Towards an Intelligent Acoustic Front End for Automatic Speech Recognition: Built-in Speaker Normalization
© U.H. Yapanel and J.H.L. Hansen. 2008
Received: 27 December 2007
Accepted: 29 May 2008
Published: 19 June 2008
A proven method for achieving effective automatic speech recognition (ASR) due to speaker differences is to perform acoustic feature speaker normalization. More effective speaker normalization methods are needed which require limited computing resources for real-time performance. The most popular speaker normalization technique is vocal-tract length normalization (VTLN), despite the fact that it is computationally expensive. In this study, we propose a novel online VTLN algorithm entitled built-in speaker normalization (BISN), where normalization is performed on-the-fly within a newly proposed PMVDR acoustic front end. The novel algorithm aspect is that in conventional frontend processing with PMVDR and VTLN, two separating warping phases are needed; while in the proposed BISN method only one single speaker dependent warp is used to achieve both the PMVDR perceptual warp and VTLN warp simultaneously. This improved integration unifies the nonlinear warping performed in the front end and reduces simultaneously. This improved integration unifies the nonlinear warping performed in the front end and reduces computational requirements, thereby offering advantages for real-time ASR systems. Evaluations are performed for (i) an in-car extended digit recognition task, where an on-the-fly BISN implementation reduces the relative word error rate (WER) by 24%, and (ii) for a diverse noisy speech task (SPINE 2), where the relative WER improvement was 9%, both relative to the baseline speaker normalization method.
Current speaker-independent automatic speech recognition (ASR) systems perform well in most of the real-world applications but the performance gap between speaker-dependent and speaker-independent settings is still significant. Although a reasonable amount of progress have occurred in recent years in the general ASR technology by exploiting more complex algorithms with the help of faster computing , little progress has been reported in the development of core speech processing algorithms. Many speech researchers would agree that there is still a significant potential in formulating an acoustic representation of the speech signal that will successfully maintain information needed for efficient speech recognition, especially in noise, while eliminating irrelevant speaker-dependent information . The perceptual MVDR (PMVDR) coefficients have proven to be more effective than the MFCC front end on a number of tasks, especially in noisy environments [2, 3]. This paper introduces a new and computationally efficient speaker normalization algorithm within the PMVDR [2, 3] framework which we call built-in speaker normalization (BISN). BISN is computationally efficient and can be completely integrated into the front-end.
There are different ways to address speaker variability for automatic speech recognition. One approach is to normalize speaker variabilities in the feature space prior to employing an HMM acoustic recognizer framework. A number of effective algorithms have been developed to compensate for such variabilities due to speaker stress and emotion (see  for an overview). Probably, the most successful approach is the adaptive cepstral compensation (ACC)  which was shown to significantly reduce the impact of speaker variability for ASR. This approach uses a low-level voiced/transitional/unvoiced segmentation scheme followed by a source generator framework to compensate the MFCC cepstral feature sequence prior to ASR. More recent approaches have focused on reducing the impact of vocal-tract length differences in the spectral domain [6, 7].
Basic likelihood-based warp estimation was first introduced by Andreou et al. . However, it was computationally cumbersome and required a substantial amount of speech from each speaker in order to estimate the best warp factor. Their basic motivation was to extract acoustic features that have reduced speaker dependency. In order to achieve this, they linearly warped the frequency axis. The degree of this linear warping is in fact a speaker-dependent factor and must be estimated for each speaker. For the estimation of the warp factor, they proposed a set of maximum likelihood-based procedures. Unfortunately, these procedures were computationally very expensive.
Lee and Rose [6, 7] proposed a set of speaker normalization procedures using maximum likelihood estimates of the best warp for each speaker. There was no attempt to recover the underlying vocal-tract shape. Instead, their motivation was to use an optimization criterion directly related to the one used in the recognizer. They revised the set of maximum likelihood estimation procedures proposed by Andreou  to estimate the warp factors for each speaker. These procedures are now widely known as vocal-tract length normalization (VTLN). The most popular way of estimating VTLN warps is to use likelihood-based estimation techniques [6, 7] in which a set of HMM models trained on a large population of speakers by placing 1 Gaussian per state is scored against warped features. Afterwards, incoming features are extracted using different VTLN warps, and the warp producing the maximum likelihood (given the HMMs and transcription) is used as the best VTLN warp for that speaker. VTLN is shown to be effective for a number of tasks but the computational load of determining the best warp for each speaker, especially at the time of recognition, is not tractable. They also proposed computationally more efficient variants of the VTLN based on the GMM modeling of each VTLN warp [6, 7]. However, these variants are less accurate due to the loss of temporal information (this stems from the use of GMMs in the modeling) buried in the speech signal. As a result, although a good method for offline simulations, classical VTLN is rarely used in practical systems where computational efficiency is of primary concern. Therefore, there is a need for achieving on-the-fly speaker normalization by introducing computationally more efficient algorithms.
Eide and Gish  proposed a waveform-based algorithm, in which they estimate the warping factors by using the average position of the third formant. Their idea is that the third formant is not affected as much as the first and second formants from the context and therefore more closely related to the speaker's vocal-tract length. By using the ratio of the average third-formant location for a particular speaker to the average third-formant location for a large population of speakers, they were able to determine reasonable normalization factors, which helped reduce interspeaker variations. Although this approach has the advantage of estimating the speaker-normalization warps directly from the speech signal, the difficulty of estimating the third formant reliably even for clean speech is apparent, as some speakers may not even have clear third-formant locations.
Acero  proposed a speaker-dependent bilinear transform (BLT) to account for interspeaker variations. In that study, an LPC-based front end is used with the FFT spectrum warped before the computation of the cepstral coefficients. A vector quantization distortion measure is computed to estimate the best BLT warp for each speaker. Substantial performance improvements were obtained with the LPC-based cepstral coefficients (LPCCs). The proposed BISN algorithm has some similarities with Acero's approach . In both methods, a first-order all-pass system (or a BLT) is used to incorporate the perceptual scale into the feature extraction process. A fixed BLT warp factor, is used to approximate Mel and Bark scales as needed. In order to reduce the speaker differences, a best BLT warp factor, is specifically estimated for each speaker, which in some sense, integrates perceptual BLT warp and speaker normalization BLT warp into a single speaker-dependent BLT warp factor. The procedure employed to estimate the best BLT warp factor for each speaker, on the other hand, has substantial differences. As mentioned above, Acero used a vector quantization distortion measure in order to estimate the best BLT warp factor for each speaker. Our approach in BISN is fundamentally different in the sense that each best BLT warp factor is estimated within the VTLN framework proposed by Lee and Rose [6, 7]. Moreover, several other algorithms are also integrated within the search process in order to reduce the computational load down to manageable levels for real-time implementations.
The feasibility of bilinear and all-pass transforms (BLT, APT) has also been extensively studied by McDonough [11, 12]. In that study, the BLT is implemented in the cepstral domain. The best BLT parameters were estimated by a Gaussian mixture model (GMM) as the one maximizing the likelihood of the incoming data [11, 12]. The BISN approach is somehow related to this method, however relation is merely in the use of a BLT for speaker normalization. McDonough did not make any attempt to integrate perceptual warp and speaker normalization BLT warp into a single warp (which BISN does). Rather, he used cepstrum transformation matrices (which are derived from the BLT) on the final MFCC vectors to achieve the speaker normalization. This means that still the perceptual and speaker normalization warps are performed in two separate steps, perceptual warp is achieved through use of a nonlinearly distributed Mel-filterbank whereas speaker normalization is achieved through the use of an appropriate matrix transformation after the Mel cepstra have been computed.
In this paper, we integrate BLT-based speaker normalization within the perceptual MVDR (PMVDR) coefficients framework [2, 3]. First, we demonstrate that the perceptual warp is actually meant to remove some of the existing speaker differences. By estimating a specific perceptual warp factor for each speaker, it is possible to further remove these speaker-dependent differences. Then, the warp estimation process is computationally improved by integrating a binary tree search (BTS)  approach which reduces the computation 67% with respect to the classical VTLN. Next, performing the best warp search in the model space rather than in the feature space  further reduces the necessary computational resources for real-time applicability and performance. Finally, a configuration for on-the-fly implementation of this built-in speaker normalization (BISN) algorithm is proposed for an in-car speech recognition task which reduces the word error rate (WER) 24% relative to the baseline PMVDR-based system.
In Section 2, we summarize the theoretical background for the PMVDR front end which is the basis for the BISN algorithm. In Section 3, we consider the underlying meaning of so-called perceptual warping. We show via a modified LDA-based analysis [15, 16] that perceptual warping successfully removes a substantial amount of interspeaker variability. This observation leads to the idea of using a specific self-normalization warp factor for each speaker. The offline approach for the vocal-tract length normalization (VTLN) is summarized in Section 4 with its disadvantages in terms of computational efficiency. Section 5 formulates the built-in speaker normalization (BISN) algorithm in detail. Improvements to the search are introduced in Sections 5.1 and 5.2. We summarize our evaluation results in Section 6 for two different tasks, CU-Move extended digit recognition task and the speech in noisy environment (SPINE-2) task. Section 7 explains how one can easily integrate the BISN algorithm within the PMVDR framework for a real-world application. After summarizing computational considerations for the different algorithms proposed in this paper in Section 8, we make concluding remarks in Section 9.
2. The PMVDR Acoustic Front End
PMVDR is a new acoustic front end which does not use a nonlinearly spaced filterbank to incorporate perceptual considerations. Instead of using a filterbank, the FFT spectrum is directly warped before the envelope extraction stage [2, 3]. The envelope is extracted via a low-order all-pole MVDR spectrum which is shown to be superior to the linear prediction- (LP-) based envelopes . Utilizing direct warping on the FFT power spectrum by removing filterbank processing avoids the smoothing effect of a filterbank and leads to preservation of almost all information that exits in the short-term speech spectrum. Also, using the MVDR method to extract the envelope contributes greatly to superior performance in noisy conditions [2, 3]. We now shortly summarize the MVDR spectrum estimation to extract the spectral envelope and the warping via interpolation algorithm to directly warp the FFT spectrum. For the details of the PMVDR computation we refer readers to [2, 3].
2.1. Minimum Variance Distortionless Response (MVDR) Spectrum Estimation
2.2. Direct Warping of FFT Spectrum
where represents the linear frequency, while represents the warped frequency. Here, the value of controls the degree of warping. We are more interested in the nonlinear phase response through which we implement the perceptual warping. For 16 kHz sampled signals, we set and to approximate the Mel and Bark scales, respectively. For 8 kHz, these values are adjusted to and . Bark scale performs more warping in the lower frequencies when compared to the Mel scale.
2.3. Implementation of Direct Warping
where is the lower nearest linear FFT bin, is the nearest upper linear FFT bin, and is the value of the warped power spectrum that corresponds to FFT bin . Thus, the spectral value , at the warped frequency index , is computed as the linear interpolation of nearest upper, , and lower, , spectral values in the linear frequency space.
2.4. Implementation of PMVDR
In utilizing a filterbank for incorporating perceptual scales, the filterbank has two tasks, (i) warping the spectrum nonlinearly and (ii) smoothing out excitation details. In using direct warping, on the other hand, no averaging of the FFT power spectrum is used to achieve smoothing, only warping of the spectrum is performed. The smoothing is achieved through a low-order MVDR analysis that follows the warping step. Therefore, in the direct warping of the spectrum, little information is lost.
Obtain the perceptually warped FFT power spectrum via interpolation.
Compute the "perceptual autocorrelation lags" by taking the IFFT of the "perceptually warped" power spectrum.
Calculate the th-order MVDR spectrum using (2) from the LP coefficients .
Obtain the final cepstrum coefficients using the straightforward FFT-based approach . In this implementation, after obtaining the MVDR coefficients from the perceptually warped spectrum, we take the FFT of the parametrically expressible MVDR spectrum. After applying the log operation, we apply IFFT to return back to the cepstral domain.
3. The "Meaning" of Perceptual Warping
4. Offline VTLN
The most popular method for speaker normalization is vocal-tract length normalization (VTLN) in which the speech spectrum is linearly warped with an optimal warp factor ( ) [6, 7, 27]. The warping can also be performed by rearranging the position of the Mel filters [6, 7]. However, in the PMVDR front end, we no longer use a filterbank structure, and therefore warping is directly performed on the FFT power spectrum. In the offline VTLN application, a two-step warp needs to be performed. The first warp is called perceptual warp and applied during the extraction of acoustic features. VTLN warp also needs to be performed in cascade to the perceptual warp within the acoustic front end. The speaker-dependent parameter, , is generally determined by conducting likelihood computations for different values within the range [0.84â€“1.16] (for our purpose we extend the range slightly to facilitate the binary search algorithm described in Section 5.1). Generally a single-Gaussian HMM set which is trained on all available training data is used to estimate the warp factor.
4.1. Warping Factor Estimation
Assume that we have utterances from speaker and would like to estimate the warp factor for this speaker. Here, we define the following terms as in :
Obtaining a closed-form solution for is difficult since the frequency warping corresponds to a highly nonlinear transformation of the speech features. Therefore, the best warp is estimated by searching over a grid of 33 points spaced evenly in the range of [0.84â€“1.16]. The goal of training is to obtain a canonical (normalized) set of HMMs, , in the sense that each speaker's utterance is warped with an appropriate warping factor and the resulting HMM is defined over a frequency-normalized feature set. Initially, the HMM set is trained from unwarped utterances, and this model is used to estimate the best warp factor for each speaker. Afterwards, every speaker's utterances are parameterized with the estimated best warp factor and then the HMM model set is re-estimated from this warped feature set. In theory, this new canonical model can be used to re-estimate the optimal warp factors, and another HMM can be trained and the procedure iterated several times. However, during our experimentation with offline VTLN, we observed that further iterating did not yield significant improvements over the first iteration, therefore we only estimate the optimal warps once and train the canonical HMMs from the feature set parameterized with these optimal warps.
Finally, we warp the utterance with the estimated warp factor, and redecode using the normalized HMM model, . The output of the recognizer is our final recognition result. For offline VTLN experiments reported in this paper, however, we used all the available data from each test speaker to estimate the best warps in an offline setting (i.e., warp factors are not estimated for every single utterance).
Typically, we parameterize speech within the range of [0.84â€“1.16] and with a step size of 0.01 yielding a 33-point search space. Using the monotonic property, we compare the likelihoods at the current warp and at the previous warp. When the difference is negative, the best warp is found. On the average, the estimation of the best VTLN warp for a speaker requires 18 times the computational resources for one feature extraction and one likelihood computation. During the test, we must perform recognition twice in order to obtain an initial transcription to estimate the optimal warp.
5. Built-In Speaker Normalization (BISN)
Our earlier interspeaker variability analysis yielded the fact that so-called perceptual warping is in fact a speaker-normalization warping too. Motivated by this outcome, we can adjust the perceptual warp parameter specifically for each speaker and call this new warp the self-normalization warp. This should, in turn, normalize the vocal-tract length differences. Since this procedure does not require 2 applications of warping to the spectrum (one for the perceptual warp and one for the VTLN warp), as in offline VTLN, it is more efficient. Moreover, the normalization is achieved by only adjusting an internal parameter of the acoustic front end (i.e., the perceptual warp factor ), making it a built-in procedure, hence the name built-in speaker normalization (BISN). The self-normalization warp ( ) in the BISN context refers to a nonlinear mapping (as defined by (3) and (4)) whereas in the VTLN context the speaker normalization warp ( ) refers to a linear mapping of the frequency axis.
The estimation of the self-normalization warp, , for speaker , is done in a manner similar to offline VTLN. Here, is estimated as the one which maximizes the total likelihood of the data given a single-Gaussian HMM set. Another advantage of BISN is the reduced search space. While in classical VTLN, the search space is generally a 33-point grid, for the BISN case, using a 17-point search space yields sufficient accuracy. (In our implementation, the search was over this range, but one may reduce the dimension of the search space at the expense of performance.) In a typical setting with a perceptual warp factor of (Bark scale at 16 kHz), the search space for the self-normalization warps can be chosen as [0.49, 0.65] reducing the search space by half versus that for VTLN. The search for the self-normalization warp within the BISN framework requires 10 times the computational resources for one feature extraction and one likelihood computation, which is still computationally expensive. The search is a computationally intensive procedure. This disadvantage has been noticed by other researchers . Taking advantage of the monotonic property of the likelihood function, one can use a binary tree search  rather than linear search which reduces the computational load substantially with no performance loss (i.e., by producing exactly the same warp factors).
5.1. Binary Tree Search (BTS) Approach
Using this monotonic property of the likelihood function, it is possible to devise a much more efficient search algorithm than the linear search approach . In , a Brent search was used in order to efficiently obtain the best warp factor. Without loss of generality, we will call the efficient search process as binary tree search (BTS) in this paper.
Compute for , if , then disregard the upper region, and consider the lower region as the new search space whose middle warp is and return to Step (2). If , then compute , for . If then disregard the lower region, and consider the upper region as the new search space whose middle warp is and return to Step (2). For the last case where , take the new search space to be ], whose middle warp is and return to Step (2). In all the cases, the search space is reduced by half.
By recursively repeating Steps (2) and (3), we compute the optimal warp for a speaker with an average of 6 times the computational resources for one feature extraction and one likelihood computation (with the example settings above). Thus, the BTS algorithm summarized above reduces the number of likelihood computations from 10 to 6 for the BISN algorithm, exactly producing the same self-normalization warps. For BTS approach integrated within the BISN algorithm (considering a 17-point search space), the number of feature extraction and likelihood computations is 6, hence when compared with classical VTLN, it estimates the self-normalization warps with a 67% relative reduction in the computational load.
5.2. Model Versus Feature Space Search
where is the Jacobian of the transformation taking to . In conventional speaker normalization, the contribution of the Jacobian is not taken into account since this may cause some systematic errors in optimal warp factor estimation. When the search is conducted in the model space, the need to compensate for the Jacobian of the transformation is eliminated .
For the estimation of the optimal warp, extract the features with self-normalization warp, (this generally can be chosen as , which is the center of our search space) and then select the model (trained with ) yielding the maximum likelihood given the warped features. The search is again performed with the BTS approach to quickly find the warped model giving the largest likelihood, .
After determining the self normalization warps by using the model space search approach summarized above, the rest of the normalization is similar to the offline VTLN. The canonical HMMs are trained from warped features which are extracted using appropriate self-normalization warps. During the test, same model-based approach is used to determine the self-normalization warp factors, and a two-pass recognition is performed.
Changing the search space from the feature to model space helps reducing the computational load further for estimating the optimal self-normalization warps. Now for the MS-BTS-based BISN, we need to extract the features only once and then perform 6 likelihood computations on the average to obtain the optimal self-normalization warp.
6. Experimental Framework
In order to test the effectiveness of the proposed BISN algorithm, recognition experiments were performed on two different databases that address different adverse conditions. We believe that it is important to test the speaker normalization algorithms for actual adverse environments, in order to determine if they have practical value. The databases used in the simulations are (a) CU-Move database-extended digits Portion , for real noisy in-car environments, (b) speech in noisy environments (SPINEs) , for simulated noisy military task conditions. These databases reflect good examples of environments where reliable and efficient speaker normalization is needed.
6.1. General System Description
For all experiments, we used SONIC [32, 33], the University of Colorado's HMM-based large vocabulary speech recognition system. We used a window length of 25 milliseconds and a skip rate of 10 milliseconds by Hamming windowing the frame data before further processing. The 39-dimensional feature set contains 12 statics, deltas and delta-deltas along with normalized-log energy, delta and delta-delta energy. Cepstral mean normalization (CMN) was utilized on the final feature vectors.
For both VTLN and BISN experiments, a single best warp is estimated for each speaker offline using all available data. We re-extracted PMVDR features using these best warps and retrained the HMM model set in order to obtain canonical models. During the test, a two-pass strategy was used. First, all utterances from a single speaker are recognized with noncanonical HMM set, and best warp factors are estimated using the result of this recognition. In the second step, the utterances for that speaker are extracted incorporating the best warps obtained in the first step, and a second recognition is performed with the canonical models to get the final hypothesis.
6.2. Experiments for CU-Move Extended Digits Task
A total of 60 speakers balanced across gender and age (18â€“70 years old) were used in the training set. (Note that  summarizes recommended training development and test sets for the CU-Move corpus.) The test set contained another 50 speakers, again gender and age balanced. The HMMs were trained using SONIC's decision-tree HMM trainer [32, 33] resulting in a model set with approximately 10 K total Gaussians. The 40-word vocabulary is very convenient for telephone dialing applications since it contains many necessary words like "dash", "pound", "sign" in addition to numbers. We used the optimized settings ( and ) for PMVDR on the CU-Move task .
Although there is no substantial improvement in the WER performance of the BISN-based techniques with respect to VTLN baseline, there is a computational gain and the convenience of performing the recognition within the acoustic front end merely changing an internal parameter. BISN-based normalization can be easily integrated into embedded systems, such as in-car speech-based navigation systems, without increasing the computational cost significantly.
6.3. Experiments for the SPINE Task
7. Application of BISN in a Real-Time Scenario
Pass through a recursive averaging block with a forgetting factor( ) to obtain an averaged version (i.e., ). Here, the forgetting factor was set to 0.6, an optimization experiment is presented in this chapter later on.
In summary, the front end estimates the self-normalization warp for the incoming utterance by using the self-normalization warp estimated from the earlier utterances via a recursive averaging with a forgetting factor. After performing recognition with the estimated self-normalization warp, the recognizer feeds back the alignment information so that the self-normalization warp for the next utterance can be estimated (and updated).
8. Computational Considerations
Computational complexity for different speaker normalization algorithms. (NWs: number of warpings, NFEs: number of feature extractions, NLCs: number of likelihood computations, NRPs: number of recognition passes).
In this paper, we have proposed a new and efficient algorithm for performing online and efficient VTLN which can easily be implemented within the PMVDR front end. In VTLN, we need to perform warping on the spectrum twice, to accommodate perceptual considerations and to normalize for speaker differences. The proposed BISN algorithm, on the other hand, estimates a self-normalization warp for each speaker which performs both the perceptual warp and speaker normalization in a single warp. The use of a single warp to achieve both perceptual warp and VTLN warp unifies these two concepts. The model space-binary tree search (MS-BTS) algorithm was integrated to reduce the computational load in the search stage for the estimation of self-normalization warps. Moving the search base from the feature space to the model space  reduced the need to extract the features for each point in the search space, which in turn eliminated the need for high computational resources. A sequential on-the-fly implementation of the BISN w/MS-BTS algorithm also eliminated the need to perform multipass recognition which makes it possible to integrate this scheme with low-resource speech recognition systems.
We have shown that the BISN approach is effective for two different databases, the CU-Move in-vehicle dialog (extended digits portion) database and the SPINE military noisy speech database. The on-the-fly implementation of the BISN w/MS-BTS algorithm was also shown to be slightly more accurate than the offline version with a considerable savings in computational resources. Integrated with the BISN approach, the PMVDR front end can now be considered an intelligent front end which cooperates with the recognizer in order to automatically normalize itself with respect to the incoming speaker/speech. Since it can quickly adapt to the changing vocal-tract characteristics, it does not require any detection of speaker changes whatsoever. We believe that the PMVDR front end integrated with the strong BISN algorithm is an ideal front end for use in every system requiring noise robustness and a measurable level of speaker normalization (especially for embedded systems). It can perform acoustic feature extraction with moderate computational requirements and achieve self-normalization with respect to changing speakers very efficiently, yielding a sound acoustic front end that can be used in today's demanding speech recognition applications.
This work was supported by US Air Force Research Laboratory, Rome NY, under Contract no.FA8750-04-1-0058.
- Hunt MJ: Spectral signal processing for ASR. Proceedings of the IEEE Workshop on Automatic Speech Recognition and Understanding (ASRU '99), December 1999, Keystone, Colo, USA 1: 17-26.Google Scholar
- Yapanel UH, Hansen JHL: A new perspective on feature extraction for robust in-vehicle speech recognition. Proceedings of the 8th European Conference on Speech Communication and Technology (EUROSPEECH '03), September 2003, Geneva, Switzerland 1281-1284.Google Scholar
- Yapanel UH, Hansen JHL: A new perceptually motivated MVDR-based acoustic front-end (PMVDR) for robust automatic speech recognition. Speech Communication 2008,50(2):142-152. 10.1016/j.specom.2007.07.006View ArticleGoogle Scholar
- Hansen JHL: Analysis and compensation of speech under stress and noise for environmental robustness in speech recognition. Speech Communication 1996,20(1-2):151-173. 10.1016/S0167-6393(96)00050-7View ArticleGoogle Scholar
- Hansen JHL: Morphological constrained feature enhancement with adaptive cepstral compensation (MCE-ACC) for speech recognition in noise and Lombard effect. IEEE Transactions on Speech and Audio Processing 1994,2(4):598-614. 10.1109/89.326618View ArticleGoogle Scholar
- Lee L, Rose RC: Speaker normalization using efficient frequency warping procedures. Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '96), May 1996, Atlanta, Ga, USA 1: 353-356.Google Scholar
- Lee L, Rose RC: A frequency warping approach to speaker normalization. IEEE Transactions on Speech and Audio Processing 1998,6(1):49-60. 10.1109/89.650310View ArticleGoogle Scholar
- Andreou A, Kamm T, Cohen J: Experiments in vocal tract normalization. Proceedings of the CAIP Workshop: Frontiers in Speech Recognition II, July-August 1994, Piscataway, NJ, USAGoogle Scholar
- Eide E, Gish H: A parametric approach to vocal tract length normalization. Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '96), May 1996, Atlanta, Ga, USA 1: 346-348.Google Scholar
- Acero A: Acoustical and environmental robustness in automatic speech recognition, Ph.D. thesis. Carnegie Mellon University, Pittsburgh, Pa, USA; 1990.Google Scholar
- McDonough J: Speaker compensation with all-pass transforms, Ph.D. thesis. The John Hopkins University, Baltimore, Md, USA; 2000.Google Scholar
- McDonough J, Byrne W, Luo X: Speaker adaptation with all-pass transforms. Proceedings of the 5th International Conference on Spoken Language Processing (ICSLP '98), November-December 1998, Sydney, Australia 6: 2307-2310.Google Scholar
- Hain T, Woodland PC, Niesler TR, Whittaker EWD: The 1998 HTK system for transcription of conversational telephone speech. Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '99), March 1999, Phoenix, Ariz, USA 57-60.Google Scholar
- Sinha R, Umesh S: A method for compensation of Jacobian in speaker normalization. Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '03), April 2003, Hong Kong 1: 560-563.View ArticleGoogle Scholar
- Haeb-Umbach R: Investigations on inter-speaker variability in the feature space. Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '99), March 1999, Phoenix, Ariz, USA 1: 397-400.View ArticleGoogle Scholar
- Kim Y: Signal modeling for robust speech recognition with frequency warping and convex optimization, Ph.D. thesis. Department of Electrical Engineering, Stanford Univerity, Palo Alto, Calif, USA; 2000.Google Scholar
- Murthi MN, Rao BD: All-pole modeling of speech based on the minimum variance distortionless response spectrum. IEEE Transactions on Speech and Audio Processing 2000,8(3):221-239. 10.1109/89.841206View ArticleGoogle Scholar
- Musicus BR: Fast MLM power spectrum estimation from uniformly spaced correlations. IEEE Transactions on Acoustics, Speech, and Signal Processing 1985,33(5):1333-1335. 10.1109/TASSP.1985.1164696View ArticleGoogle Scholar
- Smith JO III, Abel JS: Bark and ERB bilinear transforms. IEEE Transactions on Speech and Audio Processing 1999,7(6):697-708. 10.1109/89.799695View ArticleGoogle Scholar
- Tokuda K, Masuko T, Kobayashi T, Imai S: Mel-generalized cepstral analysis-a unified approach to speech spectral estimation. Proceedings of the International Conference on Spoken Language Processing (ICSLP '94), September 1994, Yokohama, Japan 1043-1046.Google Scholar
- Haykin S: Adaptive Filter Theory. Prentice-Hall, Englewood Cliffs, NJ, USA; 1991.MATHGoogle Scholar
- Makhoul J: Linear prediction: a tutorial review. Proceedings of the IEEE 1975,63(4):561-580.View ArticleGoogle Scholar
- Oppenheim AV, Schafer RW: Discrete-Time Signal Processing. Prentice-Hall, Englewood Cliffs, NJ, USA; 1989.MATHGoogle Scholar
- Yapanel UH: Acoustic modeling and speaker normalization strategies with application to robust in-vehicle speech recognition and dialect classification, Ph.D. thesis. Center for Spoken Language Research, Department of Electrical and Computer Engineering, Univerity of Colorado at Boulder, Boulder, Colo, USA; 2005.Google Scholar
- Yapanel UH, Dharanipragada S, Hansen JHL: Perceptual MVDR-based cepstral coefficients (PMCCs) for high accuracy speech recognition. Proceedings of the 8th European Conference on Speech Communication and Technology (EUROSPEECH '03), September 2003, Geneva, Switzerland 1829-1832.Google Scholar
- Hansen JHL, Zhang X, Akbacak M, et al.: CU-MOVE: advanced in-vehicle speech systems for route navigation. In DSP for In-Vehicle and Mobile Systems. Kluwer Academic Publishers, Dordrecht, The Netherlands; 2004:19-45.Google Scholar
- Zhan P, Westphal M: Speaker normalization based on frequency warping. Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '97), April 1997, Atlanta, Ga, USA 2: 1039-1042.Google Scholar
- LDC, http://wave.ldc.upenn.edu/catalog/catalogentry.jsp?catalogid=ldc93s6a
- Pitz M, Molau S, Schluter R, Ney H: Vocal tract normalization equals linear transformation in cepstral space. Proceedings of the 7th European Conference on Speech Communication and Technology (EUROSPEECH '01), September 2001, Aalborg, DenmarkGoogle Scholar
- CSLR, http://www.cumove.colorado.edu.com
- Hansen JHL, Sarikaya R, Yapanel UH, Pellom B: Robust speech recognition in noise: an evaluation using the SPINE corpus. Proceedings of the 7th European Conference on Speech Communication and Technology (EUROSPEECH '01), September 2001, Aalborg, Denmark 2: 905-908.Google Scholar
- Pellom B: SONIC: the university of colorado continuous speech recognizer. Center for Spoken Language Research, University of Colorado at Boulder, Boulder, Colo, USA; 2001.Google Scholar
- Pellom B, Hacioglu K: Recent improvements in the CU Sonic ASR system for noisy speech: the SPINE task. Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '03), April 2003, Hong Kong 1: 4-7.View ArticleGoogle Scholar
- Hansen JHL: Getting started with the cu-move database. The Center For Spoken Language Research, Boulder, Colo, USA; 2002.http://www.cumove.colorado.edu.comGoogle Scholar
- Hansen JHL, Angkititrakul P, Plucienkowski J, et al.: CU-move: analysis & corpus development for interactive in-vehicle speech systems. Proceedings of the 7th European Conference on Speech Communication and Technology (EUROSPEECH '01), September 2001, Aalborg, Denmark 209-212.Google Scholar
- Bou-Ghazale SE, Hansen JHL: A comparative study of traditional and newly proposed features for recognition of speech under stress. IEEE Transactions on Speech and Audio Processing 2000,8(4):429-442. 10.1109/89.848224View ArticleGoogle Scholar
This article is published under license to BioMed Central Ltd. This is an open access article distributed under the Creative Commons Attribution License, which permits unrestricted use, distribution, and reproduction in any medium, provided the original work is properly cited.