Improve Speech Recognition Accuracy Research Articles

Implementation of Sound Direction Detection and Mixed Source Separation in Embedded Systems.

In recent years, embedded system technologies and products for sensor networks and wearable devices used for monitoring people's activities and health have become the focus of the global IT industry. In order to enhance the speech recognition capabilities of wearable devices, this article discusses the implementation of audio positioning and enhancement in embedded systems using embedded algorithms for direction detection and mixed source separation. The two algorithms are implemented using different embedded systems: direction detection developed using TI TMS320C6713 DSK and mixed source separation developed using Raspberry Pi 2. For mixed source separation, in the first experiment, the average signal-to-interference ratio (SIR) at 1 m and 2 m distances was 16.72 and 15.76, respectively. In the second experiment, when evaluated using speech recognition, the algorithm improved speech recognition accuracy to 95%.

ЗАСТОСУВАННЯ ГЛИБОКИХ ШТУЧНИХ НЕЙРОННИХ МЕРЕЖ ДЛЯ КЛАСИФІКАЦІЇ МУЛЬТИМОДАЛЬНИХ ДАНИХ

Multimodal data analysis is gaining attention in recent research. Pu Liang et al. (2023) provide a comprehensive overview on multimodal machine learning, highlighting its founda-tions, challenges and achievements in recent years. More problem-oriented works propose new methods and applications for multimodal ML, such a Ngiam et al. (2011) propose to use joint audio and video data to improve speech recognition accuracy; Sun, Wand and Li (2018) describe application of multimodal classification for breast cancer prognosis prediction; Mao et al. (2014) propose an architecture of multimodal recurrent network to generate text de-scription of images and so on. However, such works usually focus on the task itself and meth-ods therein, and not on integrating multimodal data processing into other software systems. The goal of this research is to propose a way to conduct multimodal data processing, specifically as a part of a digital twin systems, thus efficiency and near-real-time operation are required. The paper presents an approach to conduct parallel multimodal data classification, adapting to available computing power. The method is modular and scalable and intended for in digital twin application as a part of analysis and modeling tools. Later, the detailed example of such a software module is discussed. It uses multimodal data from open datasets to detect and classify the behavior of pets using deep learning mod-els. Videos are processed using two artificial neural networks: YOLOv3 object detection net-work to process individual frames of the video and a relatively simple convolutional network to classify sounds based on their frequency spectra. Constructed module uses a producer-consumer parallel processing pattern and allows processing 5 frames per second of a video on available hardware, which can be sufficiently improved by using GPU acceleration or more paralleled processing threads.

Integrated visual transformer and flash attention for lip-to-speech generation GAN

Lip-to-Speech (LTS) generation is an emerging technology that is highly visible, widely supported, and rapidly evolving. LTS has a wide range of promising applications, including assisting speech impairment and improving speech interaction in virtual assistants and robots. However, the technique faces the following challenges: (1) Chinese lip-to-speech generation is poorly recognized. (2) The wide range of variation in lip-speaking is poorly aligned with lip movements. Addressing these challenges will contribute to advancing Lip-to-Speech (LTS) technology, enhancing the communication abilities, and improving the quality of life for individuals with disabilities. Currently, lip-to-speech generation techniques usually employ the GAN architecture but suffer from the following problems: The primary issue lies in the insufficient joint modeling of local and global lip movements, resulting in visual ambiguities and inadequate image representations. To solve these problems, we design Flash Attention GAN (FA-GAN) with the following features: (1) Vision and audio are separately coded, and lip motion is jointly modelled to improve speech recognition accuracy. (2) A multilevel Swin-transformer is introduced to improve image representation. (3) A hierarchical iterative generator is introduced to improve speech generation. (4) A flash attention mechanism is introduced to improve computational efficiency. Many experiments have indicated that FA-GAN can recognize Chinese and English datasets better than existing architectures, especially the recognition error rate of Chinese, which is only 43.19%, the lowest among the same type.

A novel channel estimate for noise robust speech recognition

We propose a novel technique to estimate the channel characteristics for robust speech recognition. The method focuses on reliable time–frequency speech patches which are highly independent of the noise condition. Combined with a root-based approximation of the logarithm in the MFCC computation, this reduces the variance caused by the noise on the spectral features, and therefore also the constrain on the acoustic model in a multi-style training setup. We show that compared to the standard mean normalization, the proposed method estimates the channel equally well under clean conditions and better under noisy conditions. When integrated in the feature extraction pipeline, we show improvements in speech recognition accuracy on noisy speech and a status quo on clean speech. Our experiments reveal that this method helps the most for generative models that need to model the complex noise variability, and less so for discriminative models, which can learn to ignore noise instead of accurately modeling it. Our approach outperforms the state of the art on the noisy Aurora4 task.

Application of Speech Recognition in English PronunciationCorrection

In the context of economic globalization, countries are beginning to conduct ordinary trade transactions, so English, asthe most frequently used language, has been incorporated into the education system of each government and studentsare being taught English systematically, but the fastest way to improve English must be for a student to have a one-onone lesson with a native English speaker. However, with the world’s advanced educational resources, teaching Englishto students on a one-to-one basis is obviously not practical. Therefore, implanting a speech recognition system intoeducational software to help the software track or correct students’ English learning in real time is possible. However,replacing a real English teacher with the current speech recognition accuracy rate is still a challenge. Therefore, thispaper explains in detail how to improve speech recognition accuracy through lip motion tracking and MFCC and to givelearners the corresponding vocabulary ratings and suggestions for improvement through this model. In the long run, theknowledge tracking model is used to help students find the most suitable learning style.

Improving N-Best Rescoring in Under-Resourced Code-Switched Speech Recognition Using Pretraining and Data Augmentation

In this study, we present improvements in N-best rescoring of code-switched speech achieved by n-gram augmentation as well as optimised pretraining of long short-term memory (LSTM) language models with larger corpora of out-of-domain monolingual text. Our investigation specifically considers the impact of the way in which multiple monolingual datasets are interleaved prior to being presented as input to a language model. In addition, we consider the application of large pretrained transformer-based architectures, and present the first investigation employing these models in English-Bantu code-switched speech recognition. Our experimental evaluation is performed on an under-resourced corpus of code-switched speech comprising four bilingual code-switched sub-corpora, each containing a Bantu language (isiZulu, isiXhosa, Sesotho, or Setswana) and English. We find in our experiments that, by combining n-gram augmentation with the optimised pretraining strategy, speech recognition errors are reduced for each individual bilingual pair by 3.51% absolute on average over the four corpora. Importantly, we find that even speech recognition at language boundaries improves by 1.14% even though the additional data is monolingual. Utilising the augmented n-grams for lattice generation, we then contrast these improvements with those achieved after fine-tuning pretrained transformer-based models such as distilled GPT-2 and M-BERT. We find that, even though these language models have not been trained on any of our target languages, they can improve speech recognition performance even in zero-shot settings. After fine-tuning on in-domain data, these large architectures offer further improvements, achieving a 4.45% absolute decrease in overall speech recognition errors and a 3.52% improvement over language boundaries. Finally, a combination of the optimised LSTM and fine-tuned BERT models achieves a further gain of 0.47% absolute on average for three of the four language pairs compared to M-BERT. We conclude that the careful optimisation of the pretraining strategy used for neural network language models can offer worthwhile improvements in speech recognition accuracy even at language switches, and that much larger state-of-the-art architectures such as GPT-2 and M-BERT promise even further gains.

Talking Points: A Modulating Circle Increases Listening Effort Without Improving Speech Recognition in Young Adults.

Speech recognition is improved when the acoustic input is accompanied by visual cues provided by a talking face (Erber in Journal of Speech and Hearing Research, 12(2), 423-425, 1969; Sumby & Pollack in The Journal of the Acoustical Society of America, 26(2), 212-215, 1954). One way that the visual signal facilitates speech recognition is by providing the listener with information about fine phonetic detail that complements information from the auditory signal. However, given that degraded face stimuli can still improve speech recognition accuracy (Munhall, Kroos, Jozan, & Vatikiotis-Bateson in Perception & Psychophysics, 66(4), 574-583, 2004), and static or moving shapes can improve speech detection accuracy (Bernstein, Auer, & Takayanagi in Speech Communication, 44(1-4), 5-18, 2004), aspects of the visual signal other than fine phonetic detail may also contribute to the perception of speech. In two experiments, we show that a modulating circle providing information about the onset, offset, and acoustic amplitude envelope of the speech does not improve recognition of spoken sentences (Experiment 1) or words (Experiment 2). Further, contrary to our hypothesis, the modulating circle increased listening effort despite subjective reports that it made the word recognition task seem easier to complete (Experiment 2). These results suggest that audiovisual speech processing, even when the visual stimulus only conveys temporal information about the acoustic signal, may be a cognitively demanding process.

Speech Sanitizer: Speech Content Desensitization and Voice Anonymization

Voice input users’ speech recordings are being collected by service providers and shared with third parties, who may abuse users’ voiceprints, identify them by voice, and learn their sensitive speech content. In this work, we design <italic xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">Speech Sanitizer</i> to perturb users’ speech recordings so that the sanitized speech can be safely shared with third parties. First, we desensitize speech content by identifying sensitive words, localizing them in the audio using DTW-based keyword spotting, and substituting them with safe words. Both common and personalized sensitive words are identified and replaced. Then, we anonymize users’ voiceprints with a carefully designed voice conversion mechanism that is resistant to de-anonymization attacks. Meanwhile, we try to preserve the utility of the sanitized speech, measured by the accuracy of speech recognition performed on it. We implement Speech Sanitizer and present extensive experimental results that validate the effectiveness and efficiency of our algorithms. It is demonstrated that we are able to reduce the chance of a user's voice being identified from 50 people by 83.7 percent while keeping the drop of speech recognition accuracy within 19.1 percent. We can also easily relax the privacy level to improve speech recognition accuracy.

Ensemble of jointly trained deep neural network-based acoustic models for reverberant speech recognition

Distant speech recognition is a challenge, particularly due to the corruption of speech signals by reverberation caused by large distances between the speaker and microphone. In order to cope with a wide range of reverberations in real-world situations, we present novel approaches for acoustic modeling including an ensemble of deep neural networks (DNNs) and an ensemble of jointly trained DNNs. First, multiple DNNs are firstly designed, each of which copes with a different reverberation time (RT60) in a setup step. Also, each model in the ensemble of DNN acoustic models is further jointly trained, including both feature mapping and acoustic modeling, where feature mapping is designed for dereverberation as a front-end. In a testing phase, ensemble of DNNs are combined by weighted averaging of the prediction probabilities of the RT60 estimates, which is obtained by the convolutional neural network (CNN). In other words, the posterior probability outputs from DNNs are combined using the CNN-based weights as a weighted average. Extensive experiments demonstrate that the proposed approach leads to substantial improvements in speech recognition accuracy over the conventional DNN baseline systems under diverse reverberant conditions. In this paper, experiments are performed on Aurora-4 and CHiME-4 databases.

Talking points: A modulating circle reduces listening effort without improving speech recognition.

Speech recognition is improved when the acoustic input is accompanied by visual cues provided by a talking face (Erber in Journal of Speech and Hearing Research, 12(2), 423-425 1969; Sumby & Pollack in The Journal of the Acoustical Society of America, 26(2), 212-215, 1954). One way that the visual signal facilitates speech recognition is by providing the listener with information about fine phonetic detail that complements information from the auditory signal. However, given that degraded face stimuli can still improve speech recognition accuracy (Munhall et al. in Perception & Psychophysics, 66(4), 574-583, 2004), and static or moving shapes can improve speech detection accuracy (Bernstein et al. in Speech Communication, 44(1/4), 5-18, 2004), aspects of the visual signal other than fine phonetic detail may also contribute to the perception of speech. In two experiments, we show that a modulating circle providing information about the onset, offset, and acoustic amplitude envelope of the speech does not improve recognition of spoken sentences (Experiment 1) or words (Experiment 2), but does reduce the effort necessary to recognize speech. These results suggest that although fine phonetic detail may be required for the visual signal to benefit speech recognition, low-level features of the visual signal may function to reduce the cognitive effort associated with processing speech.

Synchrony-Based Feature Extraction for Robust Automatic Speech Recognition

This letter discusses the application of models of temporal patterns of auditory-nerve firings to enhance robustness of automatic speech recognition systems. Most conventional feature extraction schemes (such as mel-frequency cepstral coefficients and perceptual linear processing coefficients) are based on short-time energy in each frequency band, and the temporal patterns of auditory-nerve activity are discarded. We compare the impact on speech recognition accuracy of several types of feature extraction schemes based on the putative synchrony of auditory-nerve activity, including feature extraction based on a modified version of the generalized synchrony detector proposed by Seneff, and a modified version of the averaged localized synchrony response proposed by Young and Sachs. It was found that the use of features based on auditory-nerve synchrony can indeed improve speech recognition accuracy in the presence of additive noise based on experiments using multiple standard speech databases. Recognition accuracy obtained using the synchrony-based features is further increased if some form of noise removal is applied to the signal before the synchrony measure is estimated. Signal processing for noise removal based on the noise suppression that is a part of PNCC feature extraction is more effective toward this end than conventional spectral subtraction.

Evaluation of the Vocal Tract Length Normalization Based Classifiers for Speaker Verification

This paper proposes and evaluates classifiers based on Vocal Tract Length Normalization (VTLN) in a text-dependent speaker verification (SV) task with short testing utterances. This type of tasks is important in commercial applications and is not easily addressed with methods designed for long utterances such as JFA and i-Vectors. In contrast, VTLN is a speaker compensation scheme that can lead to significant improvements in speech recognition accuracy with just a few seconds of speech samples. A novel scheme to generate new classifiers is employed by incorporating the observation vector sequence compensated with VTLN. The modified sequence of feature vectors and the corresponding warping factors are used to generate classifiers whose scores are combined by a Support Vector Machine (SVM) based SV system. The proposed scheme can provide an average reduction in EER equal to 14% when compared with the baseline system based on the likelihood of observation vectors.

Feature mapping using far-field microphones for distant speech recognition

Acoustic modeling based on deep architectures has recently gained remarkable success, with substantial improvement of speech recognition accuracy in several automatic speech recognition (ASR) tasks. For distant speech recognition, the multi-channel deep neural network based approaches rely on the powerful modeling capability of deep neural network (DNN) to learn suitable representation of distant speech directly from its multi-channel source. In this model-based combination of multiple microphones, features from each channel are concatenated and used together as an input to DNN. This allows integrating the multi-channel audio for acoustic modeling without any pre-processing steps. Despite powerful modeling capabilities of DNN, an environmental mismatch due to noise and reverberation may result in severe performance degradation when features are simply fed to a DNN without a feature enhancement step. In this paper, we introduce the nonlinear bottleneck feature mapping approach using DNN, to transform the noisy and reverberant features to its clean version. The bottleneck features derived from the DNN are used as a teacher signal because they contain relevant information to phoneme classification, and the mapping is performed with the objective of suppressing noise and reverberation. The individual and combined impacts of beamforming and speaker adaptation techniques along with the feature mapping are examined for distant large vocabulary speech recognition, using a single and multiple far-field microphones. As an alternative to beamforming, experiments with concatenating multiple channel features are conducted. The experimental results on the AMI meeting corpus show that the feature mapping, used in combination with beamforming and speaker adaptation yields a distant speech recognition performance below 50% word error rate (WER), using DNN for acoustic modeling.

An Unsupervised Adaptation Approach to Leveraging Feedback Loop Data by Using i-Vector for Data Clustering and Selection

We present a study of using unsupervised adaptation approaches to improve speech recognition accuracy of a deployed speech service by leveraging large-scale untranscribed speech data collected from a feedback loop (FBL). For a regular user with lots of adaptation utterances, conventional CMLLR-based adaptation can be used for personalization directly. For a casual user with a few adaptation utterances, we propose to use CMLLR-based adaptation by augmenting his / her adaptation utterances with utterances acoustically close to the user, which are selected from the FBL data by an i-vector based approach. For a new user, we propose to perform a CMLLR-based recognition of an unknown utterance by selecting a set of CMLLR transforms from the most similar cluster, which are pre-trained by using the utterances from the corresponding cluster generated by an i-vector based utterance clustering method from the FBL data. The effectiveness of the above approaches are confirmed by our experiments on a short message dictation task on smart phones.

Intra- and Inter-frame Features for Automatic Speech Recognition

In this paper, alternative dynamic features for speech recognition are proposed. The goal of this work is to improve speech recognition accuracy by deriving the representation of distinctive dynamic characteristics from a speech spectrum. This work was inspired by two temporal dynamics of a speech signal. One is the highly non-stationary nature of speech, and the other is the inter-frame change of a speech spectrum. We adopt the use of a sub-frame spectrum analyzer to capture very rapid spectral changes within a speech analysis frame. In addition, we attempt to measure spectral fluctuations of a more complex manner as opposed to traditional dynamic features such as delta or double-delta. To evaluate the proposed features, speech recognition tests over smartphone environments were conducted. The experimental results show that the feature streams simply combined with the proposed features are effective for an improvement in the recognition accuracy of a hidden Markov model-based speech recognizer.

시변 잡음에 강인한 음성 인식을 위한 PCA 기반의 Variational 모델 생성 기법

본 논문에서는 시간에 따라 변하는 잡음 환경에 강인한 음성 인식을 위해 효과적인 특징 보상 기법을 제안한다. 제안하는 기법에서는 기존의 Variational 모델 생성 기법의 모델 정확도를 향상시키고자 PCA를 도입한다. 제안된 기법은 다중 모델을 사용하는 PCGMM 기반의 특징 보상에 적용된다. 실험 결과는 제안한 PCA 기반의 Variational 모델 생성 기법이 배경 음악 환경의 다양한 SNR 조건에서 기존의 전처리 기법에 비하여 음성 인식 성능을 향상 시키는데 우수함을 입증한다. 제안한 모델 생성 기법이 기존의 Variational 모델 생성 방법에 비해 배경 음악 환경에서 평균 12.14%의 상대적 인식 성능 향상률을 나타낸다. This paper proposes an effective feature compensation method to improve speech recognition performance in time-varying background noise condition. The proposed method employs principal component analysis to improve the variational model composition method. The proposed method is employed to generate multiple environmental models for the PCGMM-based feature compensation scheme. Experimental results prove that the proposed scheme is more effective at improving speech recognition accuracy in various SNR conditions of background music, compared to the conventional front-end methods. It shows 12.14% of average relative improvement in WER compared to the previous variational model composition method.

Intra-frame cepstral sub-band weighting and histogram equalization for noise-robust speech recognition

In this paper, we propose a novel noise-robustness method known as weighted sub-band histogram equalization (WS-HEQ) to improve speech recognition accuracy in noise-corrupted environments. Considering the observations that high- and low-pass portions of the intra-frame cepstral features possess unequal importance for noise-corrupted speech recognition, WS-HEQ is intended to reduce the high-pass components of the cepstral features. Furthermore, we provide four types of WS-HEQ, which partially refers to the structure of spatial histogram equalization (S-HEQ). In the experiments conducted on the Aurora-2 noisy-digit database, the presented WS-HEQ yields significant recognition improvements relative to the Mel-scaled filter-bank cepstral coefficient (MFCC) baseline and to cepstral histogram normalization (CHN) in various noise-corrupted situations and exhibits a behavior superior to that of S-HEQ in most cases.

Multiple cameras for audio-visual speech recognition in an automotive environment

Audio-visual speech recognition, or the combination of visual lip-reading with traditional acoustic speech recognition, has been previously shown to provide a considerable improvement over acoustic-only approaches in noisy environments, such as that present in an automotive cabin. The research presented in this paper will extend upon the established audio-visual speech recognition literature to show that further improvements in speech recognition accuracy can be obtained when multiple frontal or near-frontal views of a speaker's face are available. A series of visual speech recognition experiments using a four-stream visual synchronous hidden Markov model (SHMM) are conducted on the four-camera AVICAR automotive audio-visual speech database. We study the relative contribution between the side and central orientated cameras in improving visual speech recognition accuracy. Finally combination of the four visual streams with a single audio stream in a five-stream SHMM demonstrates a relative improvement of over 56% in word recognition accuracy when compared to the acoustic-only approach in the noisiest conditions of the AVICAR database.

Mask classification for missing-feature reconstruction for robust speech recognition in unknown background noise

“Missing-feature” techniques to improve speech recognition accuracy are based on the blind determination of which cells in a spectrogram-like display of speech are corrupted by the effects of noise or other types of disturbance (and hence are “missing”). In this paper we present three new approaches that improve the speech recognition accuracy obtained using missing-feature techniques. It had been found in previous studies (e.g. Seltzer et al., 2004) that Bayesian approaches to missing-feature classification are effective in ameliorating the effects of various types of additive noise. While Seltzer et al. primarily used white noise for training their Bayesian classifier, we have found that this is not the best type of training signal when noise with greater spectral and/or temporal variation is encountered in the testing environment. The first innovation introduced in this paper, referred to as frequency-dependent classification, involves independent classification in each of the various frequency bands in which the incoming speech is analyzed based on parallel sets of frequency-dependent features. The second innovation, referred to as colored-noise generation using multi-band partitioning, involves the use of masking noises with artificially-introduced spectral and temporal variation in training the Bayesian classifier used to determine which spectro-temporal components of incoming speech are corrupted by noise in unknown testing environments. The third innovation consists of an adaptive method to estimate the a priori values of the mask classifier that determines whether a particular time-frequency segment of the test data should be considered to be reliable or not. It is shown that these innovations provide improved speech recognition accuracy on a small vocabulary test when missing-feature restoration is applied to incoming speech that is corrupted by additive noise of an unknown nature, especially at lower signal-to-noise ratios.

Using Laplacian eigenmaps latent variable model and manifold learning to improve speech recognition accuracy

This paper demonstrates the application of the Laplacian eigenmaps latent variable model (LELVM) to the task of speech recognition. LELVM is a new dimension reduction method that combines the benefits of latent variable models—a multimodal probability density for latent and observed variables, and globally differentiable nonlinear mappings for the tasks of reconstruction and dimensionality reduction—with spectral manifold learning methods—no local optimum, ability to unfold nonlinear manifolds, and excellent practical scaling to latent spaces of high dimensions. LELVM is achieved by defining an out-of-sample mapping for Laplacian eigenmaps using a semi-supervised learning procedure. LELVM is simple, non-parametric and computationally inexpensive. In this research, LELVM is used to project MFCC features to a new subspace which leads to more discrimination among different phonetic categories. To evaluate the performance of the proposed feature modification system, a HMM-based speech recognition system and TIMIT speech database are employed. The experiments represent about 5% of the accuracy improvement in an isolated phoneme recognition task. The experiments imply the superiority of the proposed method to the usual PCA methods. Moreover, the proposed method keeps its benefits in noisy environments and does not degrade in such conditions.