship. Finally, the proposed ME-HMM based phonological disambiguation scheme is used to process ASR N-best lists in a speech recognition experiment. II. G OVERNMENT P HONOLOGY S YSTEM In this section, we briefly review the basics of Government Phonology (GP) theory employed within the scope of this study. A. Government Phonology The GP theory is built on a small set of primes (articulation properties), and rules that govern their interactions [3]. A GP prime produces phones in speech by operating in isolation or in combination with other primes. The primes are broadly categorized into three groups, namely, resonance, manner and source primes. In general, resonance primes govern vowels, manner primes govern articulation of consonants, and source

1

INPUT UTTERANCE

"cereal grains have been used for centuries to prepare fermented beverages"

2a

Phonological Features Extraction

GP Elements Phonological Dimensions

GP State in ON (=1)

2b

Standard ASR Output

GP State in OFF (=0) N-Best List

E

cereal cereal cereal cereal cereal

U I H

Ambiguous

Unambiguous

A

grains grains grains grains grains

have have have have have

been been been been been

use use used used used

force force for for for

increased up it increased up in centuries to pair centuries but in centuries to prepare

fermented beverages fermented beverages fermented beverages fermented beverages fermented beverages

Word Mesh

h

pair

S

but

N

use

have

cereal

for

grains

i

it

increased to

a been

used

force centuries

fermented

in

beverages

up

u

Time (speech frames)

3

prepare

Proposed Phonological Features based Disambiguation Scheme

Proposed Maximum Entropy-Hidden Markov Model (ME-HMM) captures relationship between phonological features and phones

Proposed Methodology exploits phonological features-phone relationship knowlege to choose correct alternatives among ambiguous outputs

(1) Identify ambiguous words information e.g. "for" vs. "force" in above word-mesh (2) Use canonical phone expansions for ambiguous words e.g. "for" = /f/ /aw/ /r/ (3) Use ME-HMM to assign ''phonological score'' to ambiguous words e.g. "for" = -0.1 and "force" = -0.3 (4) Use new scores to generate new recognition hypotheses e.g. "for" is more likely due to higher score

Fig. 1. Proposed Phonological Features (PFs) based disambiguation scheme for speech recognition. For a speech utterance (1), a parallel PF extraction (2a) is used to disambiguate standard ASR (automatic speech recognition) output (2b) by means of the proposed ME-HMM model (3).

primes dictate voicing in speech. Table I lists the GP primes along with examples of how primes generate phonemes. B. Integrating GP in the Proposed Scheme The extraction of GP features is performed using an HMMbased scheme described in [1]. Using the GP recognition scheme described above, 11 binary numbers are obtained for each speech frame in the utterance corresponding to the 11 GP elements. The binary representation stems from the fact that each GP element can either be off (0) or on (1) for each speech frame. It is useful to note that since each GP type is decoded independently, the system allows for asynchronous behavior in terms of on/off switching action of the GP elements (e.g., if an unvoiced stop follows a nasal phone, then nasalization and voicing are correctly allowed to turn off at different times). In this manner, 2048 (= 211 ) unique combinations of GP

elements per speech frame are possible. Here, a small subset of combinations represent the canonical form of the phones, while a majority of combinations represent the phoneticvariants. In this study, the canonical GP forms of the phones are taken from [3]. III. P ROPOSED ME-HMM F RAMEWORK In this section, we develop the proposed ME-HMM framework for our phonology-based disambiguation scheme. In the proposed scheme, the ME-HMM models the relationship between a GP-sequence and a phone-sequence. This process is illustrated in Fig. 2, where the ME-HMMs corresponding for words “for” and “force” are shown. The phone sequences for words “for” (/f/,/ao/,/r/) and “force” (/f/,/ao/,/r/,/s/) are shown on the Y-axes, and the corresponding ith GP-sequence is shown on the X-axes. Due to the ambiguity faced by standard

ASR, the ith GP-sequence could have been generated by either “for” or “force” (see 2b in Fig. I). In order to resolve such ambiguity, the proposed ME-HMM computes the likelihood of a word (or equivalent phone-sequence) generating the GP sequence. In this manner, 11 likelihoods are computed per word corresponding to 11 GP-elements. By summing the 11 likelihoods, a “phonological score” is obtained for each word. As a result of employing ME-HMMs, 3 scores are now assigned to each ambiguous word in the standard ASR output: acoustic score, language score (by the standard ASR), and phonological score (by the proposed ME-HMM system). Using the newly generated phonological scores, previously confusable words can now reassessed and therefore resolved (e.g., “force” vs. “for”). In this manner, the phonological score offers a new dimension of separability among otherwise ambiguous words by leveraging the unique knowledge of speech production.

5) The initial state distribution given by Π is defined below. In order to define the initial state distribution (Π) and state transition matrix (A), it is assumed that (i) every valid path in the trellis is initiated at the first phone and terminated at the last phone, and (ii) transitions only occur from a phone to itself or the succeeding phone-state. Now, if the probability of a self transition is given by β; then the probability of transition to the next phone-state is given by 1−β. Therefore, the transition matrix A for the trellis becomes,   β 1−β 0 ··· 0  0 β 1 − β ··· 0    A= . . ..  , .. ..  .. . .  0

1) Number of HMM states are equal to the number of canonical phones in w (i.e., M ), 2) Number of distinct observation symbols for the ith GP sequence are equal to 2, (0 or 1) 3) The state transition matrix given by A is defined below, 4) The observation symbol probability distribution given by b is modeled by the MEM, and is developed in Sec. III-B, and,

···

0

1

where amk is the (m, k)th element of A. Furthermore, based on the above arguments, the initial state vector is given by,

A. Development of ME-HMM Let w be a candidate word, and w ≡ {p1 p2 ... pM } be the canonical phone-sequence. In other words, w is composed of M phones p1 p2 ... pM in this order. In Fig. 2, this is illustrated for an example word “force” composed of phones p1 =/f/, p2 =/ao/, p3 =/r/, and p4 =/s/. Let the word w span over N speech-frames, where the j th speech frame is denoted by fj . Furthermore, let the binary state of the ith GP-element (i = 1, . . . , 11) and j th speech frame for word w be given by gij . The above-defined variables are shown in Fig. 2, where phones are in circles, and GP-states are in squares. Within the production of w, the exact temporal evolution of phones p1 -pM is unknown due to the inherent uncertainity in articulation. This uncertainity is captured within the phonetrellis where each path in the trellis is one possible articulation of w. In Fig. 2, one choice of articulation is shown by the path in bold. Furthermore, each path in the trellis determines the phones responsible for the observed GP states (gij , j = 1, . . . , N ). For example, for the bold path shown in Fig. 2: gij−2 = 1 is generated by /f/, gij−1 = 1 is generated by /ao/, and so on. This generative relationship between phones and corresponding GP states is captured by the MEM in the proposed scheme, which is developed in greater detail in Sec. III-B. Next, if all articulation possibilities of w are considered, the forward algorithm [5] can be used to compute the total likelihood of w generating the observed ith GP sequence. In order to use the forward algorithm, the various elements of the HMM framework are first enumerated below:

0

Π = [1 0 · · · 0]T .

(1)

Finally, let the likelihood that the w generated the ith GP sequence be given by Λi (w), which is computed by the forward algorithm as follows, Initialization : α1 (k)

= Π(k)b(gi1 , pk ), k = 1, . . . , M , (2)

Recursion : αj+1 (k)

M X

= b(gij+1 , pk )

αj (m)amk ,

(3)

m=1

k = 1, . . . , M , T ermination : Λi (w)

=

M X

αM (m),

(4)

m=1

where α is the likelihood of partial observations. The “phonological-score” ΛP F for the word w can now be obtained by summing the likelihoods over all individual GP sequences, 11 X ΛP F (w) = Λi (w). (5) i=1

The phonological-score ΛP F may also be viewed as a production score for the word w, where a high score requires production characteristics of w in the GP space to conform with observed statistical variability. Herein, it is important to note that the lexical output from standard ASR and GP sequence from the GP extraction system are two manifestations of the same underlying speech event. Therefore, it is intuitive to believe that in a list of words, the correct word choice is most likely to show maximum agreement with the corresponding phonology. The proposed ME-HMM system exploits this intuition by leveraging PFs to comprehensively model observed variability in phones. This newly obtained statistical knowledge is then used to score the claim of each word for the list of alternatives.

Alternative Word 1: "for"

for = /f/ /ao/ /r/

Phone sequence:

ME-HMM f

f

f

f

f

ao

ao

ao

ao

ao

r

r

r

r

r MEM

GP sequence g ij-2

g ij

g ij-1

1

1

g ij+1

0

i th GP sequence for the utterance e.g. 'A'

0

Alternative Word 2: "force" ME-HMM

GP sequence

force = /f/ /ao/ /r/ /s/

Phone sequence:

MEM f

f

f

f

ao

ao

ao

ao

p3 = r

r

r

r

r

p4 = s

s

s

s

s

f j-2

f j-1

f

f j+1

p1 = f p2 = ao

j

speech frames for the utterance

Fig. 2. Relationship between observed ith GP element states and the generative phone trellis corresponding to the articulated word. Due to ambiguity in standard ASR decoding, either ”for” or ”force” is responsible for observed phonological sequence. The likelihood that each trellis generated observed GP sequence is computed using proposed ME-HMM model and forward algorithm.

In the next section, we develop the MEM which serves the role of computing observation symbol generating probabilites b in the above-described HMM structure.



B. Maximum Entropy Modeling Maximum Entropy (ME) modeling is an evidence based discrete modeling technique. It has been successfully employed in many speech and language tasks such as part-ofspeech (POS) tagging [6], machine translation (MT) [7], and acoustic modeling [8]. The ME model (MEM) is extremely flexible since it can relate a variable number of events to the observation. It provides an attractive methodology for relating phones, phonetic context, and high-level information with the GP state of a given frame. The objective of the MEM is to model the output symbol generation probability distribution (b) conditioned on various levels of knowledge. Particularly, the MEM views the different levels of knowledge as evidence. Upon training, the MEM is able to learn weight parameters that quantify the relative importance of different evidences. Within our MEM framework, evidence is implemented as features. Specifically, the feature value is 1 if the evidence is present, and 0 if not. The different evidence-types are shown in Fig. 3 and listed below: • (I) Co-occurring phone p(j): For a GP state gij , the phone that occupies that j th frame in the phone-trellis is the









co-occurring phone. In Fig. 2, for word “force” /r/ is the co-occurring phone for gij . (II) Preceding phone: The preceding phone occurs before the co-occurring phone in the phone-sequence. For example, /ao/ is the preceding phone for the co-occurring phone /r/ in the previous case. (III) Succeeding phone: The succeeding phone occurs after the co-occurring phone in the phone-sequence. For example, /s/ is the succeeding phone for the co-occurring phone /r/. (IV) Static Co-phonology (Active): The complementary active GP states (gi0 j = 1, i0 6= i) for the same j th frame. In Fig. 3, for frame fj , the co-phonological states are S,H,A for GP state of I. (V) Static Co-phonology (Inactive): The complementary inactive GP states (gi0 j = 0, i0 6= i) for the same j th frame. In Fig. 3, for frame fj , the co-phonological states are E,U,h,N,a,i,u for GP state of I. (VI) Dynamic Co-phonology (“Steady”): The dynamic knowledge in co-phonological states captures the knowledge of recent state-transitions (0-to-1 or 1-to-0). As shown in Fig. 3, the state-transition of interest are in the neighborhood of 2 frames of the j th frame (fj−2 to fj+2 ). If no state-transitions are in this neighborhood, then the co-phonological sequence is in the “steady” state (e.g.,

(2) preceding phone phone sequence

target i th GP sequence

f

ao

(1) co-occurring phone p(j)

g ij = 1

I

(7) dynamic co-phonology "steady"

s

r

(5) static co-phonology OFF state

co-phonological GP states

(3) suceeding phone

(6) dynamic co-phonology "transient"

(4) static co-phonology ON state

S

0

0

1

1

1

transient

H

0

0

1

1

1

transient

A

1

1

1

1

1

steady

E

0

0

0

0

0

steady

U

0

0

0

0

0

steady

h

0

0

0

0

0

steady

N

0

0

0

0

0

steady

a

0

0

0

0

0

i

0 0

0 0

0 0

0 0

0 0

steady steady

f j-2

f j-1

f

f j+1

f j+2

u

j

steady

Frames

Fig. 3. MEM models the conditional probability of observing the GP state gij given the various phonetic and co-phonological evidence. The (1) co-occuring phone, (2) previous phone, and (3) succeeding phone constitute phonetic evidence. The (4) co-phonological OFF states, (5) co-phonological ON states, (6) co-phonological “transient” states, and (7) co-phonological “steady” states form the co-phonological evidence.



E,U,N,H,h,a,i,u in Fig. 3). (VII) Dynamic Co-phonology (“Transient”): If statetransitions are in this neighborhood, then the cophonological sequence is in the “transient” state (e.g., A,S,H in Fig. 3).

Evidences (I), (II), and (III) represent the triphone-context. Furthermore, evidences (IV) and (V) represent the static cophonological information. Finally, evidences (VI) and (VII) represent the dynamic knowledge in co-phonology. It is noted that moving from evidence types (I) through (VII) constitutes a growing body of evidence. It is expected that MEM quality would improve as newer evidence is incorporated into the modeling. Finally, we formalize the development of the MEM. Let Ej be the set of evidence available at the j th frame, and el ∈ Ej be the lth evidence as discussed above. Furthermore, let the ME feature for the ith GP element be given by µi . Each ME feature is a binary operator on the evidence el , (i.e., if the feature is observed, it produces a value of unity, otherwise it is zero),  1 if evidence el is present, µi (el ) = 0 otherwise. For example, in the case of evidence (I): µi (e1 ) = 1 for cooccurring phone only and µi (e1 ) = 0 for all other phones.

Now, the MEM can be used to compute the observation symbol probabilites (b) in the ME-HMM model (see Sec. III). In particular, the observation symbol generation computes the probability of observing GP state gij given the evidence set Ej . Let bij be the likelihood of observing gij given Ej , given by: bij = p(gij |Ej ) =

L X 1 exp ( λil µi (el )), Zλ (E)

(6)

l=1

where Zλ (E) is a normalization term, and λil are the weights assigned to the ME feature. As mentioned earlier, the weights correspond to the importance of a feature in estimating the likelihood in question. The MEM parameters are learned offline during the training phase. In this study, the ME models were trained using the “Maximum Entropy Modeling Toolkit” [9]. IV. R ESULTS AND D ISCUSSION The proposed PF-based disambiguation scheme is applied to the task of re-processing N-Best lists. Particularly, we work with 20-Best lists generated from the test sections of the TIMIT, NTIMIT, and SPINE corpora. As shown in Table II the baseline WERs (word error rates) obtained for the TIMIT, NTIMIT, and SPINE corpora are 8%, 12.1%, and 37.8% respectively.

TABLE II TIMIT, NTIMIT, AND SPINE: S PEECH R ECOGNITION P ERFORMANCE Sub

Del

Baseline ME-HMM

6.6 4.7

1.3 1.5

Baseline ME-HMM

10.0 7.6

2.1 2.5

Baseline ME-HMM

28.1 28.1

9.6 8.1

Ins WRA TIMIT 2.1 92.1 1.7 93.8 NTIMIT 2.2 87.9 1.5 89.8 SPINE 10.1 62.8 12.4 63.8

WER

Oracle WER

10.0 7.9

3.6 3.6

14.4 11.7

5.27 5.27

47.9 48.7

16.44 16.44

TABLE III TIMIT, NTIMIT, AND SPINE: W ORD R ECOGNITION ACCURACIES FOR DIFFERENT PART- OF - SPEECH ELEMENTS Adjective Baseline ME-HMM

94.7 94.7

Baseline ME-HMM

92.9 93.6

Baseline ME-HMM

62.7 62.1

Adverb TIMIT 94.2 95.0 NTIMIT 91.2 93.7 SPINE 76.6 76.0

Noun

Verb

Others

87.2 90.0

90.5 93.3

92.1 93.5

80.9 83.9

86.2 88.4

89.0 90.7

49.3 49.6

52.9 56.4

60.8 64.6

In order to obtain the state-sequences for all GP elements, a separate HMM-based GP extraction scheme was trained for TIMIT, NTIMIT, and SPINE. Furthermore, a separate MEM was also trained for TIMIT, NTIMIT, and SPINE using data from the train-sets of the respective corpora. During test, the HMM-based GP extraction scheme was used to generate the GP state-sequences for the test-sets from TIMIT, NTIMIT, and SPINE. As a result, we obtain the GP state-sequences corresponding to the same utterances for which the 20-Best lists were generated using the above baseline ASR system. Next, each 20-best list was processed as follows: First the wordlevel time-segmentation, acoustic-score and language-score for each ambiguous word within the N-best list was identified. As shown in Fig. I, the ambiguous words in an N-best list are readily identified. Using the word-level timing information, the corresponding 11 GP state-sequences for each ambiguous word was identified. Subsequently, by using the proposed MEHMM model and the forward algorithm described in Sec. III, a “phonological score” for each ambiguous word was computed. Furthermore, the phonological score for each sentence in the 20-Best list was computed by taking a sum of the constituent ambiguous word phonological-scores. Finally, the 20-Best list was re-ranked by using a total hypothesis score which was a simple linear combination of the phonological and language scores. As shown in Table II, the proposed ME-HMM based PF disambiguation scheme achieved an absolute WRA (word recognition accuracy) improvement of 1.7%, 1.9%, and 1% for TIMIT, NTIMIT, and SPINE, respectively as a result of the 20-Best list re-ranking process. In order to illustrate the nature of improvement obtained by the proposed scheme, we show the part-of-speech (POS) tags for the correctly detected words from the baseline ASR

and proposed ME-HMM scheme in Table III. This splitanalysis of word recognition serves to illustrate the nature of the improvement obtained by the proposed scheme. The POS tags for the analysis are obtained by means of the tree-tagger tool [10]. From the table, it is observed that the proposed system improves word recognition in all POS categories (adverbs, adjectives, nouns, verbs and others) for both TIMIT and NTIMIT. For the case of SPINE, the proposed scheme achieves improved noun and verb word recognition rates. Herein, the gain in word recognition rates of nouns and verbs is significant, since they tend to be more informationbearing within the utterances, and of particular importance for applications like spoken document retrieval (SDR) [11]. V. C ONCLUSION In this paper, a novel methodology for speech recognition disambiguation based on the ME-HMM framework was proposed. The proposed ME-HMM framework served to exploit the relationship between low-level signal phonology and higher-level speech phonetics. Subsequently, the ME model was adapted into an HMM framework to form a MEHMM system which was employed as a tool to compute the likelihood of observing speech segments conditioned on phonological knowledge. In our experiments, words were chosen as the logical speech segments, but the system is just as easily applied to supra or sub-lexical structures. By computing phonological scores of N-best lists, we were able to resolve ambiguity by achieving a relative WER reduction of 22.5% and 15.7% in the TIMIT and NTIMIT corpora. R EFERENCES [1] A. Sangwan and J. H. L. Hansen, “Evidence of coarticulation in a phonological feature detection system,” in Interspeech’08, 2008. [2] O. Scharenborg, V. Wan, and R. K. Moore, “Towards capturing fine phonetic variation in speech using articulatory features,” Speech Communication, vol. 49, no. 10-11, pp. 811–826, Nov. 2007. [3] S. King and P. Taylor, “Detection of phonological features in continuous speech using neural networks,” Computer Speech and Language, vol. 14, no. 4, pp. 333–353, 2000. [4] S. King, J. Frankel, K. Livescu, E. McDermott, K. Richmond, and M. Wester, “Speech production knowledge in automatic speech recognition,” JASA, vol. 121, no. 2, pp. 723–742, Feb. 2007. [5] L. R. Rabiner, “A tutorial on hidden Markov models and selected applications in speech recognition,” Proceedings of the IEEE, vol. 77, no. 2, pp. 257–286, Feb 1989. [6] A. Ratnaparkhi, “A maximum entropy model for part-of-speech tagging,” in Conference on Empirical Methods in Natural Language Processing, 1996. [7] L. Gu, Y. Gao, L. Fu-Hua, and M. Picheny, “Concept-based speech-tospeech translation using maximum entropy models for statistical natural concept generation,” IEEE Transactions on Audio, Speech and Language Processing, vol. 14, no. 2, pp. 377–392, 2006. [8] H.-K. Kuo and Y. Gao, “Maximum entropy direct models for speech recognition,” IEEE Transactions on audio, speech, and language processing, vol. 14, no. 3, pp. 873–881, 2006. [9] Z. Le, Maximum Entropy Modeling Toolkit for Python and C++, http://homepages.inf.ed.ac.uk/s0450736/maxent toolkit.html. [10] H. Schmid, “Probabilistic part-of-speech tagging using decision trees,” in Proceedings of the International Conference on New Methods in Language Processing, 1994. [11] J. H. L. Hansen, R. Huang, B. Zhou, M. Seadle, J. R. Deller, A. R. Gurijala, and P. Angkititrakul, “SpeechFind: Advances in spoken document retrieval for a national gallery of the spoken word,” IEEE Trans. Speech and Audio Processing, Special Issue on Data Mining, vol. 13, no. 5, pp. 712–730, September 2005.

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nutritional supplements (7), and new protein polymers with both medical and ... tion of a plant species that is grown hydroponically or in in vitro systems so that ... harvested material has a limited shelf life and must be processed .... benefits on

Learning improved linear transforms for speech ... - Semantic Scholar
class to be discriminated and trains a dimensionality-reducing lin- ear transform .... Algorithm 1 Online LTGMM Optimization .... analysis,” Annals of Statistics, vol.

CASA Based Speech Separation for Robust ... - Semantic Scholar
and the dominant pitch is used as a main cue to find the target speech. Next, the time frequency (TF) units are merged into many segments. These segments are ...

Estimation for Speech Processing with Matlab or ... - Semantic Scholar
This shows the dimension of M. Try to find the size of y and z. What does the answer tell you about how Octave represents vectors internally? Type size(1) ... how ...

Yedalog: Exploring Knowledge at Scale - Semantic Scholar
neck when analyzing large repositories of data. We introduce Yedalog, a declarative programming language that allows programmers to mix data-parallel ...

Ontologies and Scenarios as Knowledge ... - Semantic Scholar
using it throughout the systems development phases. The paper emphasizes the need for better approaches for representing knowledge in information systems, ...

Ontologies and Scenarios as Knowledge ... - Semantic Scholar
using it throughout the systems development phases. The paper emphasizes the need for better approaches for representing knowledge in information systems, ...